--- /dev/null
+*~
+/build
+/_build
+/b/
--- /dev/null
+include: "https://gitlab.freedesktop.org/gstreamer/gst-ci/raw/master/gitlab/ci_template.yml"
--- /dev/null
+Wim Taymans <wim.taymans@collabora.co.uk>
--- /dev/null
+ GNU LESSER GENERAL PUBLIC LICENSE
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--- /dev/null
+ GNU LESSER GENERAL PUBLIC LICENSE
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+That's all there is to it!
+
--- /dev/null
+=== release 1.19.2 ===
+
+2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.19.2
+
+2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ Protection against early RTCP packets.
+ When receiving RTCP packets early the funnel is not ready yet and
+ GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
+ This causes the thread that handle RTCP packets to go to pause mode.
+ Since this thread is in pause mode there will be no further callbacks to
+ handle keep-alive for incoming RTCP packets. This will make the session
+ time out if the client is not using another keep-alive mechanism.
+ Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
+
+2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
+
+ * COPYING:
+ * COPYING.LIB:
+ Update COPYING.LIB, COPYING files
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
+
+2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * meson.build:
+ Back to development
+
+=== release 1.19.1 ===
+
+2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * docs/gst_plugins_cache.json:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.19.1
+
+2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: use new gst_buffer_new_memdup()
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
+
+2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media: fix leak when adding converter
+ Free the previous caps before reusing the variable for the converter caps.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
+
+2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix leak adding headers
+ gst_rtsp_message_add_header() makes a copy of the header, instead
+ of taking ownership.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
+
+2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ Use gst_element_request_pad_simple...
+ Instead of the deprecated gst_element_get_request_pad.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
+
+2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Ensure the bus watch is removed during unprepare
+ It's possible for the destruction of the source to be delayed.
+ Instead of relying on the dispose() to remove the bus watch, do
+ it ourselves.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
+
+2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
+
+ * docs/README:
+ docs: minor spelling correction in README
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
+
+2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
+
+ * examples/test-replay-server.c:
+ test-replay-server: minor spelling corrections
+ Bumped on these while investigating the example code.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
+
+2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * tests/check/gst/stream.c:
+ tests: Don't fail tests if IPv6 not available.
+ On computers with IPv6 disabled it shouldn't result in a test failure.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
+
+2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Add one more case to seek avoidance
+ This is an extension to the previous commit. There can also be cases where the
+ start position is not specified, in those cases we should also avoid doing
+ seeking unless it's forced.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
+
+2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Improve skipping trickmode seek.
+ We can also skip the seek if the end range is already
+ correct.
+ Avoids initial seek on play start if playing full stream.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
+
+2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Don't run signal class handlers during the CLEANUP stage
+ It's sufficient to run them during the FIRST stage instead of in both.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
+
+2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspclientsink.c:
+ tests: rtspclientsink: fix some leaks
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
+
+2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
+
+2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspclientsink.c:
+ rtspclientsink: add unit test for potential shutdown deadlock
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
+
+2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: fix deadlock on shutdown before preroll
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
+
+2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: avoid deadlock in send_func
+ Currently the send_func() runs in a thread of its own which is started
+ the first time we enter handle_new_sample(). It runs in an outer loop
+ until priv->continue_sending is FALSE, which happens when a TEARDOWN
+ request is received. We use a local variable, cont, which is initialized
+ to TRUE, meaning that we will always enter the outer loop, and at the
+ end of the outer loop we assign it the value of priv->continue_sending.
+ Within the outer loop there is an inner loop, where we wait to be
+ signaled when there is more data to send. The inner loop is exited when
+ priv->send_cookie has changed value, which it does when more data is
+ available or when a TEARDOWN has been received.
+ But if we get a TEARDOWN before send_func() is entered we will get stuck
+ in the inner loop because no one will increase priv->session_cookie
+ anymore.
+ By not entering the outer loop in send_func() if priv->continue_sending
+ is FALSE we make sure that we do not get stuck in send_func()'s inner
+ loop should we receive a TEARDOWN before the send thread has started.
+ Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
+
+2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: cleanup transports during TEARDOWN
+ When tunneling RTP over RTSP the stream transports are stored in a hash
+ table in the GstRTSPClientPrivate struct. They are used for, among other
+ things, mapping channel id to stream transports when receiving data from
+ the client. The stream tranports are created and added to the hash table
+ in handle_setup_request(), but unfortuately they are not removed in
+ handle_teardown_request(). This means that if the client sends data on
+ the RTSP connection after it has sent the TEARDOWN, which is often the
+ case when audio backchannel is enabled, handle_data() will still be able
+ to map the channel to a session transport and pass the data along to it.
+ Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
+ because the stream is no longer joined to a bin.
+ We avoid this by removing the stream transports from the hash table when
+ we handle the TEARDOWN request.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
+
+2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
+
+2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
+
+ * tests/check/gst/client.c:
+ Add test cases for mountpoint of '/'
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
+
+2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ Make a mount point of "/" work correctly.
+ As far as I can tell, this is neither explicitly allowed nor
+ forbidden by RFC 7826.
+ Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
+ use in the wild (presumably with non-GStreamer servers).
+ GStreamer's prior behavior was confusing, in that
+ gst_rtsp_mount_points_add_factory() would appear to accept a mount
+ path of "" or "/", but later connection attempts would fail with a
+ "media not found" error.
+ This commit makes a mount path of "/" work for either form of URL,
+ while an empty mount path ("") is rejected and logs a warning.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
+
+2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
+
+2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only count senders when counting blocked streams
+ Only sender streams sends the GstRTSPStreamBlocking message, so only
+ these should be counted before setting media status to prepared.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
+
+2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink add proper support for uri queries
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
+
+2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Only unref client watch context on finalize, to avoid deadlock
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
+
+2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: collect a clock_rate when blocking
+ This lets us provide a clock_rate in a fashion similar to the
+ other code paths in get_rtpinfo()
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
+
+2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Use guint64 for setting the size-time property on rtpstorage
+ Otherwise this will cause memory corruption as the property expects a 64
+ bit integer.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
+
+2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
+ To prevent cases with prerolling when the inactive stream prerolls first
+ and the server proceeds without waiting for the active stream, we will
+ ignore GstRTSPStreamBlocking messages from incomplete streams. When
+ there are no complete streams (during DESCRIBE), we will listen to all
+ streams.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
+
+2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
+
+ * tests/check/gst/media.c:
+ * tests/check/meson.build:
+ * tests/files/test.avi:
+ media test: Add test for seeking one active stream with a demuxer
+ Add another seek_one_active_stream test but with a demuxer. The demuxer
+ will flush both streams in opposed to the existing test which only
+ flushes the active stream. This will help exposing problems with the
+ prerolling process after a flushing seek.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
+
+2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/meson.build:
+ * meson.build:
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp-server.pc.in:
+ * pkgconfig/meson.build:
+ Meson: Use pkg-config generator
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
+
+2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * meson.build:
+ meson: update glib minimum version to 2.56
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
+
+2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * examples/test-launch.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server-internal.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ rtsp-media-factory: expose API to disable RTCP
+ This is supported by the RFC, and can be useful on systems where
+ allocating two consecutive ports is problematic, and RTCP is not
+ necessary.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
+
+2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * hooks/pre-commit.hook:
+ * meson.build:
+ git: use our standard pre commit hook
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
+
+2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: make use of blocked_running_time in query_position
+ When blocking, the sink element will not have received a buffer
+ yet and the position query will fail. Instead, we make use of
+ the running time of the buffer we blocked on.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
+
+2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: collect rtp info when blocking
+ We don't unblock the stream anymore before replying to the
+ play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
+ so the sinks don't have a last-sample after potentially flush
+ seeking. seek_trickmode waits for preroll however, which means
+ the stream will block and wait for a first buffer. Subsequent
+ calls to get_rtpinfo() can thus make use of the information.
+ See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
+
+2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
+
+ * examples/meson.build:
+ * examples/test-replay-server.c:
+ * examples/test-replay-server.h:
+ examples: Add an example for loop playback
+ This demo example shows a way of file loop playback of a given source.
+ Note that client seek request is not properly implemented yet.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
+
+2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Plug memory leak
+ The get-storage signal of rtpbin increases the ref count of the storage.
+ So we have to unref it after usage.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
+
+2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Get rates only on sender streams
+ When play a media with both sender and receiver stream, like ONVIF
+ back channel audio in, gst_rtsp_media_get_rates call
+ gst_rtsp_stream_get_rates for each stream to set the rates. But
+ gst_rtsp_stream_get_rates return false for the receiver steam, which
+ lead a g_assert crash.
+ Instead to get rates on all streams, now just get rates on sender
+ streams.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
+
+2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server-internal.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-media: set a 0 storage size for TCP receivers
+ ulpfec correction is obviously useless when receiving a stream
+ over TCP, and in TCP modes the rtp storage receives non
+ timestamped buffers, causing it to queue buffers indefinitely,
+ until the queue grows so large that sanity checks kick in and
+ warnings start to get emitted.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
+
+2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: preroll on gap events
+ This allows negotiating a SDP with all streams present, but only
+ start sending packets at some later point in time
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
+
+2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: do not unblock on unsuspend
+ rtsp_media_unsuspend() is called from handle_play_request()
+ before sending the play response. Unblocking the streams here
+ was causing data to be sent out before the client was ready
+ to handle it, with obvious side effects such as initial packets
+ getting discarded, causing decoding errors.
+ Instead we can simply let the media streams be unblocked when
+ the state of the media is set to PLAYING, which occurs after
+ sending the play response.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
+
+2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitlab-ci.yml:
+ ci: include template from gst-ci master branch again
+
+2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * meson.build:
+ Back to development
+
+=== release 1.18.0 ===
+
+2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitlab-ci.yml:
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * docs/gst_plugins_cache.json:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.18.0
+
+=== release 1.17.90 ===
+
+2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * docs/gst_plugins_cache.json:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.17.90
+
+2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ rtsp-thread-pool.c: fix clang 10 warning
+ clang 10 is complaining about incompatible types due to the
+ glib typesystem.
+ ```
+ ../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]
+ ```
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
+
+2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ rtsp-thread-pool.c: fix clang 10 warning
+ clang 10 is complaining about incompatible types due to the
+ glib typesystem.
+ ```
+ ../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]
+ ```
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
+
+2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Fix resource leak in mikey messsage
+ Fixed a resource leak for mikey message while adding crypto session
+ failed.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
+
+2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ * scripts/extract-release-date-from-doap-file.py:
+ meson: set release date from .doap file for releases
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
+
+2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: explicitly set caps on udpsrc elements
+ This causes them to send caps events before data flow, which is
+ usually a pretty correct thing to do!
+ Not doing so manifested in a bug where ssrcdemux wouldn't forward
+ the caps it had received with an extra ssrc field, as it hadn't
+ received any caps event.
+ Fixes #85
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
+
+2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * meson.build:
+ Back to development
+
+=== release 1.17.2 ===
+
+2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * docs/gst_plugins_cache.json:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.17.2
+
+2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
+
+ * docs/gst_plugins_cache.json:
+ doc: Stop documenting properties from parents
+
+2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: Fix version in the plugins cache
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
+
+2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Don't call gst_ghost_pad_construct() anymore
+ It's deprecated, unneeded and doesn't do anything anymore.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
+
+2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ Back to development
+
+=== release 1.17.1 ===
+
+2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * docs/gst_plugins_cache.json:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.17.1
+
+2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Add/configure transports when completing the pipeline
+ Otherwise the transports are not set up yet during the PLAY request
+ handling when unsuspending (and thus unblocking) the media.
+ In case of live pipelines this then causes the first few packets to go
+ to the sinks before they know what to do with them, and they simply
+ discard them which is rather suboptimal in case of keyframes.
+ For non-live pipelines this is not a problem because the sink will still
+ be PAUSED and as such not send out the data yet but wait until it goes
+ to PLAYING, which is late enough.
+ Adding the transports multiple times is not a problem: if the transport
+ is already added it won't be added another time and TRUE will be
+ returned.
+ This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
+ before 1.14.0.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
+
+2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix misleading comment
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
+
+2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
+ The pad probes are not needed anymore at this point and later when
+ reaching buffering 100% only the state is changed, no unblocking
+ happens.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
+
+2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Remove duplicated media_unblock() function
+ It does literally the same as media_streams_set_blocked(FALSE).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
+
+2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
+
+ * examples/test-onvif-server.c:
+ test-onvif-server: cast ntp-offset property value to 64 bit
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
+
+2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: Update plugins cache
+
+2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * examples/test-onvif-server.c:
+ * examples/test-onvif-server.h:
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ onvif-media-factory: define autoptr cleanup function
+ And have the factory in the onvif-server example inherit from
+ GstRTSPOnvifMediaFactory.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
+
+2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: Update plugins cache
+
+2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: enforce I420 format
+ Test was not enforcing a video format on videotestsrc. I420 was picked as it
+ was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
+ true (gst-plugins-base!689).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
+
+2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ plugins: uddate gst_type_mark_as_plugin_api() calls
+
+2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
+
+ * docs/meson.build:
+ doc: Require hotdoc >= 0.11.0
+
+2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: Update gst_plugins_cache.json
+
+2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
+
+2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/meson.build:
+ meson: gir: remove bogus sources_top_dir kwarg
+ Doesn't actually exist. Was fixed differently in Meson
+ so that the user doesn't have to specify it.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
+
+2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/meson.build:
+ tests: put registry into tests/check not the gst/ subdir
+ Underscorify the test name before setting GST_REGISTRY,
+ so the registry actually ends up in the current build dir
+ and not some subdir.
+ For consistency with the other modules, but should also
+ avoid problems on windows.
+ Also fix indentation of environment block.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
+
+2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/meson.build:
+ tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
+ If core is built as a subproject (e.g. as in gst-build), make sure to use
+ the gst-plugin-scanner from the built subproject. Without this, gstreamer
+ might accidentally use the gst-plugin-scanner from the install prefix if
+ that exists, which in turn might drag in gst library versions we didn't
+ mean to drag in. Those gst library versions might then be older than
+ what our current build needs, and might cause our newly-built plugins
+ to get blacklisted in the test registry because they rely on a symbol
+ that the wrongly-pulled in gst lib doesn't have.
+ This should fix running of unit tests in gst-build when invoking
+ meson test or ninja test from outside the devenv for the case where
+ there is an older or different-version gst-plugin-scanner installed
+ in the install prefix.
+ In case no gst-plugin-scanner is installed in the install prefix, this
+ will fix "GStreamer-WARNING: External plugin loader failed. This most
+ likely means that the plugin loader helper binary was not found or
+ could not be run. You might need to set the GST_PLUGIN_SCANNER
+ environment variable if your setup is unusual." warnings when running
+ the unit tests.
+ In the case where we find GStreamer core via pkg-config we use
+ a newly-added pkg-config var "pluginscannerdir" to get the right
+ directory. This has the benefit of working transparently for both
+ installed and uninstalled pkg-config files/setups.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
+
+2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/meson.build:
+ tests: gst-plugins-base and -bad plugins are required for the unit tests
+ Make hard requirement until we have more fine-grained control
+ in the unit tests. Of course the presence of the .pc file doesn't
+ imply that the plugins we need are actually there, but it's at
+ least a step in the right direction.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
+
+2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/meson.build:
+ tests: pick up rtsp-server plugins from build directory only
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
+
+2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: wait for all GstRTSPStreamBlocking messages
+ Make sure rtsp-media have received a GstRTSPStreamBlocking message from
+ each active stream when checking if all streams are blocked.
+ Without this change there will be a race condition when using two or
+ more streams and rtsp-media receives a GstRTSPStreamBlocking message
+ from one of the streams. This is because rtsp-media then checks if all
+ streams are blocked by calling gst_rtsp_stream_is_blocking() for each
+ stream. This function call returns TRUE if the stream has sent a
+ GstRTSPStreamBlocking message, however, rtsp-media may have yet to
+ receive this message. This would then result in that rtsp-media
+ erroneously thinks it is blocking all streams which could result in
+ rtsp-media changing state, from PREPARING to PREPARED. In the case of a
+ preroll, this could result in that rtsp-media thinks that the pipeline
+ is prerolled even though that might not be the case.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
+
+2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: update expected_async_done during suspend
+ Set expected_async_done to FALSE in default_suspend() if a state change
+ occurs and the return value from set_target_state() is something other
+ than GST_STATE_CHANGE_ASYNC.
+ Without this change there is a risk that expected_async_done will be
+ TRUE even though no asynchronous state change is taking place. This
+ could happen if the pipeline is set to PAUSED using
+ media_set_pipeline_state_locked(), an asynchronous state change starts
+ and then the media is suspended (which could result in a state change,
+ aborting the asynchronous state change). If the media is suspended
+ before the asynchronous state change ends then expected_async_done will
+ be TRUE but no asynchronous state change is taking place.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
+
+2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
+ There was a race condition where client was being finalized and
+ concurrently in some other thread the rtsp ctrl timout was relying on
+ client data that was being freed.
+ When rtsp ctrl timeout is setup, a WeakRef on Client is set.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
+
+2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media-factory: complete DSCP QoS setting support
+ add dscp_qos setting support at factory and media level to setup IP DSCP
+ field of bounded UDP sinks.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
+
+2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix some race conditions around timeout source removal
+ We always need to take the lock while accessing it as otherwise another
+ thread might've removed it in the meantime. Also when destroying and
+ creating a new one, ensure that the mutex is not shortly unlocked in
+ between as during that time another one might potentially be created
+ already.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
+
+2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
+ And the same for gst_rtsp_stream_get_rates().
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
+
+2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-onvif-server.c:
+ examples: test-onvif-server: fix compiler warnings on raspbian
+ Fix printf format for 64-bit variables.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
+
+2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
+ The old API is preserved now and new API was added that provides the
+ additional parameter to the callback.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
+
+2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Store the timeout source by pointer instead of id
+ That way we don't have to retrieve it again from the main context when
+ destroying it but can directly do so.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
+
+2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Clean up watch/watch context and related state consistently
+ And assert that it was cleaned up properly before the client is
+ finalized. If something is still around when the client is shut down
+ then something went very wrong before.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
+
+2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-client: Combine the pre-session and post-session timeout
+ They previously used the same state but different mechanisms and
+ functions, which was difficult to follow, error prone and simply
+ confusing.
+ Also adjust the test for the post-session timeout a bit to be less racy
+ now that the timing has slightly changed.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
+
+2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Don't ever close the client connection directly when a session is torn down
+ There might be other sessions that are running over the same RTSP
+ connection and we should not simply close the client directly if one of
+ them is torn down.
+ By default the connection will be closed once the client closes it or
+ the OS does. This behaviour can be adjusted with the
+ post-session-timeout property, which allows to close it automatically
+ from the server side after all sessions are gone and the given timeout
+ is reached.
+ This reverts the previous commit.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
+
+2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
+ Instead of closing it never at all. Previously there was only code that
+ closed the client asynchronously if sending the response happened
+ asynchrously at a later time.
+ Thanks to Christian M for debugging this issue.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
+
+2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
+ Otherwise no sink is found for multicast sreams and the less accurate
+ fallback is used to determine the current sequence number and timestamp.
+
+2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
+ When using the basic authentication scheme, we wouldn't validate that
+ the authorization field of the credentials is not NULL and pass it on
+ to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
+ dereference the NULL pointer and crash.
+ A specially crafted (read: invalid) RTSP header can cause this to
+ happen.
+ As a solution, check for the authorization to be not NULL before
+ continuing processing it and if it is simply fail authentication.
+ This fixes CVE-2020-6095 and TALOS-2020-1018.
+ Discovered by Peter Wang of Cisco ASIG.
+
+2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Use watch_context before unref
+ Move the usage of priv->watch_context to beginning of function
+ gst_rtsp_client_finalize. Instead of use it after
+ g_main_context_unref (priv->watch_context).
+
+2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fix deadlock on transport removal
+ We cannot take the RTSPStream lock while holding a transport backlog
+ lock, as remove_transport may be called externally, which will
+ take first the RTSPStream lock then the transport backlog lock.
+
+2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-server-internal.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: clear backlog when removing transport
+ This ensures we don't end up calling any of transports' callbacks
+ with a potentially unreffed user_data (in practice, a client that
+ may have been removed)
+
+2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: marshal calls to send_tcp_message to a single thread
+ In order to address the race condition pointed out at
+ https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
+ we get rid of the send thread pool, and instead spawn and manage
+ a single thread to pull samples from app sinks and add them to
+ the transport's backlogs.
+ Additionally, we now also always go through the backlogs in order
+ to simplify the logic.
+
+2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-server-internal.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: properly protect TCP backlog access
+ Fixes #97
+ We cannot hold stream->lock while pushing data, but need
+ to consistently check the state of the backlog both from
+ the send_tcp_message function and the on_message_sent function,
+ which may or may not be called from the same thread.
+ This commit introduces internal API to allow for potentially
+ recursive locking of transport streams, addressing a race
+ condition where the RTSP stream could push items out of order
+ when popping them from the backlog.
+
+2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
+ It's taken ownership of by the media, and returned with `transfer none`
+ from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
+ first then any bindings will wrongly take ownership of the pipeline once
+ it arrives in bindings code.
+
+2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
+
+ * examples/test-onvif-client.c:
+ Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
+
+2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix default latency
+
+2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: make closing more thread safe
+ + Take the watch lock prior to using priv->watch
+ + Flush both the watch and connection before closing / unreffing
+ gst_rtsp_connection_close() is not threadsafe on its own, this is
+ a workaround at the client level, where we control both the watch
+ and the connection
+
+2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
+
+ * gst/rtsp-server/rtsp-latency-bin.c:
+ rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
+ from glib
+ ```
+ Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
+ `your_type_get_instance_private()` function instead
+ ```
+
+2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-client: add property post-session-timeout
+ This is a TCP connection timeout for client connections, in seconds.
+ If a positive value is set for this property, the client connection
+ will be kept alive for this amount of seconds after the last session
+ timeout. For negative values of this property the connection timeout
+ handling is delegated to the system (just as it was before).
+ Fixes #83
+
+2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: check for NULL transports prior to ref'ing
+
+2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-server-internal.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fix checking of TCP backpressure
+ The internal index of our appsinks, while it can be used to
+ determine whether a message is RTP or RTCP, is not necessarily
+ the same as the interleaved channel. Let the stream-transport
+ determine the channel to check backpressure for, the same way
+ it determines the channel according to whether it is sending
+ RTP or RTCP.
+
+2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: Butcher the file to please gst-indent in the CI
+ This should be reverted once the CI has an updated gst-indent.
+
+2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ rtsp-session & client: Remove deprecated GTimeVal
+ GTimeVal won't work past 2038
+
+2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ rtsp-auth: fix default token leak
+
+2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ gstrtspclientsink: unref transports when closing bin
+ Fixes #91
+
+2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Force seek when flush flag is set
+ The commit "rtsp-client: define all seek accuracy flags from
+ setup_play_mode" changed the behaviour of when doing a seek.
+ Before that commit, having the flush flag set would result in a seek
+ (forced seek).
+ Even if no seek was needed. One reason to force seek is to flush old buffers
+ created in Describe requests.
+ Thus adding force seek also for flush flag will result in play request
+ with fresh buffers.
+
+2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Revitalize dead code
+ Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
+ CID: 1455379
+
+2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Don't try to use non-initialized values
+ Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
+ returns TRUE. Also avoid the whole clock signalling block if we're not
+ dealing with senders.
+ CID: 1439524
+ CID: 1439536
+ CID: 1439520
+
+2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/stream.c:
+ rtsp-stream: Removing invalid transports returns false
+ When removing transports an assertion was that the transports passed in
+ for removal are present in the list, however that can't be assumed.
+ As an example if a transport was removed from a thread running
+ send_tcp_message, the main thread can try to remove the same transport
+ again if it gets a handle_pause_request. This will not effect the
+ transport list but it will effect n_tcp_transports as it will be
+ decrement and then have the wrong value.
+
+2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
+
+ * tests/check/gst/client.c:
+ client test: add scale and speed negative tests
+ Negative tests for scale and speed should be done as well, verify that
+ the response code is "400 Bad request" when a bad request is done.
+
+2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ Don't pass default GLib marshallers for signals
+ By passing NULL to `g_signal_new` instead of a marshaller, GLib will
+ actually internally optimize the signal (if the marshaller is available
+ in GLib itself) by also setting the valist marshaller. This makes the
+ signal emission a bit more performant than the regular marshalling,
+ which still needs to box into `GValue` and call libffi in case of a
+ generic marshaller.
+ Note that for custom marshallers, one would use
+ `g_signal_set_va_marshaller()` with the valist marshaller instead.
+
+2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ GstRTSPMountPoints: Remove any existing factory before adding a new one
+ The documentation of gst_rtsp_mount_points_add_factory() says "Any
+ previous mount point will be freed" which was true when it was
+ implemented using a GHashTable. But in 2012 it got rewrote using a
+ GSequence and since then it could have 2 factories for the same path.
+ Which one gets used is random, depending on the sorting order of 2
+ identical items.
+
+2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server-internal.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: refactor TCP backpressure handling
+ The previous implementation stopped sending TCP messages to
+ all clients when a single one stopped consuming them, which
+ obviously created problems for shared media.
+ Instead, we now manage a backlog in stream-transport, and slow
+ clients are removed once this backlog exceeds a maximum duration,
+ currently hardcoded.
+ Fixes #80
+
+2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: build gir even when cross-compiling if introspection was enabled explicitly
+ This can be made to work in certain circumstances when
+ cross-compiling, so default to not building g-i stuff
+ when cross-compiling, but allow it if introspection was
+ enabled explicitly via -Dintrospection=enabled.
+ See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
+
+2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: clean up comment extra-timeout
+
+2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
+ Instead of hardcoding the URI, take the actual URI (and especially the correct port)
+ from the RTSP context.
+ Fixes #84
+
+2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-client: Lock shared media
+ For shared media we got race conditions. Concurrently rtsp clients might
+ suspend or unsuspend the shared media and thus change the state without
+ the clients expecting that.
+ By introducing a lock that can be taken by callers such as rtsp_client
+ one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
+ to handle the media sequentially thus allowing one client to finish its
+ rtsp call before another client calls on the same media.
+ https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
+ Fixes #86
+
+2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: add property extra-timeout
+ Extra time to add to the timeout, in seconds. This only
+ affects the time until a session is considered timed out
+ and is not signalled in the RTSP request responses.
+ Only the value of the timeout property is signalled in the
+ request responses.
+
+2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream : fix race condition in send_tcp_message
+ If one thread is inside the send_tcp_message function and are done
+ sending rtp or rtcp messages so the n_outstanding variable is zero
+ however have not exit the loop sending the messages. While sending its
+ messages, transports have been added or removed to the transport list,
+ so the cache should be updated. If now an additional thread comes to
+ the function send_tcp_message and trying to send rtp messages it will
+ first destroy the rtp cache that is still being iterated trough by the
+ first thread.
+ Fixes #81
+
+2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ * .gitmodules:
+ * Makefile.am:
+ * autogen.sh:
+ * common:
+ * configure.ac:
+ * docs/.gitignore:
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * gst/Makefile.am:
+ * gst/rtsp-server/.gitignore:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-sink/Makefile.am:
+ * pkgconfig/.gitignore:
+ * pkgconfig/Makefile.am:
+ * tests/.gitignore:
+ * tests/Makefile.am:
+ * tests/check/Makefile.am:
+ Remove autotools build
+ Replaced by Meson.
+ Maybe we can now use the meson pkgconfig module
+ for .pc files? (Does it support uninstalled now?)
+
+2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * tests/check/gst/client.c:
+ client: fix test mem leak in attach_rate_tweaking_probe
+
+2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * tests/check/gst/media.c:
+ media: remove memleak in test test_media_seek
+
+2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ rtspserver: Remove memleak in test test_double_play
+
+2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Use lock in gst_rtsp_media_is_receive_only
+
+2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-media: Unblock all streams
+ When unsuspending and going to PLAYING, unblock all streams instead of
+ only those that are linked (the linked streams are the ones for which
+ SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
+ pushing buffers on unlinked streams.
+ This change is because playback using single-threaded demuxers like
+ matroska-demux could be blocked if SETUP was not called for all media.
+ Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
+ gstflvdemux, qtdemux, and matroska-demux) will handle
+ GST_FLOW_NOT_LINKED automatically.
+ Fixes #39
+
+2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-media: Wait on async when needed.
+ Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
+ In the unit test the pause from adjust_play_mode will cause a preroll
+ and after that async-done will be produced.
+ Without this patch there are no one consuming this async-done and when
+ later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
+ wait for async-done. But then it wrongly find the async-done prodused by
+ adjus_play_mode and continue executing without waiting for the preroll
+ to finish.
+
+2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: RTP Info when completed_sender
+ Change condition that should be fulfilled regarding RTPInfo.
+ Replace !gst_rtsp_media_is_receive_only with
+ gst_rtsp_media_has_completed_sender. It is more correct to actually look
+ for a sender pipeline that is complete. Only then a RTPInfo should
+ exist.
+ gst_rtsp_media_is_receive_only gives different answears depending on
+ state of server.
+ If Describe is called wth URL+options for backchannel SDP will give only
+ audio and only backchannel a=sendonly
+ If Describe is called on URL+options that gives both audio and video
+ direction from server to client, pipelines are created. Thus
+ receive_only will return false, even though Setup only would setup
+ backchannel.
+ RTP-Info is only for outgoing streams. Thus one should look if outgoing
+ streams are complete.
+
+2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ rtsp-client: RTP Info exists conditionally in PLAY
+ If RTP Info is missing and it is not a receiver only, eg. audio
+ backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
+ In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
+ Since 1.14 there is audio backchannel support. Thus RTP-info is
+ conditional now. When audio backchannel only mode, there is no RTP-info.
+ Fixes #82
+
+2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * examples/test-onvif-client.c:
+ test-onvif-client: remove unused query
+
+2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: RTP Info must exist in PLAY response
+ If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
+ Fixes #76
+
+2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * examples/test-onvif-client.c:
+ test-onvif-client: perform accurate seeks
+ See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
+ Also, modify how we compute the position: position queries in
+ PAUSED mode fail to account for the newly-prerolled frame, leading
+ to frame skips when performing seeks in that state. Instead,
+ compute the current position from the last sample.
+
+2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * tests/check/gst/rtspserver.c:
+ Use complete streams for scale and speed.
+ Without this patch it's always stream0 that is used to get segment event
+ that is used to set scale and speed. This even if client not doing SETUP
+ for stream0. At least in suspend mode reset this not working since then
+ it's just random if send_rtp_sink have got any segment event. There are
+ no check if send_rtp_sink for stream0 got any data before media is
+ prerolled after PLAY request.
+
+2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
+
+ * examples/test-onvif-server.c:
+ * examples/test-onvif-server.h:
+ examples/onvif-server: fix werror build with clang
+ ../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]
+ self->incoming_segment->format, self->incoming_segment->flags,
+ ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
+ ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
+ G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
+ ^
+ /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
+ static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
+ ^
+ <scratch space>:77:1: note: expanded from here
+ REPLAY_IS_BIN
+ ^
+ ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
+ G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
+ ^
+ /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
+ static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
+ ^
+ <scratch space>:9:1: note: expanded from here
+ ONVIF_FACTORY
+ ^
+ ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
+ /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
+ static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
+ ^
+ <scratch space>:12:1: note: expanded from here
+ ONVIF_IS_FACTORY
+ ^
+
+2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
+
+ * docs/meson.build:
+ meson: Don't generate doc cache when no plugins are enabled
+ Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
+
+2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * examples/test-onvif-client.c:
+ test-onvif-client: stdin is not defined in MSVC
+
+2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: add missing Since tag
+
+2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * examples/test-onvif-client.c:
+ test-onvif-client: STDIN_FILENO is not portable
+ If not defined, define it to _fileno(stdin) on Windows, 0
+ everywhere else
+
+2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * examples/test-onvif-server.c:
+ test-onvif-server: downgrade logging
+
+2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * examples/meson.build:
+ * examples/test-onvif-client.c:
+ * examples/test-onvif-server.c:
+ examples: add ONVIF client / server example
+
+2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-client: define all seek accuracy flags from setup_play_mode
+ We then pass those to adjust_play_mode, which needs to operate
+ on the "final" seek flags, as previously the code in rtsp-media
+ was assuming that accuracy seek flags (accurate / key_unit) should
+ not be set if the flags passed to the seek method were already set.
+
+2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Try to get dynamic payloaders by name from their bin first
+ First try "pay", then "pay_%s" (where %s == pad name). And only then
+ fall back to the code that simply takes the first payloader that is
+ found.
+ The current code usually works (but is racy) because it will always take
+ the payloader that was last added (due to g_list_prepend() when adding
+ elements) in pad-added and that's usually the correct one. But if a new
+ payloader is added between pad-added and us trying to get it, we would
+ get the wrong payloader.
+
+2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * tests/check/gst/client.c:
+ client test: expect any port in transport
+ setup_multicast_client sets a 5000-5010 range for the client
+ ports, it is incorrect to expect the transport to always use
+ 5000-5001
+ Fixes #73
+
+2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * tests/check/gst/onvif.c:
+ onvif tests: use g_cond_wait() correctly
+ g_cond_wait() has to be called in a loop until required conditions
+ are met
+ Fixes #71
+
+2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Not wait on receiver streams when pre-rolling
+ Without this patch there are problem pre-rolling when using audio back
+ channel.
+ Without this patch a probe will be created for all streams including
+ the stream for audio backchannel. To pre-roll all this pads have to
+ receive data. Since the stream for audio backchannel is a receiver this
+ will never happen.
+ The solution is to never create any probes for streams that are for
+ incomming data and instead set them as blocking already from beginning.
+
+2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-media-factory.c:
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ onvif-media: fix "void function returning a value" compiler warning
+
+2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: make sure streams are blocked when sending seek
+ The recent ONVIF work exposed a race condition when dealing with
+ multiple streams: one of the sinks may preroll before other streams
+ have started flushing. This led to the pipeline posting async-done
+ prematurely, when some streams were actually still in the middle
+ of performing a flushing seek. The newly-added code looks up a
+ sticky segment event on the first stream in order to respond to
+ the PLAY request with accurate Scale and Speed headers. In the
+ failure condition, the first stream was flushing, and thus had
+ no sticky segment event, leading to the PLAY request failing,
+ and in turn the test.
+
+2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
+
+ * docs/README:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ Fix typos
+
+2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-onvif-client.c:
+ * gst/rtsp-server/rtsp-onvif-client.h:
+ * gst/rtsp-server/rtsp-onvif-media-factory.c:
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ * gst/rtsp-server/rtsp-onvif-server.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/media.c:
+ * tests/check/gst/onvif.c:
+ * tests/check/meson.build:
+ onvif: Implement and test the Streaming Specification
+ https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
+
+2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: add gst_rtsp_client_get_stream_transport()
+ This will be used in the onvif tests in order to validate the
+ data transmitted over TCP: for streaming to continue after a
+ data message has been provided to client->send_func, the client
+ is responsible for marking the message as sent on the relevant
+ stream transport.
+
+2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Scale implies TRICK_MODE
+
+2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: compare booleans, not pointers to them
+
+2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/media.c:
+ Reverse playback support
+ GStreamer plays segment from stop to start when doing reverse playback.
+ RTSP implies that media should be played from start of Range header to
+ its stop. Hence we swap start and stop times before passing them to
+ gst_element_seek.
+ Also make gst_rtsp_stream_query_stop always return value that can be
+ used as stop time of Range header.
+
+2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * tests/check/gst/client.c:
+ rtsp-client: add support for Scale and Speed header
+ Add support for the RTSP Scale and Speed headers by setting the rate in
+ the seek to (scale*speed). We then check the resulting segment for rate
+ and applied rate, and use them as values for the Speed and Scale headers
+ respectively.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754575
+
+2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: allow sub classes to adjust the seek
+ Adds a new virtual function, adjust_play_mode(), that allows
+ sub classes to adjust the seek done on the media. The sub class can
+ modify the values of the the seek flags and the rate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754575
+
+2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/media.c:
+ rtsp-media: allow specifying rate when seeking
+ Add new function gst_rtsp_media_seek_full_with_rate() which allows the
+ caller to specify the rate for the seek. Also added functions in
+ rtsp-stream and rtsp-media for retreiving current rate and applied rate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754575
+
+2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
+
+ * configure.ac:
+ * meson.build:
+ meson: Bump minimal GLib version to 2.44
+ This means we can use some newer features and get rid of some
+ boilerplate code using the G_DECLARE_* macros.
+ As discussed on IRC, 2.44 is old enough by now to start depending on it.
+
+2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/.gitignore:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * docs/libs/gst-rtsp-server.types:
+ docs: remove obsolete gtk-doc related files
+
+2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ doc: remove xml from comments
+
+2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
+
+ * docs/gst_plugins_cache.json:
+ * docs/meson.build:
+ docs: Stop building the doc cache by default
+ And update the cache
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
+
+2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: Update plugins documentation cache
+
+2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
+
+ * docs/meson.build:
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ doc: Fix some docstrings
+
+2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ * docs/Makefile.am:
+ * docs/gst_plugins_cache.json:
+ * docs/index.md:
+ * docs/meson.build:
+ * docs/plugin-index.md:
+ * docs/plugin-sitemap.txt:
+ * docs/sitemap.md:
+ * docs/sitemap.txt:
+ * docs/version.entities.in:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-sink/meson.build:
+ * meson.build:
+ * meson_options.txt:
+ docs: Port to hotdoc
+
+2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-server: Fix various Since markers
+
+2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Add various Since: 1.14 markers
+
+2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Add various missing Since: 1.16 markers
+
+2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Set async-handling=false for the internal bins
+ Without this we can easily run into a race condition with async state changes:
+ - the pipeline is doing an async state change
+ - we set the internal bins to PLAYING but that's ignored because an
+ async state change is currently pending
+ - the async state change finishes but does not change the state of the
+ internal bins because of locked_state==TRUE
+ - the internal bins stay in PAUSED forever
+
+2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Use write_messages() API to send buffer lists in one go
+ And to write messages with multiple memories also via writev().
+
+2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server-object.h:
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-client: Handle Content-Length limitation
+ Add functionality to limit the Content-Length.
+ API addition, Enhancement.
+ Define an appropriate request size limit and reject requests
+ exceeding the limit with response status 413 Request Entity Too Large
+ Related to !182
+
+2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * RELEASE:
+ * configure.ac:
+ * meson.build:
+ Back to development
+
+=== release 1.16.0 ===
+
+2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.16.0
+
+2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Notify the stream transport about each written message
+ Otherwise it will never try to send us the next one: it tries to keep
+ exactly one message in-flight all the time.
+ In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
+ in the client sink we always write data out synchronously.
+
+2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp_server: Free thread pool before clean transport cache
+ If not waiting for free thread pool before clean transport caches, there
+ can be a crash if a thread is executing in transport list loop in
+ function send_tcp_message.
+ Also add a check if priv->send_pool in on_message_sent to avoid that a
+ new thread is pushed during wait of free thread pool. This is possible
+ since when waiting for free thread pool mutex have to be unlocked.
+
+=== release 1.15.90 ===
+
+2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.15.90
+
+2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Add support for GCM (RFC 7714)
+ Follow-up to !198
+
+2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session pool: fix missing klass-> in klass->create_session
+
+2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ g-i: pass --quiet to g-ir-scanner
+ This suppresses the annoying 'g-ir-scanner: link: cc ..' output
+ that we get even if everything works just fine.
+ We still get g-ir-scanner warnings and compiler warnings if
+ we pass this option.
+
+2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ g-i: silence 'nested extern' compiler warnings when building scanner binary
+ We need a nested extern in our init section for the scanner binary
+ so we can call gst_init to make sure GStreamer types are initialised
+ (they are not all lazy init via get_type functions, but some are in
+ exported variables). There doesn't seem to be any other mechanism to
+ achieve this, so just remove that warning, it's not important at all.
+
+2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
+
+2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: Handle set state when preparing.
+ Handle the situation when a call to gst_rtsp_media_set_state is done
+ when media status is preparing.
+ Also add unit test for this scenario.
+ The unit test simulate on a media level when two clients share a (live)
+ media.
+ Both clients have done SETUP and got responses. Now client 1 is doing
+ play and client 2 is just closing the connection.
+ Then without patch there are a problem when
+ client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
+ And client2 is doing closing connection we can end up in a call
+ to gst_rtsp_media_set_state when
+ priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
+ shut down media is jumped over .
+ With this patch and this scenario we wait until
+ priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
+ execute after that and now we will execute the logic for
+ shut down media.
+
+2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * meson.build:
+ Back to development
+
+=== release 1.15.2 ===
+
+2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.15.2
+
+2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/client.c:
+ rtsp-media: Fix multicast use case with common media
+ Use case
+ client 1: SETUP
+ client 1: PLAY
+ client 2: SETUP
+ client 1: TEARDOWN
+ client 2: PLAY
+ client 2: TEARDOWN
+
+2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: remove recursive behavior
+ Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
+
+2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Only allow to set either a send_func or send_messages_func but not both
+ And route all messages through the send_func if no send_messages_func
+ was provided.
+ We otherwise break backwards compatibility.
+
+2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-client: Add support for sending buffer lists directly
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
+
+2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtsp-server: Add support for buffer lists
+ This adds new functions for passing buffer lists through the different
+ layers without breaking API/ABI, and enables the appsink to actually
+ provide buffer lists.
+ This should already reduce CPU usage and potentially context switches a
+ bit by passing a whole buffer list from the appsink instead of
+ individual buffers. As a next step it would be necessary to
+ a) Add support for a vector of data for the GstRTSPMessage body
+ b) Add support for sending multiple messages at once to the
+ GstRTSPWatch and let it be handled internally
+ c) Adding API to GOutputStream that works like writev()
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
+
+2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Fix crash in close handler
+ The close handler could trigger a crash because it invalidated the
+ watch_context while still leaving a source attached to it which would be
+ cleaned up at a later point.
+
+2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Use cached address when allocating sockets
+ If an address/port was previously decided upon (ex: multicast in the
+ SDP), then use that instead of re-creating another one
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
+
+2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix race codition in finish_unprepare
+ The previous fix for race condition around finish_unprepare where the
+ function could be called twice assumed that the status wouldn't change
+ during execution of the function. This assumption is incorrect as the
+ state may change, for example if an error message arrives from the
+ pipeline bus.
+ Instead a flag keeping track on whether the finish_unprepare function
+ is currently executing is introduced and checked.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
+
+=== release 1.15.1 ===
+
+2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.15.1
+
+2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ Add source elements to the pipeline before activation
+ In plug_src we changed the element state before adding it to
+ the owner container. This prevented the pipeline from intercepting
+ a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
+ to assign a custom task pool.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
+
+2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * common:
+ Automatic update of common submodule
+ From ed78bee to 59cb678
+
+2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
+
+ * examples/test-appsrc.c:
+ examples: test-appsrc: fix coding style error
+
+2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
+
+ * examples/test-appsrc.c:
+ examples: test-appsrc: fix buffer leak
+
+2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Update priv->blocked when linked streams are unblocked.
+ Media is considered to be blocked when all streams that belong to
+ that media are blocked.
+ This patch solves the problem of inconsistent updates of
+ priv->blocked that are not synchronized with the media state.
+
+2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't block streams before seeking
+ Before the seek operation is performed on media, it's required that
+ its pipeline is prepared <=> the pipeline is in the PAUSED state.
+ At this stage, all transport parts (transport sinks) have been successfully
+ added to the pipeline and there is no need for blocking the streams.
+
+2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: rtspserver: Add shared media test case for TCP
+
+2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Use seqnum-offset for rtpinfo
+ The sequence number in the rtpinfo is supposed to be the first RTP
+ sequence number. The "seqnum" property on a payloader is supposed to be
+ the number from the last processed RTP packet. The sequence number for
+ payloaders that inherit gstrtpbasepayload will not be correct in case of
+ buffer lists. In order to fix the seqnum property on the payloaders
+ gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
+ "seqnum-offset" from the "stats" property contains the value of the
+ very first RTP packet in a stream. The server will, however, try to look
+ at the last simple in the sink element and only use properties on the
+ payloader in case there no sink elements yet, and by looking at the last
+ sample of the sink gives the server full control of which RTP packet it
+ looks at. If the payloader does not have the "stats" property, "seqnum"
+ is still used since "seqnum-offset" is only present in as part of
+ "stats" and this is still an issue not solved with this patch.
+ Needed for gst-plugins-base!17
+
+2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Plug memory leak
+ Attaching a GSource to a context will increase the refcount. The idle
+ source will never be free'd since the initial reference is never
+ dropped.
+
+2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
+
+ * .gitlab-ci.yml:
+ Add Gitlab CI configuration
+ This commit adds a .gitlab-ci.yml file, which uses a feature
+ to fetch the config from a centralized repository. The intent is
+ to have all the gstreamer modules use the same configuration.
+ The configuration is currently hosted at the gst-ci repository
+ under the gitlab/ci_template.yml path.
+ Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
+
+2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
+
+ * .gitmodules:
+ * gst-rtsp-server.doap:
+ Update git locations to gitlab
+
+2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/meson.build:
+ meson: add new onvif types
+
+2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/meson.build:
+ Add ONVIF subclass headers to the installed headers in meson.build too
+
+2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-server-object.h:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: Declare GstRTSPServer struct before anything else
+ It's needed by all kinds of other headers, including the ones that are
+ required for defining the GstRTSPServer struct itself and its API.
+
+2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-client.h:
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ * gst/rtsp-server/rtsp-onvif-media.h:
+ * gst/rtsp-server/rtsp-onvif-server.h:
+ Mark all ONVIF-specific subclasses as Since 1.14
+
+2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-onvif-server.c:
+ * gst/rtsp-server/rtsp-onvif-server.h:
+ * gst/rtsp-server/rtsp-server-object.h:
+ * gst/rtsp-server/rtsp-server-prelude.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session.h:
+ Include ONVIF types from single-include rtsp-server.h
+ ... by actually making it a single-include header and moving everything
+ related to the GstRTSPServer type to rtsp-server-object.h instead.
+ Otherwise there are too many circular includes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=797361
+
+2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-latency-bin.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: use idle source in on_message_sent
+ When the underlying layers are running on_message_sent, this sometimes
+ causes the underlying layer to send more data, which will cause the
+ underlying layer to run callback on_message_sent again. This can go on
+ and on.
+ To break this chain, we introduce an idle source that takes care of
+ sending data if there are more to send when running callback
+ https://bugzilla.gnome.org/show_bug.cgi?id=797289
+
+2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Remove timeout GSource on cleanup
+ Avoids ending up with races where a timeout would still be around
+ *after* a client was gone. This could happen rather easily in
+ RTSP-over-HTTP mode on a local connection, where each RTSP message
+ would be sent as a different HTTP connection with the same tunnelid.
+ If not properly removed, that timeout would then try to free again
+ a client (and its contents).
+
+2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ autotools: fix distcheck
+
+2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-server/rtsp-latency-bin.c:
+ * gst/rtsp-server/rtsp-latency-bin.h:
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ onvif: encapsulate onvif part into a bin
+ ...and thus do not let onvif affect pipelines latency
+ https://bugzilla.gnome.org/show_bug.cgi?id=797174
+
+2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * tests/check/gst/client.c:
+ tests: client: Avoid bind() failures in tests
+ https://bugzilla.gnome.org/show_bug.cgi?id=797059
+
+2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/mediafactory.c:
+ New property for socket binding to mcast addresses
+ By default the multicast sockets are bound to INADDR_ANY,
+ as it's not allowed to bind sockets to multicast addresses
+ in Windows. This default behaviour can be changed by setting
+ bind-mcast-address property on the media-factory object.
+ https://bugzilla.gnome.org/show_bug.cgi?id=797059
+
+2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server-prelude.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-token.c:
+ * meson.build:
+ libs: fix API export/import and 'inconsistent linkage' on MSVC
+ Export rtsp-server library API in headers when we're building the
+ library itself, otherwise import the API from the headers.
+ This fixes linker warnings on Windows when building with MSVC.
+ Fix up some missing config.h includes when building the lib which
+ is needed to get the export api define from config.h
+ https://bugzilla.gnome.org/show_bug.cgi?id=797185
+
+2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: Add missing break statements
+ This resulted in warnings/assertions whenever one accessed the
+ max-mcast-ttl property.
+ CID #1439515
+ CID #1439523
+
+2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ meson: add gobject-cast-checks, glib-asserts, glib-checks options
+
+2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/meson.build:
+ * meson_options.txt:
+ * tests/check/meson.build:
+ meson: add option to disable build of rtspclientsink plugin
+
+2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson_options.txt:
+ meson: re-arrange options
+
+2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ * tests/check/meson.build:
+ * tests/meson.build:
+ meson: Use feature option for tests option
+ This was somehow missed the last time around.
+
+2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp-server/meson.build:
+ * meson.build:
+ meson: Maintain macOS ABI through dylib versioning
+ Requires Meson 0.48, but the feature will be ignored on older versions
+ so it's safe to add it without bumping the requirement.
+ Documentation:
+ https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
+
+2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/rtsp-sink/meson.build:
+ * meson.build:
+ meson: add pkg-config file for the rtspclientsink plugin
+
+2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ rtsp-client: Avoid reuse of channel numbers for interleaved
+ If a (strange) client would reuse interleaved channel numbers in
+ multiple SETUP requests, we should not accept them. The channel
+ numbers are used for looking up stream transports in the
+ priv->transports hash table, and transports disappear from the table
+ if channel numbers are reused.
+ RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
+ server to change the channel numbers suggested by the client.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796988
+
+2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/client.c:
+ rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
+ Allow regex for matching transport header against expected pattern.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796988
+
+2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * tests/check/meson.build:
+ meson: There is no gstreamer-plugins-good-1.0.pc
+ There is no installed version of that, only an uninstalled version.
+
+2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/stream.c:
+ Fix indentation again
+
+2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/stream.c:
+ stream: Added a list of multicast client addresses
+ When media is shared, the same media stream can be sent
+ to multiple multicast groups. Currently, there is no API
+ to retrieve multicast addresses from the stream.
+ When calling gst_rtsp_stream_get_multicast_address() function,
+ only the first multicast address is returned.
+ With this patch, each multicast destination requested in SETUP
+ will be stored in an internal list (call to
+ gst_rtsp_stream_add_multicast_client_address()).
+ The list of multicast groups requested by the clients can be
+ retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
+ There still exist some problems with the current implementation
+ in the multicast case:
+ 1) The receiving part is currently only configured with
+ regard to the first multicast client (see
+ https://bugzilla.gnome.org/show_bug.cgi?id=796917).
+ 2) Secondly, of security reasons, some constraints should be
+ put on the requested multicast destinations (see
+ https://bugzilla.gnome.org/show_bug.cgi?id=796916).
+ Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ stream: Choose the maximum ttl value provided by multicast clients
+ The maximum ttl value provided so far by the multicast clients
+ will be chosen and reported in the response to the current
+ client request.
+ Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ rtsp-stream: Don't require address pool in the transport specific case
+ If "transport.client-settings" parameter is set to true, the client is
+ allowed to specify destination, ports and ttl.
+ There is no need for pre-configured address pool.
+ Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ client: Don't reserve multicast address in the client setting case
+ When two multicast clients request specific transport
+ configurations, and "transport.client-settings" parameter is
+ set to true, it's wrong to actually require that these two
+ clients request the same multicast group.
+ Removed test_client_multicast_invalid_transport_specific test
+ cases as they wrongly require that the requested destination
+ address is supposed to be present in the address pool, also in
+ the case when "transport.client-settings" parameter is set to true.
+ Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/mediafactory.c:
+ Add new API for setting/getting maximum multicast ttl value
+ Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: avoid duplicating the first multicast client
+ In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
+ clients were dynamically added and removed to the multicast
+ udp sinks, as such we should no longer add a first client in
+ set_multicast_socket_for_udpsink
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ Revert "rtsp-stream: avoid duplicating the first multicast client"
+ This reverts commit 33570944401747f44d8ebfec535350651413fb92.
+ Commits where accidentially squashed together
+
+2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/mediafactory.c:
+ Revert "Add new API for setting/getting maximum multicast ttl value"
+ This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
+ Commits where accidentially squashed together
+
+2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ Revert "rtsp-stream: Don't require address pool in the transport specific case"
+ This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
+ Commits where accidentially squashed together
+
+2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/stream.c:
+ Revert "stream: Choose the maximum ttl value provided by multicast clients"
+ This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
+ Commits where accidentially squashed together
+
+2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-auth-digest.c:
+ examples: Fix indentation
+
+2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/stream.c:
+ stream: Choose the maximum ttl value provided by multicast clients
+ The maximum ttl value provided so far by the multicast clients
+ will be chosen and reported in the response to the current
+ client request.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ rtsp-stream: Don't require address pool in the transport specific case
+ If "transport.client-settings" parameter is set to true, the client is
+ allowed to specify destination, ports and ttl.
+ There is no need for pre-configured address pool.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/mediafactory.c:
+ Add new API for setting/getting maximum multicast ttl value
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: avoid duplicating the first multicast client
+ In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
+ clients were dynamically added and removed to the multicast
+ udp sinks, as such we should no longer add a first client in
+ set_multicast_socket_for_udpsink
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Add gstreamer-base gir dir in autotools
+
+2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-client: always allocate both IPV4 and IPV6 sockets
+ multiudpsink does not support setting the socket* properties
+ after it has started, which meant that rtsp-server could no
+ longer serve on both IPV4 and IPV6 sockets since the patches
+ from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
+ merged.
+ When first connecting an IPV6 client then an IPV4 client,
+ multiudpsink fell back to using the IPV6 socket.
+ When first connecting an IPV4 client, then an IPV6 client,
+ multiudpsink errored out, released the IPV4 socket, then
+ crashed when trying to send a message on NULL nevertheless,
+ that is however a separate issue.
+ This could probably be fixed by handling the setting of
+ sockets in multiudpsink after it has started, that will
+ however be a much more significant effort.
+ For now, this commit simply partially reverts the behaviour
+ of rtsp-stream: it will continue to only create the udpsinks
+ when needed, as was the case since the patches were merged,
+ it will however when creating them, always allocate both
+ sockets and set them on the sink before it starts, as was
+ the case prior to the patches.
+ Transport configuration will only error out if the allocation
+ of UDP sockets fails for the actual client's family, this
+ also downgrades the GST_ERRORs in alloc_ports_one_family
+ to GST_WARNINGs, as failing to allocate is no longer
+ necessarily fatal.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796875
+
+2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ meson: Convert common options to feature options
+ These are necessary for gst-build to set options correctly. The
+ remaining automagic option is cgroup support in examples.
+ https://bugzilla.gnome.org/show_bug.cgi?id=795107
+
+2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Slightly simplify locking
+
+2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ Limit queued TCP data messages to one per stream
+ Before, the watch backlog size in GstRTSPClient was changed
+ dynamically between unlimited and a fixed size, trying to avoid both
+ unlimited memory usage and deadlocks while waiting for place in the
+ queue. (Some of the deadlocks were described in a long comment in
+ handle_request().)
+ In the previous commit, we changed to a fixed backlog size of 100.
+ This is possible, because we now handle RTP/RTCP data messages differently
+ from RTSP request/response messages.
+ The data messages are messages tunneled over TCP. We allow at most one
+ queued data message per stream in GstRTSPClient at a time, and
+ successfully sent data messages are acked by sending a "message-sent"
+ callback from the GstStreamTransport. Until that ack comes, the
+ GstRTSPStream does not call pull_sample() on its appsink, and
+ therefore the streaming thread in the pipeline will not be blocked
+ inside GstRTSPClient, waiting for a place in the queue.
+ pull_sample() is called when we have both an ack and a "new-sample"
+ signal from the appsink. Then, we know there is a buffer to write.
+ RTSP request/response messages are not acked in the same way as data
+ messages. The rest of the 100 places in the queue are used for
+ them. If the queue becomes full of request/response messages, we
+ return an error and close the connection to the client.
+ Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
+
+2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Use fixed backlog size
+ Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
+ Preparation for the next commit, which changes to a different way of
+ avoiding both deadlocks and unlimited memory usage with the watch
+ backlog.
+
+2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: unref clock (if set) when finalizing
+ https://bugzilla.gnome.org/show_bug.cgi?id=796814
+
+2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ rtsp-media: add gst_rtsp_media_*_set_clock to docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=796814
+
+2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: unref old clock when setting new clock
+ https://bugzilla.gnome.org/show_bug.cgi?id=796724
+
+2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: unref clock in finalize
+ https://bugzilla.gnome.org/show_bug.cgi?id=796724
+
+2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ rtsp-onvif-media: fix g-ir-scanner warnings
+
+2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ .gitignore: add another example binary
+
+2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/meson.build:
+ meson: add new test-appsrc2 example to meson build
+
+2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/Makefile.am:
+ examples: fix build of new test-appsrc2 example
+ Need to link against libgstapp-1.0.
+
+2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-appsrc2.c:
+ examples: Add test-appsrc2
+ Add an example of feeding both audio and video into an RTSP
+ pipeline via appsrc.
+
+2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Strip transport parts as whitespaces could be around commas
+ https://bugzilla.gnome.org/show_bug.cgi?id=758428
+
+2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
+ Fix race when setting up source elements.
+ Since we set the source element(s) to PLAYING state before hooking
+ them up to the downstream funnel, it's possible for the source element
+ to receive packets before we actually get to linking it to the funnel,
+ in which case buffers would be pushed out on an unlinked pad, causing
+ it to error out and stop receiving more data.
+ We fix this by blocking the source's srcpad until we have linked it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796160
+
+2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix mismatch between allowed and configured protocols
+ https://bugzilla.gnome.org/show_bug.cgi?id=796679
+
+2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Emit a signal when the SRTP decoder is created
+ https://bugzilla.gnome.org/show_bug.cgi?id=778080
+
+2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't require presence of sinks in _get_*_socket()
+ Transport specific sink elements are added to the pipeline
+ in PLAY request and sockets are already created in SETUP so
+ it's actually wrong to require the presence of sinks in
+ _get_*_socket() functions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Update transport for multicast clients as well
+ If a multicast client requests different transport settings
+ than the existing one make sure that this new transport
+ configuruation is propagated to the multicast udp sink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
+ And not on unicast udp sinks
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ Update for g_type_class_add_private() deprecation in recent GLib
+
+2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ Fix indentation
+
+2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/Makefile.am:
+ * examples/test-video-disconnect.c:
+ examples: Add test-video-disconnect example
+ Simple example which cuts off all clients 10 seconds
+ after the first one connects.
+
+2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * examples/test-auth-digest.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ rtsp-auth: Add support for parsing .htdigest files
+ Passwords are usually not stored in clear text, but instead
+ stored already hashed in a .htdigest file.
+ Add support for parsing such files, add API to allow setting
+ a custom realm in RTSPAuth, and update the digest example.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796637
+
+2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ rtspclientsink: fix waiting for multiple streams
+ We were previously only ever waiting for a single stream to notify it's
+ blocked status through GstRTSPStreamBlocking. Actually count streams to
+ wait for.
+ Fixes rtspclientsink sending SDP's without out some of the input
+ streams.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796624
+
+2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: add missing auth methods
+
+2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: only create funnel if it didn't exist already.
+ This precented using multiple protocols for the same stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796634
+
+2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * examples/meson.build:
+ meson: build auth-digest example
+
+2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ Get payloader stats only for the sending streams
+ Get/set payloader properties only for streams that actually
+ contain a payloader element.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796523
+
+2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile: Don't hardcode libtool for g-i build
+ Similar to the other commits in core/base/bad
+
+2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
+
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
+ https://bugzilla.gnome.org/show_bug.cgi?id=796229
+
+2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Don't deadlock in preroll on early close
+ If the connection is closed very early, the flushing
+ marker might not get set and rtspclientsink can get
+ deadlocked waiting for preroll forever.
+ https://bugzilla.gnome.org/show_bug.cgi?id=786961
+
+2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ meson: Update option names to omit disable_ and with- prefixes
+ Also yield common options to the outer project (gst-build in our case)
+ so that they don't have to be set manually.
+
+2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: use -Wl,-Bsymbolic-functions where supported
+ Just like the autotools build.
+
+2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ * tests/check/Makefile.am:
+ configure: check for -good and -bad plugins only in uninstalled setup
+ Avoids confusing configure messages looking or a -good .pc file
+ that doesn't exist.
+ Also use plugindir variables that common macros set while at it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=795466
+
+2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix session timeout
+ When streaming data over TCP then is not the keep-alive
+ functionality working.
+ The reason is that the function do_send_data have changed
+ to boolean but the code is still checking the received result
+ from send_func with GST_RTSP_OK.
+ The result is that a successful send_func will always lead to
+ that do_send_data is returning false and the keep-alive will
+ not be updated.
+ https://bugzilla.gnome.org/show_bug.cgi?id=795321
+
+2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ Implement support for ULP Forward Error Correction
+ In this initial commit, interface is only exposed for RECORD,
+ further work will be needed in rtspsrc to support this for
+ PLAY.
+ https://bugzilla.gnome.org/show_bug.cgi?id=794911
+
+2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ Revert "rtsp-server: Switch around sendonly/recvonly attributes"
+ This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
+ While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
+ the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
+ the opposite, just like the ONVIF standard.
+ Let's follow those RFCs as we're doing RTSP here, and add a property at
+ a later time if needed to switch to the SDP RFC behaviour.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793964
+
+2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 3fa2c9e to ed78bee
+
+2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/rtspclientsink.c:
+ gst: Run everything through gst-indent again
+
+2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: query the position on active streams if media is complete
+ If the media is complete, i.e. one or more streams have been configured
+ with sinks, then we want to query the position on those streams only.
+ A query on an incomplete stream may return a position that originates from
+ an earlier preroll.
+ https://bugzilla.gnome.org/show_bug.cgi?id=794964
+
+2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: make sure not to use freed string
+ Set transport string to NULL after freeing it, so that
+ at worst we get a NULL pointer if constructing a new
+ transport string fails (which shouldn't really fail here).
+ Also check return value of that, just in case.
+ CID 1433768.
+
+2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: do not free string passed to take_header
+
+2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: do not take lock in request_aux_receiver
+ Added it right before pushing the previous commit, it is
+ incorrect and deadlocks because this function gets called
+ from the join_bin thread, which already holds the lock,
+ that's the reason why request_aux_sender didn't take the
+ lock either.
+
+2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: add API to enable retransmission requests
+ "do-retransmission" was previously set when rtx-time != 0,
+ which made no sense as do-retransmission is used to enable
+ the sending of retransmission requests, where as rtx-time
+ is used by the peer to enable storing of buffers in order
+ to respond to retransmission requests.
+ rtsp-media now also provides a callback for the
+ request-aux-receiver signal.
+ https://bugzilla.gnome.org/show_bug.cgi?id=794822
+
+2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: add rtx ssrc to mikey's crypto sessions
+ https://bugzilla.gnome.org/show_bug.cgi?id=794813
+
+2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
+ This in order to be able to decrypt the RTCP backchannel
+ https://bugzilla.gnome.org/show_bug.cgi?id=794813
+
+2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Send KeyMgmt header in ANNOUNCE response
+ When sending back an encrypted RTCP back channel, it is useful
+ for the client to know the encryption key.
+ https://bugzilla.gnome.org/show_bug.cgi?id=794813
+
+2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: extract handle_keymgmt from rtsp-client
+ rtspclientsink will also need to parse KeyMgmt headers
+ sent by the server to decrypt the RTCP backchannel stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=794813
+
+2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * tests/check/gst/rtspclientsink.c:
+ rtspclientsink: Fix client ports for the RTCP backchannel
+ This was broken since the work for delayed transport creation
+ was merged: the creation of the transports string depends on
+ calling stream_get_server_port, which only starts returning
+ something meaningful after a call to stream_allocate_udp_sockets
+ has been made, this function expects a transport that we parse
+ from the transport string ...
+ Significant refactoring is in order, but does not look entirely
+ trivial, for now we put a band aid on and create a second transport
+ string after the stream has been completed, to pass it in
+ the request headers instead of the previous, incomplete one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=794789
+
+2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client:Error handling when equal http session cookie
+ There are some clients that are sending same session cookie on random
+ basis.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753616
+
+2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: Fix compilation with latest GLib
+ rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
+ rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
+ data->factory = g_object_ref (factory);
+ ^
+
+2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * meson.build:
+ Back to development
+
+=== release 1.14.0 ===
+
+2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.14.0
+
+=== release 1.13.91 ===
+
+2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.13.91
+
+2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-onvif-client.h:
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ * gst/rtsp-server/rtsp-onvif-media.h:
+ * gst/rtsp-server/rtsp-onvif-server.h:
+ * gst/rtsp-server/rtsp-params.h:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server-prelude.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * gst/rtsp-server/rtsp-token.h:
+ rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
+ We need different export decorators for the different libs.
+ For now no actual change though, just rename before the release,
+ and add prelude headers to define the new decorator to GST_EXPORT.
+
+2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-media-factory.c:
+ rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
+ https://bugzilla.gnome.org/show_bug.cgi?id=794143
+
+=== release 1.13.90 ===
+
+2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.13.90
+
+2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: add Since tags and example for new API
+
+2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * tests/check/gst/permissions.c:
+ permissions: more bindings-friendly API
+ https://bugzilla.gnome.org/show_bug.cgi?id=793975
+
+2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * meson.build:
+ meson: enable more warnings
+
+2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Place netaddress meta on packets received via TCP
+ This allows us to later map signals from rtpbin/rtpsource back to the
+ corresponding stream transport, and allows to do keep-alive based on
+ RTCP packets in case of TCP media transport.
+ https://bugzilla.gnome.org/show_bug.cgi?id=789646
+
+2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: if OPEN failed, unqueue next command
+ As READY_TO_PAUSED can no longer return async, the RECORD
+ command will be queued before the OPEN command fails
+ (for example in case the server could not be connected),
+ and record then waits for ever.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793896
+
+2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: fix retrieval of custom payloader caps
+ If a bin is passed as the custom payloader, the caps of
+ its factory will be empty, the correct way to obtain the caps
+ is to query its sinkpad.
+
+2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: fix extra unref of custom payloader
+
+2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rspclientsink: fix recent code indentation
+
+2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: add missing get_type prototype
+
+2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: allow setting payloader as pad property
+ This was a FIXME item, and can be quite useful, also
+ allowing to specify payloader properties from the command
+ line, which is always nice.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793776
+
+2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Replace g_print() log line
+ https://bugzilla.gnome.org/show_bug.cgi?id=793838
+
+2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/rtspclientsink.c:
+ rtsp-media: fix RECORD getting stuck
+ The test_record case was working because async=false had
+ been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
+ but that was incorrect, as it should not be needed.
+ Removing async=false made the test fail as expected, this is
+ fixed by not trying to preroll when preparing the media for
+ RECORD, as start_prepare is called upon receiving ANNOUNCE,
+ and our peer will not start sending media until it has received
+ a response to that request, and sent and received a response
+ to RECORD as well, thus obviously preventing preroll.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793738
+
+2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ rtsp-auth: fix set_tls_authentication_mode annotation
+
+2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ rtp-server: remove redefined variable
+ res is a boolean variable which is defined in the function scope and
+ redefined, with no reason, in the loop scope. This patch removes the
+ redefinition.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793592
+
+2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: Add functions for checking if stream is receiver or sender
+ ...and replace all checks for RECORD in GstRTSPMedia which are really
+ for "sender-only". This way the code becomes more generic and introducing
+ support for onvif-backchannel later on will require no changes in
+ GstRTSPMedia.
+
+2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-onvif-media-factory.c:
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ onvif: Make requires_backchannel() public
+ ...in order to let subclasses building the onvif part of the pipeline
+ check whether backchannel shall be included or not.
+
+2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ rtsp-server: Switch around sendonly/recvonly attributes
+ They are wrong in the ONVIF streaming spec. The backchannel should be
+ recvonly and the normal media should be sendonly: direction is always
+ from the point of view of the SDP offerer (the server) according to
+ RFC 3264.
+
+2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-onvif-backchannel.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-onvif-client.c:
+ * gst/rtsp-server/rtsp-onvif-client.h:
+ * gst/rtsp-server/rtsp-onvif-media-factory.c:
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ * gst/rtsp-server/rtsp-onvif-media.h:
+ * gst/rtsp-server/rtsp-onvif-server.c:
+ * gst/rtsp-server/rtsp-onvif-server.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ rtsp: Add support for ONVIF backchannel
+ This adds a new RTSP server, client, media-factory and media subclass
+ for handling the specifics of the backchannel. Ideally this later can be
+ extended with other ONVIF specific features.
+
+2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Add support for sending+receiving medias
+ We need to add an appsrc/appsink in that case because otherwise the
+ media bin will be a sink and a source for rtpbin, causing a pipeline
+ loop.
+ https://bugzilla.gnome.org/show_bug.cgi?id=788950
+
+2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ * meson.build:
+ Back to development
+
+=== release 1.13.1 ===
+
+2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.13.1
+
+2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session-pool: remove nullable return annotation
+ create_watch can only return NULL from the API guards, no
+ need for nullable.
+
+2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ set_clock functions: Add nullable annotations
+
+2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ All around: add annotations and API guards
+
+2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * tests/test-cleanup.c:
+ test-cleanup: bind any port
+ The meson test suite runs tests in parallel, trying to bind
+ a single port made the test fail.
+
+2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: make version numbers ints and fix int/string comparison
+ WARNING: Trying to compare values of different types (str, int).
+ The result of this is undefined and will become a hard error
+ in a future Meson release.
+
+2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-context.c:
+ gst_rtsp_context_get_current: add (skip) annotation
+ The return value type is defined with G_DEFINE_POINTER_TYPE,
+ and gi emits the following warning:
+ Invalid non-constant return of bare structure or union; register as
+ boxed type or (skip)
+
+2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: add type annotations
+ gi doesn't seem to be able to figure out the type of the
+ signal parameters when defined with G_DEFINE_POINTER_TYPE
+
+2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ autotools: use -fno-strict-aliasing where supported
+ https://bugzilla.gnome.org/show_bug.cgi?id=769183
+
+2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: use -fno-strict-aliasing where supported
+ https://bugzilla.gnome.org/show_bug.cgi?id=769183
+
+2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mount-points: bail out of loop again when matching mount points
+ Previous patch led to us iterating the entire sequence. Bail out
+ of the loop again if we have a match but are moving away from it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771555
+
+2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/mountpoints.c:
+ tests: mountpoints: add more checks for mount point path matching
+ https://bugzilla.gnome.org/show_bug.cgi?id=771555
+
+2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mount-points: fix matching of paths where there's also an entry with a common prefix
+ e.g. with the following mount points
+ /raw
+ /raw/snapshot
+ /raw/video
+ _match() would not match /raw/video and /raw/snapshot correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771555
+
+2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * tests/check/gst/permissions.c:
+ permissions: add some new API to make this usable from bindings
+ https://bugzilla.gnome.org/show_bug.cgi?id=787073
+
+2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-token.c:
+ rtsp-token: annotate constructors for bindings
+ This maps _new_empty() to _new(), which also makes RTSPToken()
+ work properly now. Since this API wasn't usable from bindings
+ before, this should hopefully be fine.
+ https://bugzilla.gnome.org/show_bug.cgi?id=787073
+
+2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ * tests/check/gst/token.c:
+ rtsp-token: add some API to set fields from bindings
+ The existing functions are all vararg-based and as such
+ not usable from bindings.
+ https://bugzilla.gnome.org/show_bug.cgi?id=787073
+
+2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspclientsink.c:
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/sessionpool.c:
+ * tests/check/gst/stream.c:
+ tests: fix indentation
+ Fix and "fix".
+
+2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: rtspserver: fix another ref leak
+ Even if this didn't show up in valgrind.
+
+2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspclientsink.c:
+ tests: rtspclientsink: fix leak
+
+2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ test: rtspserver: plug memory leak in test_no_session_timeout
+ In test_no_session_timeout, unref the rtsp session object when the
+ test is done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=792127
+
+2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtpsclientsink: Initialize and clear newly added mutex and cond
+ While it *did* work, glib would automatically create new mutex and cond
+ ... which never got freed
+
+2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set multicast TTL on the multicast sockets
+ And not if we do unicast UDP.
+ https://bugzilla.gnome.org/show_bug.cgi?id=791743
+
+2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
+ In the multicast case (as in test-multicast, not test-multicast2), the
+ address could be allocated/reserved (and thus set) already without
+ allocating the actual socket. We need to allocate the socket here still
+ instead of just claiming that it was already allocated.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
+
+2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ rtspclientsink: Use the new rtsp-stream API
+ https://bugzilla.gnome.org/show_bug.cgi?id=790412
+
+2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ rtspclientsink: Wait until OPEN has been scheduled
+ Make sure that the sink thread has started opening connection
+ to the server before continuing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=790412
+
+2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From e8c7a71 to 3fa2c9e
+
+2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Minor doc fixes
+ Mostly for g-i
+
+2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ * tests/Makefile.am:
+ tests: disable all tests when --disable-tests is used
+ Move conditional subdir include into top level.
+ Based on patch by: Joel Holdsworth
+ https://bugzilla.gnome.org/show_bug.cgi?id=757703
+
+2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ * tests/meson.build:
+ meson: build more tests and add options to disable tests and examples
+
+2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/rtsp-server/rtsp-session.c:
+ Fix build when -Werror=deprecated-declarations is on
+ As gst_rtsp_session_next_timeout is deprecated.
+ ```
+ ../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]
+ res = (gst_rtsp_session_next_timeout (session, now) == 0);
+ ^~~
+ ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
+ gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
+ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ ```
+
+2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 3f4aa96 to e8c7a71
+
+2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * tests/check/gst/media.c:
+ check/media: Add seekability test case: not all streams are active
+ Media contains two streams but only one is complete and prepared
+ for playing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Do not reset 'blocking' if stream is already blocked
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: remove vs_module_defs_dir variable which is no longer needed
+
+2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp: fix distcheck
+
+2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * win32/MANIFEST:
+ * win32/common/libgstrtspserver.def:
+ win32: remove .def file with exports
+ They're no longer needed, symbol exporting is now explicit
+ via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
+
+2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ autotools: stop controlling symbol visibility with -export-symbols-regex
+ Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
+ This should result in consistent behaviour for the autotools and
+ Meson builds.
+
+2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: add missing GST_EXPORT and export deprecated funcs
+
+2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/check/gst/media.c:
+ check: Add seekability testing on medias
+ Make sure that once GstRTSPMedia are prepared they returned
+ the expected seekability results
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * win32/common/libgstrtspserver.def:
+ rtsp-media: Enable seeking query before pipeline is complete
+ SDP are now provided *before* the pipeline is fully complete. In order
+ to know whether a media is seekable or not therefore requires asking
+ the invididual streams.
+ API: gst_rtsp_stream_seekable
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix handling in default_unsuspend()
+ Handle the case when streams are not blocked and media
+ is suspended from PAUSED.
+ Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/media.c:
+ check/media: Fix thread pool leak.
+ Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Removed fakesink elements
+ There is not need of adding fakesink elements to the media
+ pipeline in the dynamic-payloader case.
+ The media pipeline itself is dynamically updated with
+ the receiver and sender parts that are based on the client
+ transport information known after SETUP has been received.
+ Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Corrected ASYNC_DONE handling
+ Media is complete when all the transport based parts are
+ added to the media pipeline. At this point ASYNC_DONE is
+ posted by the media pipeline and media is ready to enter
+ the PREPARED state.
+ Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/gst/media.c:
+ check/media: Check that prepared media can provide a SDP
+ Whenever a RTSPMedia is prepared, it should be able to provide a SDP
+
+2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Don't leak addr
+ CID #1422260
+
+2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ Run gst-indent
+
+2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't unblock with remaining dynamic payloaders
+ If we still have some dynamic paylaoders which haven't posted
+ no-more-pads yet, don't go to PREPARED if one of the streams
+ blocked.
+ The risk was that we would end up not exposing/using all specified
+ streams.
+ The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
+ then it will take a bit more time to start. But only if those 3
+ conditions are present.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769521
+
+2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix doc
+
+2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't set float on a gint64 variable
+ Just use 0. Fixes 'undefined' behaviour from clang
+
+2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix previous commit
+ We only want to count dynamic payloaders
+
+2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: Handle multiple dynamic elements
+ If we have more than one dynamic payloader in the pipeline, we need
+ to wait until the *last* one emits 'no-more-pads' before switching
+ to PREPARED.
+ Failure to do so would result in a race where some of the streams
+ wouldn't properly be prepared
+ https://bugzilla.gnome.org/show_bug.cgi?id=769521
+
+2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstrtspserver.def:
+ win32: Fix exported symbols list
+
+2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only update the RTP udpsink if it actually exists
+ For send-only streams it does not exist, but the RTCP udpsink might.
+
+2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstrtspserver.def:
+ win32: Update exports
+
+2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: seek on media pipelines that are complete
+ Make sure that a seek is performed on pipelines that
+ contain at least one sink element.
+ Change-Id: Icf398e10add3191d104b1289de612412da326819
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/media.c:
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/stream.c:
+ Dynamically reconfigure pipeline in PLAY based on transports
+ The initial pipeline does not contain specific transport
+ elements. The receiver and the sender parts are added
+ after PLAY.
+ If the media is shared, the streams are dynamically
+ reconfigured after each PLAY.
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: obtain stream position from pad
+ If no sinks have been added yet, obtain the current and
+ the stop position of the stream from the send_src pad.
+ Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ rtsp-session-media: add function to get a list of transports
+ Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: add functions to get rtp and rtcp multicast sockets
+ Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set async=sync=false only for RTCP appsink
+ Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: return minimum value in query position case
+ The minimum position should be returned as we are interested
+ in the whole interval.
+ Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-session: Handle the case when timeout=0
+ According to the documentation, a timeout of value 0 means
+ that the session never timeouts. This adds handling of that.
+ If timeout=0 we just return with a -1 from
+ gst_rtsp_session_next_timeout_usec ().
+ https://bugzilla.gnome.org/show_bug.cgi?id=785058
+
+2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
+ https://bugzilla.gnome.org/show_bug.cgi?id=785024
+
+2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ docs: add media factory transport mode accessors
+ and fix the documentation for the return value of the getter
+
+2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: unref 'pipelined_requests' in finalize
+ The hash table priv->pipelined_requests is not unref:ed in the
+ finalize funktion. Make sure it is.
+ https://bugzilla.gnome.org/show_bug.cgi?id=788704
+
+2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Initialize scalar variable
+ CID 1418985
+
+2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
+
+ * win32/common/libgstrtspserver.def:
+ win32: Update export file
+
+2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Start support for RTSP 2.0
+ This adds basic support for new 2.0 features, though the protocol is
+ subposdely backward incompatible, most semantics are the sames.
+ This commit adds:
+ - features:
+ * version negotiation
+ * pipelined requests support
+ * Media-Properties support
+ * Accept-Ranges support
+ - APIs:
+ * gst_rtsp_media_seekable
+ The RTSP methods that have been removed when using 2.0 now return
+ BAD_REQUEST.
+ https://bugzilla.gnome.org/show_bug.cgi?id=781446
+
+2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Use stream duration as stream-stop if segment was not configured with a stop
+ Allowing client to know stream duration when no seeking happened.
+ https://bugzilla.gnome.org/show_bug.cgi?id=783435
+
+2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: Don't cache any media if NULL was returned as key
+ The docs already mentioned this, but we actually stored it in the hash
+ table with key==NULL and leaked its reference forever.
+
+2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ rtspclientsink: Use a mutex for protecting against concurrent send/receives
+ This is a simple port of:
+ * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
+ * c438545dc9e2f14f657bc0ef261fff726449867b
+ * cd17c71dcea5c9310d21f1347c7520983e5869ac
+ in gst-plugins-good.
+
+2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: fix Memory leak in error case
+ https://bugzilla.gnome.org/show_bug.cgi?id=787059
+
+2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * pkgconfig/meson.build:
+ meson: don't install -uninstalled.pc file
+ https://bugzilla.gnome.org/show_bug.cgi?id=786457
+
+2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 48a5d85 to 3f4aa96
+
+2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix typo in debug message
+
+2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: hide symbols by default unless explicitly exported
+
+2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
+ Fixes meson warning about undefined @srcdir@.
+
+2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/meson.build:
+ meson: skip tests on windows for now
+ As we do in the other modules. As libgstcheck is currently not
+ built on windows. Fixes "Fallback variable 'gst_check_dep' in
+ the subproject 'gstreamer' does not exist"" Meson error.
+
+2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fix connection delay due to wrong assumption on last-sample
+ Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
+ multiudpsink's last-sample always comes from the payloader. Which
+ is wrong if auxiliary streams are multiplexed in the same stream.
+ So check the buffer's ssrc against the caps'ssrc before to use its
+ seqnum. If not the same ssrc just use the payloader as done prior
+ the commit above or when there is no last-sample yet.
+ https://bugzilla.gnome.org/show_bug.cgi?id=784094
+
+2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * meson.build:
+ meson: Allow using glib as a subproject
+
+2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: fix with-package-name option
+ https://bugzilla.gnome.org/show_bug.cgi?id=784082
+
+2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * Makefile.am:
+ Distribute meson_options.txt
+
+2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * Makefile.am:
+ And config.h.meson is no longer dist either
+
+2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * config.h.meson:
+ * meson.build:
+ meson: config.h.meson is no longer needed
+
+2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/meson.build:
+ * tests/meson.build:
+ meson: Fix building tests and activate them again
+
+2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/meson.build:
+ meson: Do not use path separator in test names
+ Avoiding warnings like:
+ WARNING: Target "elements/audioamplify" has a path separator in its name.
+
+2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ meson: add options to set package name and origin
+ https://bugzilla.gnome.org/show_bug.cgi?id=782172
+
+2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-params.h:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * gst/rtsp-server/rtsp-token.h:
+ Mark symbols explicitly for export with GST_EXPORT
+
+2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ * gst/rtsp-sink/Makefile.am:
+ Remove plugin specific static build option
+ Static and dynamic plugins now have the same interface. The standard
+ --enable-static/--enable-shared toggle are sufficient.
+
+2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ * meson.build:
+ Back to development
+
+=== release 1.12.0 ===
+
+2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.12.0
+
+=== release 1.11.91 ===
+
+2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.11.91
+
+2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 60aeef6 to 48a5d85
+
+2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ gi: Fix some annotations and docstrings
+
+2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/rtsp-server/meson.build:
+ * meson.build:
+ * meson_options.txt:
+ meson: Build gir
+
+2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From 39ac2f5 to 60aeef6
+
+=== release 1.11.90 ===
+
+2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.11.90
+
+2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-launch.c:
+ examples: make test-launch pipeline shared by default as well
+
+2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ gstreamer-rtsp-server: Add both srcdir and builddir to the include path
+ Just the build dir is not going to work for srcdir!=builddir.
+
+2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * meson.build:
+ meson: Update version
+
+2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.11.2 ===
+
+2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.11.2
+
+2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ meson: dist meson build files
+ Ship meson build files in tarballs, so people who use tarballs
+ in their builds can start playing with meson already.
+
+2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record.c:
+ examples/test-record: Add extra line to initial printout
+ Add an example line of how to deliver a stream to the
+ RTSP RECORD example
+
+2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
+ If there is no Content-Length header, no body would be allocated and the
+ '\0' would also not be appended to the body.
+
+2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
+ While they logically have 0 bytes length, GstRTSPConnection is appending
+ a '\0' to everything making the size be 1 instead.
+
+2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: bump version
+
+2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
+ gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
+ affected.
+
+2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.11.1 ===
+
+2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * win32/common/libgstrtspserver.def:
+ Release 1.11.1
+
+2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: corrected if-statement in _get_server_port()
+ This bug was accidentally introduced while fixing a segfault
+ in _get_server_port() function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776345
+
+2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/stream.c:
+ rtsp-stream: fixed segmenation fault in _get_server_port()
+ Calling function gst_rtsp_stream_get_server_port() results in
+ segmenation fault in the RTP/RTSP/TCP case.
+ Port that the server will use to receive RTCP makes only
+ sense in the UDP case, however the function should handle
+ the TCP case in a nicer way.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776345
+
+2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ dosc: Fix a little typo
+ https://bugzilla.gnome.org/show_bug.cgi?id=777037
+
+2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/meson.build:
+ meson: generate pkg-config -uninstalled pc files
+ Generating those files is useful for users building the GStreamer stack
+ using meson and having to link it to another project which is still
+ using the autotools.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776810
+
+2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ pkgconfig: fix -uninstalled pc file
+ pcfiledir was never defined so the paths were wrong.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776867
+
+2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-stream: Fixed TCP transport case
+ Make sure that the appsink element is actually added to
+ the bin before trying to link it with the elements in it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776343
+
+2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ * gst-rtsp.spec.in:
+ Remove generated .spec file
+ Likely extremely bitrotten, and we should not ship this anyway.
+
+2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From f980fd9 to 39ac2f5
+
+2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Fix pt map caps
+ Since decryption is handled within rtpbin, all outcoming stream
+ caps will be application/x-rtp (i.e. regular rtp)
+ Fixes RECORD with SRTP streams
+
+2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: Create media objects with the proper transport mode
+ The function called immediately afterwards (collect_streams()) will
+ need it to work properly
+
+2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
+
+2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: Don't create a pipeline for the media pipeline string
+ We're going to put a pipeline into a pipeline otherwise, which is not
+ exactly ideal.
+
+2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Fix race condition around finish_unprepare() if called multiple time
+ https://bugzilla.gnome.org/show_bug.cgi?id=755329
+
+2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Don't leave stale pointer after unref
+ Fix a warning on shutdown - don't keep a pointer to an
+ alread-unreffed object.
+
+2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitmodules:
+ common: use https protocol for common submodule
+ https://bugzilla.gnome.org/show_bug.cgi?id=775110
+
+2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: block the output of rtpbin instead of the source pipeline
+ 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
+ detection of the srtp rollover counter to add to the SDP.
+ Unfortunately, it was incomplete for live pipelines where the logic
+ blocks the source bin before creating the SDP and thus would never have
+ the necessary informaiton to create a correct SDP with srtp encryption.
+ Move the pad blocks to rtpbin's output pads instead so that the
+ necessary information can be created before we need the information for
+ the SDP.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770239
+
+2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: add IDLE timeout, before session exists
+ The RTSP server will not timeout an idle RTSP connection
+ (note this is different from doing timeout on a RTSP
+ session).
+ At least for Apache this is a problem when running RTSP over
+ HTTPS since it uses one of the threads (there is a rather
+ limited number) that are available for handling requests.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771830
+
+2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ .gitignore more
+
+2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set close-socket FALSE on UDP src:es
+ With this RTSP server can use the sockets independent on the udpsrc
+ state.
+ When the udp src is finalized it will unref socket and when g_socket
+ is finalized the socket will be closed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765673
+
+2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Move to new helper function to parse authentication responses
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ * examples/test-auth-digest.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * win32/common/libgstrtspserver.def:
+ rtsp-auth: Add support for Digest authentication
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * meson.build:
+ * tests/check/meson.build:
+ * win32/MANIFEST:
+ * win32/common/libgstrtspserver.def:
+ Enable building with MSVC
+ https://bugzilla.gnome.org/show_bug.cgi?id=774640
+
+2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * meson.build:
+ meson: gstreamer gst_check_dep does not exist on windows
+
+2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: update do_send_message to match type GstRTSPClientSendFunc
+ This type mismatch fails building with MSVC
+ https://bugzilla.gnome.org/show_bug.cgi?id=774640
+
+2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Fix indentation
+
+2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only signal "new-state" if the state has actually changed
+ https://bugzilla.gnome.org/show_bug.cgi?id=774173
+
+2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: emit signal in the beginning of each rtsp request
+ These signals let the application validate the requests, configure the
+ media/stream in a certain way and also generate error status code in
+ case of error or bad request.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758062
+
+2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: update version
+
+=== release 1.11.0 ===
+
+2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.10.0 ===
+
+2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.10.0
+
+2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/stream.c:
+ tests: try to avoid using the same ports in different tests
+ Causes problems with client multicast tests otherwise if
+ tests are run in parallel.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773640
+
+2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/client.c:
+ tests: client: use fail_unless_equals_foo() for better failure reporting
+
+2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Session filter in unwatch session
+ Call session filter with filter_session_media as paramer in
+ client_unwatch_session if using drop_backlog = FALSE.
+ In client_unwatch_session its allowed to grow the watchs backlog.
+ If using drop_backlog = FALSE and the backlog is full it will cause
+ a deadlock when setting session media state to NULL
+ if the backlog is not allowed to grow.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771983
+
+2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: add fallbacks for gst modules
+ For gst-all.
+
+2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix factory leaking in find_media() in error cases
+ https://bugzilla.gnome.org/show_bug.cgi?id=771488
+
+2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Fix randomly missing streams from SDP with dynamic elements
+ When using dynamic elements, gst_rtsp_stream_join_bin() is called from
+ "pad-added" signal. In that case priv->srcpad could already have its caps,
+ and they'll be sent to priv->send_src[0] pad. That means that when it
+ connects "notify::caps" signal, that pad could already have received its
+ caps and the signal won't be emitted anymore.
+ In that case priv->caps stay to NULL and when building the SDP that stream
+ gets ignored. Leading to missing video or audio when playing in client side.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772478
+
+2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: update version
+
+=== release 1.9.90 ===
+
+2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.9.90
+
+2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Hint that set_multicast_iface expects the name of the interface
+ To prevent any possibly confusion with IPs or anything else.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771530
+
+2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
+ https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
+
+2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Depend on gstreamer 1.9.2.1
+
+2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From b18d820 to f980fd9
+
+2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From 6f2d209 to b18d820
+
+2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Remove unused _locked() variant of a function
+ It was added during refactoring.
+
+2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: cosmetic cleanup
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Compare IP addresses case insensitive in more places
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * common:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Fix leaked joined_bin
+ There is no need to keep a strong ref on it, and _leave_bin() was
+ setting it to NULL before calling g_clear_object() so it was leaked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Compare IP address strings case insensitive
+ Otherwise IPv6 addresses might fail this comparision.
+
+2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Bind multicast sockets to ANY as before
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
+
+2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: Fix segfault when doing keep-alive after removing the session
+ If keep-alive happens after removing the session but before finalizing the
+ stream transport, we would segfault.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750544
+
+2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Always create multicast UDP elements if the protocol flag is set
+ Adding them later will cause deadlocks due to
+ 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
+ 2) adding the multicast sink
+ 3) waiting for it to get data to preroll again
+ 3) never happens because the queues after the tee are full.
+
+2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix up various multicast related issues
+
+2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/gst/stream.c:
+ tests: Fix compilation
+
+2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/stream.c:
+ stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
+ This is basically reverting changes introduced in commit f62a9a7,
+ because it was introducing various regressions:
+ - It introduces a leak of udpsrc elements that got wrongly fixed by adding
+ an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
+ ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
+ - If a mcast client connects, it creates a new socket in SETUP to try to respect
+ the destination/port given by the client in the transport, and overrides the
+ socket already set on the udpsink element. That means that if we already had a
+ client connected, the source address on the udp packets it receives suddenly
+ changes.
+ - If a 2nd mcast client connects, the destination/port in its transport is
+ ignored but its transport wasn't updated.
+ What this patch does:
+ - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
+ - Always have a tee+queue when udp is enabled. This could be optimized
+ again in a later patch, but is more complicated. If no unicast clients
+ connects then those elements are useless, this could be also optimized
+ in a later patch.
+ - When mcast transport is added, it creates a new set of udpsrc/udpsink,
+ seperated from those for unicast clients. Since we already support only
+ one mcast address, we also create only one set of elements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: factor our plug_src function
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: factor out plug_sink function
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: small documentation clarification
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Keep a ref on joined bin
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: code cleanup
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: small fix in error code path
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
+ This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
+ but keeps unit tests.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.9.2 ===
+
+2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.9.2
+
+2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * config.h.meson:
+ * examples/meson.build:
+ * gst/meson.build:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-sink/meson.build:
+ * meson.build:
+ * pkgconfig/meson.build:
+ * tests/check/meson.build:
+ * tests/meson.build:
+ Add support for Meson as alternative/parallel build system
+ https://github.com/mesonbuild/meson
+
+2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * configure.ac:
+ * tests/check/Makefile.am:
+ build: silence error about pthread for 'make check' in osx
+ Fixes "clang: error: argument unused during compilation: '-pthread'"
+
+2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix leaking of media in error cases
+ With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
+ and myself to make the media refcounting a bit easier to follow.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755632
+
+2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix leaking of session in error cases
+ https://bugzilla.gnome.org/show_bug.cgi?id=755632
+
+2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f363b32 to f49c55e
+
+2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.9.1 ===
+
+2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.9.1
+
+2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * configure.ac:
+ configure: Need to add -DGST_STATIC_COMPILATION when building only statically
+ https://bugzilla.gnome.org/show_bug.cgi?id=767463
+
+2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * common:
+ Automatic update of common submodule
+ From ac2f647 to f363b32
+
+2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ sdp: add rollover counters for all sender SSRC
+ We add different crypto sessions in MIKEY, one for each sender
+ SSRC. Currently, all of them will have the same security policy, 0.
+ The rollover counters are obtained from the srtpenc element using the
+ "stats" property.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730539
+
+2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-server.h:
+ docs: fix some typos
+
+2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ g-i: pass compiler env to g-ir-scanner
+ It's what introspection.mak does as well. Should
+ fix spurious build failures on gnome-continuous
+ (caused by g-ir-scanner getting compiler details
+ via python which is broken in some environments
+ so passing the compiler details bypasses that).
+
+2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
+ This works with rtspsrc and live555, but fails with e.g. ffmpeg.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766619
+
+2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Check return value of sscanf
+ And just make sure we always have 0/0 if we have an error
+ CID #1352031
+
+2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/stream.c:
+ rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
+ - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
+ - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
+ - Create unit test for shared media.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764744
+
+2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
+ For IPv6 addresses, binding to a multicast group does not work on Linux
+ either. Always bind to ANY and then later join the multicast group.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764679
+
+2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
+
+ * common:
+ Automatic update of common submodule
+ From 6f2d209 to ac2f647
+
+2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ rtsp-thread-pool: explained why GSource is a part of ThreadImpl
+ Clarified why it is necessary to add source information to
+ GstRTSPThreadImpl. See the reported bug in GLib:
+ https://bugzilla.gnome.org/show_bug.cgi?id=720186
+ for more information.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761702
+
+2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ examples: Clean up CFLAGS/LDADD even more
+ The internal .la should come first and is part of LDADD, as is
+ GST_CFLAGS/LIBS.
+
+2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
+
+2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
+
+2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: Implement clock signalling according to RFC7273
+ For NTP and PTP clocks we signal the actual clock that is used and signal
+ the direct media clock offset.
+ For all other clocks we at least signal that it's the local sender clock.
+ This allows receivers to know which clock was used to generate the media and
+ its RTP timestamps. Receivers can then implement network synchronization,
+ either absolute or at least relative by getting the sender clock rate directly
+ via NTP/PTP instead of estimating it from RTP timestamps and packet receive
+ times.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760005
+
+2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Add support for setting the multicast interface
+ https://bugzilla.gnome.org/show_bug.cgi?id=763000
+
+2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Add support for setting the multicast interface
+ https://bugzilla.gnome.org/show_bug.cgi?id=763000
+
+2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: use new gst_element_class_add_static_pad_template()
+ https://bugzilla.gnome.org/show_bug.cgi?id=763196
+
+2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.8.0 ===
+
+2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.8.0
+
+2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
+ This would get us NO_PREROLL in the bin again and break seeking.
+ Thanks to Carlos Rafael Giani for helping to debug this!
+ https://bugzilla.gnome.org/show_bug.cgi?id=740509
+
+=== release 1.7.91 ===
+
+2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.7.91
+
+2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
+ Without this, RECORD pipelines are broken because
+ a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
+ added later. Previously it was there earlier and due to NO_PREROLL caused the
+ pipeline to preroll immediately
+ b) the udpsrc for the pipeline is added later and never set to PLAYING state,
+ as the corresponding code previously was only for PLAY pipelines.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763281
+
+2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix typo in the docstring
+ gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
+
+2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Disable multicast loopback for all our sockets
+ On Windows this is a receiver-side setting, on Linux a sender-side setting. As
+ we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
+ loopback setting on the socket... while udpsink does which unfortunately has
+ no effect here on Windows but on Linux.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/stream.c:
+ stream tests: added new tests
+ Test a case when the address pool only contains multicast addresses
+ and the client is requesting unicast udp.
+ Added tests for multicast ports allocation.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only bind multicast sockets to ANY on Windows
+ On Linux it is still needed to bind to the multicast address
+ to filter out random other packets, while on Windows binding
+ to multicast addresses just fails.
+
+2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
+ Otherwise we fail to allocate UDP ports if the pool only contains multicast
+ addresses, which is something that used to work before. For unicast addresses
+ if the pool contains none, we just allocate them as if there is no pool at
+ all.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Fix indentation
+
+2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't bind the sockets to multicast addresses
+ This works on Linux but fails completely on Windows. You're supposed
+ to bind to ANY and then join the multicast group.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+=== release 1.7.90 ===
+
+2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.7.90
+
+2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From b64f03f to 6f2d209
+
+2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * tests/check/gst/rtspclientsink.c:
+ rtspsink: Fix some leaks in rtspclientsink and the unit test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762525
+
+2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/media.c:
+ * tests/check/gst/rtspclientsink.c:
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/stream.c:
+ tests: unit test fixes
+ Removed port allocation test from the media suite.
+ The port allocation failure is now in the stream suite.
+ rtspserver:
+ Make sure that the media is suspended after the DESCRIBE request
+ before reconfiguring the UDP sinks.
+ rtspclientsink:
+ In the RECORD case we have to set async property to false
+ for the appsink element in the test in order to make sure
+ that the media pipeline doesn't hang in start_preroll().
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: postpone UDP socket allocation until SETUP
+ Postpone the allocation of the UDP sockets until we know
+ what transport has been chosen by the client.
+ Both unicast and multicast UDP sources are created in one
+ function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: postpone the creation of the UDP sources
+ Code refactoring: allocate the UDP ports after the sender and
+ the reciver parts have been created.
+ We postpone the creation of the UDP sources until the UDP
+ ports have been allocated.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: added function for setting UDP sources to PLAYING state
+ Code refactoring: Introduced a function for setting UDP sources
+ to PLAYING state.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: added function for creating and configuring UDP sources
+ Code refactoring: create and configure UDP sources in a separate function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: added function for RTP/RTCP socket configuration
+ Code refactoring: configure RTP and RTCP sockets for UDP sinks
+ in a separate function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: added function for creating and configuring UDP sinks
+ Code refactoring: create and configure UDP sinks in a separate function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: added helper function for creating the sender/receiver parts
+ Code refactoring: introduced helper function for creating
+ the receiver and the sender parts of the streaming pipeline.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.7.2 ===
+
+2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.7.2
+
+2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
+
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ uninstalled.pc: add support for non libtool build systems
+ Currently the .la path is provided which requires to use libtool as
+ mentioned in the GStreamer manual section-helloworld-compilerun.html.
+ It is fine as long as the application is built using libtool.
+ So currently it is not possible to compile a GStreamer application
+ within gst-uninstalled with CMake or other build system different
+ than autotools.
+ This patch allows to do the following in gst-uninstalled env:
+ gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
+ gstreamer-rtsp-server-1.0)
+ Previously it required to prepend libtool --mode=link
+ https://bugzilla.gnome.org/show_bug.cgi?id=720778
+
+2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: remove check for impossible condition
+ Goto error label checks stream to see if it needs to be unreferenced before
+ returning, but this goto jumps happens before the stream is ever set, so it
+ will always be NULL in this error label.
+ CID #1352034
+
+2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: clean switch statements
+ Coverity demands for fallthrough statements to be clearly commented,
+ to distinguish from accidental fall throughs. And it also needs all
+ cases to finish with a break, even if the break is never going to be
+ executed like in the case of a continue jump.
+ CID #1352039
+ CID #1352040
+
+2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/Makefile.am:
+ tests: extend the AM_TESTS_ENVIRONMENT from check.mak
+ To get the CK_DEFAULT_TIMEOUT defined for all tests
+ Also removes a 120 seconds timeout that was set as default
+ explicitly in this module
+ https://bugzilla.gnome.org/show_bug.cgi?id=761472
+
+2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From 86e4663 to b64f03f
+
+2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix state_lock not locked again when preroll fails
+ https://bugzilla.gnome.org/show_bug.cgi?id=761399
+
+2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Move plugin specific flags below all the others
+ They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
+ -no-undefined. And -no-undefined is required on Windows to build DLLs.
+
+2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Simplify slightly using new -base API
+ Use the new Mikey and SDP API in the base plugins libs
+ to simplify some code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * .gitignore:
+ * configure.ac:
+ * gst/Makefile.am:
+ * gst/rtsp-sink/Makefile.am:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ * gst/rtsp-sink/plugin.c:
+ * tests/check/Makefile.am:
+ * tests/check/gst/rtspclientsink.c:
+ rtspsink: Add rtspclientsink element
+ Add an rtspclientsink element that accepts streams for which
+ there is a registered payloader and sends them to
+ an RTSP server using RECORD.
+ Sending is synchronised to the pipeline clock. Payload-types
+ are automatically selected. The 'new-payloader' signal is fired
+ for custom configuration of payloaders when they are created.
+ Can now stream a movie like this:
+ receiver:
+ ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
+ decodebin name=depay1 ! audioconvert ! autoaudiosink )"
+ sender:
+ gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
+ queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
+ https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: Add functions for using rtsp-stream from the client
+ Add a boolean to indicate that the rtsp-stream is running on the
+ 'client' side of an RTSP connection, for sending streams via
+ RECORD. In that case, the roles of the client/server ports
+ in transport setup are swapped.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ rtsp-sdp: Add gst_rtsp_sdp_from_stream()
+ A new function that adds info from a GstRTSPStream into an SDP message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix mutex beeing unlocked while they should be locked
+ https://bugzilla.gnome.org/show_bug.cgi?id=761226
+
+2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: add missing break in "clock" property setter
+ CID 1348453
+
+2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fixed assert during update transport
+ When RTSP server trying update transport during multicast, it throws an
+ assert. The assert is thrown because it is trying to get the parent of
+ an non-existing funnel element.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760150
+
+2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * gst/rtsp-server/rtsp-token.h:
+ docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
+ gtk-doc can handle static inline functions just fine these days,
+ there's no need for this stuff any more.
+
+2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: replace duplicated codes to call new base sdp apis
+ https://bugzilla.gnome.org/show_bug.cgi?id=745880
+
+2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock.c:
+ test-netclock: Use the new API to configure a clock directly
+
+2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Add API to directly configure a clock on the media pipelines
+
+2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
+
+2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: Add FIXME for 2.0
+
+2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix indentation
+
+2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Do not prepare media after media times out
+ Deferred calls to start_prepare() can be deferred past the point until
+ which wait_preroll() and by proxy gst_rtsp_media_get_status() is
+ prepared to wait. Previously there was no lock and no check for this
+ situation. This meant that a media could be prepared and unprepared
+ simultaneously by two different threads. Now a lock is in place and a
+ suitable check is done.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
+
+2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
+ Without TEARDOWN it might be desireable to keep the media running and continue
+ sending data to the client, even if the RTSP connection itself is
+ disconnected.
+ Only do this for session medias that have only UDP transports. If there's at
+ least on TCP transport, it will stop working and cause problems when the
+ connection is disconnected.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758999
+
+2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.7.1 ===
+
+2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.7.1
+
+2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
+
+ * configure.ac:
+ configure: Make -Bsymbolic check work with clang.
+ Update the -Bsymbolic check with the version glib has. This version
+ works with clang.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759713
+
+2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-session-pool: Avoid dollar sign ($) in session ids
+ Live555 in VLC strips off dollar signs and then gets very confused,
+ we don't loose too much entropy by just skipping it.
+
+2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * gst/rtsp-server/rtsp-token.h:
+ rtsp-server: Add g_autoptr() support to all types
+ https://bugzilla.gnome.org/show_bug.cgi?id=754464
+
+2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fixed valgrind error
+ Fixed the valgrind error in unit test. The UDP source created during
+ gst_rtsp_stream_join_bin() was not released while destroying the rtp
+ bin.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759010
+
+2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From b319909 to 86e4663
+
+2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: suspend media during setup request
+ SETUP request from clients needs to suspend the media to clear the
+ prerolled buffers. Otherwise it will not affect the prerolled buffer
+ and the prerolled buffers will be incorrect (for example block-size
+ from setup request will not affect the prerolled buffer unless the
+ media is suspended).
+ https://bugzilla.gnome.org/show_bug.cgi?id=758268
+
+2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: create stream pipeline based on transport
+ Based on the protocol, create the rtsp stream pipeline. If only TCP or
+ only UDP is set as the transport protocol, it will not add the extra tee
+ or queue element to the pipeline. Both these elements will be added, if
+ it supports both TCP and UDP protocols. This improves the pipeline
+ performance when one protocol is present.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758179
+
+2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
+ Adding them when not needed will start some logic inside rtpbin that might be
+ problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
+ would start up a rtpjitterbuffer and behave in weird ways.
+ We still set up the UDP sources for RTP receiving for a sender media to be
+ able to receive any packets sent by the client for NAT traversal. They will
+ all go to a fakesink though.
+ Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
+ NO_PREROLL, which will cause deadlocks when seeking the media as it will never
+ receive ASYNC_DONE after a seek.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758319
+
+2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Disable multicast loopback for the multicast udp sources too
+ On POSIX this setting is for sender sockets, on Windows for receiver sockets.
+ Previously we were only setting this for sender sockets, which caused looped
+ back packets to be received on Windows if a multicast transport was used.
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record-auth.c:
+ * examples/test-record.c:
+ examples: Actually use the provided port in the record examples
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record-auth.c:
+ test-record-auth: Add the option to build in TLS support
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-auth.c:
+ test-auth: Use an 'anonymous' user for unauthenticated default
+ There's a comment on one of the resources that 'user' and 'admin'
+ shouldn't even be able to see it, but they can if the default
+ token is 'admin2', since that gives them access anyway.
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-record-auth.c:
+ Add test-record-auth example
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
+
+2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: Change the logic so we don't pop a NULL context
+ When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
+ will sometimes fail. This call is made before any context is pushed
+ resulting in an attempt to pop a NULL context.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757949
+
+2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ rtspserver: Add udp-mcast transport SETUP test
+ Refactor utility functions in the test file so they can handle
+ more than UDP and TCP as lower transport.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756969
+
+2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Always unref return value of gst_object_get_parent()
+ Fixes a leak of a GstBin in the udp-mcast case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756968
+
+2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From b99800a to b319909
+
+2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Use new GST_ENABLE_EXTRA_CHECKS #define
+ https://bugzilla.gnome.org/show_bug.cgi?id=756870
+
+2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 6babecd to b99800a
+
+2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Update GLib dependency to 2.40.0
+
+2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * examples/test-mp4.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: listen to sender ssrc signals
+ https://bugzilla.gnome.org/show_bug.cgi?id=746747
+
+2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ common: update for new suppression
+ Makes check-valgrind pass with glib 2.46
+
+2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Take reference to media that will be prepared
+ default_prepare() takes a transfer-none reference GstRTSPMedia object.
+ Later on a g_idle_source_new() is created and a pointer to the media
+ object is passed as user data. If the media is freed before the idle
+ source is dispatched the media object pointer is invalid, but the idle
+ source callback expects it to still be valid. To fix this a reference to
+ the media object is taken when registering the source callback function
+ and a corresponding release of the reference is done when the souce is
+ destroyed.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
+
+2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-record.c:
+ * examples/test-uri.c:
+ rtsp-server: Fix memory leaks when context parse fails
+ When g_option_context_parse fails, context and error variables are not getting free'd
+ which results in memory leaks. Free'ing the same.
+ And replacing g_error_free with g_clear_error, which checks if the error being passed
+ is not NULL and sets the variable to NULL on free'ing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753863
+
+2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.6.0 ===
+
+2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.6.0
+
+=== release 1.5.91 ===
+
+2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.91
+
+2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: fix docs for recently-added get/set_buffer_size API
+ https://bugzilla.gnome.org/show_bug.cgi?id=749095
+
+2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't crash on encrypted RTX SDP
+ In parse_keymgmt(), don't mutate the input string that's been passed
+ as const, especially since we might need the original value again if
+ the same key info applies to multiple streams (RTX, for example).
+ https://bugzilla.gnome.org/show_bug.cgi?id=754753
+
+2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-mp4.c:
+ test-mp4: Support filenames with spaces in them. Error out on too few arguments
+
+2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Check parameter count and print out help
+ If no launch pipeline was supplied, print out some help
+
+2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: Implement UDP buffer size setting.
+ Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
+ UDP TX buffer size.
+ Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
+
+2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Fix small typo causing gtk-doc to complain
+
+=== release 1.5.90 ===
+
+2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.90
+
+2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: get port number through gst_rtsp_url_get_port
+ https://bugzilla.gnome.org/show_bug.cgi?id=753473
+
+2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
+
+ * tests/check/gst/media.c:
+ media-test: Removing unnecessary assertion
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Document that source keeps a ref on server until it's destroyed
+ https://bugzilla.gnome.org/show_bug.cgi?id=749227
+
+2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * tests/check/gst/media.c:
+ media-test: Test for multiple dynamic payload
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Only add fakesink once per pipeline
+ The intention is to prevent going PLAYING state before pads are created.
+ If there was mutilple dynamic payload, it would leak few fakesink and
+ actually prevent from ever reaching playing state.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Revert "rtsp-media: Only add 1 fakesink per pipeline"
+ This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
+
+2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only add 1 fakesink per pipeline
+ There should be only one fakesink per pipeline, not per dynpay. This
+ would lead to element naming clash.
+
+2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: assertion error due to wrong condition check
+ In media to caps function, reserved_keys array is being used for variable i,
+ leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
+ changed it to variable j
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Strip keys from the fmtp that we use internally in our caps
+ Skip keys from the fmtp, which we already use ourselves for the
+ caps. Some software is adding random things like clock-rate into
+ the fmtp, and we would otherwise here set a string-typed clock-rate
+ in the caps... and thus fail to create valid RTP caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
+ https://bugzilla.gnome.org/show_bug.cgi?id=752640
+
+2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f74b2df to 9aed1d7
+
+2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.5.2 ===
+
+2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.2
+
+2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * tests/check/gst/client.c:
+ rtsp-client: allow application to decide what requirements are supported
+ Add "check-requirements" signal and vfunc to allow application
+ (and subclasses) to check the requirements.
+ Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=749417
+
+2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6015d26 to f74b2df
+
+2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Always use real payloader when creating streams
+ A bin that contains the real payloader might be used as payloader. In this
+ case we have to get the real payloader for the various properties it provides.
+ Example use cases for this are bins that payload some media and then have
+ additional elements that add metadata or RTP extension headers to the stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750800
+
+2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
+
+2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ * examples/test-netclock.c:
+ test-netclock: Use new ntp-time-source property on rtpbin
+ Select the clock time to be used as NTP time source. This allows proper
+ synchronization between receivers, independent of sharing base times, and just
+ requires them to use the same clock.
+
+2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ * examples/test-netclock.c:
+ test-netclock: Setting the same base time on sender and receiver is not necessary
+ It's going to be fixed up by rtpbin when using ntp-sync=TRUE
+
+2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
+ https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * docs/libs/gst-rtsp-server.types:
+ docs: add missing types
+ https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: add missing apis
+ https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
+
+2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ GstRTSPAuth: Add client certificate authentication support
+ https://bugzilla.gnome.org/show_bug.cgi?id=750471
+
+2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock-client: Use new GstClock API to wait for clock synchronization
+
+2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock-client: Use a GMainLoop and playbin's source-setup signal
+ A mainloop is needed to get glimagesink to display something on OSX, and
+ the source-setup signal just makes things a little bit easier.
+
+2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From d9a3353 to 6015d26
+
+2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From d37af32 to d9a3353
+
+2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 21ba2e5 to d37af32
+
+2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From c408583 to 21ba2e5
+
+2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/libs/Makefile.am:
+ docs: remove variables that we define in the snippet from common
+ This is syncing our Makefile.am with upstream gtkdoc.
+
+2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 44a3517 to c408583
+
+2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.5.1 ===
+
+2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.1
+
+2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: No flush during Teardown.
+ When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
+ backlog is empty it can happen that just a part of a message will be
+ sent and rest is in backlog queue. If then flush during teardown
+ just a part of message will be sent.This can lead to client miss
+ teardown response since it expect to get the last part of message.
+ The flushing during teardown was introduced to fix a deadlock that now
+ is fixed more generally in handle_request by temporary setting backlog
+ size to unlimited.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
+
+2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: Use AM_TESTS_ENVIRONMENT
+ Needed by the new automake test runner and the
+ current version of the common submodule.
+
+2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: Use single-include rtsp header to make sure we get all definitions
+
+2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Mark some more functions static
+
+2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only unblock the media in suspend() when actually changing the state
+ Otherwise we're going to lose a few packets for live streams during DESCRIBE.
+
+2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-video-rtx.c:
+ examples: Use AVPF profile for the RTX example
+
+2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Only add RTX to the SDP when using a feedback profile
+
+2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: get valid clock-rate from last-sample
+ clock-rate in last-sample's caps is integer, not unsigned.
+ To get this value properly, variable needs to be type-casted to int.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747614
+
+2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ autogen.sh: only run autopoint if gettext requested in configure.ac
+ Not just because there happens to be a po directory.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ Revert "configure.ac: uncomment gettext version setup"
+ This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
+ We don't need a gettext setup here and there's no po
+ directory either, so no reason why autopoint would be
+ run in the first place.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
+
+ * examples/test-multicast.c:
+ * examples/test-multicast2.c:
+ * examples/test-sdp.c:
+ * examples/test-video-rtx.c:
+ * examples/test-video.c:
+ * tests/test-cleanup.c:
+ * tests/test-reuse.c:
+ Fix timeout function signatures across tests and examples
+
+2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
+ Make sure the test environment is set up.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: bump automake requirement to 1.14 and autoconf to 2.69
+ This is only required for builds from git, people can still
+ build tarballs if they only have older autotools.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * configure.ac:
+ configure.ac: uncomment gettext version setup
+ Fixes autogen.sh. It would run autopoint, which would complain
+ that it could not find the gettext version in configure.ac.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * examples/test-video-rtx.c:
+ test-video-rtx: set exact payload type to PCMA payloader
+ Setting wrong payload type causes failure to do retransmission through audio stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=747839
+
+2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: fix to get valid each stream data for request-aux-sender signal
+ Because of duplicated g_signal_connect for request-aux-sender signal,
+ wrong stream pointer is passed to the signal handler.
+ Instead of passing each stream, pass stream array and get the relevant stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747839
+
+2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * acinclude.m4:
+ * autogen.sh:
+ Update autogen.sh to latest version from common
+ Fixes build after aclocal_check etc. helpers have been removed.
+
+2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From bc76a8b to c8fb372
+
+2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Limit the queues to 1 buffer
+ We only need them to be able to pre-roll, queueing up more data here
+ is only going to harm latency and memory usage.
+
+2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Update comment and ASCII art to the latest code
+ We have a queue in front of the udpsink too to prevent the pipeline from
+ locking up.
+
+2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-media: Properly return first rtptime
+ Instead we where returning first GstBuffer timestamp. This would result
+ in clock skew and unwanted behaviour in RTSP playback.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746479
+
+2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't leave buffer mapped
+ If the seq is NULL, the RTP buffer was left mapped. We should always
+ unmap the buffer.
+
+2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
+
+ * README:
+ Fix typo in README
+
+2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * tests/check/gst/client.c:
+ Fix double semicolons
+
+2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
+ This gives more accurate values than asking the payloader. There might be
+ queueing happening between the payloader and the sink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745704
+
+2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't seek for PLAY if the position will not change
+ https://bugzilla.gnome.org/show_bug.cgi?id=745704
+
+2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't include payload type in the caps for framesize
+ When the sdp media attribute framesize are converted to caps
+ the <payload> should not be included.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
+ Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
+
+2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: add payload type to the sdp framesize attribute
+ The sdp framesize attribute is desribed in RFC6064. It is specified
+ for payloading of H263 and has the following form
+ a=framesize:<payload type> <width>-<height>. The <width>-<height> part
+ should be added to the caps in a payloader and the <payload type> should
+ be added by the rtsp-server.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
+
+2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: fix tainted variable
+ Insignificant but this keeps Coverity happy.
+ CID #1268404
+
+2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-netclock-client.c:
+ * examples/test-netclock.c:
+ examples: Add a simple example of network synch for live streams.
+ An example server and client that works for synchronising live streams
+ only - as it can't support pause/play.
+
+2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ rtsp-media-factory: Add functions to set/get the media gtype
+ Allow specifying the GType of a GstRtspMedia subclass to create
+ as a simpler way to get the factory to create a custom
+ GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
+
+2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix double unlock in _get_buffer_size()
+ Fixes an abort when calling gst_rtsp_media_get_buffer_size()
+ because of double g_mutex_unlock () usage.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745434
+
+2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-session: Use monotonic time for RTSP session timeout
+ Changed RTSP session timeout handling to monotonic time
+ and deprecating the API for current system time.
+ This fixes timeouts when the system time changes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743346
+
+2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-client: Only error out in PLAY if seeking actually failed
+ If the media was just not seekable, we continue from whatever position we are
+ and let the client decide if that is what is wanted or not.
+ Only if the actual seek failed, we can't really recover and should error out.
+
+2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Add necessary queues between tee and multiudpsink
+ https://bugzilla.gnome.org/show_bug.cgi?id=744379
+
+2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: If seeking fails, don't wait forever for the media to preroll again
+ Instead error out properly the same way as if the SEEKING query already
+ failed.
+
+2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: minor code formatting fix
+
+2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix logic for collect_streams
+ Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
+ all streams it knows if it got any, and can check if the transport mode is OK.
+ CID #1268400
+
+2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't set the transport mode based on what elements we find
+ Just print a warning if the one that was set before disagrees with what
+ elements we found. It must already be set to something before as this
+ function is called after we received the SDP from ANNOUNCE in RECORD mode,
+ and we would reject ANNOUNCE if the RECORD flag was not set.
+
+2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: rtspserver: rename shadowed variable
+ We have two different 'sink' variables here,
+ rename one of them for clarity.
+
+2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix awkward if clause
+
+2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: improve uri argument handling and accept file names
+ Print an error if the argument passed is not a URI and can't
+ be converted into one, or no arguments have been provided.
+
+2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: don't remove mount point after 10 seconds
+ It's very irritating when trying to test stuff repeatedly
+ and serves no real purpose other than showing that it can
+ be done.
+
+2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/.gitignore:
+ examples: add new test-record to .gitignore
+
+2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * tests/check/gst/rtspserver.c:
+ rtsp-media: Use flags to distinguish between PLAY and RECORD media
+
+2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Set latency for playback-style example to 2s instead of 200ms
+
+2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: add some unit tests for ANNOUNCE and RECORD
+ https://bugzilla.gnome.org/show_bug.cgi?id=743175
+
+2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix a couple of leaks in handle_announce
+
+2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Expose latency setting for setting the rtpbin latency
+
+2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
+
+2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
+
+2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ * examples/test-record.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ Add initial support for RECORD
+ We currently only support media that is RECORD or PLAY only, not both at once.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743175
+
+2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: RTCP and RTP transport cache cookies seperated
+ RTCP packets were not sent because the same tr_cache_cookie was used for
+ both RTP and RTCP. So only one of the tr_cache lists were populated
+ depending on which one was sent first. If the tr_cache list is not
+ populated then no packets can be sent. Most often this happened to be
+ RTCP. Now seperate RTCP and RTP transport cache cookies are added which
+ resulted in both the tr_cache_lists to be populated regardless of which
+ one was sent first.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
+
+2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fix false compiler warning
+ rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
+
+2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: log interleaved data received
+
+2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
+
+2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
+
+2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Use a random session ID in the SDP
+ RFC4566 Section 5.2 says that it should make the username, session id,
+ nettype, addrtype and unicast address tuple globally unique. Always using
+ 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
+ Instead let's create a 64 bit random number, which at least brings us
+ closer to the goal of global uniqueness.
+ https://tools.ietf.org/html/rfc4566#section-5.2
+
+2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-uri.c:
+ examples: Don't call gst_init() and gst_get_option_group()
+ The latter calls the former at the appropriate time.
+
+2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Drop trailing \0 of RTSP DATA messages
+ We add a trailing \0 in GstRTSPConnection to make parsing of
+ string message bodies easier (e.g. the SDP from DESCRIBE) but
+ for actual data this means we have to drop it or otherwise
+ create invalid data.
+
+2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
+ Fixes crash when two threads access handle_new_sample() at the same
+ time, one for RTP, one for RTCP.
+ Otherwise, when iterating over the transports cache, it might be modified by
+ another thread at the same time if the transports cookie has changed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742954
+
+2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set format=TIME on our app sources for TCP
+
+2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
+ This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
+ RFC 2326 states that session IDs may consist of alphanumeric as well as
+ the safe characters $-_.+ -- N.B. the percent character is not allowed.
+ Previously the session ID was URI-escaped, this meant that any character
+ which was not alphanumeric or any of the characters +-._~ would be
+ percent encoded. While the RFC (surprisingly) mentions that linear white
+ space in session IDs should be URI-escaped, it does not say anything
+ about other characters. Moreover no white space is allowed in the
+ session ID. Finally the percent character which is the result of
+ URI-escaping is not allowed in a session ID.
+ So there is no reason to do any URI-escaping, and now it is removed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742869
+
+2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f2c6b95 to bc76a8b
+
+2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Fix 'make check' from top-level directory
+
+2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-uri.c:
+ examples: Add command-line parsing and take a 'port' argument
+ This allows users to run multiple servers on different ports for testing.
+ Only done for examples that actually take arguments and hence are capable of
+ outputting different streams for each instance on each port.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742115
+
+2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: Add a send_message default signal handler
+ This allows subclasses to easily hook into the response sending
+ mechanism without doing everything from a signal, which seems
+ awkward from subclasses.
+
+2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From ef1ffdc to f2c6b95
+
+2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * Makefile.am:
+ * configure.ac:
+ configure: add --disable-examples switch
+ https://bugzilla.gnome.org/show_bug.cgi?id=741678
+
+2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-video-rtx.c:
+ examples: add a retransmisison example implementing RFC4588
+ Currently only SSRC-multiplexed rtx streams are supported
+
+2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix some minor memory leaks
+
+2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Some minor cleanup
+
+2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix compiler warnings
+ rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ ^
+ rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ ^
+
+2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ media: implement ssrc-multiplexed retransmission support
+ based off RFC 4588 and the server-rtpaux example in -good
+
+2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp: Ref transports in hash table.
+ Also ref streams for transports.
+ This solves a crash when reciving a rtcp after teardown but before
+ client finalize.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
+
+2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From 7bb2bce to ef1ffdc
+
+2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: refactor cleanup of cached media
+
+2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/client.c:
+ tests: Remove FIXME
+ The session leak is now fixed, lets remove those FIXME comments.
+
+2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test to setup two sessions on one connection
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test setup with tcp transport
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Configure transport after creating session media
+ The default implementation of configure_client_transport() in
+ rtsp-client uses the session media when it chooses channels for
+ interleaved traffic.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ client: Stop caching media in client when doing setup
+ If the media has been managed by a session media, it should not be
+ cached in the client any longer. The GstRTSPSessionMedia object is now
+ responsible for unpreparing the GstRTSPMedia object using
+ gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
+ session media.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: unref srtp decoder when leaving bin
+ https://bugzilla.gnome.org/show_bug.cgi?id=739481
+
+2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: mikey memory leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=739383
+
+2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 84d06cd to 7bb2bce
+
+2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Parallelise 'make check-valgrind'
+
+2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From a8c8939 to 84d06cd
+
+2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 36388a1 to a8c8939
+
+2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: deactivate media when shutting down from paused
+ This was only done when going directly from playing.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
+
+2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-context.h:
+ rtsp-client: add stream transport to context
+ We add the stream transport to the context so we can get the configured
+ client stream transport in the setup request signal.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
+
+2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: release lock even not all transports have been removed
+ We don't want to keep the lock even we return FALSE because not all the
+ transports have been removed. This could lead into a deadlock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737797
+
+2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
+ These were renamed in GstRTPBasePayload in 1.0
+
+2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: set session media to NULL without the lock
+ We need to set session medias to NULL without the client lock otherwise
+ we can end up in a deadlock if another thread is waiting for the lock
+ and media unprepare is also waiting for that thread to end.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737690
+
+2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Set state to UNPREPARING in all cases
+
+2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: set state to unpreparing when unprepare is initiated
+ https://bugzilla.gnome.org/show_bug.cgi?id=737675
+
+2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Remove backlog limit while processings requests
+ If the backlog limit is kept two cases of deadlocks may be
+ encountered when streaming over TCP. Without the backlog
+ limit this deadlocks can not happen, at the expence of
+ memory usage.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
+
+2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: do not free main context before rtsp watch
+ https://bugzilla.gnome.org/show_bug.cgi?id=737110
+
+2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Extend unit test timeout to accomodate for valgrind
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ rtsp-*: Treat sending packets to clients as keepalive
+ As long as gst-rtsp-server can successfully send RTP/RTCP data to
+ clients then the client must be reading. This change makes the server
+ timeout the connection if the client stops reading.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Allow backlog to grow while expiring session
+ Allow the send backlog in the RTSP watch to grow to unlimited size while
+ attempting to bring the media pipeline to NULL due to a session
+ expiring. Without this change the appsink element cannot change state
+ because it is blocked while rendering data in the new_sample callback.
+ This callback will block until it has successfully put the data into the
+ send backlog. There is a chance that the send backlog is full at this
+ point which means that the callback may block for a long time, possibly
+ forever. Therefore the media pipeline may also be prevented from
+ changing state for a long time.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Make old compilers happy
+ rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
+ Just in case that guint8 doesn't fit in a pointer. Just in case ...
+
+2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: raise the backlog limits before pausing
+ We need to raise the backlog limits before pausing the pipeline or else
+ the appsink might be blocking in the render method in wait_backlog() and
+ we would deadlock waiting for paused.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
+
+2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: make define for the WATCH_BACKLOG
+ See https://bugzilla.gnome.org/show_bug.cgi?id=736322
+
+2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: simplify session transport handling
+ link/unlink of the transport in a session was done to keep track of all
+ TCP transports and to send RTP/RTCP data to the streams. We can simplify
+ that by putting all the TCP transports in a hashtable indexed with the
+ channel number.
+ We also don't need to link/unlink the transports when we pause/resume
+ the streams. The same effect is already achieved when we pause/play the
+ media. Indeed, when we pause the media, the transport is removed from
+ the media and the callbacks will not be called anymore.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=736041
+
+2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ stream-transport: make method to handle received data
+ Make a method to handle the data received on a channel. It sends the
+ data to the stream of the transport on the RTP or RTCP pads based on
+ the channel number.
+
+2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-mp4.c:
+ test: add example of dumping RTCP reports
+
+2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Make sure that sequence numbers are monotonic after pause
+ The sequence number is not monotonic for RTP packets after pause. The
+ reason is basepayloader generates a randon sequence number when the
+ pipeline goes from ready to pause. With this fix generation of sequence
+ number will be monotonic when going from pause to play request.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736017
+
+2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Protect saved clients watch with a mutex
+ Fixes a crash when close() is called while merging clients
+ in handle_tunnel(). In that case close() would destroy the
+ watch while it is still being used in handle_tunnel().
+ https://bugzilla.gnome.org/show_bug.cgi?id=735570
+
+2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Remove the multicast group udp sources when removing from the bin
+
+2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Query position and stop time only on the RTP parts of the pipeline
+ The RTCP parts, in specific the RTCP udpsinks, are not flushed when
+ seeking and will always continue counting the time. This leads to
+ the NPT after a backwards seek to be something completely different
+ to the actual seek position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732644
+
+2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-appsrc.c:
+ examples: fix another reference leak
+ gst_rtsp_media_get_element() returns a new ref.
+
+2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * examples/test-appsrc.c:
+ examples: unref element after usage
+ gst_bin_get_by_name_recurse_up() returns an element
+ reference that must be unreffed after usage.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734546
+
+2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
+
+ * gst/rtsp-server/rtsp-media.c:
+ signals: Fix copy-pasto in target-state signal offset
+
+2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
+
+ * Makefile.am:
+ * common:
+ Makefile: Add usage of build-checks step
+ Allows building checks without running them
+
+2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Listen on the multicast group for RTP/RTCP packets
+ When a UDP multicast transport is used it is expected that the server listens
+ for RTP and RTCP packets on the multicast group with the corresponding port.
+ Without this we will never get RTCP packets from clients in multicast mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732238
+
+2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.4.0 ===
+
+2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.4.0
+
+2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ media: correct misspelled words in description
+ https://bugzilla.gnome.org/show_bug.cgi?id=733244
+
+=== release 1.3.91 ===
+
+2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.91
+
+2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: update docs
+
+2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: implement client REMOVE filter
+
+2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: expose _close() method
+ Expose a previously internal close method to close the client
+ connection.
+
+2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session-pool: signal session-removed outside of the lock
+ Release the lock before emiting the session-removed signal.
+
+2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ filter: Release lock in filter functions
+ Release the object lock before calling the filter functions. We need to
+ keep a cookie to detect when the list changed during the filter
+ callback. We also keep a hashtable to make sure we only call the filter
+ function once for each object in case of concurrent modification.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
+
+2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: check if watch is set in handle_teardown()
+ The unit tests run without a watch
+
+2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/client.c:
+ client tests: send teardown to cleanup session
+
+2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ server tests: send teardown to cleanup session
+
+2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: keep ref to client for the session removed handler
+ This extra ref will be dropped when all client sessions have been
+ removed. A session is removed when a client sends teardown, closes its
+ endpoint of the TCP connection or the sessions expires.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * tests/check/gst/client.c:
+ client: manage media in session as a last step
+ Once we manage a media in a session, we can't unmanage it anymore
+ without destroying it. Therefore, first check everything before we
+ manage the media, otherwise if something is wrong we have no way to
+ unmanage the media.
+ If we created a new session and something went wrong, remove the session
+ again. Fixes a leak in the unit test.
+
+2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ examples: print 'stream ready at url' for mp4 and ogg example
+
+2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp: fix for MIKEY api change
+
+2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: free watch context only once
+ The watch context is freed when the source is destroyed. Avoids
+ a CRITICAL when we try to unref the context twice.
+
+2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix build
+
+2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: protect sessions with lock
+ Protect the list of sessions with the lock.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Client: keep a ref to the session
+ Don't just keep a weak ref to the session objects but use a hard ref. We
+ will be notified when a session is removed from the pool (expired) with
+ the new session-removed signal.
+ Don't automatically close the RTSP connection when all the sessions of
+ a client are removed, a client can continue to operate and it can create
+ a new session if it wants. If you want to remove the client from the
+ server, you have to use gst_rtsp_server_client_filter() now.
+ Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ session-pool: add session-removed signal
+ Add a signal to be notified when a session is removed from the pool.
+
+2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-server.h:
+ Make rtsp-server.h a single-include header, use it for G-I
+ https://bugzilla.gnome.org/show_bug.cgi?id=732411
+
+=== release 1.3.90 ===
+
+2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.90
+
+2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: crypto can be NULL
+
+2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ introspection: add missing allow-none annotations
+ https://bugzilla.gnome.org/show_bug.cgi?id=730952
+
+2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-token.c:
+ introspection: add (nullable) annotations to return values
+ https://bugzilla.gnome.org/show_bug.cgi?id=730952
+
+2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ gi: improve annotations
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
+
+2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ signals: use generic marshal function
+ Use the generic C marshal function.
+ Use more explicit type instead of G_TYPE_POINTER
+
+2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-context.h:
+ context: add type macro
+
+2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: hide key length defines
+ They don't have a namespace.
+
+2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.3.3 ===
+
+2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.3
+
+2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ mikey: add different key length parameters
+ Add encryption and authentication key length parameters to MIKEY. For
+ the encoders, the key lengths are obtained from the cipher and auth
+ algorithms set in the caps. For the decoders, they are obtained while
+ parsing the key management from the client.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
+
+2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
+
+ * tests/check/gst/stream.c:
+ stream tests: Make sure we get right multicast address from stream
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
+
+2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: ref the context until rtsp watch is alive
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
+
+2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Destroy the rtsp watch after connection close
+
+2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix confusing comment
+
+2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: Timeout in header.
+ Adding the possbilty to always have timout in header.
+ This is configurabe with setting "timeout-always-visible".
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
+
+2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.3.2 ===
+
+2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * common:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.2
+
+2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 211fa5f to 1f5d3c3
+
+2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: store TCP ports in transport
+ Store the TCP ports in the transport when we are doing RTSP over TCP.
+ This way, we can easily get to the ports from the transport.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
+
+2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: add signals for new RTP/RTCP encoders
+ New signals to allow the user to configure the dynamically created
+ encoders.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730228
+
+2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: Make suspend()/unsuspend() virtual
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
+
+2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix send-message signal marshaller
+ Use generic marshalling for the send-message signal. It has
+ two POINTER arguments, not just one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=729900
+
+2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/gst/media.c:
+ tests: add and remove pads only once
+ In this test we simulate a dynamic pad by watching the caps event.
+ Because of renegotiation in the base payloader now, this caps is sent
+ multiple times but we can only deal with 1 invocation, use a variable to
+ only 'add and remove' the pad once.
+
+2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: add unit test for correct handling of Require headers
+ https://bugzilla.gnome.org/show_bug.cgi?id=729426
+
+2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
+ Servers must handle Require headers and must report a failure
+ if they don't handle any of the Required options, see RFC 2326,
+ section 12.32: https://tools.ietf.org/html/rfc2326#page-54
+ https://bugzilla.gnome.org/show_bug.cgi?id=729426
+
+2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.3.1 ===
+
+2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.1
+
+2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From bcb1518 to 211fa5f
+
+2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ Update .gitignore
+
+2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/sessionmedia.c:
+ tests: fix memory leak in sessionmedia unit test
+
+2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: emit a signal before sending a message
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
+
+2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: pass context to send_message
+ Pass the current context to send_message, we will need it later.
+
+2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix typo in comment
+
+2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Do not stop thread twice if default_prepare() fails
+
+2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: set the watch to flushing before going to NULL
+ First set the watch to flushing so that we unblock any current and
+ future attempt to send data on the watch, Then set the pipeline to
+ NULL.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
+
+2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * tests/check/gst/sessionpool.c:
+ rtsp-session-pool: Fixes annotation
+ Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
+ in the sessionpool test.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
+
+2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: make media_prepare virtual
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
+
+2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: stop the thread in more error cases
+
+2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: allow NULL as the thread
+ Use the default context whan passing a NULL thread.
+
+2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: indent cleanup
+ Coverity was moaning about unreachable code, and I think it was just
+ confused by { being before the label. We'll see if it pops up again.
+ Coverity 1197705
+
+2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ client: Add drop-backlog property
+ When we have too many messages queued for a client (currently hardcoded
+ to 100) we overflow and drop the messages. Add a drop-backlog property
+ to control this behaviour. Setting this property to FALSE will retry
+ to send the messages to the client by waiting for more room in the
+ backlog.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
+
+2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: support for POST before GET when setting up a tunnel
+
+2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: remove watch of the second client after http tunnel setup
+ The second client will be freed after the HTTP tunnel has been set up.
+ Make sure it's RTSP watch is never dispatched again.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
+
+2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: Make media_prepare() fail if port allocation fails
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
+
+2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/media.c:
+ media test: cleanup the thread pool in tests
+
+2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: Unblock blocked streams in unprepare
+ The streams will be blocked when a live media is prepared.
+ The streams should be unblocked in gst_rtsp_media_unprepare.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
+
+2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: release the state lock when going to NULL
+ Set our state to UNPREPARING and release the state-lock before
+ setting the pipeline to the NULL state. This way, any pad-added
+ callback will be able to take the state-lock and check that we are now
+ unpreparing instead of deadlocking.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
+
+2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: protect status with lock
+ Make sure we only update the status with the lock.
+
+2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp: update for MIKEY API changes
+
+2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: parse the mikey response from the client
+ Parse the mikey response from the client and update the policy for
+ each SSRC.
+
+2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to set crypto info
+ Make a method to configure the crypto information of a stream.
+ Set udpsrc in READY instead of PAUSED so that we can configure caps
+ later.
+
+2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: cleanup error paths
+
+2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix docs
+
+2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-video.c:
+ test: enable SRTP only on RTSPS
+ We only want to enable SRTP when doing rtsp over TLS so that we can
+ exchange the keys in a secure way.
+
+2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-video.c:
+ test: print an error on failure
+
+2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * configure.ac:
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/Makefile.am:
+ stream: add SRTP support
+ Install srtp encoder and decoder elements in rtpbin
+ Add MIKEY in SDP
+
+2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/sessionpool.c:
+ tests: Add unit tests for sessionpool
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
+
+2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/threadpool.c:
+ tests: Improve code coverage of rtsp-threadpool tests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
+
+2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/sessionmedia.c:
+ tests: Improve code coverage for rtsp-session-media
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
+
+2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ gobject-introspection: Add annotations to support language bindings
+ In addition a few cosmetic changes:
+ * Adjust the order of arguments
+ * Fix typo: occured -> occurred
+ * Fix indentation after Return:-clauses
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
+
+2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't mix IPv4 and IPv6 addresses
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
+
+2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: take caps after the session manager
+ Take the caps for the SDP after they leave the rtpbin so that we can
+ also get the properties added by rtpbin elements.
+
+2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: release lock while pushing out packets
+ Keep a cache of the transports and use this to iterate the transport
+ while pushing packets. This allows us to release the lock early.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=725898
+
+2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: vmethod for modifying tunnel GET response
+ Add a vmethod tunnel_http_response where the response to the HTTP GET
+ for tunneled connections can be modified.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
+
+2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: make 1 media line per profile
+ If we have multiple profiles (AVP or AVPF) for a stream, make one m=
+ line in the SDP for each profile. The client is then supposed to pick
+ one of the profiles in the SETUP request. Because the m= lines have the
+ same pt, the client also knows that only 1 option is possible.
+
+2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ factory: add profile property and pass to media and streams
+
+2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-multicast.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: pass multicast connection for multicast-only stream
+ Pass the multicast address of the stream in the connection info in the
+ SDP so that clients try a multicast connection first.
+ Only allow multicast connections in the test-multicast example. Also
+ increase the TTL a little.
+
+2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * .gitignore:
+ .gitignore: Ignore gcov intermediate files
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
+
+2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: release some locks in error cases
+
+2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ docs: Enable and fix gtk-doc warnings
+ * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
+ * addresspool/mediafactory: Add missing annotation colon
+ * stream: Annotate return value
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
+
+2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From fe1672e to bcb1518
+
+2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 1a07da9 to fe1672e
+
+2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/Makefile.am:
+ examples: use LDADD for libs instead of LDFLAGS
+
+2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: make sure releases are in .doap file
+
+2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-cgroups.c:
+ examples: test-cgroups: don't put code with side effects into g_assert()
+ The g_assert() might get compiled out with the right
+ compiler/preprocessor flags.
+
+2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/.gitignore:
+ examples: add cgroup test binary to .gitignore
+
+2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-cgroups.c:
+ examples: fix cgroup test build
+ Fixes build failure caused by compiler warning:
+ test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
+
+2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ .gitignore: ignore temp files created in the course of 'make check'
+
+2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: don't loose frames handling new PLAY request
+ If client supplied a range check if the range specifies the start point.
+ If not, then do an accurate seek to the current position. If a start
+ point was specified do do a key unit seek to make sure the streaming
+ starts with decodeable frames.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
+
+2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Revert "media: only flush when setting a new start position"
+ This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
+ We need to do the flush in all cases, demuxer block currently for
+ non-flushing seeks.
+
+2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: only flush when setting a new start position
+ Only flush the pipeline when we change the start position with
+ a seek.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=724611
+
+2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set ttl-mc before adding the socket
+ Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
+ never be set on socket.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
+
+2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: stop thread if media is already prepared
+ in gst_rtsp_media_prepare() the thread is not used if media is already
+ prepared (e.g. media shared) so we want to stop the thread. otherwise, a
+ leak occurs.
+ https://bugzilla.gnome.org/show_bug.cgi?id=724182
+
+2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * Makefile.am:
+ build: Ship gst-rtsp-server.doap file
+
+2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Fix another compiler warning with gcc
+
+2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ rtsp-server: Fix lots of compiler warnings with clang
+
+2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * tests/Makefile.am:
+ configure: Synchronise with the configure scripts of the other modules
+
+2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
+
+2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ Revert "rtsp-server: support build against last stable release"
+ This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
+ Let us require 1.2.3 now, which is going to be released in a few
+ minutes.
+
+2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ session: improve RTP-Info
+ Ignore streams that can't generate RTP-Info instead of failing.
+ Don't return the empty string when all streams are unconfigured but
+ return NULL so that we don't generate and empty RTP-Info header.
+ Improve docs a little.
+
+2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ Don't free rtpinfo GString when it is NULL
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
+
+2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: only set keyframe flag when modifying start
+ Only set the keyframe flag when we modify the start position. The
+ keyframe flag should probably be ignored when no change is requested but
+ until we can claim this is all documented properly and all demuxer
+ implement this, avoid setting the flag.
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
+
+2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: Unref source after mainloop has quit to avoid races in GLib
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
+
+2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: handle NULL seqnum and rtptime arguments
+
+2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * tests/check/gst/threadpool.c:
+ thread-pool: Unref reused threads in gst_rtsp_thread_stop()
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
+
+2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: add fallback for missing stats property
+ Use a fallback when the payloader does not have a stats property
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
+
+2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From f7bc1c3 to 1a07da9
+
+2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: don't leak stats structure
+ Don't leak the stats structure and deal with NULL stats.
+
+2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Get rtpinfo properties atomically from payloader
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
+
+2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: refactor state change functions and signals
+ Make functions to set the target state and the pipeline state and emit
+ the signals from those functions.
+
+2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal to notify of pending state changes
+
+2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: support build against last stable release
+ Until 1.2.3 is out with the new get_type function and we
+ can require that.
+
+2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: fix compilation
+
+2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add property to configure profiles
+
+2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: let stream check supported transport
+ Delegate the check if a transport is allowed to the stream.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=720696
+
+2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to check supported transport
+ Add a method to check if a transport is supported
+
+2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure.ac: Only check for gstreamer-check, not check
+ We include check in gstreamer-check since quite some time now.
+
+2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: return clock-rate from get_rtpinfo
+ And use it to correct the rtptime to the requested start-time.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=712198
+
+2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ session-media: calculate start-time
+
+2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: also return the running-time
+ Return the running-time in the rtpinfo as well.
+
+2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ session-media: let the session-media make the RTPInfo
+ Add method to create the RTPInfo for a stream-transport.
+ Add method to create the RTPInfo for all stream-transports in a
+ session-media.
+ Use the session-media RTPInfo code in client. This allows us to refactor
+ another method to link the TCP callbacks.
+
+2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ mount-points: sort sequence before g_sequence_lookup
+ * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
+ sort sequence if dirty, otherwise lookup will fail.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
+
+2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: rename package from gst-rtsp to gst-rtsp-server
+ To match git module name and avoid confusion with the
+ rtsp lib in gst-plugins-base and rtsp plugin in -good.
+
+2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: bump core/base/good requirement to 1.2.0
+ Bump to released stable version and make implicit
+ requirements explicit.
+
+2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ * configure.ac:
+ Fix broken gettext setup which is not used anyway
+
+2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From dbedaa0 to d48bed3
+
+2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add setup_sdp vmethod
+ gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
+ gst_rtsp_media_setup_sdp.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
+
+2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Check return value of sscanf
+ streamid is only valid if sscanf matched something.
+
+2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix iteration
+ Wouldn't even enter the code block otherwise (i++ was used as the check
+ and not the postfix).
+
+2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add vmethod to configure media and streams
+ Implement a vmethod that can be used to configure the media and the
+ streams based on the current context. Handle the blocksize handling in
+ the default handler.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=720667
+
+2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ Make git ignore more unit test binaries
+
+2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * gst/rtsp-server/rtsp-token.h:
+ rtsp-server: add padding to many public structures
+ Not mini objects though, since they are not subclassable
+ anyway, nor kept on the stack or inlined in a structure.
+
+2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ media: add new create_rtpbin vmethod
+ * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
+ https://bugzilla.gnome.org/show_bug.cgi?id=719734
+
+2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
+
+ * tests/check/gst/media.c:
+ tests: fix memory leak, free test's thread pool
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
+
+2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ stream-transport: free url in finalize
+
+2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: also do state change in suspended state
+
+2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ media: also handle prepare and range in suspended state
+ When we are suspended, we are already prepared.
+ We can get the range in the suspended state.
+
+2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/sessionmedia.c:
+ check: add test for uri in setup
+ Added unit tests for the new functionality in GstRTSPStreamTransport.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
+
+2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: store setup uri and use in PLAY response
+ Store the uri used when doing the setup and use that in the PLAY
+ response.
+ fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
+
+2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ stream-transport: add method to get/set url
+
+2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: suspend after SDP and unsuspend before PLAYING
+ Based on patches by Ognyan Tonchev <ognyan@axis.com>
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * tests/check/gst/media.c:
+ * tests/check/gst/mediafactory.c:
+ media: add suspend modes
+ Add support for different suspend modes. The stream is suspended right after
+ producing the SDP and after PAUSE. Different suspend modes are available that
+ affect the state of the pipeline. NONE leaves the pipeline state unchanged and
+ is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
+ state and RESET will bring the pipeline to the NULL state.
+ A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
+ this means that the pipeline needs to be prerolled again.
+ Base on patches by Ognyan Tonchev <ognyan@axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: start live streams in blocked state
+ Start live streams in the blocked state and make them preroll using the
+ messages. This ensure that no data is played by the sink until we explicitly
+ unblock the stream right before going to PLAYING.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: refactor starting and waiting for preroll
+ Based on patches from Ognyan Tonchev <ognyan@axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add API to block streams
+ Add an API to block on the streams and make it post a message.
+ Based on patch by Ognyan Tonchev <ognyan@axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
+
+ * docs/libs/Makefile.am:
+ docs: Specify the override file
+ Even if it's empty (for now) it avoids make distcheck complaining
+
+2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: move default implementations to where they are used
+
+2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: take the right lock in gst_rtsp_media_set_pipeline_state()
+ We need to take the state_lock when calling this method.
+
+2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: handle add-added on non-bins too
+ Handle dynamic payloaders that are not bins, as used in the unit-test.
+
+2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media/-factory: Fix request pad name comments
+ These must be escaped for gtk-doc to parse the comments without warnings.
+
+2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ rtsp-media: remove transports if media is in error status
+ * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
+ trying to change to GST_STATE_NULL and media is in error status, we
+ remove all transports.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
+
+2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: use element metadata to find payloader
+ Use the element metadata to find the payloader instead of checking
+ for the base class.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
+
+2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ rtsp-stream: add getter for payload type
+ * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
+ * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
+ element and create the stream with this one instead of the dynpay%d
+ element.
+ https://bugzilla.gnome.org/show_bug.cgi?id=712396
+
+2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-token.c:
+ rtsp-*: Refer to NULL as a constant in comments
+ Plus one typo fix.
+ https://bugzilla.gnome.org/show_bug.cgi?id=714988
+
+2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ rtsp-*: Fix type name typos in comments
+ * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
+ * rtsp-auth: Refer to part of constant name as text
+ * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
+ * rtsp-session-media: Fix GstRTSPSessionMedia typo
+ * rtsp-stream: Fix typo when refering to GstBin
+ https://bugzilla.gnome.org/show_bug.cgi?id=714988
+
+2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * docs/README:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: Improve documentation
+ * Include annotation-glossary to quiet gtk-doc
+ * Rename remaining ClientState -> Context
+ * Rename object hierarchy file
+ * Remove stale chapter references
+ * Add missing function and object references
+ * Include missing GstRTSPAddressPoolResult
+ https://bugzilla.gnome.org/show_bug.cgi?id=714988
+
+2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: sprinkle some allow-none annotations for g-i
+
+2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to filter transports
+ Add a method to safely iterate and collect the stream transports
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
+
+2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp: allow NULL func in filters
+ Passing a null function make the filters return a list of
+ refcounted objects.
+
+2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * tests/check/gst/addresspool.c:
+ address-pool: fix address increment
+ Use a guint instead of guint8 to increment the address. It's still not
+ completely correct because a guint might not be able to hold the complete
+ address range, but that's an enhacement for later.
+ Add unit test to test improved behaviour.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708237
+
+2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ client: allow absolute path in requests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
+
+2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: make make_path_from_uri a vmethod
+
+2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/Makefile.am:
+ * tests/check/gst/stream.c:
+ stream: Add functions to get rtp and rtcp sockets
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
+
+2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-context.h:
+ context: defing a GType for the context
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
+
+2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ Fixed several GIR warnings
+
+2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: small typos
+
+2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/token.c:
+ tests: Add unit tests for token
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
+
+2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-token.c:
+ token: Validate args for gst_rtsp_token_is_allowed
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710520
+
+2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-token.c:
+ token: Fix bug when creating empty token
+ We always want to have a valid GstStructure in the token.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
+
+2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: avoid race in shutdown
+ If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
+ don't actually stop the mainloop ever. Solve this race by adding an idle source
+ to the mainloop that calls the _quit. This way we immediately exit the mainloop
+ if quit was called before we started it.
+
+2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/permissions.c:
+ tests: Add unit tests for permissions
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/mediafactory.c:
+ tests: Test mediafactory permissions
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: Fix refcounting when adding/removing roles
+ Previously a role that was removed was unreffed twice, and when
+ replacing an existing role the replaced role was freed while still being
+ referenced. Both bugs are now fixed.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/media.c:
+ * tests/check/gst/mediafactory.c:
+ * tests/check/gst/rtspserver.c:
+ tests: Check gst_rtsp_url_parse return value
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 865aa20 to dbedaa0
+
+2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: Fix socket leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=710088
+
+2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-session-pool: Make sure session IDs are properly URI-escaped
+ https://bugzilla.gnome.org/show_bug.cgi?id=643812
+
+2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * examples/.gitignore:
+ * examples/test-video.c:
+ examples: fix compilation when WITH_AUTH is defined
+ https://bugzilla.gnome.org/show_bug.cgi?id=710228
+
+2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * .gitignore:
+ gitignore: Add new test binary
+
+2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/threadpool.c:
+ thread-pool: Add unit test for the thread pools
+ https://bugzilla.gnome.org/show_bug.cgi?id=710228
+
+2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: Fix thread leak when reusing threads
+ https://bugzilla.gnome.org/show_bug.cgi?id=709730
+
+2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * tests/check/gst/rtspserver.c:
+ tests: fixed racy behavior in rtspserver tests
+ https://bugzilla.gnome.org/show_bug.cgi?id=710078
+
+2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/addresspool.c:
+ tests: Improve address pool unit tests
+ Add a range with mixed IPV4 and IPV6 addresses to pool.
+ Get an IPV4 address from an IPV6-only pool.
+ Get an IPV6 address from an IPV4-only pool.
+ Reserve a IPV6 address from an IPV4-only pool.
+ Check for unicast addresses in multicast-only pool.
+ Check for unicast addresses in uni-/multicast-mixed pool.
+ https://bugzilla.gnome.org/show_bug.cgi?id=710128
+
+2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: append query string in PAUSE/PLAY/TEARDOWN as well
+
+2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Add query to control path
+ If the SETUP url contains a query it must be appended to the control
+ path so that it matches any already created stream in the media. The
+ query will also be appended to the session media path.
+
+2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: remove old line
+
+2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Correct control comparison
+ https://bugzilla.gnome.org/show_bug.cgi?id=709176
+
+2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Check dynamically if the pipeline supports seeking
+ We should not depend on whether or not the pipeline state change
+ returned NO_PREROLL or not. A media could dynamically change its
+ element and switch from seekable to non seekable so it's best to test
+ the seekable nature of the pipeline dynamically when we try to do a seek.
+
+2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Return FALSE if seeking is not supported
+
+2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: don't seek accurate by default
+ Accurate seeking is perhaps a little overkill in the most common situation and
+ causes some formats (mp3) over slow media to seek extremely slowly.
+
+2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: fix unit test
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
+
+2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Reply 400 if media cannot be constructed
+ Reply 400 Bad Request instead of 503 Service Unavailable if media
+ cannot be constructed in SETUP.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
+
+2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Send setup reply once only
+ If find_media() failed in handle_setup_request() two replies was sent.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
+
+2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6b03ba7 to 865aa20
+
+2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: Emit client-connected signal earlier
+ Emit client-connected before the client ref is given to a GSource,
+ otherwise client-connected can be emitted after the client object has
+ been freed.
+
+2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/addresspool.c:
+ addresspool: return reason of failure
+ Let gst_rtsp_address_pool_reserve_address() return the reason why
+ the address could not be reserved.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
+
+2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
+
+ * autogen.sh:
+ autogen.sh: Sync behaviour with other GStreamer modules
+ Allows building from outside of tree amongst other things
+
+2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
+
+ * common:
+ Automatic update of common submodule
+ From b613661 to 6b03ba7
+
+2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 74a6857 to b613661
+
+2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 01a7a46 to 74a6857
+
+2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Do not read beyond end of path string
+ If the setup was done without a control url, make sure we don't try to read the
+ non-existing control string and crash.
+
+2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Fix RTPInfo header
+ Refactor the method to make the content_base.
+ Use the content-base and the control url to construct the RTPInfo
+ url.
+
+2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: map url to path only in describe
+ Only map the request url to a path in the DESCRIBE method. The SDP then
+ contains the base and control urls that should be used to SETUP/PAUSE/
+ PLAY/TEARDOWN the media.
+
+2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Revert "client: map URL to path in requests"
+ This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
+ This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
+ contains the base and control urls which are used in the SETUP, PLAY,
+ PAUSE and TEARDOWN requests.
+
+2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: map URL to path in requests
+
+2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ mount-points: make vmethod to make path from uri
+ Make a vmethod to transform an url into a path. The path is then used to lookup
+ the factory. This makes it possible to also use other bits of the url, such as
+ the query parameters, to locate the factory.
+
+2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: Add cleanup to wait for the threadpool to finish
+ Also fix race condition if two threads are asking for the first
+ thread from the thread pool at once. This would case two internal
+ GThreadPools to be created.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707753
+
+2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ client: free threadpool
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * tests/check/gst/mountpoints.c:
+ mountpoints tests: unref matched factories
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * tests/check/gst/media.c:
+ media tests: unref thread pool and caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ auth, media, media-factory: unref permissions
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ Makefile: add rule for appsrc example
+
+2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-appsrc.c:
+ tests: add appsrc example
+ Add an example on how to use appsrc to feed the server pipeline with data.
+
+2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: remove query part from content-base string
+ Make sure that after the control url has been resolved, it's
+ not a part of the query-string.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
+
+2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: don't check url in response
+ There is no url or method in the response to check
+
+2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Add handle-response signal for when we receive a GET_PARAMETER response
+
+2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Fix gst_rtsp_server_client_filter, using wrong variable type
+
+2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
+ For AAC we need to check for framed=true instead of parsed=true.
+ https://bugzilla.gnome.org/show_bug.cgi?id=701384
+
+2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: optimize pipeline for protocols
+ When TCP is not an allowed protocol for the stream, avoid creating the
+ appsrc/appsink/queue and tee elements.
+
+2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: set protocols on streams
+
+2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use protocols supported by stream
+
+2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ media-factory: allow all protocols
+
+2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: configure protocols in new streams
+
+2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add protocols property
+
+2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: send state in "new-state" signal
+ https://bugzilla.gnome.org/show_bug.cgi?id=705110
+
+2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
+
+ * configure.ac:
+ build: add subdir-objects to AM_INIT_AUTOMAKE
+ Fixes warnings with automake 1.14
+ https://bugzilla.gnome.org/show_bug.cgi?id=705350
+
+2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add method to iterate clients of server
+
+2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add vmethod for rtsp-media subclass to access rtpbin
+
+2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ small documentation fix
+
+2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Do not take range header if range is invalid
+
+2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media.c:
+ media: add docs for new method
+
+2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add API to rtsp-media set the pipeline's state
+
+2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Update current position/duration when gst_rtsp_media_get_range_string is called
+
+2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-cgroups.c:
+ tests: add some more docs
+
+2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * tests/check/gst/client.c:
+ ClientState -> Context
+ Rename the clientstate to context and put the code in a separate file.
+
+2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ auth: add support for default token
+ The default token is used when the user is not authenticated and can be used to
+ give minimal permissions.
+
+2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: use defines when possible
+
+2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ address-pool: improve docs
+
+2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: add the role to the copy
+
+2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: Also copy the roles
+
+2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: Make it build
+
+2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.h:
+ docs: small fixes
+
+2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ docs: improve docs
+
+2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * tests/check/gst/addresspool.c:
+ * tests/check/gst/rtspserver.c:
+ address-pool: cleanups
+ Remove redundant method, improve docs.
+
+2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-token.c:
+ docs: improve docs
+
+2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: implement _remove_role
+
+2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: update docs
+
+2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: simplify tests
+ Client settings are now disabled by default so we don't need an auth
+ module to disable them.
+
+2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: add default authorizations
+ When no auth module is specified, use our table of defaults to look up the
+ default value of the check instead of always allowing everything. This was
+ we can disallow client settings by default.
+
+2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ README: update readme
+
+2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: add more docs
+
+2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: fix race in thread reuse
+ If we try to reuse a thread right after we made it stop, we end up using a
+ stopped thread. Catch this case and only reuse threads that are not stopping.
+
+2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: add small debug
+
+2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ client: fix test
+ Add some permissions to media so we can use the auth and enable
+ client settings.
+
+2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: support pushed context in handle_request
+ If we already have a pushed state, reuse it and add our own things. This makes
+ it easier to write tests.
+
+2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: don't auth on methods
+ Don't authorize on methods anymore but on the resources that we
+ try to access, this is more flexible.
+ Move the authorization checks to where they are needed and let the
+ check return the response on error.
+
+2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mount-points: add some debug
+
+2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: almost fix test
+
+2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ auth: let the auth module check client_settings
+ Let the auth module decide if client settings are allowed for the
+ current client.
+
+2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ token: add method to check boolean permission
+
+2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ token: simplify token constructor
+ Use variable arguments to make easier API.
+
+2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add convenience API for factory
+
+2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ permissions: simplify API a little
+ Avoid passing GstStructure in the add_role method, use varargs instead
+ to construct the structure behind the scenes. We can then also use the
+ structure name as the role and simplify some more logic.
+
+2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: fix typo
+
+2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ auth: handle unauthorized response
+ Move handling of the unauthorized response to the auth module, it can add
+ the appropriate headers to request authorization for the required method
+ much better than the client.
+
+2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: allow for sending any message, not only requests
+ Change the _send_request() method to _send_message() so that we
+ can both send requests and replies.
+
+2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-server.h:
+ docs: fix docs
+
+2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ auth: move TLS handling to auth module
+ Remove the TLS settings on the server and move it to the auth module because
+ that is where security related bits go.
+
+2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add state push/pop
+
+2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add connection to state
+
+2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mount-points: fix debug
+
+2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/media.c:
+ tests: fix media test
+
+2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: we don't require a state
+
+2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: let context ref the server
+ So that we don't risk losing the server object early anc crash.
+
+2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: fix client test
+
+2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-token.c:
+ docs: improve docs
+
+2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ session-pool: make vmethod to create a session
+ Make a vmethod to create a sessions so that subclasses can create
+ custom session objects
+
+2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ docs: more updates
+
+2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ docs: update docs
+
+2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * examples/Makefile.am:
+ configure: compile cgroup example conditionally
+ Only compile the cgroup example when we have libcgroup
+
+2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * examples/Makefile.am:
+ * examples/test-cgroups.c:
+ examples: add cgroups example
+
+2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/rtspserver.c:
+ tests: fix compilation
+
+2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: fix vmethod invocation
+
+2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: store thread type in thread
+
+2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: pass thread from pool to media _prepare
+ Get a thread from the configured threadpool and pass it to the prepare method of
+ the media.
+
+2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: Accept a thread in _prepare
+ Remove out own threadpool handling and use the provided thread and
+ maincontext for the bus messages and the state changes.
+
+2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: configure client thread pool
+
+2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add method to configure thread pool
+
+2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: use thread pool
+ Use the thread pool instead of doing our own thing.
+
+2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: add object to manage threads
+ Add an object to manage the client and media threads.
+
+2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: debug authorization check
+
+2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: start media pipeline in context
+ Start the media pipeline in the provided context (or our default one
+ when NULL). This makes sure that we run the bus thread in this context and that
+ all media threads are children of this context.
+
+2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: pass permissions to media by default
+
+2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ test: add permissions to auth test
+ Ass some permissions to the media factory in the test.
+
+2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ auth: simplify auth checks
+ Remove client from methods, it's now in the state
+ Perform the check specified by the string, use the information from the
+ thread local context.
+
+2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add state to current thread
+ Add the client to the ClientState object.
+ Place the ClientState on the current thread.
+
+2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: make it possible to set permissions
+ Make it possible to set permissions on media and media factory objects
+
+2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ permissions: add permissions object
+ Add a mini object to store permissions based on a role.
+
+2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ auth: add auth checks
+ Add an enum with auth checks and implement the checks in the auth object.
+ Perform the checks from the client.
+
+2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ auth: use the token after authentication
+ After we authenticated a user, keep the Token around in the state.
+
+2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * tests/check/gst/media.c:
+ media: add optional context for bus messages
+ Add an optional mainloop to _prepare that will handle the bus messages instead
+ of always using the shared mainloop.
+
+2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ token: add authorization token
+ Add a simply miniobject that contains the authorizations. The object contains a
+ GstStructure that hold all authorization fields. When a user is authenticated,
+ the auth module will create a Token for the user. The token is then used to
+ check what operations the user is allowed to do and various other configuration
+ values.
+
+2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ auth: remove auth from media and factory
+ Remove the auth object from media and factory. We want to have the RTSPClient
+ authenticate and authorize resources, there is no need to place another auth
+ manager on the media/factory.
+
+2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ auth: add support for multiple basic auth tokens
+ Make it possible to add multiple basic authorisation tokens to one authorization
+ object. Associate with each token an authorization group that will define what
+ capabilities are allowed.
+
+2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: error out on non-aggregate control
+ We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
+
+2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: rework setup request a little
+ Cache the media in DESCRIBE based on the longest matching path with the uri
+ that we can find in the mount points.
+ Rework the setup request a little to get the media from the session or from
+ the longest matching path, this way we can derive the control string as
+ everything after the path instead of hardcoding it.
+ Find the stream based on the control string and only open a session when all
+ this can be done.
+
+2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add method to find a stream by control url
+
+2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to check control url of stream
+
+2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: use path matching for session media
+ Use a path string instead of a uri to lookup session media in the sessions. Also
+ use path matching to find the largest possible path that matches.
+
+2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * tests/check/gst/mountpoints.c:
+ mount-points: remove useless vmethod
+ Making lookups in the mount points should not be done with a URL, if there is a
+ mapping to be done from URL to mount points, we'll need to do it somewhere
+ else.
+
+2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * tests/check/gst/mountpoints.c:
+ mount-points: improve mount point searching
+ Use a GSequence to keep track of the mount points.
+ Match a URL to the longest matching registered mount point. This should be the
+ URL to perform aggreagate control and the remainder is the stream specific
+ control part.
+ Add some unit tests for this.
+
+2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Allow building of static library
+
+2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/mediafactory.c:
+ tests: fix compilation
+
+2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: get control string from stream
+ Use the control string as configured in the stream.
+
+2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add methods and property to set control string
+
+2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: cleanups
+ Rename variables for clarity
+ Keep media in state when we can
+
+2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add more support for IPv6
+ Rename _get_address to _get_multicast_address in GstRTSPStream to
+ make it clear that this function only deals with multicast.
+ Make it possible to have both an IPv4 and IPv6 multicast address on
+ a stream. Give the client an IPv4 or IPv6 address depending on the
+ address it used to connect to the server.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
+
+2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix comment
+
+2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: handle failed port allocation
+ Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
+ can't allocate any family at all. Also keep track of what port families we
+ allocated.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
+
+2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: improve docs
+
+2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ stream-transport: remove old if 0 block
+
+2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/client.c:
+ tests: fix tests
+ gst_rtsp_client_get_uri() has been removed
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
+
+2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add method to filter managed sessions
+ Add a method to filter the sessions managed by this client connection.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=703016
+
+2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: remove _get_uri() method
+ Remove the get_uri() method on the client. A client has no uri, the uri
+ property is an internal property to manage the last cached media for
+ the client.
+
+2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: fix typo
+
+2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Do not leak the query in default_query_stop
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
+
+2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't unlock when conversion fails
+ Don't unlock the state lock when conversion fails because it was not locked.
+
+2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add query_position and query_stop vmethods to rtsp-media
+
+2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Fix typo in property install for rtsp-media's time-provider
+
+2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: clean some variables
+ Clean some variables and add some guards to _send_request()
+
+2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Add gst_rtsp_client_send_request API
+ This makes it possible to send arbitrary messages to a client, such as
+ SET_PARAMETER or GET_PARAMETER
+
+2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add _get_element() method
+ Add method to get the element used when creating the media.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
+
+2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix docs
+
+2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: allow access to the rtp session
+ https://bugzilla.gnome.org/show_bug.cgi?id=703004
+
+2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ dscp qos support in gst-rtsp-stream
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
+
+2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/rtspserver.c:
+ tests: fix test
+ Actually do what the comment says. Also keep the old code around, not sure what
+ should happen when you get a 454 from a TEARDOWN, does it close the connection?
+ it currently doesn't.
+
+2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: also watch newly created session
+ When we newly created a session, start watching it immediately instead of
+ on the next request.
+
+2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/client.c:
+ tests: add unit test for new-session
+ See https://bugzilla.gnome.org/show_bug.cgi?id=701587
+
+2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: emit new-session when new session is created
+ Only emit new-session when we created a new session for a client, not when a
+ client picked up a previous session.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
+
+2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: handle asterisk as path in requests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
+
+2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: handle segment query format mismatch
+ It's possible that the segment query returns with a different format than what
+ we asked for, handle this case also.
+
+2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: use segment stop in collect_media_stats
+ Use segment stop instead of duration as range end point.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
+
+2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: Do not leak the element in take_pipeline
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
+
+2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: Make configure_client_transport virtual
+ This patch makes configure_client_transport virtual. The functionality is
+ needed to handle some weird clients sending multicast transport settings as url
+ options.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
+
+2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: Make param_set and param_get virtual
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
+
+2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: convert_range replaces get_range_times
+ get_range_times worked for handling UTC ranges for seeks, but we also
+ need to convert back from NPT to the requested unit in
+ get_range_string. convert_range is now used for both.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
+
+2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: cleanup sdp info
+ We don't need to pass the proto, we can more easily check a boolean.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
+
+2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ use 0.0.0.0 or :: for c= line instead of server address
+
+2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ use local address, not remote, in SDP
+ See https://bugzilla.gnome.org/show_bug.cgi?id=702063
+
+2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 098c0d7 to 01a7a46
+
+2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: possibility to override range time conversion
+ Make it possible to override the conversion from GstRTSPTimeRange to
+ GstClockTimes, that is done before seeking on the media
+ pipeline. Overriding can be useful for UTC ranges, where the default
+ conversion gives nanoseconds since 1900.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
+
+2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: Expose the use_client_settings API
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
+
+2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtspstream: handle both ipv4 and ipv6 clients
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
+
+2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
+ This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
+ We already have a way to place extra attributes in the SDP by using a string
+ property with prefix x- or a- in the caps.
+
+2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
+ This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
+ We already have a way to place extra attributes in the SDP, just make a string
+ property in the payloader with a- or x- prefix.
+
+2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp: place a- and x- properties as attributes
+ application/x-rtp has properties with a- and x- prefixes that should be
+ placed as attributes in the SDP for the media instead of being added to the
+ fmtp.
+
+2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-video.c:
+ example: add TLS example
+
+2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add support for TLS
+ Add methods to set and get a TLS certificate.
+ Add vmethod to configure a new connection. By default, configure the TLS
+ certificate in a new connection if needed.
+
+2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: remove accept_client vmethod
+ This vmethod is not very useful so remove it.
+
+2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: don't crash on NULL GError
+
+2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-session-pool: corrected session timeout detection
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
+
+2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve debug
+
+2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ server: refactor connection setup
+ Let the server accept the socket connection and construct a GstRTSPConnection
+ from it. Remove the code from the client and let the client only deal with
+ a fully configure GstRTSPConnection object.
+ We will need this later when the server will configure the connection for
+ TLS.
+
+2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: keep the transport object alive
+ Keep the transport object alive while we have it as qdata on the
+ source.
+
+2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: Do not crash on nmapping of server
+ * generate error when gst_rtsp_connection_accept fails
+ * do not stop accepting incoming connections because
+ accepting a client fails
+ https://bugzilla.gnome.org/show_bug.cgi?id=701072
+
+2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
+ https://bugzilla.gnome.org/show_bug.cgi?id=700953
+
+2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Parse framerate caps field and set SDP attribute
+ The SDP attribute and its format is described in RFC4566.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
+
+2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Parse width/height from caps and set SDP attribute
+ The SDP attribute and its format is described in RFC6064.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
+
+2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ * tests/check/gst/client.c:
+ rtsp-sdp: add bandwidth line
+ https://bugzilla.gnome.org/show_bug.cgi?id=699220
+
+2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 5edcd85 to 098c0d7
+
+2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/media.c:
+ tests: add dynamic payloader prepare/unprepare check
+
+2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: release lock when removing fakesink
+
+2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set elements to NULL before removing
+ When removing a stream, set the elements to NULL first. This avoids
+ element-is-not-in-NULL-state errors when we dispose the elements.
+
+2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 3cb3d3c to 5edcd85
+
+2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: listen to pad-removed signals
+ Listen to the pad-removed signal and remove the stream associated with the
+ removed pad.
+ Add signal to be notified of the removed pad.
+ Remove the fakesink in unprepare()
+ Fix signatures of the signal methods
+
+2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-sdp.c:
+ tests: add example of reusable pipelines
+
+2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to get the srcpad
+
+2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/media.c:
+ check: add media prepare/unprepare test
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: disconnect from signal handlers in unprepare()
+ We connected to the pad-added and no-more-pads signals in prepare() so
+ we need to disconnect from them in unprepare().
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't free streams array
+ Don't free the streams array in the unprepare() method, they were not
+ added in prepare().
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't unref the pipeline in unprepare
+ Unprepare() should undo what prepare() does. Because the pipeline is
+ not created in prepare(), we should not unref it in unprepare()
+
+2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: clear session and caps for reuse
+ Set the session and caps to NULL after unref otherwise we might unref
+ them again later.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: send out teardown signal before tearing down
+ The advantage is that in the signal handler you get direct access to
+ information about what streams are about to get torn down (in the
+ GstRTSPClientState).
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
+
+2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: expose connection
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
+
+2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From aed87ae to 3cb3d3c
+
+2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ media: add method to get the base_time of the pipeline
+ Together with a shared clock, this base-time could eventually be sent to
+ the client so that it can reconstruct the exact running-time of the clock
+ on the server.
+
+2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ media: add GstNetTimeProvider support
+ Add a property to let the media provide a GstNetTimeProvider for its clock.
+ Make methods to get the clock and nettimeprovider
+ Add a x-gst-clock property to the SDP with the IP and port number of the nettime
+ provider and also the current time of the clock. This should make it possible
+ for (GStreamer) clients to slave their clock to the server clock.
+
+2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 04c7a1e to aed87ae
+
+2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: wait for buffering to complete
+ Wait for buffering to complete before changing the state to the target state.
+
+2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: small cleanup
+
+2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: remove extra unref in test_setup_non_existing_stream
+ The unref is not needed anymore, teardown runs without it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=696542
+
+2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: GSocketService cleanup in test_bind_already_in_use
+ Use g_socket_service_stop so the rtspserver test stops listening for
+ incoming connections in test_bind_already_in_use.
+ https://bugzilla.gnome.org/show_bug.cgi?id=696541
+
+2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
+ Instead use a GWeakRef which is safe to use
+ This is a known GLib bug, see:
+ https://bugzilla.gnome.org/show_bug.cgi?id=667145
+
+2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * tests/check/gst/media.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-media/client: Reply to PLAY request with same type of Range
+ Remember the type of Range from the PLAY request and use the same type for
+ the reply.
+
+2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * tests/check/gst/client.c:
+ rtsp-client: expose uri
+
+2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/mediafactory.c:
+ tests: Hold ref while creating second media
+ To test if the media aren't shared, make sure we keep the first one while creating a second
+ otherwise the same memory address may be reused.
+
+2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * configure.ac:
+ configure: remove out-of-date comment
+
+2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * .gitignore:
+ .gitignore: ignore more build files
+
+2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * tests/check/Makefile.am:
+ tests: use right _LIBS variable for gst-plugins-base libs
+
+2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ check: add librtp to libs
+
+2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Add test to check selecting a port the server will send from
+
+2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Make sure packets are actually received
+
+2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Select unicast address from pool if appropriate
+
+2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Properties are always there in Gst 1.0
+
+2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/addresspool.c:
+ tests: Add tests for unicast addresses in pool
+
+2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * tests/check/gst/addresspool.c:
+ address-pool: Verify that multicast addresses are used for multicast and vice-versa
+
+2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/addresspool.c:
+ address-pool: Add unicast addresses
+
+2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * configure.ac:
+ * gst/rtsp-server/rtsp-server.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-server: Limit the number of threads per server instance
+ If we exceed the maximum, just round robin the clients over the existing
+ threads.
+
+2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: No need to store the GMainContext in the client context
+
+2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Add test for client disconnection
+
+2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test client and session timeouts with multiple threads
+
+2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ Document locking and its order
+
+2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test that slow DESCRIBE don't block other clients
+
+2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/client.c:
+ tests: Add tests for client-requested multicast address
+
+2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: Put the various functions in the right sections
+
+2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ docs: Generate docs for GstRTSPAddressPool
+
+2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ client: Check client provided addresses against the address pool
+
+2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * tests/check/gst/addresspool.c:
+ address-pool: Add API to request a specific address from the pool
+ Also add relevant unit tests.
+
+2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/mediafactory.c:
+ tests: Check the passing around of a RTSPAddressPool
+ Make sure the RTSPAddressPool is propagated from the MediaFactory all the
+ way down to the stream.
+
+2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/addresspool.c:
+ tests: Add more tests for the address pool
+
+2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ address-pool: Fix off by one error
+ When splitting a port range, the port after a skip is not part of range.
+
+2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 2de221c to 04c7a1e
+
+2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
+
+ * configure.ac:
+ configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
+ AM_CONFIG_HEADER was removed in automake 1.13
+ https://bugzilla.gnome.org/show_bug.cgi?id=693368
+
+2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From a942293 to 2de221c
+
+2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: make sure the watch exists while sending data
+ Protect the send_func with a lock. This allows us to wait for sending
+ to complete before changing the send_func and user_data. We add an
+ extra ref to the watch to make sure that it remains valid during
+ sending.
+ When closing the connection, set the send_func to NULL
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
+
+2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ tests: use GST_*_1_0 environment variables everywhere
+ The _1_0 suffixed environment variables override the
+ non-suffixed ones, so if we're in an environment that
+ sets the _1_0 suffixed ones, such as jhbuild, we need
+ to set those to make sure ours actually always get
+ used.
+
+2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From acb04d9 to a942293
+
+2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: set the client backlog
+ Set the client backlog to a reasonable default
+
+2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Make the element a constructor parameter
+ https://bugzilla.gnome.org/show_bug.cgi?id=689594
+
+2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * docs/libs/Makefile.am:
+ docs: Link with gcov library when gcov is enabled
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
+
+2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: match prepare with unprepare
+ Really unprepare when there were an equal amount of prepare calls.
+
+2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: media has to be unprepared in finalize
+ Because unprepare takes away the last ref on the media.
+
+2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
+ This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
+ We can't use the refcount to trigger unprepare because it is the unprepare call
+ that removes the last refcount after all messages are consumed. What we should
+ probably do is make a prepared refcount and only unprepare when the refcount
+ reaches 0.
+
+2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: let the source unref the last media ref
+ the last ref to the media is held by the source so we don't need to add more ref
+ and unrefs, we simply destroy the media when the source is gone.
+
+2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: improve debug
+
+2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: check state
+ Make sure we are in the right state when collecting the position and duration.
+ Only make ourselves PREPARED when we were previously PREPARING.
+
+2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: use g_object_ref/unref for GObjects
+
+2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
+ Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
+ GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
+ isn't being used anymore.
+
+2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Fix compiler warning
+
+2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
+
+2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-media.h:
+ small cleanup
+
+2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: avoid element leak
+
+2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: require an element in media constructor
+
+2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Revert "client: TEARDOWN brings that state to Init again"
+ This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
+ The object is already disposed, there is no point in setting the state.
+
+2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: TEARDOWN brings that state to Init again
+
+2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/media.c:
+ rtsp: make object details private
+ Make all object details private
+ Add methods to access private bits
+
+2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/media.c:
+ tests: add media tests
+
+2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: check if prepared for some methods
+ Check that the media object is prepared before doing seek and getting the
+ current position etc.
+ Add some g_return checks.
+
+2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/mediafactory.c:
+ tests: add mediafactory test
+
+2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: improve debug
+
+2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: unref pipeline in finalize to avoid leaking it
+
+2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp: use gst_object_unref on GstObjects
+
+2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: require an url
+
+2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ examples: fix include
+
+2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.h:
+ server: remove unused include
+
+2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/mountpoints.c:
+ tests: add test for mountpoints
+
+2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix factory leak
+ Keep the factory in the state object only for authorization checks and make
+ sure we unref it on failure. Also don't keep invalid objects in the state
+ object.
+
+2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mounts: add g_return_if guards
+
+2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: add more tests
+
+2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve debug
+
+2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve debug and fix leaks
+ Cleanup the uri and session when there is a bad request.
+
+2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * common:
+ update common
+
+2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ test: add test for session in options request
+
+2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use 454 when session can't be found
+ We should use 454 when a session can't be found because there was no session
+ pool configured in the server. This is not a server configuration problem
+ because the server on which the request is done might not be the same one that
+ will keep the sessions for us and so it does not need to support sessions.
+
+2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: only free connection when there is one
+ It's possible that the client doesn't have a connection when we try to free it.
+
+2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/client.c:
+ tests: add unit test for the client object
+
+2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: small cleanup
+
+2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ client: remove unused include
+
+2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix compilation
+
+2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: call destroy without the lock
+
+2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: make the client usable without a socket
+ Make a method to let the client handle a message and a callback when the client
+ wants us to send a response message back. This makes it possible to also use the
+ client object without the sockets, which should make it easier to test.
+
+2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: small cleanup
+
+2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ client: remove reference to server
+ We don't need to keep a ref to the server
+
+2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add locking
+ Also add some g_return_if()
+
+2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: log more errors
+
+2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix compilation
+
+2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add generic close-after-send support
+ Add a property to send_response() to close the connection after the response has
+ been sent to the client.
+
+2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * docs/libs/gst-rtsp-server.types:
+ * examples/test-auth.c:
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-multicast.c:
+ * examples/test-multicast2.c:
+ * examples/test-ogg.c:
+ * examples/test-readme.c:
+ * examples/test-sdp.c:
+ * examples/test-uri.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * tests/check/gst/rtspserver.c:
+ MediaMapping -> MountPoints
+ Describes better what the object manages.
+
+2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump required version of -base
+
+2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix seeking
+
+2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: support more Range formats
+ Use the new -base methods to convert the Range string into a seek start and stop
+ value.
+
+2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-launch.c:
+ examples: fix whitespace
+
+2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ test-auth: add example of how to remove sessions
+ Add an example of the session filter api.
+
+2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ test-uri: remove mapping example
+
+2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ test-uri: fix callback signature
+
+2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: keep ref to factory while media active
+ While the media from a factory is alive, keep a ref to the factory.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
+
+2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: add some debug
+
+2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set udp sources to PLAYING
+ Set the UDP sources to PLAYING and locked state before we add it to the pipeline
+ so that it doesn't cause our pipeline to produce ASYNC-DONE.
+
+2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: take ref to factory
+ Take a ref to the factory that we place in our list.
+
+2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/Makefile.am:
+ * tests/test-reuse.c:
+ test: add test for server reuse
+ See https://bugzilla.gnome.org/show_bug.cgi?id=688395
+
+2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: start and stop multiple times
+ Stop listening on the RTSP port when the GSource is removed, so clients
+ can't connect and the server can be started again.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
+
+2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: fix small leak
+
+2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: unref source in finish_unprepare
+ The source is created in prepare, unref it in finish_unprepare.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=688707
+
+2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: remove bus watch before finalizing
+ * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
+ * An extra media ref is added for the bus watch. This extra ref is unreffed by
+ the GDestroyNotify function.
+ * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
+ * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
+ gst_rtsp_media_unprepare before unreffing the media.
+ This way, the bus watch will be removed before the media is finalized.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
+
+2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: wait until the TEARDOWN response is sent to close the connection
+ Responses can be sent async so we need to wait until the TEARDOWN response has
+ been written before we close the connection to the client. This avoids the risk
+ of writing/polling closed sockets.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
+
+2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: plug socket leak
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
+
+2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6bb6951 to a72faea
+
+2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-server: don't use deprecated API
+
+2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix unused-but-set-variable compiler warning
+ rtsp-client.c:1260:21: error: variable 'protocols' set but not used
+
+2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * TODO:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp: cleanups
+
+2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-multicast2.c:
+ examples: add another multicast example
+ Add an example for how to configure separate multicast ranges for each media
+ stream.
+
+2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-multicast.c:
+ test: set shared
+
+2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ stream: use the address managed by the stream
+ Use the address managed by the stream for multicast. This allows us to have 1
+ multicast address for each stream.
+ Because the address is now managed by the stream we don't have to pass it around
+ anymore.
+ Set the address pool on the streams.
+
+2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp: improve debug
+
+2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal for new streams
+ This allows applications to listen for new streams and configure properties on
+ them, like the address pool.
+
+2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: configure address pool in new streams
+
+2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add methods to deal with address pool
+ Add methods to get and set the address pool for the stream
+ Add method to allocate and get the multicast addresses for this stream.
+
+2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: remove MTU property
+ It is a stream property
+
+2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: set blocksize only on stream
+ Set the blocksize only on the current stream.
+
+2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: share src and sink sockets
+ the allocated socket is in the used-socket property, not socket.
+
+2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * tests/check/gst/addresspool.c:
+ rtsp: make address-pool return an address object
+ Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
+ store more info in the structure and allows us to more easily return the address
+ to the right pool when no longer needed.
+ Pass the address to the StreamTransport so that we can return it to the pool
+ when the stream transport is freed or changed.
+
+2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-multicast.c:
+ examples: add multicast example
+ Show how to set up the multicast address pool so that media can be
+ server with multicast.
+
+2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp: use AddressPool
+ Remove the multicast_group property.
+ Use the configured addresspool to allocate multicast addresses.
+
+2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ address-pool: add clear method
+
+2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ address-pool: small cleanups
+
+2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/addresspool.c:
+ tests: add addresspool unit test
+
+2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ address-pool: add object to manage multicast addresses
+ Make an object that can manage a rage of multicast addresses and ports.
+
+2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: set default max-threads property
+
+2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: wait for concurrent _prepare
+ If a prepare is busy, wait for the result.
+
+2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: add lock around message handler
+ We don't want to dispatch messages while we are still processing the result of
+ the state change.
+
+2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add lock to protect state changes
+
+2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add locking
+
+2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream-transport: add keep-alive method
+
+2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream-transport: add method to handle RTP/RTCP
+ Call new methods instead of poking into the structures directly.
+
+2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ session-media: add locking
+
+2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: add locking
+
+2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: free old socket
+
+2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ mapping: add locking
+
+2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: add locking
+
+2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ auth: add locking
+
+2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add max-thread property
+
+2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: use a threadpool for the mainloops
+
+2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: rename method
+ gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
+ don't really create the client from the socket, we use the socket for the
+ client.
+
+2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ server: rework maincontext handling in clients
+ Make a separate method to attach a client to a MainContext.
+ Let the server decide in what GMainContext the client will operate and give this
+ context to the client in attach. Then the server can later decide to use a
+ separate thread for each client or just use the mainthread.
+
+2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: move session header code in session object
+
+2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * COPYING:
+ * COPYING.LIB:
+ * examples/test-auth.c:
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-readme.c:
+ * examples/test-sdp.c:
+ * examples/test-uri.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/rtspserver.c:
+ * tests/test-cleanup.c:
+ Fix FSF address
+
+2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-server: added annotations to indicate type of ownership transfer of return values
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * configure.ac:
+ No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
+
+2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * Makefile.am:
+ * bindings/Makefile.am:
+ * bindings/vala/Makefile.am:
+ * bindings/vala/gst-rtsp-server-0.10.deps:
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.deps:
+ * bindings/vala/packages/gst-rtsp-server-0.10.files:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
+ * configure.ac:
+ bindings: remove vala bindings
+ They'll be reunited with the other GStreamer bindings
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ rtsp: only create transport when needed
+ Only create the StreamTransport when configured.
+
+2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: small cleanup
+
+2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ rtsp: refactor configuration of transport
+ Move the configuration of the transport to a place where it makes
+ more sense.
+
+2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: refactor transport parsing
+
+2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: refuse to change the MTU on shared media
+ If we change the MTU of chared media, it changes for all clients.
+ We don't want to set the MTU to something large for clients that
+ stream over UDP.
+
+2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-mp4.c:
+ * gst/rtsp-server/rtsp-media.c:
+ small fixes to docs and debug
+
+2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: transports must already have been removed
+
+2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: improve join and leave of the pipeline
+ simplify code
+ Do the cleanup properly
+ Add some docs
+
+2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: move unprepare below default implementation
+ Makes it easier to find the default implementation
+
+2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: signal unprepared when we actually finish
+
+2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: no need to unlock, unprepare does that when needed
+
+2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ docs: update docs
+
+2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp: fix MTU setting
+ Fix setting of the MTU. There is no need for a vmethod.
+
+2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ docs: update docs
+
+2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump version number after refactoring
+
+2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp: massive refactoring
+ Make GObjects from the remaining simple structures.
+ Remove GstRTSPSessionStream, it's not needed.
+ Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
+ Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
+ a GstRTSPStream should be transported to a client.
+ Rename GstRTSPMediaFactory::get_element -> create_element because that
+ more accurately describes what it does.
+ Make nice methods instead of poking in the structures.
+ Move some methods inside the relevant object source code.
+ Use GPtrArray to store objects instead of plain arrays, it is more
+ natural and allows us to more easily clean up.
+ Move the allocation of udp ports to the Stream object. The Stream object
+ contains the elements needed to stream the media to a client.
+ Improve the prepare and unprepare methods. Unprepare should now undo
+ everything prepare did. Improve also async unprepare when doing EOS on
+ shutdown. Make sure we always unprepare correctly.
+
+2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Unref server address clients connected to
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
+
+2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: don't ref server socket if it is NULL
+ Fixes test_bind_already_in_use unit test again after commit 6a497440.
+ https://bugzilla.gnome.org/show_bug.cgi?id=686644
+
+2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * tests/check/Makefile.am:
+ tests: Add libgio link dependency
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
+
+2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ rtsp-media-mapping: rename find_media vfunc to find_factory
+ The virtual method and class method should have the same name
+ so it is correctly represented in GIR file
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-server: fixed comments and GIR annotations
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
+
+2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: allow binding on port 0 (binds on a random port)
+
+2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: add bound-port property
+ bound-port can be used to retrieve the port number when the server is bound on
+ port 0, which binds on a random port.
+
+2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ rtsp-media-factory: make ::get_element overridable by GI bindings
+ The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
+ for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
+ as the invoker for ::get_element(), making it overridable by GI generated
+ bindings.
+
+2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: don't autoplug parsers in a loop
+ Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
+ h264parse forever.
+
+2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/Makefile.am:
+ Explicitly link against gio. Fix link error on mac.
+
+2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: add ttl to the transport header in SETUP
+ See https://bugzilla.gnome.org/show_bug.cgi?id=685561
+
+2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ client: Use client transport settings for multicast if allowed.
+ This patch makes it possible for the client to send transport settings for
+ multicast (destination && ttl). Client settings must be explicitly allowed or
+ the server will use its own settings.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
+
+2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6c0b52c to 6bb6951
+
+2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: do not destroy the rtsp watch
+ Don't destroy the client watch while dispatching. The rtsp watch is
+ automatically destroyed after the rtsp watch function closed() has
+ been called.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
+
+2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 4f962f7 to 6c0b52c
+
+2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix check for seekability
+
+2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use more GIO
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
+
+2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: remove obsolete includes
+
+2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
+ * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
+ be available in "on_new_ssrc". The transports are added in
+ gst_rtsp_media_set_state when going to PLAYING state. However,
+ "on_new_ssrc" might be called before this happens.
+ https://bugzilla.gnome.org/show_bug.cgi?id=683304
+
+2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: add signals for rtsp requests (fixes #683287)
+
+2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ add new-session signal to rtsp-client (fixes #683058)
+
+2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 668acee to 4f962f7
+
+2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-server: fixed segfault in gst_rtsp_server_create_socket
+ Do not assume that *error is set in g_socket_address_enumerator_next.
+ Added test_bind_already_in_use unit-test.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
+
+2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 94ccf4c to 668acee
+
+2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: make create_sdp virtual method
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
+
+2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 98e386f to 94ccf4c
+
+2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix docs
+
+2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: use an existing socket to establish HTTP tunnel
+ Make it possible to transfer a socket from an HTTP server to be used as
+ an RTSP over HTTP tunnel.
+
+2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp: Handle the blocksize parameter
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
+
+2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/rtspserver.c:
+ Have unit test get header from source dir, not installed dir
+ This makes compilation of unit tests work in a build directory other
+ than the source directory.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
+
+2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: update for gst_element_make_from_uri() changes
+
+2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * configure.ac:
+ * tests/Makefile.am:
+ * tests/check/Makefile.am:
+ * tests/check/gst/rtspserver.c:
+ rtsp: add unit test
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
+
+2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: don't collect media stats when going to NULL
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
+
+2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: don't leak transports
+
+2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: free transport on no_stream in SETUP handler
+
+2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: changed session media iteration
+ In client_unlink_session: now don't iterate in session->medias
+ list where items are removed by gst_rtsp_session_release_media.
+ Instead, repeatedly remove the first item.
+
+2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
+ GstRTSPSessionMedia is not a GObject type. When the
+ GstRTSPSession is freed, it will free the media.
+
+2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: plug pad leak in collect_streams
+ In gst_rtsp_media_factory_collect_streams: unref the srcpad that
+ was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
+ will take one reference, and the other reference will otherwise
+ give a memory leak.
+
+2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * configure.ac:
+ configure: suppress some warnings when debug is disabled
+ Warnings about unused variables should be suppressed if core has the
+ debug system disabled.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
+
+2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/libs/Makefile.am:
+ docs: fix build in uninstalled setup
+ Include gst-plugins-base libs properly.
+
+2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * docs/libs/gst-rtsp-server.types:
+ docs: include headers defining rtsp-server object types
+ Fixes compiler warnings during docs build.
+ https://bugzilla.gnome.org/show_bug.cgi?id=676824
+
+2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * configure.ac:
+ configure: Add warning flags for compiler when configuring
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
+
+2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 03a0e57 to 98e386f
+
+2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 1fab359 to 03a0e57
+
+2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix GSocketAddress leak in gst_rtsp_client_accept
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
+
+2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From f1b5a96 to 1fab359
+
+2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 92b7266 to f1b5a96
+
+2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From ec1c4a8 to 92b7266
+
+2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 3429ba6 to ec1c4a8
+
+2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp: fix compiler warnings
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
+
+2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From dc70203 to 3429ba6
+
+2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ rtsp-server: port to new thread API
+
+2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6db25be to dc70203
+
+2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-server: Fix compilation and compiler warnings
+
+2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * autogen.sh:
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ configure: Modernize autotools setup a bit
+ Also we now only create tar.bz2 and tar.xz tarballs.
+
+2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 464fe15 to 6db25be
+
+2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 7fda524 to 464fe15
+
+2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ * docs/libs/Makefile.am:
+ * docs/version.entities.in:
+ * gst-rtsp.spec.in:
+ * gst/rtsp-server/Makefile.am:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp-server.pc.in:
+ * tests/Makefile.am:
+ rtsp-server: Update versioning
+
+2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/0.10'
+ Conflicts:
+ gst/rtsp-server/rtsp-session-pool.c
+
+2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-server: Don't use deprecated GLib API
+
+2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Replace master with 0.11
+
+2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * docs/README:
+ A couple minor typo fixes
+
+2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix state of the appqueue
+
+2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory: use videoconvert
+
+2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory: change to new style caps
+
+2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-server: port to GIO
+ Port to GIO
+
+2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: fix build
+
+2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/README:
+ docs: fix for gst_rtsp_server_set_port() -> _set_service()
+ https://bugzilla.gnome.org/show_bug.cgi?id=666548
+
+2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * examples/Makefile.am:
+ First rule of gst-rtsp-server club: don't talk about gst-phonon
+
+2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp-server.pc.in:
+ pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
+ For consistency with all other modules.
+
+2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: update for new map API
+
+2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * .gitignore:
+ * bindings/Makefile.am:
+ * bindings/python/Makefile.am:
+ * bindings/python/arg-types.py:
+ * bindings/python/codegen/Makefile.am:
+ * bindings/python/codegen/__init__.py:
+ * bindings/python/codegen/argtypes.py:
+ * bindings/python/codegen/code-coverage.py:
+ * bindings/python/codegen/codegen.py:
+ * bindings/python/codegen/definitions.py:
+ * bindings/python/codegen/defsparser.py:
+ * bindings/python/codegen/docextract.py:
+ * bindings/python/codegen/docgen.py:
+ * bindings/python/codegen/fileprefix.override:
+ * bindings/python/codegen/fileprefixmodule.c:
+ * bindings/python/codegen/h2def.py:
+ * bindings/python/codegen/mergedefs.py:
+ * bindings/python/codegen/mkskel.py:
+ * bindings/python/codegen/override.py:
+ * bindings/python/codegen/reversewrapper.py:
+ * bindings/python/codegen/scmexpr.py:
+ * bindings/python/rtspserver-types.defs:
+ * bindings/python/rtspserver.defs:
+ * bindings/python/rtspserver.override:
+ * bindings/python/rtspservermodule.c:
+ * bindings/python/test.py:
+ * configure.ac:
+ python: remove pygst-based python bindings
+ pygi is the future, apparently.
+
+2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+
+ * common:
+ Automatic update of common submodule
+ From c463bc0 to 7fda524
+
+2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 2a59016 to c463bc0
+
+2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 0807187 to 2a59016
+
+2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 11f0cd5 to 0807187
+
+2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ example: update for new caps
+
+2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: port some more to 0.11
+ Fix caps.
+ Remove bufferlist stuff
+ Update for new API.
+ Add queue before appsink now that preroll-queue-len is gone.
+ Update for request pad changes.
+
+2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add a seekable boolean
+ Maintain the seekable state with a new variable instead of reusing the
+ is_live variable.
+
+2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Disallow seek in live media
+
+2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
+
+ * gst/rtsp-server/rtsp-server.c:
+ #ifdef statements for windows socket creation were missing
+
+2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From a39eb83 to 11f0cd5
+
+2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 605cd9a to a39eb83
+
+2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use method to access property
+
+2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add protocols property
+ Add a property to configure the allowed protocols in the media created from the
+ factory.
+
+2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add media-configure signal
+ Add signal to allow the application to configure the media after it was created
+ from the factory.
+
+2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use method to access property
+
+2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add protocols property
+ Add a property to configure the allowed protocols in the media created from the
+ factory.
+
+2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add media-configure signal
+ Add signal to allow the application to configure the media after it was created
+ from the factory.
+
+2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use media multicast group
+
+2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ retab some .h
+
+2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: copy and free the server ip address
+ Copy and free the server ip address to make memory management easier later.
+
+2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: configure multicast in media
+
+2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add property for multicast group
+ Add a property to configure the multicast group in the media.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add property for multicast group
+ Add a property to configure the multicast group in the media factory.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: do configuration of transport in one place
+ Move the configuration of the transport destination address to where we also
+ configure the other bits.
+
+2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use media multicast group
+
+2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ retab some .h
+
+2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: copy and free the server ip address
+ Copy and free the server ip address to make memory management easier later.
+
+2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: configure multicast in media
+
+2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add property for multicast group
+ Add a property to configure the multicast group in the media.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add property for multicast group
+ Add a property to configure the multicast group in the media factory.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: do configuration of transport in one place
+ Move the configuration of the transport destination address to where we also
+ configure the other bits.
+
+2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: destroy pipeline on client disconnect with no prior TEARDOWN.
+ The problem occurs when the client abruptly closes the connection without
+ issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
+ server is where the pipeline gets torn down. Since this handler is not called,
+ the pipeline remains and is up and running. Subsequent clients get their own
+ pipelines and if the do not issue TEARDOWNs then those pipelines will also
+ remain up and running. This is a resource leak.
+
+2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
+ For example, it can be used to retrieve source elements like appsrc, in a more
+ convenient way than subclassing get_element.
+
+2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: hold on to reference while using object
+
+2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: use new api
+
+2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: use unstable api
+
+2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix reference counting
+
+2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ fix compiler warnings about unused variables
+
+2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
+
+ * examples/test-launch.c:
+ * examples/test-readme.c:
+ * examples/test-uri.c:
+ * examples/test-video.c:
+ examples: tell rtsp uri when ready
+
+2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
+
+ * common:
+ Automatic update of common submodule
+ From 69b981f to 605cd9a
+
+2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: update for buffer API change
+
+2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile.am: 0.10 => @GST_MAJORMINOR@
+
+2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
+
+2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/.gitignore:
+ .gitignore: 0.10 => 0.11
+
+2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile.am: 0.10 => @GST_MAJORMINOR@
+
+2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 9e5bbd5 to 69b981f
+
+2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From fd35073 to 9e5bbd5
+
+2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 46dfcea to fd35073
+
+2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ media: port to new caps API
+
+2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ Updated Vala bindings.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Add a signal for newly connected clients.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/rtspserver.override:
+ python: override gst_rtsp_media_mapping_add_factory to fix refcounting
+
+2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-funnel.c:
+ * gst/rtsp-server/rtsp-funnel.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-server: port to 0.11
+
+2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * common:
+ add common
+
+2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ common
+ configure.ac
+
+2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From c3cafe1 to 46dfcea
+
+2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/Makefile.am:
+ * bindings/python/rtspserver.defs:
+ python bindings: wrap GstRTSPMediaFactoryClass vfuncs
+
+2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/arg-types.py:
+ python bindings: add GstRTSPUrlParam
+ Needed to implement MediaFactory virtual proxies
+
+2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/arg-types.py:
+ python bindings: fix returning GstRTSPUrl types
+
+2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/arg-types.py:
+ python bindings: add arg type for GstRTSPUrl
+
+2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/rtspserver.defs:
+ python bindings: fix the definition of MediaFactory.collect_stream
+
+2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 1ccbe09 to c3cafe1
+
+2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 193b717 to 1ccbe09
+
+2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From b77e2bf to 193b717
+
+2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * Makefile.am:
+ build: Include lcov.mak to allow test coverage report generation
+
+2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From d8814b6 to b77e2bf
+
+2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6aaa286 to d8814b6
+
+2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6aec6b9 to 6aaa286
+
+2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * autogen.sh:
+ autogen: wingo signed comment
+
+2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session: use full charset for RTSP session ID
+ As specified in RFC 2326 section 3.4 use full valid charset to make guessing
+ session ID more difficult.
+ https://bugzilla.gnome.org/show_bug.cgi?id=643812
+
+2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Don't install the funnel header
+
+2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 1de7f6a to 6aec6b9
+
+2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: require core/base 0.10.31
+ Needed at least for gst_plugin_feature_rank_compare_func().
+
+2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f94d739 to 1de7f6a
+
+2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: remove more unused code
+
+2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: remove duplicate filtering
+ Remove the duplicate filtering code now that we have a released -good version.
+ Give a warning instead.
+
+2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix default buffer size
+
+2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add property to configure the buffer-size
+ Add a property to configure the kernel UDP buffer size.
+
+2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add property to configure kernel buffer sizes
+ Add a property to configure the kernel UDP buffer size.
+
+2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: set PYGOBJECT_REQ before using it
+ https://bugzilla.gnome.org/show_bug.cgi?id=640641
+
+2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/Makefile.am:
+ docs: recursive into sub-directories on 'make upload'
+
+2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/version.entities.in:
+ docs: mention full version these docs are for, not just major-minor
+
+2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.8 ===
+
+2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.8
+
+2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: clarify docs a little
+
+2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: init debug category before starting thread
+
+2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: add realm to make it more spec compliant
+
+2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add locking
+
+2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ example: improve example docs a little
+
+2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: ensure the watch has a ref to the server
+
+2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: simpify channel function
+
+2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: simplify management of channel and source
+ We don't need to keep around the channel and source objects. Let the mainloop
+ and the source manage the source and channel respectively.
+
+2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * Makefile.am:
+ * configure.ac:
+ build tests
+
+2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/.gitignore:
+ * tests/Makefile.am:
+ * tests/test-cleanup.c:
+ tests: add tests directory and cleanup test
+
+2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ server: improve debugging in various objects
+
+2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: chain up to the parent finalize
+
+2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
+
+ * bindings/python/rtspserver-types.defs:
+ * bindings/python/rtspserver.defs:
+ * bindings/python/rtspserver.override:
+ * bindings/python/test.py:
+ gst-rtsp-server: update python bindings
+
+2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use the response from the clientstate
+ Create the response object only once and store in the client state.
+ Make all methods use the state response,
+
+2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: use signal to keep track of clients
+ Keep track of all the clients that the server creates and remove them when they
+ fire the 'closed' signal.
+
+2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: emit signal when closing
+
+2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-auth.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ media: enable per factory authorisations
+ Allow for adding a GstRTSPAuth on the factory and media level and check
+ permissions when accessing the factory.
+ Add hints to the auth methods for future more fine grained authorisation.
+ Add example application for per factory authentication.
+
+2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ rtsp-server: Pass ClientState structure arround
+ Pass the collected information for the ongoing request in a GstRTSPClientState
+ structure that we can then pass around to simplify the method arguments. This
+ will also be handy when we implement logging functionality.
+
+2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add methods to configure authorisation
+
+2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: unref auth in finalize
+
+2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: unref auth in finalize
+
+2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * docs/libs/gst-rtsp-server.types:
+ docs: add more docs
+
+2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: separate create and accept
+ Create separate create and accept methods so that subclasses can create custom
+ client object.
+ Configure the server in the client object and prepare for keeping track of
+ connected clients.
+
+2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add support for setting the server.
+ Add support for keeping a ref to the server that started this client
+ connection.
+
+2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: fix memleak and add some docs
+ Fix a memleak of the basic auth token.
+ Add docs for the helper function
+
+2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ client: delegate setup of auth to the manager
+ Delegate the configuration of the authentication tokens to the manager object
+ when configured.
+
+2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ auth: add authentication object
+ Add an object that can check the authorization of requests.
+ Implement basic authentication.
+ Add example authentication to test-video
+
+2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: move includes back
+ the includes are needed for sockaddr_in.
+
+2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp: move network includes where they are needed
+
+2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media.h: Minor corrections in comments.
+ Fixes #638944
+
+2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From e572c87 to f94d739
+
+2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * .gitignore:
+ * docs/.gitignore:
+ * docs/libs/.gitignore:
+ * examples/.gitignore:
+ * gst/rtsp-server/.gitignore:
+ gitignore: updates
+
+2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * docs/libs/Makefile.am:
+ docs: We don't build ps/pdf for API reference docs
+
+2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From ccbaa85 to e572c87
+
+2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 46445ad to ccbaa85
+
+2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-funnel.c:
+ * gst/rtsp-server/rtsp-funnel.h:
+ * gst/rtsp-server/rtsp-media.c:
+ funnel: rename fsfunnel to rtspfunnel
+ Rename the funnel to avoid conflicts with the farsight one.
+
+2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/fs-funnel.c:
+ * gst/rtsp-server/fs-funnel.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: add and use fsfunnel
+ Add a copy of fsfunnel to the build because input-selector removed the (broken)
+ select-all property that we need.
+
+2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ gobject-introspection: use PKG_CONFIG_PATH specified at configure time
+ Use PKG_CONFIG_PATH specified at configure time (if any) as well
+ for the g-ir-compiler, rather than just assuming the env var has
+ been set.
+
+2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ * m4/Makefile.am:
+ * m4/codeset.m4:
+ build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
+
+2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ gobject-introspection: fix g-i build for uninstalled setup
+ Requires gst-plugins-base git (> 0.10.31.2).
+
+2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ examples: add some more options and comments
+
+2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: use right property type
+
+2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: attempt to configure buffer-lists
+ Attempt to configure buffer lists in the payloader for improved performance.
+
+2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: attempt to configure bigger UDP buffers
+ Attempt to configure bigger udp kernel send buffers to avoid overflowing the
+ send buffers with high bitrate streams.
+
+2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use the socket length from getsockname
+ Use the length returned by getsockname to perform the getnameinfo call because
+ the size can depend on the socket type and platform.
+ Fixes #638723
+
+2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: add uri factory to the docs
+
+2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.h:
+ docs: improve docs
+
+2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: add support for buffer lists
+ Add support for sending bufferlists received from appsink.
+ Fixes #635832
+
+2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ media: make method to retrieve the play range
+ Make a method to retrieve the playback range so that we can conditionally create
+ a different range for the SDP and the PLAY requests.
+
+2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal to notify of state changes
+
+2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ client: cleanup headers
+
+2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix typo
+
+2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ factory-uri: add support for gstpay
+ Add an option to prefer gstpay over decoder + raw payloader.
+
+2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ factory-uri: rework the autoplugger.
+ Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
+ before payloaders.
+
+2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: use better factory filter
+ Make better payloader filter based on autoplug rank and RTP use case.
+
+2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 169462a to 46445ad
+
+2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: set SO_REUSEADDR before bind
+ Set the SO_REUSEADDR _before_ bind() to make it actually work.
+
+2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: emit prepared signal when prepared
+ Make a 'prepared' signal and emit it when we successfully prepared the element.
+ This signal can be used to configure the media object after it has been prepared
+ for streaming.
+
+2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 011bcc8 to 169462a
+
+2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
+
+ python an optional dependency
+ * configure.ac: Move up valgrind and g-i checks. Make the python
+ dependency optional, as it was before.
+
+2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ common
+ configure.ac
+
+2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: update range when active clients changed
+ When we changed the number of active clients, update the current range
+ information because we want the second client connecting to a shared resource
+ continue from where the stream currently.
+
+2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ factory-uri: add colorspace and fix pt
+ Rework the way we pass data to the autoplugger.
+ When we have raw caps, plug a converter element to make pluggin to raw
+ payloaders more successful.
+ Make sure all dynamically plugged payloaders have a unique payload types.
+
+2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-uri.c:
+ example: add example of the uri factory
+
+2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-server.h:
+ factory-uri: add a factory to stream any URI
+ Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
+ when we have one.
+
+2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: ignore spurious ASYNC_DONE messages
+ When we are dynamically adding pads, the addition of the udpsrc elements will
+ trigger an ASYNC_DONE. We have to ignore this because we only want to react to
+ the real ASYNC_DONE when everything is prerolled.
+
+2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: make lock macro
+
+2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-server: Remove unused variable and dead assignment
+
+2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-readme.c:
+ * examples/test-sdp.c:
+ * examples/test-video.c:
+ examples: Run gst-indent
+
+2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-server: Run gst-indent
+ Since it wasn't using the upstream common previously, there was no
+ indentation check before commiting.
+
+2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: Some more doc fixups
+
+2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * Makefile.am:
+ Makefile: Add cruft-cleaning support
+
+2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * Makefile.am:
+ * configure.ac:
+ * docs/Makefile.am:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * docs/libs/gst-rtsp-server.types:
+ * docs/version.entities.in:
+ docs: Add gtk-doc build system
+
+2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile.am: Use standard GIR make behaviour
+
+2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * autogen.sh:
+ * configure.ac:
+ autogen/configure: Bring more in sync to standard gst module behaviour
+
+2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: warn and fail when gstrtpbin is not found
+
+2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: open 0.11 branch
+
+2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * .gitmodules:
+ * common:
+ Add common submodule
+
+2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common/ChangeLog:
+ * common/Makefile.am:
+ * common/c-to-xml.py:
+ * common/check.mak:
+ * common/coverage/coverage-report-entry.pl:
+ * common/coverage/coverage-report.pl:
+ * common/coverage/coverage-report.xsl:
+ * common/coverage/lcov.mak:
+ * common/gettext.patch:
+ * common/glib-gen.mak:
+ * common/gst-autogen.sh:
+ * common/gst-xmlinspect.py:
+ * common/gst.supp:
+ * common/gstdoc-scangobj:
+ * common/gtk-doc-plugins.mak:
+ * common/gtk-doc.mak:
+ * common/m4/.gitignore:
+ * common/m4/Makefile.am:
+ * common/m4/README:
+ * common/m4/as-ac-expand.m4:
+ * common/m4/as-auto-alt.m4:
+ * common/m4/as-compiler-flag.m4:
+ * common/m4/as-compiler.m4:
+ * common/m4/as-docbook.m4:
+ * common/m4/as-libtool-tags.m4:
+ * common/m4/as-libtool.m4:
+ * common/m4/as-python.m4:
+ * common/m4/as-scrub-include.m4:
+ * common/m4/as-version.m4:
+ * common/m4/ax_create_stdint_h.m4:
+ * common/m4/check.m4:
+ * common/m4/glib-gettext.m4:
+ * common/m4/gst-arch.m4:
+ * common/m4/gst-args.m4:
+ * common/m4/gst-check.m4:
+ * common/m4/gst-debuginfo.m4:
+ * common/m4/gst-default.m4:
+ * common/m4/gst-doc.m4:
+ * common/m4/gst-error.m4:
+ * common/m4/gst-feature.m4:
+ * common/m4/gst-function.m4:
+ * common/m4/gst-gettext.m4:
+ * common/m4/gst-glib2.m4:
+ * common/m4/gst-libxml2.m4:
+ * common/m4/gst-plugindir.m4:
+ * common/m4/gst-valgrind.m4:
+ * common/m4/gtk-doc.m4:
+ * common/m4/introspection.m4:
+ * common/m4/pkg.m4:
+ * common/mangle-tmpl.py:
+ * common/plugins.xsl:
+ * common/po.mak:
+ * common/release.mak:
+ * common/scangobj-merge.py:
+ * common/upload.mak:
+ common: Remove static version
+
+2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
+
+ * common/m4/introspection.m4:
+ Update introspection.m4 to match usage
+
+2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * README:
+ README: update
+ Remove old stuff from the README
+
+2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.7 ===
+
+2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.7
+
+2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-ogg.c:
+ test-ogg: remove parsers
+ Remove the parsers, they are not needed anymore as oggdemux now outputs normal
+ buffers with timestamps. Using the parsers also seems to break things.
+
+2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated Vala bindings
+
+2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * common/m4/introspection.m4:
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ Added initial gobject-introspection support
+
+2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: don't use host for shared hash key
+ When we generate the key to share made between connections, don't include the
+ host used to connect so that we can share media even if between clients that
+ connected with localhost and ones with the ip address.
+
+2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * bindings/vala/Makefile.am:
+ build: fix distcheck
+
+2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Update Vala bindings
+
+2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * bindings/vala/Makefile.am:
+ * configure.ac:
+ Fix configure checks and installation location for Vala bindings
+ Fixes bug #628676.
+
+2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.6 ===
+
+2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: release 0.10.6
+
+2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: help the compiler a little
+
+2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ media: cleanup media transport before freeing
+ Cleanup the media transport data before freeing. In particular, remove the qdata
+ from the rtpsource object.
+
+2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media-factory: add eos-shutdown property
+ Add an eos-shutdown property that will send an EOS to the pipeline before
+ shutting it down. This allows for nice cleanup in case of a muxer.
+ Fixes #625597
+
+2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: use multiudpsink send-duplicates when we can
+ If we have a new enough multiudpsink with the send-duplicates property, use this
+ instead of doing our own filtering. Our custom filtering code should eventually
+ be removed when we can depend on a released -good.
+
+2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't leak destinations
+ Refactor and cleanup the destinations array when the stream is destroyed.
+
+2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: don't add udp addresses multiple times
+ Keep track of the udp addresses we added to udpsink and never add the same udp
+ destination twice. This avoids duplicate packets when using multicast.
+
+2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: disable use of SO_LINGER
+ SO_LINGER cause the client to fail to receive a TEARDOWN message because the
+ server close()s the connection.
+
+2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: use 5 second linger period in SO_LINGER
+ Wait 5 seconds before clearing the send buffers and reseting the connection with
+ the client when we do a close. This should be enough time to get the message to
+ the client.
+ See #622757
+
+2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: use SO_LINGER
+ SO_LINGER on the socket will make sure that any pending data on the socket is
+ flushed ASAP and that the socket connection is reset. This makes sure that the
+ socket can be reused immediately.
+ Fixes 622757
+
+2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ README: add blurb about shared media factories
+
+2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Add stdlib.h for atoi()
+
+2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * bindings/python/Makefile.am:
+ * bindings/vala/Makefile.am:
+ build: distcheck fixes
+ Fix 'make distcheck', somewhat (it still fails because it tries to
+ install files into /usr/share/vala/vapi/ irrespective of the
+ configured prefix).
+
+2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump core/base requirements to released version
+ Makes things less confusing for people.
+
+2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: fail if GStreamer core/base requirements are not met
+
+2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve client cleanups
+ Make sure the session does not timeout when using TCP. We need to do this
+ because quicktime player does not send RTCP for some reason in tunneled
+ mode.
+ Refactor some cleanup code.
+ Fixes #612915
+
+2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: add support for prevent session timeouts
+ Add an atomix counter to prevent session timeouts when we are, for example,
+ streaming over TCP.
+
+2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix unlink on session timeouts
+ When our session times out, make sure we unlink all streams in this
+ session.
+ Remove the tunnelid when closing the connection.
+
+2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: small cleanups
+
+2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: handle lost_tunnel callbacks
+ Handle lost_tunnel callbacks and use it to store the tunnelid back into the
+ hashtable so that we can reuse it for when the client reopens the POST
+ socket.
+ Close the connection after a TEARDOWN.
+ Make sure or watchid is cleared when the watch is removed.
+ Fixes #612915
+
+2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-server: add more support for multicast
+
+2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: allow configuration of allowed lower transport
+
+2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp: keep track of server ip and ipv6
+ Keep track of how the client connected to the server and setup the udp ports
+ with the same protocol.
+ Copy the server ip address in the SDP so that clients can send RTCP back to
+ us.
+
+2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: indent
+
+2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use right size for malloc
+
+2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: comment ipv6 server listening address
+
+2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: allow for ipv6 sockets
+
+2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: rework server part
+ Allow setting a bind address, make sure we can deal with ipv6.
+ Remove the port property and change with the service property.
+
+2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.h:
+ media: update comments a little
+
+2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: make content-base better
+ Use the URI formatting functions to make a content-base. Also make sure that
+ there is a trailing / at the end.
+
+2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: guard against invalid paths
+
+2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ test: catch server bind errors
+
+2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtspmedia: emit "unprepared" if _prepare fails.
+ Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
+ media object is removed from its factory's cache.
+
+2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: collect media position when seek completes
+
+2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: call unlink_streams in client finalize
+ Fixes #599027
+
+2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: limit the time to wait to something huge
+ Avoid waiting forever but limit the timeout to 20 seconds.
+
+2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: reindent and check for prepared status
+
+2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ media: avoid doing _get_state() for state changes
+ When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
+ until the media is prerolled or in error. This avoids doing a blocking call of
+ gst_element_get_state() that can cause lockups when there is an error.
+ Fixes #611899
+
+2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: reindent
+
+2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: better error handling
+ Improve the error handling a bit.
+
+2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: rework transport parsing
+ Rework the transport parsing code so that we can ignore transports we don't
+ support instead of just picking the first one we can parse.
+ Configure a (for now hardcoded) destination for multicast transports.
+
+2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: set multicast sink parameters
+ Disable loop and automatic multicast join on the udpsink elements.
+ Add some more debug info.
+ Reset some state variables in the right place.
+ Use the right port numbers for multicast.
+
+2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: handle transport setup correctly
+ Handle UDP, MCAST and TCP transport negotiation more correctly.
+ Store the server session SSRC in the transport.
+
+2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: implement error_full
+ Implement error_full to avoid some segfaults when the rtspconnection calls it.
+ See #608245
+
+2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ docs: update docs and comments
+
+2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: make server work better when behind a proxy
+
+2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
+
+2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ Use GStreamer's debugging subsystem
+
+2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
+
+2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.5 ===
+
+2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.5
+
+2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump required versions
+
+2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: call weak-unref on client->sessions from finalize
+ Fixes bug #596305
+
+2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Fixed crasher where caps got unref'ed too often
+
+2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * configure.ac:
+ * pkgconfig/.gitignore:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
+ Added pkg-config file to use gst-rtsp-server uninstalled
+
+2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: add some docs
+
+2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp: Use gst_rtsp_watch_send_message().
+ Use gst_rtsp_watch_send_message() since the old API which used
+ gst_rtsp_watch_queue_message() has been deprecated.
+
+2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.4 ===
+
+2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ Release 0.10.4
+
+2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp: allocate channels in TCP mode
+ When the client does not provide us with channels in TCP mode, allocate channels
+ ourselves.
+
+2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: don't crash when tunnelid is missing
+ When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
+ don't crash but return an error response to the client.
+ Fixes #589489
+
+2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ bindings: update vala bindings with new method
+
+2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ sessionpool: add function to filter sessions
+ Add generic function to retrieve/remove sessions.
+
+2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump core/base requirements to release
+
+2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix indentation
+
+2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
+
+2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ set state and remove elements of media in for loop
+
+2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ Added gst_rtsp_media_remove_elements function to Vala bindings
+
+2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Added gst_rtsp_media_remove_elements function
+
+2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Don't use name for gstrtpbin so we can add multiple instances to the pipeline
+
+2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated Vala bindings
+
+2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Added vmethod unprepare to GstRTSPMedia
+ The default implementation sets the state of the pipeline to GST_STATE_NULL
+
+2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ Made collect_streams function public
+
+2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ Added vmethod create_pipeline to GstRTSPMediaFactory
+ The pipeline is created in this method and the GstRTSPMedia's element is added to it
+
+2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use g_source_destroy()
+ We need to use g_source_destroy() because we might have added the source to a
+ different main context than the default one.
+
+2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ rtsp: prepare for handling GET/SET_PARAMETER
+ Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
+ is a body now.
+ Fix return codes of handlers.
+
+2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't leak session pads
+
+2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: clean up the messages a bit
+
+2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: warn and skip streams without media
+
+2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ vala: Fixed typo in header file of RTSPMediaStream
+
+2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix message
+ Fix a debug message
+ Make dumping RTCP stats configurable
+
+2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: be less verbose and leak less
+
+2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't leak the destination address
+
+2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp: use RTCP to keep the session alive
+ Use the RTCP rtcp-from stats field to find the associated session and use this
+ to keep the session alive.
+
+2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: add 5sec to the real session timeout
+ Allow the session to live 5sec longer before really timing out. This should give
+ clients some extra time to keep the session active.
+
+2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: replay OK to GET/SET_PARAMETER
+ Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
+ so that we return OK for those requests.
+
+2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: keep track of active transports
+ Keep track of which transport is active to avoid closing the connection too
+ soon.
+ Remove the destination transport also when going to NULL.
+ Print some stats about the SDES and other RTCP messages we receive from the
+ clients.
+
+2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-sdp.c:
+ example: add SDP relay example
+
+2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: also count active TCP connections
+
+2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp: add support for dynamic elements
+ Add support for dynamic elements.
+ Don't set live pipelines back to paused.
+
+2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: don't add encoding name when absent in caps
+
+2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: warn when we can't do RTP-Info
+
+2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: factor out the stream construction
+
+2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: only add RTP-Info when we have the info
+ Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
+ depayloader.
+
+2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.3 ===
+
+2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release: 0.10.3
+ - Fixes a bug where it put the wrong verion in pkgconfig
+ - Link RTP and RTCP sources
+
+2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: link the RTP udpsrc to the session manager
+ Link the RTP udpsrc and the appsrc to the session manager so that they don't
+ shut down when the client sends a packet to open firewalls.
+
+2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * pkgconfig/gst-rtsp-server.pc.in:
+ Don't use hard-coded version number in pkg-config file
+
+2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.2 ===
+
+2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.2
+
+2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * .gitignore:
+ * common/m4/.gitignore:
+ * examples/.gitignore:
+ * pkgconfig/.gitignore:
+ add some .gitignore files
+
+2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: seek to key frames
+
+2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: emit the unprepared signal by id
+ Emit the unprepared signal by id instead of name and set the media as
+ reused.
+
+2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
+
+2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Added finalize function to GstRTPSPServer to unref session pool and media mapping
+
+2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated vala bindings
+
+2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ server: use appsink and appsrc with the API
+ Use the appsink/appsrc API instead of the signals for higher
+ performance.
+
+2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-ogg.c:
+ tests: set the payload type correctly
+
+2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: connect to the unprepare signal
+ Connect to the unprepare signal for non-reusable media so that we can remove
+ them from the cache.
+
+2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal to notify of unprepare
+
+2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: more work on making the media shared
+ Add a reusable flag to medias, indicating that they can be reused after a state
+ change to NULL.
+ Small cleanups.
+
+2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-readme.c:
+ examples: mark the example as shared for testing
+
+2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ client: support shared media
+ Always perform the state actions even if the target state of the pipeline is
+ already correct, we still want to add/remove the transports when we are dealing
+ with shared media.
+ Keep a counter of the number of active transports for a media so that we can use
+ this to perform a state change when needed.
+ Perform a state change of the pipeline only when the first transport was added
+ or when there are no active transports.
+
+2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix refcounting crasher
+ Don't need to remove the weak refs in the finalize methods, they are already
+ removed in the dispose.
+ Don't register the callback with a DestroyNofity.
+
+2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Fix rtsp client refcount management in TCP mode.
+ Don't unref a client ref we never had. Fixes an unref
+ of an already-free client object after a client
+ teardown request for me.
+
+2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ docs: fix typo in API docs
+
+2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ More seeking fixes.
+ Keep the udp sources in playing even if we go to paused. unlock the sources when
+ we shut down.
+ Add some more debug info.
+ Only seek when we need to.
+ Keep track of the position when we go to paused.
+
+2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add beginnings of seeking.
+ Parse the Range header and perform a seek on the pipeline for the requested
+ position. It's disabled currently until I figure out what's going wrong.
+
+2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ allow pause requests for now.
+ --
+
+2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Remove weak ref on the session in teardown
+ We need to remove our weakref from the session when we do a teardown because
+ else we close the TCP connection prematurely.
+
+2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Do some more session cleanup
+ Make session timeout kill the TCP connection that currently watches the
+ session.
+ Remove the client timeout property.
+
+2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add TCP transports
+ Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
+ connection.
+
+2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-launch.c:
+ Add example server that takes launch lines
+ Add an example server that streams any -launch line.
+
+2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-readme.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add support for live streams
+ Add support for live streams and ranges
+ Start on handling TCP data transfer.
+
+2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Free the pipeline before other things
+ ---
+
+2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Only free the pending tunnel if there is one
+ --
+
+2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-server: Add support for tunneling
+ Add support for tunneling over HTTP.
+ Use new connection methods to retrieve the url.
+ Dispatch messages based on the message type instead of blindly
+ assuming it's always a request.
+ Keep track of the watch id so that we can remove it later.
+ Set the media pipeline to NULL before unreffing the pipeline.
+
+2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Fix for channel -> watch rename in gstreamer
+ Rename the RTSPChannel to RTSPWatch and remove an unused variable.
+
+2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Use ASYNC RTSP io
+ Use the async RTSP channels instead of spawning a new thread for each client.
+ If a sessionid is specified in a request, fail if we don't have the session.
+
+2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Add better debug info
+ Add some better debug info.
+
+2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ Time out sessions
+ Add support for session timeouts in the example.
+
+2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ Pass GTimeVal around for performance reasons
+ Get the current time only once and pass it around so that sessions don't have to
+ get the current time anymore.
+ Add experimental support for a GSource that dispatches when the session needs to
+ be cleaned up.
+
+2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add better support for session timeouts
+ Add a method to request the number of milliseconds when a session will timeout.
+
+2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add suport for RTP manager monitoring
+ Add the first stage in monitoring the rtp manager.
+ Make sure we don't update the state to something we don't want.
+
+2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Add support for session keepalive
+ Get and update the session timeout for all requests. get the session as early as
+ possible.
+
+2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Handle media bus messages
+ Handle media bus messages in a custom mainloop and dispatch them to the
+ RTSPMedia objects. Let the default implementation handle some common messages.
+
+2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ Some more session timeout handling
+ Move the session header setting code to a central place so that we always add
+ the timeout parameter too.
+ Handle timeouts by running the session cleanup code.
+ Stop media before cleaning up.
+
+2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Add timeout property
+ Add a timeout property ot the client and make the other properties into GObject
+ properties.
+
+2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Use getters and setters in property code
+ Use the getters and setters for the timeout property instead of locking
+ ourselves.
+
+2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
+
+2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add more timeout stuff
+ Add method to check if a session is expired.
+ Add method to perform cleanup on a session pool.
+
+2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add beginnings of session timeouts and limits
+ Add the timeout value to the Session header for unusual timeout values.
+ Allow us to configure a limit to the amount of active sessions in a pool. Set a
+ limit on the amount of retry we do after a sessionid collision.
+ Add properties to the sessionid and the timeout of a session. Keep track of
+ creation time and last access time for sessions.
+
+2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Cleanup of sessions and more
+ Fix the refcounting of media and sessions in the client. Properly clean up the
+ session data when the client performs a teardown.
+ Add Server header to responses.
+ Allow for multiple uri setups in one session.
+ Add Range header to the PLAY response and add the range attribute to the SDP
+ message.
+ Fix the session pool remove method, it used the wrong key in the hashtable. Also
+ give the ownership of the sessionid to the session object.
+
+2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Rename a variable
+ Rename the 'server_port' variable to simply 'port'.
+
+2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Rework the way we handle transports for streams
+ Make the media accept an array of transports for the streams that we have
+ configured for the play/pause requests.
+ Implement server states for a client and its media.
+ Require 0.10.22.1 (git HEAD) of gstreamer.
+
+2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ Drop const from functions dealing with urls
+ Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
+ have the right const in them.
+
+2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ Fix various leaks
+ Fix some leaks.
+
+2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ More cleanups
+ Don't keep a reference to the GstRTSPMedia in the stream.
+ Free more things when freeing the GstRTSPMedia.
+
+2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ More docs and small cleanups
+ Add some more docs and update the README
+ Cleanup some method names.
+ Remove an unneeded idx field in the GstRTSPMediaStream
+
+2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * examples/Makefile.am:
+ * examples/test-readme.c:
+ Add a README and more example code
+ Add a README file that contains a small introduction on how to use the server
+ along with the example code explained in the readme.
+
+2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ Fix some leaks and change default port
+ Fix some memory leaks by setting the udpsrc elements to the unlocked state after
+ we finished the initial preroll. If we keep them locked, setting the pipeline to
+ NULL will not stop and clean up the sources correctly.
+ Change the default RTSP port to 8554 aka the official alternative RTSP port.
+
+2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Cleanups to the session object
+ Remove some unneeded variables in the session state of a stream such as the
+ owner media and the server transport.
+ Get the configuration of a media stream in a session based on the media_stream
+ in the original object instead of our cached index.
+ Free more data in the finalize method.
+
+2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Cleanups and reuse media from DESCRIBE
+ Handle thread create errors.
+ Rename some internal methods to better match what they actually do.
+ Handle misconfiguration of session_pool and media_mapping gracefully.
+ Cache the DESCRIBE media and uri in the client connection and reuse them when
+ we receive a SETUP request in the same connection for the same uri.
+ Cleanup the client connection object.
+
+2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add shared properties to media and factory
+ Add the shared property to media.
+ Implement some simple caching in the factory depending on if the media is shared
+ or not.
+
+2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Add a little comment
+ Add some comment about the content-base header.
+
+2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Reorganize things, prepare for media sharing
+ Added various other test server examples
+ Move the SDP message generation to a separate helper.
+ Refactor common code for finding the session.
+ Add content-base for realplayer compatibility
+ Clean up request uris before processing for better vlc compatibility.
+ Move prerolling and pipeline construction to the RTSPMedia object.
+ Use multiudpsink for future pipeline reuse.
+
+2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ Back to development
+ Back to 0.10.1.1
+
+=== release 0.10.1 ===
+
+2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ Make 0.10.1 release
+ Release 0.10.1
+
+2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * bindings/vala/Makefile.am:
+ Fix make dist
+ Add more directories and files to the dist.
+
+2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/python/Makefile.am:
+ * bindings/python/rtspserver.override:
+ Fixed compile error of python bindings
+
+2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Marked values as nullable accordingly
+
+2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated Vala bindings
+
+2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ Cleanups and doc updates
+ Add some more documentation and do some minor cleanups here and there.
+
+2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ More improvements
+ Rename GstRTSPMediaBin to GstRTSPMedia
+ Parse the request url into a GstRTSPUri object and pass this object to the
+ various handlers and methods that require the uri.
+
+2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/main.c:
+ Update example
+ Add some more docs and remove some old code from the example.
+
+2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Handle state change failures better
+ Handle state change failures better when changing the state of the pipeline to
+ determine the SDP.
+
+2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ Make element creation more extendible
+ Add get_element vmethod to the default MediaFactory so that subclasses can just
+ override that method and still use the default logic for making a MediaBin from
+ that.
+
+2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/main.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Make the server handle arbitrary pipelines
+ Make GstMediaFactory an object that can instantiate GstMediaBin objects.
+ The GstMediaBin object has a handle to a bin with elements and to a list of
+ GstMediaStream objects that this bin produces.
+ Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
+ with methods to register and remove those mappings.
+ Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
+ used by the server instance.
+ Modify the example application so that it shows how to create custom pipelines
+ attached to a specific mount point.
+ Various misc cleanps.
+
+2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Allow setting a custom media factory for a server
+
+2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Allow setting a custom media factory for a client.
+
+2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Add Makefile entry for the media factory
+
+2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ Add media factory to map urls to media pipeline objects.
+
+2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add comments. Remove unused field
+
+2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ Allow custom session pools to override the session id allocation algorithms Add some comments.
+
+2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.h:
+ Add some comments.
+
+2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Move the connection code in one place Add some comments
+
+2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Make vmethod to create and accept new clients. Add some docs.
+
+2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
+
+2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Name the parameters more appropriately.
+
+2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Do some more cleanup of the session pool.
+
+2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ Check if return value of gst_rtsp_session_get_media is not NULL
+
+2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Install rtsp-session and rtsp-session-pool headers
+
+2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * .gitignore:
+ * Makefile.am:
+ * acinclude.m4:
+ * bindings/python/Makefile.am:
+ * bindings/python/arg-types.py:
+ * bindings/python/codegen/Makefile.am:
+ * bindings/python/codegen/__init__.py:
+ * bindings/python/codegen/argtypes.py:
+ * bindings/python/codegen/code-coverage.py:
+ * bindings/python/codegen/codegen.py:
+ * bindings/python/codegen/definitions.py:
+ * bindings/python/codegen/defsparser.py:
+ * bindings/python/codegen/docextract.py:
+ * bindings/python/codegen/docgen.py:
+ * bindings/python/codegen/fileprefix.override:
+ * bindings/python/codegen/fileprefixmodule.c:
+ * bindings/python/codegen/h2def.py:
+ * bindings/python/codegen/mergedefs.py:
+ * bindings/python/codegen/mkskel.py:
+ * bindings/python/codegen/override.py:
+ * bindings/python/codegen/reversewrapper.py:
+ * bindings/python/codegen/scmexpr.py:
+ * bindings/python/rtspserver-types.defs:
+ * bindings/python/rtspserver.defs:
+ * bindings/python/rtspserver.override:
+ * bindings/python/rtspservermodule.c:
+ * configure.ac:
+ Add python bindings.
+
+2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * bindings/Makefile.am:
+ * configure.ac:
+ Don't go into python dir when requirements for python bindings are missing
+
+2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * bindings/Makefile.am:
+ * bindings/vala/Makefile.am:
+ * configure.ac:
+ Install Vala bindings if vala is available
+
+2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.deps:
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.deps:
+ * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
+ * bindings/vala/packages/gst-rtsp-server-0.10.files:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
+ Regenerated Vala bindings
+
+2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server.vapi:
+ * bindings/vala/packages/gst-rtsp-server.metadata:
+ Fixed typo in included headers for vala bindings
+
+2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * Makefile.am:
+ * configure.ac:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gst-rtsp-server.pc.in:
+ Added pkgconfig file
+
+2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server.vapi:
+ * bindings/vala/packages/gst-rtsp-server.excludes:
+ * bindings/vala/packages/gst-rtsp-server.gi:
+ * bindings/vala/packages/gst-rtsp-server.metadata:
+ Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
+
+2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server.vapi:
+ * bindings/vala/packages/gst-rtsp-server.deps:
+ * bindings/vala/packages/gst-rtsp-server.files:
+ * bindings/vala/packages/gst-rtsp-server.gi:
+ * bindings/vala/packages/gst-rtsp-server.metadata:
+ * bindings/vala/packages/gst-rtsp-server.namespace:
+ Added Vala bindings
+
+2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
+
+2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
+
+ * examples/Makefile.am:
+ * gst/rtsp-server/Makefile.am:
+ Put GStreamer version in library name
+
+2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * gst/rtsp-server/Makefile.am:
+ Fix some issues to pass distcheck
+
+2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Added port property to GstRTSPServer class.
+
+2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * Makefile.am:
+ * autogen.sh:
+ * configure.ac:
+ * examples/Makefile.am:
+ * examples/main.c:
+ * gst/Makefile.am:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * src/Makefile.am:
+ Split in library and example program
+
+2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
+
+ * src/rtsp-client.h:
+ Removed obsolete variable
+
+2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
+
+ * src/rtsp-client.c:
+ * src/rtsp-client.h:
+ Removed pipeline variable GstRTSPClient, because it's only used in one function
+
+2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * src/rtsp-media.c:
+ Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
+
+2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
+
+ * src/rtsp-session.c:
+ Initialize some more vars.
+
+2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
+
+ * src/rtsp-session.c:
+ Initialize variable to avoid compiler warning.
+
+2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
+
+ * .gitignore:
+ Add a reasonable generic .gitignore
+
--- /dev/null
+GStreamer 1.20 Release Notes
+
+GStreamer 1.20 has not been released yet. It is scheduled for release
+around October/November 2021.
+
+1.19.x is the unstable development version that is being developed in
+the git main branch and which will eventually result in 1.20, and 1.19.2
+is the current development release in that series
+
+It is expected that feature freeze will be in early October 2021,
+followed by one or two 1.19.9x pre-releases and the new 1.20 stable
+release around October/November 2021.
+
+1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
+1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+
+See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
+version of this document.
+
+Last updated: Wednesday 22 September 2021, 18:00 UTC (log)
+
+Introduction
+
+The GStreamer team is proud to announce a new major feature release in
+the stable 1.x API series of your favourite cross-platform multimedia
+framework!
+
+As always, this release is again packed with many new features, bug
+fixes and other improvements.
+
+Highlights
+
+- this section will be completed in due course
+
+Major new features and changes
+
+Noteworthy new features and API
+
+- this section will be filled in in due course
+
+New elements
+
+- this section will be filled in in due course
+
+New element features and additions
+
+- this section will be filled in in due course
+
+Plugin and library moves
+
+- this section will be filled in in due course
+
+- There were no plugin moves or library moves in this cycle.
+
+Plugin removals
+
+The following elements or plugins have been removed:
+
+- this section will be filled in in due course
+
+Miscellaneous API additions
+
+- this section will be filled in in due course
+
+Miscellaneous performance, latency and memory optimisations
+
+- this section will be filled in in due course
+
+Miscellaneous other changes and enhancements
+
+- this section will be filled in in due course
+
+Tracing framework and debugging improvements
+
+- this section will be filled in in due course
+
+Tools
+
+- this section will be filled in in due course
+
+GStreamer RTSP server
+
+- this section will be filled in in due course
+
+GStreamer VAAPI
+
+- this section will be filled in in due course
+
+GStreamer OMX
+
+- this section will be filled in in due course
+
+GStreamer Editing Services and NLE
+
+- this section will be filled in in due course
+
+GStreamer validate
+
+- this section will be filled in in due course
+
+GStreamer Python Bindings
+
+- this section will be filled in in due course
+
+GStreamer C# Bindings
+
+- this section will be filled in in due course
+
+GStreamer Rust Bindings and Rust Plugins
+
+The GStreamer Rust bindings are released separately with a different
+release cadence that’s tied to gtk-rs, but the latest release has
+already been updated for the upcoming new GStreamer 1.20 API.
+
+gst-plugins-rs, the module containing GStreamer plugins written in Rust,
+has also seen lots of activity with many new elements and plugins.
+
+What follows is a list of elements and plugins available in
+gst-plugins-rs, so people don’t miss out on all those potentially useful
+elements that have no C equivalent.
+
+- FIXME: add new elements
+
+Rust audio plugins
+
+- audiornnoise: New element for audio denoising which implements the
+ noise removal algorithm of the Xiph RNNoise library, in Rust
+- rsaudioecho: Port of the audioecho element from gst-plugins-good
+ rsaudioloudnorm: Live audio loudness normalization element based on
+ the FFmpeg af_loudnorm filter
+- claxondec: FLAC lossless audio codec decoder element based on the
+ pure-Rust claxon implementation
+- csoundfilter: Audio filter that can use any filter defined via the
+ Csound audio programming language
+- lewtondec: Vorbis audio decoder element based on the pure-Rust
+ lewton implementation
+
+Rust video plugins
+
+- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
+ on a pure-Rust CD+G implementation, used for example by karaoke CDs
+- cea608overlay: CEA-608 Closed Captions overlay element
+- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
+ subtitles) converter
+- tttocea608: CEA-608 Closed Captions from timed-text converter
+- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
+- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
+- dav1dec: AV1 video decoder based on the dav1d decoder implementation
+ by the VLC project
+- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
+ encoder implementation
+- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
+ gst-plugins-good, not feature-equivalent yet
+- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
+ implementations by the image-rs project
+
+Rust text plugins
+
+- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
+ better screen-fitting, including hyphenation support for some
+ languages
+
+Rust network plugins
+
+- reqwesthttpsrc: HTTP(S) source element based on the Rust
+ reqwest/hyper HTTP implementations and almost feature-equivalent
+ with the main GStreamer HTTP source souphttpsrc
+- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
+- awstranscriber: Live audio to timed text transcription element using
+ the Amazon AWS Transcribe API
+
+Generic Rust plugins
+
+- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
+ on libsodium/NaCl
+- togglerecord: Recording element that allows to pause/resume
+ recordings easily and considers keyframe boundaries
+- fallbackswitch/fallbacksrc: Elements for handling potentially
+ failing (network) sources, restarting them on errors/timeout and
+ showing a fallback stream instead
+- threadshare: Set of elements that provide alternatives for various
+ existing GStreamer elements but allow to share the streaming threads
+ between each other to reduce the number of threads
+- rsfilesrc/rsfilesink: File source/sink elements as replacements for
+ the existing filesrc/filesink elements
+
+Build and Dependencies
+
+- this section will be filled in in due course
+
+gst-build
+
+- this section will be filled in in due course
+
+Cerbero
+
+Cerbero is a meta build system used to build GStreamer plus dependencies
+on platforms where dependencies are not readily available, such as
+Windows, Android, iOS and macOS.
+
+General improvements
+
+- this section will be filled in in due course
+
+macOS / iOS
+
+- this section will be filled in in due course
+
+Windows
+
+- this section will be filled in in due course
+
+Windows MSI installer
+
+- this section will be filled in in due course
+
+Linux
+
+- this section will be filled in in due course
+
+Android
+
+- this section will be filled in in due course
+
+Platform-specific changes and improvements
+
+Android
+
+- this section will be filled in in due course
+
+macOS and iOS
+
+- this section will be filled in in due course
+
+Windows
+
+- this section will be filled in in due course
+
+Linux
+
+- this section will be filled in in due course
+
+Documentation improvements
+
+- this section will be filled in in due course
+
+Possibly Breaking Changes
+
+- this section will be filled in in due course
+- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
+- gst_parse_launch() and friends now error out on non-existing
+ properties on top-level bins where they would silently fail and
+ ignore those before.
+
+Known Issues
+
+- this section will be filled in in due course
+
+- There are a couple of known WebRTC-related regressions/blockers:
+
+ - webrtc: DTLS setup with Chrome is broken
+ - webrtcbin: First keyframe is usually lost
+
+Contributors
+
+- this section will be filled in in due course
+
+… and many others who have contributed bug reports, translations, sent
+suggestions or helped testing.
+
+Stable 1.20 branch
+
+After the 1.20.0 release there will be several 1.20.x bug-fix releases
+which will contain bug fixes which have been deemed suitable for a
+stable branch, but no new features or intrusive changes will be added to
+a bug-fix release usually. The 1.20.x bug-fix releases will be made from
+the git 1.20 branch, which will be a stable branch.
+
+1.20.0
+
+1.20.0 is scheduled to be released around October/November 2021.
+
+Schedule for 1.22
+
+Our next major feature release will be 1.22, and 1.21 will be the
+unstable development version leading up to the stable 1.22 release. The
+development of 1.21/1.22 will happen in the git main branch.
+
+The plan for the 1.22 development cycle is yet to be confirmed.
+
+1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
+1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
+
+------------------------------------------------------------------------
+
+These release notes have been prepared by Tim-Philipp Müller with
+contributions from …
+
+License: CC BY-SA 4.0
--- /dev/null
+gst-rtsp-server is a library on top of GStreamer for building an RTSP server
+
+There are some examples in the examples/ directory and more comprehensive
+documentation in docs/README.
--- /dev/null
+This is GStreamer gst-rtsp-server 1.19.2.
+
+GStreamer 1.19 is the development branch leading up to the next major
+stable version which will be 1.20.
+
+The 1.19 development series adds new features on top of the 1.18 series and is
+part of the API and ABI-stable 1.x release series of the GStreamer multimedia
+framework.
+
+Full release notes will one day be found at:
+
+ https://gstreamer.freedesktop.org/releases/1.20/
+
+Binaries for Android, iOS, Mac OS X and Windows will usually be provided
+shortly after the release.
+
+This module will not be very useful by itself and should be used in conjunction
+with other GStreamer modules for a complete multimedia experience.
+
+ - gstreamer: provides the core GStreamer libraries and some generic plugins
+
+ - gst-plugins-base: a basic set of well-supported plugins and additional
+ media-specific GStreamer helper libraries for audio,
+ video, rtsp, rtp, tags, OpenGL, etc.
+
+ - gst-plugins-good: a set of well-supported plugins under our preferred
+ license
+
+ - gst-plugins-ugly: a set of well-supported plugins which might pose
+ problems for distributors
+
+ - gst-plugins-bad: a set of plugins of varying quality that have not made
+ their way into one of core/base/good/ugly yet, for one
+ reason or another. Many of these are are production quality
+ elements, but may still be missing documentation or unit
+ tests; others haven't passed the rigorous quality testing
+ we expect yet.
+
+ - gst-libav: a set of codecs plugins based on the ffmpeg library. This is
+ where you can find audio and video decoders and encoders
+ for a wide variety of formats including H.264, AAC, etc.
+
+ - gstreamer-vaapi: hardware-accelerated video decoding and encoding using
+ VA-API on Linux. Primarily for Intel graphics hardware.
+
+ - gst-omx: hardware-accelerated video decoding and encoding, primarily for
+ embedded Linux systems that provide an OpenMax
+ implementation layer such as the Raspberry Pi.
+
+ - gst-rtsp-server: library to serve files or streaming pipelines via RTSP
+
+ - gst-editing-services: library an plugins for non-linear editing
+
+==== Download ====
+
+You can find source releases of gstreamer in the download
+directory: https://gstreamer.freedesktop.org/src/gstreamer/
+
+The git repository and details how to clone it can be found at
+https://gitlab.freedesktop.org/gstreamer/
+
+==== Homepage ====
+
+The project's website is https://gstreamer.freedesktop.org/
+
+==== Support and Bugs ====
+
+We have recently moved from GNOME Bugzilla to GitLab on freedesktop.org
+for bug reports and feature requests:
+
+ https://gitlab.freedesktop.org/gstreamer
+
+Please submit patches via GitLab as well, in form of Merge Requests. See
+
+ https://gstreamer.freedesktop.org/documentation/contribute/
+
+for more details.
+
+For help and support, please subscribe to and send questions to the
+gstreamer-devel mailing list (see below for details).
+
+There is also a #gstreamer IRC channel on the Freenode IRC network.
+
+==== Developers ====
+
+GStreamer source code repositories can be found on GitLab on freedesktop.org:
+
+ https://gitlab.freedesktop.org/gstreamer
+
+and can also be cloned from there and this is also where you can submit
+Merge Requests or file issues for bugs or feature requests.
+
+Interested developers of the core library, plugins, and applications should
+subscribe to the gstreamer-devel list:
+
+ https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
--- /dev/null
+You need to have GStreamer. You can use an installed version of
+GStreamer or from its build dir.
+
--- /dev/null
+
+ - use a config file to configure the server
+ - error recovery
--- /dev/null
+README
+------
+
+(Last updated on Mon 15 jul 2013, version 0.11.90.1)
+
+This HOWTO describes the basic usage of the GStreamer RTSP libraries and how you
+can build simple server applications with it.
+
+* General
+
+ The server relies heavily on the RTSP infrastructure of GStreamer. This includes
+ all of the media acquisition, decoding, encoding, payloading and UDP/TCP
+ streaming. We use the rtpbin element for all the session management. Most of
+ the RTSP message parsing and construction in the server is done using the RTSP
+ library that comes with gst-plugins-base.
+
+ The result is that the server is rather small (a few 11000 lines of code) and easy
+ to understand and extend. In its current state of development, things change
+ fast, API and ABI are unstable. We encourage people to use it for their various
+ use cases and participate by suggesting changes/features.
+
+ Most of the server is built as a library containing a bunch of GObject objects
+ that provide reasonable default functionality but has a fair amount of hooks
+ to override the default behaviour.
+
+ The server currently integrates with the glib mainloop nicely. It's currently
+ not meant to be used in high-load scenarios and because no security audit has
+ been done, you should probably not put it on a public IP address.
+
+* Initialisation
+
+ You need to initialize GStreamer before using any of the RTSP server functions.
+
+ #include <gst/gst.h>
+
+ int
+ main (int argc, char *argv[])
+ {
+ gst_init (&argc, &argv);
+
+ ...
+ }
+
+ The server itself currently does not have any specific initialisation function
+ but that might change in the future.
+
+
+* Creating the server
+
+ The first thing you want to do is create a new GstRTSPServer object. This object
+ will handle all the new client connections to your server once it is added to a
+ GMainLoop. You can create a new server object like this:
+
+ #include <gst/rtsp-server/rtsp-server.h>
+
+ GstRTSPServer *server;
+
+ server = gst_rtsp_server_new ();
+
+ The server will by default listen on port 8554 for new connections. This can be
+ changed by calling gst_rtsp_server_set_service() or with the 'service' GObject
+ property. This makes it possible to run multiple server instances listening on
+ multiple ports on one machine.
+
+ We can make the server start listening on its default port by attaching it to a
+ mainloop. The following example shows how this is done and will start a server
+ on the default 8554 port. For any request we make, we will get a NOT_FOUND
+ error code because we need to configure more things before the server becomes
+ useful.
+
+ #include <gst/gst.h>
+ #include <gst/rtsp-server/rtsp-server.h>
+
+ int
+ main (int argc, char *argv[])
+ {
+ GstRTSPServer *server;
+ GMainLoop *loop;
+
+ gst_init (&argc, &argv);
+
+ server = gst_rtsp_server_new ();
+
+ /* make a mainloop for the default context */
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_main_loop_run (loop);
+ }
+
+ The server manages four other objects: GstRTSPSessionPool,
+ GstRTSPMountPoints, GstRTSPAuth and GstRTSPThreadPool.
+
+ The GstRTSPSessionPool is an object that keeps track of all the active sessions
+ in the server. A session will usually be kept for each client that performed a
+ SETUP request for a certain media stream. It contains the configuration that
+ the client negotiated with the server to receive the particular stream, ie. the
+ transport used and port pairs for UDP along with the state of the streaming.
+ The default implementation of the session pool is usually sufficient but
+ alternative implementation can be used by the server.
+
+ The GstRTSPMountPoints object is more interesting and needs more configuration
+ before the server object is useful. This object manages the mapping from a
+ request URL to a specific stream and its configuration. We explain in the next
+ topic how to configure this object.
+
+ GstRTSPAuth is an object that authenticates users and authorizes actions
+ performed by users. By default, a server does not have a GstRTSPAuth object and
+ thus does not try to perform any authentication or authorization.
+
+ GstRTSPThreadPool manages the threads used for client connections and media
+ pipelines. The server has a default implementation of a threadpool that should
+ be sufficient in most cases.
+
+
+* Making url mount points
+
+ Next we need to define what media is attached to a particular URL. What we want
+ to achieve is that when the user asks our server for a specific URL, say /test,
+ that we create (or reuse) a GStreamer pipeline that produces one or more RTP
+ streams.
+
+ The object that can create such pipeline is called a GstRTSPMediaFactory object.
+ The default implementation of GstRTSPMediaFactory allows you to easily create
+ GStreamer pipelines using the gst-launch syntax. It is possible to create a
+ GstRTSPMediaFactory subclass that uses different methods for constructing
+ pipelines.
+
+ The default GstRTSPMediaFactory can be configured with a gst-launch line that
+ produces a toplevel bin (use '(' and ')' around the pipeline description to
+ force a toplevel GstBin instead of the default GstPipeline toplevel element).
+ The pipeline description should contain elements named payN, one for each
+ stream (ex. pay0, pay1, ...). Also, for increased compatibility each stream
+ should have a different payload type which can be configured on the payloader.
+
+ The following code snippet illustrates how to create a media factory that
+ creates an RTP feed of an H264 encoded test video signal.
+
+ GstRTSPMediaFactory *factory;
+
+ factory = gst_rtsp_media_factory_new ();
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! x264enc ! rtph264pay pt=96 name=pay0 )");
+
+ Now that we have the media factory, we can attach it to a specific url. To do
+ this we get the default GstRTSPMountPoints from our server and add the url to
+ factory mount points to it like this:
+
+ GstRTSPMountPoints *mounts;
+
+ ...create server..create factory..
+
+ /* get the default mount points from the server */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* attach the video test signal to the "/test" URL */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+ g_object_unref (mounts);
+
+ When starting the server now and directing an RTP client to the URL (like with
+ vlc, mplayer or gstreamer):
+
+ rtsp://localhost:8554/test
+
+ a test signal will be streamed to the client. The full example code can be
+ found in the examples/test-readme.c file.
+
+ Note that by default the factory will create a new pipeline for each client. If
+ you want to share a pipeline between clients, use
+ gst_rtsp_media_factory_set_shared().
+
+
+* more on GstRTSPMediaFactory
+
+ The GstRTSPMediaFactory is responsible for creating and caching GstRTSPMedia
+ objects.
+
+ A freshly created GstRTSPMedia object from the factory initially only contains a
+ GstElement containing the elements to produce the RTP streams for the media and
+ a GPtrArray of GstRTSPStream objects describing the payloader and its source
+ pad. The media is unprepared in this state.
+
+ Usually the url will determine what kind of pipeline should be created. You can
+ for example use query parameters to configure certain parts of the pipeline or
+ select encoders and payloaders based on some url pattern.
+
+ When dealing with a live stream from, for example, a webcam, it can be
+ interesting to share the pipeline with multiple clients. This must be done when
+ only one instance of the video capture element can be used at a time. In this
+ case, the shared property of GstRTSPMedia must be used to instruct the default
+ GstRTSPMediaFactory implementation to cache the media.
+
+ When all objects created from a factory can be shared, you can set the shared
+ property directly on the factory.
+
+* more on GstRTSPMedia
+
+ After creating the GstRTSPMedia object from the factory, it can be prepared
+ with gst_rtsp_media_prepare(). This method will put those objects in a
+ GstPipeline and will construct and link the streaming elements and the
+ rtpbin session manager object.
+
+ The _prepare() method will then preroll the pipeline in order to figure out the
+ caps on the payloaders. After the GstRTSPMedia prerolled it will be in the
+ prepared state and can be used for creating SDP files or for streaming to
+ clients.
+
+ The prepare method will also create 2 UDP ports for each stream that can be
+ used for sending and receiving RTP/RTCP from clients. These port numbers will
+ have to be negotiated with the client in the SETUP requests.
+
+ When preparing a GstRTSPMedia, an appsink and asppsrc is also constructed
+ for streaming the stream over TCP when requested.
+
+ Media is prepared by the server when DESCRIBE or SETUP requests are received
+ from the client.
+
+
+* the GstRTSPClient object
+
+ When a server detects a new client connection on its port, it will accept the
+ connection, check if the connection is allowed and then call the vmethod
+ create_client. The default implementation of this function will create
+ a new GstRTCPClient object, will configure the session pool, mount points,
+ auth and thread pool objects in it.
+
+ The server will then attach the new client to a server mainloop to let it
+ handle further communication with the client. In RTSP it is usual to keep
+ the connection open between multiple RTSP requests. The client watch will
+ be dispatched by the server mainloop when a new GstRTSPMessage is received,
+ which will then be handled and a response will be sent.
+
+ The GstRTSPClient object remains alive for as long as a client has a TCP
+ connection open with the server. Since is possible for a client to open and close
+ the TCP connection between requests, we cannot store the state related
+ to the configured RTSP session in the GstRTSPClient object. This server state
+ is instead stored in the GstRTSPSession object, identified with the session
+ id.
+
+
+* GstRTSPSession
+
+ This object contains state about a specific RTSP session identified with a
+ session id. This state contains the configured streams and their associated
+ transports.
+
+ When a GstRTSPClient performs a SETUP request, the server will allocate a new
+ GstRTSPSession with a unique session id from the GstRTSPSessionPool. The pool
+ maintains a list of all existing sessions and makes sure that no session id is
+ used multiple times. The session id is sent to the client so that the client
+ can refer to its previously configured state by sending the session id in
+ further requests.
+
+ A client will then use the session id to configure one or more
+ GstRTSPSessionMedia objects, identified by their url. This SessionMedia object
+ contains the configuration of a GstRTSPMedia and its configured
+ GstRTSPStreamTransport.
+
+
+* GstRTSPSessionMedia and GstRTSPStreamTransport
+
+ A GstRTSPSessionMedia is identified by a URL and is referenced by a
+ GstRTSPSession. It is created as soon as a client performs a SETUP operation on
+ a particular URL. It will contain a link to the GstRTSPMedia object associated
+ with the URL along with the state of the media and the configured transports
+ for each of the streams in the media.
+
+ Each SETUP request performed by the client will configure a
+ GstRTSPStreamTransport object linked to by the GstRTSPSessionMedia structure.
+ It will contain the transport information needed to send this stream to the
+ client. The GstRTSPStreamTransport also contains a link to the GstRTSPStream
+ object that generates the actual data to be streamed to the client.
+
+ Note how GstRTSPMedia and GstRTSPStream (the providers of the data to
+ stream) are decoupled from GstRTSPSessionMedia and GstRTSPStreamTransport (the
+ configuration of how to send this stream to a client) in order to be able to
+ send the data of one GstRTSPMedia to multiple clients.
+
+
+* media control
+
+ After a client has configured the transports for a GstRTSPMedia and its
+ GstRTSPStreams, the client can play/pause/stop the stream.
+
+ The GstRTSPMedia object was prepared in the DESCRIBE call (or during SETUP when
+ the client skipped the DESCRIBE request). As seen earlier, this configures a
+ couple of udpsink and udpsrc elements to respectively send and receive the
+ media to clients.
+
+ When a client performs a PLAY request, its configured destination UDP ports are
+ added to the GstRTSPStream target destinations, at which point data will
+ be sent to the client. The corresponding GstRTSPMedia object will be set to the
+ PLAYING state if it was not already in order to send the data to the
+ destination.
+
+ The server needs to prepare an RTP-Info header field in the PLAY response,
+ which consists of the sequence number and the RTP timestamp of the next RTP
+ packet. In order to achive this, the server queries the payloaders for this
+ information when it prerolled the pipeline.
+
+ When a client performs a PAUSE request, the destination UDP ports are removed
+ from the GstRTSPStream object and the GstRTSPMedia object is set to PAUSED
+ if no other destinations are configured anymore.
+
+
+* seeking
+
+ A seek is performed when a client sends a Range header in the PLAY request.
+ This only works when not dealing with shared (live) streams.
+
+ The server performs a GStreamer flushing seek on the media, waits for the
+ pipeline to preroll again and then responds to the client after collecting the
+ new RTP sequence number and timestamp from the payloaders.
+
+
+* session management
+
+ The server has to react to clients that suddenly disappear because of network
+ problems or otherwise. It needs to make sure that it can reasonably free the
+ resources that are used by the various objects in use for streaming when the
+ client appears to be gone.
+
+ Each of the GstRTSPSession objects managed by a GstRTSPSessionPool has
+ therefore a last_access field that contains the timestamp of when activity from
+ a client was last recorded.
+
+ Various ways exist to detect activity from a client:
+
+ - RTSP keepalive requests. When a client is receiving RTP data, the RTSP TCP
+ connection is largely unused. It is the client's responsibility to
+ periodically send keep-alive requests over the TCP channel.
+
+ Whenever a keep-alive request is received by the server (any request that
+ contains a session id, usually an OPTION or GET_PARAMETER request) the
+ last_access of the session is updated.
+
+ - Since it is not required for a client to keep the RTSP TCP connection open
+ while streaming, gst-rtsp-server also detects activity from clients by
+ looking at the RTCP messages it receives.
+
+ When an RTCP message is received from a client, the server looks in its list
+ of active ports if this message originates from a known host/port pair that
+ is currently active in a GstRTSPSession. If this is the case, the session is
+ kept alive.
+
+ Since the server does not know anything about the port number that will be
+ used by the client to send RTCP, this method does not always work. Later
+ RTSP RFCs will include support for negotiating this port number with the
+ server. Most clients however use the same port number for sending and
+ receiving RTCP exactly for this reason.
+
+ If there was no activity in a particular session for a long time (by default 60
+ seconds), the application should remove the session from the pool. For this,
+ the application should periodically (say every 2 seconds) check if no sessions
+ expired and call gst_rtsp_session_pool_cleanup() to remove them.
+
+ When a session is removed from the sessionpool and its last reference is
+ unreffed, all related objects and media are destroyed as if a TEARDOWN happened
+ from the client.
+
+
+* TEARDOWN
+
+ A TEARDOWN request will first locate the GstRTSPSessionMedia of the URL. It
+ will then remove all transports from the streams, making sure that streaming
+ stops to the clients. It will then remove the GstRTSPSessionMedia and
+ GstRTSPStreamTransport objects. Finally the GstRTSPSession is released back
+ into the pool.
+
+ When there are no more references to the GstRTSPMedia, the media pipeline is
+ shut down (with _unprepare) and destroyed. This will then also destroy the
+ GstRTSPStream objects.
+
+
+* Security
+
+ The security of the server and the policy is implemented in a GstRTSPAuth
+ object. The object is reponsible for:
+
+ - authenticate the user of the server.
+
+ - check if the current user is authorized to perform an operation.
+
+ For critical operations, the server will call gst_rtsp_auth_check() with
+ a string describing the operation which should be validated. The installed
+ GstRTSPAuth object is then responsible for checking if the operation is
+ allowed.
+
+ Implementations of GstRTSPAuth objects can use the following infrastructure
+ bits of the rtsp server to implement these checks:
+
+ - GstRTSPToken: a generic structure describing roles and permissions granted
+ to a user.
+
+ - GstRTSPPermissions: a generic list of roles and matching permissions. These
+ can be attached to media and factories currently.
+
+ An Auth implementation will usually authenticate a user, using a method such as
+ Basic authentication or client certificates or perhaps simply use the IP address.
+ The result of the authentication of the user will be a GstRTSPToken that is made
+ current in the context of the ongoing request.
+
+ The auth module can then implement the various checks in the server by looking
+ at the current token and, if needed, compare it to the required GstRTSPPermissions
+ of the current object.
+
+ The security is deliberately kept generic with a default implementation of the
+ GstRTSPAuth object providing a usable and simple implementaion. To make more
+ complicated security modules, the auth object should be subclassed and new
+ implementations for the checks needs to be made.
+
+
+Objects
+-------
+
+GstRTSPServer
+ - Toplevel object listening for connections and creating new
+ GstRTSPClient objects
+
+GstRTSPClient
+ - Handle RTSP Requests from connected clients. All other objects
+ are called by this object.
+
+GstRTSPContext
+ - Helper structure containing the current state of the request
+ handled by the client.
+
+
+GstRTSPMountPoints
+ - Maps a url to a GstRTSPMediaFactory implementation. The default
+ implementation uses a simple hashtable to map a url to a factory.
+
+GstRTSPMediaFactory
+ - Creates and caches GstRTSPMedia objects. The default implementation
+ can create GstRTSPMedia objects based on gst-launch syntax.
+
+GstRTSPMediaFactoryURI
+ - Specialized GstRTSPMediaFactory that can stream the content of any
+ URI.
+
+GstRTSPMedia
+ - The object that contains the media pipeline and various GstRTSPStream
+ objects that produce RTP packets
+
+GstRTSPStream
+ - Manages the elements to stream a stream of a GstRTSPMedia to one or
+ more GstRTSPStreamTransports.
+
+
+GstRTSPSessionPool
+ - Creates and manages GstRTSPSession objects identified by an id.
+
+GstRTSPSession
+ - An object containing the various GstRTSPSessionMedia objects managed
+ by this session.
+
+GstRTSPSessionMedia
+ - The state of a GstRTSPMedia and the configuration of a GstRTSPStream
+ objects. The configuration for the GstRTSPStream is stored in
+ GstRTSPStreamTransport objects.
+
+GstRTSPStreamTransport
+ - Configuration of how a GstRTSPStream is send to a particular client. It
+ contains the transport that was negotiated with the client in the SETUP
+ request.
+
+
+GstRTSPSDP
+ - helper functions for creating SDP messages from gstRTSPMedia
+
+GstRTSPAddressPool
+ - a pool of multicast and unicast addresses used in streaming
+
+GstRTSPThreadPool
+ - a pool of threads used for various server tasks such as handling clients and
+ managing media pipelines.
+
+
+GstRTSPAuth
+ - Hooks for checking authorizations, all client activity will call this
+ object with the GstRTSPContext structure. By default it supports
+ basic authentication.
+
+GstRTSPToken
+ - Credentials of a user. This contrains the roles that the user is allowed
+ to assume and other permissions or capabilities of the user.
+
+GstRTSPPermissions
+ - A list of permissions for each role. The permissions are usually attached
+ to objects to describe what roles have what permissions.
+
+GstRTSPParams
+ - object to handle get and set parameter requests.
+
--- /dev/null
+RTSP server
+-----------
+
+This directory contains an example RTSP server built with various GStreamer
+components and libraries. It also uses GStreamer for all of the multimedia
+procesing and RTP bits. The following features are implemented:
+
+ -
+
+Server Design
+-------------
+
+The toplevel component of the server is a GstRTSPServer object. This object
+creates and binds on the server socket and attaches into the mainloop.
+
+For each request a new GstRTSPClient object is created that will accept the
+request and a thread is started to handle further communication with the
+client until the connection is closed.
+
+When a client issues a SETUP request we create a GstRTSPSession object,
+identified with a sessionid, that will keep track of the state of a client.
+The object is destroyed when a TEARDOWN request is made for that sessionid.
+
+We also maintain a pool of URL to media pipeline mappings. Each url is mapped to
+an object that is able to provide a pipeline for that media. We provide
+pipelines to stream live captured data, on-demand file streaming or on-demand
+transcoding of a file or stream.
+
+A pool of currently active pipelines is also maintained. Usually the active
+pipelines are in use by one or more GstRTSPSession objects. An active pipeline
+becomes inactive when no more sessions refer to it.
+
+A client can choose to start a new pipeline or join a currently active pipeline.
+Some active pipeline cannot be joined (such as on-demand streams) but a new
+instance of that pipeline can be created.
--- /dev/null
+{
+ "rtspclientsink": {
+ "description": "RTSP client sink element",
+ "elements": {
+ "rtspclientsink": {
+ "author": "Jan Schmidt <jan@centricular.com>",
+ "description": "Send data over the network via RTSP RECORD(RFC 2326)",
+ "hierarchy": [
+ "GstRTSPClientSink",
+ "GstBin",
+ "GstElement",
+ "GstObject",
+ "GInitiallyUnowned",
+ "GObject"
+ ],
+ "interfaces": [
+ "GstChildProxy",
+ "GstURIHandler"
+ ],
+ "klass": "Sink/Network",
+ "long-name": "RTSP RECORD client",
+ "pad-templates": {
+ "sink_%%u": {
+ "caps": "ANY",
+ "direction": "sink",
+ "presence": "request",
+ "type": "GstRtspClientSinkPad"
+ }
+ },
+ "properties": {
+ "debug": {
+ "blurb": "Dump request and response messages to stdout",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "false",
+ "mutable": "null",
+ "readable": true,
+ "type": "gboolean",
+ "writable": true
+ },
+ "do-rtsp-keep-alive": {
+ "blurb": "Send RTSP keep alive packets, disable for old incompatible server.",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "true",
+ "mutable": "null",
+ "readable": true,
+ "type": "gboolean",
+ "writable": true
+ },
+ "latency": {
+ "blurb": "Amount of ms to buffer",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "2000",
+ "max": "-1",
+ "min": "0",
+ "mutable": "null",
+ "readable": true,
+ "type": "guint",
+ "writable": true
+ },
+ "location": {
+ "blurb": "Location of the RTSP url to read",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "NULL",
+ "mutable": "null",
+ "readable": true,
+ "type": "gchararray",
+ "writable": true
+ },
+ "multicast-iface": {
+ "blurb": "The network interface on which to join the multicast group",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "NULL",
+ "mutable": "null",
+ "readable": true,
+ "type": "gchararray",
+ "writable": true
+ },
+ "ntp-time-source": {
+ "blurb": "NTP time source for RTCP packets",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "ntp (0)",
+ "mutable": "null",
+ "readable": true,
+ "type": "GstRTSPClientSinkNtpTimeSource",
+ "writable": true
+ },
+ "port-range": {
+ "blurb": "Client port range that can be used to receive RTCP data, eg. 3000-3005 (NULL = no restrictions)",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "NULL",
+ "mutable": "null",
+ "readable": true,
+ "type": "gchararray",
+ "writable": true
+ },
+ "profiles": {
+ "blurb": "Allowed RTSP profiles",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "avp",
+ "mutable": "null",
+ "readable": true,
+ "type": "GstRTSPProfile",
+ "writable": true
+ },
+ "protocols": {
+ "blurb": "Allowed lower transport protocols",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "tcp+udp-mcast+udp",
+ "mutable": "null",
+ "readable": true,
+ "type": "GstRTSPLowerTrans",
+ "writable": true
+ },
+ "proxy": {
+ "blurb": "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "NULL",
+ "mutable": "null",
+ "readable": true,
+ "type": "gchararray",
+ "writable": true
+ },
+ "proxy-id": {
+ "blurb": "HTTP proxy URI user id for authentication",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "NULL",
+ "mutable": "null",
+ "readable": true,
+ "type": "gchararray",
+ "writable": true
+ },
+ "proxy-pw": {
+ "blurb": "HTTP proxy URI user password for authentication",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "NULL",
+ "mutable": "null",
+ "readable": true,
+ "type": "gchararray",
+ "writable": true
+ },
+ "retry": {
+ "blurb": "Max number of retries when allocating RTP ports.",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "20",
+ "max": "65535",
+ "min": "0",
+ "mutable": "null",
+ "readable": true,
+ "type": "guint",
+ "writable": true
+ },
+ "rtp-blocksize": {
+ "blurb": "RTP package size to suggest to server (0 = disabled)",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "0",
+ "max": "65536",
+ "min": "0",
+ "mutable": "null",
+ "readable": true,
+ "type": "guint",
+ "writable": true
+ },
+ "rtx-time": {
+ "blurb": "Amount of ms to buffer for retransmission. 0 disables retransmission",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "500",
+ "max": "-1",
+ "min": "0",
+ "mutable": "null",
+ "readable": true,
+ "type": "guint",
+ "writable": true
+ },
+ "sdes": {
+ "blurb": "The SDES items of this session",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "mutable": "null",
+ "readable": true,
+ "type": "GstStructure",
+ "writable": true
+ },
+ "tcp-timeout": {
+ "blurb": "Fail after timeout microseconds on TCP connections (0 = disabled)",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "20000000",
+ "max": "18446744073709551615",
+ "min": "0",
+ "mutable": "null",
+ "readable": true,
+ "type": "guint64",
+ "writable": true
+ },
+ "timeout": {
+ "blurb": "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "5000000",
+ "max": "18446744073709551615",
+ "min": "0",
+ "mutable": "null",
+ "readable": true,
+ "type": "guint64",
+ "writable": true
+ },
+ "tls-database": {
+ "blurb": "TLS database with anchor certificate authorities used to validate the server certificate",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "mutable": "null",
+ "readable": true,
+ "type": "GTlsDatabase",
+ "writable": true
+ },
+ "tls-interaction": {
+ "blurb": "A GTlsInteraction object to prompt the user for password or certificate",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "mutable": "null",
+ "readable": true,
+ "type": "GTlsInteraction",
+ "writable": true
+ },
+ "tls-validation-flags": {
+ "blurb": "TLS certificate validation flags used to validate the server certificate",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "validate-all",
+ "mutable": "null",
+ "readable": true,
+ "type": "GTlsCertificateFlags",
+ "writable": true
+ },
+ "udp-buffer-size": {
+ "blurb": "Size of the kernel UDP receive buffer in bytes, 0=default",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "524288",
+ "max": "2147483647",
+ "min": "0",
+ "mutable": "null",
+ "readable": true,
+ "type": "gint",
+ "writable": true
+ },
+ "udp-reconnect": {
+ "blurb": "Reconnect to the server if RTSP connection is closed when doing UDP",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "true",
+ "mutable": "null",
+ "readable": true,
+ "type": "gboolean",
+ "writable": true
+ },
+ "user-agent": {
+ "blurb": "The User-Agent string to send to the server",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "GStreamer/1.19.2",
+ "mutable": "null",
+ "readable": true,
+ "type": "gchararray",
+ "writable": true
+ },
+ "user-id": {
+ "blurb": "RTSP location URI user id for authentication",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "NULL",
+ "mutable": "null",
+ "readable": true,
+ "type": "gchararray",
+ "writable": true
+ },
+ "user-pw": {
+ "blurb": "RTSP location URI user password for authentication",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "NULL",
+ "mutable": "null",
+ "readable": true,
+ "type": "gchararray",
+ "writable": true
+ }
+ },
+ "rank": "none",
+ "signals": {
+ "accept-certificate": {
+ "args": [
+ {
+ "name": "arg0",
+ "type": "GTlsConnection"
+ },
+ {
+ "name": "arg1",
+ "type": "GTlsCertificate"
+ },
+ {
+ "name": "arg2",
+ "type": "GTlsCertificateFlags"
+ }
+ ],
+ "return-type": "gboolean",
+ "when": "last"
+ },
+ "handle-request": {
+ "args": [
+ {
+ "name": "arg0",
+ "type": "GstRTSPMessage"
+ },
+ {
+ "name": "arg1",
+ "type": "GstRTSPMessage"
+ }
+ ],
+ "return-type": "void"
+ },
+ "new-manager": {
+ "args": [
+ {
+ "name": "arg0",
+ "type": "GstElement"
+ }
+ ],
+ "return-type": "void",
+ "when": "first"
+ },
+ "new-payloader": {
+ "args": [
+ {
+ "name": "arg0",
+ "type": "GstElement"
+ }
+ ],
+ "return-type": "void",
+ "when": "first"
+ },
+ "request-rtcp-key": {
+ "args": [
+ {
+ "name": "arg0",
+ "type": "guint"
+ }
+ ],
+ "return-type": "GstCaps",
+ "when": "last"
+ },
+ "update-sdp": {
+ "args": [
+ {
+ "name": "arg0",
+ "type": "GstSDPMessage"
+ }
+ ],
+ "return-type": "void"
+ }
+ }
+ }
+ },
+ "filename": "gstrtspclientsink",
+ "license": "LGPL",
+ "other-types": {
+ "GstRTSPClientSinkNtpTimeSource": {
+ "kind": "enum",
+ "values": [
+ {
+ "desc": "NTP time based on realtime clock",
+ "name": "ntp",
+ "value": "0"
+ },
+ {
+ "desc": "UNIX time based on realtime clock",
+ "name": "unix",
+ "value": "1"
+ },
+ {
+ "desc": "Running time based on pipeline clock",
+ "name": "running-time",
+ "value": "2"
+ },
+ {
+ "desc": "Pipeline clock time",
+ "name": "clock-time",
+ "value": "3"
+ }
+ ]
+ },
+ "GstRtspClientSinkPad": {
+ "hierarchy": [
+ "GstRtspClientSinkPad",
+ "GstGhostPad",
+ "GstProxyPad",
+ "GstPad",
+ "GstObject",
+ "GInitiallyUnowned",
+ "GObject"
+ ],
+ "kind": "object",
+ "properties": {
+ "payloader": {
+ "blurb": "The payloader element to use (NULL = default automatically selected)",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "mutable": "null",
+ "readable": true,
+ "type": "GstElement",
+ "writable": true
+ },
+ "ulpfec-percentage": {
+ "blurb": "The percentage of ULP redundancy to apply",
+ "conditionally-available": false,
+ "construct": false,
+ "construct-only": false,
+ "controllable": false,
+ "default": "0",
+ "max": "100",
+ "min": "0",
+ "mutable": "null",
+ "readable": true,
+ "type": "guint",
+ "writable": true
+ }
+ }
+ }
+ },
+ "package": "GStreamer RTSP Server Library",
+ "source": "gst-rtsp-server",
+ "tracers": {},
+ "url": "Unknown package origin"
+ }
+}
\ No newline at end of file
--- /dev/null
+# GStreamer RTSP Server
--- /dev/null
+build_hotdoc = false
+
+if meson.is_cross_build()
+ if get_option('doc').enabled()
+ error('Documentation enabled but building the doc while cross building is not supported yet.')
+ endif
+
+ message('Documentation not built as building it while cross building is not supported yet.')
+ subdir_done()
+endif
+
+required_hotdoc_extensions = ['gi-extension', 'gst-extension']
+if gst_dep.type_name() == 'internal'
+ gst_proj = subproject('gstreamer')
+ plugins_cache_generator = gst_proj.get_variable('plugins_cache_generator')
+else
+ required_hotdoc_extensions += ['gst-extension']
+ plugins_cache_generator = find_program(join_paths(gst_dep.get_pkgconfig_variable('libexecdir'), 'gstreamer-' + api_version, 'gst-plugins-doc-cache-generator'),
+ required: false)
+endif
+
+plugins_cache = join_paths(meson.current_source_dir(), 'gst_plugins_cache.json')
+if plugins.length() == 0
+ message('All rtsp-server plugins have been disabled')
+elif plugins_cache_generator.found()
+ plugins_doc_dep = custom_target('rtsp-server-plugins-doc-cache',
+ command: [plugins_cache_generator, plugins_cache, '@OUTPUT@', '@INPUT@'],
+ input: plugins,
+ output: 'gst_plugins_cache.json',
+ )
+else
+ warning('GStreamer plugin inspector for documentation not found, can\'t update the cache')
+endif
+
+hotdoc_p = find_program('hotdoc', required: get_option('doc'))
+if not hotdoc_p.found()
+ message('Hotdoc not found, not building the documentation')
+ subdir_done()
+endif
+
+hotdoc_req = '>= 0.11.0'
+hotdoc_version = run_command(hotdoc_p, '--version').stdout()
+if not hotdoc_version.version_compare(hotdoc_req)
+ if get_option('doc').enabled()
+ error('Hotdoc version @0@ not found, got @1@'.format(hotdoc_req, hotdoc_version))
+ else
+ message('Hotdoc version @0@ not found, got @1@'.format(hotdoc_req, hotdoc_version))
+ subdir_done()
+ endif
+endif
+
+hotdoc = import('hotdoc')
+foreach extension: required_hotdoc_extensions
+ if not hotdoc.has_extensions(extension)
+ if get_option('doc').enabled()
+ error('Documentation enabled but @0@ missing'.format(extension))
+ endif
+
+ message('@0@ extension not found, not building documentation'.format(extension))
+ subdir_done()
+ endif
+endforeach
+
+if not build_gir
+ if get_option('doc').enabled()
+ error('Documentation enabled but introspection not built.')
+ endif
+
+ message('Introspection not built, can\'t build the documentation')
+ subdir_done()
+endif
+
+build_hotdoc = true
+hotdoc = import('hotdoc')
+
+libs_doc = [hotdoc.generate_doc('gst-rtsp-server',
+ project_version: api_version,
+ gi_c_sources: ['../gst/rtsp-server/*.[hc]'],
+ gi_sources: rtsp_server_gir[0].full_path(),
+ sitemap: 'sitemap.txt',
+ index: 'index.md',
+ gi_index: 'index.md',
+ gi_smart_index: true,
+ gi_order_generated_subpages: true,
+)]
+
+plugins_doc = [hotdoc.generate_doc('rtspclientsink',
+ project_version: api_version,
+ sitemap: 'plugin-sitemap.txt',
+ index: 'plugin-index.md',
+ gst_index: 'plugin-index.md',
+ gst_c_sources: ['../gst/rtsp-sink/*.[ch]'],
+ gst_dl_sources: [rtspsink.full_path()],
+ gst_smart_index: true,
+ dependencies: gst_rtsp_server_deps + [rtspsink],
+ gst_cache_file: plugins_cache,
+ gst_plugin_name: 'rtspclientsink',
+)]
+doc = libs_doc[0]
--- /dev/null
+# rtspclientsink
--- /dev/null
+gi-index
+ gst-index
--- /dev/null
+examples = [
+ 'test-appsrc',
+ 'test-appsrc2',
+ 'test-auth',
+ 'test-auth-digest',
+ 'test-launch',
+ 'test-mp4',
+ 'test-multicast2',
+ 'test-multicast',
+ 'test-netclock',
+ 'test-netclock-client',
+ 'test-ogg',
+ 'test-onvif-client',
+ 'test-onvif-server',
+ 'test-readme',
+ 'test-record-auth',
+ 'test-record',
+ 'test-replay-server',
+ 'test-sdp',
+ 'test-uri',
+ 'test-video',
+ 'test-video-rtx',
+]
+
+foreach example : examples
+ executable(example, '@0@.c'.format(example),
+ c_args : rtspserver_args,
+ include_directories : rtspserver_incs,
+ dependencies : [glib_dep, gst_dep, gstapp_dep, gstnet_dep, gst_rtsp_server_dep],
+ install: false)
+endforeach
+
+cgroup_dep = dependency('libcgroup', version : '>= 0.26', required : false)
+if cgroup_dep.found()
+ executable('test-cgroups', 'test-cgroups.c',
+ c_args : rtspserver_args,
+ include_directories : rtspserver_incs,
+ dependencies : [glib_dep, gst_dep, gstnet_dep, gst_rtsp_server_dep, cgroup_dep],
+ install: false)
+endif
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+typedef struct
+{
+ gboolean white;
+ GstClockTime timestamp;
+} MyContext;
+
+/* called when we need to give data to appsrc */
+static void
+need_data (GstElement * appsrc, guint unused, MyContext * ctx)
+{
+ GstBuffer *buffer;
+ guint size;
+ GstFlowReturn ret;
+
+ size = 385 * 288 * 2;
+
+ buffer = gst_buffer_new_allocate (NULL, size, NULL);
+
+ /* this makes the image black/white */
+ gst_buffer_memset (buffer, 0, ctx->white ? 0xff : 0x0, size);
+
+ ctx->white = !ctx->white;
+
+ /* increment the timestamp every 1/2 second */
+ GST_BUFFER_PTS (buffer) = ctx->timestamp;
+ GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND, 2);
+ ctx->timestamp += GST_BUFFER_DURATION (buffer);
+
+ g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
+ gst_buffer_unref (buffer);
+}
+
+/* called when a new media pipeline is constructed. We can query the
+ * pipeline and configure our appsrc */
+static void
+media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media,
+ gpointer user_data)
+{
+ GstElement *element, *appsrc;
+ MyContext *ctx;
+
+ /* get the element used for providing the streams of the media */
+ element = gst_rtsp_media_get_element (media);
+
+ /* get our appsrc, we named it 'mysrc' with the name property */
+ appsrc = gst_bin_get_by_name_recurse_up (GST_BIN (element), "mysrc");
+
+ /* this instructs appsrc that we will be dealing with timed buffer */
+ gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
+ /* configure the caps of the video */
+ g_object_set (G_OBJECT (appsrc), "caps",
+ gst_caps_new_simple ("video/x-raw",
+ "format", G_TYPE_STRING, "RGB16",
+ "width", G_TYPE_INT, 384,
+ "height", G_TYPE_INT, 288,
+ "framerate", GST_TYPE_FRACTION, 0, 1, NULL), NULL);
+
+ ctx = g_new0 (MyContext, 1);
+ ctx->white = FALSE;
+ ctx->timestamp = 0;
+ /* make sure ther datais freed when the media is gone */
+ g_object_set_data_full (G_OBJECT (media), "my-extra-data", ctx,
+ (GDestroyNotify) g_free);
+
+ /* install the callback that will be called when a buffer is needed */
+ g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
+ gst_object_unref (appsrc);
+ gst_object_unref (element);
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "( appsrc name=mysrc ! videoconvert ! x264enc ! rtph264pay name=pay0 pt=96 )");
+
+ /* notify when our media is ready, This is called whenever someone asks for
+ * the media and a new pipeline with our appsrc is created */
+ g_signal_connect (factory, "media-configure", (GCallback) media_configure,
+ NULL);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mounts anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/app/app.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+typedef struct
+{
+ GstElement *generator_pipe;
+ GstElement *vid_appsink;
+ GstElement *vid_appsrc;
+ GstElement *aud_appsink;
+ GstElement *aud_appsrc;
+} MyContext;
+
+/* called when we need to give data to an appsrc */
+static void
+need_data (GstElement * appsrc, guint unused, MyContext * ctx)
+{
+ GstSample *sample;
+ GstFlowReturn ret;
+
+ if (appsrc == ctx->vid_appsrc)
+ sample = gst_app_sink_pull_sample (GST_APP_SINK (ctx->vid_appsink));
+ else
+ sample = gst_app_sink_pull_sample (GST_APP_SINK (ctx->aud_appsink));
+
+ if (sample) {
+ GstBuffer *buffer = gst_sample_get_buffer (sample);
+ GstSegment *seg = gst_sample_get_segment (sample);
+ GstClockTime pts, dts;
+
+ /* Convert the PTS/DTS to running time so they start from 0 */
+ pts = GST_BUFFER_PTS (buffer);
+ if (GST_CLOCK_TIME_IS_VALID (pts))
+ pts = gst_segment_to_running_time (seg, GST_FORMAT_TIME, pts);
+
+ dts = GST_BUFFER_DTS (buffer);
+ if (GST_CLOCK_TIME_IS_VALID (dts))
+ dts = gst_segment_to_running_time (seg, GST_FORMAT_TIME, dts);
+
+ if (buffer) {
+ /* Make writable so we can adjust the timestamps */
+ buffer = gst_buffer_copy (buffer);
+ GST_BUFFER_PTS (buffer) = pts;
+ GST_BUFFER_DTS (buffer) = dts;
+ g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
+ }
+
+ /* we don't need the appsink sample anymore */
+ gst_sample_unref (sample);
+ }
+}
+
+static void
+ctx_free (MyContext * ctx)
+{
+ gst_element_set_state (ctx->generator_pipe, GST_STATE_NULL);
+
+ gst_object_unref (ctx->generator_pipe);
+ gst_object_unref (ctx->vid_appsrc);
+ gst_object_unref (ctx->vid_appsink);
+ gst_object_unref (ctx->aud_appsrc);
+ gst_object_unref (ctx->aud_appsink);
+
+ g_free (ctx);
+}
+
+/* called when a new media pipeline is constructed. We can query the
+ * pipeline and configure our appsrc */
+static void
+media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media,
+ gpointer user_data)
+{
+ GstElement *element, *appsrc, *appsink;
+ GstCaps *caps;
+ MyContext *ctx;
+
+ ctx = g_new0 (MyContext, 1);
+ /* This pipeline generates H264 video and PCM audio. The appsinks are kept small so that if delivery is slow,
+ * encoded buffers are dropped as needed. There's slightly more buffers (32) allowed for audio */
+ ctx->generator_pipe =
+ gst_parse_launch
+ ("videotestsrc is-live=true ! x264enc speed-preset=superfast tune=zerolatency ! h264parse ! appsink name=vid max-buffers=1 drop=true "
+ "audiotestsrc is-live=true ! appsink name=aud max-buffers=32 drop=true",
+ NULL);
+
+ /* make sure the data is freed when the media is gone */
+ g_object_set_data_full (G_OBJECT (media), "rtsp-extra-data", ctx,
+ (GDestroyNotify) ctx_free);
+
+ /* get the element (bin) used for providing the streams of the media */
+ element = gst_rtsp_media_get_element (media);
+
+ /* Find the 2 app sources (video / audio), and configure them, connect to the
+ * signals to request data */
+ /* configure the caps of the video */
+ caps = gst_caps_new_simple ("video/x-h264",
+ "stream-format", G_TYPE_STRING, "byte-stream",
+ "alignment", G_TYPE_STRING, "au",
+ "width", G_TYPE_INT, 384, "height", G_TYPE_INT, 288,
+ "framerate", GST_TYPE_FRACTION, 15, 1, NULL);
+ ctx->vid_appsrc = appsrc =
+ gst_bin_get_by_name_recurse_up (GST_BIN (element), "videosrc");
+ ctx->vid_appsink = appsink =
+ gst_bin_get_by_name (GST_BIN (ctx->generator_pipe), "vid");
+ gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
+ g_object_set (G_OBJECT (appsrc), "caps", caps, NULL);
+ g_object_set (G_OBJECT (appsink), "caps", caps, NULL);
+ /* install the callback that will be called when a buffer is needed */
+ g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
+ gst_caps_unref (caps);
+
+ caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S24BE",
+ "layout", G_TYPE_STRING, "interleaved", "rate", G_TYPE_INT, 48000,
+ "channels", G_TYPE_INT, 2, NULL);
+ ctx->aud_appsrc = appsrc =
+ gst_bin_get_by_name_recurse_up (GST_BIN (element), "audiosrc");
+ ctx->aud_appsink = appsink =
+ gst_bin_get_by_name (GST_BIN (ctx->generator_pipe), "aud");
+ gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
+ g_object_set (G_OBJECT (appsrc), "caps", caps, NULL);
+ g_object_set (G_OBJECT (appsink), "caps", caps, NULL);
+ g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
+ gst_caps_unref (caps);
+
+ gst_element_set_state (ctx->generator_pipe, GST_STATE_PLAYING);
+ gst_object_unref (element);
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "( appsrc name=videosrc ! h264parse ! rtph264pay name=pay0 pt=96 "
+ " appsrc name=audiosrc ! audioconvert ! rtpL24pay name=pay1 pt=97 )");
+
+ /* notify when our media is ready, This is called whenever someone asks for
+ * the media and a new pipeline with our appsrc is created */
+ g_signal_connect (factory, "media-configure", (GCallback) media_configure,
+ NULL);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mounts anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+static gchar *htdigest_path = NULL;
+static gchar *realm = NULL;
+
+static GOptionEntry entries[] = {
+ {"htdigest-path", 'h', 0, G_OPTION_ARG_STRING, &htdigest_path,
+ "Path to an htdigest file to parse (default: None)", "PATH"},
+ {"realm", 'r', 0, G_OPTION_ARG_STRING, &realm,
+ "Authentication realm (default: None)", "REALM"},
+ {NULL}
+};
+
+
+static gboolean
+remove_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
+ GstRTSPServer * server)
+{
+ return GST_RTSP_FILTER_REMOVE;
+}
+
+static gboolean
+remove_sessions (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ g_print ("removing all sessions\n");
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_filter (pool,
+ (GstRTSPSessionPoolFilterFunc) remove_func, server);
+ g_object_unref (pool);
+
+ return FALSE;
+}
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ GOptionContext *optctx;
+ GError *error = NULL;
+
+ optctx = g_option_context_new (NULL);
+ g_option_context_add_main_entries (optctx, entries, NULL);
+ g_option_context_add_group (optctx, gst_init_get_option_group ());
+ if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
+ g_printerr ("Error parsing options: %s\n", error->message);
+ g_option_context_free (optctx);
+ g_clear_error (&error);
+ return -1;
+ }
+ g_option_context_free (optctx);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mounts for this server, every server has a default mapper object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* allow user and admin to access this resource */
+ gst_rtsp_media_factory_add_role (factory, "user",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_media_factory_add_role (factory, "admin",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+ /* admin2 can look at the media but not construct so he gets a
+ * 401 Unauthorized */
+ gst_rtsp_media_factory_add_role (factory, "admin2",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, FALSE, NULL);
+ /* Anonymous user can do the same things as admin2 on this resource */
+ gst_rtsp_media_factory_add_role (factory, "anonymous",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, FALSE, NULL);
+
+ /* make another factory */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=30/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 )");
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test2", factory);
+
+ /* allow admin2 to access this resource */
+ /* user and admin have no permissions so they can't even see the
+ * media and get a 404 Not Found */
+ gst_rtsp_media_factory_add_role (factory, "admin2",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* make a new authentication manager */
+ auth = gst_rtsp_auth_new ();
+ gst_rtsp_auth_set_supported_methods (auth, GST_RTSP_AUTH_DIGEST);
+
+ /* make default token, it has no permissions */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "anonymous", NULL);
+ gst_rtsp_auth_set_default_token (auth, token);
+ gst_rtsp_token_unref (token);
+
+ /* make user token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_auth_add_digest (auth, "user", "password", token);
+ gst_rtsp_token_unref (token);
+
+ if (htdigest_path) {
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+
+ if (!gst_rtsp_auth_parse_htdigest (auth, htdigest_path, token)) {
+ g_printerr ("Could not parse htdigest at %s\n", htdigest_path);
+ gst_rtsp_token_unref (token);
+ goto failed;
+ }
+
+ gst_rtsp_token_unref (token);
+ }
+
+ if (realm)
+ gst_rtsp_auth_set_realm (auth, realm);
+
+ /* make admin token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "admin", NULL);
+ gst_rtsp_auth_add_digest (auth, "admin", "power", token);
+ gst_rtsp_token_unref (token);
+
+ /* make admin2 token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "admin2", NULL);
+ gst_rtsp_auth_add_digest (auth, "admin2", "power2", token);
+ gst_rtsp_token_unref (token);
+
+ /* set as the server authentication manager */
+ gst_rtsp_server_set_auth (server, auth);
+ g_object_unref (auth);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+ g_timeout_add_seconds (10, (GSourceFunc) remove_sessions, server);
+
+ /* start serving */
+ g_print ("stream with user:password ready at rtsp://127.0.0.1:8554/test\n");
+ g_print ("stream with admin:power ready at rtsp://127.0.0.1:8554/test\n");
+ g_print ("stream with admin2:power2 ready at rtsp://127.0.0.1:8554/test2\n");
+
+ if (htdigest_path)
+ g_print
+ ("stream with htdigest users ready at rtsp://127.0.0.1:8554/test\n");
+
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+static gboolean
+remove_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
+ GstRTSPServer * server)
+{
+ return GST_RTSP_FILTER_REMOVE;
+}
+
+static gboolean
+remove_sessions (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ g_print ("removing all sessions\n");
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_filter (pool,
+ (GstRTSPSessionPoolFilterFunc) remove_func, server);
+ g_object_unref (pool);
+
+ return FALSE;
+}
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ gchar *basic;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mounts for this server, every server has a default mapper object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* allow user and admin to access this resource */
+ gst_rtsp_media_factory_add_role (factory, "user",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_media_factory_add_role (factory, "admin",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+ /* admin2 can look at the media but not construct so he gets a
+ * 401 Unauthorized */
+ gst_rtsp_media_factory_add_role (factory, "admin2",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, FALSE, NULL);
+ /* Anonymous user can do the same things as admin2 on this resource */
+ gst_rtsp_media_factory_add_role (factory, "anonymous",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, FALSE, NULL);
+
+ /* make another factory */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=30/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 )");
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test2", factory);
+
+ /* allow admin2 to access this resource */
+ /* user and admin have no permissions so they can't even see the
+ * media and get a 404 Not Found */
+ gst_rtsp_media_factory_add_role (factory, "admin2",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* make a new authentication manager */
+ auth = gst_rtsp_auth_new ();
+
+ /* make default token, it has no permissions */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "anonymous", NULL);
+ gst_rtsp_auth_set_default_token (auth, token);
+ gst_rtsp_token_unref (token);
+
+ /* make user token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ basic = gst_rtsp_auth_make_basic ("user", "password");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* make admin token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "admin", NULL);
+ basic = gst_rtsp_auth_make_basic ("admin", "power");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* make admin2 token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "admin2", NULL);
+ basic = gst_rtsp_auth_make_basic ("admin2", "power2");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* set as the server authentication manager */
+ gst_rtsp_server_set_auth (server, auth);
+ g_object_unref (auth);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+ g_timeout_add_seconds (10, (GSourceFunc) remove_sessions, server);
+
+ /* start serving */
+ g_print ("stream with user:password ready at rtsp://127.0.0.1:8554/test\n");
+ g_print ("stream with admin:power ready at rtsp://127.0.0.1:8554/test\n");
+ g_print ("stream with admin2:power2 ready at rtsp://127.0.0.1:8554/test2\n");
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/* Runs a pipeline and clasifies the media pipelines based on the
+ * authenticated user.
+ *
+ * This test requires 2 cpu cgroups to exist named 'user' and 'admin'.
+ * The rtsp server should have permission to add its threads to the
+ * cgroups.
+ *
+ * sudo cgcreate -t uid:gid -g cpu:/user
+ * sudo cgcreate -t uid:gid -g cpu:/admin
+ *
+ * With -t you can give the user and group access to the task file to
+ * write the thread ids. The user running the server can be used.
+ *
+ * Then you would want to change the cpu shares assigned to each group:
+ *
+ * sudo cgset -r cpu.shares=100 user
+ * sudo cgset -r cpu.shares=1024 admin
+ *
+ * Then start clients for 'user' until the stream is degraded because of
+ * lack of CPU. Then start a client for 'admin' and check that the stream
+ * is not degraded.
+ */
+
+#include <libcgroup.h>
+
+#include <gst/gst.h>
+#include <gst/rtsp-server/rtsp-server.h>
+
+typedef struct _GstRTSPCGroupPool GstRTSPCGroupPool;
+typedef struct _GstRTSPCGroupPoolClass GstRTSPCGroupPoolClass;
+
+#define GST_TYPE_RTSP_CGROUP_POOL (gst_rtsp_cgroup_pool_get_type ())
+#define GST_IS_RTSP_CGROUP_POOL(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CGROUP_POOL))
+#define GST_IS_RTSP_CGROUP_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CGROUP_POOL))
+#define GST_RTSP_CGROUP_POOL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CGROUP_POOL, GstRTSPCGroupPoolClass))
+#define GST_RTSP_CGROUP_POOL(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CGROUP_POOL, GstRTSPCGroupPool))
+#define GST_RTSP_CGROUP_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CGROUP_POOL, GstRTSPCGroupPoolClass))
+#define GST_RTSP_CGROUP_POOL_CAST(obj) ((GstRTSPCGroupPool*)(obj))
+#define GST_RTSP_CGROUP_POOL_CLASS_CAST(klass) ((GstRTSPCGroupPoolClass*)(klass))
+
+struct _GstRTSPCGroupPool
+{
+ GstRTSPThreadPool parent;
+
+ struct cgroup *user;
+ struct cgroup *admin;
+};
+
+struct _GstRTSPCGroupPoolClass
+{
+ GstRTSPThreadPoolClass parent_class;
+};
+
+static GQuark thread_cgroup;
+
+static void gst_rtsp_cgroup_pool_finalize (GObject * obj);
+
+static void default_thread_enter (GstRTSPThreadPool * pool,
+ GstRTSPThread * thread);
+static void default_configure_thread (GstRTSPThreadPool * pool,
+ GstRTSPThread * thread, GstRTSPContext * ctx);
+
+static GType gst_rtsp_cgroup_pool_get_type (void);
+
+G_DEFINE_TYPE (GstRTSPCGroupPool, gst_rtsp_cgroup_pool,
+ GST_TYPE_RTSP_THREAD_POOL);
+
+static void
+gst_rtsp_cgroup_pool_class_init (GstRTSPCGroupPoolClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstRTSPThreadPoolClass *tpool_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ tpool_class = GST_RTSP_THREAD_POOL_CLASS (klass);
+
+ gobject_class->finalize = gst_rtsp_cgroup_pool_finalize;
+
+ tpool_class->configure_thread = default_configure_thread;
+ tpool_class->thread_enter = default_thread_enter;
+
+ thread_cgroup = g_quark_from_string ("cgroup.pool.thread.cgroup");
+
+ cgroup_init ();
+}
+
+static void
+gst_rtsp_cgroup_pool_init (GstRTSPCGroupPool * pool)
+{
+ pool->user = cgroup_new_cgroup ("user");
+ if (cgroup_add_controller (pool->user, "cpu") == NULL)
+ g_error ("Failed to add cpu controller to user cgroup");
+ pool->admin = cgroup_new_cgroup ("admin");
+ if (cgroup_add_controller (pool->admin, "cpu") == NULL)
+ g_error ("Failed to add cpu controller to admin cgroup");
+}
+
+static void
+gst_rtsp_cgroup_pool_finalize (GObject * obj)
+{
+ GstRTSPCGroupPool *pool = GST_RTSP_CGROUP_POOL (obj);
+
+ GST_INFO ("finalize pool %p", pool);
+
+ cgroup_free (&pool->user);
+ cgroup_free (&pool->admin);
+
+ G_OBJECT_CLASS (gst_rtsp_cgroup_pool_parent_class)->finalize (obj);
+}
+
+static void
+default_thread_enter (GstRTSPThreadPool * pool, GstRTSPThread * thread)
+{
+ struct cgroup *cgroup;
+
+ cgroup = gst_mini_object_get_qdata (GST_MINI_OBJECT (thread), thread_cgroup);
+ if (cgroup) {
+ gint res = 0;
+
+ res = cgroup_attach_task (cgroup);
+
+ if (res != 0)
+ GST_ERROR ("error: %d (%s)", res, cgroup_strerror (res));
+ }
+}
+
+static void
+default_configure_thread (GstRTSPThreadPool * pool,
+ GstRTSPThread * thread, GstRTSPContext * ctx)
+{
+ GstRTSPCGroupPool *cpool = GST_RTSP_CGROUP_POOL (pool);
+ const gchar *cls;
+ struct cgroup *cgroup;
+
+ if (ctx->token)
+ cls = gst_rtsp_token_get_string (ctx->token, "cgroup.pool.media.class");
+ else
+ cls = NULL;
+
+ GST_DEBUG ("manage cgroup %s", cls);
+
+ if (!g_strcmp0 (cls, "admin"))
+ cgroup = cpool->admin;
+ else
+ cgroup = cpool->user;
+
+ /* attach the cgroup to the thread */
+ gst_mini_object_set_qdata (GST_MINI_OBJECT (thread), thread_cgroup,
+ cgroup, NULL);
+}
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ gchar *basic;
+ GstRTSPThreadPool *thread_pool;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mounts for this server, every server has a default mapper object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=640,height=480,framerate=50/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* allow user and admin to access this resource */
+ gst_rtsp_media_factory_add_role (factory, "user",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_media_factory_add_role (factory, "admin",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* make a new authentication manager */
+ auth = gst_rtsp_auth_new ();
+
+ /* make user token */
+ token = gst_rtsp_token_new ("cgroup.pool.media.class", G_TYPE_STRING, "user",
+ "media.factory.role", G_TYPE_STRING, "user", NULL);
+ basic = gst_rtsp_auth_make_basic ("user", "password");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* make admin token */
+ token = gst_rtsp_token_new ("cgroup.pool.media.class", G_TYPE_STRING, "admin",
+ "media.factory.role", G_TYPE_STRING, "admin", NULL);
+ basic = gst_rtsp_auth_make_basic ("admin", "power");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* set as the server authentication manager */
+ gst_rtsp_server_set_auth (server, auth);
+ g_object_unref (auth);
+
+ thread_pool = g_object_new (GST_TYPE_RTSP_CGROUP_POOL, NULL);
+ gst_rtsp_server_set_thread_pool (server, thread_pool);
+ g_object_unref (thread_pool);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving */
+ g_print ("stream with user:password ready at rtsp://127.0.0.1:8554/test\n");
+ g_print ("stream with admin:power ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+#define DEFAULT_RTSP_PORT "8554"
+#define DEFAULT_DISABLE_RTCP FALSE
+
+static char *port = (char *) DEFAULT_RTSP_PORT;
+static gboolean disable_rtcp = DEFAULT_DISABLE_RTCP;
+
+static GOptionEntry entries[] = {
+ {"port", 'p', 0, G_OPTION_ARG_STRING, &port,
+ "Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
+ {"disable-rtcp", '\0', 0, G_OPTION_ARG_NONE, &disable_rtcp,
+ "Whether RTCP should be disabled (default false)", NULL},
+ {NULL}
+};
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GOptionContext *optctx;
+ GError *error = NULL;
+
+ optctx = g_option_context_new ("<launch line> - Test RTSP Server, Launch\n\n"
+ "Example: \"( videotestsrc ! x264enc ! rtph264pay name=pay0 pt=96 )\"");
+ g_option_context_add_main_entries (optctx, entries, NULL);
+ g_option_context_add_group (optctx, gst_init_get_option_group ());
+ if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
+ g_printerr ("Error parsing options: %s\n", error->message);
+ g_option_context_free (optctx);
+ g_clear_error (&error);
+ return -1;
+ }
+ g_option_context_free (optctx);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+ g_object_set (server, "service", port, NULL);
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, argv[1]);
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ gst_rtsp_media_factory_set_enable_rtcp (factory, !disable_rtcp);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", port);
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+#define DEFAULT_RTSP_PORT "8554"
+
+static char *port = (char *) DEFAULT_RTSP_PORT;
+
+static GOptionEntry entries[] = {
+ {"port", 'p', 0, G_OPTION_ARG_STRING, &port,
+ "Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
+ {NULL}
+};
+
+/* called when a stream has received an RTCP packet from the client */
+static void
+on_ssrc_active (GObject * session, GObject * source, GstRTSPMedia * media)
+{
+ GstStructure *stats;
+
+ GST_INFO ("source %p in session %p is active", source, session);
+
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ gchar *sstr;
+
+ sstr = gst_structure_to_string (stats);
+ g_print ("structure: %s\n", sstr);
+ g_free (sstr);
+
+ gst_structure_free (stats);
+ }
+}
+
+static void
+on_sender_ssrc_active (GObject * session, GObject * source,
+ GstRTSPMedia * media)
+{
+ GstStructure *stats;
+
+ GST_INFO ("source %p in session %p is active", source, session);
+
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ gchar *sstr;
+
+ sstr = gst_structure_to_string (stats);
+ g_print ("Sender stats:\nstructure: %s\n", sstr);
+ g_free (sstr);
+
+ gst_structure_free (stats);
+ }
+}
+
+/* signal callback when the media is prepared for streaming. We can get the
+ * session manager for each of the streams and connect to some signals. */
+static void
+media_prepared_cb (GstRTSPMedia * media)
+{
+ guint i, n_streams;
+
+ n_streams = gst_rtsp_media_n_streams (media);
+
+ GST_INFO ("media %p is prepared and has %u streams", media, n_streams);
+
+ for (i = 0; i < n_streams; i++) {
+ GstRTSPStream *stream;
+ GObject *session;
+
+ stream = gst_rtsp_media_get_stream (media, i);
+ if (stream == NULL)
+ continue;
+
+ session = gst_rtsp_stream_get_rtpsession (stream);
+ GST_INFO ("watching session %p on stream %u", session, i);
+
+ g_signal_connect (session, "on-ssrc-active",
+ (GCallback) on_ssrc_active, media);
+ g_signal_connect (session, "on-sender-ssrc-active",
+ (GCallback) on_sender_ssrc_active, media);
+ }
+}
+
+static void
+media_configure_cb (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+ /* connect our prepared signal so that we can see when this media is
+ * prepared for streaming */
+ g_signal_connect (media, "prepared", (GCallback) media_prepared_cb, factory);
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GOptionContext *optctx;
+ GError *error = NULL;
+ gchar *str;
+
+ optctx = g_option_context_new ("<filename.mp4> - Test RTSP Server, MP4");
+ g_option_context_add_main_entries (optctx, entries, NULL);
+ g_option_context_add_group (optctx, gst_init_get_option_group ());
+ if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
+ g_printerr ("Error parsing options: %s\n", error->message);
+ g_option_context_free (optctx);
+ g_clear_error (&error);
+ return -1;
+ }
+
+ if (argc < 2) {
+ g_print ("%s\n", g_option_context_get_help (optctx, TRUE, NULL));
+ return 1;
+ }
+ g_option_context_free (optctx);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+ g_object_set (server, "service", port, NULL);
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ str = g_strdup_printf ("( "
+ "filesrc location=\"%s\" ! qtdemux name=d "
+ "d. ! queue ! rtph264pay pt=96 name=pay0 "
+ "d. ! queue ! rtpmp4apay pt=97 name=pay1 " ")", argv[1]);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, str);
+ g_signal_connect (factory, "media-configure", (GCallback) media_configure_cb,
+ factory);
+ g_free (str);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", port);
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAddressPool *pool;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+
+ /* make a new address pool */
+ pool = gst_rtsp_address_pool_new ();
+ gst_rtsp_address_pool_add_range (pool,
+ "224.3.0.0", "224.3.0.10", 5000, 5010, 16);
+ gst_rtsp_media_factory_set_address_pool (factory, pool);
+ /* only allow multicast */
+ gst_rtsp_media_factory_set_protocols (factory,
+ GST_RTSP_LOWER_TRANS_UDP_MCAST);
+ g_object_unref (pool);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+static void
+media_constructed (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+ guint i, n_streams;
+
+ n_streams = gst_rtsp_media_n_streams (media);
+
+ for (i = 0; i < n_streams; i++) {
+ GstRTSPAddressPool *pool;
+ GstRTSPStream *stream;
+ gchar *min, *max;
+
+ stream = gst_rtsp_media_get_stream (media, i);
+
+ /* make a new address pool */
+ pool = gst_rtsp_address_pool_new ();
+
+ min = g_strdup_printf ("224.3.0.%d", (2 * i) + 1);
+ max = g_strdup_printf ("224.3.0.%d", (2 * i) + 2);
+ gst_rtsp_address_pool_add_range (pool, min, max,
+ 5000 + (10 * i), 5010 + (10 * i), 1);
+ g_free (min);
+ g_free (max);
+
+ gst_rtsp_stream_set_address_pool (stream, pool);
+ g_object_unref (pool);
+ }
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+
+ g_signal_connect (factory, "media-constructed", (GCallback)
+ media_constructed, NULL);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2014 Jan Schmidt <jan@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <stdlib.h>
+
+#include <gst/gst.h>
+#include <gst/net/gstnet.h>
+
+#define PLAYBACK_DELAY_MS 40
+
+static void
+source_created (GstElement * pipe, GstElement * source)
+{
+ g_object_set (source, "latency", PLAYBACK_DELAY_MS,
+ "ntp-time-source", 3, "buffer-mode", 4, "ntp-sync", TRUE, NULL);
+}
+
+static gboolean
+message (GstBus * bus, GstMessage * message, gpointer user_data)
+{
+ GMainLoop *loop = user_data;
+
+ switch (GST_MESSAGE_TYPE (message)) {
+ case GST_MESSAGE_ERROR:{
+ GError *err = NULL;
+ gchar *name, *debug = NULL;
+
+ name = gst_object_get_path_string (message->src);
+ gst_message_parse_error (message, &err, &debug);
+
+ g_printerr ("ERROR: from element %s: %s\n", name, err->message);
+ if (debug != NULL)
+ g_printerr ("Additional debug info:\n%s\n", debug);
+
+ g_error_free (err);
+ g_free (debug);
+ g_free (name);
+
+ g_main_loop_quit (loop);
+ break;
+ }
+ case GST_MESSAGE_WARNING:{
+ GError *err = NULL;
+ gchar *name, *debug = NULL;
+
+ name = gst_object_get_path_string (message->src);
+ gst_message_parse_warning (message, &err, &debug);
+
+ g_printerr ("ERROR: from element %s: %s\n", name, err->message);
+ if (debug != NULL)
+ g_printerr ("Additional debug info:\n%s\n", debug);
+
+ g_error_free (err);
+ g_free (debug);
+ g_free (name);
+ break;
+ }
+ case GST_MESSAGE_EOS:
+ g_print ("Got EOS\n");
+ g_main_loop_quit (loop);
+ break;
+ default:
+ break;
+ }
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GstClock *net_clock;
+ gchar *server;
+ gint clock_port;
+ GstElement *pipe;
+ GMainLoop *loop;
+
+ gst_init (&argc, &argv);
+
+ if (argc < 2) {
+ g_print ("usage: %s rtsp://URI clock-IP clock-PORT\n"
+ "example: %s rtsp://localhost:8554/test 127.0.0.1 8554\n",
+ argv[0], argv[0]);
+ return -1;
+ }
+
+ server = argv[2];
+ clock_port = atoi (argv[3]);
+
+ net_clock = gst_net_client_clock_new ("net_clock", server, clock_port, 0);
+ if (net_clock == NULL) {
+ g_print ("Failed to create net clock client for %s:%d\n",
+ server, clock_port);
+ return 1;
+ }
+
+ /* Wait for the clock to stabilise */
+ gst_clock_wait_for_sync (net_clock, GST_CLOCK_TIME_NONE);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ pipe = gst_element_factory_make ("playbin", NULL);
+ g_object_set (pipe, "uri", argv[1], NULL);
+ g_signal_connect (pipe, "source-setup", G_CALLBACK (source_created), NULL);
+
+ gst_pipeline_use_clock (GST_PIPELINE (pipe), net_clock);
+
+ /* Set this high enough so that it's higher than the minimum latency
+ * on all receivers */
+ gst_pipeline_set_latency (GST_PIPELINE (pipe), 500 * GST_MSECOND);
+
+ if (gst_element_set_state (pipe,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
+ g_print ("Failed to set state to PLAYING\n");
+ goto exit;
+ };
+
+ gst_bus_add_signal_watch (GST_ELEMENT_BUS (pipe));
+ g_signal_connect (GST_ELEMENT_BUS (pipe), "message", G_CALLBACK (message),
+ loop);
+
+ g_main_loop_run (loop);
+
+exit:
+ gst_element_set_state (pipe, GST_STATE_NULL);
+ gst_object_unref (pipe);
+ g_main_loop_unref (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2014 Jan Schmidt <jan@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/net/gstnettimeprovider.h>
+#include <gst/rtsp-server/rtsp-server.h>
+
+GstClock *global_clock;
+
+#define TEST_TYPE_RTSP_MEDIA_FACTORY (test_rtsp_media_factory_get_type ())
+#define TEST_TYPE_RTSP_MEDIA (test_rtsp_media_get_type ())
+
+GType test_rtsp_media_get_type (void);
+
+typedef struct TestRTSPMediaClass TestRTSPMediaClass;
+typedef struct TestRTSPMedia TestRTSPMedia;
+
+struct TestRTSPMediaClass
+{
+ GstRTSPMediaClass parent;
+};
+
+struct TestRTSPMedia
+{
+ GstRTSPMedia parent;
+};
+
+static gboolean custom_setup_rtpbin (GstRTSPMedia * media, GstElement * rtpbin);
+
+G_DEFINE_TYPE (TestRTSPMedia, test_rtsp_media, GST_TYPE_RTSP_MEDIA);
+
+static void
+test_rtsp_media_class_init (TestRTSPMediaClass * test_klass)
+{
+ GstRTSPMediaClass *klass = (GstRTSPMediaClass *) (test_klass);
+ klass->setup_rtpbin = custom_setup_rtpbin;
+}
+
+static void
+test_rtsp_media_init (TestRTSPMedia * media)
+{
+}
+
+static gboolean
+custom_setup_rtpbin (GstRTSPMedia * media, GstElement * rtpbin)
+{
+ g_object_set (rtpbin, "ntp-time-source", 3, NULL);
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ if (argc < 2) {
+ g_print ("usage: %s <launch line> \n"
+ "example: %s \"( videotestsrc is-live=true ! x264enc ! rtph264pay name=pay0 pt=96 )\"\n"
+ "Pipeline must be live for synchronisation to work properly with this method!\n",
+ argv[0], argv[0]);
+ return -1;
+ }
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ global_clock = gst_system_clock_obtain ();
+ gst_net_time_provider_new (global_clock, "0.0.0.0", 8554);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, argv[1]);
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ gst_rtsp_media_factory_set_media_gtype (factory, TEST_TYPE_RTSP_MEDIA);
+ gst_rtsp_media_factory_set_clock (factory, global_clock);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+#define DEFAULT_RTSP_PORT "8554"
+
+static char *port = (char *) DEFAULT_RTSP_PORT;
+
+static GOptionEntry entries[] = {
+ {"port", 'p', 0, G_OPTION_ARG_STRING, &port,
+ "Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
+ {NULL}
+};
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GOptionContext *optctx;
+ GError *error = NULL;
+ gchar *str;
+
+ optctx = g_option_context_new ("<filename.ogg> - Test RTSP Server, OGG");
+ g_option_context_add_main_entries (optctx, entries, NULL);
+ g_option_context_add_group (optctx, gst_init_get_option_group ());
+ if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
+ g_printerr ("Error parsing options: %s\n", error->message);
+ g_option_context_free (optctx);
+ g_clear_error (&error);
+ return -1;
+ }
+ g_option_context_free (optctx);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+ g_object_set (server, "service", port, NULL);
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ str = g_strdup_printf ("( "
+ "filesrc location=%s ! oggdemux name=d "
+ "d. ! queue ! rtptheorapay name=pay0 pt=96 "
+ "d. ! queue ! rtpvorbispay name=pay1 pt=97 " ")", argv[1]);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, str);
+ g_free (str);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", port);
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp-server/rtsp-onvif-server.h>
+
+#include <string.h>
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_onvif_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_onvif_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc is-live=true ! x264enc ! rtph264pay name=pay0 pt=96 audiotestsrc is-live=true ! mulawenc ! rtppcmupay name=pay1 )");
+ gst_rtsp_onvif_media_factory_set_backchannel_launch
+ (GST_RTSP_ONVIF_MEDIA_FACTORY (factory),
+ "( capsfilter caps=\"application/x-rtp, media=audio, payload=0, clock-rate=8000, encoding-name=PCMU\" name=depay_backchannel ! rtppcmudepay ! fakesink async=false )");
+ gst_rtsp_media_factory_set_shared (factory, FALSE);
+ gst_rtsp_media_factory_set_media_gtype (factory, GST_TYPE_RTSP_ONVIF_MEDIA);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2019 Mathieu Duponchelle <mathieu@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <stdio.h>
+
+#include <gst/gst.h>
+#include <gst/rtsp/rtsp.h>
+
+typedef struct
+{
+ gchar *range;
+ gdouble speed;
+ gchar *frames;
+ gchar *rate_control;
+ gboolean reverse;
+} SeekParameters;
+
+typedef struct
+{
+ GstElement *src;
+ GstElement *sink;
+ GstElement *pipe;
+ SeekParameters *seek_params;
+ GMainLoop *loop;
+ GIOChannel *io;
+ gboolean new_range;
+ guint io_watch_id;
+ gboolean reset_sync;
+} Context;
+
+typedef struct
+{
+ const gchar *name;
+ gboolean has_argument;
+ const gchar *help;
+ gboolean (*func) (Context * ctx, gchar * arg, gboolean * async);
+} Command;
+
+static gboolean cmd_help (Context * ctx, gchar * arg, gboolean * async);
+static gboolean cmd_pause (Context * ctx, gchar * arg, gboolean * async);
+static gboolean cmd_play (Context * ctx, gchar * arg, gboolean * async);
+static gboolean cmd_reverse (Context * ctx, gchar * arg, gboolean * async);
+static gboolean cmd_range (Context * ctx, gchar * arg, gboolean * async);
+static gboolean cmd_speed (Context * ctx, gchar * arg, gboolean * async);
+static gboolean cmd_frames (Context * ctx, gchar * arg, gboolean * async);
+static gboolean cmd_rate_control (Context * ctx, gchar * arg, gboolean * async);
+static gboolean cmd_step_forward (Context * ctx, gchar * arg, gboolean * async);
+
+static Command commands[] = {
+ {"help", FALSE, "Display list of valid commands", cmd_help},
+ {"pause", FALSE, "Pause playback", cmd_pause},
+ {"play", FALSE, "Resume playback", cmd_play},
+ {"reverse", FALSE, "Reverse playback direction", cmd_reverse},
+ {"range", TRUE,
+ "Seek to the specified range, example: \"range: 19000101T000000Z-19000101T000200Z\"",
+ cmd_range},
+ {"speed", TRUE, "Set the playback speed, example: \"speed: 1.0\"", cmd_speed},
+ {"frames", TRUE,
+ "Set the frames trickmode, example: \"frames: intra\", \"frames: predicted\", \"frames: intra/1000\"",
+ cmd_frames},
+ {"rate-control", TRUE,
+ "Set the rate control mode, example: \"rate-control: no\"",
+ cmd_rate_control},
+ {"s", FALSE, "Step to the following frame (in current playback direction)",
+ cmd_step_forward},
+ {NULL},
+};
+
+static gchar *rtsp_address;
+
+#define MAKE_AND_ADD(var, pipe, name, label, elem_name) \
+G_STMT_START { \
+ if (G_UNLIKELY (!(var = (gst_element_factory_make (name, elem_name))))) { \
+ GST_ERROR ("Could not create element %s", name); \
+ goto label; \
+ } \
+ if (G_UNLIKELY (!gst_bin_add (GST_BIN_CAST (pipe), var))) { \
+ GST_ERROR ("Could not add element %s", name); \
+ goto label; \
+ } \
+} G_STMT_END
+
+#define DEFAULT_RANGE "19000101T000000Z-19000101T000200Z"
+#define DEFAULT_SPEED 1.0
+#define DEFAULT_FRAMES "none"
+#define DEFAULT_RATE_CONTROL "yes"
+#define DEFAULT_REVERSE FALSE
+
+static void
+pad_added_cb (GstElement * src, GstPad * srcpad, GstElement * peer)
+{
+ GstPad *sinkpad = gst_element_get_static_pad (peer, "sink");
+
+ gst_pad_link (srcpad, sinkpad);
+
+ gst_object_unref (sinkpad);
+}
+
+static gboolean
+setup (Context * ctx)
+{
+ GstElement *onvifparse, *queue, *vdepay, *vdec, *vconv, *toverlay, *tee,
+ *vqueue;
+ gboolean ret = FALSE;
+
+ MAKE_AND_ADD (ctx->src, ctx->pipe, "rtspsrc", done, NULL);
+ MAKE_AND_ADD (queue, ctx->pipe, "queue", done, NULL);
+ MAKE_AND_ADD (onvifparse, ctx->pipe, "rtponvifparse", done, NULL);
+ MAKE_AND_ADD (vdepay, ctx->pipe, "rtph264depay", done, NULL);
+ MAKE_AND_ADD (vdec, ctx->pipe, "avdec_h264", done, NULL);
+ MAKE_AND_ADD (vconv, ctx->pipe, "videoconvert", done, NULL);
+ MAKE_AND_ADD (toverlay, ctx->pipe, "timeoverlay", done, NULL);
+ MAKE_AND_ADD (tee, ctx->pipe, "tee", done, NULL);
+ MAKE_AND_ADD (vqueue, ctx->pipe, "queue", done, NULL);
+ MAKE_AND_ADD (ctx->sink, ctx->pipe, "xvimagesink", done, NULL);
+
+ g_object_set (ctx->src, "location", rtsp_address, NULL);
+ g_object_set (ctx->src, "onvif-mode", TRUE, NULL);
+ g_object_set (ctx->src, "tcp-timeout", 0, NULL);
+ g_object_set (toverlay, "show-times-as-dates", TRUE, NULL);
+
+ g_object_set (toverlay, "datetime-format", "%a %d, %b %Y - %T", NULL);
+
+ g_signal_connect (ctx->src, "pad-added", G_CALLBACK (pad_added_cb), queue);
+
+ if (!gst_element_link_many (queue, onvifparse, vdepay, vdec, vconv, toverlay,
+ tee, vqueue, ctx->sink, NULL)) {
+ goto done;
+ }
+
+ g_object_set (ctx->src, "onvif-rate-control", FALSE, "is-live", FALSE, NULL);
+
+ if (!g_strcmp0 (ctx->seek_params->rate_control, "no")) {
+ g_object_set (ctx->sink, "sync", FALSE, NULL);
+ }
+
+ ret = TRUE;
+
+done:
+ return ret;
+}
+
+static GstClockTime
+get_current_position (Context * ctx, gboolean reverse)
+{
+ GstSample *sample;
+ GstBuffer *buffer;
+ GstClockTime ret;
+
+ g_object_get (ctx->sink, "last-sample", &sample, NULL);
+
+ buffer = gst_sample_get_buffer (sample);
+
+ ret = GST_BUFFER_PTS (buffer);
+
+ if (reverse && GST_CLOCK_TIME_IS_VALID (GST_BUFFER_DURATION (buffer)))
+ ret += GST_BUFFER_DURATION (buffer);
+
+ gst_sample_unref (sample);
+
+ return ret;
+}
+
+static GstEvent *
+translate_seek_parameters (Context * ctx, SeekParameters * seek_params)
+{
+ GstEvent *ret = NULL;
+ gchar *range_str = NULL;
+ GstRTSPTimeRange *rtsp_range;
+ GstSeekType start_type, stop_type;
+ GstClockTime start, stop;
+ gdouble rate;
+ GstSeekFlags flags;
+ gchar **split = NULL;
+ GstClockTime trickmode_interval = 0;
+
+ range_str = g_strdup_printf ("clock=%s", seek_params->range);
+
+ if (gst_rtsp_range_parse (range_str, &rtsp_range) != GST_RTSP_OK) {
+ GST_ERROR ("Failed to parse range %s", range_str);
+ goto done;
+ }
+
+ gst_rtsp_range_get_times (rtsp_range, &start, &stop);
+
+ if (start > stop) {
+ GST_ERROR ("Invalid range, start > stop: %s", seek_params->range);
+ goto done;
+ }
+
+ start_type = GST_SEEK_TYPE_SET;
+ stop_type = GST_SEEK_TYPE_SET;
+
+ if (!ctx->new_range) {
+ GstClockTime current_position =
+ get_current_position (ctx, seek_params->reverse);
+
+ if (seek_params->reverse) {
+ stop_type = GST_SEEK_TYPE_SET;
+ stop = current_position;
+ } else {
+ start_type = GST_SEEK_TYPE_SET;
+ start = current_position;
+ }
+ }
+
+ ctx->new_range = FALSE;
+
+ flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE;
+
+ split = g_strsplit (seek_params->frames, "/", 2);
+
+ if (!g_strcmp0 (split[0], "intra")) {
+ if (split[1]) {
+ guint64 interval;
+ gchar *end;
+
+ interval = g_ascii_strtoull (split[1], &end, 10);
+
+ if (!end || *end != '\0') {
+ GST_ERROR ("Unexpected interval value %s", split[1]);
+ goto done;
+ }
+
+ trickmode_interval = interval * GST_MSECOND;
+ }
+ flags |= GST_SEEK_FLAG_TRICKMODE_KEY_UNITS;
+ } else if (!g_strcmp0 (split[0], "predicted")) {
+ if (split[1]) {
+ GST_ERROR ("Predicted frames mode does not allow an interval (%s)",
+ seek_params->frames);
+ goto done;
+ }
+ flags |= GST_SEEK_FLAG_TRICKMODE_FORWARD_PREDICTED;
+ } else if (g_strcmp0 (split[0], "none")) {
+ GST_ERROR ("Invalid frames mode (%s)", seek_params->frames);
+ goto done;
+ }
+
+ if (seek_params->reverse) {
+ rate = -1.0 * seek_params->speed;
+ } else {
+ rate = 1.0 * seek_params->speed;
+ }
+
+ ret = gst_event_new_seek (rate, GST_FORMAT_TIME, flags,
+ start_type, start, stop_type, stop);
+
+ if (trickmode_interval)
+ gst_event_set_seek_trickmode_interval (ret, trickmode_interval);
+
+done:
+ if (split)
+ g_strfreev (split);
+ g_free (range_str);
+ return ret;
+}
+
+static void prompt_on (Context * ctx);
+static void prompt_off (Context * ctx);
+
+static gboolean
+cmd_help (Context * ctx, gchar * arg, gboolean * async)
+{
+ gboolean ret = TRUE;
+ guint i;
+
+ *async = FALSE;
+
+ for (i = 0; commands[i].name; i++) {
+ g_print ("%s: %s\n", commands[i].name, commands[i].help);
+ }
+
+ return ret;
+}
+
+static gboolean
+cmd_pause (Context * ctx, gchar * arg, gboolean * async)
+{
+ gboolean ret;
+ GstStateChangeReturn state_ret;
+
+ g_print ("Pausing\n");
+
+ state_ret = gst_element_set_state (ctx->pipe, GST_STATE_PAUSED);
+
+ *async = state_ret == GST_STATE_CHANGE_ASYNC;
+ ret = state_ret != GST_STATE_CHANGE_FAILURE;
+
+ return ret;
+}
+
+static gboolean
+cmd_play (Context * ctx, gchar * arg, gboolean * async)
+{
+ gboolean ret;
+ GstStateChangeReturn state_ret;
+
+ g_print ("Playing\n");
+
+ state_ret = gst_element_set_state (ctx->pipe, GST_STATE_PLAYING);
+
+ *async = state_ret == GST_STATE_CHANGE_ASYNC;
+ ret = state_ret != GST_STATE_CHANGE_FAILURE;
+
+ return ret;
+}
+
+static gboolean
+do_seek (Context * ctx)
+{
+ gboolean ret = FALSE;
+ GstEvent *event;
+
+ if (!(event = translate_seek_parameters (ctx, ctx->seek_params))) {
+ GST_ERROR ("Failed to create seek event");
+ goto done;
+ }
+
+ if (ctx->seek_params->reverse)
+ g_object_set (ctx->src, "onvif-rate-control", FALSE, NULL);
+
+ if (ctx->reset_sync) {
+ g_object_set (ctx->sink, "sync", TRUE, NULL);
+ ctx->reset_sync = FALSE;
+ }
+
+ if (!gst_element_send_event (ctx->src, event)) {
+ GST_ERROR ("Failed to seek rtspsrc");
+ g_main_loop_quit (ctx->loop);
+ goto done;
+ }
+
+ ret = TRUE;
+
+done:
+ return ret;
+}
+
+static gboolean
+cmd_reverse (Context * ctx, gchar * arg, gboolean * async)
+{
+ gboolean ret = TRUE;
+
+ g_print ("Reversing playback direction\n");
+
+ ctx->seek_params->reverse = !ctx->seek_params->reverse;
+
+ ret = do_seek (ctx);
+
+ *async = ret == TRUE;
+
+ return ret;
+}
+
+static gboolean
+cmd_range (Context * ctx, gchar * arg, gboolean * async)
+{
+ gboolean ret = TRUE;
+
+ g_print ("Switching to new range\n");
+
+ g_free (ctx->seek_params->range);
+ ctx->seek_params->range = g_strdup (arg);
+ ctx->new_range = TRUE;
+
+ ret = do_seek (ctx);
+
+ *async = ret == TRUE;
+
+ return ret;
+}
+
+static gboolean
+cmd_speed (Context * ctx, gchar * arg, gboolean * async)
+{
+ gboolean ret = FALSE;
+ gchar *endptr = NULL;
+ gdouble new_speed;
+
+ new_speed = g_ascii_strtod (arg, &endptr);
+
+ g_print ("Switching gears\n");
+
+ if (endptr == NULL || *endptr != '\0' || new_speed <= 0.0) {
+ GST_ERROR ("Invalid value for speed: %s", arg);
+ goto done;
+ }
+
+ ctx->seek_params->speed = new_speed;
+ ret = do_seek (ctx);
+
+done:
+ *async = ret == TRUE;
+ return ret;
+}
+
+static gboolean
+cmd_frames (Context * ctx, gchar * arg, gboolean * async)
+{
+ gboolean ret = TRUE;
+
+ g_print ("Changing Frames trickmode\n");
+
+ g_free (ctx->seek_params->frames);
+ ctx->seek_params->frames = g_strdup (arg);
+ ret = do_seek (ctx);
+ *async = ret == TRUE;
+
+ return ret;
+}
+
+static gboolean
+cmd_rate_control (Context * ctx, gchar * arg, gboolean * async)
+{
+ gboolean ret = FALSE;
+
+ *async = FALSE;
+
+ if (!g_strcmp0 (arg, "no")) {
+ g_object_set (ctx->sink, "sync", FALSE, NULL);
+ ret = TRUE;
+ } else if (!g_strcmp0 (arg, "yes")) {
+ /* TODO: there probably is a solution that doesn't involve sending
+ * a request to the server to reset our position */
+ ctx->reset_sync = TRUE;
+ ret = do_seek (ctx);
+ *async = TRUE;
+ } else {
+ GST_ERROR ("Invalid rate-control: %s", arg);
+ goto done;
+ }
+
+ ret = TRUE;
+
+done:
+ return ret;
+}
+
+static gboolean
+cmd_step_forward (Context * ctx, gchar * arg, gboolean * async)
+{
+ gboolean ret = FALSE;
+ GstEvent *event;
+
+ event = gst_event_new_step (GST_FORMAT_BUFFERS, 1, 1.0, TRUE, FALSE);
+
+ g_print ("Stepping\n");
+
+ if (!gst_element_send_event (ctx->sink, event)) {
+ GST_ERROR ("Failed to step forward");
+ goto done;
+ }
+
+ ret = TRUE;
+
+done:
+ *async = ret == TRUE;
+ return ret;
+}
+
+static void
+handle_command (Context * ctx, gchar * cmd)
+{
+ gchar **split;
+ guint i;
+ gboolean valid_command = FALSE;
+
+ split = g_strsplit (cmd, ":", 0);
+
+ cmd = g_strstrip (split[0]);
+
+ if (cmd == NULL || *cmd == '\0') {
+ g_print ("> ");
+ goto done;
+ }
+
+ for (i = 0; commands[i].name; i++) {
+ if (!g_strcmp0 (commands[i].name, cmd)) {
+ valid_command = TRUE;
+ if (commands[i].has_argument && g_strv_length (split) != 2) {
+ g_print ("Command %s expects exactly one argument:\n%s: %s\n", cmd,
+ commands[i].name, commands[i].help);
+ } else if (!commands[i].has_argument && g_strv_length (split) != 1) {
+ g_print ("Command %s expects no argument:\n%s: %s\n", cmd,
+ commands[i].name, commands[i].help);
+ } else {
+ gboolean async = FALSE;
+
+ if (commands[i].func (ctx,
+ commands[i].has_argument ? g_strstrip (split[1]) : NULL, &async)
+ && async)
+ prompt_off (ctx);
+ else
+ g_print ("> ");
+ }
+ break;
+ }
+ }
+
+ if (!valid_command) {
+ g_print ("Invalid command %s\n> ", cmd);
+ }
+
+done:
+ g_strfreev (split);
+}
+
+static gboolean
+io_callback (GIOChannel * io, GIOCondition condition, Context * ctx)
+{
+ gboolean ret = TRUE;
+ gchar *str;
+ GError *error = NULL;
+
+ switch (condition) {
+ case G_IO_PRI:
+ case G_IO_IN:
+ switch (g_io_channel_read_line (io, &str, NULL, NULL, &error)) {
+ case G_IO_STATUS_ERROR:
+ GST_ERROR ("Failed to read commands from stdin: %s", error->message);
+ g_clear_error (&error);
+ g_main_loop_quit (ctx->loop);
+ break;
+ case G_IO_STATUS_EOF:
+ g_print ("EOF received, bye\n");
+ g_main_loop_quit (ctx->loop);
+ break;
+ case G_IO_STATUS_AGAIN:
+ break;
+ case G_IO_STATUS_NORMAL:
+ handle_command (ctx, str);
+ g_free (str);
+ break;
+ }
+ break;
+ case G_IO_ERR:
+ case G_IO_HUP:
+ GST_ERROR ("Failed to read commands from stdin");
+ g_main_loop_quit (ctx->loop);
+ break;
+ case G_IO_OUT:
+ default:
+ break;
+ }
+
+ return ret;
+}
+
+#ifndef STDIN_FILENO
+#ifdef G_OS_WIN32
+#define STDIN_FILENO _fileno(stdin)
+#else /* !G_OS_WIN32 */
+#define STDIN_FILENO 0
+#endif /* G_OS_WIN32 */
+#endif /* STDIN_FILENO */
+
+static void
+prompt_on (Context * ctx)
+{
+ g_assert (!ctx->io);
+ ctx->io = g_io_channel_unix_new (STDIN_FILENO);
+ ctx->io_watch_id =
+ g_io_add_watch (ctx->io, G_IO_IN, (GIOFunc) io_callback, ctx);
+ g_print ("> ");
+}
+
+static void
+prompt_off (Context * ctx)
+{
+ g_assert (ctx->io);
+ g_source_remove (ctx->io_watch_id);
+ g_io_channel_unref (ctx->io);
+ ctx->io = NULL;
+}
+
+static gboolean
+bus_message_cb (GstBus * bus, GstMessage * message, Context * ctx)
+{
+ switch (GST_MESSAGE_TYPE (message)) {
+ case GST_MESSAGE_STATE_CHANGED:{
+ GstState olds, news, pendings;
+
+ if (GST_MESSAGE_SRC (message) == GST_OBJECT (ctx->pipe)) {
+ gst_message_parse_state_changed (message, &olds, &news, &pendings);
+ GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (ctx->pipe),
+ GST_DEBUG_GRAPH_SHOW_ALL, "playing");
+ }
+ break;
+ }
+ case GST_MESSAGE_ERROR:{
+ GError *error = NULL;
+ gchar *debug;
+
+ gst_message_parse_error (message, &error, &debug);
+
+ gst_printerr ("Error: %s (%s)\n", error->message, debug);
+ g_clear_error (&error);
+ g_free (debug);
+ g_main_loop_quit (ctx->loop);
+ break;
+ }
+ case GST_MESSAGE_LATENCY:{
+ gst_bin_recalculate_latency (GST_BIN (ctx->pipe));
+ break;
+ }
+ case GST_MESSAGE_ASYNC_DONE:{
+ prompt_on (ctx);
+ }
+ default:
+ break;
+ }
+
+ return TRUE;
+}
+
+int
+main (int argc, char **argv)
+{
+ GOptionContext *optctx;
+ Context ctx = { 0, };
+ GstBus *bus;
+ gint ret = 1;
+ GError *error = NULL;
+ const gchar *range = NULL;
+ const gchar *frames = NULL;
+ const gchar *rate_control = NULL;
+ gchar *default_speed =
+ g_strdup_printf ("Speed to request (default: %.1f)", DEFAULT_SPEED);
+ SeekParameters seek_params =
+ { NULL, DEFAULT_SPEED, NULL, NULL, DEFAULT_REVERSE };
+ GOptionEntry entries[] = {
+ {"range", 0, 0, G_OPTION_ARG_STRING, &range,
+ "Range to seek (default: " DEFAULT_RANGE ")", "RANGE"},
+ {"speed", 0, 0, G_OPTION_ARG_DOUBLE, &seek_params.speed,
+ default_speed, "SPEED"},
+ {"frames", 0, 0, G_OPTION_ARG_STRING, &frames,
+ "Frames to request (default: " DEFAULT_FRAMES ")", "FRAMES"},
+ {"rate-control", 0, 0, G_OPTION_ARG_STRING, &rate_control,
+ "Apply rate control on the client side (default: "
+ DEFAULT_RATE_CONTROL ")", "RATE_CONTROL"},
+ {"reverse", 0, 0, G_OPTION_ARG_NONE, &seek_params.reverse,
+ "Playback direction", ""},
+ {NULL}
+ };
+
+ optctx = g_option_context_new ("<rtsp-url> - ONVIF RTSP Client");
+ g_option_context_add_main_entries (optctx, entries, NULL);
+ g_option_context_add_group (optctx, gst_init_get_option_group ());
+ if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
+ g_printerr ("Error parsing options: %s\n", error->message);
+ g_option_context_free (optctx);
+ g_clear_error (&error);
+ return -1;
+ }
+ if (argc < 2) {
+ g_print ("%s\n", g_option_context_get_help (optctx, TRUE, NULL));
+ return 1;
+ }
+ rtsp_address = argv[1];
+ g_option_context_free (optctx);
+
+ seek_params.range = g_strdup (range ? range : DEFAULT_RANGE);
+ seek_params.frames = g_strdup (frames ? frames : DEFAULT_FRAMES);
+ seek_params.rate_control =
+ g_strdup (rate_control ? rate_control : DEFAULT_RATE_CONTROL);
+
+ if (seek_params.speed <= 0.0) {
+ GST_ERROR ("SPEED must be a positive number");
+ return 1;
+ }
+
+ ctx.seek_params = &seek_params;
+ ctx.new_range = TRUE;
+ ctx.reset_sync = FALSE;
+
+ ctx.pipe = gst_pipeline_new (NULL);
+ if (!setup (&ctx)) {
+ g_printerr ("Damn\n");
+ goto done;
+ }
+
+ g_print ("Type help for the list of available commands\n");
+
+ do_seek (&ctx);
+
+ ctx.loop = g_main_loop_new (NULL, FALSE);
+
+ bus = gst_pipeline_get_bus (GST_PIPELINE (ctx.pipe));
+ gst_bus_add_watch (bus, (GstBusFunc) bus_message_cb, &ctx);
+
+ /* This will make rtspsrc progress to the OPEN state, at which point we can seek it */
+ if (!gst_element_set_state (ctx.pipe, GST_STATE_PLAYING))
+ goto done;
+
+ g_main_loop_run (ctx.loop);
+
+ g_main_loop_unref (ctx.loop);
+
+ gst_bus_remove_watch (bus);
+ gst_object_unref (bus);
+ gst_element_set_state (ctx.pipe, GST_STATE_NULL);
+ gst_object_unref (ctx.pipe);
+
+ ret = 0;
+
+done:
+ g_free (seek_params.range);
+ g_free (seek_params.frames);
+ g_free (seek_params.rate_control);
+ g_free (default_speed);
+ return ret;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2019 Mathieu Duponchelle <mathieu@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+#include "test-onvif-server.h"
+
+GST_DEBUG_CATEGORY_STATIC (onvif_server_debug);
+#define GST_CAT_DEFAULT (onvif_server_debug)
+
+#define MAKE_AND_ADD(var, pipe, name, label, elem_name) \
+G_STMT_START { \
+ if (G_UNLIKELY (!(var = (gst_element_factory_make (name, elem_name))))) { \
+ GST_ERROR ("Could not create element %s", name); \
+ goto label; \
+ } \
+ if (G_UNLIKELY (!gst_bin_add (GST_BIN_CAST (pipe), var))) { \
+ GST_ERROR ("Could not add element %s", name); \
+ goto label; \
+ } \
+} G_STMT_END
+
+/* This simulates an archive of recordings running from 01-01-1900 to 01-01-2000.
+ *
+ * This is implemented by repeating the file provided at the command line, with
+ * an empty interval of 5 seconds in-between. We intercept relevant events to
+ * translate them, and update the timestamps on the output buffers.
+ */
+
+#define INTERVAL (5 * GST_SECOND)
+
+/* January the first, 2000 */
+#define END_DATE 3155673600 * GST_SECOND
+
+static gchar *filename;
+
+struct _ReplayBin
+{
+ GstBin parent;
+
+ GstEvent *incoming_seek;
+ GstEvent *outgoing_seek;
+ GstClockTime trickmode_interval;
+
+ GstSegment segment;
+ const GstSegment *incoming_segment;
+ gboolean sent_segment;
+ GstClockTime ts_offset;
+ gint64 remainder;
+ GstClockTime min_pts;
+};
+
+G_DEFINE_TYPE (ReplayBin, replay_bin, GST_TYPE_BIN);
+
+static void
+replay_bin_init (ReplayBin * self)
+{
+ self->incoming_seek = NULL;
+ self->outgoing_seek = NULL;
+ self->trickmode_interval = 0;
+ self->ts_offset = 0;
+ self->sent_segment = FALSE;
+ self->min_pts = GST_CLOCK_TIME_NONE;
+}
+
+static void
+replay_bin_class_init (ReplayBinClass * klass)
+{
+}
+
+static GstElement *
+replay_bin_new (void)
+{
+ return GST_ELEMENT (g_object_new (replay_bin_get_type (), NULL));
+}
+
+static void
+demux_pad_added_cb (GstElement * demux, GstPad * pad, GstGhostPad * ghost)
+{
+ GstCaps *caps = gst_pad_get_current_caps (pad);
+ GstStructure *s = gst_caps_get_structure (caps, 0);
+
+ if (gst_structure_has_name (s, "video/x-h264")) {
+ gst_ghost_pad_set_target (ghost, pad);
+ }
+
+ gst_caps_unref (caps);
+}
+
+static void
+query_seekable (GstPad * ghost, gint64 * start, gint64 * stop)
+{
+ GstPad *target;
+ GstQuery *query;
+ GstFormat format;
+ gboolean seekable;
+
+ target = gst_ghost_pad_get_target (GST_GHOST_PAD (ghost));
+
+ query = gst_query_new_seeking (GST_FORMAT_TIME);
+
+ gst_pad_query (target, query);
+
+ gst_query_parse_seeking (query, &format, &seekable, start, stop);
+
+ g_assert (seekable);
+
+ gst_object_unref (target);
+}
+
+static GstEvent *
+translate_seek (ReplayBin * self, GstPad * pad, GstEvent * ievent)
+{
+ GstEvent *oevent = NULL;
+ gdouble rate;
+ GstFormat format;
+ GstSeekFlags flags;
+ GstSeekType start_type, stop_type;
+ gint64 start, stop;
+ gint64 istart, istop; /* Incoming */
+ gint64 ustart, ustop; /* Upstream */
+ gint64 ostart, ostop; /* Outgoing */
+ guint32 seqnum = gst_event_get_seqnum (ievent);
+
+ gst_event_parse_seek (ievent, &rate, &format, &flags, &start_type, &start,
+ &stop_type, &stop);
+
+ if (!GST_CLOCK_TIME_IS_VALID (stop))
+ stop = END_DATE;
+
+ gst_event_parse_seek_trickmode_interval (ievent, &self->trickmode_interval);
+
+ istart = start;
+ istop = stop;
+
+ query_seekable (pad, &ustart, &ustop);
+
+ if (rate > 0) {
+ /* First, from where we should seek the file */
+ ostart = istart % (ustop + INTERVAL);
+
+ /* This may end up in our empty interval */
+ if (ostart > ustop) {
+ istart += ostart - ustop;
+ ostart = 0;
+ }
+
+ /* Then, up to where we should seek it */
+ ostop = MIN (ustop, ostart + (istop - istart));
+ } else {
+ /* First up to where we should seek the file */
+ ostop = istop % (ustop + INTERVAL);
+
+ /* This may end up in our empty interval */
+ if (ostop > ustop) {
+ istop -= ostop - ustop;
+ ostop = ustop;
+ }
+
+ ostart = MAX (0, ostop - (istop - istart));
+ }
+
+ /* We may be left with nothing to actually play, in this
+ * case we won't seek upstream, and emit the expected events
+ * ourselves */
+ if (istart > istop) {
+ GstSegment segment;
+ GstEvent *event;
+ gboolean update;
+
+ event = gst_event_new_flush_start ();
+ gst_event_set_seqnum (event, seqnum);
+ gst_pad_push_event (pad, event);
+
+ event = gst_event_new_flush_stop (TRUE);
+ gst_event_set_seqnum (event, seqnum);
+ gst_pad_push_event (pad, event);
+
+ gst_segment_init (&segment, format);
+ gst_segment_do_seek (&segment, rate, format, flags, start_type, start,
+ stop_type, stop, &update);
+
+ event = gst_event_new_segment (&segment);
+ gst_event_set_seqnum (event, seqnum);
+ gst_pad_push_event (pad, event);
+
+ event = gst_event_new_eos ();
+ gst_event_set_seqnum (event, seqnum);
+ gst_pad_push_event (pad, event);
+
+ goto done;
+ }
+
+ /* Lastly, how much will remain to play back (this remainder includes the interval) */
+ if (stop - start > ostop - ostart)
+ self->remainder = (stop - start) - (ostop - ostart);
+
+ flags |= GST_SEEK_FLAG_SEGMENT;
+
+ oevent =
+ gst_event_new_seek (rate, format, flags, start_type, ostart, stop_type,
+ ostop);
+ gst_event_set_seek_trickmode_interval (oevent, self->trickmode_interval);
+ gst_event_set_seqnum (oevent, seqnum);
+
+ GST_DEBUG ("Translated event to %" GST_PTR_FORMAT
+ " (remainder: %" G_GINT64_FORMAT ")", oevent, self->remainder);
+
+done:
+ return oevent;
+}
+
+static gboolean
+replay_bin_event_func (GstPad * pad, GstObject * parent, GstEvent * event)
+{
+ ReplayBin *self = REPLAY_BIN (parent);
+ gboolean ret = TRUE;
+ gboolean forward = TRUE;
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEEK:
+ {
+ GST_DEBUG ("Processing seek event %" GST_PTR_FORMAT, event);
+
+ self->incoming_seek = event;
+
+ gst_event_replace (&self->outgoing_seek, NULL);
+ self->sent_segment = FALSE;
+
+ event = translate_seek (self, pad, event);
+
+ if (!event)
+ forward = FALSE;
+ else
+ self->outgoing_seek = gst_event_ref (event);
+ break;
+ }
+ default:
+ break;
+ }
+
+ if (forward)
+ return gst_pad_event_default (pad, parent, event);
+ else
+ return ret;
+}
+
+static gboolean
+replay_bin_query_func (GstPad * pad, GstObject * parent, GstQuery * query)
+{
+ ReplayBin *self = REPLAY_BIN (parent);
+ gboolean ret = TRUE;
+ gboolean forward = TRUE;
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_SEEKING:
+ /* We are seekable from the beginning till the end of time */
+ gst_query_set_seeking (query, GST_FORMAT_TIME, TRUE, 0,
+ GST_CLOCK_TIME_NONE);
+ forward = FALSE;
+ break;
+ case GST_QUERY_SEGMENT:
+ gst_query_set_segment (query, self->segment.rate, self->segment.format,
+ self->segment.start, self->segment.stop);
+ forward = FALSE;
+ default:
+ break;
+ }
+
+ GST_DEBUG ("Processed query %" GST_PTR_FORMAT, query);
+
+ if (forward)
+ return gst_pad_query_default (pad, parent, query);
+ else
+ return ret;
+}
+
+static GstEvent *
+translate_segment (GstPad * pad, GstEvent * ievent)
+{
+ ReplayBin *self = REPLAY_BIN (GST_OBJECT_PARENT (pad));
+ GstEvent *ret;
+ gdouble irate, orate;
+ GstFormat iformat, oformat;
+ GstSeekFlags iflags, oflags;
+ GstSeekType istart_type, ostart_type, istop_type, ostop_type;
+ gint64 istart, ostart, istop, ostop;
+ gboolean update;
+
+ gst_event_parse_segment (ievent, &self->incoming_segment);
+
+ if (!self->outgoing_seek) {
+ GstSegment segment;
+ gboolean update;
+
+ gst_segment_init (&segment, GST_FORMAT_TIME);
+
+ gst_segment_do_seek (&segment, 1.0, GST_FORMAT_TIME, 0, GST_SEEK_TYPE_SET,
+ 0, GST_SEEK_TYPE_SET, END_DATE, &update);
+
+ ret = gst_event_new_segment (&segment);
+ gst_event_unref (ievent);
+ goto done;
+ }
+
+ if (!self->sent_segment) {
+ gst_event_parse_seek (self->incoming_seek, &irate, &iformat, &iflags,
+ &istart_type, &istart, &istop_type, &istop);
+ gst_event_parse_seek (self->outgoing_seek, &orate, &oformat, &oflags,
+ &ostart_type, &ostart, &ostop_type, &ostop);
+
+ if (istop == -1)
+ istop = END_DATE;
+
+ if (self->incoming_segment->rate > 0)
+ self->ts_offset = istart - ostart;
+ else
+ self->ts_offset = istop - ostop;
+
+ istart += self->incoming_segment->start - ostart;
+ istop += self->incoming_segment->stop - ostop;
+
+ gst_segment_init (&self->segment, self->incoming_segment->format);
+
+ gst_segment_do_seek (&self->segment, self->incoming_segment->rate,
+ self->incoming_segment->format,
+ (GstSeekFlags) self->incoming_segment->flags, GST_SEEK_TYPE_SET,
+ (guint64) istart, GST_SEEK_TYPE_SET, (guint64) istop, &update);
+
+ self->min_pts = istart;
+
+ ret = gst_event_new_segment (&self->segment);
+
+ self->sent_segment = TRUE;
+
+ gst_event_unref (ievent);
+
+ GST_DEBUG ("Translated segment: %" GST_PTR_FORMAT ", "
+ "ts_offset: %" G_GUINT64_FORMAT, ret, self->ts_offset);
+ } else {
+ ret = NULL;
+ }
+
+done:
+ return ret;
+}
+
+static void
+handle_segment_done (ReplayBin * self, GstPad * pad)
+{
+ GstEvent *event;
+
+ if (self->remainder < INTERVAL) {
+ self->remainder = 0;
+ event = gst_event_new_eos ();
+ gst_event_set_seqnum (event, gst_event_get_seqnum (self->incoming_seek));
+ gst_pad_push_event (pad, event);
+ } else {
+ gint64 ustart, ustop;
+ gint64 ostart, ostop;
+ GstPad *target;
+ GstStructure *s;
+
+ /* Signify the end of a contiguous section of recording */
+ s = gst_structure_new ("GstNtpOffset",
+ "ntp-offset", G_TYPE_UINT64, 0, "discont", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM, s);
+
+ gst_pad_push_event (pad, event);
+
+ query_seekable (pad, &ustart, &ustop);
+
+ self->remainder -= INTERVAL;
+
+ if (self->incoming_segment->rate > 0) {
+ ostart = 0;
+ ostop = MIN (ustop, self->remainder);
+ } else {
+ ostart = MAX (ustop - self->remainder, 0);
+ ostop = ustop;
+ }
+
+ self->remainder = MAX (self->remainder - ostop - ostart, 0);
+
+ event =
+ gst_event_new_seek (self->segment.rate, self->segment.format,
+ self->segment.flags & ~GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, ostart,
+ GST_SEEK_TYPE_SET, ostop);
+ gst_event_set_seek_trickmode_interval (event, self->trickmode_interval);
+
+ if (self->incoming_segment->rate > 0)
+ self->ts_offset += INTERVAL + ustop;
+ else
+ self->ts_offset -= INTERVAL + ustop;
+
+ GST_DEBUG ("New offset: %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (self->ts_offset));
+
+ GST_DEBUG ("Seeking to %" GST_PTR_FORMAT, event);
+ target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
+ gst_pad_send_event (target, event);
+ gst_object_unref (target);
+ }
+}
+
+static GstPadProbeReturn
+replay_bin_event_probe (GstPad * pad, GstPadProbeInfo * info, gpointer unused)
+{
+ ReplayBin *self = REPLAY_BIN (GST_OBJECT_PARENT (pad));
+ GstPadProbeReturn ret = GST_PAD_PROBE_OK;
+
+ GST_DEBUG ("Probed %" GST_PTR_FORMAT, info->data);
+
+ switch (GST_EVENT_TYPE (info->data)) {
+ case GST_EVENT_SEGMENT:
+ {
+ GstEvent *translated;
+
+ GST_DEBUG ("Probed segment %" GST_PTR_FORMAT, info->data);
+
+ translated = translate_segment (pad, GST_EVENT (info->data));
+ if (translated)
+ info->data = translated;
+ else
+ ret = GST_PAD_PROBE_HANDLED;
+
+ break;
+ }
+ case GST_EVENT_SEGMENT_DONE:
+ {
+ handle_segment_done (self, pad);
+ ret = GST_PAD_PROBE_HANDLED;
+ break;
+ }
+ default:
+ break;
+ }
+
+ return ret;
+}
+
+static GstPadProbeReturn
+replay_bin_buffer_probe (GstPad * pad, GstPadProbeInfo * info, gpointer unused)
+{
+ ReplayBin *self = REPLAY_BIN (GST_OBJECT_PARENT (pad));
+ GstPadProbeReturn ret = GST_PAD_PROBE_OK;
+
+ if (GST_BUFFER_PTS (info->data) > self->incoming_segment->stop) {
+ ret = GST_PAD_PROBE_DROP;
+ goto done;
+ }
+
+ if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_PTS (info->data)))
+ GST_BUFFER_PTS (info->data) += self->ts_offset;
+ if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_DTS (info->data)))
+ GST_BUFFER_DTS (info->data) += self->ts_offset;
+
+ GST_LOG ("Pushing buffer %" GST_PTR_FORMAT, info->data);
+
+done:
+ return ret;
+}
+
+static GstElement *
+create_replay_bin (GstElement * parent)
+{
+ GstElement *ret, *src, *demux;
+ GstPad *ghost;
+
+ ret = replay_bin_new ();
+ if (!gst_bin_add (GST_BIN (parent), ret)) {
+ gst_object_unref (ret);
+ goto fail;
+ }
+
+ MAKE_AND_ADD (src, ret, "filesrc", fail, NULL);
+ MAKE_AND_ADD (demux, ret, "qtdemux", fail, NULL);
+
+ ghost = gst_ghost_pad_new_no_target ("src", GST_PAD_SRC);
+ gst_element_add_pad (ret, ghost);
+
+ gst_pad_set_event_function (ghost, replay_bin_event_func);
+ gst_pad_add_probe (ghost, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
+ replay_bin_event_probe, NULL, NULL);
+ gst_pad_add_probe (ghost, GST_PAD_PROBE_TYPE_BUFFER, replay_bin_buffer_probe,
+ NULL, NULL);
+ gst_pad_set_query_function (ghost, replay_bin_query_func);
+
+ if (!gst_element_link (src, demux))
+ goto fail;
+
+ g_object_set (src, "location", filename, NULL);
+ g_signal_connect (demux, "pad-added", G_CALLBACK (demux_pad_added_cb), ghost);
+
+done:
+ return ret;
+
+fail:
+ ret = NULL;
+ goto done;
+}
+
+/* A simple factory to set up our replay bin */
+
+struct _OnvifFactory
+{
+ GstRTSPOnvifMediaFactory parent;
+};
+
+G_DEFINE_TYPE (OnvifFactory, onvif_factory, GST_TYPE_RTSP_MEDIA_FACTORY);
+
+static void
+onvif_factory_init (OnvifFactory * factory)
+{
+}
+
+static GstElement *
+onvif_factory_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstElement *replay_bin, *q1, *parse, *pay, *onvifts, *q2;
+ GstElement *ret = gst_bin_new (NULL);
+ GstElement *pbin = gst_bin_new ("pay0");
+ GstPad *sinkpad, *srcpad;
+
+ if (!(replay_bin = create_replay_bin (ret)))
+ goto fail;
+
+ MAKE_AND_ADD (q1, pbin, "queue", fail, NULL);
+ MAKE_AND_ADD (parse, pbin, "h264parse", fail, NULL);
+ MAKE_AND_ADD (pay, pbin, "rtph264pay", fail, NULL);
+ MAKE_AND_ADD (onvifts, pbin, "rtponviftimestamp", fail, NULL);
+ MAKE_AND_ADD (q2, pbin, "queue", fail, NULL);
+
+ gst_bin_add (GST_BIN (ret), pbin);
+
+ if (!gst_element_link_many (q1, parse, pay, onvifts, q2, NULL))
+ goto fail;
+
+ sinkpad = gst_element_get_static_pad (q1, "sink");
+ gst_element_add_pad (pbin, gst_ghost_pad_new ("sink", sinkpad));
+ gst_object_unref (sinkpad);
+
+ if (!gst_element_link (replay_bin, pbin))
+ goto fail;
+
+ srcpad = gst_element_get_static_pad (q2, "src");
+ gst_element_add_pad (pbin, gst_ghost_pad_new ("src", srcpad));
+ gst_object_unref (srcpad);
+
+ g_object_set (onvifts, "set-t-bit", TRUE, "set-e-bit", TRUE, "ntp-offset",
+ G_GUINT64_CONSTANT (0), "drop-out-of-segment", FALSE, NULL);
+
+ gst_element_set_clock (onvifts, gst_system_clock_obtain ());
+
+done:
+ return ret;
+
+fail:
+ gst_object_unref (ret);
+ ret = NULL;
+ goto done;
+}
+
+static void
+onvif_factory_class_init (OnvifFactoryClass * klass)
+{
+ GstRTSPMediaFactoryClass *mf_class = GST_RTSP_MEDIA_FACTORY_CLASS (klass);
+
+ mf_class->create_element = onvif_factory_create_element;
+}
+
+static GstRTSPMediaFactory *
+onvif_factory_new (void)
+{
+ GstRTSPMediaFactory *result;
+
+ result =
+ GST_RTSP_MEDIA_FACTORY (g_object_new (onvif_factory_get_type (), NULL));
+
+ return result;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GOptionContext *optctx;
+ GError *error = NULL;
+ gchar *service;
+
+ optctx = g_option_context_new ("<filename.mp4> - ONVIF RTSP Server, MP4");
+ g_option_context_add_group (optctx, gst_init_get_option_group ());
+ if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
+ g_printerr ("Error parsing options: %s\n", error->message);
+ g_option_context_free (optctx);
+ g_clear_error (&error);
+ return -1;
+ }
+ if (argc < 2) {
+ g_print ("%s\n", g_option_context_get_help (optctx, TRUE, NULL));
+ return 1;
+ }
+ filename = argv[1];
+ g_option_context_free (optctx);
+
+ GST_DEBUG_CATEGORY_INIT (onvif_server_debug, "onvif-server", 0,
+ "ONVIF server");
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ server = gst_rtsp_onvif_server_new ();
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = onvif_factory_new ();
+ gst_rtsp_media_factory_set_media_gtype (factory, GST_TYPE_RTSP_ONVIF_MEDIA);
+
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ g_object_unref (mounts);
+
+ gst_rtsp_server_attach (server, NULL);
+
+ service = gst_rtsp_server_get_service (server);
+ g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", service);
+ g_free (service);
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2019 Mathieu Duponchelle <mathieu@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+G_BEGIN_DECLS
+
+G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
+
+G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
+ GstRTSPOnvifMediaFactory);
+
+G_END_DECLS
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc is-live=1 ! x264enc ! rtph264pay name=pay0 pt=96 )");
+
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2015 Centricular Ltd
+ * Author: Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+/* define this if you want the server to use TLS */
+//#define WITH_TLS
+
+#define DEFAULT_RTSP_PORT "8554"
+
+static char *port = (char *) DEFAULT_RTSP_PORT;
+
+static GOptionEntry entries[] = {
+ {"port", 'p', 0, G_OPTION_ARG_STRING, &port,
+ "Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
+ {NULL}
+};
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GOptionContext *optctx;
+ GError *error = NULL;
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ gchar *basic;
+#ifdef WITH_TLS
+ GTlsCertificate *cert;
+#endif
+
+ optctx = g_option_context_new ("<launch line> - Test RTSP Server, Launch\n\n"
+ "Example: \"( decodebin name=depay0 ! autovideosink )\"");
+ g_option_context_add_main_entries (optctx, entries, NULL);
+ g_option_context_add_group (optctx, gst_init_get_option_group ());
+ if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
+ g_printerr ("Error parsing options: %s\n", error->message);
+ return -1;
+ }
+
+ if (argc < 2) {
+ g_print ("%s\n", g_option_context_get_help (optctx, TRUE, NULL));
+ return 1;
+ }
+ g_option_context_free (optctx);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+ g_object_set (server, "service", port, NULL);
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named depay%d. Each
+ * element with depay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_transport_mode (factory,
+ GST_RTSP_TRANSPORT_MODE_RECORD);
+ gst_rtsp_media_factory_set_launch (factory, argv[1]);
+ gst_rtsp_media_factory_set_latency (factory, 2000);
+#ifdef WITH_TLS
+ gst_rtsp_media_factory_set_profiles (factory,
+ GST_RTSP_PROFILE_SAVP | GST_RTSP_PROFILE_SAVPF);
+#else
+ gst_rtsp_media_factory_set_profiles (factory,
+ GST_RTSP_PROFILE_AVP | GST_RTSP_PROFILE_AVPF);
+#endif
+
+ /* allow user to access this resource */
+ gst_rtsp_media_factory_add_role (factory, "user",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+ /* Anonymous users can see but not construct, so get UNAUTHORIZED */
+ gst_rtsp_media_factory_add_role (factory, "anonymous",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, FALSE, NULL);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* Set up the auth for user account */
+ /* make a new authentication manager */
+ auth = gst_rtsp_auth_new ();
+#ifdef WITH_TLS
+ cert = g_tls_certificate_new_from_pem ("-----BEGIN CERTIFICATE-----"
+ "MIICJjCCAY+gAwIBAgIBBzANBgkqhkiG9w0BAQUFADCBhjETMBEGCgmSJomT8ixk"
+ "ARkWA0NPTTEXMBUGCgmSJomT8ixkARkWB0VYQU1QTEUxHjAcBgNVBAsTFUNlcnRp"
+ "ZmljYXRlIEF1dGhvcml0eTEXMBUGA1UEAxMOY2EuZXhhbXBsZS5jb20xHTAbBgkq"
+ "hkiG9w0BCQEWDmNhQGV4YW1wbGUuY29tMB4XDTExMDExNzE5NDcxN1oXDTIxMDEx"
+ "NDE5NDcxN1owSzETMBEGCgmSJomT8ixkARkWA0NPTTEXMBUGCgmSJomT8ixkARkW"
+ "B0VYQU1QTEUxGzAZBgNVBAMTEnNlcnZlci5leGFtcGxlLmNvbTBcMA0GCSqGSIb3"
+ "DQEBAQUAA0sAMEgCQQDYScTxk55XBmbDM9zzwO+grVySE4rudWuzH2PpObIonqbf"
+ "hRoAalKVluG9jvbHI81eXxCdSObv1KBP1sbN5RzpAgMBAAGjIjAgMAkGA1UdEwQC"
+ "MAAwEwYDVR0lBAwwCgYIKwYBBQUHAwEwDQYJKoZIhvcNAQEFBQADgYEAYx6fMqT1"
+ "Gvo0jq88E8mc+bmp4LfXD4wJ7KxYeadQxt75HFRpj4FhFO3DOpVRFgzHlOEo3Fwk"
+ "PZOKjvkT0cbcoEq5whLH25dHoQxGoVQgFyAP5s+7Vp5AlHh8Y/vAoXeEVyy/RCIH"
+ "QkhUlAflfDMcrrYjsmwoOPSjhx6Mm/AopX4="
+ "-----END CERTIFICATE-----"
+ "-----BEGIN PRIVATE KEY-----"
+ "MIIBVAIBADANBgkqhkiG9w0BAQEFAASCAT4wggE6AgEAAkEA2EnE8ZOeVwZmwzPc"
+ "88DvoK1ckhOK7nVrsx9j6TmyKJ6m34UaAGpSlZbhvY72xyPNXl8QnUjm79SgT9bG"
+ "zeUc6QIDAQABAkBRFJZ32VbqWMP9OVwDJLiwC01AlYLnka0mIQZbT/2xq9dUc9GW"
+ "U3kiVw4lL8v/+sPjtTPCYYdzHHOyDen6znVhAiEA9qJT7BtQvRxCvGrAhr9MS022"
+ "tTdPbW829BoUtIeH64cCIQDggG5i48v7HPacPBIH1RaSVhXl8qHCpQD3qrIw3FMw"
+ "DwIga8PqH5Sf5sHedy2+CiK0V4MRfoU4c3zQ6kArI+bEgSkCIQCLA1vXBiE31B5s"
+ "bdHoYa1BXebfZVd+1Hd95IfEM5mbRwIgSkDuQwV55BBlvWph3U8wVIMIb4GStaH8"
+ "W535W8UBbEg=" "-----END PRIVATE KEY-----", -1, &error);
+ if (cert == NULL) {
+ g_printerr ("failed to parse PEM: %s\n", error->message);
+ return -1;
+ }
+ gst_rtsp_auth_set_tls_certificate (auth, cert);
+ g_object_unref (cert);
+#endif
+
+ /* make default token - anonymous unauthenticated access */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "anonymous", NULL);
+ gst_rtsp_auth_set_default_token (auth, token);
+ gst_rtsp_token_unref (token);
+
+ /* make user token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ basic = gst_rtsp_auth_make_basic ("user", "password");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* set as the server authentication manager */
+ gst_rtsp_server_set_auth (server, auth);
+ g_object_unref (auth);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+#ifdef WITH_TLS
+ g_print ("stream ready at rtsps://127.0.0.1:%s/test\n", port);
+#else
+ g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", port);
+#endif
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2015 Centricular Ltd
+ * Author: Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+#define DEFAULT_RTSP_PORT "8554"
+
+static char *port = (char *) DEFAULT_RTSP_PORT;
+
+static GOptionEntry entries[] = {
+ {"port", 'p', 0, G_OPTION_ARG_STRING, &port,
+ "Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
+ {NULL}
+};
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GOptionContext *optctx;
+ GError *error = NULL;
+
+ optctx = g_option_context_new ("<launch line> - Test RTSP Server, Launch\n\n"
+ "Example: \"( decodebin name=depay0 ! autovideosink )\"");
+ g_option_context_add_main_entries (optctx, entries, NULL);
+ g_option_context_add_group (optctx, gst_init_get_option_group ());
+ if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
+ g_printerr ("Error parsing options: %s\n", error->message);
+ g_option_context_free (optctx);
+ g_clear_error (&error);
+ return -1;
+ }
+
+ if (argc < 2) {
+ g_print ("%s\n", g_option_context_get_help (optctx, TRUE, NULL));
+ return 1;
+ }
+ g_option_context_free (optctx);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ g_object_set (server, "service", port, NULL);
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named depay%d. Each
+ * element with depay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_transport_mode (factory,
+ GST_RTSP_TRANSPORT_MODE_RECORD);
+ gst_rtsp_media_factory_set_launch (factory, argv[1]);
+ gst_rtsp_media_factory_set_latency (factory, 2000);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", port);
+ g_print ("On the sender, send a stream with rtspclientsink:\n"
+ " gst-launch-1.0 videotestsrc ! x264enc ! rtspclientsink location=rtsp://127.0.0.1:%s/test\n",
+ port);
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2019 Mathieu Duponchelle <mathieu@centricular.com>
+ * Copyright (C) 2020 Seungha Yang <seungha@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+#include "test-replay-server.h"
+
+GST_DEBUG_CATEGORY_STATIC (replay_server_debug);
+#define GST_CAT_DEFAULT (replay_server_debug)
+
+static GstStaticCaps raw_video_caps = GST_STATIC_CAPS ("video/x-raw");
+static GstStaticCaps raw_audio_caps = GST_STATIC_CAPS ("audio/x-raw");
+
+static GList
+ * gst_rtsp_media_factory_replay_get_demuxers (GstRTSPMediaFactoryReplay *
+ factory);
+static GList
+ * gst_rtsp_media_factory_replay_get_payloaders (GstRTSPMediaFactoryReplay *
+ factory);
+static GList
+ * gst_rtsp_media_factory_replay_get_decoders (GstRTSPMediaFactoryReplay *
+ factory);
+
+typedef struct
+{
+ GstPad *srcpad;
+ gulong block_id;
+} GstReplayBinPad;
+
+static void
+gst_replay_bin_pad_unblock_and_free (GstReplayBinPad * pad)
+{
+ if (pad->srcpad && pad->block_id) {
+ GST_DEBUG_OBJECT (pad->srcpad, "Unblock");
+ gst_pad_remove_probe (pad->srcpad, pad->block_id);
+ pad->block_id = 0;
+ }
+
+ gst_clear_object (&pad->srcpad);
+ g_free (pad);
+}
+
+/* NOTE: this bin implementation is almost completely taken from rtsp-media-factory-uri
+ * but this example doesn't use the GstRTSPMediaFactoryURI object so that
+ * we can handle events and messages ourselves.
+ * Specifically,
+ * - Handle segment-done message for looping given source
+ * - Drop all incoming seek event because client seek is not implemented
+ * and do initial segment seeking on no-more-pads signal
+ */
+struct _GstReplayBin
+{
+ GstBin parent;
+
+ gint64 num_loops;
+
+ GstCaps *raw_vcaps;
+ GstCaps *raw_acaps;
+
+ guint pt;
+
+ /* without ref */
+ GstElement *uridecodebin;
+ GstElement *inner_bin;
+
+ /* holds ref */
+ GstRTSPMediaFactoryReplay *factory;
+
+ GMutex lock;
+
+ GList *srcpads;
+};
+
+static void gst_replay_bin_dispose (GObject * object);
+static void gst_replay_bin_finalize (GObject * object);
+static void gst_replay_bin_handle_message (GstBin * bin, GstMessage * message);
+
+static gboolean autoplug_continue_cb (GstElement * dbin, GstPad * pad,
+ GstCaps * caps, GstReplayBin * self);
+static void pad_added_cb (GstElement * dbin, GstPad * pad, GstReplayBin * self);
+static void no_more_pads_cb (GstElement * uribin, GstReplayBin * self);
+
+#define gst_replay_bin_parent_class bin_parent_class
+G_DEFINE_TYPE (GstReplayBin, gst_replay_bin, GST_TYPE_BIN);
+
+static void
+gst_replay_bin_class_init (GstReplayBinClass * klass)
+{
+ GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+ GstBinClass *bin_class = GST_BIN_CLASS (klass);
+
+ gobject_class->dispose = gst_replay_bin_dispose;
+ gobject_class->finalize = gst_replay_bin_finalize;
+
+ bin_class->handle_message = GST_DEBUG_FUNCPTR (gst_replay_bin_handle_message);
+}
+
+static void
+gst_replay_bin_init (GstReplayBin * self)
+{
+ self->raw_vcaps = gst_static_caps_get (&raw_video_caps);
+ self->raw_acaps = gst_static_caps_get (&raw_audio_caps);
+
+ self->uridecodebin = gst_element_factory_make ("uridecodebin", NULL);
+ if (!self->uridecodebin) {
+ GST_ERROR_OBJECT (self, "uridecodebin is unavailable");
+ return;
+ }
+
+ /* our bin will dynamically expose payloaded pads */
+ self->inner_bin = gst_bin_new ("dynpay0");
+ gst_bin_add (GST_BIN_CAST (self), self->inner_bin);
+ gst_bin_add (GST_BIN_CAST (self->inner_bin), self->uridecodebin);
+
+ g_signal_connect (self->uridecodebin, "autoplug-continue",
+ G_CALLBACK (autoplug_continue_cb), self);
+ g_signal_connect (self->uridecodebin, "pad-added",
+ G_CALLBACK (pad_added_cb), self);
+ g_signal_connect (self->uridecodebin, "no-more-pads",
+ G_CALLBACK (no_more_pads_cb), self);
+
+ self->pt = 96;
+
+ g_mutex_init (&self->lock);
+}
+
+static void
+gst_replay_bin_dispose (GObject * object)
+{
+ GstReplayBin *self = GST_REPLAY_BIN (object);
+
+ GST_DEBUG_OBJECT (self, "dispose");
+
+ gst_clear_caps (&self->raw_vcaps);
+ gst_clear_caps (&self->raw_acaps);
+ gst_clear_object (&self->factory);
+
+ if (self->srcpads) {
+ g_list_free_full (self->srcpads,
+ (GDestroyNotify) gst_replay_bin_pad_unblock_and_free);
+ self->srcpads = NULL;
+ }
+
+ G_OBJECT_CLASS (bin_parent_class)->dispose (object);
+}
+
+static void
+gst_replay_bin_finalize (GObject * object)
+{
+ GstReplayBin *self = GST_REPLAY_BIN (object);
+
+ g_mutex_clear (&self->lock);
+
+ G_OBJECT_CLASS (bin_parent_class)->finalize (object);
+}
+
+static gboolean
+send_eos_foreach_srcpad (GstElement * element, GstPad * pad, gpointer user_data)
+{
+ GST_DEBUG_OBJECT (pad, "Sending EOS to downstream");
+ gst_pad_push_event (pad, gst_event_new_eos ());
+
+ return TRUE;
+}
+
+static void
+gst_replay_bin_do_segment_seek (GstElement * element, GstReplayBin * self)
+{
+ gboolean ret;
+
+ ret = gst_element_seek (element, 1.0, GST_FORMAT_TIME,
+ GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_SEGMENT,
+ GST_SEEK_TYPE_SET, 0, GST_SEEK_TYPE_NONE, -1);
+
+ if (!ret) {
+ GST_WARNING_OBJECT (self, "segment seeking failed");
+ gst_element_foreach_src_pad (element,
+ (GstElementForeachPadFunc) send_eos_foreach_srcpad, NULL);
+ }
+}
+
+static void
+gst_replay_bin_handle_message (GstBin * bin, GstMessage * message)
+{
+ GstReplayBin *self = GST_REPLAY_BIN (bin);
+
+ if (GST_MESSAGE_TYPE (message) == GST_MESSAGE_SEGMENT_DONE) {
+ gboolean next_loop = TRUE;
+
+ GST_DEBUG_OBJECT (self, "Have segment done message");
+
+ g_mutex_lock (&self->lock);
+ if (self->num_loops != -1) {
+ self->num_loops--;
+
+ if (self->num_loops < 1)
+ next_loop = FALSE;
+ }
+
+ if (next_loop) {
+ /* Send seek event from non-streaming thread */
+ gst_element_call_async (GST_ELEMENT_CAST (self->uridecodebin),
+ (GstElementCallAsyncFunc) gst_replay_bin_do_segment_seek, self, NULL);
+ } else {
+ gst_element_foreach_src_pad (GST_ELEMENT_CAST (self->uridecodebin),
+ (GstElementForeachPadFunc) send_eos_foreach_srcpad, NULL);
+ }
+
+ g_mutex_unlock (&self->lock);
+ }
+
+ GST_BIN_CLASS (bin_parent_class)->handle_message (bin, message);
+}
+
+static GstElementFactory *
+find_payloader (GstReplayBin * self, GstCaps * caps)
+{
+ GList *list;
+ GstElementFactory *factory = NULL;
+ gboolean autoplug_more = FALSE;
+ GList *demuxers = NULL;
+ GList *payloaders = NULL;
+
+ demuxers = gst_rtsp_media_factory_replay_get_demuxers (self->factory);
+
+ /* first find a demuxer that can link */
+ list = gst_element_factory_list_filter (demuxers, caps, GST_PAD_SINK, FALSE);
+
+ if (list) {
+ GstStructure *structure = gst_caps_get_structure (caps, 0);
+ gboolean parsed = FALSE;
+ gint mpegversion = 0;
+
+ if (!gst_structure_get_boolean (structure, "parsed", &parsed) &&
+ gst_structure_has_name (structure, "audio/mpeg") &&
+ gst_structure_get_int (structure, "mpegversion", &mpegversion) &&
+ (mpegversion == 2 || mpegversion == 4)) {
+ /* for AAC it's framed=true instead of parsed=true */
+ gst_structure_get_boolean (structure, "framed", &parsed);
+ }
+
+ /* Avoid plugging parsers in a loop. This is not 100% correct, as some
+ * parsers don't set parsed=true in caps. We should do something like
+ * decodebin does and track decode chains and elements plugged in those
+ * chains...
+ */
+ if (parsed) {
+ GList *walk;
+ const gchar *klass;
+
+ for (walk = list; walk; walk = walk->next) {
+ factory = GST_ELEMENT_FACTORY (walk->data);
+ klass = gst_element_factory_get_metadata (factory,
+ GST_ELEMENT_METADATA_KLASS);
+ if (strstr (klass, "Parser"))
+ /* caps have parsed=true, so skip this parser to avoid loops */
+ continue;
+
+ autoplug_more = TRUE;
+ break;
+ }
+ } else {
+ /* caps don't have parsed=true set and we have a demuxer/parser */
+ autoplug_more = TRUE;
+ }
+
+ gst_plugin_feature_list_free (list);
+ }
+
+ if (autoplug_more)
+ /* we have a demuxer, try that one first */
+ return NULL;
+
+ payloaders = gst_rtsp_media_factory_replay_get_payloaders (self->factory);
+
+ /* no demuxer try a depayloader */
+ list = gst_element_factory_list_filter (payloaders,
+ caps, GST_PAD_SINK, FALSE);
+
+ if (list == NULL) {
+ GList *decoders =
+ gst_rtsp_media_factory_replay_get_decoders (self->factory);
+ /* no depayloader, try a decoder, we'll get to a payloader for a decoded
+ * video or audio format, worst case. */
+ list = gst_element_factory_list_filter (decoders,
+ caps, GST_PAD_SINK, FALSE);
+
+ if (list != NULL) {
+ /* we have a decoder, try that one first */
+ gst_plugin_feature_list_free (list);
+ return NULL;
+ }
+ }
+
+ if (list != NULL) {
+ factory = GST_ELEMENT_FACTORY_CAST (list->data);
+ g_object_ref (factory);
+ gst_plugin_feature_list_free (list);
+ }
+
+ return factory;
+}
+
+static gboolean
+autoplug_continue_cb (GstElement * dbin, GstPad * pad, GstCaps * caps,
+ GstReplayBin * self)
+{
+ GstElementFactory *factory;
+
+ GST_DEBUG_OBJECT (self, "found pad %s:%s of caps %" GST_PTR_FORMAT,
+ GST_DEBUG_PAD_NAME (pad), caps);
+
+ if (!(factory = find_payloader (self, caps)))
+ goto no_factory;
+
+ /* we found a payloader, stop autoplugging so we can plug the
+ * payloader. */
+ GST_DEBUG_OBJECT (self, "found factory %s",
+ gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (factory)));
+ gst_object_unref (factory);
+
+ return FALSE;
+
+no_factory:
+ {
+ /* no payloader, continue autoplugging */
+ GST_DEBUG_OBJECT (self, "no payloader found for caps %" GST_PTR_FORMAT,
+ caps);
+ return TRUE;
+ }
+}
+
+static GstPadProbeReturn
+replay_bin_sink_probe (GstPad * pad, GstPadProbeInfo * info,
+ GstReplayBin * self)
+{
+ GstPadProbeReturn ret = GST_PAD_PROBE_OK;
+
+ if (GST_IS_EVENT (GST_PAD_PROBE_INFO_DATA (info))) {
+ GstEvent *event = GST_PAD_PROBE_INFO_EVENT (info);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEEK:
+ /* Ideally this shouldn't happen because we are responding
+ * seeking query with non-seekable */
+ GST_DEBUG_OBJECT (pad, "Drop seek event");
+ ret = GST_PAD_PROBE_DROP;
+ break;
+ default:
+ break;
+ }
+ } else if (GST_IS_QUERY (GST_PAD_PROBE_INFO_DATA (info))) {
+ GstQuery *query = GST_PAD_PROBE_INFO_QUERY (info);
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_SEEKING:
+ {
+ /* FIXME: client seek is not implemented */
+ gst_query_set_seeking (query, GST_FORMAT_TIME, FALSE, 0,
+ GST_CLOCK_TIME_NONE);
+ ret = GST_PAD_PROBE_HANDLED;
+ break;
+ }
+ case GST_QUERY_SEGMENT:
+ /* client seeking is not considered in here */
+ gst_query_set_segment (query,
+ 1.0, GST_FORMAT_TIME, 0, GST_CLOCK_TIME_NONE);
+ ret = GST_PAD_PROBE_HANDLED;
+ break;
+ default:
+ break;
+ }
+ }
+
+ return ret;
+}
+
+static GstPadProbeReturn
+replay_bin_src_block (GstPad * pad, GstPadProbeInfo * info, GstReplayBin * self)
+{
+ GST_DEBUG_OBJECT (pad, "Block pad");
+
+ return GST_PAD_PROBE_OK;
+}
+
+static void
+pad_added_cb (GstElement * dbin, GstPad * pad, GstReplayBin * self)
+{
+ GstElementFactory *factory;
+ GstElement *payloader;
+ GstCaps *caps;
+ GstPad *sinkpad, *srcpad, *ghostpad;
+ GstPad *dpad = pad;
+ GstElement *convert;
+ gchar *padname, *payloader_name;
+ GstElement *inner_bin = self->inner_bin;
+ GstReplayBinPad *bin_pad;
+
+ GST_DEBUG_OBJECT (self, "added pad %s:%s", GST_DEBUG_PAD_NAME (pad));
+
+ /* ref to make refcounting easier later */
+ gst_object_ref (pad);
+ padname = gst_pad_get_name (pad);
+
+ /* get pad caps first, then call get_caps, then fail */
+ if ((caps = gst_pad_get_current_caps (pad)) == NULL)
+ if ((caps = gst_pad_query_caps (pad, NULL)) == NULL)
+ goto no_caps;
+
+ /* check for raw caps */
+ if (gst_caps_can_intersect (caps, self->raw_vcaps)) {
+ /* we have raw video caps, insert converter */
+ convert = gst_element_factory_make ("videoconvert", NULL);
+ } else if (gst_caps_can_intersect (caps, self->raw_acaps)) {
+ /* we have raw audio caps, insert converter */
+ convert = gst_element_factory_make ("audioconvert", NULL);
+ } else {
+ convert = NULL;
+ }
+
+ if (convert) {
+ gst_bin_add (GST_BIN_CAST (inner_bin), convert);
+ gst_element_sync_state_with_parent (convert);
+
+ sinkpad = gst_element_get_static_pad (convert, "sink");
+ gst_pad_link (pad, sinkpad);
+ gst_object_unref (sinkpad);
+
+ /* unref old pad, we reffed before */
+ gst_object_unref (pad);
+
+ /* continue with new pad and caps */
+ pad = gst_element_get_static_pad (convert, "src");
+ if ((caps = gst_pad_get_current_caps (pad)) == NULL)
+ if ((caps = gst_pad_query_caps (pad, NULL)) == NULL)
+ goto no_caps;
+ }
+
+ if (!(factory = find_payloader (self, caps)))
+ goto no_factory;
+
+ gst_caps_unref (caps);
+
+ /* we have a payloader now */
+ GST_DEBUG_OBJECT (self, "found payloader factory %s",
+ gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (factory)));
+
+ payloader_name = g_strdup_printf ("pay_%s", padname);
+ payloader = gst_element_factory_create (factory, payloader_name);
+ g_free (payloader_name);
+ if (payloader == NULL)
+ goto no_payloader;
+
+ g_object_set (payloader, "pt", self->pt, NULL);
+ self->pt++;
+
+ if (g_object_class_find_property (G_OBJECT_GET_CLASS (payloader),
+ "buffer-list"))
+ g_object_set (payloader, "buffer-list", TRUE, NULL);
+
+ /* add the payloader to the pipeline */
+ gst_bin_add (GST_BIN_CAST (inner_bin), payloader);
+ gst_element_sync_state_with_parent (payloader);
+
+ /* link the pad to the sinkpad of the payloader */
+ sinkpad = gst_element_get_static_pad (payloader, "sink");
+ gst_pad_link (pad, sinkpad);
+ gst_object_unref (pad);
+
+ /* Add pad probe to handle events */
+ gst_pad_add_probe (sinkpad,
+ GST_PAD_PROBE_TYPE_EVENT_UPSTREAM | GST_PAD_PROBE_TYPE_QUERY_UPSTREAM,
+ (GstPadProbeCallback) replay_bin_sink_probe, self, NULL);
+ gst_object_unref (sinkpad);
+
+ /* block data for initial segment seeking */
+ bin_pad = g_new0 (GstReplayBinPad, 1);
+
+ /* Move ownership of pad to this struct */
+ bin_pad->srcpad = gst_object_ref (dpad);
+ bin_pad->block_id =
+ gst_pad_add_probe (dpad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM,
+ (GstPadProbeCallback) replay_bin_src_block, self, NULL);
+ g_mutex_lock (&self->lock);
+ self->srcpads = g_list_append (self->srcpads, bin_pad);
+ g_mutex_unlock (&self->lock);
+
+ /* now expose the srcpad of the payloader as a ghostpad with the same name
+ * as the uridecodebin pad name. */
+ srcpad = gst_element_get_static_pad (payloader, "src");
+ ghostpad = gst_ghost_pad_new (padname, srcpad);
+ gst_object_unref (srcpad);
+ g_free (padname);
+
+ gst_pad_set_active (ghostpad, TRUE);
+ gst_element_add_pad (inner_bin, ghostpad);
+
+ return;
+
+ /* ERRORS */
+no_caps:
+ {
+ GST_WARNING ("could not get caps from pad");
+ g_free (padname);
+ gst_object_unref (pad);
+ return;
+ }
+no_factory:
+ {
+ GST_DEBUG ("no payloader found");
+ g_free (padname);
+ gst_caps_unref (caps);
+ gst_object_unref (pad);
+ return;
+ }
+no_payloader:
+ {
+ GST_ERROR ("could not create payloader from factory");
+ g_free (padname);
+ gst_caps_unref (caps);
+ gst_object_unref (pad);
+ return;
+ }
+}
+
+static void
+gst_replay_bin_do_initial_segment_seek (GstElement * element,
+ GstReplayBin * self)
+{
+ gboolean ret;
+ GstQuery *query;
+ gboolean seekable;
+
+ query = gst_query_new_seeking (GST_FORMAT_TIME);
+ ret = gst_element_query (element, query);
+
+ if (!ret) {
+ GST_WARNING_OBJECT (self, "Cannot query seeking");
+ gst_query_unref (query);
+ goto done;
+ }
+
+ gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL);
+ gst_query_unref (query);
+
+ if (!seekable) {
+ GST_WARNING_OBJECT (self, "Source is not seekable");
+ ret = FALSE;
+ goto done;
+ }
+
+ ret = gst_element_seek (element, 1.0, GST_FORMAT_TIME,
+ GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
+ GST_SEEK_TYPE_SET, 0, GST_SEEK_TYPE_NONE, -1);
+
+ if (!ret)
+ GST_WARNING_OBJECT (self, "segment seeking failed");
+
+done:
+ /* Unblock all pads then */
+ g_mutex_lock (&self->lock);
+ if (self->srcpads) {
+ g_list_free_full (self->srcpads,
+ (GDestroyNotify) gst_replay_bin_pad_unblock_and_free);
+ self->srcpads = NULL;
+ }
+ g_mutex_unlock (&self->lock);
+
+ if (!ret) {
+ GST_WARNING_OBJECT (self, "Sending eos to all pads");
+ gst_element_foreach_src_pad (element,
+ (GstElementForeachPadFunc) send_eos_foreach_srcpad, NULL);
+ }
+}
+
+static void
+no_more_pads_cb (GstElement * uribin, GstReplayBin * self)
+{
+ GST_DEBUG_OBJECT (self, "no-more-pads");
+ gst_element_no_more_pads (GST_ELEMENT_CAST (self->inner_bin));
+
+ /* Flush seeking from streaming thread might not be good idea.
+ * Do this from another (non-streaming) thread */
+ gst_element_call_async (GST_ELEMENT_CAST (self->uridecodebin),
+ (GstElementCallAsyncFunc) gst_replay_bin_do_initial_segment_seek,
+ self, NULL);
+}
+
+static GstElement *
+gst_replay_bin_new (const gchar * uri, gint64 num_loops,
+ GstRTSPMediaFactoryReplay * factory, const gchar * name)
+{
+ GstReplayBin *self;
+
+ g_return_val_if_fail (uri != NULL, NULL);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+ if (!name)
+ name = "GstRelayBin";
+
+ self = GST_REPLAY_BIN (g_object_new (GST_TYPE_REPLAY_BIN,
+ "name", name, NULL));
+
+ if (!self->uridecodebin) {
+ gst_object_unref (self);
+ return NULL;
+ }
+
+ g_object_set (self->uridecodebin, "uri", uri, NULL);
+ self->factory = g_object_ref (factory);
+ self->num_loops = num_loops;
+
+ return GST_ELEMENT_CAST (self);
+}
+
+struct _GstRTSPMediaFactoryReplay
+{
+ GstRTSPMediaFactory parent;
+
+ gchar *uri;
+
+ GList *demuxers;
+ GList *payloaders;
+ GList *decoders;
+
+ gint64 num_loops;
+};
+
+enum
+{
+ PROP_0,
+ PROP_URI,
+ PROP_NUM_LOOPS,
+};
+
+#define DEFAULT_NUM_LOOPS (-1)
+
+static void gst_rtsp_media_factory_replay_get_property (GObject * object,
+ guint propid, GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_replay_set_property (GObject * object,
+ guint propid, const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_replay_finalize (GObject * object);
+
+static GstElement
+ * gst_rtsp_media_factory_replay_create_element (GstRTSPMediaFactory *
+ factory, const GstRTSPUrl * url);
+
+typedef struct
+{
+ GList *demux;
+ GList *payload;
+ GList *decode;
+} FilterData;
+
+static gboolean
+payloader_filter (GstPluginFeature * feature, FilterData * self);
+
+#define gst_rtsp_media_factory_replay_parent_class parent_class
+G_DEFINE_TYPE (GstRTSPMediaFactoryReplay,
+ gst_rtsp_media_factory_replay, GST_TYPE_RTSP_MEDIA_FACTORY);
+
+static void
+gst_rtsp_media_factory_replay_class_init (GstRTSPMediaFactoryReplayClass
+ * klass)
+{
+ GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+ GstRTSPMediaFactoryClass *mf_class = GST_RTSP_MEDIA_FACTORY_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_media_factory_replay_get_property;
+ gobject_class->set_property = gst_rtsp_media_factory_replay_set_property;
+ gobject_class->finalize = gst_rtsp_media_factory_replay_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_URI,
+ g_param_spec_string ("uri", "URI",
+ "The URI of the resource to stream", NULL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_NUM_LOOPS,
+ g_param_spec_int64 ("num-loops", "Num Loops",
+ "The number of loops (-1 = infinite)", -1, G_MAXINT64,
+ DEFAULT_NUM_LOOPS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ mf_class->create_element =
+ GST_DEBUG_FUNCPTR (gst_rtsp_media_factory_replay_create_element);
+}
+
+static void
+gst_rtsp_media_factory_replay_init (GstRTSPMediaFactoryReplay * self)
+{
+ FilterData data = { NULL, };
+
+ /* get the feature list using the filter */
+ gst_registry_feature_filter (gst_registry_get (), (GstPluginFeatureFilter)
+ payloader_filter, FALSE, &data);
+
+ /* sort */
+ self->demuxers =
+ g_list_sort (data.demux, gst_plugin_feature_rank_compare_func);
+ self->payloaders =
+ g_list_sort (data.payload, gst_plugin_feature_rank_compare_func);
+ self->decoders =
+ g_list_sort (data.decode, gst_plugin_feature_rank_compare_func);
+
+ self->num_loops = DEFAULT_NUM_LOOPS;
+}
+
+static void
+gst_rtsp_media_factory_replay_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMediaFactoryReplay *self = GST_RTSP_MEDIA_FACTORY_REPLAY (object);
+
+ switch (propid) {
+ case PROP_URI:
+ g_value_take_string (value, self->uri);
+ break;
+ case PROP_NUM_LOOPS:
+ g_value_set_int64 (value, self->num_loops);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_media_factory_replay_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMediaFactoryReplay *self = GST_RTSP_MEDIA_FACTORY_REPLAY (object);
+
+ switch (propid) {
+ case PROP_URI:
+ g_free (self->uri);
+ self->uri = g_value_dup_string (value);
+ break;
+ case PROP_NUM_LOOPS:
+ self->num_loops = g_value_get_int64 (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_media_factory_replay_finalize (GObject * object)
+{
+ GstRTSPMediaFactoryReplay *self = GST_RTSP_MEDIA_FACTORY_REPLAY (object);
+
+ g_free (self->uri);
+
+ gst_plugin_feature_list_free (self->demuxers);
+ gst_plugin_feature_list_free (self->payloaders);
+ gst_plugin_feature_list_free (self->decoders);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static GstElement *
+gst_rtsp_media_factory_replay_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMediaFactoryReplay *self = GST_RTSP_MEDIA_FACTORY_REPLAY (factory);
+
+ return gst_replay_bin_new (self->uri, self->num_loops, self,
+ "GstRTSPMediaFactoryReplay");
+}
+
+static gboolean
+payloader_filter (GstPluginFeature * feature, FilterData * data)
+{
+ const gchar *klass;
+ GstElementFactory *fact;
+ GList **list = NULL;
+
+ /* we only care about element factories */
+ if (G_UNLIKELY (!GST_IS_ELEMENT_FACTORY (feature)))
+ return FALSE;
+
+ if (gst_plugin_feature_get_rank (feature) < GST_RANK_MARGINAL)
+ return FALSE;
+
+ fact = GST_ELEMENT_FACTORY_CAST (feature);
+
+ klass = gst_element_factory_get_metadata (fact, GST_ELEMENT_METADATA_KLASS);
+
+ if (strstr (klass, "Decoder"))
+ list = &data->decode;
+ else if (strstr (klass, "Demux"))
+ list = &data->demux;
+ else if (strstr (klass, "Parser") && strstr (klass, "Codec"))
+ list = &data->demux;
+ else if (strstr (klass, "Payloader") && strstr (klass, "RTP"))
+ list = &data->payload;
+
+ if (list) {
+ GST_LOG ("adding %s", GST_OBJECT_NAME (fact));
+ *list = g_list_prepend (*list, gst_object_ref (fact));
+ }
+
+ return FALSE;
+}
+
+static GList *
+gst_rtsp_media_factory_replay_get_demuxers (GstRTSPMediaFactoryReplay * factory)
+{
+ return factory->demuxers;
+}
+
+static GList *
+gst_rtsp_media_factory_replay_get_payloaders (GstRTSPMediaFactoryReplay *
+ factory)
+{
+ return factory->payloaders;
+}
+
+static GList *
+gst_rtsp_media_factory_replay_get_decoders (GstRTSPMediaFactoryReplay * factory)
+{
+ return factory->decoders;
+}
+
+static GstRTSPMediaFactory *
+gst_rtsp_media_factory_replay_new (const gchar * uri, gint64 num_loops)
+{
+ GstRTSPMediaFactory *factory;
+
+ factory =
+ GST_RTSP_MEDIA_FACTORY (g_object_new
+ (GST_TYPE_RTSP_MEDIA_FACTORY_REPLAY, "uri", uri, "num-loops", num_loops,
+ NULL));
+
+ return factory;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GOptionContext *optctx;
+ GError *error = NULL;
+ gchar *service;
+ gchar *uri = NULL;
+ gint64 num_loops = -1;
+ GOptionEntry options[] = {
+ {"num-loops", 0, 0, G_OPTION_ARG_INT64, &num_loops,
+ "The number of loops (default = -1, infinite)", NULL},
+ {NULL}
+ };
+
+ optctx = g_option_context_new ("RTSP Replay Server");
+ g_option_context_add_main_entries (optctx, options, NULL);
+ g_option_context_add_group (optctx, gst_init_get_option_group ());
+ if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
+ g_printerr ("Error parsing options: %s\n", error->message);
+ g_option_context_free (optctx);
+ g_clear_error (&error);
+ return -1;
+ }
+ if (argc < 2) {
+ g_print ("%s\n", g_option_context_get_help (optctx, TRUE, NULL));
+ return 1;
+ }
+
+ g_option_context_free (optctx);
+
+ /* check if URI is valid, otherwise convert filename to URI if it's a file */
+ if (gst_uri_is_valid (argv[1])) {
+ uri = g_strdup (argv[1]);
+ } else if (g_file_test (argv[1], G_FILE_TEST_EXISTS)) {
+ uri = gst_filename_to_uri (argv[1], NULL);
+ } else {
+ g_printerr ("Unrecognised command line argument '%s'.\n"
+ "Please pass an URI or file as argument!\n", argv[1]);
+ return -1;
+ }
+
+ if (num_loops < -1 || num_loops == 0) {
+ g_printerr ("num-loop should be non-zero or -1");
+ return -1;
+ }
+
+ GST_DEBUG_CATEGORY_INIT (replay_server_debug, "replay-server", 0,
+ "RTSP replay server");
+
+ if (num_loops != -1)
+ g_print ("Run loop %" G_GINT64_FORMAT " times\n", num_loops);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ server = gst_rtsp_server_new ();
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+ factory = gst_rtsp_media_factory_replay_new (uri, num_loops);
+ g_free (uri);
+
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ g_object_unref (mounts);
+
+ gst_rtsp_server_attach (server, NULL);
+
+ service = gst_rtsp_server_get_service (server);
+ g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", service);
+ g_free (service);
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2019 Mathieu Duponchelle <mathieu@centricular.com>
+ * Copyright (C) 2020 Seungha Yang <seungha@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_REPLAY_BIN (gst_replay_bin_get_type ())
+G_DECLARE_FINAL_TYPE (GstReplayBin, gst_replay_bin, GST, REPLAY_BIN, GstBin);
+
+#define GST_TYPE_RTSP_MEDIA_FACTORY_REPLAY (gst_rtsp_media_factory_replay_get_type ())
+G_DECLARE_FINAL_TYPE (GstRTSPMediaFactoryReplay,
+ gst_rtsp_media_factory_replay, GST, RTSP_MEDIA_FACTORY_REPLAY,
+ GstRTSPMediaFactory);
+
+G_END_DECLS
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2009 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+static void
+media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+ gst_rtsp_media_set_reusable (media, TRUE);
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ gchar *str;
+
+ gst_init (&argc, &argv);
+
+ if (argc < 2) {
+ g_message ("usage: %s <filename.sdp>", argv[0]);
+ return -1;
+ }
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+
+ str =
+ g_strdup_printf ("( filesrc location=%s ! sdpdemux name=dynpay0 )",
+ argv[1]);
+ gst_rtsp_media_factory_set_launch (factory, str);
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ g_signal_connect (factory, "media-configure", (GCallback) media_configure,
+ NULL);
+ g_free (str);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving */
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+#include <gst/rtsp-server/rtsp-media-factory-uri.h>
+
+#define DEFAULT_RTSP_PORT "8554"
+
+static char *port = (char *) DEFAULT_RTSP_PORT;
+
+static GOptionEntry entries[] = {
+ {"port", 'p', 0, G_OPTION_ARG_STRING, &port,
+ "Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
+ {NULL}
+};
+
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+#if 0
+static gboolean
+remove_map (GstRTSPServer * server)
+{
+ GstRTSPMountPoints *mounts;
+
+ g_print ("removing /test mount point\n");
+ mounts = gst_rtsp_server_get_mount_points (server);
+ gst_rtsp_mount_points_remove_factory (mounts, "/test");
+ g_object_unref (mounts);
+
+ return FALSE;
+}
+#endif
+
+int
+main (int argc, gchar * argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactoryURI *factory;
+ GOptionContext *optctx;
+ GError *error = NULL;
+ gchar *uri;
+
+ optctx = g_option_context_new ("<uri> - Test RTSP Server, URI");
+ g_option_context_add_main_entries (optctx, entries, NULL);
+ g_option_context_add_group (optctx, gst_init_get_option_group ());
+ if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
+ g_printerr ("Error parsing options: %s\n", error->message);
+ g_option_context_free (optctx);
+ g_clear_error (&error);
+ return -1;
+ }
+ g_option_context_free (optctx);
+
+ if (argc < 2) {
+ g_printerr ("Please pass an URI or file as argument!\n");
+ return -1;
+ }
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+ g_object_set (server, "service", port, NULL);
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a URI media factory for a test stream. */
+ factory = gst_rtsp_media_factory_uri_new ();
+
+ /* when using GStreamer as a client, one can use the gst payloader, which is
+ * more efficient when there is no payloader for the compressed format */
+ /* g_object_set (factory, "use-gstpay", TRUE, NULL); */
+
+ /* check if URI is valid, otherwise convert filename to URI if it's a file */
+ if (gst_uri_is_valid (argv[1])) {
+ uri = g_strdup (argv[1]);
+ } else if (g_file_test (argv[1], G_FILE_TEST_EXISTS)) {
+ uri = gst_filename_to_uri (argv[1], NULL);
+ } else {
+ g_printerr ("Unrecognised command line argument '%s'.\n"
+ "Please pass an URI or file as argument!\n", argv[1]);
+ return -1;
+ }
+
+ gst_rtsp_media_factory_uri_set_uri (factory, uri);
+ g_free (uri);
+
+ /* if you want multiple clients to see the same video, set the shared property
+ * to TRUE */
+ /* gst_rtsp_media_factory_set_shared ( GST_RTSP_MEDIA_FACTORY (factory), TRUE); */
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test",
+ GST_RTSP_MEDIA_FACTORY (factory));
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ /* do session cleanup every 2 seconds */
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+#if 0
+ /* remove the mount point after 10 seconds, new clients won't be able to use
+ * the /test url anymore */
+ g_timeout_add_seconds (10, (GSourceFunc) remove_map, server);
+#endif
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", port);
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2018 Jan Schmidt <jan at centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/* This example disconnects any clients and exits 10 seconds
+ * after the first client connects */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+guint exit_timeout_id = 0;
+
+/* define this if you want the resource to only be available when using
+ * user/password as the password */
+#undef WITH_AUTH
+
+/* define this if you want the server to use TLS (it will also need WITH_AUTH
+ * to be defined) */
+#undef WITH_TLS
+
+/* this timeout is periodically run to clean up the expired sessions from the
+ * pool. This needs to be run explicitly currently but might be done
+ * automatically as part of the mainloop. */
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+static GstRTSPFilterResult
+client_filter (GstRTSPServer * server, GstRTSPClient * client,
+ gpointer user_data)
+{
+ /* Simple filter that shuts down all clients. */
+ return GST_RTSP_FILTER_REMOVE;
+}
+
+/* Timeout that runs 10 seconds after the first client connects and triggers
+ * the shutdown of the server */
+static gboolean
+shutdown_timeout (GstRTSPServer * server)
+{
+ GstRTSPMountPoints *mounts;
+ g_print ("Time for everyone to go. Removing mount point\n");
+ /* Remove the mount point to prevent new clients connecting */
+ mounts = gst_rtsp_server_get_mount_points (server);
+ gst_rtsp_mount_points_remove_factory (mounts, "/test");
+ g_object_unref (mounts);
+
+ /* Filter existing clients and remove them */
+ g_print ("Disconnecting existing clients\n");
+ gst_rtsp_server_client_filter (server, client_filter, NULL);
+ return FALSE;
+}
+
+static void
+client_connected (GstRTSPServer * server, GstRTSPClient * client)
+{
+ if (exit_timeout_id == 0) {
+ g_print ("First Client connected. Disconnecting everyone in 10 seconds\n");
+ exit_timeout_id =
+ g_timeout_add_seconds (10, (GSourceFunc) shutdown_timeout, server);
+ }
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+#ifdef WITH_AUTH
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ gchar *basic;
+ GstRTSPPermissions *permissions;
+#endif
+#ifdef WITH_TLS
+ GTlsCertificate *cert;
+ GError *error = NULL;
+#endif
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+#ifdef WITH_AUTH
+ /* make a new authentication manager. it can be added to control access to all
+ * the factories on the server or on individual factories. */
+ auth = gst_rtsp_auth_new ();
+#ifdef WITH_TLS
+ cert = g_tls_certificate_new_from_pem ("-----BEGIN CERTIFICATE-----"
+ "MIICJjCCAY+gAwIBAgIBBzANBgkqhkiG9w0BAQUFADCBhjETMBEGCgmSJomT8ixk"
+ "ARkWA0NPTTEXMBUGCgmSJomT8ixkARkWB0VYQU1QTEUxHjAcBgNVBAsTFUNlcnRp"
+ "ZmljYXRlIEF1dGhvcml0eTEXMBUGA1UEAxMOY2EuZXhhbXBsZS5jb20xHTAbBgkq"
+ "hkiG9w0BCQEWDmNhQGV4YW1wbGUuY29tMB4XDTExMDExNzE5NDcxN1oXDTIxMDEx"
+ "NDE5NDcxN1owSzETMBEGCgmSJomT8ixkARkWA0NPTTEXMBUGCgmSJomT8ixkARkW"
+ "B0VYQU1QTEUxGzAZBgNVBAMTEnNlcnZlci5leGFtcGxlLmNvbTBcMA0GCSqGSIb3"
+ "DQEBAQUAA0sAMEgCQQDYScTxk55XBmbDM9zzwO+grVySE4rudWuzH2PpObIonqbf"
+ "hRoAalKVluG9jvbHI81eXxCdSObv1KBP1sbN5RzpAgMBAAGjIjAgMAkGA1UdEwQC"
+ "MAAwEwYDVR0lBAwwCgYIKwYBBQUHAwEwDQYJKoZIhvcNAQEFBQADgYEAYx6fMqT1"
+ "Gvo0jq88E8mc+bmp4LfXD4wJ7KxYeadQxt75HFRpj4FhFO3DOpVRFgzHlOEo3Fwk"
+ "PZOKjvkT0cbcoEq5whLH25dHoQxGoVQgFyAP5s+7Vp5AlHh8Y/vAoXeEVyy/RCIH"
+ "QkhUlAflfDMcrrYjsmwoOPSjhx6Mm/AopX4="
+ "-----END CERTIFICATE-----"
+ "-----BEGIN PRIVATE KEY-----"
+ "MIIBVAIBADANBgkqhkiG9w0BAQEFAASCAT4wggE6AgEAAkEA2EnE8ZOeVwZmwzPc"
+ "88DvoK1ckhOK7nVrsx9j6TmyKJ6m34UaAGpSlZbhvY72xyPNXl8QnUjm79SgT9bG"
+ "zeUc6QIDAQABAkBRFJZ32VbqWMP9OVwDJLiwC01AlYLnka0mIQZbT/2xq9dUc9GW"
+ "U3kiVw4lL8v/+sPjtTPCYYdzHHOyDen6znVhAiEA9qJT7BtQvRxCvGrAhr9MS022"
+ "tTdPbW829BoUtIeH64cCIQDggG5i48v7HPacPBIH1RaSVhXl8qHCpQD3qrIw3FMw"
+ "DwIga8PqH5Sf5sHedy2+CiK0V4MRfoU4c3zQ6kArI+bEgSkCIQCLA1vXBiE31B5s"
+ "bdHoYa1BXebfZVd+1Hd95IfEM5mbRwIgSkDuQwV55BBlvWph3U8wVIMIb4GStaH8"
+ "W535W8UBbEg=" "-----END PRIVATE KEY-----", -1, &error);
+ if (cert == NULL) {
+ g_printerr ("failed to parse PEM: %s\n", error->message);
+ return -1;
+ }
+ gst_rtsp_auth_set_tls_certificate (auth, cert);
+ g_object_unref (cert);
+#endif
+
+ /* make user token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ basic = gst_rtsp_auth_make_basic ("user", "password");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* configure in the server */
+ gst_rtsp_server_set_auth (server, auth);
+#endif
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+#ifdef WITH_AUTH
+ /* add permissions for the user media role */
+ permissions = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (permissions, "user",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_media_factory_set_permissions (factory, permissions);
+ gst_rtsp_permissions_unref (permissions);
+#ifdef WITH_TLS
+ gst_rtsp_media_factory_set_profiles (factory, GST_RTSP_PROFILE_SAVP);
+#endif
+#endif
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ g_signal_connect (server, "client-connected", (GCallback) client_connected,
+ NULL);
+
+ /* add a timeout for the session cleanup */
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving, this never stops */
+#ifdef WITH_TLS
+ g_print ("stream ready at rtsps://127.0.0.1:8554/test\n");
+#else
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+#endif
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+/* this timeout is periodically run to clean up the expired sessions from the
+ * pool. This needs to be run explicitly currently but might be done
+ * automatically as part of the mainloop. */
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=8 " ")");
+
+ gst_rtsp_media_factory_set_profiles (factory, GST_RTSP_PROFILE_AVPF);
+
+ /* store up to 0.4 seconds of retransmission data */
+ gst_rtsp_media_factory_set_retransmission_time (factory, 400 * GST_MSECOND);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ /* add a timeout for the session cleanup */
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving, this never stops */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+/* define this if you want the resource to only be available when using
+ * user/password as the password */
+#undef WITH_AUTH
+
+/* define this if you want the server to use TLS (it will also need WITH_AUTH
+ * to be defined) */
+#undef WITH_TLS
+
+/* this timeout is periodically run to clean up the expired sessions from the
+ * pool. This needs to be run explicitly currently but might be done
+ * automatically as part of the mainloop. */
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+#ifdef WITH_AUTH
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ gchar *basic;
+ GstRTSPPermissions *permissions;
+#endif
+#ifdef WITH_TLS
+ GTlsCertificate *cert;
+ GError *error = NULL;
+#endif
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+#ifdef WITH_AUTH
+ /* make a new authentication manager. it can be added to control access to all
+ * the factories on the server or on individual factories. */
+ auth = gst_rtsp_auth_new ();
+#ifdef WITH_TLS
+ cert = g_tls_certificate_new_from_pem ("-----BEGIN CERTIFICATE-----"
+ "MIICJjCCAY+gAwIBAgIBBzANBgkqhkiG9w0BAQUFADCBhjETMBEGCgmSJomT8ixk"
+ "ARkWA0NPTTEXMBUGCgmSJomT8ixkARkWB0VYQU1QTEUxHjAcBgNVBAsTFUNlcnRp"
+ "ZmljYXRlIEF1dGhvcml0eTEXMBUGA1UEAxMOY2EuZXhhbXBsZS5jb20xHTAbBgkq"
+ "hkiG9w0BCQEWDmNhQGV4YW1wbGUuY29tMB4XDTExMDExNzE5NDcxN1oXDTIxMDEx"
+ "NDE5NDcxN1owSzETMBEGCgmSJomT8ixkARkWA0NPTTEXMBUGCgmSJomT8ixkARkW"
+ "B0VYQU1QTEUxGzAZBgNVBAMTEnNlcnZlci5leGFtcGxlLmNvbTBcMA0GCSqGSIb3"
+ "DQEBAQUAA0sAMEgCQQDYScTxk55XBmbDM9zzwO+grVySE4rudWuzH2PpObIonqbf"
+ "hRoAalKVluG9jvbHI81eXxCdSObv1KBP1sbN5RzpAgMBAAGjIjAgMAkGA1UdEwQC"
+ "MAAwEwYDVR0lBAwwCgYIKwYBBQUHAwEwDQYJKoZIhvcNAQEFBQADgYEAYx6fMqT1"
+ "Gvo0jq88E8mc+bmp4LfXD4wJ7KxYeadQxt75HFRpj4FhFO3DOpVRFgzHlOEo3Fwk"
+ "PZOKjvkT0cbcoEq5whLH25dHoQxGoVQgFyAP5s+7Vp5AlHh8Y/vAoXeEVyy/RCIH"
+ "QkhUlAflfDMcrrYjsmwoOPSjhx6Mm/AopX4="
+ "-----END CERTIFICATE-----"
+ "-----BEGIN PRIVATE KEY-----"
+ "MIIBVAIBADANBgkqhkiG9w0BAQEFAASCAT4wggE6AgEAAkEA2EnE8ZOeVwZmwzPc"
+ "88DvoK1ckhOK7nVrsx9j6TmyKJ6m34UaAGpSlZbhvY72xyPNXl8QnUjm79SgT9bG"
+ "zeUc6QIDAQABAkBRFJZ32VbqWMP9OVwDJLiwC01AlYLnka0mIQZbT/2xq9dUc9GW"
+ "U3kiVw4lL8v/+sPjtTPCYYdzHHOyDen6znVhAiEA9qJT7BtQvRxCvGrAhr9MS022"
+ "tTdPbW829BoUtIeH64cCIQDggG5i48v7HPacPBIH1RaSVhXl8qHCpQD3qrIw3FMw"
+ "DwIga8PqH5Sf5sHedy2+CiK0V4MRfoU4c3zQ6kArI+bEgSkCIQCLA1vXBiE31B5s"
+ "bdHoYa1BXebfZVd+1Hd95IfEM5mbRwIgSkDuQwV55BBlvWph3U8wVIMIb4GStaH8"
+ "W535W8UBbEg=" "-----END PRIVATE KEY-----", -1, &error);
+ if (cert == NULL) {
+ g_printerr ("failed to parse PEM: %s\n", error->message);
+ return -1;
+ }
+ gst_rtsp_auth_set_tls_certificate (auth, cert);
+ g_object_unref (cert);
+#endif
+
+ /* make user token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ basic = gst_rtsp_auth_make_basic ("user", "password");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* configure in the server */
+ gst_rtsp_server_set_auth (server, auth);
+#endif
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+#ifdef WITH_AUTH
+ /* add permissions for the user media role */
+ permissions = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (permissions, "user",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_media_factory_set_permissions (factory, permissions);
+ gst_rtsp_permissions_unref (permissions);
+#ifdef WITH_TLS
+ gst_rtsp_media_factory_set_profiles (factory, GST_RTSP_PROFILE_SAVP);
+#endif
+#endif
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ /* add a timeout for the session cleanup */
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving, this never stops */
+#ifdef WITH_TLS
+ g_print ("stream ready at rtsps://127.0.0.1:8554/test\n");
+#else
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+#endif
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+<Project
+ xmlns:rdf="http://www.w3.org/1999/02/22-rdf-syntax-ns#"
+ xmlns:rdfs="http://www.w3.org/2000/01/rdf-schema#"
+ xmlns="http://usefulinc.com/ns/doap#"
+ xmlns:foaf="http://xmlns.com/foaf/0.1/"
+ xmlns:admin="http://webns.net/mvcb/">
+
+ <name>GStreamer RTSP Server</name>
+ <shortname>gst-rtsp-server</shortname>
+ <homepage rdf:resource="http://gstreamer.freedesktop.org/modules/gst-rtsp-server.html" />
+ <created>1999-10-31</created>
+ <shortdesc xml:lang="en">
+RTSP server library based on GStreamer
+</shortdesc>
+ <description xml:lang="en">
+RTSP server library based on GStreamer
+ </description>
+ <category></category>
+ <bug-database rdf:resource="https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/" />
+ <screenshots></screenshots>
+ <mailing-list rdf:resource="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" />
+ <programming-language>C</programming-language>
+ <license rdf:resource="http://usefulinc.com/doap/licenses/lgpl" />
+ <download-page rdf:resource="http://gstreamer.freedesktop.org/download/" />
+
+ <repository>
+ <GitRepository>
+ <location rdf:resource="https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server"/>
+ <browse rdf:resource="https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server"/>
+ </GitRepository>
+</repository>
+
+ <release>
+ <Version>
+ <revision>1.19.2</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2021-09-23</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.19.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.19.1</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2021-06-01</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.19.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.18.0</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2020-09-08</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.18.0.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.17.90</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2020-08-20</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.17.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.17.2</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2020-07-03</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.17.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.17.1</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2020-06-19</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.17.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.16.0</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2019-04-19</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.16.0.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.15.90</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2019-04-11</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.15.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.15.2</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2019-02-26</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.15.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.15.1</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2019-01-17</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.15.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.14.0</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2018-03-19</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.14.0.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.13.91</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2018-03-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.13.91.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.13.90</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2018-03-03</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.13.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.13.1</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2018-02-15</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.13.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.12.4</revision>
+ <branch>1.12</branch>
+ <name></name>
+ <created>2017-12-07</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.12.4.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.12.3</revision>
+ <branch>1.12</branch>
+ <name></name>
+ <created>2017-09-18</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.12.3.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.12.2</revision>
+ <branch>1.12</branch>
+ <name></name>
+ <created>2017-07-14</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.12.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.12.1</revision>
+ <branch>1.12</branch>
+ <name></name>
+ <created>2017-06-20</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.12.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.12.0</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2017-05-04</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.12.0.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.11.91</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2017-04-27</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.11.91.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.11.90</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2017-04-07</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.11.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.11.2</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2017-02-24</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.11.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.11.1</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2017-01-12</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.11.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.10.0</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2016-11-01</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.10.0.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.9.90</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2016-09-30</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.9.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.9.2</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2016-09-01</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.9.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.9.1</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2016-06-06</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.9.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.8.0</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2016-03-24</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.8.0.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.7.91</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2016-03-15</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.7.91.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.7.90</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2016-03-01</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.7.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.7.2</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2016-02-19</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.7.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.7.1</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2015-12-24</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.7.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.6.2</revision>
+ <branch>1.6</branch>
+ <name></name>
+ <created>2015-12-14</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.6.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.6.1</revision>
+ <branch>1.6</branch>
+ <name></name>
+ <created>2015-10-30</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.6.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.6.0</revision>
+ <branch>1.6</branch>
+ <name></name>
+ <created>2015-09-25</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.6.0.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.5.91</revision>
+ <branch>1.5</branch>
+ <name></name>
+ <created>2015-09-18</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.5.91.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.5.90</revision>
+ <branch>1.5</branch>
+ <name></name>
+ <created>2015-08-19</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.5.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.5.2</revision>
+ <branch>1.5</branch>
+ <name></name>
+ <created>2015-06-24</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.5.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.5.1</revision>
+ <branch>1.5</branch>
+ <name></name>
+ <created>2015-06-07</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.5.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.4.0</revision>
+ <branch>1.4</branch>
+ <name></name>
+ <created>2014-07-19</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.4.0.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.3.91</revision>
+ <branch>1.3</branch>
+ <name></name>
+ <created>2014-07-11</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.91.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.3.90</revision>
+ <branch>1.3</branch>
+ <name></name>
+ <created>2014-06-28</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.3.3</revision>
+ <branch>1.3</branch>
+ <name></name>
+ <created>2014-06-22</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.3.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.3.2</revision>
+ <branch>1.3</branch>
+ <name></name>
+ <created>2014-05-21</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.3.1</revision>
+ <branch>1.3</branch>
+ <name></name>
+ <created>2014-05-03</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.1.90</revision>
+ <branch>1.1</branch>
+ <name></name>
+ <created>2014-02-09</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.1.90.tar.xz" />
+ </Version>
+ </release>
+
+ <maintainer>
+ <foaf:Person>
+ <foaf:name>Wim Taymans</foaf:name>
+ <foaf:mbox_sha1sum>0d93fde052812d51a05fd86de9bdbf674423daa2</foaf:mbox_sha1sum>
+ </foaf:Person>
+ </maintainer>
+
+</Project>
--- /dev/null
+subdir('rtsp-server')
+
+if not get_option('rtspclientsink').disabled()
+ subdir('rtsp-sink')
+endif
--- /dev/null
+rtsp_server_sources = [
+ 'rtsp-address-pool.c',
+ 'rtsp-auth.c',
+ 'rtsp-client.c',
+ 'rtsp-context.c',
+ 'rtsp-latency-bin.c',
+ 'rtsp-media.c',
+ 'rtsp-media-factory.c',
+ 'rtsp-media-factory-uri.c',
+ 'rtsp-mount-points.c',
+ 'rtsp-params.c',
+ 'rtsp-permissions.c',
+ 'rtsp-sdp.c',
+ 'rtsp-server.c',
+ 'rtsp-session.c',
+ 'rtsp-session-media.c',
+ 'rtsp-session-pool.c',
+ 'rtsp-stream.c',
+ 'rtsp-stream-transport.c',
+ 'rtsp-thread-pool.c',
+ 'rtsp-token.c',
+ 'rtsp-onvif-server.c',
+ 'rtsp-onvif-client.c',
+ 'rtsp-onvif-media-factory.c',
+ 'rtsp-onvif-media.c',
+]
+
+rtsp_server_headers = [
+ 'rtsp-auth.h',
+ 'rtsp-address-pool.h',
+ 'rtsp-context.h',
+ 'rtsp-params.h',
+ 'rtsp-sdp.h',
+ 'rtsp-thread-pool.h',
+ 'rtsp-media.h',
+ 'rtsp-media-factory.h',
+ 'rtsp-media-factory-uri.h',
+ 'rtsp-mount-points.h',
+ 'rtsp-permissions.h',
+ 'rtsp-stream.h',
+ 'rtsp-stream-transport.h',
+ 'rtsp-session.h',
+ 'rtsp-session-media.h',
+ 'rtsp-session-pool.h',
+ 'rtsp-token.h',
+ 'rtsp-client.h',
+ 'rtsp-server.h',
+ 'rtsp-server-object.h',
+ 'rtsp-server-prelude.h',
+ 'rtsp-onvif-server.h',
+ 'rtsp-onvif-client.h',
+ 'rtsp-onvif-media-factory.h',
+ 'rtsp-onvif-media.h',
+]
+
+install_headers(rtsp_server_headers, subdir : 'gstreamer-1.0/gst/rtsp-server')
+
+gst_rtsp_server_deps = [gstrtsp_dep, gstrtp_dep, gstsdp_dep, gstnet_dep, gstapp_dep]
+gst_rtsp_server = library('gstrtspserver-@0@'.format(api_version),
+ rtsp_server_sources,
+ include_directories : rtspserver_incs,
+ c_args: rtspserver_args + ['-DBUILDING_GST_RTSP_SERVER'],
+ version : libversion,
+ soversion : soversion,
+ darwin_versions : osxversion,
+ install : true,
+ dependencies : gst_rtsp_server_deps)
+
+pkgconfig.generate(gst_rtsp_server,
+ libraries : [gst_dep],
+ subdirs : pkgconfig_subdirs,
+ name : 'gstreamer-rtsp-server-1.0',
+ description : 'GStreamer based RTSP server',
+)
+
+rtsp_server_gen_sources = []
+if build_gir
+ gst_gir_extra_args = gir_init_section + ['--c-include=gst/rtsp-server/rtsp-server.h']
+ rtsp_server_gir = gnome.generate_gir(gst_rtsp_server,
+ sources : rtsp_server_headers + rtsp_server_sources,
+ namespace : 'GstRtspServer',
+ nsversion : api_version,
+ identifier_prefix : 'Gst',
+ symbol_prefix : 'gst',
+ export_packages : 'gstreamer-rtsp-server-' + api_version,
+ install : true,
+ extra_args : gst_gir_extra_args,
+ includes : ['Gst-1.0', 'GstRtsp-1.0', 'GstNet-1.0'],
+ dependencies : gst_rtsp_server_deps,
+ )
+ rtsp_server_gen_sources += [rtsp_server_gir]
+endif
+
+gst_rtsp_server_dep = declare_dependency(link_with : gst_rtsp_server,
+ include_directories : rtspserver_incs,
+ sources : rtsp_server_gen_sources,
+ dependencies : [gstrtsp_dep, gstrtp_dep, gstsdp_dep, gstnet_dep, gstapp_dep])
+
+meson.override_dependency('gstreamer-rtsp-server-1.0', gst_rtsp_server_dep)
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-address-pool
+ * @short_description: A pool of network addresses
+ * @see_also: #GstRTSPStream, #GstRTSPStreamTransport
+ *
+ * The #GstRTSPAddressPool is an object that maintains a collection of network
+ * addresses. It is used to allocate server ports and server multicast addresses
+ * but also to reserve client provided destination addresses.
+ *
+ * A range of addresses can be added with gst_rtsp_address_pool_add_range().
+ * Both multicast and unicast addresses can be added.
+ *
+ * With gst_rtsp_address_pool_acquire_address() an unused address and port range
+ * can be acquired from the pool. With gst_rtsp_address_pool_reserve_address() a
+ * specific address can be retrieved. Both methods return a boxed
+ * #GstRTSPAddress that should be freed with gst_rtsp_address_free() after
+ * usage, which brings the address back into the pool.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include <gio/gio.h>
+
+#include "rtsp-address-pool.h"
+
+/**
+ * gst_rtsp_address_copy:
+ * @addr: a #GstRTSPAddress
+ *
+ * Make a copy of @addr.
+ *
+ * Returns: a copy of @addr.
+ */
+GstRTSPAddress *
+gst_rtsp_address_copy (GstRTSPAddress * addr)
+{
+ GstRTSPAddress *copy;
+
+ g_return_val_if_fail (addr != NULL, NULL);
+
+ copy = g_slice_dup (GstRTSPAddress, addr);
+ /* only release to the pool when the original is freed. It's a bit
+ * weird but this will do for now as it avoid us to use refcounting. */
+ copy->pool = NULL;
+ copy->address = g_strdup (copy->address);
+
+ return copy;
+}
+
+static void gst_rtsp_address_pool_release_address (GstRTSPAddressPool * pool,
+ GstRTSPAddress * addr);
+
+/**
+ * gst_rtsp_address_free:
+ * @addr: a #GstRTSPAddress
+ *
+ * Free @addr and releasing it back into the pool when owned by a
+ * pool.
+ */
+void
+gst_rtsp_address_free (GstRTSPAddress * addr)
+{
+ g_return_if_fail (addr != NULL);
+
+ if (addr->pool) {
+ /* unrefs the pool and sets it to NULL */
+ gst_rtsp_address_pool_release_address (addr->pool, addr);
+ }
+ g_free (addr->address);
+ g_slice_free (GstRTSPAddress, addr);
+}
+
+G_DEFINE_BOXED_TYPE (GstRTSPAddress, gst_rtsp_address,
+ (GBoxedCopyFunc) gst_rtsp_address_copy,
+ (GBoxedFreeFunc) gst_rtsp_address_free);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_address_pool_debug);
+#define GST_CAT_DEFAULT rtsp_address_pool_debug
+
+#define GST_RTSP_ADDRESS_POOL_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPoolPrivate))
+
+struct _GstRTSPAddressPoolPrivate
+{
+ GMutex lock; /* protects everything in this struct */
+ GList *addresses;
+ GList *allocated;
+
+ gboolean has_unicast_addresses;
+};
+
+#define ADDR_IS_IPV4(a) ((a)->size == 4)
+#define ADDR_IS_IPV6(a) ((a)->size == 16)
+#define ADDR_IS_EVEN_PORT(a) (((a)->port & 1) == 0)
+
+typedef struct
+{
+ guint8 bytes[16];
+ gsize size;
+ guint16 port;
+} Addr;
+
+typedef struct
+{
+ Addr min;
+ Addr max;
+ guint8 ttl;
+} AddrRange;
+
+#define RANGE_IS_SINGLE(r) (memcmp ((r)->min.bytes, (r)->max.bytes, (r)->min.size) == 0)
+
+#define gst_rtsp_address_pool_parent_class parent_class
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPAddressPool, gst_rtsp_address_pool,
+ G_TYPE_OBJECT);
+
+static void gst_rtsp_address_pool_finalize (GObject * obj);
+
+static void
+gst_rtsp_address_pool_class_init (GstRTSPAddressPoolClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->finalize = gst_rtsp_address_pool_finalize;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_address_pool_debug, "rtspaddresspool", 0,
+ "GstRTSPAddressPool");
+}
+
+static void
+gst_rtsp_address_pool_init (GstRTSPAddressPool * pool)
+{
+ pool->priv = gst_rtsp_address_pool_get_instance_private (pool);
+
+ g_mutex_init (&pool->priv->lock);
+}
+
+static void
+free_range (AddrRange * range)
+{
+ g_slice_free (AddrRange, range);
+}
+
+static void
+gst_rtsp_address_pool_finalize (GObject * obj)
+{
+ GstRTSPAddressPool *pool;
+
+ pool = GST_RTSP_ADDRESS_POOL (obj);
+
+ g_list_free_full (pool->priv->addresses, (GDestroyNotify) free_range);
+ g_list_free_full (pool->priv->allocated, (GDestroyNotify) free_range);
+ g_mutex_clear (&pool->priv->lock);
+
+ G_OBJECT_CLASS (parent_class)->finalize (obj);
+}
+
+/**
+ * gst_rtsp_address_pool_new:
+ *
+ * Make a new #GstRTSPAddressPool.
+ *
+ * Returns: (transfer full): a new #GstRTSPAddressPool
+ */
+GstRTSPAddressPool *
+gst_rtsp_address_pool_new (void)
+{
+ GstRTSPAddressPool *pool;
+
+ pool = g_object_new (GST_TYPE_RTSP_ADDRESS_POOL, NULL);
+
+ return pool;
+}
+
+/**
+ * gst_rtsp_address_pool_clear:
+ * @pool: a #GstRTSPAddressPool
+ *
+ * Clear all addresses in @pool. There should be no outstanding
+ * allocations.
+ */
+void
+gst_rtsp_address_pool_clear (GstRTSPAddressPool * pool)
+{
+ GstRTSPAddressPoolPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool));
+ g_return_if_fail (pool->priv->allocated == NULL);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_list_free_full (priv->addresses, (GDestroyNotify) free_range);
+ priv->addresses = NULL;
+ g_mutex_unlock (&priv->lock);
+}
+
+static gboolean
+fill_address (const gchar * address, guint16 port, Addr * addr,
+ gboolean is_multicast)
+{
+ GInetAddress *inet;
+
+ inet = g_inet_address_new_from_string (address);
+ if (inet == NULL)
+ return FALSE;
+
+ if (is_multicast != g_inet_address_get_is_multicast (inet)) {
+ g_object_unref (inet);
+ return FALSE;
+ }
+
+ addr->size = g_inet_address_get_native_size (inet);
+ memcpy (addr->bytes, g_inet_address_to_bytes (inet), addr->size);
+ g_object_unref (inet);
+ addr->port = port;
+
+ return TRUE;
+}
+
+static gchar *
+get_address_string (Addr * addr)
+{
+ gchar *res;
+ GInetAddress *inet;
+
+ inet = g_inet_address_new_from_bytes (addr->bytes,
+ addr->size == 4 ? G_SOCKET_FAMILY_IPV4 : G_SOCKET_FAMILY_IPV6);
+ res = g_inet_address_to_string (inet);
+ g_object_unref (inet);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_address_pool_add_range:
+ * @pool: a #GstRTSPAddressPool
+ * @min_address: a minimum address to add
+ * @max_address: a maximum address to add
+ * @min_port: the minimum port
+ * @max_port: the maximum port
+ * @ttl: a TTL or 0 for unicast addresses
+ *
+ * Adds the addresses from @min_addess to @max_address (inclusive)
+ * to @pool. The valid port range for the addresses will be from @min_port to
+ * @max_port inclusive.
+ *
+ * When @ttl is 0, @min_address and @max_address should be unicast addresses.
+ * @min_address and @max_address can be set to
+ * #GST_RTSP_ADDRESS_POOL_ANY_IPV4 or #GST_RTSP_ADDRESS_POOL_ANY_IPV6 to bind
+ * to all available IPv4 or IPv6 addresses.
+ *
+ * When @ttl > 0, @min_address and @max_address should be multicast addresses.
+ *
+ * Returns: %TRUE if the addresses could be added.
+ */
+gboolean
+gst_rtsp_address_pool_add_range (GstRTSPAddressPool * pool,
+ const gchar * min_address, const gchar * max_address,
+ guint16 min_port, guint16 max_port, guint8 ttl)
+{
+ AddrRange *range;
+ GstRTSPAddressPoolPrivate *priv;
+ gboolean is_multicast;
+
+ g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool), FALSE);
+ g_return_val_if_fail (min_port <= max_port, FALSE);
+
+ priv = pool->priv;
+
+ is_multicast = ttl != 0;
+
+ range = g_slice_new0 (AddrRange);
+
+ if (!fill_address (min_address, min_port, &range->min, is_multicast))
+ goto invalid;
+ if (!fill_address (max_address, max_port, &range->max, is_multicast))
+ goto invalid;
+
+ if (range->min.size != range->max.size)
+ goto invalid;
+ if (memcmp (range->min.bytes, range->max.bytes, range->min.size) > 0)
+ goto invalid;
+
+ range->ttl = ttl;
+
+ GST_DEBUG_OBJECT (pool, "adding %s-%s:%u-%u ttl %u", min_address, max_address,
+ min_port, max_port, ttl);
+
+ g_mutex_lock (&priv->lock);
+ priv->addresses = g_list_prepend (priv->addresses, range);
+
+ if (!is_multicast)
+ priv->has_unicast_addresses = TRUE;
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+ /* ERRORS */
+invalid:
+ {
+ GST_ERROR_OBJECT (pool, "invalid address range %s-%s", min_address,
+ max_address);
+ g_slice_free (AddrRange, range);
+ return FALSE;
+ }
+}
+
+static void
+inc_address (Addr * addr, guint count)
+{
+ gint i;
+ guint carry;
+
+ carry = count;
+ for (i = addr->size - 1; i >= 0 && carry > 0; i--) {
+ carry += addr->bytes[i];
+ addr->bytes[i] = carry & 0xff;
+ carry >>= 8;
+ }
+}
+
+/* tells us the number of addresses between min_addr and max_addr */
+static guint
+diff_address (Addr * max_addr, Addr * min_addr)
+{
+ gint i;
+ guint result = 0;
+
+ g_return_val_if_fail (min_addr->size == max_addr->size, 0);
+
+ for (i = 0; i < min_addr->size; i++) {
+ g_return_val_if_fail (result < (1 << 24), result);
+
+ result <<= 8;
+ result += max_addr->bytes[i] - min_addr->bytes[i];
+ }
+
+ return result;
+}
+
+static AddrRange *
+split_range (GstRTSPAddressPool * pool, AddrRange * range, guint skip_addr,
+ guint skip_port, gint n_ports)
+{
+ GstRTSPAddressPoolPrivate *priv = pool->priv;
+ AddrRange *temp;
+
+ if (skip_addr) {
+ temp = g_slice_dup (AddrRange, range);
+ memcpy (temp->max.bytes, temp->min.bytes, temp->min.size);
+ inc_address (&temp->max, skip_addr - 1);
+ priv->addresses = g_list_prepend (priv->addresses, temp);
+
+ inc_address (&range->min, skip_addr);
+ }
+
+ if (!RANGE_IS_SINGLE (range)) {
+ /* min and max are not the same, we have more than one address. */
+ temp = g_slice_dup (AddrRange, range);
+ /* increment the range min address */
+ inc_address (&temp->min, 1);
+ /* and store back in pool */
+ priv->addresses = g_list_prepend (priv->addresses, temp);
+
+ /* adjust range with only the first address */
+ memcpy (range->max.bytes, range->min.bytes, range->min.size);
+ }
+
+ /* range now contains only one single address */
+ if (skip_port > 0) {
+ /* make a range with the skipped ports */
+ temp = g_slice_dup (AddrRange, range);
+ temp->max.port = temp->min.port + skip_port - 1;
+ /* and store back in pool */
+ priv->addresses = g_list_prepend (priv->addresses, temp);
+
+ /* increment range port */
+ range->min.port += skip_port;
+ }
+ /* range now contains single address with desired port number */
+ if (range->max.port - range->min.port + 1 > n_ports) {
+ /* make a range with the remaining ports */
+ temp = g_slice_dup (AddrRange, range);
+ temp->min.port += n_ports;
+ /* and store back in pool */
+ priv->addresses = g_list_prepend (priv->addresses, temp);
+
+ /* and truncate port */
+ range->max.port = range->min.port + n_ports - 1;
+ }
+ return range;
+}
+
+/**
+ * gst_rtsp_address_pool_acquire_address:
+ * @pool: a #GstRTSPAddressPool
+ * @flags: flags
+ * @n_ports: the amount of ports
+ *
+ * Take an address and ports from @pool. @flags can be used to control the
+ * allocation. @n_ports consecutive ports will be allocated of which the first
+ * one can be found in @port.
+ *
+ * Returns: (nullable): a #GstRTSPAddress that should be freed with
+ * gst_rtsp_address_free after use or %NULL when no address could be
+ * acquired.
+ */
+GstRTSPAddress *
+gst_rtsp_address_pool_acquire_address (GstRTSPAddressPool * pool,
+ GstRTSPAddressFlags flags, gint n_ports)
+{
+ GstRTSPAddressPoolPrivate *priv;
+ GList *walk, *next;
+ AddrRange *result;
+ GstRTSPAddress *addr;
+
+ g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool), NULL);
+ g_return_val_if_fail (n_ports > 0, NULL);
+
+ priv = pool->priv;
+ result = NULL;
+ addr = NULL;
+
+ g_mutex_lock (&priv->lock);
+ /* go over available ranges */
+ for (walk = priv->addresses; walk; walk = next) {
+ AddrRange *range;
+ gint ports, skip;
+
+ range = walk->data;
+ next = walk->next;
+
+ /* check address type when given */
+ if (flags & GST_RTSP_ADDRESS_FLAG_IPV4 && !ADDR_IS_IPV4 (&range->min))
+ continue;
+ if (flags & GST_RTSP_ADDRESS_FLAG_IPV6 && !ADDR_IS_IPV6 (&range->min))
+ continue;
+ if (flags & GST_RTSP_ADDRESS_FLAG_MULTICAST && range->ttl == 0)
+ continue;
+ if (flags & GST_RTSP_ADDRESS_FLAG_UNICAST && range->ttl != 0)
+ continue;
+
+ /* check for enough ports */
+ ports = range->max.port - range->min.port + 1;
+ if (flags & GST_RTSP_ADDRESS_FLAG_EVEN_PORT
+ && !ADDR_IS_EVEN_PORT (&range->min))
+ skip = 1;
+ else
+ skip = 0;
+ if (ports - skip < n_ports)
+ continue;
+
+ /* we found a range, remove from the list */
+ priv->addresses = g_list_delete_link (priv->addresses, walk);
+ /* now split and exit our loop */
+ result = split_range (pool, range, 0, skip, n_ports);
+ priv->allocated = g_list_prepend (priv->allocated, result);
+ break;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (result) {
+ addr = g_slice_new0 (GstRTSPAddress);
+ addr->pool = g_object_ref (pool);
+ addr->address = get_address_string (&result->min);
+ addr->n_ports = n_ports;
+ addr->port = result->min.port;
+ addr->ttl = result->ttl;
+ addr->priv = result;
+
+ GST_DEBUG_OBJECT (pool, "got address %s:%u ttl %u", addr->address,
+ addr->port, addr->ttl);
+ }
+
+ return addr;
+}
+
+/**
+ * gst_rtsp_address_pool_release_address:
+ * @pool: a #GstRTSPAddressPool
+ * @id: an address id
+ *
+ * Release a previously acquired address (with
+ * gst_rtsp_address_pool_acquire_address()) back into @pool.
+ */
+static void
+gst_rtsp_address_pool_release_address (GstRTSPAddressPool * pool,
+ GstRTSPAddress * addr)
+{
+ GstRTSPAddressPoolPrivate *priv;
+ GList *find;
+ AddrRange *range;
+
+ g_return_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool));
+ g_return_if_fail (addr != NULL);
+ g_return_if_fail (addr->pool == pool);
+
+ priv = pool->priv;
+ range = addr->priv;
+
+ /* we don't want to free twice */
+ addr->priv = NULL;
+ addr->pool = NULL;
+
+ g_mutex_lock (&priv->lock);
+ find = g_list_find (priv->allocated, range);
+ if (find == NULL)
+ goto not_found;
+
+ priv->allocated = g_list_delete_link (priv->allocated, find);
+
+ /* FIXME, merge and do something clever */
+ priv->addresses = g_list_prepend (priv->addresses, range);
+ g_mutex_unlock (&priv->lock);
+
+ g_object_unref (pool);
+
+ return;
+
+ /* ERRORS */
+not_found:
+ {
+ g_warning ("Released unknown address %p", addr);
+ g_mutex_unlock (&priv->lock);
+ return;
+ }
+}
+
+static void
+dump_range (AddrRange * range, GstRTSPAddressPool * pool)
+{
+ gchar *addr1, *addr2;
+
+ addr1 = get_address_string (&range->min);
+ addr2 = get_address_string (&range->max);
+ g_print (" address %s-%s, port %u-%u, ttl %u\n", addr1, addr2,
+ range->min.port, range->max.port, range->ttl);
+ g_free (addr1);
+ g_free (addr2);
+}
+
+/**
+ * gst_rtsp_address_pool_dump:
+ * @pool: a #GstRTSPAddressPool
+ *
+ * Dump the free and allocated addresses to stdout.
+ */
+void
+gst_rtsp_address_pool_dump (GstRTSPAddressPool * pool)
+{
+ GstRTSPAddressPoolPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool));
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_print ("free:\n");
+ g_list_foreach (priv->addresses, (GFunc) dump_range, pool);
+ g_print ("allocated:\n");
+ g_list_foreach (priv->allocated, (GFunc) dump_range, pool);
+ g_mutex_unlock (&priv->lock);
+}
+
+static GList *
+find_address_in_ranges (GList * addresses, Addr * addr, guint port,
+ guint n_ports, guint ttl)
+{
+ GList *walk, *next;
+
+ /* go over available ranges */
+ for (walk = addresses; walk; walk = next) {
+ AddrRange *range;
+
+ range = walk->data;
+ next = walk->next;
+
+ /* Not the right type of address */
+ if (range->min.size != addr->size)
+ continue;
+
+ /* Check that the address is in the interval */
+ if (memcmp (range->min.bytes, addr->bytes, addr->size) > 0 ||
+ memcmp (range->max.bytes, addr->bytes, addr->size) < 0)
+ continue;
+
+ /* Make sure the requested ports are inside the range */
+ if (port < range->min.port || port + n_ports - 1 > range->max.port)
+ continue;
+
+ if (ttl != range->ttl)
+ continue;
+
+ break;
+ }
+
+ return walk;
+}
+
+/**
+ * gst_rtsp_address_pool_reserve_address:
+ * @pool: a #GstRTSPAddressPool
+ * @ip_address: The IP address to reserve
+ * @port: The first port to reserve
+ * @n_ports: The number of ports
+ * @ttl: The requested ttl
+ * @address: (out): storage for a #GstRTSPAddress
+ *
+ * Take a specific address and ports from @pool. @n_ports consecutive
+ * ports will be allocated of which the first one can be found in
+ * @port.
+ *
+ * If @ttl is 0, @address should be a unicast address. If @ttl > 0, @address
+ * should be a valid multicast address.
+ *
+ * Returns: #GST_RTSP_ADDRESS_POOL_OK if an address was reserved. The address
+ * is returned in @address and should be freed with gst_rtsp_address_free
+ * after use.
+ */
+GstRTSPAddressPoolResult
+gst_rtsp_address_pool_reserve_address (GstRTSPAddressPool * pool,
+ const gchar * ip_address, guint port, guint n_ports, guint ttl,
+ GstRTSPAddress ** address)
+{
+ GstRTSPAddressPoolPrivate *priv;
+ Addr input_addr;
+ GList *list;
+ AddrRange *addr_range;
+ GstRTSPAddress *addr;
+ gboolean is_multicast;
+ GstRTSPAddressPoolResult result;
+
+ g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool),
+ GST_RTSP_ADDRESS_POOL_EINVAL);
+ g_return_val_if_fail (ip_address != NULL, GST_RTSP_ADDRESS_POOL_EINVAL);
+ g_return_val_if_fail (port > 0, GST_RTSP_ADDRESS_POOL_EINVAL);
+ g_return_val_if_fail (n_ports > 0, GST_RTSP_ADDRESS_POOL_EINVAL);
+ g_return_val_if_fail (address != NULL, GST_RTSP_ADDRESS_POOL_EINVAL);
+
+ priv = pool->priv;
+ addr_range = NULL;
+ addr = NULL;
+ is_multicast = ttl != 0;
+
+ if (!fill_address (ip_address, port, &input_addr, is_multicast))
+ goto invalid;
+
+ g_mutex_lock (&priv->lock);
+ list = find_address_in_ranges (priv->addresses, &input_addr, port, n_ports,
+ ttl);
+ if (list != NULL) {
+ AddrRange *range = list->data;
+ guint skip_port, skip_addr;
+
+ skip_addr = diff_address (&input_addr, &range->min);
+ skip_port = port - range->min.port;
+
+ GST_DEBUG_OBJECT (pool, "diff 0x%08x/%u", skip_addr, skip_port);
+
+ /* we found a range, remove from the list */
+ priv->addresses = g_list_delete_link (priv->addresses, list);
+ /* now split and exit our loop */
+ addr_range = split_range (pool, range, skip_addr, skip_port, n_ports);
+ priv->allocated = g_list_prepend (priv->allocated, addr_range);
+ }
+
+ if (addr_range) {
+ addr = g_slice_new0 (GstRTSPAddress);
+ addr->pool = g_object_ref (pool);
+ addr->address = get_address_string (&addr_range->min);
+ addr->n_ports = n_ports;
+ addr->port = addr_range->min.port;
+ addr->ttl = addr_range->ttl;
+ addr->priv = addr_range;
+
+ result = GST_RTSP_ADDRESS_POOL_OK;
+ GST_DEBUG_OBJECT (pool, "reserved address %s:%u ttl %u", addr->address,
+ addr->port, addr->ttl);
+ } else {
+ /* We failed to reserve the address. Check if it was because the address
+ * was already in use or if it wasn't in the pool to begin with */
+ list = find_address_in_ranges (priv->allocated, &input_addr, port, n_ports,
+ ttl);
+ if (list != NULL) {
+ result = GST_RTSP_ADDRESS_POOL_ERESERVED;
+ } else {
+ result = GST_RTSP_ADDRESS_POOL_ERANGE;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ *address = addr;
+ return result;
+
+ /* ERRORS */
+invalid:
+ {
+ GST_ERROR_OBJECT (pool, "invalid address %s:%u/%u/%u", ip_address,
+ port, n_ports, ttl);
+ *address = NULL;
+ return GST_RTSP_ADDRESS_POOL_EINVAL;
+ }
+}
+
+/**
+ * gst_rtsp_address_pool_has_unicast_addresses:
+ * @pool: a #GstRTSPAddressPool
+ *
+ * Used to know if the pool includes any unicast addresses.
+ *
+ * Returns: %TRUE if the pool includes any unicast addresses, %FALSE otherwise
+ */
+
+gboolean
+gst_rtsp_address_pool_has_unicast_addresses (GstRTSPAddressPool * pool)
+{
+ GstRTSPAddressPoolPrivate *priv;
+ gboolean has_unicast_addresses;
+
+ g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool), FALSE);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ has_unicast_addresses = priv->has_unicast_addresses;
+ g_mutex_unlock (&priv->lock);
+
+ return has_unicast_addresses;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_ADDRESS_POOL_H__
+#define __GST_RTSP_ADDRESS_POOL_H__
+
+#include <gst/gst.h>
+#include "rtsp-server-prelude.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_ADDRESS_POOL (gst_rtsp_address_pool_get_type ())
+#define GST_IS_RTSP_ADDRESS_POOL(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_ADDRESS_POOL))
+#define GST_IS_RTSP_ADDRESS_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_ADDRESS_POOL))
+#define GST_RTSP_ADDRESS_POOL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPoolClass))
+#define GST_RTSP_ADDRESS_POOL(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPool))
+#define GST_RTSP_ADDRESS_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPoolClass))
+#define GST_RTSP_ADDRESS_POOL_CAST(obj) ((GstRTSPAddressPool*)(obj))
+#define GST_RTSP_ADDRESS_POOL_CLASS_CAST(klass) ((GstRTSPAddressPoolClass*)(klass))
+
+/**
+ * GstRTSPAddressPoolResult:
+ * @GST_RTSP_ADDRESS_POOL_OK: no error
+ * @GST_RTSP_ADDRESS_POOL_EINVAL:invalid arguments were provided to a function
+ * @GST_RTSP_ADDRESS_POOL_ERESERVED: the addres has already been reserved
+ * @GST_RTSP_ADDRESS_POOL_ERANGE: the address is not in the pool
+ * @GST_RTSP_ADDRESS_POOL_ELAST: last error
+ *
+ * Result codes from RTSP address pool functions.
+ */
+typedef enum {
+ GST_RTSP_ADDRESS_POOL_OK = 0,
+ /* errors */
+ GST_RTSP_ADDRESS_POOL_EINVAL = -1,
+ GST_RTSP_ADDRESS_POOL_ERESERVED = -2,
+ GST_RTSP_ADDRESS_POOL_ERANGE = -3,
+
+ GST_RTSP_ADDRESS_POOL_ELAST = -4,
+} GstRTSPAddressPoolResult;
+
+
+typedef struct _GstRTSPAddress GstRTSPAddress;
+
+typedef struct _GstRTSPAddressPool GstRTSPAddressPool;
+typedef struct _GstRTSPAddressPoolClass GstRTSPAddressPoolClass;
+typedef struct _GstRTSPAddressPoolPrivate GstRTSPAddressPoolPrivate;
+
+/**
+ * GstRTSPAddress:
+ * @pool: the #GstRTSPAddressPool owner of this address
+ * @address: the address
+ * @port: the port number
+ * @n_ports: number of ports
+ * @ttl: TTL or 0 for unicast addresses
+ *
+ * An address
+ */
+struct _GstRTSPAddress {
+ GstRTSPAddressPool *pool;
+
+ gchar *address;
+ guint16 port;
+ gint n_ports;
+ guint8 ttl;
+
+ /*<private >*/
+ gpointer priv;
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_address_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPAddress * gst_rtsp_address_copy (GstRTSPAddress *addr);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_address_free (GstRTSPAddress *addr);
+
+/**
+ * GstRTSPAddressFlags:
+ * @GST_RTSP_ADDRESS_FLAG_NONE: no flags
+ * @GST_RTSP_ADDRESS_FLAG_IPV4: an IPv4 address
+ * @GST_RTSP_ADDRESS_FLAG_IPV6: and IPv6 address
+ * @GST_RTSP_ADDRESS_FLAG_EVEN_PORT: address with an even port
+ * @GST_RTSP_ADDRESS_FLAG_MULTICAST: a multicast address
+ * @GST_RTSP_ADDRESS_FLAG_UNICAST: a unicast address
+ *
+ * Flags used to control allocation of addresses
+ */
+typedef enum {
+ GST_RTSP_ADDRESS_FLAG_NONE = 0,
+ GST_RTSP_ADDRESS_FLAG_IPV4 = (1 << 0),
+ GST_RTSP_ADDRESS_FLAG_IPV6 = (1 << 1),
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT = (1 << 2),
+ GST_RTSP_ADDRESS_FLAG_MULTICAST = (1 << 3),
+ GST_RTSP_ADDRESS_FLAG_UNICAST = (1 << 4),
+} GstRTSPAddressFlags;
+
+/**
+ * GST_RTSP_ADDRESS_POOL_ANY_IPV4:
+ *
+ * Used with gst_rtsp_address_pool_add_range() to bind to all
+ * IPv4 addresses
+ */
+#define GST_RTSP_ADDRESS_POOL_ANY_IPV4 "0.0.0.0"
+
+/**
+ * GST_RTSP_ADDRESS_POOL_ANY_IPV6:
+ *
+ * Used with gst_rtsp_address_pool_add_range() to bind to all
+ * IPv6 addresses
+ */
+#define GST_RTSP_ADDRESS_POOL_ANY_IPV6 "::"
+
+/**
+ * GstRTSPAddressPool:
+ * @parent: the parent GObject
+ *
+ * An address pool, all member are private
+ */
+struct _GstRTSPAddressPool {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPAddressPoolPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPAddressPoolClass:
+ *
+ * Opaque Address pool class.
+ */
+struct _GstRTSPAddressPoolClass {
+ GObjectClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_address_pool_get_type (void);
+
+/* create a new address pool */
+
+GST_RTSP_SERVER_API
+GstRTSPAddressPool * gst_rtsp_address_pool_new (void);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_address_pool_clear (GstRTSPAddressPool * pool);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_address_pool_dump (GstRTSPAddressPool * pool);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_address_pool_add_range (GstRTSPAddressPool * pool,
+ const gchar *min_address,
+ const gchar *max_address,
+ guint16 min_port,
+ guint16 max_port,
+ guint8 ttl);
+
+GST_RTSP_SERVER_API
+GstRTSPAddress * gst_rtsp_address_pool_acquire_address (GstRTSPAddressPool * pool,
+ GstRTSPAddressFlags flags,
+ gint n_ports);
+
+GST_RTSP_SERVER_API
+GstRTSPAddressPoolResult gst_rtsp_address_pool_reserve_address (GstRTSPAddressPool * pool,
+ const gchar *ip_address,
+ guint port,
+ guint n_ports,
+ guint ttl,
+ GstRTSPAddress ** address);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_address_pool_has_unicast_addresses (GstRTSPAddressPool * pool);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPAddress, gst_rtsp_address_free)
+#endif
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPAddressPool, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_ADDRESS_POOL_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-auth
+ * @short_description: Authentication and authorization
+ * @see_also: #GstRTSPPermissions, #GstRTSPToken
+ *
+ * The #GstRTSPAuth object is responsible for checking if the current user is
+ * allowed to perform requested actions. The default implementation has some
+ * reasonable checks but subclasses can implement custom security policies.
+ *
+ * A new auth object is made with gst_rtsp_auth_new(). It is usually configured
+ * on the #GstRTSPServer object.
+ *
+ * The RTSP server will call gst_rtsp_auth_check() with a string describing the
+ * check to perform. The possible checks are prefixed with
+ * GST_RTSP_AUTH_CHECK_*. Depending on the check, the default implementation
+ * will use the current #GstRTSPToken, #GstRTSPContext and
+ * #GstRTSPPermissions on the object to check if an operation is allowed.
+ *
+ * The default #GstRTSPAuth object has support for basic authentication. With
+ * gst_rtsp_auth_add_basic() you can add a basic authentication string together
+ * with the #GstRTSPToken that will become active when successfully
+ * authenticated.
+ *
+ * When a TLS certificate has been set with gst_rtsp_auth_set_tls_certificate(),
+ * the default auth object will require the client to connect with a TLS
+ * connection.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-auth.h"
+
+struct _GstRTSPAuthPrivate
+{
+ GMutex lock;
+
+ /* the TLS certificate */
+ GTlsCertificate *certificate;
+ GTlsDatabase *database;
+ GTlsAuthenticationMode mode;
+ GHashTable *basic; /* protected by lock */
+ GHashTable *digest, *nonces; /* protected by lock */
+ guint64 last_nonce_check;
+ GstRTSPToken *default_token;
+ GstRTSPMethod methods;
+ GstRTSPAuthMethod auth_methods;
+ gchar *realm;
+};
+
+typedef struct
+{
+ GstRTSPToken *token;
+ gchar *pass;
+ gchar *md5_pass;
+} GstRTSPDigestEntry;
+
+typedef struct
+{
+ gchar *nonce;
+ gchar *ip;
+ guint64 timestamp;
+ gpointer client;
+} GstRTSPDigestNonce;
+
+static void
+gst_rtsp_digest_entry_free (GstRTSPDigestEntry * entry)
+{
+ gst_rtsp_token_unref (entry->token);
+ g_free (entry->pass);
+ g_free (entry->md5_pass);
+ g_free (entry);
+}
+
+static void
+gst_rtsp_digest_nonce_free (GstRTSPDigestNonce * nonce)
+{
+ g_free (nonce->nonce);
+ g_free (nonce->ip);
+ g_free (nonce);
+}
+
+enum
+{
+ PROP_0,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_ACCEPT_CERTIFICATE,
+ SIGNAL_LAST
+};
+
+static guint signals[SIGNAL_LAST] = { 0 };
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_auth_debug);
+#define GST_CAT_DEFAULT rtsp_auth_debug
+
+static void gst_rtsp_auth_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_auth_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_auth_finalize (GObject * obj);
+
+static gboolean default_authenticate (GstRTSPAuth * auth, GstRTSPContext * ctx);
+static gboolean default_check (GstRTSPAuth * auth, GstRTSPContext * ctx,
+ const gchar * check);
+static void default_generate_authenticate_header (GstRTSPAuth * auth,
+ GstRTSPContext * ctx);
+
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPAuth, gst_rtsp_auth, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_auth_class_init (GstRTSPAuthClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_auth_get_property;
+ gobject_class->set_property = gst_rtsp_auth_set_property;
+ gobject_class->finalize = gst_rtsp_auth_finalize;
+
+ klass->authenticate = default_authenticate;
+ klass->check = default_check;
+ klass->generate_authenticate_header = default_generate_authenticate_header;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_auth_debug, "rtspauth", 0, "GstRTSPAuth");
+
+ /**
+ * GstRTSPAuth::accept-certificate:
+ * @auth: a #GstRTSPAuth
+ * @connection: a #GTlsConnection
+ * @peer_cert: the peer's #GTlsCertificate
+ * @errors: the problems with @peer_cert.
+ *
+ * Emitted during the TLS handshake after the client certificate has
+ * been received. See also gst_rtsp_auth_set_tls_authentication_mode().
+ *
+ * Returns: %TRUE to accept @peer_cert (which will also
+ * immediately end the signal emission). %FALSE to allow the signal
+ * emission to continue, which will cause the handshake to fail if
+ * no one else overrides it.
+ *
+ * Since: 1.6
+ */
+ signals[SIGNAL_ACCEPT_CERTIFICATE] = g_signal_new ("accept-certificate",
+ G_TYPE_FROM_CLASS (gobject_class),
+ G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPAuthClass, accept_certificate),
+ g_signal_accumulator_true_handled, NULL, NULL,
+ G_TYPE_BOOLEAN, 3, G_TYPE_TLS_CONNECTION, G_TYPE_TLS_CERTIFICATE,
+ G_TYPE_TLS_CERTIFICATE_FLAGS);
+}
+
+static void
+gst_rtsp_auth_init (GstRTSPAuth * auth)
+{
+ GstRTSPAuthPrivate *priv;
+
+ auth->priv = priv = gst_rtsp_auth_get_instance_private (auth);
+
+ g_mutex_init (&priv->lock);
+
+ priv->basic = g_hash_table_new_full (g_str_hash, g_str_equal, g_free,
+ (GDestroyNotify) gst_rtsp_token_unref);
+ priv->digest = g_hash_table_new_full (g_str_hash, g_str_equal, g_free,
+ (GDestroyNotify) gst_rtsp_digest_entry_free);
+ priv->nonces = g_hash_table_new_full (g_str_hash, g_str_equal, g_free,
+ (GDestroyNotify) gst_rtsp_digest_nonce_free);
+
+ /* bitwise or of all methods that need authentication */
+ priv->methods = 0;
+ priv->auth_methods = GST_RTSP_AUTH_BASIC;
+ priv->realm = g_strdup ("GStreamer RTSP Server");
+}
+
+static void
+gst_rtsp_auth_finalize (GObject * obj)
+{
+ GstRTSPAuth *auth = GST_RTSP_AUTH (obj);
+ GstRTSPAuthPrivate *priv = auth->priv;
+
+ GST_INFO ("finalize auth %p", auth);
+
+ if (priv->certificate)
+ g_object_unref (priv->certificate);
+ if (priv->database)
+ g_object_unref (priv->database);
+ g_hash_table_unref (priv->basic);
+ g_hash_table_unref (priv->digest);
+ g_hash_table_unref (priv->nonces);
+ if (priv->default_token)
+ gst_rtsp_token_unref (priv->default_token);
+ g_mutex_clear (&priv->lock);
+ g_free (priv->realm);
+
+ G_OBJECT_CLASS (gst_rtsp_auth_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_auth_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ switch (propid) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_auth_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ switch (propid) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_auth_new:
+ *
+ * Create a new #GstRTSPAuth instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPAuth
+ */
+GstRTSPAuth *
+gst_rtsp_auth_new (void)
+{
+ GstRTSPAuth *result;
+
+ result = g_object_new (GST_TYPE_RTSP_AUTH, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_auth_set_tls_certificate:
+ * @auth: a #GstRTSPAuth
+ * @cert: (transfer none) (allow-none): a #GTlsCertificate
+ *
+ * Set the TLS certificate for the auth. Client connections will only
+ * be accepted when TLS is negotiated.
+ */
+void
+gst_rtsp_auth_set_tls_certificate (GstRTSPAuth * auth, GTlsCertificate * cert)
+{
+ GstRTSPAuthPrivate *priv;
+ GTlsCertificate *old;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+
+ priv = auth->priv;
+
+ if (cert)
+ g_object_ref (cert);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->certificate;
+ priv->certificate = cert;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_auth_get_tls_certificate:
+ * @auth: a #GstRTSPAuth
+ *
+ * Get the #GTlsCertificate used for negotiating TLS @auth.
+ *
+ * Returns: (transfer full) (nullable): the #GTlsCertificate of @auth. g_object_unref() after
+ * usage.
+ */
+GTlsCertificate *
+gst_rtsp_auth_get_tls_certificate (GstRTSPAuth * auth)
+{
+ GstRTSPAuthPrivate *priv;
+ GTlsCertificate *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), NULL);
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->certificate))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_auth_set_tls_database:
+ * @auth: a #GstRTSPAuth
+ * @database: (transfer none) (allow-none): a #GTlsDatabase
+ *
+ * Sets the certificate database that is used to verify peer certificates.
+ * If set to %NULL (the default), then peer certificate validation will always
+ * set the %G_TLS_CERTIFICATE_UNKNOWN_CA error.
+ *
+ * Since: 1.6
+ */
+void
+gst_rtsp_auth_set_tls_database (GstRTSPAuth * auth, GTlsDatabase * database)
+{
+ GstRTSPAuthPrivate *priv;
+ GTlsDatabase *old;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+
+ priv = auth->priv;
+
+ if (database)
+ g_object_ref (database);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->database;
+ priv->database = database;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_auth_get_tls_database:
+ * @auth: a #GstRTSPAuth
+ *
+ * Get the #GTlsDatabase used for verifying client certificate.
+ *
+ * Returns: (transfer full) (nullable): the #GTlsDatabase of @auth. g_object_unref() after
+ * usage.
+ * Since: 1.6
+ */
+GTlsDatabase *
+gst_rtsp_auth_get_tls_database (GstRTSPAuth * auth)
+{
+ GstRTSPAuthPrivate *priv;
+ GTlsDatabase *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), NULL);
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->database))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_auth_set_tls_authentication_mode:
+ * @auth: a #GstRTSPAuth
+ * @mode: a #GTlsAuthenticationMode
+ *
+ * The #GTlsAuthenticationMode to set on the underlying GTlsServerConnection.
+ * When set to another value than %G_TLS_AUTHENTICATION_NONE,
+ * #GstRTSPAuth::accept-certificate signal will be emitted and must be handled.
+ *
+ * Since: 1.6
+ */
+void
+gst_rtsp_auth_set_tls_authentication_mode (GstRTSPAuth * auth,
+ GTlsAuthenticationMode mode)
+{
+ GstRTSPAuthPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->mode = mode;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_auth_get_tls_authentication_mode:
+ * @auth: a #GstRTSPAuth
+ *
+ * Get the #GTlsAuthenticationMode.
+ *
+ * Returns: (transfer full): the #GTlsAuthenticationMode.
+ */
+GTlsAuthenticationMode
+gst_rtsp_auth_get_tls_authentication_mode (GstRTSPAuth * auth)
+{
+ GstRTSPAuthPrivate *priv;
+ GTlsAuthenticationMode result;
+
+ g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), G_TLS_AUTHENTICATION_NONE);
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = priv->mode;
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_auth_set_default_token:
+ * @auth: a #GstRTSPAuth
+ * @token: (transfer none) (allow-none): a #GstRTSPToken
+ *
+ * Set the default #GstRTSPToken to @token in @auth. The default token will
+ * be used for unauthenticated users.
+ */
+void
+gst_rtsp_auth_set_default_token (GstRTSPAuth * auth, GstRTSPToken * token)
+{
+ GstRTSPAuthPrivate *priv;
+ GstRTSPToken *old;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+
+ priv = auth->priv;
+
+ if (token)
+ gst_rtsp_token_ref (token);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->default_token;
+ priv->default_token = token;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ gst_rtsp_token_unref (old);
+}
+
+/**
+ * gst_rtsp_auth_get_default_token:
+ * @auth: a #GstRTSPAuth
+ *
+ * Get the default token for @auth. This token will be used for unauthenticated
+ * users.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPToken of @auth. gst_rtsp_token_unref() after
+ * usage.
+ */
+GstRTSPToken *
+gst_rtsp_auth_get_default_token (GstRTSPAuth * auth)
+{
+ GstRTSPAuthPrivate *priv;
+ GstRTSPToken *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), NULL);
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->default_token))
+ gst_rtsp_token_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_auth_add_basic:
+ * @auth: a #GstRTSPAuth
+ * @basic: the basic token
+ * @token: (transfer none): authorisation token
+ *
+ * Add a basic token for the default authentication algorithm that
+ * enables the client with privileges listed in @token.
+ */
+void
+gst_rtsp_auth_add_basic (GstRTSPAuth * auth, const gchar * basic,
+ GstRTSPToken * token)
+{
+ GstRTSPAuthPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+ g_return_if_fail (basic != NULL);
+ g_return_if_fail (GST_IS_RTSP_TOKEN (token));
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_hash_table_replace (priv->basic, g_strdup (basic),
+ gst_rtsp_token_ref (token));
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_auth_remove_basic:
+ * @auth: a #GstRTSPAuth
+ * @basic: (transfer none): the basic token
+ *
+ * Removes @basic authentication token.
+ */
+void
+gst_rtsp_auth_remove_basic (GstRTSPAuth * auth, const gchar * basic)
+{
+ GstRTSPAuthPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+ g_return_if_fail (basic != NULL);
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_hash_table_remove (priv->basic, basic);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_auth_add_digest:
+ * @auth: a #GstRTSPAuth
+ * @user: the digest user name
+ * @pass: the digest password
+ * @token: (transfer none): authorisation token
+ *
+ * Add a digest @user and @pass for the default authentication algorithm that
+ * enables the client with privileges listed in @token.
+ *
+ * Since: 1.12
+ */
+void
+gst_rtsp_auth_add_digest (GstRTSPAuth * auth, const gchar * user,
+ const gchar * pass, GstRTSPToken * token)
+{
+ GstRTSPAuthPrivate *priv;
+ GstRTSPDigestEntry *entry;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+ g_return_if_fail (user != NULL);
+ g_return_if_fail (pass != NULL);
+ g_return_if_fail (GST_IS_RTSP_TOKEN (token));
+
+ priv = auth->priv;
+
+ entry = g_new0 (GstRTSPDigestEntry, 1);
+ entry->token = gst_rtsp_token_ref (token);
+ entry->pass = g_strdup (pass);
+
+ g_mutex_lock (&priv->lock);
+ g_hash_table_replace (priv->digest, g_strdup (user), entry);
+ g_mutex_unlock (&priv->lock);
+}
+
+/* With auth lock taken */
+static gboolean
+update_digest_cb (gchar * key, GstRTSPDigestEntry * entry, GHashTable * digest)
+{
+ g_hash_table_replace (digest, key, entry);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_auth_parse_htdigest:
+ * @path: (type filename): Path to the htdigest file
+ * @token: (transfer none): authorisation token
+ *
+ * Parse the contents of the file at @path and enable the privileges
+ * listed in @token for the users it describes.
+ *
+ * The format of the file is expected to match the format described by
+ * <https://en.wikipedia.org/wiki/Digest_access_authentication#The_.htdigest_file>,
+ * as output by the `htdigest` command.
+ *
+ * Returns: %TRUE if the file was successfully parsed, %FALSE otherwise.
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_auth_parse_htdigest (GstRTSPAuth * auth, const gchar * path,
+ GstRTSPToken * token)
+{
+ GstRTSPAuthPrivate *priv;
+ gboolean ret = FALSE;
+ gchar *line = NULL;
+ gchar *eol = NULL;
+ gchar *contents = NULL;
+ GError *error = NULL;
+ GHashTable *new_entries =
+ g_hash_table_new_full (g_str_hash, g_str_equal, g_free,
+ (GDestroyNotify) gst_rtsp_digest_entry_free);
+
+
+ g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), FALSE);
+ g_return_val_if_fail (path != NULL, FALSE);
+ g_return_val_if_fail (GST_IS_RTSP_TOKEN (token), FALSE);
+
+ priv = auth->priv;
+ if (!g_file_get_contents (path, &contents, NULL, &error)) {
+ GST_ERROR_OBJECT (auth, "Could not parse htdigest: %s", error->message);
+ goto done;
+ }
+
+ for (line = contents; line && *line; line = eol ? eol + 1 : NULL) {
+ GstRTSPDigestEntry *entry;
+ gchar **strv;
+ eol = strchr (line, '\n');
+
+ if (eol)
+ *eol = '\0';
+
+ strv = g_strsplit (line, ":", -1);
+
+ if (!(strv[0] && strv[1] && strv[2] && !strv[3])) {
+ GST_ERROR_OBJECT (auth, "Invalid htdigest format");
+ g_strfreev (strv);
+ goto done;
+ }
+
+ if (strlen (strv[2]) != 32) {
+ GST_ERROR_OBJECT (auth,
+ "Invalid htdigest format, hash is expected to be 32 characters long");
+ g_strfreev (strv);
+ goto done;
+ }
+
+ entry = g_new0 (GstRTSPDigestEntry, 1);
+ entry->token = gst_rtsp_token_ref (token);
+ entry->md5_pass = g_strdup (strv[2]);
+ g_hash_table_replace (new_entries, g_strdup (strv[0]), entry);
+ g_strfreev (strv);
+ }
+
+ ret = TRUE;
+
+ /* We only update digest if the file was entirely valid */
+ g_mutex_lock (&priv->lock);
+ g_hash_table_foreach_steal (new_entries, (GHRFunc) update_digest_cb,
+ priv->digest);
+ g_mutex_unlock (&priv->lock);
+
+done:
+ if (error)
+ g_clear_error (&error);
+ g_free (contents);
+ g_hash_table_unref (new_entries);
+ return ret;
+}
+
+/**
+ * gst_rtsp_auth_remove_digest:
+ * @auth: a #GstRTSPAuth
+ * @user: (transfer none): the digest user name
+ *
+ * Removes a digest user.
+ *
+ * Since: 1.12
+ */
+void
+gst_rtsp_auth_remove_digest (GstRTSPAuth * auth, const gchar * user)
+{
+ GstRTSPAuthPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+ g_return_if_fail (user != NULL);
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_hash_table_remove (priv->digest, user);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_auth_set_supported_methods:
+ * @auth: a #GstRTSPAuth
+ * @methods: supported methods
+ *
+ * Sets the supported authentication @methods for @auth.
+ *
+ * Since: 1.12
+ */
+void
+gst_rtsp_auth_set_supported_methods (GstRTSPAuth * auth,
+ GstRTSPAuthMethod methods)
+{
+ GstRTSPAuthPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->auth_methods = methods;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_auth_get_supported_methods:
+ * @auth: a #GstRTSPAuth
+ *
+ * Gets the supported authentication methods of @auth.
+ *
+ * Returns: The supported authentication methods
+ *
+ * Since: 1.12
+ */
+GstRTSPAuthMethod
+gst_rtsp_auth_get_supported_methods (GstRTSPAuth * auth)
+{
+ GstRTSPAuthPrivate *priv;
+ GstRTSPAuthMethod methods;
+
+ g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), 0);
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ methods = priv->auth_methods;
+ g_mutex_unlock (&priv->lock);
+
+ return methods;
+}
+
+typedef struct
+{
+ GstRTSPAuth *auth;
+ GstRTSPDigestNonce *nonce;
+} RemoveNonceData;
+
+static void
+remove_nonce (gpointer data, GObject * object)
+{
+ RemoveNonceData *remove_nonce_data = data;
+
+ g_mutex_lock (&remove_nonce_data->auth->priv->lock);
+ g_hash_table_remove (remove_nonce_data->auth->priv->nonces,
+ remove_nonce_data->nonce->nonce);
+ g_mutex_unlock (&remove_nonce_data->auth->priv->lock);
+
+ g_object_unref (remove_nonce_data->auth);
+ g_free (remove_nonce_data);
+}
+
+static gboolean
+default_digest_auth (GstRTSPAuth * auth, GstRTSPContext * ctx,
+ GstRTSPAuthParam ** param)
+{
+ const gchar *realm = NULL, *user = NULL, *nonce = NULL;
+ const gchar *response = NULL, *uri = NULL;
+ GstRTSPDigestNonce *nonce_entry = NULL;
+ GstRTSPDigestEntry *digest_entry;
+ gchar *expected_response = NULL;
+ gboolean ret = FALSE;
+
+ GST_DEBUG_OBJECT (auth, "check Digest auth");
+
+ if (!param)
+ return ret;
+
+ while (*param) {
+ if (!realm && strcmp ((*param)->name, "realm") == 0 && (*param)->value)
+ realm = (*param)->value;
+ else if (!user && strcmp ((*param)->name, "username") == 0
+ && (*param)->value)
+ user = (*param)->value;
+ else if (!nonce && strcmp ((*param)->name, "nonce") == 0 && (*param)->value)
+ nonce = (*param)->value;
+ else if (!response && strcmp ((*param)->name, "response") == 0
+ && (*param)->value)
+ response = (*param)->value;
+ else if (!uri && strcmp ((*param)->name, "uri") == 0 && (*param)->value)
+ uri = (*param)->value;
+
+ param++;
+ }
+
+ if (!realm || !user || !nonce || !response || !uri)
+ return FALSE;
+
+ g_mutex_lock (&auth->priv->lock);
+ digest_entry = g_hash_table_lookup (auth->priv->digest, user);
+ if (!digest_entry)
+ goto out;
+ nonce_entry = g_hash_table_lookup (auth->priv->nonces, nonce);
+ if (!nonce_entry)
+ goto out;
+
+ if (strcmp (nonce_entry->ip, gst_rtsp_connection_get_ip (ctx->conn)) != 0)
+ goto out;
+ if (nonce_entry->client && nonce_entry->client != ctx->client)
+ goto out;
+
+ if (digest_entry->md5_pass) {
+ expected_response = gst_rtsp_generate_digest_auth_response_from_md5 (NULL,
+ gst_rtsp_method_as_text (ctx->method), digest_entry->md5_pass,
+ uri, nonce);
+ } else {
+ expected_response =
+ gst_rtsp_generate_digest_auth_response (NULL,
+ gst_rtsp_method_as_text (ctx->method), realm, user,
+ digest_entry->pass, uri, nonce);
+ }
+
+ if (!expected_response || strcmp (response, expected_response) != 0)
+ goto out;
+
+ ctx->token = digest_entry->token;
+ ret = TRUE;
+
+out:
+ if (nonce_entry && !nonce_entry->client) {
+ RemoveNonceData *remove_nonce_data = g_new (RemoveNonceData, 1);
+
+ nonce_entry->client = ctx->client;
+ remove_nonce_data->nonce = nonce_entry;
+ remove_nonce_data->auth = g_object_ref (auth);
+ g_object_weak_ref (G_OBJECT (ctx->client), remove_nonce, remove_nonce_data);
+ }
+ g_mutex_unlock (&auth->priv->lock);
+
+ g_free (expected_response);
+
+ return ret;
+}
+
+static gboolean
+default_authenticate (GstRTSPAuth * auth, GstRTSPContext * ctx)
+{
+ GstRTSPAuthPrivate *priv = auth->priv;
+ GstRTSPAuthCredential **credentials, **credential;
+
+ GST_DEBUG_OBJECT (auth, "authenticate");
+
+ g_mutex_lock (&priv->lock);
+ /* FIXME, need to ref but we have no way to unref when the ctx is
+ * popped */
+ ctx->token = priv->default_token;
+ g_mutex_unlock (&priv->lock);
+
+ credentials =
+ gst_rtsp_message_parse_auth_credentials (ctx->request,
+ GST_RTSP_HDR_AUTHORIZATION);
+ if (!credentials)
+ goto no_auth;
+
+ /* parse type */
+ credential = credentials;
+ while (*credential) {
+ if ((*credential)->scheme == GST_RTSP_AUTH_BASIC) {
+ GstRTSPToken *token;
+
+ GST_DEBUG_OBJECT (auth, "check Basic auth");
+ g_mutex_lock (&priv->lock);
+ if ((*credential)->authorization && (token =
+ g_hash_table_lookup (priv->basic,
+ (*credential)->authorization))) {
+ GST_DEBUG_OBJECT (auth, "setting token %p", token);
+ ctx->token = token;
+ g_mutex_unlock (&priv->lock);
+ break;
+ }
+ g_mutex_unlock (&priv->lock);
+ } else if ((*credential)->scheme == GST_RTSP_AUTH_DIGEST) {
+ if (default_digest_auth (auth, ctx, (*credential)->params))
+ break;
+ }
+
+ credential++;
+ }
+
+ gst_rtsp_auth_credentials_free (credentials);
+ return TRUE;
+
+no_auth:
+ {
+ GST_DEBUG_OBJECT (auth, "no authorization header found");
+ return TRUE;
+ }
+}
+
+static void
+default_generate_authenticate_header (GstRTSPAuth * auth, GstRTSPContext * ctx)
+{
+ if (auth->priv->auth_methods & GST_RTSP_AUTH_BASIC) {
+ gchar *auth_header =
+ g_strdup_printf ("Basic realm=\"%s\"", auth->priv->realm);
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_WWW_AUTHENTICATE,
+ auth_header);
+ g_free (auth_header);
+ }
+
+ if (auth->priv->auth_methods & GST_RTSP_AUTH_DIGEST) {
+ GstRTSPDigestNonce *nonce;
+ gchar *nonce_value, *auth_header;
+
+ nonce_value =
+ g_strdup_printf ("%08x%08x", g_random_int (), g_random_int ());
+
+ auth_header =
+ g_strdup_printf
+ ("Digest realm=\"%s\", nonce=\"%s\"", auth->priv->realm, nonce_value);
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_WWW_AUTHENTICATE,
+ auth_header);
+ g_free (auth_header);
+
+ nonce = g_new0 (GstRTSPDigestNonce, 1);
+ nonce->nonce = g_strdup (nonce_value);
+ nonce->timestamp = g_get_monotonic_time ();
+ nonce->ip = g_strdup (gst_rtsp_connection_get_ip (ctx->conn));
+ g_mutex_lock (&auth->priv->lock);
+ g_hash_table_replace (auth->priv->nonces, nonce_value, nonce);
+
+ if (auth->priv->last_nonce_check == 0)
+ auth->priv->last_nonce_check = nonce->timestamp;
+
+ /* 30 second nonce timeout */
+ if (nonce->timestamp - auth->priv->last_nonce_check >= 30 * G_USEC_PER_SEC) {
+ GHashTableIter iter;
+ gpointer key, value;
+
+ g_hash_table_iter_init (&iter, auth->priv->nonces);
+ while (g_hash_table_iter_next (&iter, &key, &value)) {
+ GstRTSPDigestNonce *tmp = value;
+
+ if (!tmp->client
+ && nonce->timestamp - tmp->timestamp >= 30 * G_USEC_PER_SEC)
+ g_hash_table_iter_remove (&iter);
+ }
+ auth->priv->last_nonce_check = nonce->timestamp;
+ }
+
+ g_mutex_unlock (&auth->priv->lock);
+ }
+}
+
+static void
+send_response (GstRTSPAuth * auth, GstRTSPStatusCode code, GstRTSPContext * ctx)
+{
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ if (code == GST_RTSP_STS_UNAUTHORIZED) {
+ GstRTSPAuthClass *klass;
+
+ klass = GST_RTSP_AUTH_GET_CLASS (auth);
+
+ if (klass->generate_authenticate_header)
+ klass->generate_authenticate_header (auth, ctx);
+ }
+ gst_rtsp_client_send_message (ctx->client, ctx->session, ctx->response);
+}
+
+static gboolean
+ensure_authenticated (GstRTSPAuth * auth, GstRTSPContext * ctx)
+{
+ GstRTSPAuthClass *klass;
+
+ klass = GST_RTSP_AUTH_GET_CLASS (auth);
+
+ /* we need a token to check */
+ if (ctx->token == NULL) {
+ if (klass->authenticate) {
+ if (!klass->authenticate (auth, ctx))
+ goto authenticate_failed;
+ }
+ }
+ if (ctx->token == NULL)
+ goto no_auth;
+
+ return TRUE;
+
+/* ERRORS */
+authenticate_failed:
+ {
+ GST_DEBUG_OBJECT (auth, "authenticate failed");
+ send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+ return FALSE;
+ }
+no_auth:
+ {
+ GST_DEBUG_OBJECT (auth, "no authorization token found");
+ send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+ return FALSE;
+ }
+}
+
+static gboolean
+accept_certificate_cb (GTlsConnection * conn, GTlsCertificate * peer_cert,
+ GTlsCertificateFlags errors, GstRTSPAuth * auth)
+{
+ gboolean ret = FALSE;
+
+ g_signal_emit (auth, signals[SIGNAL_ACCEPT_CERTIFICATE], 0,
+ conn, peer_cert, errors, &ret);
+
+ return ret;
+}
+
+/* new connection */
+static gboolean
+check_connect (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+ GstRTSPAuthPrivate *priv = auth->priv;
+ GTlsConnection *tls;
+
+ /* configure the connection */
+
+ if (priv->certificate) {
+ tls = gst_rtsp_connection_get_tls (ctx->conn, NULL);
+ g_tls_connection_set_certificate (tls, priv->certificate);
+ }
+
+ if (priv->mode != G_TLS_AUTHENTICATION_NONE) {
+ tls = gst_rtsp_connection_get_tls (ctx->conn, NULL);
+ g_tls_connection_set_database (tls, priv->database);
+ g_object_set (tls, "authentication-mode", priv->mode, NULL);
+ g_signal_connect (tls, "accept-certificate",
+ G_CALLBACK (accept_certificate_cb), auth);
+ }
+
+ return TRUE;
+}
+
+/* check url and methods */
+static gboolean
+check_url (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+ GstRTSPAuthPrivate *priv = auth->priv;
+
+ if ((ctx->method & priv->methods) != 0)
+ if (!ensure_authenticated (auth, ctx))
+ goto not_authenticated;
+
+ return TRUE;
+
+ /* ERRORS */
+not_authenticated:
+ {
+ return FALSE;
+ }
+}
+
+/* check access to media factory */
+static gboolean
+check_factory (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+ const gchar *role;
+ GstRTSPPermissions *perms;
+
+ if (!ensure_authenticated (auth, ctx))
+ return FALSE;
+
+ if (!(role = gst_rtsp_token_get_string (ctx->token,
+ GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE)))
+ goto no_media_role;
+ if (!(perms = gst_rtsp_media_factory_get_permissions (ctx->factory)))
+ goto no_permissions;
+
+ if (g_str_equal (check, GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS)) {
+ if (!gst_rtsp_permissions_is_allowed (perms, role,
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS))
+ goto no_access;
+ } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT)) {
+ if (!gst_rtsp_permissions_is_allowed (perms, role,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT))
+ goto no_construct;
+ }
+
+ gst_rtsp_permissions_unref (perms);
+
+ return TRUE;
+
+ /* ERRORS */
+no_media_role:
+ {
+ GST_DEBUG_OBJECT (auth, "no media factory role found");
+ send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+ return FALSE;
+ }
+no_permissions:
+ {
+ GST_DEBUG_OBJECT (auth, "no permissions on media factory found");
+ send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+ return FALSE;
+ }
+no_access:
+ {
+ GST_DEBUG_OBJECT (auth, "no permissions to access media factory");
+ gst_rtsp_permissions_unref (perms);
+ send_response (auth, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_construct:
+ {
+ GST_DEBUG_OBJECT (auth, "no permissions to construct media factory");
+ gst_rtsp_permissions_unref (perms);
+ send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+ return FALSE;
+ }
+}
+
+static gboolean
+check_client_settings (GstRTSPAuth * auth, GstRTSPContext * ctx,
+ const gchar * check)
+{
+ if (!ensure_authenticated (auth, ctx))
+ return FALSE;
+
+ return gst_rtsp_token_is_allowed (ctx->token,
+ GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS);
+}
+
+static gboolean
+default_check (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+ gboolean res = FALSE;
+
+ /* FIXME, use hastable or so */
+ if (g_str_equal (check, GST_RTSP_AUTH_CHECK_CONNECT)) {
+ res = check_connect (auth, ctx, check);
+ } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_URL)) {
+ res = check_url (auth, ctx, check);
+ } else if (g_str_has_prefix (check, "auth.check.media.factory.")) {
+ res = check_factory (auth, ctx, check);
+ } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS)) {
+ res = check_client_settings (auth, ctx, check);
+ }
+ return res;
+}
+
+static gboolean
+no_auth_check (const gchar * check)
+{
+ gboolean res;
+
+ if (g_str_equal (check, GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS))
+ res = FALSE;
+ else
+ res = TRUE;
+
+ return res;
+}
+
+/**
+ * gst_rtsp_auth_check:
+ * @check: the item to check
+ *
+ * Check if @check is allowed in the current context.
+ *
+ * Returns: FALSE if check failed.
+ */
+gboolean
+gst_rtsp_auth_check (const gchar * check)
+{
+ gboolean result = FALSE;
+ GstRTSPAuthClass *klass;
+ GstRTSPContext *ctx;
+ GstRTSPAuth *auth;
+
+ g_return_val_if_fail (check != NULL, FALSE);
+
+ if (!(ctx = gst_rtsp_context_get_current ()))
+ goto no_context;
+
+ /* no auth, we don't need to check */
+ if (!(auth = ctx->auth))
+ return no_auth_check (check);
+
+ klass = GST_RTSP_AUTH_GET_CLASS (auth);
+
+ GST_DEBUG_OBJECT (auth, "check authorization '%s'", check);
+
+ if (klass->check)
+ result = klass->check (auth, ctx, check);
+
+ return result;
+
+ /* ERRORS */
+no_context:
+ {
+ GST_ERROR ("no context found");
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_auth_make_basic:
+ * @user: a userid
+ * @pass: a password
+ *
+ * Construct a Basic authorisation token from @user and @pass.
+ *
+ * Returns: (transfer full): the base64 encoding of the string @user:@pass.
+ * g_free() after usage.
+ */
+gchar *
+gst_rtsp_auth_make_basic (const gchar * user, const gchar * pass)
+{
+ gchar *user_pass;
+ gchar *result;
+
+ g_return_val_if_fail (user != NULL, NULL);
+ g_return_val_if_fail (pass != NULL, NULL);
+
+ user_pass = g_strjoin (":", user, pass, NULL);
+ result = g_base64_encode ((guchar *) user_pass, strlen (user_pass));
+ g_free (user_pass);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_auth_set_realm:
+ *
+ * Set the @realm of @auth
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_auth_set_realm (GstRTSPAuth * auth, const gchar * realm)
+{
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+ g_return_if_fail (realm != NULL);
+
+ if (auth->priv->realm)
+ g_free (auth->priv->realm);
+
+ auth->priv->realm = g_strdup (realm);
+}
+
+/**
+ * gst_rtsp_auth_get_realm:
+ *
+ * Returns: (transfer full): the @realm of @auth
+ *
+ * Since: 1.16
+ */
+gchar *
+gst_rtsp_auth_get_realm (GstRTSPAuth * auth)
+{
+ g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), NULL);
+
+ return g_strdup (auth->priv->realm);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_AUTH_H__
+#define __GST_RTSP_AUTH_H__
+
+typedef struct _GstRTSPAuth GstRTSPAuth;
+typedef struct _GstRTSPAuthClass GstRTSPAuthClass;
+typedef struct _GstRTSPAuthPrivate GstRTSPAuthPrivate;
+
+#include "rtsp-server-prelude.h"
+#include "rtsp-client.h"
+#include "rtsp-token.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_AUTH (gst_rtsp_auth_get_type ())
+#define GST_IS_RTSP_AUTH(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_AUTH))
+#define GST_IS_RTSP_AUTH_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_AUTH))
+#define GST_RTSP_AUTH_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_AUTH, GstRTSPAuthClass))
+#define GST_RTSP_AUTH(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_AUTH, GstRTSPAuth))
+#define GST_RTSP_AUTH_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_AUTH, GstRTSPAuthClass))
+#define GST_RTSP_AUTH_CAST(obj) ((GstRTSPAuth*)(obj))
+#define GST_RTSP_AUTH_CLASS_CAST(klass) ((GstRTSPAuthClass*)(klass))
+
+/**
+ * GstRTSPAuth:
+ *
+ * The authentication structure.
+ */
+struct _GstRTSPAuth {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPAuthPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPAuthClass:
+ * @authenticate: check the authentication of a client. The default implementation
+ * checks if the authentication in the header matches one of the basic
+ * authentication tokens. This function should set the authgroup field
+ * in the context.
+ * @check: check if a resource can be accessed. this function should
+ * call authenticate to authenticate the client when needed. The method
+ * should also construct and send an appropriate response message on
+ * error.
+ *
+ * The authentication class.
+ */
+struct _GstRTSPAuthClass {
+ GObjectClass parent_class;
+
+ gboolean (*authenticate) (GstRTSPAuth *auth, GstRTSPContext *ctx);
+ gboolean (*check) (GstRTSPAuth *auth, GstRTSPContext *ctx,
+ const gchar *check);
+ void (*generate_authenticate_header) (GstRTSPAuth *auth, GstRTSPContext *ctx);
+ gboolean (*accept_certificate) (GstRTSPAuth *auth,
+ GTlsConnection *connection,
+ GTlsCertificate *peer_cert,
+ GTlsCertificateFlags errors);
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING - 1];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_auth_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPAuth * gst_rtsp_auth_new (void);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_auth_set_tls_certificate (GstRTSPAuth *auth, GTlsCertificate *cert);
+
+GST_RTSP_SERVER_API
+GTlsCertificate * gst_rtsp_auth_get_tls_certificate (GstRTSPAuth *auth);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_auth_set_tls_database (GstRTSPAuth *auth, GTlsDatabase *database);
+
+GST_RTSP_SERVER_API
+GTlsDatabase * gst_rtsp_auth_get_tls_database (GstRTSPAuth *auth);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_auth_set_tls_authentication_mode (GstRTSPAuth *auth, GTlsAuthenticationMode mode);
+
+GST_RTSP_SERVER_API
+GTlsAuthenticationMode gst_rtsp_auth_get_tls_authentication_mode (GstRTSPAuth *auth);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_auth_set_default_token (GstRTSPAuth *auth, GstRTSPToken *token);
+
+GST_RTSP_SERVER_API
+GstRTSPToken * gst_rtsp_auth_get_default_token (GstRTSPAuth *auth);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_auth_add_basic (GstRTSPAuth *auth, const gchar * basic,
+ GstRTSPToken *token);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_auth_remove_basic (GstRTSPAuth *auth, const gchar * basic);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_auth_add_digest (GstRTSPAuth *auth, const gchar *user,
+ const gchar *pass, GstRTSPToken *token);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_auth_remove_digest (GstRTSPAuth *auth, const gchar *user);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_auth_set_supported_methods (GstRTSPAuth *auth, GstRTSPAuthMethod methods);
+
+GST_RTSP_SERVER_API
+GstRTSPAuthMethod gst_rtsp_auth_get_supported_methods (GstRTSPAuth *auth);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_auth_check (const gchar *check);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_auth_parse_htdigest (GstRTSPAuth *auth, const gchar *path, GstRTSPToken *token);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_auth_set_realm (GstRTSPAuth *auth, const gchar *realm);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_auth_get_realm (GstRTSPAuth *auth);
+
+/* helpers */
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_auth_make_basic (const gchar * user, const gchar * pass);
+
+/* checks */
+/**
+ * GST_RTSP_AUTH_CHECK_CONNECT:
+ *
+ * Check a new connection
+ */
+#define GST_RTSP_AUTH_CHECK_CONNECT "auth.check.connect"
+/**
+ * GST_RTSP_AUTH_CHECK_URL:
+ *
+ * Check the URL and methods
+ */
+#define GST_RTSP_AUTH_CHECK_URL "auth.check.url"
+/**
+ * GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS:
+ *
+ * Check if access is allowed to a factory.
+ * When access is not allowed an 404 Not Found is sent in the response.
+ */
+#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS "auth.check.media.factory.access"
+/**
+ * GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT:
+ *
+ * Check if media can be constructed from a media factory
+ * A response should be sent on error.
+ */
+#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT "auth.check.media.factory.construct"
+/**
+ * GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS:
+ *
+ * Check if the client can specify TTL, destination and
+ * port pair in multicast. No response is sent when the check returns
+ * %FALSE.
+ */
+#define GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS "auth.check.transport.client-settings"
+
+
+/* tokens */
+/**
+ * GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE:
+ *
+ * G_TYPE_STRING, the role to use when dealing with media factories
+ *
+ * The default #GstRTSPAuth object uses this string in the token to find the
+ * role of the media factory. It will then retrieve the #GstRTSPPermissions of
+ * the media factory and retrieve the role with the same name.
+ */
+#define GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE "media.factory.role"
+/**
+ * GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS:
+ *
+ * G_TYPE_BOOLEAN, %TRUE if the client can specify TTL, destination and
+ * port pair in multicast.
+ */
+#define GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS "transport.client-settings"
+
+/* permissions */
+/**
+ * GST_RTSP_PERM_MEDIA_FACTORY_ACCESS:
+ *
+ * G_TYPE_BOOLEAN, %TRUE if the media can be accessed, %FALSE will
+ * return a 404 Not Found error when trying to access the media.
+ */
+#define GST_RTSP_PERM_MEDIA_FACTORY_ACCESS "media.factory.access"
+/**
+ * GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT:
+ *
+ * G_TYPE_BOOLEAN, %TRUE if the media can be constructed, %FALSE will
+ * return a 404 Not Found error when trying to access the media.
+ */
+#define GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT "media.factory.construct"
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPAuth, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_AUTH_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2015 Centricular Ltd
+ * Author: Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-client
+ * @short_description: A client connection state
+ * @see_also: #GstRTSPServer, #GstRTSPThreadPool
+ *
+ * The client object handles the connection with a client for as long as a TCP
+ * connection is open.
+ *
+ * A #GstRTSPClient is created by #GstRTSPServer when a new connection is
+ * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
+ * #GstRTSPAuth and #GstRTSPThreadPool from the server.
+ *
+ * The client connection should be configured with the #GstRTSPConnection using
+ * gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
+ * using gst_rtsp_client_attach(). From then on the client will handle requests
+ * on the connection.
+ *
+ * Use gst_rtsp_client_session_filter() to iterate or modify all the
+ * #GstRTSPSession objects managed by the client object.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdio.h>
+#include <string.h>
+
+#include <gst/sdp/gstmikey.h>
+#include <gst/rtsp/gstrtsp-enumtypes.h>
+
+#include "rtsp-client.h"
+#include "rtsp-sdp.h"
+#include "rtsp-params.h"
+#include "rtsp-server-internal.h"
+
+typedef enum
+{
+ TUNNEL_STATE_UNKNOWN,
+ TUNNEL_STATE_GET,
+ TUNNEL_STATE_POST
+} GstRTSPTunnelState;
+
+/* locking order:
+ * send_lock, lock, tunnels_lock
+ */
+
+struct _GstRTSPClientPrivate
+{
+ GMutex lock; /* protects everything else */
+ GMutex send_lock;
+ GMutex watch_lock;
+ GstRTSPConnection *connection;
+ GstRTSPWatch *watch;
+ GMainContext *watch_context;
+ gchar *server_ip;
+ gboolean is_ipv6;
+
+ /* protected by send_lock */
+ GstRTSPClientSendFunc send_func;
+ gpointer send_data;
+ GDestroyNotify send_notify;
+ GstRTSPClientSendMessagesFunc send_messages_func;
+ gpointer send_messages_data;
+ GDestroyNotify send_messages_notify;
+ GArray *data_seqs;
+
+ GstRTSPSessionPool *session_pool;
+ gulong session_removed_id;
+ GstRTSPMountPoints *mount_points;
+ GstRTSPAuth *auth;
+ GstRTSPThreadPool *thread_pool;
+
+ /* used to cache the media in the last requested DESCRIBE so that
+ * we can pick it up in the next SETUP immediately */
+ gchar *path;
+ GstRTSPMedia *media;
+
+ GHashTable *transports;
+ GList *sessions;
+ guint sessions_cookie;
+
+ gboolean drop_backlog;
+ gint post_session_timeout;
+
+ guint content_length_limit;
+
+ gboolean had_session;
+ GSource *rtsp_ctrl_timeout;
+ guint rtsp_ctrl_timeout_cnt;
+
+ /* The version currently being used */
+ GstRTSPVersion version;
+
+ GHashTable *pipelined_requests; /* pipelined_request_id -> session_id */
+ GstRTSPTunnelState tstate;
+};
+
+typedef struct
+{
+ guint8 channel;
+ guint seq;
+} DataSeq;
+
+static GMutex tunnels_lock;
+static GHashTable *tunnels; /* protected by tunnels_lock */
+
+#define WATCH_BACKLOG_SIZE 100
+
+#define DEFAULT_SESSION_POOL NULL
+#define DEFAULT_MOUNT_POINTS NULL
+#define DEFAULT_DROP_BACKLOG TRUE
+#define DEFAULT_POST_SESSION_TIMEOUT -1
+
+#define RTSP_CTRL_CB_INTERVAL 1
+#define RTSP_CTRL_TIMEOUT_VALUE 60
+
+enum
+{
+ PROP_0,
+ PROP_SESSION_POOL,
+ PROP_MOUNT_POINTS,
+ PROP_DROP_BACKLOG,
+ PROP_POST_SESSION_TIMEOUT,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_CLOSED,
+ SIGNAL_NEW_SESSION,
+ SIGNAL_PRE_OPTIONS_REQUEST,
+ SIGNAL_OPTIONS_REQUEST,
+ SIGNAL_PRE_DESCRIBE_REQUEST,
+ SIGNAL_DESCRIBE_REQUEST,
+ SIGNAL_PRE_SETUP_REQUEST,
+ SIGNAL_SETUP_REQUEST,
+ SIGNAL_PRE_PLAY_REQUEST,
+ SIGNAL_PLAY_REQUEST,
+ SIGNAL_PRE_PAUSE_REQUEST,
+ SIGNAL_PAUSE_REQUEST,
+ SIGNAL_PRE_TEARDOWN_REQUEST,
+ SIGNAL_TEARDOWN_REQUEST,
+ SIGNAL_PRE_SET_PARAMETER_REQUEST,
+ SIGNAL_SET_PARAMETER_REQUEST,
+ SIGNAL_PRE_GET_PARAMETER_REQUEST,
+ SIGNAL_GET_PARAMETER_REQUEST,
+ SIGNAL_HANDLE_RESPONSE,
+ SIGNAL_SEND_MESSAGE,
+ SIGNAL_PRE_ANNOUNCE_REQUEST,
+ SIGNAL_ANNOUNCE_REQUEST,
+ SIGNAL_PRE_RECORD_REQUEST,
+ SIGNAL_RECORD_REQUEST,
+ SIGNAL_CHECK_REQUIREMENTS,
+ SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
+#define GST_CAT_DEFAULT rtsp_client_debug
+
+static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
+
+static void gst_rtsp_client_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_client_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_client_finalize (GObject * obj);
+
+static void rtsp_ctrl_timeout_remove (GstRTSPClient * client);
+
+static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
+static gboolean handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx,
+ GstRTSPMedia * media, GstSDPMessage * sdp);
+static gboolean default_configure_client_media (GstRTSPClient * client,
+ GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
+static gboolean default_configure_client_transport (GstRTSPClient * client,
+ GstRTSPContext * ctx, GstRTSPTransport * ct);
+static GstRTSPResult default_params_set (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static GstRTSPResult default_params_get (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static gchar *default_make_path_from_uri (GstRTSPClient * client,
+ const GstRTSPUrl * uri);
+static void client_session_removed (GstRTSPSessionPool * pool,
+ GstRTSPSession * session, GstRTSPClient * client);
+static GstRTSPStatusCode default_pre_signal_handler (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static gboolean pre_signal_accumulator (GSignalInvocationHint * ihint,
+ GValue * return_accu, const GValue * handler_return, gpointer data);
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_client_class_init (GstRTSPClientClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_client_get_property;
+ gobject_class->set_property = gst_rtsp_client_set_property;
+ gobject_class->finalize = gst_rtsp_client_finalize;
+
+ klass->create_sdp = create_sdp;
+ klass->handle_sdp = handle_sdp;
+ klass->configure_client_media = default_configure_client_media;
+ klass->configure_client_transport = default_configure_client_transport;
+ klass->params_set = default_params_set;
+ klass->params_get = default_params_get;
+ klass->make_path_from_uri = default_make_path_from_uri;
+
+ klass->pre_options_request = default_pre_signal_handler;
+ klass->pre_describe_request = default_pre_signal_handler;
+ klass->pre_setup_request = default_pre_signal_handler;
+ klass->pre_play_request = default_pre_signal_handler;
+ klass->pre_pause_request = default_pre_signal_handler;
+ klass->pre_teardown_request = default_pre_signal_handler;
+ klass->pre_set_parameter_request = default_pre_signal_handler;
+ klass->pre_get_parameter_request = default_pre_signal_handler;
+ klass->pre_announce_request = default_pre_signal_handler;
+ klass->pre_record_request = default_pre_signal_handler;
+
+ g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
+ g_param_spec_object ("session-pool", "Session Pool",
+ "The session pool to use for client session",
+ GST_TYPE_RTSP_SESSION_POOL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
+ g_param_spec_object ("mount-points", "Mount Points",
+ "The mount points to use for client session",
+ GST_TYPE_RTSP_MOUNT_POINTS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
+ g_param_spec_boolean ("drop-backlog", "Drop Backlog",
+ "Drop data when the backlog queue is full",
+ DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClient::post-session-timeout:
+ *
+ * An extra tcp timeout ( > 0) after session timeout, in seconds.
+ * The tcp connection will be kept alive until this timeout happens to give
+ * the client a possibility to reuse the connection.
+ * 0 means that the connection will be closed immediately after the session
+ * timeout.
+ *
+ * Default value is -1 seconds, meaning that we let the system close
+ * the connection.
+ *
+ * Since: 1.18
+ */
+ g_object_class_install_property (gobject_class, PROP_POST_SESSION_TIMEOUT,
+ g_param_spec_int ("post-session-timeout", "Post Session Timeout",
+ "An extra TCP connection timeout after session timeout", G_MININT,
+ G_MAXINT, DEFAULT_POST_SESSION_TIMEOUT,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_client_signals[SIGNAL_CLOSED] =
+ g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL, NULL,
+ G_TYPE_NONE, 0, G_TYPE_NONE);
+
+ gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
+ g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL, NULL,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
+
+ /**
+ * GstRTSPClient::pre-options-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ *
+ * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
+ * otherwise an appropriate return code
+ *
+ * Since: 1.12
+ */
+ gst_rtsp_client_signals[SIGNAL_PRE_OPTIONS_REQUEST] =
+ g_signal_new ("pre-options-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ pre_options_request), pre_signal_accumulator, NULL, NULL,
+ GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::options-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
+ g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::pre-describe-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ *
+ * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
+ * otherwise an appropriate return code
+ *
+ * Since: 1.12
+ */
+ gst_rtsp_client_signals[SIGNAL_PRE_DESCRIBE_REQUEST] =
+ g_signal_new ("pre-describe-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ pre_describe_request), pre_signal_accumulator, NULL, NULL,
+ GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::describe-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
+ g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::pre-setup-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ *
+ * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
+ * otherwise an appropriate return code
+ *
+ * Since: 1.12
+ */
+ gst_rtsp_client_signals[SIGNAL_PRE_SETUP_REQUEST] =
+ g_signal_new ("pre-setup-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ pre_setup_request), pre_signal_accumulator, NULL, NULL,
+ GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::setup-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
+ g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::pre-play-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ *
+ * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
+ * otherwise an appropriate return code
+ *
+ * Since: 1.12
+ */
+ gst_rtsp_client_signals[SIGNAL_PRE_PLAY_REQUEST] =
+ g_signal_new ("pre-play-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ pre_play_request), pre_signal_accumulator, NULL,
+ NULL, GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::play-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
+ g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::pre-pause-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ *
+ * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
+ * otherwise an appropriate return code
+ *
+ * Since: 1.12
+ */
+ gst_rtsp_client_signals[SIGNAL_PRE_PAUSE_REQUEST] =
+ g_signal_new ("pre-pause-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ pre_pause_request), pre_signal_accumulator, NULL, NULL,
+ GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::pause-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
+ g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::pre-teardown-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ *
+ * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
+ * otherwise an appropriate return code
+ *
+ * Since: 1.12
+ */
+ gst_rtsp_client_signals[SIGNAL_PRE_TEARDOWN_REQUEST] =
+ g_signal_new ("pre-teardown-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ pre_teardown_request), pre_signal_accumulator, NULL, NULL,
+ GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::teardown-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
+ g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::pre-set-parameter-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ *
+ * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
+ * otherwise an appropriate return code
+ *
+ * Since: 1.12
+ */
+ gst_rtsp_client_signals[SIGNAL_PRE_SET_PARAMETER_REQUEST] =
+ g_signal_new ("pre-set-parameter-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ pre_set_parameter_request), pre_signal_accumulator, NULL, NULL,
+ GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::set-parameter-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
+ g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ set_parameter_request), NULL, NULL, NULL,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::pre-get-parameter-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ *
+ * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
+ * otherwise an appropriate return code
+ *
+ * Since: 1.12
+ */
+ gst_rtsp_client_signals[SIGNAL_PRE_GET_PARAMETER_REQUEST] =
+ g_signal_new ("pre-get-parameter-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ pre_get_parameter_request), pre_signal_accumulator, NULL, NULL,
+ GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::get-parameter-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
+ g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ get_parameter_request), NULL, NULL, NULL,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::handle-response:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
+ g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ handle_response), NULL, NULL, NULL,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::send-message:
+ * @client: The RTSP client
+ * @session: (type GstRtspServer.RTSPSession): The session
+ * @message: (type GstRtsp.RTSPMessage): The message
+ */
+ gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
+ g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ send_message), NULL, NULL, NULL,
+ G_TYPE_NONE, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_POINTER);
+
+ /**
+ * GstRTSPClient::pre-announce-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ *
+ * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
+ * otherwise an appropriate return code
+ *
+ * Since: 1.12
+ */
+ gst_rtsp_client_signals[SIGNAL_PRE_ANNOUNCE_REQUEST] =
+ g_signal_new ("pre-announce-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ pre_announce_request), pre_signal_accumulator, NULL, NULL,
+ GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::announce-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST] =
+ g_signal_new ("announce-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, announce_request),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::pre-record-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ *
+ * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
+ * otherwise an appropriate return code
+ *
+ * Since: 1.12
+ */
+ gst_rtsp_client_signals[SIGNAL_PRE_RECORD_REQUEST] =
+ g_signal_new ("pre-record-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ pre_record_request), pre_signal_accumulator, NULL, NULL,
+ GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::record-request:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ */
+ gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST] =
+ g_signal_new ("record-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, record_request),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::check-requirements:
+ * @client: a #GstRTSPClient
+ * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
+ * @arr: a NULL-terminated array of strings
+ *
+ * Returns: a newly allocated string with comma-separated list of
+ * unsupported options. An empty string must be returned if
+ * all options are supported.
+ *
+ * Since: 1.6
+ */
+ gst_rtsp_client_signals[SIGNAL_CHECK_REQUIREMENTS] =
+ g_signal_new ("check-requirements", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ check_requirements), NULL, NULL, NULL,
+ G_TYPE_STRING, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_STRV);
+
+ tunnels =
+ g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
+ g_mutex_init (&tunnels_lock);
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
+}
+
+static void
+gst_rtsp_client_init (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = gst_rtsp_client_get_instance_private (client);
+
+ client->priv = priv;
+
+ g_mutex_init (&priv->lock);
+ g_mutex_init (&priv->send_lock);
+ g_mutex_init (&priv->watch_lock);
+ priv->data_seqs = g_array_new (FALSE, FALSE, sizeof (DataSeq));
+ priv->drop_backlog = DEFAULT_DROP_BACKLOG;
+ priv->post_session_timeout = DEFAULT_POST_SESSION_TIMEOUT;
+ priv->transports =
+ g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL,
+ g_object_unref);
+ priv->pipelined_requests = g_hash_table_new_full (g_str_hash,
+ g_str_equal, g_free, g_free);
+ priv->tstate = TUNNEL_STATE_UNKNOWN;
+ priv->content_length_limit = G_MAXUINT;
+}
+
+static GstRTSPFilterResult
+filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
+ gpointer user_data)
+{
+ gboolean *closed = user_data;
+ GstRTSPMedia *media;
+ guint i, n_streams;
+ gboolean is_all_udp = TRUE;
+
+ media = gst_rtsp_session_media_get_media (sessmedia);
+ n_streams = gst_rtsp_media_n_streams (media);
+
+ for (i = 0; i < n_streams; i++) {
+ GstRTSPStreamTransport *transport =
+ gst_rtsp_session_media_get_transport (sessmedia, i);
+ const GstRTSPTransport *rtsp_transport;
+
+ if (!transport)
+ continue;
+
+ rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
+ if (rtsp_transport
+ && rtsp_transport->lower_transport != GST_RTSP_LOWER_TRANS_UDP
+ && rtsp_transport->lower_transport != GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ is_all_udp = FALSE;
+ break;
+ }
+ }
+
+ if (!is_all_udp || gst_rtsp_media_is_stop_on_disconnect (media)) {
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
+ return GST_RTSP_FILTER_REMOVE;
+ } else {
+ *closed = FALSE;
+ return GST_RTSP_FILTER_KEEP;
+ }
+}
+
+static void
+client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* check if we already know about this session */
+ if (g_list_find (priv->sessions, session) == NULL) {
+ GST_INFO ("watching session %p", session);
+
+ priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
+ priv->sessions_cookie++;
+
+ /* connect removed session handler, it will be disconnected when the last
+ * session gets removed */
+ if (priv->session_removed_id == 0)
+ priv->session_removed_id = g_signal_connect_data (priv->session_pool,
+ "session-removed", G_CALLBACK (client_session_removed),
+ g_object_ref (client), (GClosureNotify) g_object_unref, 0);
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return;
+}
+
+/* should be called with lock */
+static void
+client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
+ GList * link)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_INFO ("client %p: unwatch session %p", client, session);
+
+ if (link == NULL) {
+ link = g_list_find (priv->sessions, session);
+ if (link == NULL)
+ return;
+ }
+
+ priv->sessions = g_list_delete_link (priv->sessions, link);
+ priv->sessions_cookie++;
+
+ /* if this was the last session, disconnect the handler.
+ * This will also drop the extra client ref */
+ if (!priv->sessions) {
+ g_signal_handler_disconnect (priv->session_pool, priv->session_removed_id);
+ priv->session_removed_id = 0;
+ }
+
+ if (!priv->drop_backlog) {
+ /* unlink all media managed in this session */
+ gst_rtsp_session_filter (session, filter_session_media, client);
+ }
+
+ /* remove the session */
+ g_object_unref (session);
+}
+
+static GstRTSPFilterResult
+cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
+ gpointer user_data)
+{
+ gboolean *closed = user_data;
+ GstRTSPClientPrivate *priv = client->priv;
+
+ if (priv->drop_backlog) {
+ /* unlink all media managed in this session. This needs to happen
+ * without the client lock, so we really want to do it here. */
+ gst_rtsp_session_filter (sess, filter_session_media, user_data);
+ }
+
+ if (*closed)
+ return GST_RTSP_FILTER_REMOVE;
+ else
+ return GST_RTSP_FILTER_KEEP;
+}
+
+static void
+clean_cached_media (GstRTSPClient * client, gboolean unprepare)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ if (priv->path) {
+ g_free (priv->path);
+ priv->path = NULL;
+ }
+ if (priv->media) {
+ if (unprepare)
+ gst_rtsp_media_unprepare (priv->media);
+ g_object_unref (priv->media);
+ priv->media = NULL;
+ }
+}
+
+/* A client is finalized when the connection is broken */
+static void
+gst_rtsp_client_finalize (GObject * obj)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (obj);
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_INFO ("finalize client %p", client);
+
+ /* the watch and related state should be cleared before finalize
+ * as the watch actually holds a strong reference to the client */
+ g_assert (priv->watch == NULL);
+ g_assert (priv->rtsp_ctrl_timeout == NULL);
+
+ if (priv->watch_context) {
+ g_main_context_unref (priv->watch_context);
+ priv->watch_context = NULL;
+ }
+
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
+
+ /* all sessions should have been removed by now. We keep a ref to
+ * the client object for the session removed handler. The ref is
+ * dropped when the last session is removed from the list. */
+ g_assert (priv->sessions == NULL);
+ g_assert (priv->session_removed_id == 0);
+
+ g_array_unref (priv->data_seqs);
+ g_hash_table_unref (priv->transports);
+ g_hash_table_unref (priv->pipelined_requests);
+
+ if (priv->connection)
+ gst_rtsp_connection_free (priv->connection);
+ if (priv->session_pool) {
+ g_object_unref (priv->session_pool);
+ }
+ if (priv->mount_points)
+ g_object_unref (priv->mount_points);
+ if (priv->auth)
+ g_object_unref (priv->auth);
+ if (priv->thread_pool)
+ g_object_unref (priv->thread_pool);
+
+ clean_cached_media (client, TRUE);
+
+ g_free (priv->server_ip);
+ g_mutex_clear (&priv->lock);
+ g_mutex_clear (&priv->send_lock);
+ g_mutex_clear (&priv->watch_lock);
+
+ G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_client_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (object);
+ GstRTSPClientPrivate *priv = client->priv;
+
+ switch (propid) {
+ case PROP_SESSION_POOL:
+ g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
+ break;
+ case PROP_MOUNT_POINTS:
+ g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
+ break;
+ case PROP_DROP_BACKLOG:
+ g_value_set_boolean (value, priv->drop_backlog);
+ break;
+ case PROP_POST_SESSION_TIMEOUT:
+ g_value_set_int (value, priv->post_session_timeout);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_client_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (object);
+ GstRTSPClientPrivate *priv = client->priv;
+
+ switch (propid) {
+ case PROP_SESSION_POOL:
+ gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
+ break;
+ case PROP_MOUNT_POINTS:
+ gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
+ break;
+ case PROP_DROP_BACKLOG:
+ g_mutex_lock (&priv->lock);
+ priv->drop_backlog = g_value_get_boolean (value);
+ g_mutex_unlock (&priv->lock);
+ break;
+ case PROP_POST_SESSION_TIMEOUT:
+ g_mutex_lock (&priv->lock);
+ priv->post_session_timeout = g_value_get_int (value);
+ g_mutex_unlock (&priv->lock);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_client_new:
+ *
+ * Create a new #GstRTSPClient instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPClient
+ */
+GstRTSPClient *
+gst_rtsp_client_new (void)
+{
+ GstRTSPClient *result;
+
+ result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
+
+ return result;
+}
+
+static void
+send_message (GstRTSPClient * client, GstRTSPContext * ctx,
+ GstRTSPMessage * message, gboolean close)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
+ "GStreamer RTSP server");
+
+ /* remove any previous header */
+ gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
+
+ /* add the new session header for new session ids */
+ if (ctx->session) {
+ gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
+ gst_rtsp_session_get_header (ctx->session));
+ }
+
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (message);
+ }
+
+ if (close)
+ gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
+
+ if (ctx->request)
+ message->type_data.response.version =
+ ctx->request->type_data.request.version;
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
+ 0, ctx, message);
+
+ g_mutex_lock (&priv->send_lock);
+ if (priv->send_messages_func) {
+ priv->send_messages_func (client, message, 1, close, priv->send_data);
+ } else if (priv->send_func) {
+ priv->send_func (client, message, close, priv->send_data);
+ }
+ g_mutex_unlock (&priv->send_lock);
+
+ gst_rtsp_message_unset (message);
+}
+
+static void
+send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
+ GstRTSPContext * ctx)
+{
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ ctx->session = NULL;
+
+ send_message (client, ctx, ctx->response, FALSE);
+}
+
+static void
+send_option_not_supported_response (GstRTSPClient * client,
+ GstRTSPContext * ctx, const gchar * unsupported_options)
+{
+ GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
+
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ if (unsupported_options != NULL) {
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
+ unsupported_options);
+ }
+
+ ctx->session = NULL;
+
+ send_message (client, ctx, ctx->response, FALSE);
+}
+
+static gboolean
+paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
+{
+ if (path1 == NULL || path2 == NULL)
+ return FALSE;
+
+ if (strlen (path1) != len2)
+ return FALSE;
+
+ if (strncmp (path1, path2, len2))
+ return FALSE;
+
+ return TRUE;
+}
+
+/* this function is called to initially find the media for the DESCRIBE request
+ * but is cached for when the same client (without breaking the connection) is
+ * doing a setup for the exact same url. */
+static GstRTSPMedia *
+find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
+ gint * matched)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ gint path_len;
+
+ /* find the longest matching factory for the uri first */
+ if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
+ path, matched)))
+ goto no_factory;
+
+ ctx->factory = factory;
+
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
+ goto no_factory_access;
+
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
+ goto not_authorized;
+
+ if (matched)
+ path_len = *matched;
+ else
+ path_len = strlen (path);
+
+ if (!paths_are_equal (priv->path, path, path_len)) {
+ /* remove any previously cached values before we try to construct a new
+ * media for uri */
+ clean_cached_media (client, TRUE);
+
+ /* prepare the media and add it to the pipeline */
+ if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
+ goto no_media;
+
+ ctx->media = media;
+
+ if (!(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_RECORD)) {
+ GstRTSPThread *thread;
+
+ thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, ctx);
+ if (thread == NULL)
+ goto no_thread;
+
+ /* prepare the media */
+ if (!gst_rtsp_media_prepare (media, thread))
+ goto no_prepare;
+ }
+
+ /* now keep track of the uri and the media */
+ priv->path = g_strndup (path, path_len);
+ priv->media = media;
+ } else {
+ /* we have seen this path before, used cached media */
+ media = priv->media;
+ ctx->media = media;
+ GST_INFO ("reusing cached media %p for path %s", media, priv->path);
+ }
+
+ g_object_unref (factory);
+ ctx->factory = NULL;
+
+ if (media)
+ g_object_ref (media);
+
+ return media;
+
+ /* ERRORS */
+no_factory:
+ {
+ GST_ERROR ("client %p: no factory for path %s", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return NULL;
+ }
+no_factory_access:
+ {
+ g_object_unref (factory);
+ ctx->factory = NULL;
+ GST_ERROR ("client %p: not authorized to see factory path %s", client,
+ path);
+ /* error reply is already sent */
+ return NULL;
+ }
+not_authorized:
+ {
+ g_object_unref (factory);
+ ctx->factory = NULL;
+ GST_ERROR ("client %p: not authorized for factory path %s", client, path);
+ /* error reply is already sent */
+ return NULL;
+ }
+no_media:
+ {
+ GST_ERROR ("client %p: can't create media", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ g_object_unref (factory);
+ ctx->factory = NULL;
+ return NULL;
+ }
+no_thread:
+ {
+ GST_ERROR ("client %p: can't create thread", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_object_unref (media);
+ ctx->media = NULL;
+ g_object_unref (factory);
+ ctx->factory = NULL;
+ return NULL;
+ }
+no_prepare:
+ {
+ GST_ERROR ("client %p: can't prepare media", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_object_unref (media);
+ ctx->media = NULL;
+ g_object_unref (factory);
+ ctx->factory = NULL;
+ return NULL;
+ }
+}
+
+static inline DataSeq *
+get_data_seq_element (GstRTSPClient * client, guint8 channel)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GArray *data_seqs = priv->data_seqs;
+ gint i = 0;
+
+ while (i < data_seqs->len) {
+ DataSeq *data_seq = &g_array_index (data_seqs, DataSeq, i);
+ if (data_seq->channel == channel)
+ return data_seq;
+ i++;
+ }
+
+ return NULL;
+}
+
+static void
+add_data_seq (GstRTSPClient * client, guint8 channel)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ DataSeq data_seq = {.channel = channel,.seq = 0 };
+
+ if (get_data_seq_element (client, channel) == NULL)
+ g_array_append_val (priv->data_seqs, data_seq);
+}
+
+static void
+set_data_seq (GstRTSPClient * client, guint8 channel, guint seq)
+{
+ DataSeq *data_seq;
+
+ data_seq = get_data_seq_element (client, channel);
+ g_assert_nonnull (data_seq);
+ data_seq->seq = seq;
+}
+
+static guint
+get_data_seq (GstRTSPClient * client, guint8 channel)
+{
+ DataSeq *data_seq;
+
+ data_seq = get_data_seq_element (client, channel);
+ g_assert_nonnull (data_seq);
+ return data_seq->seq;
+}
+
+static gboolean
+get_data_channel (GstRTSPClient * client, guint seq, guint8 * channel)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GArray *data_seqs = priv->data_seqs;
+ gint i = 0;
+
+ while (i < data_seqs->len) {
+ DataSeq *data_seq = &g_array_index (data_seqs, DataSeq, i);
+ if (data_seq->seq == seq) {
+ *channel = data_seq->channel;
+ return TRUE;
+ }
+ i++;
+ }
+
+ return FALSE;
+}
+
+static gboolean
+do_close (gpointer user_data)
+{
+ GstRTSPClient *client = user_data;
+
+ gst_rtsp_client_close (client);
+
+ return G_SOURCE_REMOVE;
+}
+
+static gboolean
+do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPMessage message = { 0 };
+ gboolean ret = TRUE;
+
+ gst_rtsp_message_init_data (&message, channel);
+
+ gst_rtsp_message_set_body_buffer (&message, buffer);
+
+ g_mutex_lock (&priv->send_lock);
+ if (get_data_seq (client, channel) != 0) {
+ GST_WARNING ("already a queued data message for channel %d", channel);
+ g_mutex_unlock (&priv->send_lock);
+ return FALSE;
+ }
+ if (priv->send_messages_func) {
+ ret =
+ priv->send_messages_func (client, &message, 1, FALSE, priv->send_data);
+ } else if (priv->send_func) {
+ ret = priv->send_func (client, &message, FALSE, priv->send_data);
+ }
+ g_mutex_unlock (&priv->send_lock);
+
+ gst_rtsp_message_unset (&message);
+
+ if (!ret) {
+ GSource *idle_src;
+
+ /* close in watch context */
+ idle_src = g_idle_source_new ();
+ g_source_set_callback (idle_src, do_close, client, NULL);
+ g_source_attach (idle_src, priv->watch_context);
+ g_source_unref (idle_src);
+ }
+
+ return ret;
+}
+
+static gboolean
+do_check_back_pressure (guint8 channel, GstRTSPClient * client)
+{
+ return get_data_seq (client, channel) != 0;
+}
+
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
+ GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ gboolean ret = TRUE;
+ guint i, n = gst_buffer_list_length (buffer_list);
+ GstRTSPMessage *messages;
+
+ g_mutex_lock (&priv->send_lock);
+ if (get_data_seq (client, channel) != 0) {
+ GST_WARNING ("already a queued data message for channel %d", channel);
+ g_mutex_unlock (&priv->send_lock);
+ return FALSE;
+ }
+
+ messages = g_newa (GstRTSPMessage, n);
+ memset (messages, 0, sizeof (GstRTSPMessage) * n);
+ for (i = 0; i < n; i++) {
+ GstBuffer *buffer = gst_buffer_list_get (buffer_list, i);
+ gst_rtsp_message_init_data (&messages[i], channel);
+ gst_rtsp_message_set_body_buffer (&messages[i], buffer);
+ }
+
+ if (priv->send_messages_func) {
+ ret =
+ priv->send_messages_func (client, messages, n, FALSE, priv->send_data);
+ } else if (priv->send_func) {
+ for (i = 0; i < n; i++) {
+ ret = priv->send_func (client, &messages[i], FALSE, priv->send_data);
+ if (!ret)
+ break;
+ }
+ }
+ g_mutex_unlock (&priv->send_lock);
+
+ for (i = 0; i < n; i++) {
+ gst_rtsp_message_unset (&messages[i]);
+ }
+
+ if (!ret) {
+ GSource *idle_src;
+
+ /* close in watch context */
+ idle_src = g_idle_source_new ();
+ g_source_set_callback (idle_src, do_close, client, NULL);
+ g_source_attach (idle_src, priv->watch_context);
+ g_source_unref (idle_src);
+ }
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_client_close:
+ * @client: a #GstRTSPClient
+ *
+ * Close the connection of @client and remove all media it was managing.
+ *
+ * Since: 1.4
+ */
+void
+gst_rtsp_client_close (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ const gchar *tunnelid;
+
+ GST_DEBUG ("client %p: closing connection", client);
+
+ g_mutex_lock (&priv->watch_lock);
+
+ /* Work around the lack of thread safety of gst_rtsp_connection_close */
+ if (priv->watch) {
+ gst_rtsp_watch_set_flushing (priv->watch, TRUE);
+ }
+
+ if (priv->connection) {
+ if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
+ g_mutex_lock (&tunnels_lock);
+ /* remove from tunnelids */
+ g_hash_table_remove (tunnels, tunnelid);
+ g_mutex_unlock (&tunnels_lock);
+ }
+ gst_rtsp_connection_flush (priv->connection, TRUE);
+ gst_rtsp_connection_close (priv->connection);
+ }
+
+ if (priv->watch) {
+ GST_DEBUG ("client %p: destroying watch", client);
+ g_source_destroy ((GSource *) priv->watch);
+ priv->watch = NULL;
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
+ rtsp_ctrl_timeout_remove (client);
+ }
+
+ g_mutex_unlock (&priv->watch_lock);
+}
+
+static gchar *
+default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
+{
+ gchar *path;
+
+ if (uri->query) {
+ path = g_strconcat (uri->abspath, "?", uri->query, NULL);
+ } else {
+ /* normalize rtsp://<IP>:<PORT> to rtsp://<IP>:<PORT>/ */
+ path = g_strdup (uri->abspath[0] ? uri->abspath : "/");
+ }
+
+ return path;
+}
+
+/* Default signal handler function for all "pre-command" signals, like
+ * pre-options-request. It just returns the RTSP return code 200.
+ * Subclasses can override this to get another default behaviour.
+ */
+static GstRTSPStatusCode
+default_pre_signal_handler (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GST_LOG_OBJECT (client, "returning GST_RTSP_STS_OK");
+ return GST_RTSP_STS_OK;
+}
+
+/* The pre-signal accumulator function checks the return value of the signal
+ * handlers. If any of them returns an RTSP status code that does not start
+ * with 2 it will return FALSE, no more signal handlers will be called, and
+ * this last RTSP status code will be the result of the signal emission.
+ */
+static gboolean
+pre_signal_accumulator (GSignalInvocationHint * ihint, GValue * return_accu,
+ const GValue * handler_return, gpointer data)
+{
+ GstRTSPStatusCode handler_value = g_value_get_enum (handler_return);
+ GstRTSPStatusCode accumulated_value = g_value_get_enum (return_accu);
+
+ if (handler_value < 200 || handler_value > 299) {
+ GST_DEBUG ("handler_value : %d, returning FALSE", handler_value);
+ g_value_set_enum (return_accu, handler_value);
+ return FALSE;
+ }
+
+ /* the accumulated value is initiated to 0 by GLib. if current handler value is
+ * bigger then use that instead
+ *
+ * FIXME: Should we prioritize the 2xx codes in a smarter way?
+ * Like, "201 Created" > "250 Low On Storage Space" > "200 OK"?
+ */
+ if (handler_value > accumulated_value)
+ g_value_set_enum (return_accu, handler_value);
+
+ return TRUE;
+}
+
+/* The cleanup_transports function is called from handle_teardown_request() to
+ * remove any stream transports from the newly closed session that were added to
+ * priv->transports in handle_setup_request(). This is done to avoid forwarding
+ * data from the client on a channel that we just closed.
+ */
+static void
+cleanup_transports (GstRTSPClient * client, GPtrArray * transports)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPStreamTransport *stream_transport;
+ const GstRTSPTransport *rtsp_transport;
+ guint i;
+
+ GST_LOG_OBJECT (client, "potentially removing %u transports",
+ transports->len);
+
+ for (i = 0; i < transports->len; i++) {
+ stream_transport = g_ptr_array_index (transports, i);
+ if (stream_transport == NULL) {
+ GST_LOG_OBJECT (client, "stream transport %u is NULL, continue", i);
+ continue;
+ }
+
+ rtsp_transport = gst_rtsp_stream_transport_get_transport (stream_transport);
+ if (rtsp_transport == NULL) {
+ GST_LOG_OBJECT (client, "RTSP transport %u is NULL, continue", i);
+ continue;
+ }
+
+ /* priv->transport only stores transports where RTP is tunneled over RTSP */
+ if (rtsp_transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+ if (!g_hash_table_remove (priv->transports,
+ GINT_TO_POINTER (rtsp_transport->interleaved.min))) {
+ GST_WARNING_OBJECT (client,
+ "failed removing transport with key '%d' from priv->transports",
+ rtsp_transport->interleaved.min);
+ }
+ if (!g_hash_table_remove (priv->transports,
+ GINT_TO_POINTER (rtsp_transport->interleaved.max))) {
+ GST_WARNING_OBJECT (client,
+ "failed removing transport with key '%d' from priv->transports",
+ rtsp_transport->interleaved.max);
+ }
+ } else {
+ GST_LOG_OBJECT (client, "transport %u not RTP/RTSP, skip it", i);
+ }
+ }
+}
+
+static gboolean
+handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPClientClass *klass;
+ GstRTSPSession *session;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
+ GstRTSPStatusCode code;
+ gchar *path;
+ gint matched;
+ gboolean keep_session;
+ GstRTSPStatusCode sig_result;
+ GPtrArray *session_media_transports;
+
+ if (!ctx->session)
+ goto no_session;
+
+ session = ctx->session;
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, ctx->uri);
+
+ /* get a handle to the configuration of the media in the session */
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
+ goto not_found;
+
+ /* only aggregate control for now.. */
+ if (path[matched] != '\0')
+ goto no_aggregate;
+
+ g_free (path);
+
+ ctx->sessmedia = sessmedia;
+
+ media = gst_rtsp_session_media_get_media (sessmedia);
+ g_object_ref (media);
+ gst_rtsp_media_lock (media);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_TEARDOWN_REQUEST],
+ 0, ctx, &sig_result);
+ if (sig_result != GST_RTSP_STS_OK) {
+ goto sig_failed;
+ }
+
+ /* get a reference to the transports in the session media so we can clean up
+ * our priv->transports before returning */
+ session_media_transports = gst_rtsp_session_media_get_transports (sessmedia);
+
+ /* we emit the signal before closing the connection */
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
+ 0, ctx);
+
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
+
+ /* unmanage the media in the session, returns false if all media session
+ * are torn down. */
+ keep_session = gst_rtsp_session_release_media (session, sessmedia);
+
+ /* construct the response now */
+ code = GST_RTSP_STS_OK;
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ send_message (client, ctx, ctx->response, TRUE);
+
+ if (!keep_session) {
+ /* remove the session */
+ gst_rtsp_session_pool_remove (priv->session_pool, session);
+ }
+
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+
+ /* remove all transports that were present in the session media which we just
+ * unmanaged from the priv->transports array, so we do not try to handle data
+ * on channels that were just closed */
+ cleanup_transports (client, session_media_transports);
+ g_ptr_array_unref (session_media_transports);
+
+ return TRUE;
+
+ /* ERRORS */
+no_session:
+ {
+ GST_ERROR ("client %p: no session", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri supplied", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+not_found:
+ {
+ GST_ERROR ("client %p: no media for uri", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
+ return FALSE;
+ }
+no_aggregate:
+ {
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
+ return FALSE;
+ }
+sig_failed:
+ {
+ GST_ERROR ("client %p: pre signal returned error: %s", client,
+ gst_rtsp_status_as_text (sig_result));
+ send_generic_response (client, sig_result, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+}
+
+static GstRTSPResult
+default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res;
+
+ res = gst_rtsp_params_set (client, ctx);
+
+ return res;
+}
+
+static GstRTSPResult
+default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res;
+
+ res = gst_rtsp_params_get (client, ctx);
+
+ return res;
+}
+
+static gboolean
+handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res;
+ guint8 *data;
+ guint size;
+ GstRTSPStatusCode sig_result;
+
+ g_signal_emit (client,
+ gst_rtsp_client_signals[SIGNAL_PRE_GET_PARAMETER_REQUEST], 0, ctx,
+ &sig_result);
+ if (sig_result != GST_RTSP_STS_OK) {
+ goto sig_failed;
+ }
+
+ res = gst_rtsp_message_get_body (ctx->request, &data, &size);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ if (size == 0 || !data || strlen ((char *) data) == 0) {
+ if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0) {
+ GST_ERROR_OBJECT (client, "Using PLAY request for keep-alive is forbidden"
+ " in RTSP 2.0");
+ goto bad_request;
+ }
+
+ /* no body (or only '\0'), keep-alive request */
+ send_generic_response (client, GST_RTSP_STS_OK, ctx);
+ } else {
+ /* there is a body, handle the params */
+ res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ send_message (client, ctx, ctx->response, FALSE);
+ }
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
+ 0, ctx);
+
+ return TRUE;
+
+ /* ERRORS */
+sig_failed:
+ {
+ GST_ERROR ("client %p: pre signal returned error: %s", client,
+ gst_rtsp_status_as_text (sig_result));
+ send_generic_response (client, sig_result, ctx);
+ return FALSE;
+ }
+bad_request:
+ {
+ GST_ERROR ("client %p: bad request", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res;
+ guint8 *data;
+ guint size;
+ GstRTSPStatusCode sig_result;
+
+ g_signal_emit (client,
+ gst_rtsp_client_signals[SIGNAL_PRE_SET_PARAMETER_REQUEST], 0, ctx,
+ &sig_result);
+ if (sig_result != GST_RTSP_STS_OK) {
+ goto sig_failed;
+ }
+
+ res = gst_rtsp_message_get_body (ctx->request, &data, &size);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ if (size == 0 || !data || strlen ((char *) data) == 0) {
+ /* no body (or only '\0'), keep-alive request */
+ send_generic_response (client, GST_RTSP_STS_OK, ctx);
+ } else {
+ /* there is a body, handle the params */
+ res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ send_message (client, ctx, ctx->response, FALSE);
+ }
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
+ 0, ctx);
+
+ return TRUE;
+
+ /* ERRORS */
+sig_failed:
+ {
+ GST_ERROR ("client %p: pre signal returned error: %s", client,
+ gst_rtsp_status_as_text (sig_result));
+ send_generic_response (client, sig_result, ctx);
+ return FALSE;
+ }
+bad_request:
+ {
+ GST_ERROR ("client %p: bad request", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPSession *session;
+ GstRTSPClientClass *klass;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
+ GstRTSPStatusCode code;
+ GstRTSPState rtspstate;
+ gchar *path;
+ gint matched;
+ GstRTSPStatusCode sig_result;
+ guint i, n;
+
+ if (!(session = ctx->session))
+ goto no_session;
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, ctx->uri);
+
+ /* get a handle to the configuration of the media in the session */
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
+ goto not_found;
+
+ if (path[matched] != '\0')
+ goto no_aggregate;
+
+ g_free (path);
+
+ media = gst_rtsp_session_media_get_media (sessmedia);
+ g_object_ref (media);
+ gst_rtsp_media_lock (media);
+ n = gst_rtsp_media_n_streams (media);
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
+
+ if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
+ GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET)
+ goto not_supported;
+ }
+
+ ctx->sessmedia = sessmedia;
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_PAUSE_REQUEST], 0,
+ ctx, &sig_result);
+ if (sig_result != GST_RTSP_STS_OK) {
+ goto sig_failed;
+ }
+
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ /* the session state must be playing or recording */
+ if (rtspstate != GST_RTSP_STATE_PLAYING &&
+ rtspstate != GST_RTSP_STATE_RECORDING)
+ goto invalid_state;
+
+ /* then pause sending */
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
+
+ /* construct the response now */
+ code = GST_RTSP_STS_OK;
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ /* the state is now READY */
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
+
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+
+ return TRUE;
+
+ /* ERRORS */
+no_session:
+ {
+ GST_ERROR ("client %p: no session", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri supplied", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+not_found:
+ {
+ GST_ERROR ("client %p: no media for uri", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
+ return FALSE;
+ }
+no_aggregate:
+ {
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
+ return FALSE;
+ }
+sig_failed:
+ {
+ GST_ERROR ("client %p: pre signal returned error: %s", client,
+ gst_rtsp_status_as_text (sig_result));
+ send_generic_response (client, sig_result, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+invalid_state:
+ {
+ GST_ERROR ("client %p: not PLAYING or RECORDING", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+not_supported:
+ {
+ GST_ERROR ("client %p: pausing not supported", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+}
+
+/* convert @url and @path to a URL used as a content base for the factory
+ * located at @path */
+static gchar *
+make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
+{
+ GstRTSPUrl tmp;
+ gchar *result;
+ const gchar *trail;
+
+ /* check for trailing '/' and append one */
+ trail = (path[strlen (path) - 1] != '/' ? "/" : "");
+
+ tmp = *url;
+ tmp.user = NULL;
+ tmp.passwd = NULL;
+ tmp.abspath = g_strdup_printf ("%s%s", path, trail);
+ tmp.query = NULL;
+ result = gst_rtsp_url_get_request_uri (&tmp);
+ g_free (tmp.abspath);
+
+ return result;
+}
+
+/* Check if the given header of type double is present and, if so,
+ * put it's value in the supplied variable.
+ */
+static GstRTSPStatusCode
+parse_header_value_double (GstRTSPClient * client, GstRTSPContext * ctx,
+ GstRTSPHeaderField header, gboolean * present, gdouble * value)
+{
+ GstRTSPResult res;
+ gchar *str;
+ gchar *end;
+
+ res = gst_rtsp_message_get_header (ctx->request, header, &str, 0);
+ if (res == GST_RTSP_OK) {
+ *value = g_ascii_strtod (str, &end);
+ if (end == str)
+ goto parse_header_failed;
+
+ GST_DEBUG ("client %p: got '%s', value %f", client,
+ gst_rtsp_header_as_text (header), *value);
+ *present = TRUE;
+ } else {
+ *present = FALSE;
+ }
+
+ return GST_RTSP_STS_OK;
+
+parse_header_failed:
+ {
+ GST_ERROR ("client %p: failed parsing '%s' header", client,
+ gst_rtsp_header_as_text (header));
+ return GST_RTSP_STS_BAD_REQUEST;
+ }
+}
+
+/* Parse scale and speed headers, if present, and set the rate to
+ * (rate * scale * speed) */
+static GstRTSPStatusCode
+parse_scale_and_speed (GstRTSPClient * client, GstRTSPContext * ctx,
+ gboolean * scale_present, gboolean * speed_present, gdouble * rate,
+ GstSeekFlags * flags)
+{
+ gdouble scale = 1.0;
+ gdouble speed = 1.0;
+ GstRTSPStatusCode status;
+
+ GST_DEBUG ("got rate %f", *rate);
+
+ status = parse_header_value_double (client, ctx, GST_RTSP_HDR_SCALE,
+ scale_present, &scale);
+ if (status != GST_RTSP_STS_OK)
+ return status;
+
+ if (*scale_present) {
+ GST_DEBUG ("got Scale %f", scale);
+ if (scale == 0)
+ goto bad_scale_value;
+ *rate *= scale;
+
+ if (ABS (scale) != 1.0)
+ *flags |= GST_SEEK_FLAG_TRICKMODE;
+ }
+
+ GST_DEBUG ("rate after parsing Scale %f", *rate);
+
+ status = parse_header_value_double (client, ctx, GST_RTSP_HDR_SPEED,
+ speed_present, &speed);
+ if (status != GST_RTSP_STS_OK)
+ return status;
+
+ if (*speed_present) {
+ GST_DEBUG ("got Speed %f", speed);
+ if (speed <= 0)
+ goto bad_speed_value;
+ *rate *= speed;
+ }
+
+ GST_DEBUG ("rate after parsing Speed %f", *rate);
+
+ return status;
+
+bad_scale_value:
+ {
+ GST_ERROR ("client %p: bad 'Scale' header value (%f)", client, scale);
+ return GST_RTSP_STS_BAD_REQUEST;
+ }
+bad_speed_value:
+ {
+ GST_ERROR ("client %p: bad 'Speed' header value (%f)", client, speed);
+ return GST_RTSP_STS_BAD_REQUEST;
+ }
+}
+
+static GstRTSPStatusCode
+setup_play_mode (GstRTSPClient * client, GstRTSPContext * ctx,
+ GstRTSPRangeUnit * unit, gboolean * scale_present, gboolean * speed_present)
+{
+ gchar *str;
+ GstRTSPResult res;
+ GstRTSPTimeRange *range = NULL;
+ gdouble rate = 1.0;
+ GstSeekFlags flags = GST_SEEK_FLAG_NONE;
+ GstRTSPClientClass *klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ GstRTSPStatusCode rtsp_status_code;
+ GstClockTime trickmode_interval = 0;
+ gboolean enable_rate_control = TRUE;
+
+ /* parse the range header if we have one */
+ res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
+ if (res == GST_RTSP_OK) {
+ gchar *seek_style = NULL;
+
+ res = gst_rtsp_range_parse (str, &range);
+ if (res != GST_RTSP_OK)
+ goto parse_range_failed;
+
+ *unit = range->unit;
+
+ /* parse seek style header, if present */
+ res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_SEEK_STYLE,
+ &seek_style, 0);
+
+ if (res == GST_RTSP_OK) {
+ if (g_strcmp0 (seek_style, "RAP") == 0)
+ flags = GST_SEEK_FLAG_ACCURATE;
+ else if (g_strcmp0 (seek_style, "CoRAP") == 0)
+ flags = GST_SEEK_FLAG_KEY_UNIT;
+ else if (g_strcmp0 (seek_style, "First-Prior") == 0)
+ flags = GST_SEEK_FLAG_KEY_UNIT & GST_SEEK_FLAG_SNAP_BEFORE;
+ else if (g_strcmp0 (seek_style, "Next") == 0)
+ flags = GST_SEEK_FLAG_KEY_UNIT & GST_SEEK_FLAG_SNAP_AFTER;
+ else
+ GST_FIXME_OBJECT (client, "Add support for seek style %s", seek_style);
+ } else if (range->min.type == GST_RTSP_TIME_END) {
+ flags = GST_SEEK_FLAG_ACCURATE;
+ } else {
+ flags = GST_SEEK_FLAG_KEY_UNIT;
+ }
+
+ if (seek_style)
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_SEEK_STYLE,
+ seek_style);
+ } else {
+ flags = GST_SEEK_FLAG_ACCURATE;
+ }
+
+ /* check for scale and/or speed headers
+ * we will set the seek rate to (speed * scale) and let the media decide
+ * the resulting scale and speed. in the response we will use rate and applied
+ * rate from the resulting segment as values for the speed and scale headers
+ * respectively */
+ rtsp_status_code = parse_scale_and_speed (client, ctx, scale_present,
+ speed_present, &rate, &flags);
+ if (rtsp_status_code != GST_RTSP_STS_OK)
+ goto scale_speed_failed;
+
+ /* give the application a chance to tweak range, flags, or rate */
+ if (klass->adjust_play_mode != NULL) {
+ rtsp_status_code =
+ klass->adjust_play_mode (client, ctx, &range, &flags, &rate,
+ &trickmode_interval, &enable_rate_control);
+ if (rtsp_status_code != GST_RTSP_STS_OK)
+ goto adjust_play_mode_failed;
+ }
+
+ gst_rtsp_media_set_rate_control (ctx->media, enable_rate_control);
+
+ /* now do the seek with the seek options */
+ gst_rtsp_media_seek_trickmode (ctx->media, range, flags, rate,
+ trickmode_interval);
+ if (range != NULL)
+ gst_rtsp_range_free (range);
+
+ if (gst_rtsp_media_get_status (ctx->media) == GST_RTSP_MEDIA_STATUS_ERROR)
+ goto seek_failed;
+
+ return GST_RTSP_STS_OK;
+
+parse_range_failed:
+ {
+ GST_ERROR ("client %p: failed parsing range header", client);
+ return GST_RTSP_STS_BAD_REQUEST;
+ }
+scale_speed_failed:
+ {
+ if (range != NULL)
+ gst_rtsp_range_free (range);
+ GST_ERROR ("client %p: failed parsing Scale or Speed headers", client);
+ return rtsp_status_code;
+ }
+adjust_play_mode_failed:
+ {
+ GST_ERROR ("client %p: sub class returned bad code (%d)", client,
+ rtsp_status_code);
+ if (range != NULL)
+ gst_rtsp_range_free (range);
+ return rtsp_status_code;
+ }
+seek_failed:
+ {
+ GST_ERROR ("client %p: seek failed", client);
+ return GST_RTSP_STS_SERVICE_UNAVAILABLE;
+ }
+}
+
+static gboolean
+handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPSession *session;
+ GstRTSPClientClass *klass;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
+ GstRTSPStatusCode code;
+ GstRTSPUrl *uri;
+ gchar *str;
+ GstRTSPState rtspstate;
+ GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
+ gchar *path, *rtpinfo = NULL;
+ gint matched;
+ GstRTSPStatusCode sig_result;
+ GPtrArray *transports;
+ gboolean scale_present;
+ gboolean speed_present;
+ gdouble rate;
+ gdouble applied_rate;
+
+ if (!(session = ctx->session))
+ goto no_session;
+
+ if (!(uri = ctx->uri))
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
+
+ /* get a handle to the configuration of the media in the session */
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
+ goto not_found;
+
+ if (path[matched] != '\0')
+ goto no_aggregate;
+
+ g_free (path);
+
+ ctx->sessmedia = sessmedia;
+ ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
+
+ g_object_ref (media);
+ gst_rtsp_media_lock (media);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_PLAY_REQUEST], 0,
+ ctx, &sig_result);
+ if (sig_result != GST_RTSP_STS_OK) {
+ goto sig_failed;
+ }
+
+ if (!(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_PLAY))
+ goto unsupported_mode;
+
+ /* the session state must be playing or ready */
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
+ goto invalid_state;
+
+ /* update the pipeline */
+ transports = gst_rtsp_session_media_get_transports (sessmedia);
+ if (!gst_rtsp_media_complete_pipeline (media, transports)) {
+ g_ptr_array_unref (transports);
+ goto pipeline_error;
+ }
+ g_ptr_array_unref (transports);
+
+ /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
+ if (!gst_rtsp_media_unsuspend (media))
+ goto unsuspend_failed;
+
+ code = setup_play_mode (client, ctx, &unit, &scale_present, &speed_present);
+ if (code != GST_RTSP_STS_OK)
+ goto invalid_mode;
+
+ /* grab RTPInfo from the media now */
+ if (gst_rtsp_media_has_completed_sender (media) &&
+ !(rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia)))
+ goto rtp_info_error;
+
+ /* construct the response now */
+ code = GST_RTSP_STS_OK;
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ /* add the RTP-Info header */
+ if (rtpinfo)
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
+ rtpinfo);
+
+ /* add the range */
+ str = gst_rtsp_media_get_range_string (media, TRUE, unit);
+ if (str)
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
+
+ if (gst_rtsp_media_has_completed_sender (media)) {
+ /* the scale and speed headers must always be added if they were present in
+ * the request. however, even if they were not, we still add them if
+ * applied_rate or rate deviate from the "normal", i.e. 1.0 */
+ if (!gst_rtsp_media_get_rates (media, &rate, &applied_rate))
+ goto get_rates_error;
+ g_assert (rate != 0 && applied_rate != 0);
+
+ if (scale_present || applied_rate != 1.0)
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_SCALE,
+ g_strdup_printf ("%1.3f", applied_rate));
+
+ if (speed_present || rate != 1.0)
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_SPEED,
+ g_strdup_printf ("%1.3f", rate));
+ }
+
+ if (klass->adjust_play_response) {
+ code = klass->adjust_play_response (client, ctx);
+ if (code != GST_RTSP_STS_OK)
+ goto adjust_play_response_failed;
+ }
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ /* start playing after sending the response */
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
+
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
+
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+
+ return TRUE;
+
+ /* ERRORS */
+no_session:
+ {
+ GST_ERROR ("client %p: no session", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri supplied", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+not_found:
+ {
+ GST_ERROR ("client %p: media not found", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_aggregate:
+ {
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
+ return FALSE;
+ }
+sig_failed:
+ {
+ GST_ERROR ("client %p: pre signal returned error: %s", client,
+ gst_rtsp_status_as_text (sig_result));
+ send_generic_response (client, sig_result, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+invalid_state:
+ {
+ GST_ERROR ("client %p: not PLAYING or READY", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+pipeline_error:
+ {
+ GST_ERROR ("client %p: failed to configure the pipeline", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+unsuspend_failed:
+ {
+ GST_ERROR ("client %p: unsuspend failed", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+invalid_mode:
+ {
+ GST_ERROR ("client %p: seek failed", client);
+ send_generic_response (client, code, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+unsupported_mode:
+ {
+ GST_ERROR ("client %p: media does not support PLAY", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+get_rates_error:
+ {
+ GST_ERROR ("client %p: failed obtaining rate and applied_rate", client);
+ send_generic_response (client, GST_RTSP_STS_INTERNAL_SERVER_ERROR, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+adjust_play_response_failed:
+ {
+ GST_ERROR ("client %p: failed to adjust play response", client);
+ send_generic_response (client, code, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+rtp_info_error:
+ {
+ GST_ERROR ("client %p: failed to add RTP-Info", client);
+ send_generic_response (client, GST_RTSP_STS_INTERNAL_SERVER_ERROR, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+}
+
+static void
+do_keepalive (GstRTSPSession * session)
+{
+ GST_INFO ("keep session %p alive", session);
+ gst_rtsp_session_touch (session);
+}
+
+/* parse @transport and return a valid transport in @tr. only transports
+ * supported by @stream are returned. Returns FALSE if no valid transport
+ * was found. */
+static gboolean
+parse_transport (const char *transport, GstRTSPStream * stream,
+ GstRTSPTransport * tr)
+{
+ gint i;
+ gboolean res;
+ gchar **transports;
+
+ res = FALSE;
+ gst_rtsp_transport_init (tr);
+
+ GST_DEBUG ("parsing transports %s", transport);
+
+ transports = g_strsplit (transport, ",", 0);
+
+ /* loop through the transports, try to parse */
+ for (i = 0; transports[i]; i++) {
+ g_strstrip (transports[i]);
+ res = gst_rtsp_transport_parse (transports[i], tr);
+ if (res != GST_RTSP_OK) {
+ /* no valid transport, search some more */
+ GST_WARNING ("could not parse transport %s", transports[i]);
+ goto next;
+ }
+
+ /* we have a transport, see if it's supported */
+ if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
+ GST_WARNING ("unsupported transport %s", transports[i]);
+ goto next;
+ }
+
+ /* we have a valid transport */
+ GST_INFO ("found valid transport %s", transports[i]);
+ res = TRUE;
+ break;
+
+ next:
+ gst_rtsp_transport_init (tr);
+ }
+ g_strfreev (transports);
+
+ return res;
+}
+
+static gboolean
+default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+ GstRTSPStream * stream, GstRTSPContext * ctx)
+{
+ GstRTSPMessage *request = ctx->request;
+ gchar *blocksize_str;
+
+ if (!gst_rtsp_stream_is_sender (stream))
+ return TRUE;
+
+ if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
+ &blocksize_str, 0) == GST_RTSP_OK) {
+ guint64 blocksize;
+ gchar *end;
+
+ blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
+ if (end == blocksize_str)
+ goto parse_failed;
+
+ /* we don't want to change the mtu when this media
+ * can be shared because it impacts other clients */
+ if (gst_rtsp_media_is_shared (media))
+ goto done;
+
+ if (blocksize > G_MAXUINT)
+ blocksize = G_MAXUINT;
+
+ gst_rtsp_stream_set_mtu (stream, blocksize);
+ }
+done:
+ return TRUE;
+
+ /* ERRORS */
+parse_failed:
+ {
+ GST_ERROR_OBJECT (client, "failed to parse blocksize");
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+}
+
+static gboolean
+default_configure_client_transport (GstRTSPClient * client,
+ GstRTSPContext * ctx, GstRTSPTransport * ct)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ /* we have a valid transport now, set the destination of the client. */
+ if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST ||
+ ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP) {
+ /* allocate UDP ports */
+ GSocketFamily family;
+ gboolean use_client_settings = FALSE;
+
+ family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
+
+ if ((ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) &&
+ gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS) &&
+ (ct->destination != NULL)) {
+
+ if (!gst_rtsp_stream_verify_mcast_ttl (ctx->stream, ct->ttl))
+ goto error_ttl;
+
+ use_client_settings = TRUE;
+ }
+
+ /* We need to allocate the sockets for both families before starting
+ * multiudpsink, otherwise multiudpsink won't accept new clients with
+ * a different family.
+ */
+ /* FIXME: could be more adequately solved by making it possible
+ * to set a socket on multiudpsink after it has already been started */
+ if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream,
+ G_SOCKET_FAMILY_IPV4, ct, use_client_settings)
+ && family == G_SOCKET_FAMILY_IPV4)
+ goto error_allocating_ports;
+
+ if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream,
+ G_SOCKET_FAMILY_IPV6, ct, use_client_settings)
+ && family == G_SOCKET_FAMILY_IPV6)
+ goto error_allocating_ports;
+
+ if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ if (use_client_settings) {
+ /* FIXME: the address has been successfully allocated, however, in
+ * the use_client_settings case we need to verify that the allocated
+ * address is the one requested by the client and if this address is
+ * an allowed destination. Verifying this via the address pool in not
+ * the proper way as the address pool should only be used for choosing
+ * the server-selected address/port pairs. */
+ GSocket *rtp_socket;
+ guint ttl;
+
+ rtp_socket =
+ gst_rtsp_stream_get_rtp_multicast_socket (ctx->stream, family);
+ if (rtp_socket == NULL)
+ goto no_socket;
+ ttl = g_socket_get_multicast_ttl (rtp_socket);
+ g_object_unref (rtp_socket);
+ if (ct->ttl < ttl) {
+ /* use the maximum ttl that is requested by multicast clients */
+ GST_DEBUG ("requested ttl %u, but keeping ttl %u", ct->ttl, ttl);
+ ct->ttl = ttl;
+ }
+
+ } else {
+ GstRTSPAddress *addr = NULL;
+
+ g_free (ct->destination);
+ addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
+ if (addr == NULL)
+ goto no_address;
+ ct->destination = g_strdup (addr->address);
+ ct->port.min = addr->port;
+ ct->port.max = addr->port + addr->n_ports - 1;
+ ct->ttl = addr->ttl;
+ gst_rtsp_address_free (addr);
+ }
+
+ if (!gst_rtsp_stream_add_multicast_client_address (ctx->stream,
+ ct->destination, ct->port.min, ct->port.max, family))
+ goto error_mcast_transport;
+
+ } else {
+ GstRTSPUrl *url;
+
+ url = gst_rtsp_connection_get_url (priv->connection);
+ g_free (ct->destination);
+ ct->destination = g_strdup (url->host);
+ }
+ } else {
+ GstRTSPUrl *url;
+
+ url = gst_rtsp_connection_get_url (priv->connection);
+ g_free (ct->destination);
+ ct->destination = g_strdup (url->host);
+
+ if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
+ GSocket *sock;
+ GSocketAddress *addr;
+
+ sock = gst_rtsp_connection_get_read_socket (priv->connection);
+ if ((addr = g_socket_get_remote_address (sock, NULL))) {
+ /* our read port is the sender port of client */
+ ct->client_port.min =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ if ((addr = g_socket_get_local_address (sock, NULL))) {
+ ct->server_port.max =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ sock = gst_rtsp_connection_get_write_socket (priv->connection);
+ if ((addr = g_socket_get_remote_address (sock, NULL))) {
+ /* our write port is the receiver port of client */
+ ct->client_port.max =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ if ((addr = g_socket_get_local_address (sock, NULL))) {
+ ct->server_port.min =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ /* check if the client selected channels for TCP */
+ if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
+ gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
+ &ct->interleaved);
+ }
+ /* alloc new channels if they are already taken */
+ while (g_hash_table_contains (priv->transports,
+ GINT_TO_POINTER (ct->interleaved.min))
+ || g_hash_table_contains (priv->transports,
+ GINT_TO_POINTER (ct->interleaved.max))) {
+ gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
+ &ct->interleaved);
+ if (ct->interleaved.max > 255)
+ goto error_allocating_channels;
+ }
+ }
+ }
+ return TRUE;
+
+ /* ERRORS */
+error_ttl:
+ {
+ GST_ERROR_OBJECT (client,
+ "Failed to allocate UDP ports: invalid ttl value");
+ return FALSE;
+ }
+error_allocating_ports:
+ {
+ GST_ERROR_OBJECT (client, "Failed to allocate UDP ports");
+ return FALSE;
+ }
+no_address:
+ {
+ GST_ERROR_OBJECT (client, "Failed to acquire address for stream");
+ return FALSE;
+ }
+no_socket:
+ {
+ GST_ERROR_OBJECT (client, "Failed to get UDP socket");
+ return FALSE;
+ }
+error_mcast_transport:
+ {
+ GST_ERROR_OBJECT (client, "Failed to add multicast client transport");
+ return FALSE;
+ }
+error_allocating_channels:
+ {
+ GST_ERROR_OBJECT (client, "Failed to allocate interleaved channels");
+ return FALSE;
+ }
+}
+
+static GstRTSPTransport *
+make_server_transport (GstRTSPClient * client, GstRTSPMedia * media,
+ GstRTSPContext * ctx, GstRTSPTransport * ct)
+{
+ GstRTSPTransport *st;
+ GInetAddress *addr;
+ GSocketFamily family;
+
+ /* prepare the server transport */
+ gst_rtsp_transport_new (&st);
+
+ st->trans = ct->trans;
+ st->profile = ct->profile;
+ st->lower_transport = ct->lower_transport;
+ st->mode_play = ct->mode_play;
+ st->mode_record = ct->mode_record;
+
+ addr = g_inet_address_new_from_string (ct->destination);
+
+ if (!addr) {
+ GST_ERROR ("failed to get inet addr from client destination");
+ family = G_SOCKET_FAMILY_IPV4;
+ } else {
+ family = g_inet_address_get_family (addr);
+ g_object_unref (addr);
+ addr = NULL;
+ }
+
+ switch (st->lower_transport) {
+ case GST_RTSP_LOWER_TRANS_UDP:
+ st->client_port = ct->client_port;
+ gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
+ break;
+ case GST_RTSP_LOWER_TRANS_UDP_MCAST:
+ st->port = ct->port;
+ st->destination = g_strdup (ct->destination);
+ st->ttl = ct->ttl;
+ break;
+ case GST_RTSP_LOWER_TRANS_TCP:
+ st->interleaved = ct->interleaved;
+ st->client_port = ct->client_port;
+ st->server_port = ct->server_port;
+ default:
+ break;
+ }
+
+ if ((gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_PLAY))
+ gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
+
+ return st;
+}
+
+static void
+rtsp_ctrl_timeout_remove_unlocked (GstRTSPClientPrivate * priv)
+{
+ if (priv->rtsp_ctrl_timeout != NULL) {
+ GST_DEBUG ("rtsp control session removed timeout %p.",
+ priv->rtsp_ctrl_timeout);
+ g_source_destroy (priv->rtsp_ctrl_timeout);
+ g_source_unref (priv->rtsp_ctrl_timeout);
+ priv->rtsp_ctrl_timeout = NULL;
+ priv->rtsp_ctrl_timeout_cnt = 0;
+ }
+}
+
+static void
+rtsp_ctrl_timeout_remove (GstRTSPClient * client)
+{
+ g_mutex_lock (&client->priv->lock);
+ rtsp_ctrl_timeout_remove_unlocked (client->priv);
+ g_mutex_unlock (&client->priv->lock);
+}
+
+static void
+rtsp_ctrl_timeout_destroy_notify (gpointer user_data)
+{
+ GWeakRef *client_weak_ref = (GWeakRef *) user_data;
+
+ g_weak_ref_clear (client_weak_ref);
+ g_free (client_weak_ref);
+}
+
+static gboolean
+rtsp_ctrl_timeout_cb (gpointer user_data)
+{
+ gboolean res = G_SOURCE_CONTINUE;
+ GstRTSPClientPrivate *priv;
+ GWeakRef *client_weak_ref = (GWeakRef *) user_data;
+ GstRTSPClient *client = (GstRTSPClient *) g_weak_ref_get (client_weak_ref);
+
+ if (client == NULL) {
+ return G_SOURCE_REMOVE;
+ }
+
+ priv = client->priv;
+ g_mutex_lock (&priv->lock);
+ priv->rtsp_ctrl_timeout_cnt += RTSP_CTRL_CB_INTERVAL;
+
+ if ((priv->rtsp_ctrl_timeout_cnt > RTSP_CTRL_TIMEOUT_VALUE)
+ || (priv->had_session
+ && priv->rtsp_ctrl_timeout_cnt > priv->post_session_timeout)) {
+ GST_DEBUG ("rtsp control session timeout %p expired, closing client.",
+ priv->rtsp_ctrl_timeout);
+ rtsp_ctrl_timeout_remove_unlocked (client->priv);
+
+ res = G_SOURCE_REMOVE;
+ }
+
+ g_mutex_unlock (&priv->lock);
+
+ if (res == G_SOURCE_REMOVE) {
+ gst_rtsp_client_close (client);
+ }
+
+ g_object_unref (client);
+
+ return res;
+}
+
+static gchar *
+stream_make_keymgmt (GstRTSPClient * client, const gchar * location,
+ GstRTSPStream * stream)
+{
+ gchar *base64, *result = NULL;
+ GstMIKEYMessage *mikey_msg;
+ GstCaps *srtcpparams;
+ GstElement *rtcp_encoder;
+ gint srtcp_cipher, srtp_cipher;
+ gint srtcp_auth, srtp_auth;
+ GstBuffer *key;
+ GType ciphertype, authtype;
+ GEnumClass *cipher_enum, *auth_enum;
+ GEnumValue *srtcp_cipher_value, *srtp_cipher_value, *srtcp_auth_value,
+ *srtp_auth_value;
+
+ rtcp_encoder = gst_rtsp_stream_get_srtp_encoder (stream);
+
+ if (!rtcp_encoder)
+ goto done;
+
+ ciphertype = g_type_from_name ("GstSrtpCipherType");
+ authtype = g_type_from_name ("GstSrtpAuthType");
+
+ cipher_enum = g_type_class_ref (ciphertype);
+ auth_enum = g_type_class_ref (authtype);
+
+ /* We need to bring the encoder to READY so that it generates its key */
+ gst_element_set_state (rtcp_encoder, GST_STATE_READY);
+
+ g_object_get (rtcp_encoder, "rtcp-cipher", &srtcp_cipher, "rtcp-auth",
+ &srtcp_auth, "rtp-cipher", &srtp_cipher, "rtp-auth", &srtp_auth, "key",
+ &key, NULL);
+ g_object_unref (rtcp_encoder);
+
+ srtcp_cipher_value = g_enum_get_value (cipher_enum, srtcp_cipher);
+ srtp_cipher_value = g_enum_get_value (cipher_enum, srtp_cipher);
+ srtcp_auth_value = g_enum_get_value (auth_enum, srtcp_auth);
+ srtp_auth_value = g_enum_get_value (auth_enum, srtp_auth);
+
+ g_type_class_unref (cipher_enum);
+ g_type_class_unref (auth_enum);
+
+ srtcpparams = gst_caps_new_simple ("application/x-srtcp",
+ "srtcp-cipher", G_TYPE_STRING, srtcp_cipher_value->value_nick,
+ "srtcp-auth", G_TYPE_STRING, srtcp_auth_value->value_nick,
+ "srtp-cipher", G_TYPE_STRING, srtp_cipher_value->value_nick,
+ "srtp-auth", G_TYPE_STRING, srtp_auth_value->value_nick,
+ "srtp-key", GST_TYPE_BUFFER, key, NULL);
+
+ mikey_msg = gst_mikey_message_new_from_caps (srtcpparams);
+ if (mikey_msg) {
+ guint send_ssrc;
+
+ gst_rtsp_stream_get_ssrc (stream, &send_ssrc);
+ gst_mikey_message_add_cs_srtp (mikey_msg, 0, send_ssrc, 0);
+
+ base64 = gst_mikey_message_base64_encode (mikey_msg);
+ gst_mikey_message_unref (mikey_msg);
+
+ if (base64) {
+ result = gst_sdp_make_keymgmt (location, base64);
+ g_free (base64);
+ }
+ }
+
+done:
+ return result;
+}
+
+static gboolean
+handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ GstRTSPUrl *uri;
+ gchar *transport, *keymgmt;
+ GstRTSPTransport *ct, *st;
+ GstRTSPStatusCode code;
+ GstRTSPSession *session;
+ GstRTSPStreamTransport *trans;
+ gchar *trans_str;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
+ GstRTSPStream *stream;
+ GstRTSPState rtspstate;
+ GstRTSPClientClass *klass;
+ gchar *path, *control = NULL;
+ gint matched;
+ gboolean new_session = FALSE;
+ GstRTSPStatusCode sig_result;
+ gchar *pipelined_request_id = NULL, *accept_range = NULL;
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ uri = ctx->uri;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
+
+ /* parse the transport */
+ res =
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
+ &transport, 0);
+ if (res != GST_RTSP_OK)
+ goto no_transport;
+
+ /* Handle Pipelined-requests if using >= 2.0 */
+ if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0)
+ gst_rtsp_message_get_header (ctx->request,
+ GST_RTSP_HDR_PIPELINED_REQUESTS, &pipelined_request_id, 0);
+
+ /* we create the session after parsing stuff so that we don't make
+ * a session for malformed requests */
+ if (priv->session_pool == NULL)
+ goto no_pool;
+
+ session = ctx->session;
+
+ if (session) {
+ g_object_ref (session);
+ /* get a handle to the configuration of the media in the session, this can
+ * return NULL if this is a new url to manage in this session. */
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ } else {
+ /* we need a new media configuration in this session */
+ sessmedia = NULL;
+ }
+
+ /* we have no session media, find one and manage it */
+ if (sessmedia == NULL) {
+ /* get a handle to the configuration of the media in the session */
+ media = find_media (client, ctx, path, &matched);
+ /* need to suspend the media, if the protocol has changed */
+ if (media != NULL) {
+ gst_rtsp_media_lock (media);
+ gst_rtsp_media_suspend (media);
+ }
+ } else {
+ if ((media = gst_rtsp_session_media_get_media (sessmedia))) {
+ g_object_ref (media);
+ gst_rtsp_media_lock (media);
+ } else {
+ goto media_not_found;
+ }
+ }
+ /* no media, not found then */
+ if (media == NULL)
+ goto media_not_found_no_reply;
+
+ if (path[matched] == '\0') {
+ if (gst_rtsp_media_n_streams (media) == 1) {
+ stream = gst_rtsp_media_get_stream (media, 0);
+ } else {
+ goto control_not_found;
+ }
+ } else {
+ /* path is what matched. */
+ gchar *newpath = g_strndup (path, matched);
+ /* control is remainder */
+ if (matched == 1 && path[0] == '/')
+ control = g_strdup (&path[1]);
+ else
+ control = g_strdup (&path[matched + 1]);
+
+ g_free (path);
+ path = newpath;
+
+ /* find the stream now using the control part */
+ stream = gst_rtsp_media_find_stream (media, control);
+ }
+
+ if (stream == NULL)
+ goto stream_not_found;
+
+ /* now we have a uri identifying a valid media and stream */
+ ctx->stream = stream;
+ ctx->media = media;
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_SETUP_REQUEST], 0,
+ ctx, &sig_result);
+ if (sig_result != GST_RTSP_STS_OK) {
+ goto sig_failed;
+ }
+
+ if (session == NULL) {
+ /* create a session if this fails we probably reached our session limit or
+ * something. */
+ if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
+ goto service_unavailable;
+
+ /* Pipelined requests should be cleared between sessions */
+ g_hash_table_remove_all (priv->pipelined_requests);
+
+ /* make sure this client is closed when the session is closed */
+ client_watch_session (client, session);
+
+ new_session = TRUE;
+ /* signal new session */
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
+ session);
+
+ ctx->session = session;
+ }
+
+ if (pipelined_request_id) {
+ g_hash_table_insert (client->priv->pipelined_requests,
+ g_strdup (pipelined_request_id),
+ g_strdup (gst_rtsp_session_get_sessionid (session)));
+ }
+ /* Remember that we had at least one session in the past */
+ priv->had_session = TRUE;
+ rtsp_ctrl_timeout_remove (client);
+
+ if (!klass->configure_client_media (client, media, stream, ctx))
+ goto configure_media_failed_no_reply;
+
+ gst_rtsp_transport_new (&ct);
+
+ /* parse and find a usable supported transport */
+ if (!parse_transport (transport, stream, ct))
+ goto unsupported_transports;
+
+ if ((ct->mode_play
+ && !(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_PLAY)) || (ct->mode_record
+ && !(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_RECORD)))
+ goto unsupported_mode;
+
+ /* parse the keymgmt */
+ if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
+ &keymgmt, 0) == GST_RTSP_OK) {
+ if (!gst_rtsp_stream_handle_keymgmt (ctx->stream, keymgmt))
+ goto keymgmt_error;
+ }
+
+ if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT_RANGES,
+ &accept_range, 0) == GST_RTSP_OK) {
+ GEnumValue *runit = NULL;
+ gint i;
+ gchar **valid_ranges;
+ GEnumClass *runit_class = g_type_class_ref (GST_TYPE_RTSP_RANGE_UNIT);
+
+ gst_rtsp_message_dump (ctx->request);
+ valid_ranges = g_strsplit (accept_range, ",", -1);
+
+ for (i = 0; valid_ranges[i]; i++) {
+ gchar *range = valid_ranges[i];
+
+ while (*range == ' ')
+ range++;
+
+ runit = g_enum_get_value_by_nick (runit_class, range);
+ if (runit)
+ break;
+ }
+ g_strfreev (valid_ranges);
+ g_type_class_unref (runit_class);
+
+ if (!runit)
+ goto unsupported_range_unit;
+ }
+
+ if (sessmedia == NULL) {
+ /* manage the media in our session now, if not done already */
+ sessmedia =
+ gst_rtsp_session_manage_media (session, path, g_object_ref (media));
+ /* if we stil have no media, error */
+ if (sessmedia == NULL)
+ goto sessmedia_unavailable;
+
+ /* don't cache media anymore */
+ clean_cached_media (client, FALSE);
+ }
+
+ ctx->sessmedia = sessmedia;
+
+ /* update the client transport */
+ if (!klass->configure_client_transport (client, ctx, ct))
+ goto unsupported_client_transport;
+
+ /* set in the session media transport */
+ trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
+
+ ctx->trans = trans;
+
+ /* configure the url used to set this transport, this we will use when
+ * generating the response for the PLAY request */
+ gst_rtsp_stream_transport_set_url (trans, uri);
+ /* configure keepalive for this transport */
+ gst_rtsp_stream_transport_set_keepalive (trans,
+ (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
+
+ if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+ /* our callbacks to send data on this TCP connection */
+ gst_rtsp_stream_transport_set_callbacks (trans,
+ (GstRTSPSendFunc) do_send_data,
+ (GstRTSPSendFunc) do_send_data, client, NULL);
+ gst_rtsp_stream_transport_set_list_callbacks (trans,
+ (GstRTSPSendListFunc) do_send_data_list,
+ (GstRTSPSendListFunc) do_send_data_list, client, NULL);
+
+ gst_rtsp_stream_transport_set_back_pressure_callback (trans,
+ (GstRTSPBackPressureFunc) do_check_back_pressure, client, NULL);
+
+ g_hash_table_insert (priv->transports,
+ GINT_TO_POINTER (ct->interleaved.min), trans);
+ g_object_ref (trans);
+ g_hash_table_insert (priv->transports,
+ GINT_TO_POINTER (ct->interleaved.max), trans);
+ g_object_ref (trans);
+ add_data_seq (client, ct->interleaved.min);
+ add_data_seq (client, ct->interleaved.max);
+ }
+
+ /* create and serialize the server transport */
+ st = make_server_transport (client, media, ctx, ct);
+ trans_str = gst_rtsp_transport_as_text (st);
+ gst_rtsp_transport_free (st);
+
+ /* construct the response now */
+ code = GST_RTSP_STS_OK;
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
+ trans_str);
+ g_free (trans_str);
+
+ if (pipelined_request_id)
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PIPELINED_REQUESTS,
+ pipelined_request_id);
+
+ if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0) {
+ GstClockTimeDiff seekable = gst_rtsp_media_seekable (media);
+ GString *media_properties = g_string_new (NULL);
+
+ if (seekable == -1)
+ g_string_append (media_properties,
+ "No-Seeking,Time-Progressing,Time-Duration=0.0");
+ else if (seekable == 0)
+ g_string_append (media_properties, "Beginning-Only");
+ else if (seekable == G_MAXINT64)
+ g_string_append (media_properties, "Random-Access");
+ else
+ g_string_append_printf (media_properties,
+ "Random-Access=%f, Unlimited, Immutable",
+ (gdouble) seekable / GST_SECOND);
+
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_MEDIA_PROPERTIES,
+ media_properties->str);
+ g_string_free (media_properties, TRUE);
+ /* TODO Check how Accept-Ranges should be filled */
+ gst_rtsp_message_add_header (ctx->request, GST_RTSP_HDR_ACCEPT_RANGES,
+ "npt, clock, smpte, clock");
+ }
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ /* update the state */
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ switch (rtspstate) {
+ case GST_RTSP_STATE_PLAYING:
+ case GST_RTSP_STATE_RECORDING:
+ case GST_RTSP_STATE_READY:
+ /* no state change */
+ break;
+ default:
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
+ break;
+ }
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
+
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ g_object_unref (session);
+ g_free (path);
+ g_free (control);
+
+ return TRUE;
+
+ /* ERRORS */
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_transport:
+ {
+ GST_ERROR ("client %p: no transport", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ goto cleanup_path;
+ }
+no_pool:
+ {
+ GST_ERROR ("client %p: no session pool configured", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ goto cleanup_path;
+ }
+media_not_found_no_reply:
+ {
+ GST_ERROR ("client %p: media '%s' not found", client, path);
+ /* error reply is already sent */
+ goto cleanup_session;
+ }
+media_not_found:
+ {
+ GST_ERROR ("client %p: media '%s' not found", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ goto cleanup_session;
+ }
+control_not_found:
+ {
+ GST_ERROR ("client %p: no control in path '%s'", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ goto cleanup_session;
+ }
+stream_not_found:
+ {
+ GST_ERROR ("client %p: stream '%s' not found", client,
+ GST_STR_NULL (control));
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ goto cleanup_session;
+ }
+sig_failed:
+ {
+ GST_ERROR ("client %p: pre signal returned error: %s", client,
+ gst_rtsp_status_as_text (sig_result));
+ send_generic_response (client, sig_result, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ goto cleanup_path;
+ }
+service_unavailable:
+ {
+ GST_ERROR ("client %p: can't create session", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ goto cleanup_session;
+ }
+sessmedia_unavailable:
+ {
+ GST_ERROR ("client %p: can't create session media", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ goto cleanup_transport;
+ }
+configure_media_failed_no_reply:
+ {
+ GST_ERROR ("client %p: configure_media failed", client);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ /* error reply is already sent */
+ goto cleanup_session;
+ }
+unsupported_transports:
+ {
+ GST_ERROR ("client %p: unsupported transports", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ goto cleanup_transport;
+ }
+unsupported_client_transport:
+ {
+ GST_ERROR ("client %p: unsupported client transport", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ goto cleanup_transport;
+ }
+unsupported_mode:
+ {
+ GST_ERROR ("client %p: unsupported mode (media play: %d, media record: %d, "
+ "mode play: %d, mode record: %d)", client,
+ ! !(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_PLAY),
+ ! !(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_RECORD), ct->mode_play, ct->mode_record);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ goto cleanup_transport;
+ }
+unsupported_range_unit:
+ {
+ GST_ERROR ("Client %p: does not support any range format we support",
+ client);
+ send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
+ goto cleanup_transport;
+ }
+keymgmt_error:
+ {
+ GST_ERROR ("client %p: keymgmt error", client);
+ send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
+ goto cleanup_transport;
+ }
+ {
+ cleanup_transport:
+ gst_rtsp_transport_free (ct);
+ if (media) {
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ }
+ cleanup_session:
+ if (new_session)
+ gst_rtsp_session_pool_remove (priv->session_pool, session);
+ if (session)
+ g_object_unref (session);
+ cleanup_path:
+ g_free (path);
+ g_free (control);
+ return FALSE;
+ }
+}
+
+static GstSDPMessage *
+create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstSDPMessage *sdp;
+ GstSDPInfo info;
+ const gchar *proto;
+ guint64 session_id_tmp;
+ gchar session_id[21];
+
+ gst_sdp_message_new (&sdp);
+
+ /* some standard things first */
+ gst_sdp_message_set_version (sdp, "0");
+
+ if (priv->is_ipv6)
+ proto = "IP6";
+ else
+ proto = "IP4";
+
+ session_id_tmp = (((guint64) g_random_int ()) << 32) | g_random_int ();
+ g_snprintf (session_id, sizeof (session_id), "%" G_GUINT64_FORMAT,
+ session_id_tmp);
+
+ gst_sdp_message_set_origin (sdp, "-", session_id, "1", "IN", proto,
+ priv->server_ip);
+
+ gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
+ gst_sdp_message_set_information (sdp, "rtsp-server");
+ gst_sdp_message_add_time (sdp, "0", "0", NULL);
+ gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
+ gst_sdp_message_add_attribute (sdp, "type", "broadcast");
+ gst_sdp_message_add_attribute (sdp, "control", "*");
+
+ info.is_ipv6 = priv->is_ipv6;
+ info.server_ip = priv->server_ip;
+
+ /* create an SDP for the media object */
+ if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
+ goto no_sdp;
+
+ return sdp;
+
+ /* ERRORS */
+no_sdp:
+ {
+ GST_ERROR ("client %p: could not create SDP", client);
+ gst_sdp_message_free (sdp);
+ return NULL;
+ }
+}
+
+/* for the describe we must generate an SDP */
+static gboolean
+handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ GstSDPMessage *sdp;
+ guint i;
+ gchar *path, *str;
+ GstRTSPMedia *media;
+ GstRTSPClientClass *klass;
+ GstRTSPStatusCode sig_result;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_DESCRIBE_REQUEST],
+ 0, ctx, &sig_result);
+ if (sig_result != GST_RTSP_STS_OK) {
+ goto sig_failed;
+ }
+
+ /* check what kind of format is accepted, we don't really do anything with it
+ * and always return SDP for now. */
+ for (i = 0;; i++) {
+ gchar *accept;
+
+ res =
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
+ &accept, i);
+ if (res == GST_RTSP_ENOTIMPL)
+ break;
+
+ if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
+ break;
+ }
+
+ if (!priv->mount_points)
+ goto no_mount_points;
+
+ if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
+ goto no_path;
+
+ /* find the media object for the uri */
+ if (!(media = find_media (client, ctx, path, NULL)))
+ goto no_media;
+
+ gst_rtsp_media_lock (media);
+
+ if (!(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_PLAY))
+ goto unsupported_mode;
+
+ /* create an SDP for the media object on this client */
+ if (!(sdp = klass->create_sdp (client, media)))
+ goto no_sdp;
+
+ /* we suspend after the describe */
+ gst_rtsp_media_suspend (media);
+
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/sdp");
+
+ /* content base for some clients that might screw up creating the setup uri */
+ str = make_base_url (client, ctx->uri, path);
+ g_free (path);
+
+ GST_INFO ("adding content-base: %s", str);
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
+
+ /* add SDP to the response body */
+ str = gst_sdp_message_as_text (sdp);
+ gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
+ gst_sdp_message_free (sdp);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
+ 0, ctx);
+
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+
+ return TRUE;
+
+ /* ERRORS */
+sig_failed:
+ {
+ GST_ERROR ("client %p: pre signal returned error: %s", client,
+ gst_rtsp_status_as_text (sig_result));
+ send_generic_response (client, sig_result, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_mount_points:
+ {
+ GST_ERROR ("client %p: no mount points configured", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_path:
+ {
+ GST_ERROR ("client %p: can't find path for url", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_media:
+ {
+ GST_ERROR ("client %p: no media", client);
+ g_free (path);
+ /* error reply is already sent */
+ return FALSE;
+ }
+unsupported_mode:
+ {
+ GST_ERROR ("client %p: media does not support DESCRIBE", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
+ g_free (path);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+no_sdp:
+ {
+ GST_ERROR ("client %p: can't create SDP", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_free (path);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPMedia * media,
+ GstSDPMessage * sdp)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPThread *thread;
+
+ /* create an SDP for the media object */
+ if (!gst_rtsp_media_handle_sdp (media, sdp))
+ goto unhandled_sdp;
+
+ thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, ctx);
+ if (thread == NULL)
+ goto no_thread;
+
+ /* prepare the media */
+ if (!gst_rtsp_media_prepare (media, thread))
+ goto no_prepare;
+
+ return TRUE;
+
+ /* ERRORS */
+unhandled_sdp:
+ {
+ GST_ERROR ("client %p: could not handle SDP", client);
+ return FALSE;
+ }
+no_thread:
+ {
+ GST_ERROR ("client %p: can't create thread", client);
+ return FALSE;
+ }
+no_prepare:
+ {
+ GST_ERROR ("client %p: can't prepare media", client);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPClientClass *klass;
+ GstSDPResult sres;
+ GstSDPMessage *sdp;
+ GstRTSPMedia *media;
+ gchar *path, *cont = NULL;
+ guint8 *data;
+ guint size;
+ GstRTSPStatusCode sig_result;
+ guint i, n_streams;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ if (!priv->mount_points)
+ goto no_mount_points;
+
+ /* check if reply is SDP */
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_CONTENT_TYPE, &cont,
+ 0);
+ /* could not be set but since the request returned OK, we assume it
+ * was SDP, else check it. */
+ if (cont) {
+ if (g_ascii_strcasecmp (cont, "application/sdp") != 0)
+ goto wrong_content_type;
+ }
+
+ /* get message body and parse as SDP */
+ gst_rtsp_message_get_body (ctx->request, &data, &size);
+ if (data == NULL || size == 0)
+ goto no_message;
+
+ GST_DEBUG ("client %p: parse SDP...", client);
+ gst_sdp_message_new (&sdp);
+ sres = gst_sdp_message_parse_buffer (data, size, sdp);
+ if (sres != GST_SDP_OK)
+ goto sdp_parse_failed;
+
+ if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
+ goto no_path;
+
+ /* find the media object for the uri */
+ if (!(media = find_media (client, ctx, path, NULL)))
+ goto no_media;
+
+ ctx->media = media;
+ gst_rtsp_media_lock (media);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_ANNOUNCE_REQUEST],
+ 0, ctx, &sig_result);
+ if (sig_result != GST_RTSP_STS_OK) {
+ goto sig_failed;
+ }
+
+ if (!(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_RECORD))
+ goto unsupported_mode;
+
+ /* Tell client subclass about the media */
+ if (!klass->handle_sdp (client, ctx, media, sdp))
+ goto unhandled_sdp;
+
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+ n_streams = gst_rtsp_media_n_streams (media);
+ for (i = 0; i < n_streams; i++) {
+ GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
+ gchar *uri, *location, *keymgmt;
+
+ uri = gst_rtsp_url_get_request_uri (ctx->uri);
+ location = g_strdup_printf ("%s/stream=%d", uri, i);
+ keymgmt = stream_make_keymgmt (client, location, stream);
+
+ if (keymgmt)
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_KEYMGMT,
+ keymgmt);
+
+ g_free (location);
+ g_free (uri);
+ }
+
+ /* we suspend after the announce */
+ gst_rtsp_media_suspend (media);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST],
+ 0, ctx);
+
+ gst_sdp_message_free (sdp);
+ g_free (path);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+
+ return TRUE;
+
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_mount_points:
+ {
+ GST_ERROR ("client %p: no mount points configured", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_path:
+ {
+ GST_ERROR ("client %p: can't find path for url", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+wrong_content_type:
+ {
+ GST_ERROR ("client %p: unknown content type", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_message:
+ {
+ GST_ERROR ("client %p: can't find SDP message", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+sdp_parse_failed:
+ {
+ GST_ERROR ("client %p: failed to parse SDP message", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+no_media:
+ {
+ GST_ERROR ("client %p: no media", client);
+ g_free (path);
+ /* error reply is already sent */
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+sig_failed:
+ {
+ GST_ERROR ("client %p: pre signal returned error: %s", client,
+ gst_rtsp_status_as_text (sig_result));
+ send_generic_response (client, sig_result, ctx);
+ gst_sdp_message_free (sdp);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ return FALSE;
+ }
+unsupported_mode:
+ {
+ GST_ERROR ("client %p: media does not support ANNOUNCE", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
+ g_free (path);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+unhandled_sdp:
+ {
+ GST_ERROR ("client %p: can't handle SDP", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE, ctx);
+ g_free (path);
+ gst_rtsp_media_unlock (media);
+ g_object_unref (media);
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_record_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPSession *session;
+ GstRTSPClientClass *klass;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
+ GstRTSPUrl *uri;
+ GstRTSPState rtspstate;
+ gchar *path;
+ gint matched;
+ GstRTSPStatusCode sig_result;
+ GPtrArray *transports;
+
+ if (!(session = ctx->session))
+ goto no_session;
+
+ if (!(uri = ctx->uri))
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
+
+ /* get a handle to the configuration of the media in the session */
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
+ goto not_found;
+
+ if (path[matched] != '\0')
+ goto no_aggregate;
+
+ g_free (path);
+
+ ctx->sessmedia = sessmedia;
+ ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_RECORD_REQUEST], 0,
+ ctx, &sig_result);
+ if (sig_result != GST_RTSP_STS_OK) {
+ goto sig_failed;
+ }
+
+ if (!(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_RECORD))
+ goto unsupported_mode;
+
+ /* the session state must be playing or ready */
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
+ goto invalid_state;
+
+ /* update the pipeline */
+ transports = gst_rtsp_session_media_get_transports (sessmedia);
+ if (!gst_rtsp_media_complete_pipeline (media, transports)) {
+ g_ptr_array_unref (transports);
+ goto pipeline_error;
+ }
+ g_ptr_array_unref (transports);
+
+ /* in record we first unsuspend, media could be suspended from SDP or PAUSED */
+ if (!gst_rtsp_media_unsuspend (media))
+ goto unsuspend_failed;
+
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ /* start playing after sending the response */
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
+
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST], 0,
+ ctx);
+
+ return TRUE;
+
+ /* ERRORS */
+no_session:
+ {
+ GST_ERROR ("client %p: no session", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri supplied", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+not_found:
+ {
+ GST_ERROR ("client %p: media not found", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_aggregate:
+ {
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
+ return FALSE;
+ }
+sig_failed:
+ {
+ GST_ERROR ("client %p: pre signal returned error: %s", client,
+ gst_rtsp_status_as_text (sig_result));
+ send_generic_response (client, sig_result, ctx);
+ return FALSE;
+ }
+unsupported_mode:
+ {
+ GST_ERROR ("client %p: media does not support RECORD", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
+ return FALSE;
+ }
+invalid_state:
+ {
+ GST_ERROR ("client %p: not PLAYING or READY", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ ctx);
+ return FALSE;
+ }
+pipeline_error:
+ {
+ GST_ERROR ("client %p: failed to configure the pipeline", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ ctx);
+ return FALSE;
+ }
+unsuspend_failed:
+ {
+ GST_ERROR ("client %p: unsuspend failed", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx,
+ GstRTSPVersion version)
+{
+ GstRTSPMethod options;
+ gchar *str;
+ GstRTSPStatusCode sig_result;
+
+ options = GST_RTSP_DESCRIBE |
+ GST_RTSP_OPTIONS |
+ GST_RTSP_PAUSE |
+ GST_RTSP_PLAY |
+ GST_RTSP_SETUP |
+ GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
+
+ if (version < GST_RTSP_VERSION_2_0) {
+ options |= GST_RTSP_RECORD;
+ options |= GST_RTSP_ANNOUNCE;
+ }
+
+ str = gst_rtsp_options_as_text (options);
+
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
+ g_free (str);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_OPTIONS_REQUEST], 0,
+ ctx, &sig_result);
+ if (sig_result != GST_RTSP_STS_OK) {
+ goto sig_failed;
+ }
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
+ 0, ctx);
+
+ return TRUE;
+
+/* ERRORS */
+sig_failed:
+ {
+ GST_ERROR ("client %p: pre signal returned error: %s", client,
+ gst_rtsp_status_as_text (sig_result));
+ send_generic_response (client, sig_result, ctx);
+ gst_rtsp_message_free (ctx->response);
+ return FALSE;
+ }
+}
+
+/* remove duplicate and trailing '/' */
+static void
+sanitize_uri (GstRTSPUrl * uri)
+{
+ gint i, len;
+ gchar *s, *d;
+ gboolean have_slash, prev_slash;
+
+ s = d = uri->abspath;
+ len = strlen (uri->abspath);
+
+ prev_slash = FALSE;
+
+ for (i = 0; i < len; i++) {
+ have_slash = s[i] == '/';
+ *d = s[i];
+ if (!have_slash || !prev_slash)
+ d++;
+ prev_slash = have_slash;
+ }
+ len = d - uri->abspath;
+ /* don't remove the first slash if that's the only thing left */
+ if (len > 1 && *(d - 1) == '/')
+ d--;
+ *d = '\0';
+}
+
+/* is called when the session is removed from its session pool. */
+static void
+client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
+ GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GSource *timer_src;
+
+ GST_INFO ("client %p: session %p removed", client, session);
+
+ g_mutex_lock (&priv->lock);
+ client_unwatch_session (client, session, NULL);
+
+ if (!priv->sessions && priv->rtsp_ctrl_timeout == NULL) {
+ if (priv->post_session_timeout > 0) {
+ GWeakRef *client_weak_ref = g_new (GWeakRef, 1);
+ timer_src = g_timeout_source_new_seconds (RTSP_CTRL_CB_INTERVAL);
+
+ g_weak_ref_init (client_weak_ref, client);
+ g_source_set_callback (timer_src, rtsp_ctrl_timeout_cb, client_weak_ref,
+ rtsp_ctrl_timeout_destroy_notify);
+ priv->rtsp_ctrl_timeout_cnt = 0;
+ g_source_attach (timer_src, priv->watch_context);
+ priv->rtsp_ctrl_timeout = timer_src;
+ GST_DEBUG ("rtsp control setting up connection timeout %p.",
+ priv->rtsp_ctrl_timeout);
+ g_mutex_unlock (&priv->lock);
+ } else if (priv->post_session_timeout == 0) {
+ g_mutex_unlock (&priv->lock);
+ gst_rtsp_client_close (client);
+ } else {
+ g_mutex_unlock (&priv->lock);
+ }
+ } else {
+ g_mutex_unlock (&priv->lock);
+ }
+}
+
+/* Check for Require headers. Returns TRUE if there are no Require headers,
+ * otherwise lets the application decide which headers are supported.
+ * By default all headers are unsupported.
+ * If there are unsupported options, FALSE will be returned together with
+ * a newly-allocated string of (comma-separated) unsupported options in
+ * the unsupported_reqs variable.
+ *
+ * There may be multiple Require headers, but we must send one single
+ * Unsupported header with all the unsupported options as response. If
+ * an incoming Require header contained a comma-separated list of options
+ * GstRtspConnection will already have split that list up into multiple
+ * headers.
+ */
+static gboolean
+check_request_requirements (GstRTSPContext * ctx, gchar ** unsupported_reqs)
+{
+ GstRTSPResult res;
+ GPtrArray *arr = NULL;
+ GstRTSPMessage *msg = ctx->request;
+ gchar *reqs = NULL;
+ gint i;
+ gchar *sig_result = NULL;
+ gboolean result = TRUE;
+
+ i = 0;
+ do {
+ res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
+
+ if (res == GST_RTSP_ENOTIMPL)
+ break;
+
+ if (arr == NULL)
+ arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
+
+ g_ptr_array_add (arr, g_strdup (reqs));
+ }
+ while (TRUE);
+
+ /* if we don't have any Require headers at all, all is fine */
+ if (i == 1)
+ return TRUE;
+
+ /* otherwise we've now processed at all the Require headers */
+ g_ptr_array_add (arr, NULL);
+
+ g_signal_emit (ctx->client,
+ gst_rtsp_client_signals[SIGNAL_CHECK_REQUIREMENTS], 0, ctx,
+ (gchar **) arr->pdata, &sig_result);
+
+ if (sig_result == NULL) {
+ /* no supported options, just report all of the required ones as
+ * unsupported */
+ *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
+ result = FALSE;
+ goto done;
+ }
+
+ if (strlen (sig_result) == 0)
+ g_free (sig_result);
+ else {
+ *unsupported_reqs = sig_result;
+ result = FALSE;
+ }
+
+done:
+ g_ptr_array_unref (arr);
+ return result;
+}
+
+static void
+handle_request (GstRTSPClient * client, GstRTSPMessage * request)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPMethod method;
+ const gchar *uristr;
+ GstRTSPUrl *uri = NULL;
+ GstRTSPVersion version;
+ GstRTSPResult res;
+ GstRTSPSession *session = NULL;
+ GstRTSPContext sctx = { NULL }, *ctx;
+ GstRTSPMessage response = { 0 };
+ gchar *unsupported_reqs = NULL;
+ gchar *sessid = NULL, *pipelined_request_id = NULL;
+
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->request = request;
+ ctx->response = &response;
+
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (request);
+ }
+
+ gst_rtsp_message_parse_request (request, &method, &uristr, &version);
+
+ GST_INFO ("client %p: received a request %s %s %s", client,
+ gst_rtsp_method_as_text (method), uristr,
+ gst_rtsp_version_as_text (version));
+
+ /* we can only handle 1.0 requests */
+ if (version != GST_RTSP_VERSION_1_0 && version != GST_RTSP_VERSION_2_0)
+ goto not_supported;
+
+ ctx->method = method;
+
+ /* we always try to parse the url first */
+ if (strcmp (uristr, "*") == 0) {
+ /* special case where we have * as uri, keep uri = NULL */
+ } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
+ /* check if the uristr is an absolute path <=> scheme and host information
+ * is missing */
+ gchar *scheme;
+
+ scheme = g_uri_parse_scheme (uristr);
+ if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
+ gchar *absolute_uristr = NULL;
+
+ GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
+ if (priv->server_ip == NULL) {
+ GST_WARNING_OBJECT (client, "host information missing");
+ goto bad_request;
+ }
+
+ absolute_uristr =
+ g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
+
+ GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
+ if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
+ g_free (absolute_uristr);
+ goto bad_request;
+ }
+ g_free (absolute_uristr);
+ } else {
+ g_free (scheme);
+ goto bad_request;
+ }
+ }
+
+ /* get the session if there is any */
+ res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_PIPELINED_REQUESTS,
+ &pipelined_request_id, 0);
+ if (res == GST_RTSP_OK) {
+ sessid = g_hash_table_lookup (client->priv->pipelined_requests,
+ pipelined_request_id);
+
+ if (!sessid)
+ res = GST_RTSP_ERROR;
+ }
+
+ if (res != GST_RTSP_OK)
+ res =
+ gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
+
+ if (res == GST_RTSP_OK) {
+ if (priv->session_pool == NULL)
+ goto no_pool;
+
+ /* we had a session in the request, find it again */
+ if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
+ goto session_not_found;
+
+ /* we add the session to the client list of watched sessions. When a session
+ * disappears because it times out, we will be notified. If all sessions are
+ * gone, we will close the connection */
+ client_watch_session (client, session);
+ }
+
+ /* sanitize the uri */
+ if (uri)
+ sanitize_uri (uri);
+ ctx->uri = uri;
+ ctx->session = session;
+
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
+ goto not_authorized;
+
+ /* handle any 'Require' headers */
+ if (!check_request_requirements (ctx, &unsupported_reqs))
+ goto unsupported_requirement;
+
+ /* now see what is asked and dispatch to a dedicated handler */
+ switch (method) {
+ case GST_RTSP_OPTIONS:
+ priv->version = version;
+ handle_options_request (client, ctx, version);
+ break;
+ case GST_RTSP_DESCRIBE:
+ handle_describe_request (client, ctx);
+ break;
+ case GST_RTSP_SETUP:
+ handle_setup_request (client, ctx);
+ break;
+ case GST_RTSP_PLAY:
+ handle_play_request (client, ctx);
+ break;
+ case GST_RTSP_PAUSE:
+ handle_pause_request (client, ctx);
+ break;
+ case GST_RTSP_TEARDOWN:
+ handle_teardown_request (client, ctx);
+ break;
+ case GST_RTSP_SET_PARAMETER:
+ handle_set_param_request (client, ctx);
+ break;
+ case GST_RTSP_GET_PARAMETER:
+ handle_get_param_request (client, ctx);
+ break;
+ case GST_RTSP_ANNOUNCE:
+ if (version >= GST_RTSP_VERSION_2_0)
+ goto invalid_command_for_version;
+ handle_announce_request (client, ctx);
+ break;
+ case GST_RTSP_RECORD:
+ if (version >= GST_RTSP_VERSION_2_0)
+ goto invalid_command_for_version;
+ handle_record_request (client, ctx);
+ break;
+ case GST_RTSP_REDIRECT:
+ goto not_implemented;
+ case GST_RTSP_INVALID:
+ default:
+ goto bad_request;
+ }
+
+done:
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
+ if (session)
+ g_object_unref (session);
+ if (uri)
+ gst_rtsp_url_free (uri);
+ return;
+
+ /* ERRORS */
+not_supported:
+ {
+ GST_ERROR ("client %p: version %d not supported", client, version);
+ send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
+ ctx);
+ goto done;
+ }
+invalid_command_for_version:
+ {
+ GST_ERROR ("client %p: invalid command for version", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ goto done;
+ }
+bad_request:
+ {
+ GST_ERROR ("client %p: bad request", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ goto done;
+ }
+no_pool:
+ {
+ GST_ERROR ("client %p: no pool configured", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ goto done;
+ }
+session_not_found:
+ {
+ GST_ERROR ("client %p: session not found", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ goto done;
+ }
+not_authorized:
+ {
+ GST_ERROR ("client %p: not allowed", client);
+ /* error reply is already sent */
+ goto done;
+ }
+unsupported_requirement:
+ {
+ GST_ERROR ("client %p: Required option is not supported (%s)", client,
+ unsupported_reqs);
+ send_option_not_supported_response (client, ctx, unsupported_reqs);
+ g_free (unsupported_reqs);
+ goto done;
+ }
+not_implemented:
+ {
+ GST_ERROR ("client %p: method %d not implemented", client, method);
+ send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
+ goto done;
+ }
+}
+
+
+static void
+handle_response (GstRTSPClient * client, GstRTSPMessage * response)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ GstRTSPSession *session = NULL;
+ GstRTSPContext sctx = { NULL }, *ctx;
+ gchar *sessid;
+
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->request = NULL;
+ ctx->uri = NULL;
+ ctx->method = GST_RTSP_INVALID;
+ ctx->response = response;
+
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (response);
+ }
+
+ GST_INFO ("client %p: received a response", client);
+
+ /* get the session if there is any */
+ res =
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
+ if (res == GST_RTSP_OK) {
+ if (priv->session_pool == NULL)
+ goto no_pool;
+
+ /* we had a session in the request, find it again */
+ if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
+ goto session_not_found;
+
+ /* we add the session to the client list of watched sessions. When a session
+ * disappears because it times out, we will be notified. If all sessions are
+ * gone, we will close the connection */
+ client_watch_session (client, session);
+ }
+
+ ctx->session = session;
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
+ 0, ctx);
+
+done:
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
+ if (session)
+ g_object_unref (session);
+ return;
+
+no_pool:
+ {
+ GST_ERROR ("client %p: no pool configured", client);
+ goto done;
+ }
+session_not_found:
+ {
+ GST_ERROR ("client %p: session not found", client);
+ goto done;
+ }
+}
+
+static void
+handle_data (GstRTSPClient * client, GstRTSPMessage * message)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ guint8 channel;
+ guint8 *data;
+ guint size;
+ GstBuffer *buffer;
+ GstRTSPStreamTransport *trans;
+
+ /* find the stream for this message */
+ res = gst_rtsp_message_parse_data (message, &channel);
+ if (res != GST_RTSP_OK)
+ return;
+
+ gst_rtsp_message_get_body (message, &data, &size);
+ if (size < 2)
+ goto invalid_length;
+
+ gst_rtsp_message_steal_body (message, &data, &size);
+
+ /* Strip trailing \0 (which GstRTSPConnection adds) */
+ --size;
+
+ buffer = gst_buffer_new_wrapped (data, size);
+
+ trans =
+ g_hash_table_lookup (priv->transports, GINT_TO_POINTER ((gint) channel));
+ if (trans) {
+ GSocketAddress *addr;
+
+ /* Only create the socket address once for the transport, we don't really
+ * want to do that for every single packet.
+ *
+ * The netaddress meta is later used by the RTP stack to know where
+ * packets came from and allows us to match it again to a stream transport
+ *
+ * In theory we could use the remote socket address of the RTSP connection
+ * here, but this would fail with a custom configure_client_transport()
+ * implementation.
+ */
+ if (!(addr =
+ g_object_get_data (G_OBJECT (trans), "rtsp-client.remote-addr"))) {
+ const GstRTSPTransport *tr;
+ GInetAddress *iaddr;
+
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+ iaddr = g_inet_address_new_from_string (tr->destination);
+ if (iaddr) {
+ addr = g_inet_socket_address_new (iaddr, tr->client_port.min);
+ g_object_unref (iaddr);
+ g_object_set_data_full (G_OBJECT (trans), "rtsp-client.remote-addr",
+ addr, (GDestroyNotify) g_object_unref);
+ }
+ }
+
+ if (addr) {
+ gst_buffer_add_net_address_meta (buffer, addr);
+ }
+
+ /* dispatch to the stream based on the channel number */
+ GST_LOG_OBJECT (client, "%u bytes of data on channel %u", size, channel);
+ gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
+ } else {
+ GST_DEBUG_OBJECT (client, "received %u bytes of data for "
+ "unknown channel %u", size, channel);
+ gst_buffer_unref (buffer);
+ }
+
+ return;
+
+/* ERRORS */
+invalid_length:
+ {
+ GST_DEBUG ("client %p: Short message received, ignoring", client);
+ return;
+ }
+}
+
+/**
+ * gst_rtsp_client_set_session_pool:
+ * @client: a #GstRTSPClient
+ * @pool: (transfer none) (nullable): a #GstRTSPSessionPool
+ *
+ * Set @pool as the sessionpool for @client which it will use to find
+ * or allocate sessions. the sessionpool is usually inherited from the server
+ * that created the client but can be overridden later.
+ */
+void
+gst_rtsp_client_set_session_pool (GstRTSPClient * client,
+ GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPool *old;
+ GstRTSPClientPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ if (pool)
+ g_object_ref (pool);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->session_pool;
+ priv->session_pool = pool;
+
+ if (priv->session_removed_id) {
+ g_signal_handler_disconnect (old, priv->session_removed_id);
+ priv->session_removed_id = 0;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ /* FIXME, should remove all sessions from the old pool for this client */
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_client_get_session_pool:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
+ *
+ * Returns: (transfer full) (nullable): a #GstRTSPSessionPool, unref after usage.
+ */
+GstRTSPSessionPool *
+gst_rtsp_client_get_session_pool (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPSessionPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->session_pool))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_client_set_mount_points:
+ * @client: a #GstRTSPClient
+ * @mounts: (transfer none) (nullable): a #GstRTSPMountPoints
+ *
+ * Set @mounts as the mount points for @client which it will use to map urls
+ * to media streams. These mount points are usually inherited from the server that
+ * created the client but can be overriden later.
+ */
+void
+gst_rtsp_client_set_mount_points (GstRTSPClient * client,
+ GstRTSPMountPoints * mounts)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPMountPoints *old;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ if (mounts)
+ g_object_ref (mounts);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->mount_points;
+ priv->mount_points = mounts;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_client_get_mount_points:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
+ *
+ * Returns: (transfer full) (nullable): a #GstRTSPMountPoints, unref after usage.
+ */
+GstRTSPMountPoints *
+gst_rtsp_client_get_mount_points (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPMountPoints *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->mount_points))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_client_set_content_length_limit:
+ * @client: a #GstRTSPClient
+ * @limit: Content-Length limit
+ *
+ * Configure @client to use the specified Content-Length limit.
+ *
+ * Define an appropriate request size limit and reject requests exceeding the
+ * limit with response status 413 Request Entity Too Large
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_client_set_content_length_limit (GstRTSPClient * client, guint limit)
+{
+ GstRTSPClientPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+ g_mutex_lock (&priv->lock);
+ priv->content_length_limit = limit;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_client_get_content_length_limit:
+ * @client: a #GstRTSPClient
+ *
+ * Get the Content-Length limit of @client.
+ *
+ * Returns: the Content-Length limit.
+ *
+ * Since: 1.18
+ */
+guint
+gst_rtsp_client_get_content_length_limit (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv;
+ glong content_length_limit;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), -1);
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ content_length_limit = priv->content_length_limit;
+ g_mutex_unlock (&priv->lock);
+
+ return content_length_limit;
+}
+
+/**
+ * gst_rtsp_client_set_auth:
+ * @client: a #GstRTSPClient
+ * @auth: (transfer none) (nullable): a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @client.
+ */
+void
+gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPAuth *old;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ if (auth)
+ g_object_ref (auth);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->auth;
+ priv->auth = auth;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+
+/**
+ * gst_rtsp_client_get_auth:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @client.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPAuth of @client.
+ * g_object_unref() after usage.
+ */
+GstRTSPAuth *
+gst_rtsp_client_get_auth (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPAuth *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->auth))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_client_set_thread_pool:
+ * @client: a #GstRTSPClient
+ * @pool: (transfer none) (nullable): a #GstRTSPThreadPool
+ *
+ * configure @pool to be used as the thread pool of @client.
+ */
+void
+gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
+ GstRTSPThreadPool * pool)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPThreadPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ if (pool)
+ g_object_ref (pool);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->thread_pool;
+ priv->thread_pool = pool;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_client_get_thread_pool:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPThreadPool used as the thread pool of @client.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPThreadPool of @client. g_object_unref() after
+ * usage.
+ */
+GstRTSPThreadPool *
+gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPThreadPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->thread_pool))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_client_set_connection:
+ * @client: a #GstRTSPClient
+ * @conn: (transfer full): a #GstRTSPConnection
+ *
+ * Set the #GstRTSPConnection of @client. This function takes ownership of
+ * @conn.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_client_set_connection (GstRTSPClient * client,
+ GstRTSPConnection * conn)
+{
+ GstRTSPClientPrivate *priv;
+ GSocket *read_socket;
+ GSocketAddress *address;
+ GstRTSPUrl *url;
+ GError *error = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
+ g_return_val_if_fail (conn != NULL, FALSE);
+
+ priv = client->priv;
+
+ gst_rtsp_connection_set_content_length_limit (conn,
+ priv->content_length_limit);
+ read_socket = gst_rtsp_connection_get_read_socket (conn);
+
+ if (!(address = g_socket_get_local_address (read_socket, &error)))
+ goto no_address;
+
+ g_free (priv->server_ip);
+ /* keep the original ip that the client connected to */
+ if (G_IS_INET_SOCKET_ADDRESS (address)) {
+ GInetAddress *iaddr;
+
+ iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
+
+ /* socket might be ipv6 but adress still ipv4 */
+ priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
+ priv->server_ip = g_inet_address_to_string (iaddr);
+ g_object_unref (address);
+ } else {
+ priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
+ priv->server_ip = g_strdup ("unknown");
+ }
+
+ GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
+ priv->server_ip, priv->is_ipv6);
+
+ url = gst_rtsp_connection_get_url (conn);
+ GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
+
+ priv->connection = conn;
+
+ return TRUE;
+
+ /* ERRORS */
+no_address:
+ {
+ GST_ERROR ("could not get local address %s", error->message);
+ g_error_free (error);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_client_get_connection:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPConnection of @client.
+ *
+ * Returns: (transfer none) (nullable): the #GstRTSPConnection of @client.
+ * The connection object returned remains valid until the client is freed.
+ */
+GstRTSPConnection *
+gst_rtsp_client_get_connection (GstRTSPClient * client)
+{
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ return client->priv->connection;
+}
+
+/**
+ * gst_rtsp_client_set_send_func:
+ * @client: a #GstRTSPClient
+ * @func: (scope notified): a #GstRTSPClientSendFunc
+ * @user_data: (closure): user data passed to @func
+ * @notify: (allow-none): called when @user_data is no longer in use
+ *
+ * Set @func as the callback that will be called when a new message needs to be
+ * sent to the client. @user_data is passed to @func and @notify is called when
+ * @user_data is no longer in use.
+ *
+ * By default, the client will send the messages on the #GstRTSPConnection that
+ * was configured with gst_rtsp_client_attach() was called.
+ *
+ * It is only allowed to set either a `send_func` or a `send_messages_func`
+ * but not both at the same time.
+ */
+void
+gst_rtsp_client_set_send_func (GstRTSPClient * client,
+ GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
+{
+ GstRTSPClientPrivate *priv;
+ GDestroyNotify old_notify;
+ gpointer old_data;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->send_lock);
+ g_assert (func == NULL || priv->send_messages_func == NULL);
+ priv->send_func = func;
+ old_notify = priv->send_notify;
+ old_data = priv->send_data;
+ priv->send_notify = notify;
+ priv->send_data = user_data;
+ g_mutex_unlock (&priv->send_lock);
+
+ if (old_notify)
+ old_notify (old_data);
+}
+
+/**
+ * gst_rtsp_client_set_send_messages_func:
+ * @client: a #GstRTSPClient
+ * @func: (scope notified): a #GstRTSPClientSendMessagesFunc
+ * @user_data: (closure): user data passed to @func
+ * @notify: (allow-none): called when @user_data is no longer in use
+ *
+ * Set @func as the callback that will be called when new messages needs to be
+ * sent to the client. @user_data is passed to @func and @notify is called when
+ * @user_data is no longer in use.
+ *
+ * By default, the client will send the messages on the #GstRTSPConnection that
+ * was configured with gst_rtsp_client_attach() was called.
+ *
+ * It is only allowed to set either a `send_func` or a `send_messages_func`
+ * but not both at the same time.
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_client_set_send_messages_func (GstRTSPClient * client,
+ GstRTSPClientSendMessagesFunc func, gpointer user_data,
+ GDestroyNotify notify)
+{
+ GstRTSPClientPrivate *priv;
+ GDestroyNotify old_notify;
+ gpointer old_data;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->send_lock);
+ g_assert (func == NULL || priv->send_func == NULL);
+ priv->send_messages_func = func;
+ old_notify = priv->send_messages_notify;
+ old_data = priv->send_messages_data;
+ priv->send_messages_notify = notify;
+ priv->send_messages_data = user_data;
+ g_mutex_unlock (&priv->send_lock);
+
+ if (old_notify)
+ old_notify (old_data);
+}
+
+/**
+ * gst_rtsp_client_handle_message:
+ * @client: a #GstRTSPClient
+ * @message: (transfer none): an #GstRTSPMessage
+ *
+ * Let the client handle @message.
+ *
+ * Returns: a #GstRTSPResult.
+ */
+GstRTSPResult
+gst_rtsp_client_handle_message (GstRTSPClient * client,
+ GstRTSPMessage * message)
+{
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+
+ switch (message->type) {
+ case GST_RTSP_MESSAGE_REQUEST:
+ handle_request (client, message);
+ break;
+ case GST_RTSP_MESSAGE_RESPONSE:
+ handle_response (client, message);
+ break;
+ case GST_RTSP_MESSAGE_DATA:
+ handle_data (client, message);
+ break;
+ default:
+ break;
+ }
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_client_send_message:
+ * @client: a #GstRTSPClient
+ * @session: (allow-none) (transfer none): a #GstRTSPSession to send
+ * the message to or %NULL
+ * @message: (transfer none): The #GstRTSPMessage to send
+ *
+ * Send a message message to the remote end. @message must be a
+ * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
+ */
+GstRTSPResult
+gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPMessage * message)
+{
+ GstRTSPContext sctx = { NULL }
+ , *ctx;
+ GstRTSPClientPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
+ message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
+
+ priv = client->priv;
+
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->session = session;
+
+ send_message (client, ctx, message, FALSE);
+
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
+
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_client_get_stream_transport:
+ *
+ * This is useful when providing a send function through
+ * gst_rtsp_client_set_send_func() when doing RTSP over TCP:
+ * the send function must call gst_rtsp_stream_transport_message_sent ()
+ * on the appropriate transport when data has been received for streaming
+ * to continue.
+ *
+ * Returns: (transfer none) (nullable): the #GstRTSPStreamTransport associated with @channel.
+ *
+ * Since: 1.18
+ */
+GstRTSPStreamTransport *
+gst_rtsp_client_get_stream_transport (GstRTSPClient * self, guint8 channel)
+{
+ return g_hash_table_lookup (self->priv->transports,
+ GINT_TO_POINTER ((gint) channel));
+}
+
+static gboolean
+do_send_messages (GstRTSPClient * client, GstRTSPMessage * messages,
+ guint n_messages, gboolean close, gpointer user_data)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ guint id = 0;
+ GstRTSPResult ret;
+ guint i;
+
+ /* send the message */
+ if (close)
+ GST_INFO ("client %p: sending close message", client);
+
+ ret = gst_rtsp_watch_send_messages (priv->watch, messages, n_messages, &id);
+ if (ret != GST_RTSP_OK)
+ goto error;
+
+ for (i = 0; i < n_messages; i++) {
+ if (gst_rtsp_message_get_type (&messages[i]) == GST_RTSP_MESSAGE_DATA) {
+ guint8 channel = 0;
+ GstRTSPResult r;
+
+ /* We assume that all data messages in the list are for the
+ * same channel */
+ r = gst_rtsp_message_parse_data (&messages[i], &channel);
+ if (r != GST_RTSP_OK) {
+ ret = r;
+ goto error;
+ }
+
+ /* check if the message has been queued for transmission in watch */
+ if (id) {
+ /* store the seq number so we can wait until it has been sent */
+ GST_DEBUG_OBJECT (client, "wait for message %d, channel %d", id,
+ channel);
+ set_data_seq (client, channel, id);
+ } else {
+ GstRTSPStreamTransport *trans;
+
+ trans =
+ g_hash_table_lookup (priv->transports,
+ GINT_TO_POINTER ((gint) channel));
+ if (trans) {
+ GST_DEBUG_OBJECT (client, "emit 'message-sent' signal");
+ g_mutex_unlock (&priv->send_lock);
+ gst_rtsp_stream_transport_message_sent (trans);
+ g_mutex_lock (&priv->send_lock);
+ }
+ }
+ break;
+ }
+ }
+
+ return ret == GST_RTSP_OK;
+
+ /* ERRORS */
+error:
+ {
+ GST_DEBUG_OBJECT (client, "got error %d", ret);
+ return FALSE;
+ }
+}
+
+static GstRTSPResult
+message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
+ gpointer user_data)
+{
+ return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
+}
+
+static GstRTSPResult
+message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPStreamTransport *trans = NULL;
+ guint8 channel = 0;
+
+ g_mutex_lock (&priv->send_lock);
+
+ if (get_data_channel (client, cseq, &channel)) {
+ trans = g_hash_table_lookup (priv->transports, GINT_TO_POINTER (channel));
+ set_data_seq (client, channel, 0);
+ }
+ g_mutex_unlock (&priv->send_lock);
+
+ if (trans) {
+ GST_DEBUG_OBJECT (client, "emit 'message-sent' signal");
+ gst_rtsp_stream_transport_message_sent (trans);
+ }
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+closed (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
+ const gchar *tunnelid;
+
+ GST_INFO ("client %p: connection closed", client);
+
+ if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
+ g_mutex_lock (&tunnels_lock);
+ /* remove from tunnelids */
+ g_hash_table_remove (tunnels, tunnelid);
+ g_mutex_unlock (&tunnels_lock);
+ }
+
+ gst_rtsp_watch_set_flushing (watch, TRUE);
+ g_mutex_lock (&priv->watch_lock);
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
+ g_mutex_unlock (&priv->watch_lock);
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ gchar *str;
+
+ str = gst_rtsp_strresult (result);
+ GST_INFO ("client %p: received an error %s", client, str);
+ g_free (str);
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+error_full (GstRTSPWatch * watch, GstRTSPResult result,
+ GstRTSPMessage * message, guint id, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ gchar *str;
+ GstRTSPContext sctx = { NULL }, *ctx;
+ GstRTSPClientPrivate *priv;
+ GstRTSPMessage response = { 0 };
+ priv = client->priv;
+
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->request = message;
+ ctx->method = GST_RTSP_INVALID;
+ ctx->response = &response;
+
+ /* only return error response if it is a request */
+ if (!message || message->type != GST_RTSP_MESSAGE_REQUEST)
+ goto done;
+
+ if (result == GST_RTSP_ENOMEM) {
+ send_generic_response (client, GST_RTSP_STS_REQUEST_ENTITY_TOO_LARGE, ctx);
+ goto done;
+ }
+ if (result == GST_RTSP_EPARSE) {
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ goto done;
+ }
+
+done:
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
+ str = gst_rtsp_strresult (result);
+ GST_INFO
+ ("client %p: error when handling message %p with id %d: %s",
+ client, message, id, str);
+ g_free (str);
+
+ return GST_RTSP_OK;
+}
+
+static gboolean
+remember_tunnel (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ const gchar *tunnelid;
+
+ /* store client in the pending tunnels */
+ tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
+ if (tunnelid == NULL)
+ goto no_tunnelid;
+
+ GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
+
+ /* we can't have two clients connecting with the same tunnelid */
+ g_mutex_lock (&tunnels_lock);
+ if (g_hash_table_lookup (tunnels, tunnelid))
+ goto tunnel_existed;
+
+ g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
+ g_mutex_unlock (&tunnels_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+no_tunnelid:
+ {
+ GST_ERROR ("client %p: no tunnelid provided", client);
+ return FALSE;
+ }
+tunnel_existed:
+ {
+ g_mutex_unlock (&tunnels_lock);
+ GST_ERROR ("client %p: tunnel session %s already existed", client,
+ tunnelid);
+ return FALSE;
+ }
+}
+
+static GstRTSPResult
+tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_WARNING ("client %p: tunnel lost (connection %p)", client,
+ priv->connection);
+
+ /* ignore error, it'll only be a problem when the client does a POST again */
+ remember_tunnel (client);
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPStatusCode
+handle_tunnel (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPClient *oclient;
+ GstRTSPClientPrivate *opriv;
+ const gchar *tunnelid;
+
+ tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
+ if (tunnelid == NULL)
+ goto no_tunnelid;
+
+ /* check for previous tunnel */
+ g_mutex_lock (&tunnels_lock);
+ oclient = g_hash_table_lookup (tunnels, tunnelid);
+
+ if (oclient == NULL) {
+ /* no previous tunnel, remember tunnel */
+ g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
+ g_mutex_unlock (&tunnels_lock);
+
+ GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
+ client, priv->connection);
+ } else {
+ /* merge both tunnels into the first client */
+ /* remove the old client from the table. ref before because removing it will
+ * remove the ref to it. */
+ g_object_ref (oclient);
+ g_hash_table_remove (tunnels, tunnelid);
+ g_mutex_unlock (&tunnels_lock);
+
+ opriv = oclient->priv;
+
+ g_mutex_lock (&opriv->watch_lock);
+ if (opriv->watch == NULL)
+ goto tunnel_closed;
+ if (opriv->tstate == priv->tstate)
+ goto tunnel_duplicate_id;
+
+ GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
+ oclient, opriv->connection, priv->connection);
+
+ gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
+ gst_rtsp_watch_reset (priv->watch);
+ gst_rtsp_watch_reset (opriv->watch);
+ g_mutex_unlock (&opriv->watch_lock);
+ g_object_unref (oclient);
+
+ /* the old client owns the tunnel now, the new one will be freed */
+ g_source_destroy ((GSource *) priv->watch);
+ priv->watch = NULL;
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
+ rtsp_ctrl_timeout_remove (client);
+ }
+
+ return GST_RTSP_STS_OK;
+
+ /* ERRORS */
+no_tunnelid:
+ {
+ GST_ERROR ("client %p: no tunnelid provided", client);
+ return GST_RTSP_STS_SERVICE_UNAVAILABLE;
+ }
+tunnel_closed:
+ {
+ GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
+ g_mutex_unlock (&opriv->watch_lock);
+ g_object_unref (oclient);
+ return GST_RTSP_STS_SERVICE_UNAVAILABLE;
+ }
+tunnel_duplicate_id:
+ {
+ GST_ERROR ("client %p: tunnel session %s was duplicate", client, tunnelid);
+ g_mutex_unlock (&opriv->watch_lock);
+ g_object_unref (oclient);
+ return GST_RTSP_STS_BAD_REQUEST;
+ }
+}
+
+static GstRTSPStatusCode
+tunnel_get (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+ GST_INFO ("client %p: tunnel get (connection %p)", client,
+ client->priv->connection);
+
+ g_mutex_lock (&client->priv->lock);
+ client->priv->tstate = TUNNEL_STATE_GET;
+ g_mutex_unlock (&client->priv->lock);
+
+ return handle_tunnel (client);
+}
+
+static GstRTSPResult
+tunnel_post (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+ GST_INFO ("client %p: tunnel post (connection %p)", client,
+ client->priv->connection);
+
+ g_mutex_lock (&client->priv->lock);
+ client->priv->tstate = TUNNEL_STATE_POST;
+ g_mutex_unlock (&client->priv->lock);
+
+ if (handle_tunnel (client) != GST_RTSP_STS_OK)
+ return GST_RTSP_ERROR;
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
+ GstRTSPMessage * response, gpointer user_data)
+{
+ GstRTSPClientClass *klass;
+
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+ if (klass->tunnel_http_response) {
+ klass->tunnel_http_response (client, request, response);
+ }
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPWatchFuncs watch_funcs = {
+ message_received,
+ message_sent,
+ closed,
+ error,
+ tunnel_get,
+ tunnel_post,
+ error_full,
+ tunnel_lost,
+ tunnel_http_response
+};
+
+static void
+client_watch_notify (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ gboolean closed = TRUE;
+
+ GST_INFO ("client %p: watch destroyed", client);
+ priv->watch = NULL;
+ /* remove all sessions if the media says so and so drop the extra client ref */
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
+ rtsp_ctrl_timeout_remove (client);
+ gst_rtsp_client_session_filter (client, cleanup_session, &closed);
+
+ if (closed)
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
+ g_object_unref (client);
+}
+
+/**
+ * gst_rtsp_client_attach:
+ * @client: a #GstRTSPClient
+ * @context: (allow-none): a #GMainContext
+ *
+ * Attaches @client to @context. When the mainloop for @context is run, the
+ * client will be dispatched. When @context is %NULL, the default context will be
+ * used).
+ *
+ * This function should be called when the client properties and urls are fully
+ * configured and the client is ready to start.
+ *
+ * Returns: the ID (greater than 0) for the source within the GMainContext.
+ */
+guint
+gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
+{
+ GstRTSPClientPrivate *priv;
+ GSource *timer_src;
+ guint res;
+ GWeakRef *client_weak_ref = g_new (GWeakRef, 1);
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
+ priv = client->priv;
+ g_return_val_if_fail (priv->connection != NULL, 0);
+ g_return_val_if_fail (priv->watch == NULL, 0);
+ g_return_val_if_fail (priv->watch_context == NULL, 0);
+
+ /* make sure noone will free the context before the watch is destroyed */
+ priv->watch_context = g_main_context_ref (context);
+
+ /* create watch for the connection and attach */
+ priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
+ g_object_ref (client), (GDestroyNotify) client_watch_notify);
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ gst_rtsp_client_set_send_messages_func (client, do_send_messages, priv->watch,
+ (GDestroyNotify) gst_rtsp_watch_unref);
+
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
+
+ GST_INFO ("client %p: attaching to context %p", client, context);
+ res = gst_rtsp_watch_attach (priv->watch, context);
+
+ /* Setting up a timeout for the RTSP control channel until a session
+ * is up where it is handling timeouts. */
+ g_mutex_lock (&priv->lock);
+
+ /* remove old timeout if any */
+ rtsp_ctrl_timeout_remove_unlocked (client->priv);
+
+ timer_src = g_timeout_source_new_seconds (RTSP_CTRL_CB_INTERVAL);
+ g_weak_ref_init (client_weak_ref, client);
+ g_source_set_callback (timer_src, rtsp_ctrl_timeout_cb, client_weak_ref,
+ rtsp_ctrl_timeout_destroy_notify);
+ g_source_attach (timer_src, priv->watch_context);
+ priv->rtsp_ctrl_timeout = timer_src;
+ GST_DEBUG ("rtsp control setting up session timeout %p.",
+ priv->rtsp_ctrl_timeout);
+
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_client_session_filter:
+ * @client: a #GstRTSPClient
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: user data passed to @func
+ *
+ * Call @func for each session managed by @client. The result value of @func
+ * determines what happens to the session. @func will be called with @client
+ * locked so no further actions on @client can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
+ * @client.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
+ *
+ * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
+ * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_client_session_filter (GstRTSPClient * client,
+ GstRTSPClientSessionFilterFunc func, gpointer user_data)
+{
+ GstRTSPClientPrivate *priv;
+ GList *result, *walk, *next;
+ GHashTable *visited;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ g_mutex_lock (&priv->lock);
+restart:
+ cookie = priv->sessions_cookie;
+ for (walk = priv->sessions; walk; walk = next) {
+ GstRTSPSession *sess = walk->data;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ next = g_list_next (walk);
+
+ if (func) {
+ /* only visit each session once */
+ if (g_hash_table_contains (visited, sess))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (sess));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (client, sess, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->sessions_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ /* stop watching the session and pretend it went away, if the list was
+ * changed, we can't use the current list position, try to see if we
+ * still have the session */
+ client_unwatch_session (client, sess, changed ? NULL : walk);
+ cookie = priv->sessions_cookie;
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (sess));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspconnection.h>
+
+#ifndef __GST_RTSP_CLIENT_H__
+#define __GST_RTSP_CLIENT_H__
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPClient GstRTSPClient;
+typedef struct _GstRTSPClientClass GstRTSPClientClass;
+typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate;
+
+#include "rtsp-server-prelude.h"
+#include "rtsp-context.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-sdp.h"
+#include "rtsp-auth.h"
+
+#define GST_TYPE_RTSP_CLIENT (gst_rtsp_client_get_type ())
+#define GST_IS_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CLIENT))
+#define GST_IS_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CLIENT))
+#define GST_RTSP_CLIENT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
+#define GST_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClient))
+#define GST_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
+#define GST_RTSP_CLIENT_CAST(obj) ((GstRTSPClient*)(obj))
+#define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass))
+
+/**
+ * GstRTSPClientSendFunc:
+ * @client: a #GstRTSPClient
+ * @message: a #GstRTSPMessage
+ * @close: close the connection
+ * @user_data: user data when registering the callback
+ *
+ * This callback is called when @client wants to send @message. When @close is
+ * %TRUE, the connection should be closed when the message has been sent.
+ *
+ * Returns: %TRUE on success.
+ */
+typedef gboolean (*GstRTSPClientSendFunc) (GstRTSPClient *client,
+ GstRTSPMessage *message,
+ gboolean close,
+ gpointer user_data);
+
+/**
+ * GstRTSPClientSendMessagesFunc:
+ * @client: a #GstRTSPClient
+ * @messages: #GstRTSPMessage
+ * @n_messages: number of messages
+ * @close: close the connection
+ * @user_data: user data when registering the callback
+ *
+ * This callback is called when @client wants to send @messages. When @close is
+ * %TRUE, the connection should be closed when the message has been sent.
+ *
+ * Returns: %TRUE on success.
+ *
+ * Since: 1.16
+ */
+typedef gboolean (*GstRTSPClientSendMessagesFunc) (GstRTSPClient *client,
+ GstRTSPMessage *messages,
+ guint n_messages,
+ gboolean close,
+ gpointer user_data);
+
+/**
+ * GstRTSPClient:
+ *
+ * The client object represents the connection and its state with a client.
+ */
+struct _GstRTSPClient {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPClientPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPClientClass:
+ * @create_sdp: called when the SDP needs to be created for media.
+ * @configure_client_media: called when the stream in media needs to be configured.
+ * The default implementation will configure the blocksize on the payloader when
+ * spcified in the request headers.
+ * @configure_client_transport: called when the client transport needs to be
+ * configured.
+ * @params_set: set parameters. This function should also initialize the
+ * RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
+ * @params_get: get parameters. This function should also initialize the
+ * RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
+ * @make_path_from_uri: called to create path from uri.
+ * @adjust_play_mode: called to give the application the possibility to adjust
+ * the range, seek flags, rate and rate-control. Since 1.18
+ * @adjust_play_response: called to give the implementation the possibility to
+ * adjust the response to a play request, for example if extra headers were
+ * parsed when #GstRTSPClientClass.adjust_play_mode was called. Since 1.18
+ * @tunnel_http_response: called when a response to the GET request is about to
+ * be sent for a tunneled connection. The response can be modified. Since: 1.4
+ *
+ * The client class structure.
+ */
+struct _GstRTSPClientClass {
+ GObjectClass parent_class;
+
+ GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
+ gboolean (*configure_client_media) (GstRTSPClient * client,
+ GstRTSPMedia * media, GstRTSPStream * stream,
+ GstRTSPContext * ctx);
+ gboolean (*configure_client_transport) (GstRTSPClient * client,
+ GstRTSPContext * ctx,
+ GstRTSPTransport * ct);
+ GstRTSPResult (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPResult (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
+ gchar * (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri);
+ GstRTSPStatusCode (*adjust_play_mode) (GstRTSPClient * client,
+ GstRTSPContext * context,
+ GstRTSPTimeRange ** range,
+ GstSeekFlags * flags,
+ gdouble * rate,
+ GstClockTime * trickmode_interval,
+ gboolean * enable_rate_control);
+ GstRTSPStatusCode (*adjust_play_response) (GstRTSPClient * client,
+ GstRTSPContext * context);
+
+ /* signals */
+ void (*closed) (GstRTSPClient *client);
+ void (*new_session) (GstRTSPClient *client, GstRTSPSession *session);
+ void (*options_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*describe_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*setup_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*play_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*pause_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*handle_response) (GstRTSPClient *client, GstRTSPContext *ctx);
+
+ void (*tunnel_http_response) (GstRTSPClient * client, GstRTSPMessage * request,
+ GstRTSPMessage * response);
+ void (*send_message) (GstRTSPClient * client, GstRTSPContext *ctx,
+ GstRTSPMessage * response);
+
+ gboolean (*handle_sdp) (GstRTSPClient *client, GstRTSPContext *ctx, GstRTSPMedia *media, GstSDPMessage *sdp);
+
+ void (*announce_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*record_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ gchar* (*check_requirements) (GstRTSPClient *client, GstRTSPContext *ctx, gchar ** arr);
+
+ GstRTSPStatusCode (*pre_options_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPStatusCode (*pre_describe_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPStatusCode (*pre_setup_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPStatusCode (*pre_play_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPStatusCode (*pre_pause_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPStatusCode (*pre_teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPStatusCode (*pre_set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPStatusCode (*pre_get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPStatusCode (*pre_announce_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPStatusCode (*pre_record_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE-18];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_client_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPClient * gst_rtsp_client_new (void);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_client_set_session_pool (GstRTSPClient *client,
+ GstRTSPSessionPool *pool);
+
+GST_RTSP_SERVER_API
+GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_client_set_mount_points (GstRTSPClient *client,
+ GstRTSPMountPoints *mounts);
+
+GST_RTSP_SERVER_API
+GstRTSPMountPoints * gst_rtsp_client_get_mount_points (GstRTSPClient *client);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_client_set_content_length_limit (GstRTSPClient *client, guint limit);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_client_get_content_length_limit (GstRTSPClient *client);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth);
+
+GST_RTSP_SERVER_API
+GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_client_set_thread_pool (GstRTSPClient *client, GstRTSPThreadPool *pool);
+
+GST_RTSP_SERVER_API
+GstRTSPThreadPool * gst_rtsp_client_get_thread_pool (GstRTSPClient *client);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_client_set_connection (GstRTSPClient *client, GstRTSPConnection *conn);
+
+GST_RTSP_SERVER_API
+GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient *client);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_client_attach (GstRTSPClient *client,
+ GMainContext *context);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_client_close (GstRTSPClient * client);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_client_set_send_func (GstRTSPClient *client,
+ GstRTSPClientSendFunc func,
+ gpointer user_data,
+ GDestroyNotify notify);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_client_set_send_messages_func (GstRTSPClient *client,
+ GstRTSPClientSendMessagesFunc func,
+ gpointer user_data,
+ GDestroyNotify notify);
+
+GST_RTSP_SERVER_API
+GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client,
+ GstRTSPMessage *message);
+
+GST_RTSP_SERVER_API
+GstRTSPResult gst_rtsp_client_send_message (GstRTSPClient * client,
+ GstRTSPSession *session,
+ GstRTSPMessage *message);
+/**
+ * GstRTSPClientSessionFilterFunc:
+ * @client: a #GstRTSPClient object
+ * @sess: a #GstRTSPSession in @client
+ * @user_data: user data that has been given to gst_rtsp_client_session_filter()
+ *
+ * This function will be called by the gst_rtsp_client_session_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed
+ * from @client.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in
+ * @client.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of
+ * gst_rtsp_client_session_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPClientSessionFilterFunc) (GstRTSPClient *client,
+ GstRTSPSession *sess,
+ gpointer user_data);
+
+GST_RTSP_SERVER_API
+GList * gst_rtsp_client_session_filter (GstRTSPClient *client,
+ GstRTSPClientSessionFilterFunc func,
+ gpointer user_data);
+
+GST_RTSP_SERVER_API
+GstRTSPStreamTransport * gst_rtsp_client_get_stream_transport (GstRTSPClient *client,
+ guint8 channel);
+
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPClient, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_CLIENT_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-context
+ * @short_description: A client request context
+ * @see_also: #GstRTSPServer, #GstRTSPClient
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "rtsp-context.h"
+
+G_DEFINE_POINTER_TYPE (GstRTSPContext, gst_rtsp_context);
+
+static GPrivate current_context;
+
+/**
+ * gst_rtsp_context_get_current: (skip):
+ *
+ * Get the current #GstRTSPContext. This object is retrieved from the
+ * current thread that is handling the request for a client.
+ *
+ * Returns: a #GstRTSPContext
+ */
+GstRTSPContext *
+gst_rtsp_context_get_current (void)
+{
+ GSList *l;
+
+ l = g_private_get (¤t_context);
+ if (l == NULL)
+ return NULL;
+
+ return (GstRTSPContext *) (l->data);
+
+}
+
+/**
+ * gst_rtsp_context_push_current:
+ * @ctx: a #GstRTSPContext
+ *
+ * Pushes @ctx onto the context stack. The current
+ * context can then be received using gst_rtsp_context_get_current().
+ **/
+void
+gst_rtsp_context_push_current (GstRTSPContext * ctx)
+{
+ GSList *l;
+
+ g_return_if_fail (ctx != NULL);
+
+ l = g_private_get (¤t_context);
+ l = g_slist_prepend (l, ctx);
+ g_private_set (¤t_context, l);
+}
+
+/**
+ * gst_rtsp_context_pop_current:
+ * @ctx: a #GstRTSPContext
+ *
+ * Pops @ctx off the context stack (verifying that @ctx
+ * is on the top of the stack).
+ **/
+void
+gst_rtsp_context_pop_current (GstRTSPContext * ctx)
+{
+ GSList *l;
+
+ l = g_private_get (¤t_context);
+
+ g_return_if_fail (l != NULL);
+ g_return_if_fail (l->data == ctx);
+
+ l = g_slist_delete_link (l, l);
+ g_private_set (¤t_context, l);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspconnection.h>
+
+#ifndef __GST_RTSP_CONTEXT_H__
+#define __GST_RTSP_CONTEXT_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_CONTEXT (gst_rtsp_context_get_type ())
+
+typedef struct _GstRTSPContext GstRTSPContext;
+
+#include "rtsp-server-prelude.h"
+#include "rtsp-server-object.h"
+#include "rtsp-media.h"
+#include "rtsp-media-factory.h"
+#include "rtsp-session-media.h"
+#include "rtsp-auth.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-token.h"
+
+/**
+ * GstRTSPContext:
+ * @server: the server
+ * @conn: the connection
+ * @client: the client
+ * @request: the complete request
+ * @uri: the complete url parsed from @request
+ * @method: the parsed method of @uri
+ * @auth: the current auth object or %NULL
+ * @token: authorisation token
+ * @session: the session, can be %NULL
+ * @sessmedia: the session media for the url can be %NULL
+ * @factory: the media factory for the url, can be %NULL
+ * @media: the media for the url can be %NULL
+ * @stream: the stream for the url can be %NULL
+ * @response: the response
+ * @trans: the stream transport, can be %NULL
+ *
+ * Information passed around containing the context of a request.
+ */
+struct _GstRTSPContext {
+ GstRTSPServer *server;
+ GstRTSPConnection *conn;
+ GstRTSPClient *client;
+ GstRTSPMessage *request;
+ GstRTSPUrl *uri;
+ GstRTSPMethod method;
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ GstRTSPSession *session;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPStream *stream;
+ GstRTSPMessage *response;
+ GstRTSPStreamTransport *trans;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING - 1];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_context_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPContext * gst_rtsp_context_get_current (void);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_context_push_current (GstRTSPContext * ctx);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_context_pop_current (GstRTSPContext * ctx);
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_CONTEXT_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2018 Ognyan Tonchev <ognyan@axis.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include "rtsp-latency-bin.h"
+
+struct _GstRTSPLatencyBinPrivate
+{
+ GstPad *sinkpad;
+ GstElement *element;
+};
+
+enum
+{
+ PROP_0,
+ PROP_ELEMENT,
+ PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_latency_bin_debug);
+#define GST_CAT_DEFAULT rtsp_latency_bin_debug
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS_ANY);
+
+static void gst_rtsp_latency_bin_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_latency_bin_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static gboolean gst_rtsp_latency_bin_element_query (GstElement * element,
+ GstQuery * query);
+static gboolean gst_rtsp_latency_bin_element_event (GstElement * element,
+ GstEvent * event);
+static void gst_rtsp_latency_bin_message_handler (GstBin * bin,
+ GstMessage * message);
+static gboolean gst_rtsp_latency_bin_add_element (GstRTSPLatencyBin *
+ latency_bin, GstElement * element);
+static GstStateChangeReturn gst_rtsp_latency_bin_change_state (GstElement *
+ element, GstStateChange transition);
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPLatencyBin, gst_rtsp_latency_bin,
+ GST_TYPE_BIN);
+
+static void
+gst_rtsp_latency_bin_class_init (GstRTSPLatencyBinClass * klass)
+{
+ GObjectClass *gobject_klass = G_OBJECT_CLASS (klass);
+ GstElementClass *gstelement_klass = GST_ELEMENT_CLASS (klass);
+ GstBinClass *gstbin_klass = GST_BIN_CLASS (klass);
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_latency_bin_debug,
+ "rtsplatencybin", 0, "GstRTSPLatencyBin");
+
+ gobject_klass->get_property = gst_rtsp_latency_bin_get_property;
+ gobject_klass->set_property = gst_rtsp_latency_bin_set_property;
+
+ g_object_class_install_property (gobject_klass, PROP_ELEMENT,
+ g_param_spec_object ("element", "The Element",
+ "The GstElement to prevent from affecting piplines latency",
+ GST_TYPE_ELEMENT, G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
+
+ gstelement_klass->change_state =
+ GST_DEBUG_FUNCPTR (gst_rtsp_latency_bin_change_state);
+ gstelement_klass->query =
+ GST_DEBUG_FUNCPTR (gst_rtsp_latency_bin_element_query);
+ gstelement_klass->send_event =
+ GST_DEBUG_FUNCPTR (gst_rtsp_latency_bin_element_event);
+
+ gstbin_klass->handle_message =
+ GST_DEBUG_FUNCPTR (gst_rtsp_latency_bin_message_handler);
+}
+
+static void
+gst_rtsp_latency_bin_init (GstRTSPLatencyBin * latency_bin)
+{
+ GST_OBJECT_FLAG_SET (latency_bin, GST_ELEMENT_FLAG_SINK);
+}
+
+static void
+gst_rtsp_latency_bin_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPLatencyBin *latency_bin = GST_RTSP_LATENCY_BIN (object);
+ GstRTSPLatencyBinPrivate *priv =
+ gst_rtsp_latency_bin_get_instance_private (latency_bin);
+
+ switch (propid) {
+ case PROP_ELEMENT:
+ g_value_set_object (value, priv->element);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_latency_bin_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPLatencyBin *latency_bin = GST_RTSP_LATENCY_BIN (object);
+
+ switch (propid) {
+ case PROP_ELEMENT:
+ if (!gst_rtsp_latency_bin_add_element (latency_bin,
+ g_value_get_object (value))) {
+ GST_WARNING_OBJECT (latency_bin, "Could not add the element");
+ }
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static gboolean
+gst_rtsp_latency_bin_add_element (GstRTSPLatencyBin * latency_bin,
+ GstElement * element)
+{
+ GstRTSPLatencyBinPrivate *priv =
+ gst_rtsp_latency_bin_get_instance_private (latency_bin);
+ GstPad *pad;
+ GstPadTemplate *templ;
+
+ GST_DEBUG_OBJECT (latency_bin, "Adding element to latencybin : %s",
+ GST_ELEMENT_NAME (element));
+
+ if (!element) {
+ goto no_element;
+ }
+
+ /* add the element to ourself */
+ gst_object_ref (element);
+ gst_bin_add (GST_BIN (latency_bin), element);
+ priv->element = element;
+
+ /* add ghost pad first */
+ templ = gst_static_pad_template_get (&sinktemplate);
+ priv->sinkpad = gst_ghost_pad_new_no_target_from_template ("sink", templ);
+ gst_object_unref (templ);
+ g_assert (priv->sinkpad);
+
+ gst_element_add_pad (GST_ELEMENT (latency_bin), priv->sinkpad);
+
+ /* and link it to our element */
+ pad = gst_element_get_static_pad (element, "sink");
+ if (!pad) {
+ goto no_sink_pad;
+ }
+
+ if (!gst_ghost_pad_set_target (GST_GHOST_PAD_CAST (priv->sinkpad), pad)) {
+ goto set_target_failed;
+ }
+
+ gst_object_unref (pad);
+
+ return TRUE;
+
+ /* ERRORs */
+no_element:
+ {
+ GST_WARNING_OBJECT (latency_bin, "No element, not adding");
+ return FALSE;
+ }
+no_sink_pad:
+ {
+ GST_WARNING_OBJECT (latency_bin, "The element has no sink pad");
+ return FALSE;
+ }
+set_target_failed:
+ {
+ GST_WARNING_OBJECT (latency_bin, "Could not set target pad");
+ gst_object_unref (pad);
+ return FALSE;
+ }
+}
+
+
+static gboolean
+gst_rtsp_latency_bin_element_query (GstElement * element, GstQuery * query)
+{
+ gboolean ret = TRUE;
+
+ GST_LOG_OBJECT (element, "got query %s", GST_QUERY_TYPE_NAME (query));
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_LATENCY:
+ /* ignoring latency query, we do not want our element to affect latency on
+ * the rest of the pipeline */
+ GST_DEBUG_OBJECT (element, "ignoring latency query");
+ gst_query_set_latency (query, FALSE, 0, -1);
+ break;
+ default:
+ ret =
+ GST_ELEMENT_CLASS (gst_rtsp_latency_bin_parent_class)->query
+ (GST_ELEMENT (element), query);
+ break;
+ }
+
+ return ret;
+}
+
+static gboolean
+gst_rtsp_latency_bin_element_event (GstElement * element, GstEvent * event)
+{
+ gboolean ret = TRUE;
+
+ GST_LOG_OBJECT (element, "got event %s", GST_EVENT_TYPE_NAME (event));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_LATENCY:
+ /* ignoring latency event, we will configure latency on our element when
+ * going to PLAYING */
+ GST_DEBUG_OBJECT (element, "ignoring latency event");
+ gst_event_unref (event);
+ break;
+ default:
+ ret =
+ GST_ELEMENT_CLASS (gst_rtsp_latency_bin_parent_class)->send_event
+ (GST_ELEMENT (element), event);
+ break;
+ }
+
+ return ret;
+}
+
+static gboolean
+gst_rtsp_latency_bin_recalculate_latency (GstRTSPLatencyBin * latency_bin)
+{
+ GstRTSPLatencyBinPrivate *priv =
+ gst_rtsp_latency_bin_get_instance_private (latency_bin);
+ GstEvent *latency;
+ GstQuery *query;
+ GstClockTime min_latency;
+
+ GST_DEBUG_OBJECT (latency_bin, "Recalculating latency");
+
+ if (!priv->element) {
+ GST_WARNING_OBJECT (latency_bin, "We do not have an element");
+ return FALSE;
+ }
+
+ query = gst_query_new_latency ();
+
+ if (!gst_element_query (priv->element, query)) {
+ GST_WARNING_OBJECT (latency_bin, "Latency query failed");
+ gst_query_unref (query);
+ return FALSE;
+ }
+
+ gst_query_parse_latency (query, NULL, &min_latency, NULL);
+ gst_query_unref (query);
+
+ GST_LOG_OBJECT (latency_bin, "Got min_latency from stream: %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (min_latency));
+
+ latency = gst_event_new_latency (min_latency);
+ if (!gst_element_send_event (priv->element, latency)) {
+ GST_WARNING_OBJECT (latency_bin, "Sending latency event to stream failed");
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+static void
+gst_rtsp_latency_bin_message_handler (GstBin * bin, GstMessage * message)
+{
+ GstRTSPLatencyBin *latency_bin = GST_RTSP_LATENCY_BIN (bin);
+
+ GST_LOG_OBJECT (bin, "Got message %s", GST_MESSAGE_TYPE_NAME (message));
+
+ switch (GST_MESSAGE_TYPE (message)) {
+ case GST_MESSAGE_LATENCY:{
+ if (!gst_rtsp_latency_bin_recalculate_latency (latency_bin)) {
+ GST_WARNING_OBJECT (latency_bin, "Could not recalculate latency");
+ }
+ break;
+ }
+ default:
+ GST_BIN_CLASS (gst_rtsp_latency_bin_parent_class)->handle_message (bin,
+ message);
+ break;
+ }
+}
+
+static GstStateChangeReturn
+gst_rtsp_latency_bin_change_state (GstElement * element, GstStateChange
+ transition)
+{
+ GstRTSPLatencyBin *latency_bin = GST_RTSP_LATENCY_BIN (element);
+ GstStateChangeReturn ret;
+
+ GST_LOG_OBJECT (latency_bin, "Changing state %s",
+ gst_state_change_get_name (transition));
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ case GST_STATE_CHANGE_PLAYING_TO_PLAYING:
+ if (!gst_rtsp_latency_bin_recalculate_latency (latency_bin)) {
+ GST_WARNING_OBJECT (latency_bin, "Could not recalculate latency");
+ }
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (gst_rtsp_latency_bin_parent_class)->change_state
+ (element, transition);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_latency_bin_new:
+ * @element: (transfer full): a #GstElement
+ *
+ * Create a bin that encapsulates an @element and prevents it from affecting
+ * latency on the whole pipeline.
+ *
+ * Returns: A newly created #GstRTSPLatencyBin element, or %NULL on failure
+ */
+GstElement *
+gst_rtsp_latency_bin_new (GstElement * element)
+{
+ GstElement *gst_rtsp_latency_bin;
+
+ g_return_val_if_fail (element, NULL);
+
+ gst_rtsp_latency_bin = g_object_new (GST_RTSP_LATENCY_BIN_TYPE, "element",
+ element, NULL);
+ gst_object_unref (element);
+ return gst_rtsp_latency_bin;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2018 Ognyan Tonchev <ognyan@axis.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_LATENCY_BIN_H__
+#define __GST_RTSP_LATENCY_BIN_H__
+
+#include <gst/gst.h>
+#include "rtsp-server-prelude.h"
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPLatencyBin GstRTSPLatencyBin;
+typedef struct _GstRTSPLatencyBinClass GstRTSPLatencyBinClass;
+typedef struct _GstRTSPLatencyBinPrivate GstRTSPLatencyBinPrivate;
+
+#define GST_RTSP_LATENCY_BIN_TYPE (gst_rtsp_latency_bin_get_type ())
+#define IS_GST_RTSP_LATENCY_BIN(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_RTSP_LATENCY_BIN_TYPE))
+#define IS_GST_RTSP_LATENCY_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_RTSP_LATENCY_BIN_TYPE))
+#define GST_RTSP_LATENCY_BIN_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_RTSP_LATENCY_BIN_TYPE, GstRTSPLatencyBinClass))
+#define GST_RTSP_LATENCY_BIN(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_RTSP_LATENCY_BIN_TYPE, GstRTSPLatencyBin))
+#define GST_RTSP_LATENCY_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_RTSP_LATENCY_BIN_TYPE, GstRTSPLatencyBinClass))
+#define GST_RTSP_LATENCY_BIN_CAST(obj) ((GstRTSPLatencyBin*)(obj))
+#define GST_RTSP_LATENCY_BIN_CLASS_CAST(klass) ((GstRTSPLatencyBinClass*)(klass))
+
+struct _GstRTSPLatencyBin {
+ GstBin parent;
+
+ GstRTSPLatencyBinPrivate *priv;
+};
+
+struct _GstRTSPLatencyBinClass {
+ GstBinClass parent_class;
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_latency_bin_get_type (void);
+
+GST_RTSP_SERVER_API
+GstElement * gst_rtsp_latency_bin_new (GstElement * element);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_LATENCY_BIN_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media-factory-uri
+ * @short_description: A factory for URI sources
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPMedia
+ *
+ * This specialized #GstRTSPMediaFactory constructs media pipelines from a URI,
+ * given with gst_rtsp_media_factory_uri_set_uri().
+ *
+ * It will automatically demux and payload the different streams found in the
+ * media at URL.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-media-factory-uri.h"
+
+struct _GstRTSPMediaFactoryURIPrivate
+{
+ GMutex lock;
+ gchar *uri; /* protected by lock */
+ gboolean use_gstpay;
+
+ GstCaps *raw_vcaps;
+ GstCaps *raw_acaps;
+ GList *demuxers;
+ GList *payloaders;
+ GList *decoders;
+};
+
+#define DEFAULT_URI NULL
+#define DEFAULT_USE_GSTPAY FALSE
+
+enum
+{
+ PROP_0,
+ PROP_URI,
+ PROP_USE_GSTPAY,
+ PROP_LAST
+};
+
+
+#define RAW_VIDEO_CAPS \
+ "video/x-raw"
+
+#define RAW_AUDIO_CAPS \
+ "audio/x-raw"
+
+static GstStaticCaps raw_video_caps = GST_STATIC_CAPS (RAW_VIDEO_CAPS);
+static GstStaticCaps raw_audio_caps = GST_STATIC_CAPS (RAW_AUDIO_CAPS);
+
+typedef struct
+{
+ GstRTSPMediaFactoryURI *factory;
+ guint pt;
+} FactoryData;
+
+static void
+free_data (FactoryData * data)
+{
+ g_object_unref (data->factory);
+ g_free (data);
+}
+
+static const gchar *factory_key = "GstRTSPMediaFactoryURI";
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_factory_uri_debug);
+#define GST_CAT_DEFAULT rtsp_media_factory_uri_debug
+
+static void gst_rtsp_media_factory_uri_get_property (GObject * object,
+ guint propid, GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_uri_set_property (GObject * object,
+ guint propid, const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_uri_finalize (GObject * obj);
+
+static GstElement *rtsp_media_factory_uri_create_element (GstRTSPMediaFactory *
+ factory, const GstRTSPUrl * url);
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPMediaFactoryURI, gst_rtsp_media_factory_uri,
+ GST_TYPE_RTSP_MEDIA_FACTORY);
+
+static void
+gst_rtsp_media_factory_uri_class_init (GstRTSPMediaFactoryURIClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstRTSPMediaFactoryClass *mediafactory_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ mediafactory_class = GST_RTSP_MEDIA_FACTORY_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_media_factory_uri_get_property;
+ gobject_class->set_property = gst_rtsp_media_factory_uri_set_property;
+ gobject_class->finalize = gst_rtsp_media_factory_uri_finalize;
+
+ /**
+ * GstRTSPMediaFactoryURI::uri:
+ *
+ * The uri of the resource that will be served by this factory.
+ */
+ g_object_class_install_property (gobject_class, PROP_URI,
+ g_param_spec_string ("uri", "URI",
+ "The URI of the resource to stream", DEFAULT_URI,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPMediaFactoryURI::use-gstpay:
+ *
+ * Allow the usage of gstpay in order to avoid decoding of compressed formats
+ * without a payloader.
+ */
+ g_object_class_install_property (gobject_class, PROP_USE_GSTPAY,
+ g_param_spec_boolean ("use-gstpay", "Use gstpay",
+ "Use the gstpay payloader to avoid decoding", DEFAULT_USE_GSTPAY,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ mediafactory_class->create_element = rtsp_media_factory_uri_create_element;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_media_factory_uri_debug, "rtspmediafactoryuri",
+ 0, "GstRTSPMediaFactoryUri");
+}
+
+typedef struct
+{
+ GList *demux;
+ GList *payload;
+ GList *decode;
+} FilterData;
+
+static gboolean
+payloader_filter (GstPluginFeature * feature, FilterData * data)
+{
+ const gchar *klass;
+ GstElementFactory *fact;
+ GList **list = NULL;
+
+ /* we only care about element factories */
+ if (G_UNLIKELY (!GST_IS_ELEMENT_FACTORY (feature)))
+ return FALSE;
+
+ if (gst_plugin_feature_get_rank (feature) < GST_RANK_MARGINAL)
+ return FALSE;
+
+ fact = GST_ELEMENT_FACTORY_CAST (feature);
+
+ klass = gst_element_factory_get_metadata (fact, GST_ELEMENT_METADATA_KLASS);
+
+ if (strstr (klass, "Decoder"))
+ list = &data->decode;
+ else if (strstr (klass, "Demux"))
+ list = &data->demux;
+ else if (strstr (klass, "Parser") && strstr (klass, "Codec"))
+ list = &data->demux;
+ else if (strstr (klass, "Payloader") && strstr (klass, "RTP"))
+ list = &data->payload;
+
+ if (list) {
+ GST_DEBUG ("adding %s", GST_OBJECT_NAME (fact));
+ *list = g_list_prepend (*list, gst_object_ref (fact));
+ }
+
+ return FALSE;
+}
+
+static void
+gst_rtsp_media_factory_uri_init (GstRTSPMediaFactoryURI * factory)
+{
+ GstRTSPMediaFactoryURIPrivate *priv =
+ gst_rtsp_media_factory_uri_get_instance_private (factory);
+ FilterData data = { NULL, NULL, NULL };
+
+ GST_DEBUG_OBJECT (factory, "new");
+
+ factory->priv = priv;
+
+ priv->uri = g_strdup (DEFAULT_URI);
+ priv->use_gstpay = DEFAULT_USE_GSTPAY;
+ g_mutex_init (&priv->lock);
+
+ /* get the feature list using the filter */
+ gst_registry_feature_filter (gst_registry_get (), (GstPluginFeatureFilter)
+ payloader_filter, FALSE, &data);
+ /* sort */
+ priv->demuxers =
+ g_list_sort (data.demux, gst_plugin_feature_rank_compare_func);
+ priv->payloaders =
+ g_list_sort (data.payload, gst_plugin_feature_rank_compare_func);
+ priv->decoders =
+ g_list_sort (data.decode, gst_plugin_feature_rank_compare_func);
+
+ priv->raw_vcaps = gst_static_caps_get (&raw_video_caps);
+ priv->raw_acaps = gst_static_caps_get (&raw_audio_caps);
+}
+
+static void
+gst_rtsp_media_factory_uri_finalize (GObject * obj)
+{
+ GstRTSPMediaFactoryURI *factory = GST_RTSP_MEDIA_FACTORY_URI (obj);
+ GstRTSPMediaFactoryURIPrivate *priv = factory->priv;
+
+ GST_DEBUG_OBJECT (factory, "finalize");
+
+ g_free (priv->uri);
+ gst_plugin_feature_list_free (priv->demuxers);
+ gst_plugin_feature_list_free (priv->payloaders);
+ gst_plugin_feature_list_free (priv->decoders);
+ gst_caps_unref (priv->raw_vcaps);
+ gst_caps_unref (priv->raw_acaps);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_media_factory_uri_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_media_factory_uri_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMediaFactoryURI *factory = GST_RTSP_MEDIA_FACTORY_URI (object);
+ GstRTSPMediaFactoryURIPrivate *priv = factory->priv;
+
+ switch (propid) {
+ case PROP_URI:
+ g_value_take_string (value, gst_rtsp_media_factory_uri_get_uri (factory));
+ break;
+ case PROP_USE_GSTPAY:
+ g_value_set_boolean (value, priv->use_gstpay);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_media_factory_uri_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMediaFactoryURI *factory = GST_RTSP_MEDIA_FACTORY_URI (object);
+ GstRTSPMediaFactoryURIPrivate *priv = factory->priv;
+
+ switch (propid) {
+ case PROP_URI:
+ gst_rtsp_media_factory_uri_set_uri (factory, g_value_get_string (value));
+ break;
+ case PROP_USE_GSTPAY:
+ priv->use_gstpay = g_value_get_boolean (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_media_factory_uri_new:
+ *
+ * Create a new #GstRTSPMediaFactoryURI instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPMediaFactoryURI object.
+ */
+GstRTSPMediaFactoryURI *
+gst_rtsp_media_factory_uri_new (void)
+{
+ GstRTSPMediaFactoryURI *result;
+
+ result = g_object_new (GST_TYPE_RTSP_MEDIA_FACTORY_URI, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_uri_set_uri:
+ * @factory: a #GstRTSPMediaFactory
+ * @uri: the uri the stream
+ *
+ * Set the URI of the resource that will be streamed by this factory.
+ */
+void
+gst_rtsp_media_factory_uri_set_uri (GstRTSPMediaFactoryURI * factory,
+ const gchar * uri)
+{
+ GstRTSPMediaFactoryURIPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY_URI (factory));
+ g_return_if_fail (uri != NULL);
+
+ priv = factory->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_free (priv->uri);
+ priv->uri = g_strdup (uri);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_factory_uri_get_uri:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the URI that will provide media for this factory.
+ *
+ * Returns: (transfer full): the configured URI. g_free() after usage.
+ */
+gchar *
+gst_rtsp_media_factory_uri_get_uri (GstRTSPMediaFactoryURI * factory)
+{
+ GstRTSPMediaFactoryURIPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY_URI (factory), NULL);
+
+ priv = factory->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = g_strdup (priv->uri);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+static GstElementFactory *
+find_payloader (GstRTSPMediaFactoryURI * urifact, GstCaps * caps)
+{
+ GstRTSPMediaFactoryURIPrivate *priv = urifact->priv;
+ GList *list;
+ GstElementFactory *factory = NULL;
+ gboolean autoplug_more = FALSE;
+
+ /* first find a demuxer that can link */
+ list = gst_element_factory_list_filter (priv->demuxers, caps,
+ GST_PAD_SINK, FALSE);
+
+ if (list) {
+ GstStructure *structure = gst_caps_get_structure (caps, 0);
+ gboolean parsed = FALSE;
+ gint mpegversion = 0;
+
+ if (!gst_structure_get_boolean (structure, "parsed", &parsed) &&
+ gst_structure_has_name (structure, "audio/mpeg") &&
+ gst_structure_get_int (structure, "mpegversion", &mpegversion) &&
+ (mpegversion == 2 || mpegversion == 4)) {
+ /* for AAC it's framed=true instead of parsed=true */
+ gst_structure_get_boolean (structure, "framed", &parsed);
+ }
+
+ /* Avoid plugging parsers in a loop. This is not 100% correct, as some
+ * parsers don't set parsed=true in caps. We should do something like
+ * decodebin does and track decode chains and elements plugged in those
+ * chains...
+ */
+ if (parsed) {
+ GList *walk;
+ const gchar *klass;
+
+ for (walk = list; walk; walk = walk->next) {
+ factory = GST_ELEMENT_FACTORY (walk->data);
+ klass = gst_element_factory_get_metadata (factory,
+ GST_ELEMENT_METADATA_KLASS);
+ if (strstr (klass, "Parser"))
+ /* caps have parsed=true, so skip this parser to avoid loops */
+ continue;
+
+ autoplug_more = TRUE;
+ break;
+ }
+ } else {
+ /* caps don't have parsed=true set and we have a demuxer/parser */
+ autoplug_more = TRUE;
+ }
+
+ gst_plugin_feature_list_free (list);
+ }
+
+ if (autoplug_more)
+ /* we have a demuxer, try that one first */
+ return NULL;
+
+ /* no demuxer try a depayloader */
+ list = gst_element_factory_list_filter (priv->payloaders, caps,
+ GST_PAD_SINK, FALSE);
+
+ if (list == NULL) {
+ if (priv->use_gstpay) {
+ /* no depayloader or parser/demuxer, use gstpay when allowed */
+ factory = gst_element_factory_find ("rtpgstpay");
+ } else {
+ /* no depayloader, try a decoder, we'll get to a payloader for a decoded
+ * video or audio format, worst case. */
+ list = gst_element_factory_list_filter (priv->decoders, caps,
+ GST_PAD_SINK, FALSE);
+
+ if (list != NULL) {
+ /* we have a decoder, try that one first */
+ gst_plugin_feature_list_free (list);
+ return NULL;
+ }
+ }
+ }
+
+ if (list != NULL) {
+ factory = GST_ELEMENT_FACTORY_CAST (list->data);
+ g_object_ref (factory);
+ gst_plugin_feature_list_free (list);
+ }
+ return factory;
+}
+
+static gboolean
+autoplug_continue_cb (GstElement * uribin, GstPad * pad, GstCaps * caps,
+ GstElement * element)
+{
+ FactoryData *data;
+ GstElementFactory *factory;
+
+ GST_DEBUG ("found pad %s:%s of caps %" GST_PTR_FORMAT,
+ GST_DEBUG_PAD_NAME (pad), caps);
+
+ data = g_object_get_data (G_OBJECT (element), factory_key);
+
+ if (!(factory = find_payloader (data->factory, caps)))
+ goto no_factory;
+
+ /* we found a payloader, stop autoplugging so we can plug the
+ * payloader. */
+ GST_DEBUG ("found factory %s",
+ gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (factory)));
+ gst_object_unref (factory);
+
+ return FALSE;
+
+ /* ERRORS */
+no_factory:
+ {
+ /* no payloader, continue autoplugging */
+ GST_DEBUG ("no payloader found");
+ return TRUE;
+ }
+}
+
+static void
+pad_added_cb (GstElement * uribin, GstPad * pad, GstElement * element)
+{
+ GstRTSPMediaFactoryURI *urifact;
+ GstRTSPMediaFactoryURIPrivate *priv;
+ FactoryData *data;
+ GstElementFactory *factory;
+ GstElement *payloader;
+ GstCaps *caps;
+ GstPad *sinkpad, *srcpad, *ghostpad;
+ GstElement *convert;
+ gchar *padname, *payloader_name;
+
+ GST_DEBUG ("added pad %s:%s", GST_DEBUG_PAD_NAME (pad));
+
+ /* link the element now and expose the pad */
+ data = g_object_get_data (G_OBJECT (element), factory_key);
+ urifact = data->factory;
+ priv = urifact->priv;
+
+ /* ref to make refcounting easier later */
+ gst_object_ref (pad);
+ padname = gst_pad_get_name (pad);
+
+ /* get pad caps first, then call get_caps, then fail */
+ if ((caps = gst_pad_get_current_caps (pad)) == NULL)
+ if ((caps = gst_pad_query_caps (pad, NULL)) == NULL)
+ goto no_caps;
+
+ /* check for raw caps */
+ if (gst_caps_can_intersect (caps, priv->raw_vcaps)) {
+ /* we have raw video caps, insert converter */
+ convert = gst_element_factory_make ("videoconvert", NULL);
+ } else if (gst_caps_can_intersect (caps, priv->raw_acaps)) {
+ /* we have raw audio caps, insert converter */
+ convert = gst_element_factory_make ("audioconvert", NULL);
+ } else {
+ convert = NULL;
+ }
+
+ if (convert) {
+ gst_bin_add (GST_BIN_CAST (element), convert);
+ gst_element_set_state (convert, GST_STATE_PLAYING);
+
+ sinkpad = gst_element_get_static_pad (convert, "sink");
+ gst_pad_link (pad, sinkpad);
+ gst_object_unref (sinkpad);
+
+ /* unref old pad, we reffed before */
+ gst_object_unref (pad);
+ gst_caps_unref (caps);
+
+ /* continue with new pad and caps */
+ pad = gst_element_get_static_pad (convert, "src");
+ if ((caps = gst_pad_get_current_caps (pad)) == NULL)
+ if ((caps = gst_pad_query_caps (pad, NULL)) == NULL)
+ goto no_caps;
+ }
+
+ if (!(factory = find_payloader (urifact, caps)))
+ goto no_factory;
+
+ gst_caps_unref (caps);
+
+ /* we have a payloader now */
+ GST_DEBUG ("found payloader factory %s",
+ gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (factory)));
+
+ payloader_name = g_strdup_printf ("pay_%s", padname);
+ payloader = gst_element_factory_create (factory, payloader_name);
+ g_free (payloader_name);
+ if (payloader == NULL)
+ goto no_payloader;
+
+ g_object_set (payloader, "pt", data->pt, NULL);
+ data->pt++;
+
+ if (g_object_class_find_property (G_OBJECT_GET_CLASS (payloader),
+ "buffer-list"))
+ g_object_set (payloader, "buffer-list", TRUE, NULL);
+
+ /* add the payloader to the pipeline */
+ gst_bin_add (GST_BIN_CAST (element), payloader);
+ gst_element_set_state (payloader, GST_STATE_PLAYING);
+
+ /* link the pad to the sinkpad of the payloader */
+ sinkpad = gst_element_get_static_pad (payloader, "sink");
+ gst_pad_link (pad, sinkpad);
+ gst_object_unref (sinkpad);
+ gst_object_unref (pad);
+
+ /* now expose the srcpad of the payloader as a ghostpad with the same name
+ * as the uridecodebin pad name. */
+ srcpad = gst_element_get_static_pad (payloader, "src");
+ ghostpad = gst_ghost_pad_new (padname, srcpad);
+ gst_object_unref (srcpad);
+ g_free (padname);
+
+ gst_pad_set_active (ghostpad, TRUE);
+ gst_element_add_pad (element, ghostpad);
+
+ return;
+
+ /* ERRORS */
+no_caps:
+ {
+ GST_WARNING ("could not get caps from pad");
+ g_free (padname);
+ gst_object_unref (pad);
+ return;
+ }
+no_factory:
+ {
+ GST_DEBUG ("no payloader found");
+ g_free (padname);
+ gst_caps_unref (caps);
+ gst_object_unref (pad);
+ return;
+ }
+no_payloader:
+ {
+ GST_ERROR ("could not create payloader from factory");
+ g_free (padname);
+ gst_caps_unref (caps);
+ gst_object_unref (pad);
+ return;
+ }
+}
+
+static void
+no_more_pads_cb (GstElement * uribin, GstElement * element)
+{
+ GST_DEBUG ("no-more-pads");
+ gst_element_no_more_pads (element);
+}
+
+static GstElement *
+rtsp_media_factory_uri_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMediaFactoryURIPrivate *priv;
+ GstElement *topbin, *element, *uribin;
+ GstRTSPMediaFactoryURI *urifact;
+ FactoryData *data;
+
+ urifact = GST_RTSP_MEDIA_FACTORY_URI_CAST (factory);
+ priv = urifact->priv;
+
+ GST_LOG ("creating element");
+
+ topbin = gst_bin_new ("GstRTSPMediaFactoryURI");
+ g_assert (topbin != NULL);
+
+ /* our bin will dynamically expose payloaded pads */
+ element = gst_bin_new ("dynpay0");
+ g_assert (element != NULL);
+
+ uribin = gst_element_factory_make ("uridecodebin", "uribin");
+ if (uribin == NULL)
+ goto no_uridecodebin;
+
+ g_object_set (uribin, "uri", priv->uri, NULL);
+
+ /* keep factory data around */
+ data = g_new0 (FactoryData, 1);
+ data->factory = g_object_ref (urifact);
+ data->pt = 96;
+
+ g_object_set_data_full (G_OBJECT (element), factory_key,
+ data, (GDestroyNotify) free_data);
+
+ /* connect to the signals */
+ g_signal_connect (uribin, "autoplug-continue",
+ (GCallback) autoplug_continue_cb, element);
+ g_signal_connect (uribin, "pad-added", (GCallback) pad_added_cb, element);
+ g_signal_connect (uribin, "no-more-pads", (GCallback) no_more_pads_cb,
+ element);
+
+ gst_bin_add (GST_BIN_CAST (element), uribin);
+ gst_bin_add (GST_BIN_CAST (topbin), element);
+
+ return topbin;
+
+no_uridecodebin:
+ {
+ g_critical ("can't create uridecodebin element");
+ gst_object_unref (element);
+ return NULL;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include "rtsp-media-factory.h"
+
+#ifndef __GST_RTSP_MEDIA_FACTORY_URI_H__
+#define __GST_RTSP_MEDIA_FACTORY_URI_H__
+
+G_BEGIN_DECLS
+
+/* types for the media factory */
+#define GST_TYPE_RTSP_MEDIA_FACTORY_URI (gst_rtsp_media_factory_uri_get_type ())
+#define GST_IS_RTSP_MEDIA_FACTORY_URI(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI))
+#define GST_IS_RTSP_MEDIA_FACTORY_URI_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA_FACTORY_URI))
+#define GST_RTSP_MEDIA_FACTORY_URI_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURIClass))
+#define GST_RTSP_MEDIA_FACTORY_URI(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURI))
+#define GST_RTSP_MEDIA_FACTORY_URI_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURIClass))
+#define GST_RTSP_MEDIA_FACTORY_URI_CAST(obj) ((GstRTSPMediaFactoryURI*)(obj))
+#define GST_RTSP_MEDIA_FACTORY_URI_CLASS_CAST(klass) ((GstRTSPMediaFactoryURIClass*)(klass))
+
+typedef struct _GstRTSPMediaFactoryURI GstRTSPMediaFactoryURI;
+typedef struct _GstRTSPMediaFactoryURIClass GstRTSPMediaFactoryURIClass;
+typedef struct _GstRTSPMediaFactoryURIPrivate GstRTSPMediaFactoryURIPrivate;
+
+/**
+ * GstRTSPMediaFactoryURI:
+ *
+ * A media factory that creates a pipeline to play any uri.
+ */
+struct _GstRTSPMediaFactoryURI {
+ GstRTSPMediaFactory parent;
+
+ /*< private >*/
+ GstRTSPMediaFactoryURIPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaFactoryURIClass:
+ *
+ * The #GstRTSPMediaFactoryURI class structure.
+ */
+struct _GstRTSPMediaFactoryURIClass {
+ GstRTSPMediaFactoryClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_media_factory_uri_get_type (void);
+
+/* creating the factory */
+
+GST_RTSP_SERVER_API
+GstRTSPMediaFactoryURI * gst_rtsp_media_factory_uri_new (void);
+
+/* configuring the factory */
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_uri_set_uri (GstRTSPMediaFactoryURI *factory,
+ const gchar *uri);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_media_factory_uri_get_uri (GstRTSPMediaFactoryURI *factory);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPMediaFactoryURI, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MEDIA_FACTORY_URI_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2015 Centricular Ltd
+ * Author: Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media-factory
+ * @short_description: A factory for media pipelines
+ * @see_also: #GstRTSPMountPoints, #GstRTSPMedia
+ *
+ * The #GstRTSPMediaFactory is responsible for creating or recycling
+ * #GstRTSPMedia objects based on the passed URL.
+ *
+ * The default implementation of the object can create #GstRTSPMedia objects
+ * containing a pipeline created from a launch description set with
+ * gst_rtsp_media_factory_set_launch().
+ *
+ * Media from a factory can be shared by setting the shared flag with
+ * gst_rtsp_media_factory_set_shared(). When a factory is shared,
+ * gst_rtsp_media_factory_construct() will return the same #GstRTSPMedia when
+ * the url matches.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "rtsp-server-internal.h"
+#include "rtsp-media-factory.h"
+
+#define GST_RTSP_MEDIA_FACTORY_GET_LOCK(f) (&(GST_RTSP_MEDIA_FACTORY_CAST(f)->priv->lock))
+#define GST_RTSP_MEDIA_FACTORY_LOCK(f) (g_mutex_lock(GST_RTSP_MEDIA_FACTORY_GET_LOCK(f)))
+#define GST_RTSP_MEDIA_FACTORY_UNLOCK(f) (g_mutex_unlock(GST_RTSP_MEDIA_FACTORY_GET_LOCK(f)))
+
+struct _GstRTSPMediaFactoryPrivate
+{
+ GMutex lock; /* protects everything but medias */
+ GstRTSPPermissions *permissions;
+ gchar *launch;
+ gboolean shared;
+ GstRTSPSuspendMode suspend_mode;
+ gboolean eos_shutdown;
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans protocols;
+ guint buffer_size;
+ gint dscp_qos;
+ GstRTSPAddressPool *pool;
+ GstRTSPTransportMode transport_mode;
+ gboolean stop_on_disconnect;
+ gchar *multicast_iface;
+ guint max_mcast_ttl;
+ gboolean bind_mcast_address;
+ gboolean enable_rtcp;
+
+ GstClockTime rtx_time;
+ guint latency;
+ gboolean do_retransmission;
+
+ GMutex medias_lock;
+ GHashTable *medias; /* protected by medias_lock */
+
+ GType media_gtype;
+
+ GstClock *clock;
+
+ GstRTSPPublishClockMode publish_clock_mode;
+};
+
+#define DEFAULT_LAUNCH NULL
+#define DEFAULT_SHARED FALSE
+#define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
+#define DEFAULT_EOS_SHUTDOWN FALSE
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+ GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_BUFFER_SIZE 0x80000
+#define DEFAULT_LATENCY 200
+#define DEFAULT_MAX_MCAST_TTL 255
+#define DEFAULT_BIND_MCAST_ADDRESS FALSE
+#define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
+#define DEFAULT_STOP_ON_DISCONNECT TRUE
+#define DEFAULT_DO_RETRANSMISSION FALSE
+#define DEFAULT_DSCP_QOS (-1)
+#define DEFAULT_ENABLE_RTCP TRUE
+
+enum
+{
+ PROP_0,
+ PROP_LAUNCH,
+ PROP_SHARED,
+ PROP_SUSPEND_MODE,
+ PROP_EOS_SHUTDOWN,
+ PROP_PROFILES,
+ PROP_PROTOCOLS,
+ PROP_BUFFER_SIZE,
+ PROP_LATENCY,
+ PROP_TRANSPORT_MODE,
+ PROP_STOP_ON_DISCONNECT,
+ PROP_CLOCK,
+ PROP_MAX_MCAST_TTL,
+ PROP_BIND_MCAST_ADDRESS,
+ PROP_DSCP_QOS,
+ PROP_ENABLE_RTCP,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_MEDIA_CONSTRUCTED,
+ SIGNAL_MEDIA_CONFIGURE,
+ SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
+#define GST_CAT_DEFAULT rtsp_media_debug
+
+static guint gst_rtsp_media_factory_signals[SIGNAL_LAST] = { 0 };
+
+static void gst_rtsp_media_factory_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_finalize (GObject * obj);
+
+static gchar *default_gen_key (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url);
+static GstElement *default_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url);
+static GstRTSPMedia *default_construct (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url);
+static void default_configure (GstRTSPMediaFactory * factory,
+ GstRTSPMedia * media);
+static GstElement *default_create_pipeline (GstRTSPMediaFactory * factory,
+ GstRTSPMedia * media);
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPMediaFactory, gst_rtsp_media_factory,
+ G_TYPE_OBJECT);
+
+static void
+gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_media_factory_get_property;
+ gobject_class->set_property = gst_rtsp_media_factory_set_property;
+ gobject_class->finalize = gst_rtsp_media_factory_finalize;
+
+ /**
+ * GstRTSPMediaFactory::launch:
+ *
+ * The gst_parse_launch() line to use for constructing the pipeline in the
+ * default prepare vmethod.
+ *
+ * The pipeline description should return a GstBin as the toplevel element
+ * which can be accomplished by enclosing the description with brackets '('
+ * ')'.
+ *
+ * The description should return a pipeline with payloaders named pay0, pay1,
+ * etc.. Each of the payloaders will result in a stream.
+ *
+ * Support for dynamic payloaders can be accomplished by adding payloaders
+ * named dynpay0, dynpay1, etc..
+ */
+ g_object_class_install_property (gobject_class, PROP_LAUNCH,
+ g_param_spec_string ("launch", "Launch",
+ "A launch description of the pipeline", DEFAULT_LAUNCH,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_SHARED,
+ g_param_spec_boolean ("shared", "Shared",
+ "If media from this factory is shared", DEFAULT_SHARED,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
+ g_param_spec_enum ("suspend-mode", "Suspend Mode",
+ "Control how media will be suspended", GST_TYPE_RTSP_SUSPEND_MODE,
+ DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
+ g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
+ "Send EOS down the pipeline before shutting down",
+ DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
+ g_param_spec_uint ("buffer-size", "Buffer Size",
+ "The kernel UDP buffer size to use", 0, G_MAXUINT,
+ DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_LATENCY,
+ g_param_spec_uint ("latency", "Latency",
+ "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
+ DEFAULT_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
+ g_param_spec_flags ("transport-mode", "Transport Mode",
+ "If media from this factory is for PLAY or RECORD",
+ GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
+ g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
+ "If media from this factory should be stopped "
+ "when a client disconnects without TEARDOWN",
+ DEFAULT_STOP_ON_DISCONNECT,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_CLOCK,
+ g_param_spec_object ("clock", "Clock",
+ "Clock to be used by the pipelines created for all "
+ "medias of this factory", GST_TYPE_CLOCK,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MAX_MCAST_TTL,
+ g_param_spec_uint ("max-mcast-ttl", "Maximum multicast ttl",
+ "The maximum time-to-live value of outgoing multicast packets", 1,
+ 255, DEFAULT_MAX_MCAST_TTL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_BIND_MCAST_ADDRESS,
+ g_param_spec_boolean ("bind-mcast-address", "Bind mcast address",
+ "Whether the multicast sockets should be bound to multicast addresses "
+ "or INADDR_ANY",
+ DEFAULT_BIND_MCAST_ADDRESS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPMediaFactory:enable-rtcp:
+ *
+ * Whether the created media should send and receive RTCP
+ *
+ * Since: 1.20
+ */
+ g_object_class_install_property (gobject_class, PROP_ENABLE_RTCP,
+ g_param_spec_boolean ("enable-rtcp", "Enable RTCP",
+ "Whether the created media should send and receive RTCP",
+ DEFAULT_ENABLE_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_DSCP_QOS,
+ g_param_spec_int ("dscp-qos", "DSCP QoS",
+ "The IP DSCP field to use", -1, 63,
+ DEFAULT_DSCP_QOS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONSTRUCTED] =
+ g_signal_new ("media-constructed", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaFactoryClass,
+ media_constructed), NULL, NULL, NULL,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_MEDIA);
+
+ gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONFIGURE] =
+ g_signal_new ("media-configure", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaFactoryClass,
+ media_configure), NULL, NULL, NULL,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_MEDIA);
+
+ klass->gen_key = default_gen_key;
+ klass->create_element = default_create_element;
+ klass->construct = default_construct;
+ klass->configure = default_configure;
+ klass->create_pipeline = default_create_pipeline;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmediafactory", 0,
+ "GstRTSPMediaFactory");
+}
+
+static void
+gst_rtsp_media_factory_init (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv =
+ gst_rtsp_media_factory_get_instance_private (factory);
+ factory->priv = priv;
+
+ priv->launch = g_strdup (DEFAULT_LAUNCH);
+ priv->shared = DEFAULT_SHARED;
+ priv->suspend_mode = DEFAULT_SUSPEND_MODE;
+ priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
+ priv->profiles = DEFAULT_PROFILES;
+ priv->protocols = DEFAULT_PROTOCOLS;
+ priv->buffer_size = DEFAULT_BUFFER_SIZE;
+ priv->latency = DEFAULT_LATENCY;
+ priv->transport_mode = DEFAULT_TRANSPORT_MODE;
+ priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
+ priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
+ priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
+ priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
+ priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
+ priv->enable_rtcp = DEFAULT_ENABLE_RTCP;
+ priv->dscp_qos = DEFAULT_DSCP_QOS;
+
+ g_mutex_init (&priv->lock);
+ g_mutex_init (&priv->medias_lock);
+ priv->medias = g_hash_table_new_full (g_str_hash, g_str_equal,
+ g_free, g_object_unref);
+ priv->media_gtype = GST_TYPE_RTSP_MEDIA;
+}
+
+static void
+gst_rtsp_media_factory_finalize (GObject * obj)
+{
+ GstRTSPMediaFactory *factory = GST_RTSP_MEDIA_FACTORY (obj);
+ GstRTSPMediaFactoryPrivate *priv = factory->priv;
+
+ if (priv->clock)
+ gst_object_unref (priv->clock);
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+ g_hash_table_unref (priv->medias);
+ g_mutex_clear (&priv->medias_lock);
+ g_free (priv->launch);
+ g_mutex_clear (&priv->lock);
+ if (priv->pool)
+ g_object_unref (priv->pool);
+ g_free (priv->multicast_iface);
+
+ G_OBJECT_CLASS (gst_rtsp_media_factory_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_media_factory_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMediaFactory *factory = GST_RTSP_MEDIA_FACTORY (object);
+
+ switch (propid) {
+ case PROP_LAUNCH:
+ g_value_take_string (value, gst_rtsp_media_factory_get_launch (factory));
+ break;
+ case PROP_SHARED:
+ g_value_set_boolean (value, gst_rtsp_media_factory_is_shared (factory));
+ break;
+ case PROP_SUSPEND_MODE:
+ g_value_set_enum (value,
+ gst_rtsp_media_factory_get_suspend_mode (factory));
+ break;
+ case PROP_EOS_SHUTDOWN:
+ g_value_set_boolean (value,
+ gst_rtsp_media_factory_is_eos_shutdown (factory));
+ break;
+ case PROP_PROFILES:
+ g_value_set_flags (value, gst_rtsp_media_factory_get_profiles (factory));
+ break;
+ case PROP_PROTOCOLS:
+ g_value_set_flags (value, gst_rtsp_media_factory_get_protocols (factory));
+ break;
+ case PROP_BUFFER_SIZE:
+ g_value_set_uint (value,
+ gst_rtsp_media_factory_get_buffer_size (factory));
+ break;
+ case PROP_LATENCY:
+ g_value_set_uint (value, gst_rtsp_media_factory_get_latency (factory));
+ break;
+ case PROP_TRANSPORT_MODE:
+ g_value_set_flags (value,
+ gst_rtsp_media_factory_get_transport_mode (factory));
+ break;
+ case PROP_STOP_ON_DISCONNECT:
+ g_value_set_boolean (value,
+ gst_rtsp_media_factory_is_stop_on_disonnect (factory));
+ break;
+ case PROP_CLOCK:
+ g_value_take_object (value, gst_rtsp_media_factory_get_clock (factory));
+ break;
+ case PROP_MAX_MCAST_TTL:
+ g_value_set_uint (value,
+ gst_rtsp_media_factory_get_max_mcast_ttl (factory));
+ break;
+ case PROP_BIND_MCAST_ADDRESS:
+ g_value_set_boolean (value,
+ gst_rtsp_media_factory_is_bind_mcast_address (factory));
+ break;
+ case PROP_DSCP_QOS:
+ g_value_set_int (value, gst_rtsp_media_factory_get_dscp_qos (factory));
+ break;
+ case PROP_ENABLE_RTCP:
+ g_value_set_boolean (value,
+ gst_rtsp_media_factory_is_enable_rtcp (factory));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_media_factory_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMediaFactory *factory = GST_RTSP_MEDIA_FACTORY (object);
+
+ switch (propid) {
+ case PROP_LAUNCH:
+ gst_rtsp_media_factory_set_launch (factory, g_value_get_string (value));
+ break;
+ case PROP_SHARED:
+ gst_rtsp_media_factory_set_shared (factory, g_value_get_boolean (value));
+ break;
+ case PROP_SUSPEND_MODE:
+ gst_rtsp_media_factory_set_suspend_mode (factory,
+ g_value_get_enum (value));
+ break;
+ case PROP_EOS_SHUTDOWN:
+ gst_rtsp_media_factory_set_eos_shutdown (factory,
+ g_value_get_boolean (value));
+ break;
+ case PROP_PROFILES:
+ gst_rtsp_media_factory_set_profiles (factory, g_value_get_flags (value));
+ break;
+ case PROP_PROTOCOLS:
+ gst_rtsp_media_factory_set_protocols (factory, g_value_get_flags (value));
+ break;
+ case PROP_BUFFER_SIZE:
+ gst_rtsp_media_factory_set_buffer_size (factory,
+ g_value_get_uint (value));
+ break;
+ case PROP_LATENCY:
+ gst_rtsp_media_factory_set_latency (factory, g_value_get_uint (value));
+ break;
+ case PROP_TRANSPORT_MODE:
+ gst_rtsp_media_factory_set_transport_mode (factory,
+ g_value_get_flags (value));
+ break;
+ case PROP_STOP_ON_DISCONNECT:
+ gst_rtsp_media_factory_set_stop_on_disconnect (factory,
+ g_value_get_boolean (value));
+ break;
+ case PROP_CLOCK:
+ gst_rtsp_media_factory_set_clock (factory, g_value_get_object (value));
+ break;
+ case PROP_MAX_MCAST_TTL:
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory,
+ g_value_get_uint (value));
+ break;
+ case PROP_BIND_MCAST_ADDRESS:
+ gst_rtsp_media_factory_set_bind_mcast_address (factory,
+ g_value_get_boolean (value));
+ break;
+ case PROP_DSCP_QOS:
+ gst_rtsp_media_factory_set_dscp_qos (factory, g_value_get_int (value));
+ break;
+ case PROP_ENABLE_RTCP:
+ gst_rtsp_media_factory_set_enable_rtcp (factory,
+ g_value_get_boolean (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_media_factory_new:
+ *
+ * Create a new #GstRTSPMediaFactory instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPMediaFactory object.
+ */
+GstRTSPMediaFactory *
+gst_rtsp_media_factory_new (void)
+{
+ GstRTSPMediaFactory *result;
+
+ result = g_object_new (GST_TYPE_RTSP_MEDIA_FACTORY, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_permissions:
+ * @factory: a #GstRTSPMediaFactory
+ * @permissions: (transfer none) (nullable): a #GstRTSPPermissions
+ *
+ * Set @permissions on @factory.
+ */
+void
+gst_rtsp_media_factory_set_permissions (GstRTSPMediaFactory * factory,
+ GstRTSPPermissions * permissions)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+ if ((priv->permissions = permissions))
+ gst_rtsp_permissions_ref (permissions);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_permissions:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the permissions object from @factory.
+ *
+ * Returns: (transfer full) (nullable): a #GstRTSPPermissions object, unref after usage.
+ */
+GstRTSPPermissions *
+gst_rtsp_media_factory_get_permissions (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPPermissions *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if ((result = priv->permissions))
+ gst_rtsp_permissions_ref (result);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_add_role:
+ * @factory: a #GstRTSPMediaFactory
+ * @role: a role
+ * @fieldname: the first field name
+ * @...: additional arguments
+ *
+ * A convenience method to add @role with @fieldname and additional arguments to
+ * the permissions of @factory. If @factory had no permissions, new permissions
+ * will be created and the role will be added to it.
+ */
+void
+gst_rtsp_media_factory_add_role (GstRTSPMediaFactory * factory,
+ const gchar * role, const gchar * fieldname, ...)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ va_list var_args;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+ g_return_if_fail (role != NULL);
+ g_return_if_fail (fieldname != NULL);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if (priv->permissions == NULL)
+ priv->permissions = gst_rtsp_permissions_new ();
+
+ va_start (var_args, fieldname);
+ gst_rtsp_permissions_add_role_valist (priv->permissions, role, fieldname,
+ var_args);
+ va_end (var_args);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_add_role_from_structure:
+ *
+ * A convenience wrapper around gst_rtsp_permissions_add_role_from_structure().
+ * If @factory had no permissions, new permissions will be created and the
+ * role will be added to it.
+ *
+ * Since: 1.14
+ */
+void
+gst_rtsp_media_factory_add_role_from_structure (GstRTSPMediaFactory * factory,
+ GstStructure * structure)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+ g_return_if_fail (GST_IS_STRUCTURE (structure));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if (priv->permissions == NULL)
+ priv->permissions = gst_rtsp_permissions_new ();
+
+ gst_rtsp_permissions_add_role_from_structure (priv->permissions, structure);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_set_launch:
+ * @factory: a #GstRTSPMediaFactory
+ * @launch: the launch description
+ *
+ *
+ * The gst_parse_launch() line to use for constructing the pipeline in the
+ * default prepare vmethod.
+ *
+ * The pipeline description should return a GstBin as the toplevel element
+ * which can be accomplished by enclosing the description with brackets '('
+ * ')'.
+ *
+ * The description should return a pipeline with payloaders named pay0, pay1,
+ * etc.. Each of the payloaders will result in a stream.
+ */
+void
+gst_rtsp_media_factory_set_launch (GstRTSPMediaFactory * factory,
+ const gchar * launch)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+ g_return_if_fail (launch != NULL);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ g_free (priv->launch);
+ priv->launch = g_strdup (launch);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_launch:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the gst_parse_launch() pipeline description that will be used in the
+ * default prepare vmethod.
+ *
+ * Returns: (transfer full) (nullable): the configured launch description. g_free() after
+ * usage.
+ */
+gchar *
+gst_rtsp_media_factory_get_launch (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = g_strdup (priv->launch);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_suspend_mode:
+ * @factory: a #GstRTSPMediaFactory
+ * @mode: the new #GstRTSPSuspendMode
+ *
+ * Configure how media created from this factory will be suspended.
+ */
+void
+gst_rtsp_media_factory_set_suspend_mode (GstRTSPMediaFactory * factory,
+ GstRTSPSuspendMode mode)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->suspend_mode = mode;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_suspend_mode:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get how media created from this factory will be suspended.
+ *
+ * Returns: a #GstRTSPSuspendMode.
+ */
+GstRTSPSuspendMode
+gst_rtsp_media_factory_get_suspend_mode (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPSuspendMode result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory),
+ GST_RTSP_SUSPEND_MODE_NONE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->suspend_mode;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_shared:
+ * @factory: a #GstRTSPMediaFactory
+ * @shared: the new value
+ *
+ * Configure if media created from this factory can be shared between clients.
+ */
+void
+gst_rtsp_media_factory_set_shared (GstRTSPMediaFactory * factory,
+ gboolean shared)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->shared = shared;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_is_shared:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get if media created from this factory can be shared between clients.
+ *
+ * Returns: %TRUE if the media will be shared between clients.
+ */
+gboolean
+gst_rtsp_media_factory_is_shared (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->shared;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_eos_shutdown:
+ * @factory: a #GstRTSPMediaFactory
+ * @eos_shutdown: the new value
+ *
+ * Configure if media created from this factory will have an EOS sent to the
+ * pipeline before shutdown.
+ */
+void
+gst_rtsp_media_factory_set_eos_shutdown (GstRTSPMediaFactory * factory,
+ gboolean eos_shutdown)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->eos_shutdown = eos_shutdown;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_is_eos_shutdown:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get if media created from this factory will have an EOS event sent to the
+ * pipeline before shutdown.
+ *
+ * Returns: %TRUE if the media will receive EOS before shutdown.
+ */
+gboolean
+gst_rtsp_media_factory_is_eos_shutdown (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->eos_shutdown;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_buffer_size:
+ * @factory: a #GstRTSPMedia
+ * @size: the new value
+ *
+ * Set the kernel UDP buffer size.
+ */
+void
+gst_rtsp_media_factory_set_buffer_size (GstRTSPMediaFactory * factory,
+ guint size)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->buffer_size = size;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_buffer_size:
+ * @factory: a #GstRTSPMedia
+ *
+ * Get the kernel UDP buffer size.
+ *
+ * Returns: the kernel UDP buffer size.
+ */
+guint
+gst_rtsp_media_factory_get_buffer_size (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ guint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), 0);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->buffer_size;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_dscp_qos:
+ * @factory: a #GstRTSPMediaFactory
+ * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
+ *
+ * Configure the media dscp qos to @dscp_qos.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_media_factory_set_dscp_qos (GstRTSPMediaFactory * factory,
+ gint dscp_qos)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ if (dscp_qos < -1 || dscp_qos > 63) {
+ GST_WARNING_OBJECT (factory, "trying to set illegal dscp qos %d", dscp_qos);
+ return;
+ }
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->dscp_qos = dscp_qos;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_dscp_qos:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the configured media DSCP QoS.
+ *
+ * Returns: the media DSCP QoS value or -1 if disabled.
+ *
+ * Since: 1.18
+ */
+gint
+gst_rtsp_media_factory_get_dscp_qos (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ guint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), 0);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->dscp_qos;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_address_pool:
+ * @factory: a #GstRTSPMediaFactory
+ * @pool: (transfer none) (nullable): a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @factory.
+ */
+void
+gst_rtsp_media_factory_set_address_pool (GstRTSPMediaFactory * factory,
+ GstRTSPAddressPool * pool)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPAddressPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if ((old = priv->pool) != pool)
+ priv->pool = pool ? g_object_ref (pool) : NULL;
+ else
+ old = NULL;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_media_factory_get_address_pool:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @factory.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @factory. g_object_unref() after
+ * usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_media_factory_get_address_pool (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPAddressPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if ((result = priv->pool))
+ g_object_ref (result);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_multicast_iface:
+ * @factory: a #GstRTSPMediaFactory
+ * @multicast_iface: (transfer none) (nullable): a multicast interface name
+ *
+ * configure @multicast_iface to be used for @factory.
+ */
+void
+gst_rtsp_media_factory_set_multicast_iface (GstRTSPMediaFactory * media_factory,
+ const gchar * multicast_iface)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gchar *old;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (media_factory));
+
+ priv = media_factory->priv;
+
+ GST_LOG_OBJECT (media_factory, "set multicast interface %s", multicast_iface);
+
+ g_mutex_lock (&priv->lock);
+ if ((old = priv->multicast_iface) != multicast_iface)
+ priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
+ else
+ old = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_free (old);
+}
+
+/**
+ * gst_rtsp_media_factory_get_multicast_iface:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the multicast interface used for @factory.
+ *
+ * Returns: (transfer full) (nullable): the multicast interface for @factory. g_free() after
+ * usage.
+ */
+gchar *
+gst_rtsp_media_factory_get_multicast_iface (GstRTSPMediaFactory * media_factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (media_factory), NULL);
+
+ priv = media_factory->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->multicast_iface))
+ result = g_strdup (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_profiles:
+ * @factory: a #GstRTSPMediaFactory
+ * @profiles: the new flags
+ *
+ * Configure the allowed profiles for @factory.
+ */
+void
+gst_rtsp_media_factory_set_profiles (GstRTSPMediaFactory * factory,
+ GstRTSPProfile profiles)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_DEBUG_OBJECT (factory, "profiles %d", profiles);
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->profiles = profiles;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_profiles:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the allowed profiles of @factory.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_media_factory_get_profiles (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory),
+ GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ res = priv->profiles;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_factory_set_protocols:
+ * @factory: a #GstRTSPMediaFactory
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @factory.
+ */
+void
+gst_rtsp_media_factory_set_protocols (GstRTSPMediaFactory * factory,
+ GstRTSPLowerTrans protocols)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_DEBUG_OBJECT (factory, "protocols %d", protocols);
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->protocols = protocols;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_protocols:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the allowed protocols of @factory.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_media_factory_get_protocols (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPLowerTrans res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory),
+ GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ res = priv->protocols;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_factory_set_stop_on_disconnect:
+ * @factory: a #GstRTSPMediaFactory
+ * @stop_on_disconnect: the new value
+ *
+ * Configure if media created from this factory should be stopped
+ * when a client disconnects without sending TEARDOWN.
+ */
+void
+gst_rtsp_media_factory_set_stop_on_disconnect (GstRTSPMediaFactory * factory,
+ gboolean stop_on_disconnect)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->stop_on_disconnect = stop_on_disconnect;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_is_stop_on_disconnect:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get if media created from this factory should be stopped when a client
+ * disconnects without sending TEARDOWN.
+ *
+ * Returns: %TRUE if the media will be stopped when a client disconnects
+ * without sending TEARDOWN.
+ */
+gboolean
+gst_rtsp_media_factory_is_stop_on_disonnect (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), TRUE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->stop_on_disconnect;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_retransmission_time:
+ * @factory: a #GstRTSPMediaFactory
+ * @time: a #GstClockTime
+ *
+ * Configure the time to store for possible retransmission
+ */
+void
+gst_rtsp_media_factory_set_retransmission_time (GstRTSPMediaFactory * factory,
+ GstClockTime time)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_DEBUG_OBJECT (factory, "retransmission time %" G_GUINT64_FORMAT, time);
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->rtx_time = time;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_retransmission_time:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the time that is stored for retransmission purposes
+ *
+ * Returns: a #GstClockTime
+ */
+GstClockTime
+gst_rtsp_media_factory_get_retransmission_time (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstClockTime res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), 0);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ res = priv->rtx_time;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_factory_set_do_retransmission:
+ *
+ * Set whether retransmission requests will be sent for
+ * receiving media
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_media_factory_set_do_retransmission (GstRTSPMediaFactory * factory,
+ gboolean do_retransmission)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_DEBUG_OBJECT (factory, "Do retransmission %d", do_retransmission);
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->do_retransmission = do_retransmission;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_do_retransmission:
+ *
+ * Returns: Whether retransmission requests will be sent for receiving media
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_media_factory_get_do_retransmission (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), 0);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ res = priv->do_retransmission;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_factory_set_latency:
+ * @factory: a #GstRTSPMediaFactory
+ * @latency: latency in milliseconds
+ *
+ * Configure the latency used for receiving media
+ */
+void
+gst_rtsp_media_factory_set_latency (GstRTSPMediaFactory * factory,
+ guint latency)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_DEBUG_OBJECT (factory, "latency %ums", latency);
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->latency = latency;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_latency:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the latency that is used for receiving media
+ *
+ * Returns: latency in milliseconds
+ */
+guint
+gst_rtsp_media_factory_get_latency (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), 0);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ res = priv->latency;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return res;
+}
+
+static gboolean
+compare_media (gpointer key, GstRTSPMedia * media1, GstRTSPMedia * media2)
+{
+ return (media1 == media2);
+}
+
+static void
+media_unprepared (GstRTSPMedia * media, GWeakRef * ref)
+{
+ GstRTSPMediaFactory *factory = g_weak_ref_get (ref);
+ GstRTSPMediaFactoryPrivate *priv;
+
+ if (!factory)
+ return;
+
+ priv = factory->priv;
+
+ g_mutex_lock (&priv->medias_lock);
+ g_hash_table_foreach_remove (priv->medias, (GHRFunc) compare_media, media);
+ g_mutex_unlock (&priv->medias_lock);
+
+ g_object_unref (factory);
+}
+
+static GWeakRef *
+weak_ref_new (gpointer obj)
+{
+ GWeakRef *ref = g_slice_new (GWeakRef);
+
+ g_weak_ref_init (ref, obj);
+ return ref;
+}
+
+static void
+weak_ref_free (GWeakRef * ref)
+{
+ g_weak_ref_clear (ref);
+ g_slice_free (GWeakRef, ref);
+}
+
+/**
+ * gst_rtsp_media_factory_construct:
+ * @factory: a #GstRTSPMediaFactory
+ * @url: the url used
+ *
+ * Construct the media object and create its streams. Implementations
+ * should create the needed gstreamer elements and add them to the result
+ * object. No state changes should be performed on them yet.
+ *
+ * One or more GstRTSPStream objects should be created from the result
+ * with gst_rtsp_media_create_stream ().
+ *
+ * After the media is constructed, it can be configured and then prepared
+ * with gst_rtsp_media_prepare ().
+ *
+ * Returns: (transfer full): a new #GstRTSPMedia if the media could be prepared.
+ */
+GstRTSPMedia *
+gst_rtsp_media_factory_construct (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gchar *key;
+ GstRTSPMedia *media;
+ GstRTSPMediaFactoryClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+ g_return_val_if_fail (url != NULL, NULL);
+
+ priv = factory->priv;
+ klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+ /* convert the url to a key for the hashtable. NULL return or a NULL function
+ * will not cache anything for this factory. */
+ if (klass->gen_key)
+ key = klass->gen_key (factory, url);
+ else
+ key = NULL;
+
+ g_mutex_lock (&priv->medias_lock);
+ if (key) {
+ /* we have a key, see if we find a cached media */
+ media = g_hash_table_lookup (priv->medias, key);
+ if (media)
+ g_object_ref (media);
+ } else
+ media = NULL;
+
+ if (media == NULL) {
+ /* nothing cached found, try to create one */
+ if (klass->construct) {
+ media = klass->construct (factory, url);
+ if (media)
+ g_signal_emit (factory,
+ gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONSTRUCTED], 0, media,
+ NULL);
+ } else
+ media = NULL;
+
+ if (media) {
+ /* configure the media */
+ if (klass->configure)
+ klass->configure (factory, media);
+
+ g_signal_emit (factory,
+ gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONFIGURE], 0, media,
+ NULL);
+
+ /* check if we can cache this media */
+ if (gst_rtsp_media_is_shared (media) && key) {
+ /* insert in the hashtable, takes ownership of the key */
+ g_object_ref (media);
+ g_hash_table_insert (priv->medias, key, media);
+ key = NULL;
+ }
+ if (!gst_rtsp_media_is_reusable (media)) {
+ /* when not reusable, connect to the unprepare signal to remove the item
+ * from our cache when it gets unprepared */
+ g_signal_connect_data (media, "unprepared",
+ (GCallback) media_unprepared, weak_ref_new (factory),
+ (GClosureNotify) weak_ref_free, 0);
+ }
+ }
+ }
+ g_mutex_unlock (&priv->medias_lock);
+
+ if (key)
+ g_free (key);
+
+ GST_INFO ("constructed media %p for url %s", media, url->abspath);
+
+ return media;
+}
+
+/**
+ * gst_rtsp_media_factory_set_media_gtype:
+ * @factory: a #GstRTSPMediaFactory
+ * @media_gtype: the GType of the class to create
+ *
+ * Configure the GType of the GstRTSPMedia subclass to
+ * create (by default, overridden construct vmethods
+ * may of course do something different)
+ *
+ * Since: 1.6
+ */
+void
+gst_rtsp_media_factory_set_media_gtype (GstRTSPMediaFactory * factory,
+ GType media_gtype)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (g_type_is_a (media_gtype, GST_TYPE_RTSP_MEDIA));
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv = factory->priv;
+ priv->media_gtype = media_gtype;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_media_gtype:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Return the GType of the GstRTSPMedia subclass this
+ * factory will create.
+ *
+ * Since: 1.6
+ */
+GType
+gst_rtsp_media_factory_get_media_gtype (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GType ret;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv = factory->priv;
+ ret = priv->media_gtype;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_media_factory_set_clock:
+ * @factory: a #GstRTSPMediaFactory
+ * @clock: (nullable): the clock to be used by the media factory
+ *
+ * Configures a specific clock to be used by the pipelines
+ * of all medias created from this factory.
+ *
+ * Since: 1.8
+ */
+void
+gst_rtsp_media_factory_set_clock (GstRTSPMediaFactory * factory,
+ GstClock * clock)
+{
+ GstClock **clock_p;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+ g_return_if_fail (GST_IS_CLOCK (clock) || clock == NULL);
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ clock_p = &factory->priv->clock;
+ gst_object_replace ((GstObject **) clock_p, (GstObject *) clock);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_clock:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Returns the clock that is going to be used by the pipelines
+ * of all medias created from this factory.
+ *
+ * Returns: (transfer full): The GstClock
+ *
+ * Since: 1.8
+ */
+GstClock *
+gst_rtsp_media_factory_get_clock (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstClock *ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv = factory->priv;
+ ret = priv->clock ? gst_object_ref (priv->clock) : NULL;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_media_factory_set_publish_clock_mode:
+ * @factory: a #GstRTSPMediaFactory
+ * @mode: the clock publish mode
+ *
+ * Sets if and how the media clock should be published according to RFC7273.
+ *
+ * Since: 1.8
+ */
+void
+gst_rtsp_media_factory_set_publish_clock_mode (GstRTSPMediaFactory * factory,
+ GstRTSPPublishClockMode mode)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv = factory->priv;
+ priv->publish_clock_mode = mode;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_publish_clock_mode:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Gets if and how the media clock should be published according to RFC7273.
+ *
+ * Returns: The GstRTSPPublishClockMode
+ *
+ * Since: 1.8
+ */
+GstRTSPPublishClockMode
+gst_rtsp_media_factory_get_publish_clock_mode (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPPublishClockMode ret;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv = factory->priv;
+ ret = priv->publish_clock_mode;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_media_factory_set_max_mcast_ttl:
+ * @factory: a #GstRTSPMedia
+ * @ttl: the new multicast ttl value
+ *
+ * Set the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: %TRUE if the requested ttl has been set successfully.
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_media_factory_set_max_mcast_ttl (GstRTSPMediaFactory * factory,
+ guint ttl)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
+ GST_WARNING_OBJECT (factory, "The requested mcast TTL value is not valid.");
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+ return FALSE;
+ }
+ priv->max_mcast_ttl = ttl;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_media_factory_get_max_mcast_ttl:
+ * @factory: a #GstRTSPMedia
+ *
+ * Get the the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Since: 1.16
+ */
+guint
+gst_rtsp_media_factory_get_max_mcast_ttl (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ guint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), 0);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->max_mcast_ttl;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_bind_mcast_address:
+ * @factory: a #GstRTSPMediaFactory
+ * @bind_mcast_addr: the new value
+ *
+ * Decide whether the multicast socket should be bound to a multicast address or
+ * INADDR_ANY.
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_media_factory_set_bind_mcast_address (GstRTSPMediaFactory * factory,
+ gboolean bind_mcast_addr)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->bind_mcast_address = bind_mcast_addr;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_is_bind_mcast_address:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Check if multicast sockets are configured to be bound to multicast addresses.
+ *
+ * Returns: %TRUE if multicast sockets are configured to be bound to multicast addresses.
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_media_factory_is_bind_mcast_address (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->bind_mcast_address;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_enable_rtcp:
+ * @factory: a #GstRTSPMediaFactory
+ * @enable: the new value
+ *
+ * Decide whether the created media should send and receive RTCP
+ *
+ * Since: 1.20
+ */
+void
+gst_rtsp_media_factory_set_enable_rtcp (GstRTSPMediaFactory * factory,
+ gboolean enable)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->enable_rtcp = enable;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_is_enable_rtcp:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Check if created media will send and receive RTCP
+ *
+ * Returns: %TRUE if created media will send and receive RTCP
+ *
+ * Since: 1.20
+ */
+gboolean
+gst_rtsp_media_factory_is_enable_rtcp (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->enable_rtcp;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+static gchar *
+default_gen_key (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
+{
+ gchar *result;
+ const gchar *pre_query;
+ const gchar *query;
+ guint16 port;
+
+ pre_query = url->query ? "?" : "";
+ query = url->query ? url->query : "";
+
+ gst_rtsp_url_get_port (url, &port);
+
+ result = g_strdup_printf ("%u%s%s%s", port, url->abspath, pre_query, query);
+
+ return result;
+}
+
+static GstElement *
+default_create_element (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
+{
+ GstRTSPMediaFactoryPrivate *priv = factory->priv;
+ GstElement *element;
+ GError *error = NULL;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ /* we need a parse syntax */
+ if (priv->launch == NULL)
+ goto no_launch;
+
+ /* parse the user provided launch line */
+ element =
+ gst_parse_launch_full (priv->launch, NULL, GST_PARSE_FLAG_PLACE_IN_BIN,
+ &error);
+ if (element == NULL)
+ goto parse_error;
+
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ if (error != NULL) {
+ /* a recoverable error was encountered */
+ GST_WARNING ("recoverable parsing error: %s", error->message);
+ g_error_free (error);
+ }
+ return element;
+
+ /* ERRORS */
+no_launch:
+ {
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+ g_critical ("no launch line specified");
+ return NULL;
+ }
+parse_error:
+ {
+ g_critical ("could not parse launch syntax (%s): %s", priv->launch,
+ (error ? error->message : "unknown reason"));
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+ if (error)
+ g_error_free (error);
+ return NULL;
+ }
+}
+
+static GstRTSPMedia *
+default_construct (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
+{
+ GstRTSPMedia *media;
+ GstElement *element, *pipeline;
+ GstRTSPMediaFactoryClass *klass;
+ GType media_gtype;
+ gboolean enable_rtcp;
+
+ klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+ if (!klass->create_pipeline)
+ goto no_create;
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ if (element == NULL)
+ goto no_element;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ media_gtype = factory->priv->media_gtype;
+ enable_rtcp = factory->priv->enable_rtcp;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ /* create a new empty media */
+ media =
+ g_object_new (media_gtype, "element", element, "transport-mode",
+ factory->priv->transport_mode, NULL);
+
+ /* We need to call this prior to collecting streams */
+ gst_rtsp_media_set_enable_rtcp (media, enable_rtcp);
+
+ gst_rtsp_media_collect_streams (media);
+
+ pipeline = klass->create_pipeline (factory, media);
+ if (pipeline == NULL)
+ goto no_pipeline;
+
+ return media;
+
+ /* ERRORS */
+no_create:
+ {
+ g_critical ("no create_pipeline function");
+ return NULL;
+ }
+no_element:
+ {
+ g_critical ("could not create element");
+ return NULL;
+ }
+no_pipeline:
+ {
+ g_critical ("can't create pipeline");
+ g_object_unref (media);
+ return NULL;
+ }
+}
+
+static GstElement *
+default_create_pipeline (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+ GstElement *pipeline;
+
+ pipeline = gst_pipeline_new ("media-pipeline");
+
+ /* FIXME 2.0: This should be done by the caller, not the vfunc. Every
+ * implementation of the vfunc has to call it otherwise at the end.
+ * Also it does not allow use to add further behaviour here that could
+ * be reused by subclasses that chain up */
+ gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+ return pipeline;
+}
+
+static void
+default_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+ GstRTSPMediaFactoryPrivate *priv = factory->priv;
+ gboolean shared, eos_shutdown, stop_on_disconnect;
+ guint size;
+ gint dscp_qos;
+ GstRTSPSuspendMode suspend_mode;
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans protocols;
+ GstRTSPAddressPool *pool;
+ GstRTSPPermissions *perms;
+ GstClockTime rtx_time;
+ guint latency;
+ GstRTSPTransportMode transport_mode;
+ GstClock *clock;
+ gchar *multicast_iface;
+ GstRTSPPublishClockMode publish_clock_mode;
+ guint ttl;
+ gboolean bind_mcast;
+
+ /* configure the sharedness */
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ suspend_mode = priv->suspend_mode;
+ shared = priv->shared;
+ eos_shutdown = priv->eos_shutdown;
+ size = priv->buffer_size;
+ dscp_qos = priv->dscp_qos;
+ profiles = priv->profiles;
+ protocols = priv->protocols;
+ rtx_time = priv->rtx_time;
+ latency = priv->latency;
+ transport_mode = priv->transport_mode;
+ stop_on_disconnect = priv->stop_on_disconnect;
+ clock = priv->clock ? gst_object_ref (priv->clock) : NULL;
+ publish_clock_mode = priv->publish_clock_mode;
+ ttl = priv->max_mcast_ttl;
+ bind_mcast = priv->bind_mcast_address;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ gst_rtsp_media_set_suspend_mode (media, suspend_mode);
+ gst_rtsp_media_set_shared (media, shared);
+ gst_rtsp_media_set_eos_shutdown (media, eos_shutdown);
+ gst_rtsp_media_set_buffer_size (media, size);
+ gst_rtsp_media_set_dscp_qos (media, dscp_qos);
+ gst_rtsp_media_set_profiles (media, profiles);
+ gst_rtsp_media_set_protocols (media, protocols);
+ gst_rtsp_media_set_retransmission_time (media, rtx_time);
+ gst_rtsp_media_set_do_retransmission (media, priv->do_retransmission);
+ gst_rtsp_media_set_latency (media, latency);
+ gst_rtsp_media_set_transport_mode (media, transport_mode);
+ gst_rtsp_media_set_stop_on_disconnect (media, stop_on_disconnect);
+ gst_rtsp_media_set_publish_clock_mode (media, publish_clock_mode);
+ gst_rtsp_media_set_max_mcast_ttl (media, ttl);
+ gst_rtsp_media_set_bind_mcast_address (media, bind_mcast);
+
+ if (clock) {
+ gst_rtsp_media_set_clock (media, clock);
+ gst_object_unref (clock);
+ }
+
+ if ((pool = gst_rtsp_media_factory_get_address_pool (factory))) {
+ gst_rtsp_media_set_address_pool (media, pool);
+ g_object_unref (pool);
+ }
+ if ((multicast_iface = gst_rtsp_media_factory_get_multicast_iface (factory))) {
+ gst_rtsp_media_set_multicast_iface (media, multicast_iface);
+ g_free (multicast_iface);
+ }
+ if ((perms = gst_rtsp_media_factory_get_permissions (factory))) {
+ gst_rtsp_media_set_permissions (media, perms);
+ gst_rtsp_permissions_unref (perms);
+ }
+}
+
+/**
+ * gst_rtsp_media_factory_create_element:
+ * @factory: a #GstRTSPMediaFactory
+ * @url: the url used
+ *
+ * Construct and return a #GstElement that is a #GstBin containing
+ * the elements to use for streaming the media.
+ *
+ * The bin should contain payloaders pay\%d for each stream. The default
+ * implementation of this function returns the bin created from the
+ * launch parameter.
+ *
+ * Returns: (transfer floating): a new #GstElement.
+ */
+GstElement *
+gst_rtsp_media_factory_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMediaFactoryClass *klass;
+ GstElement *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+ g_return_val_if_fail (url != NULL, NULL);
+
+ klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+ if (klass->create_element)
+ result = klass->create_element (factory, url);
+ else
+ result = NULL;
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_transport_mode:
+ * @factory: a #GstRTSPMediaFactory
+ * @mode: the new value
+ *
+ * Configure if this factory creates media for PLAY or RECORD modes.
+ */
+void
+gst_rtsp_media_factory_set_transport_mode (GstRTSPMediaFactory * factory,
+ GstRTSPTransportMode mode)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->transport_mode = mode;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_transport_mode:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get if media created from this factory can be used for PLAY or RECORD
+ * methods.
+ *
+ * Returns: The transport mode.
+ */
+GstRTSPTransportMode
+gst_rtsp_media_factory_get_transport_mode (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPTransportMode result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->transport_mode;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspurl.h>
+
+#include "rtsp-media.h"
+#include "rtsp-permissions.h"
+#include "rtsp-address-pool.h"
+
+#ifndef __GST_RTSP_MEDIA_FACTORY_H__
+#define __GST_RTSP_MEDIA_FACTORY_H__
+
+G_BEGIN_DECLS
+
+/* types for the media factory */
+#define GST_TYPE_RTSP_MEDIA_FACTORY (gst_rtsp_media_factory_get_type ())
+#define GST_IS_RTSP_MEDIA_FACTORY(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY))
+#define GST_IS_RTSP_MEDIA_FACTORY_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA_FACTORY))
+#define GST_RTSP_MEDIA_FACTORY_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactoryClass))
+#define GST_RTSP_MEDIA_FACTORY(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactory))
+#define GST_RTSP_MEDIA_FACTORY_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactoryClass))
+#define GST_RTSP_MEDIA_FACTORY_CAST(obj) ((GstRTSPMediaFactory*)(obj))
+#define GST_RTSP_MEDIA_FACTORY_CLASS_CAST(klass) ((GstRTSPMediaFactoryClass*)(klass))
+
+typedef struct _GstRTSPMediaFactory GstRTSPMediaFactory;
+typedef struct _GstRTSPMediaFactoryClass GstRTSPMediaFactoryClass;
+typedef struct _GstRTSPMediaFactoryPrivate GstRTSPMediaFactoryPrivate;
+
+/**
+ * GstRTSPMediaFactory:
+ *
+ * The definition and logic for constructing the pipeline for a media. The media
+ * can contain multiple streams like audio and video.
+ */
+struct _GstRTSPMediaFactory {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPMediaFactoryPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaFactoryClass:
+ * @gen_key: convert @url to a key for caching shared #GstRTSPMedia objects.
+ * The default implementation of this function will use the complete URL
+ * including the query parameters to return a key.
+ * @create_element: Construct and return a #GstElement that is a #GstBin containing
+ * the elements to use for streaming the media. The bin should contain
+ * payloaders pay\%d for each stream. The default implementation of this
+ * function returns the bin created from the launch parameter.
+ * @construct: the vmethod that will be called when the factory has to create the
+ * #GstRTSPMedia for @url. The default implementation of this
+ * function calls create_element to retrieve an element and then looks for
+ * pay\%d to create the streams.
+ * @create_pipeline: create a new pipeline or re-use an existing one and
+ * add the #GstRTSPMedia's element created by @construct to the pipeline.
+ * @configure: configure the media created with @construct. The default
+ * implementation will configure the 'shared' property of the media.
+ * @media_constructed: signal emitted when a media was constructed
+ * @media_configure: signal emitted when a media should be configured
+ *
+ * The #GstRTSPMediaFactory class structure.
+ */
+struct _GstRTSPMediaFactoryClass {
+ GObjectClass parent_class;
+
+ gchar * (*gen_key) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+
+ GstElement * (*create_element) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+ GstRTSPMedia * (*construct) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+ GstElement * (*create_pipeline) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+ void (*configure) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+
+ /* signals */
+ void (*media_constructed) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+ void (*media_configure) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_media_factory_get_type (void);
+
+/* creating the factory */
+
+GST_RTSP_SERVER_API
+GstRTSPMediaFactory * gst_rtsp_media_factory_new (void);
+
+/* configuring the factory */
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_launch (GstRTSPMediaFactory *factory,
+ const gchar *launch);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_media_factory_get_launch (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_permissions (GstRTSPMediaFactory *factory,
+ GstRTSPPermissions *permissions);
+
+GST_RTSP_SERVER_API
+GstRTSPPermissions * gst_rtsp_media_factory_get_permissions (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_add_role (GstRTSPMediaFactory *factory,
+ const gchar *role,
+ const gchar *fieldname, ...);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_add_role_from_structure (GstRTSPMediaFactory * factory,
+ GstStructure *structure);
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_shared (GstRTSPMediaFactory *factory,
+ gboolean shared);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_factory_is_shared (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_stop_on_disconnect (GstRTSPMediaFactory *factory,
+ gboolean stop_on_disconnect);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_factory_is_stop_on_disonnect (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_suspend_mode (GstRTSPMediaFactory *factory,
+ GstRTSPSuspendMode mode);
+
+GST_RTSP_SERVER_API
+GstRTSPSuspendMode gst_rtsp_media_factory_get_suspend_mode (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_eos_shutdown (GstRTSPMediaFactory *factory,
+ gboolean eos_shutdown);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_factory_is_eos_shutdown (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_profiles (GstRTSPMediaFactory *factory,
+ GstRTSPProfile profiles);
+
+GST_RTSP_SERVER_API
+GstRTSPProfile gst_rtsp_media_factory_get_profiles (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_protocols (GstRTSPMediaFactory *factory,
+ GstRTSPLowerTrans protocols);
+
+GST_RTSP_SERVER_API
+GstRTSPLowerTrans gst_rtsp_media_factory_get_protocols (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_address_pool (GstRTSPMediaFactory * factory,
+ GstRTSPAddressPool * pool);
+
+GST_RTSP_SERVER_API
+GstRTSPAddressPool * gst_rtsp_media_factory_get_address_pool (GstRTSPMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_multicast_iface (GstRTSPMediaFactory *factory, const gchar *multicast_iface);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_media_factory_get_multicast_iface (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_buffer_size (GstRTSPMediaFactory * factory,
+ guint size);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_media_factory_get_buffer_size (GstRTSPMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_retransmission_time (GstRTSPMediaFactory * factory,
+ GstClockTime time);
+
+GST_RTSP_SERVER_API
+GstClockTime gst_rtsp_media_factory_get_retransmission_time (GstRTSPMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_do_retransmission (GstRTSPMediaFactory * factory,
+ gboolean do_retransmission);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_factory_get_do_retransmission (GstRTSPMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_latency (GstRTSPMediaFactory * factory,
+ guint latency);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_media_factory_get_latency (GstRTSPMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_transport_mode (GstRTSPMediaFactory *factory,
+ GstRTSPTransportMode mode);
+
+GST_RTSP_SERVER_API
+GstRTSPTransportMode gst_rtsp_media_factory_get_transport_mode (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_media_gtype (GstRTSPMediaFactory * factory,
+ GType media_gtype);
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_media_factory_get_media_gtype (GstRTSPMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_clock (GstRTSPMediaFactory *factory,
+ GstClock * clock);
+
+GST_RTSP_SERVER_API
+GstClock * gst_rtsp_media_factory_get_clock (GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_publish_clock_mode (GstRTSPMediaFactory * factory, GstRTSPPublishClockMode mode);
+
+GST_RTSP_SERVER_API
+GstRTSPPublishClockMode gst_rtsp_media_factory_get_publish_clock_mode (GstRTSPMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_factory_set_max_mcast_ttl (GstRTSPMediaFactory * factory,
+ guint ttl);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_media_factory_get_max_mcast_ttl (GstRTSPMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_bind_mcast_address (GstRTSPMediaFactory * factory,
+ gboolean bind_mcast_addr);
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_factory_is_bind_mcast_address (GstRTSPMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_dscp_qos (GstRTSPMediaFactory * factory,
+ gint dscp_qos);
+GST_RTSP_SERVER_API
+gint gst_rtsp_media_factory_get_dscp_qos (GstRTSPMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_factory_set_enable_rtcp (GstRTSPMediaFactory * factory,
+ gboolean enable);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_factory_is_enable_rtcp (GstRTSPMediaFactory * factory);
+
+/* creating the media from the factory and a url */
+
+GST_RTSP_SERVER_API
+GstRTSPMedia * gst_rtsp_media_factory_construct (GstRTSPMediaFactory *factory,
+ const GstRTSPUrl *url);
+
+GST_RTSP_SERVER_API
+GstElement * gst_rtsp_media_factory_create_element (GstRTSPMediaFactory *factory,
+ const GstRTSPUrl *url);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPMediaFactory, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MEDIA_FACTORY_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2015 Centricular Ltd
+ * Author: Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media
+ * @short_description: The media pipeline
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
+ * #GstRTSPSessionMedia
+ *
+ * a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
+ * streaming to the clients. The actual data transfer is done by the
+ * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
+ *
+ * The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
+ * client does a DESCRIBE or SETUP of a resource.
+ *
+ * A media is created with gst_rtsp_media_new() that takes the element that will
+ * provide the streaming elements. For each of the streams, a new #GstRTSPStream
+ * object needs to be made with the gst_rtsp_media_create_stream() which takes
+ * the payloader element and the source pad that produces the RTP stream.
+ *
+ * The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
+ * prepare method will add rtpbin and sinks and sources to send and receive RTP
+ * and RTCP packets from the clients. Each stream srcpad is connected to an
+ * input into the internal rtpbin.
+ *
+ * It is also possible to dynamically create #GstRTSPStream objects during the
+ * prepare phase. With gst_rtsp_media_get_status() you can check the status of
+ * the prepare phase.
+ *
+ * After the media is prepared, it is ready for streaming. It will usually be
+ * managed in a session with gst_rtsp_session_manage_media(). See
+ * #GstRTSPSession and #GstRTSPSessionMedia.
+ *
+ * The state of the media can be controlled with gst_rtsp_media_set_state ().
+ * Seeking can be done with gst_rtsp_media_seek(), or gst_rtsp_media_seek_full()
+ * or gst_rtsp_media_seek_trickmode() for finer control of the seek.
+ *
+ * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
+ * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
+ * cleanly shut down.
+ *
+ * With gst_rtsp_media_set_shared(), the media can be shared between multiple
+ * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
+ * can be prepared again after an unprepare.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+
+#include <gst/app/gstappsrc.h>
+#include <gst/app/gstappsink.h>
+
+#include <gst/sdp/gstmikey.h>
+#include <gst/rtp/gstrtppayloads.h>
+
+#define AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
+#include "rtsp-media.h"
+#include "rtsp-server-internal.h"
+
+struct _GstRTSPMediaPrivate
+{
+ GMutex lock;
+ GCond cond;
+
+ /* the global lock is used to lock the entire media. This is needed by callers
+ such as rtsp_client to protect the media when it is shared by many clients.
+ The lock prevents that concurrenting clients messes up media.
+ Typically the lock is taken in external API calls such as SETUP */
+ GMutex global_lock;
+
+ /* protected by lock */
+ GstRTSPPermissions *permissions;
+ gboolean shared;
+ gboolean suspend_mode;
+ gboolean reusable;
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans protocols;
+ gboolean reused;
+ gboolean eos_shutdown;
+ guint buffer_size;
+ gint dscp_qos;
+ GstRTSPAddressPool *pool;
+ gchar *multicast_iface;
+ guint max_mcast_ttl;
+ gboolean bind_mcast_address;
+ gboolean enable_rtcp;
+ gboolean blocked;
+ GstRTSPTransportMode transport_mode;
+ gboolean stop_on_disconnect;
+ guint blocking_msg_received;
+
+ GstElement *element;
+ GRecMutex state_lock; /* locking order: state lock, lock */
+ GPtrArray *streams; /* protected by lock */
+ GList *dynamic; /* protected by lock */
+ GstRTSPMediaStatus status; /* protected by lock */
+ gint prepare_count;
+ gint n_active;
+ gboolean complete;
+ gboolean finishing_unprepare;
+
+ /* the pipeline for the media */
+ GstElement *pipeline;
+ GSource *source;
+ GstRTSPThread *thread;
+ GList *pending_pipeline_elements;
+
+ gboolean time_provider;
+ GstNetTimeProvider *nettime;
+
+ gboolean is_live;
+ GstClockTimeDiff seekable;
+ gboolean buffering;
+ GstState target_state;
+
+ /* RTP session manager */
+ GstElement *rtpbin;
+
+ /* the range of media */
+ GstRTSPTimeRange range; /* protected by lock */
+ GstClockTime range_start;
+ GstClockTime range_stop;
+
+ GList *payloads; /* protected by lock */
+ GstClockTime rtx_time; /* protected by lock */
+ gboolean do_retransmission; /* protected by lock */
+ guint latency; /* protected by lock */
+ GstClock *clock; /* protected by lock */
+ gboolean do_rate_control; /* protected by lock */
+ GstRTSPPublishClockMode publish_clock_mode;
+
+ /* Dynamic element handling */
+ guint nb_dynamic_elements;
+ guint no_more_pads_pending;
+ gboolean expected_async_done;
+};
+
+#define DEFAULT_SHARED FALSE
+#define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
+#define DEFAULT_REUSABLE FALSE
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+ GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_EOS_SHUTDOWN FALSE
+#define DEFAULT_BUFFER_SIZE 0x80000
+#define DEFAULT_DSCP_QOS (-1)
+#define DEFAULT_TIME_PROVIDER FALSE
+#define DEFAULT_LATENCY 200
+#define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
+#define DEFAULT_STOP_ON_DISCONNECT TRUE
+#define DEFAULT_MAX_MCAST_TTL 255
+#define DEFAULT_BIND_MCAST_ADDRESS FALSE
+#define DEFAULT_DO_RATE_CONTROL TRUE
+#define DEFAULT_ENABLE_RTCP TRUE
+
+#define DEFAULT_DO_RETRANSMISSION FALSE
+
+/* define to dump received RTCP packets */
+#undef DUMP_STATS
+
+enum
+{
+ PROP_0,
+ PROP_SHARED,
+ PROP_SUSPEND_MODE,
+ PROP_REUSABLE,
+ PROP_PROFILES,
+ PROP_PROTOCOLS,
+ PROP_EOS_SHUTDOWN,
+ PROP_BUFFER_SIZE,
+ PROP_ELEMENT,
+ PROP_TIME_PROVIDER,
+ PROP_LATENCY,
+ PROP_TRANSPORT_MODE,
+ PROP_STOP_ON_DISCONNECT,
+ PROP_CLOCK,
+ PROP_MAX_MCAST_TTL,
+ PROP_BIND_MCAST_ADDRESS,
+ PROP_DSCP_QOS,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_NEW_STREAM,
+ SIGNAL_REMOVED_STREAM,
+ SIGNAL_PREPARED,
+ SIGNAL_UNPREPARED,
+ SIGNAL_TARGET_STATE,
+ SIGNAL_NEW_STATE,
+ SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
+#define GST_CAT_DEFAULT rtsp_media_debug
+
+static void gst_rtsp_media_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_finalize (GObject * obj);
+
+static gboolean default_handle_message (GstRTSPMedia * media,
+ GstMessage * message);
+static void finish_unprepare (GstRTSPMedia * media);
+static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
+static gboolean default_unprepare (GstRTSPMedia * media);
+static gboolean default_suspend (GstRTSPMedia * media);
+static gboolean default_unsuspend (GstRTSPMedia * media);
+static gboolean default_convert_range (GstRTSPMedia * media,
+ GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
+static gboolean default_query_position (GstRTSPMedia * media,
+ gint64 * position);
+static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
+static GstElement *default_create_rtpbin (GstRTSPMedia * media);
+static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info);
+static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
+
+static gboolean wait_preroll (GstRTSPMedia * media);
+
+static GstElement *find_payload_element (GstElement * payloader, GstPad * pad);
+
+static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
+
+static gboolean check_complete (GstRTSPMedia * media);
+
+#define C_ENUM(v) ((gint) v)
+
+#define TRICKMODE_FLAGS (GST_SEEK_FLAG_TRICKMODE | GST_SEEK_FLAG_TRICKMODE_KEY_UNITS | GST_SEEK_FLAG_TRICKMODE_FORWARD_PREDICTED)
+
+GType
+gst_rtsp_suspend_mode_get_type (void)
+{
+ static gsize id = 0;
+ static const GEnumValue values[] = {
+ {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
+ {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
+ "pause"},
+ {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
+ "reset"},
+ {0, NULL, NULL}
+ };
+
+ if (g_once_init_enter (&id)) {
+ GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
+ g_once_init_leave (&id, tmp);
+ }
+ return (GType) id;
+}
+
+#define C_FLAGS(v) ((guint) v)
+
+GType
+gst_rtsp_transport_mode_get_type (void)
+{
+ static gsize id = 0;
+ static const GFlagsValue values[] = {
+ {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
+ "play"},
+ {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
+ "record"},
+ {0, NULL, NULL}
+ };
+
+ if (g_once_init_enter (&id)) {
+ GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
+ g_once_init_leave (&id, tmp);
+ }
+ return (GType) id;
+}
+
+GType
+gst_rtsp_publish_clock_mode_get_type (void)
+{
+ static gsize id = 0;
+ static const GEnumValue values[] = {
+ {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_NONE),
+ "GST_RTSP_PUBLISH_CLOCK_MODE_NONE", "none"},
+ {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK),
+ "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK",
+ "clock"},
+ {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET),
+ "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET",
+ "clock-and-offset"},
+ {0, NULL, NULL}
+ };
+
+ if (g_once_init_enter (&id)) {
+ GType tmp = g_enum_register_static ("GstRTSPPublishClockMode", values);
+ g_once_init_leave (&id, tmp);
+ }
+ return (GType) id;
+}
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_media_get_property;
+ gobject_class->set_property = gst_rtsp_media_set_property;
+ gobject_class->finalize = gst_rtsp_media_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_SHARED,
+ g_param_spec_boolean ("shared", "Shared",
+ "If this media pipeline can be shared", DEFAULT_SHARED,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
+ g_param_spec_enum ("suspend-mode", "Suspend Mode",
+ "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
+ DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_REUSABLE,
+ g_param_spec_boolean ("reusable", "Reusable",
+ "If this media pipeline can be reused after an unprepare",
+ DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
+ g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
+ "Send an EOS event to the pipeline before unpreparing",
+ DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
+ g_param_spec_uint ("buffer-size", "Buffer Size",
+ "The kernel UDP buffer size to use", 0, G_MAXUINT,
+ DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_ELEMENT,
+ g_param_spec_object ("element", "The Element",
+ "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
+ G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
+ g_param_spec_boolean ("time-provider", "Time Provider",
+ "Use a NetTimeProvider for clients",
+ DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_LATENCY,
+ g_param_spec_uint ("latency", "Latency",
+ "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
+ DEFAULT_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
+ g_param_spec_flags ("transport-mode", "Transport Mode",
+ "If this media pipeline can be used for PLAY or RECORD",
+ GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
+ g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
+ "If this media pipeline should be stopped "
+ "when a client disconnects without TEARDOWN",
+ DEFAULT_STOP_ON_DISCONNECT,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_CLOCK,
+ g_param_spec_object ("clock", "Clock",
+ "Clock to be used by the media pipeline",
+ GST_TYPE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MAX_MCAST_TTL,
+ g_param_spec_uint ("max-mcast-ttl", "Maximum multicast ttl",
+ "The maximum time-to-live value of outgoing multicast packets", 1,
+ 255, DEFAULT_MAX_MCAST_TTL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_BIND_MCAST_ADDRESS,
+ g_param_spec_boolean ("bind-mcast-address", "Bind mcast address",
+ "Whether the multicast sockets should be bound to multicast addresses "
+ "or INADDR_ANY",
+ DEFAULT_BIND_MCAST_ADDRESS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_DSCP_QOS,
+ g_param_spec_int ("dscp-qos", "DSCP QoS",
+ "The IP DSCP field to use for each related stream", -1, 63,
+ DEFAULT_DSCP_QOS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
+ g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL, NULL,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
+
+ gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
+ g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
+
+ gst_rtsp_media_signals[SIGNAL_PREPARED] =
+ g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL, NULL,
+ G_TYPE_NONE, 0, G_TYPE_NONE);
+
+ gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
+ g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL, NULL,
+ G_TYPE_NONE, 0, G_TYPE_NONE);
+
+ gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
+ g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_INT);
+
+ gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
+ g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL, NULL,
+ G_TYPE_NONE, 1, G_TYPE_INT);
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
+
+ klass->handle_message = default_handle_message;
+ klass->prepare = default_prepare;
+ klass->unprepare = default_unprepare;
+ klass->suspend = default_suspend;
+ klass->unsuspend = default_unsuspend;
+ klass->convert_range = default_convert_range;
+ klass->query_position = default_query_position;
+ klass->query_stop = default_query_stop;
+ klass->create_rtpbin = default_create_rtpbin;
+ klass->setup_sdp = default_setup_sdp;
+ klass->handle_sdp = default_handle_sdp;
+}
+
+static void
+gst_rtsp_media_init (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = gst_rtsp_media_get_instance_private (media);
+
+ media->priv = priv;
+
+ priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
+ g_mutex_init (&priv->lock);
+ g_mutex_init (&priv->global_lock);
+ g_cond_init (&priv->cond);
+ g_rec_mutex_init (&priv->state_lock);
+
+ priv->shared = DEFAULT_SHARED;
+ priv->suspend_mode = DEFAULT_SUSPEND_MODE;
+ priv->reusable = DEFAULT_REUSABLE;
+ priv->profiles = DEFAULT_PROFILES;
+ priv->protocols = DEFAULT_PROTOCOLS;
+ priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
+ priv->buffer_size = DEFAULT_BUFFER_SIZE;
+ priv->time_provider = DEFAULT_TIME_PROVIDER;
+ priv->transport_mode = DEFAULT_TRANSPORT_MODE;
+ priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
+ priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
+ priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
+ priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
+ priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
+ priv->enable_rtcp = DEFAULT_ENABLE_RTCP;
+ priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
+ priv->dscp_qos = DEFAULT_DSCP_QOS;
+ priv->expected_async_done = FALSE;
+ priv->blocking_msg_received = 0;
+}
+
+static void
+gst_rtsp_media_finalize (GObject * obj)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMedia *media;
+
+ media = GST_RTSP_MEDIA (obj);
+ priv = media->priv;
+
+ GST_INFO ("finalize media %p", media);
+
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+
+ g_ptr_array_unref (priv->streams);
+
+ g_list_free_full (priv->dynamic, gst_object_unref);
+ g_list_free_full (priv->pending_pipeline_elements, gst_object_unref);
+
+ if (priv->pipeline)
+ gst_object_unref (priv->pipeline);
+ if (priv->nettime)
+ gst_object_unref (priv->nettime);
+ gst_object_unref (priv->element);
+ if (priv->pool)
+ g_object_unref (priv->pool);
+ if (priv->payloads)
+ g_list_free (priv->payloads);
+ if (priv->clock)
+ gst_object_unref (priv->clock);
+ g_free (priv->multicast_iface);
+ g_mutex_clear (&priv->lock);
+ g_mutex_clear (&priv->global_lock);
+ g_cond_clear (&priv->cond);
+ g_rec_mutex_clear (&priv->state_lock);
+
+ G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_media_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMedia *media = GST_RTSP_MEDIA (object);
+
+ switch (propid) {
+ case PROP_ELEMENT:
+ g_value_set_object (value, media->priv->element);
+ break;
+ case PROP_SHARED:
+ g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
+ break;
+ case PROP_SUSPEND_MODE:
+ g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
+ break;
+ case PROP_REUSABLE:
+ g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
+ break;
+ case PROP_PROFILES:
+ g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
+ break;
+ case PROP_PROTOCOLS:
+ g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
+ break;
+ case PROP_EOS_SHUTDOWN:
+ g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
+ break;
+ case PROP_BUFFER_SIZE:
+ g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
+ break;
+ case PROP_TIME_PROVIDER:
+ g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
+ break;
+ case PROP_LATENCY:
+ g_value_set_uint (value, gst_rtsp_media_get_latency (media));
+ break;
+ case PROP_TRANSPORT_MODE:
+ g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
+ break;
+ case PROP_STOP_ON_DISCONNECT:
+ g_value_set_boolean (value, gst_rtsp_media_is_stop_on_disconnect (media));
+ break;
+ case PROP_CLOCK:
+ g_value_take_object (value, gst_rtsp_media_get_clock (media));
+ break;
+ case PROP_MAX_MCAST_TTL:
+ g_value_set_uint (value, gst_rtsp_media_get_max_mcast_ttl (media));
+ break;
+ case PROP_BIND_MCAST_ADDRESS:
+ g_value_set_boolean (value, gst_rtsp_media_is_bind_mcast_address (media));
+ break;
+ case PROP_DSCP_QOS:
+ g_value_set_int (value, gst_rtsp_media_get_dscp_qos (media));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_media_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMedia *media = GST_RTSP_MEDIA (object);
+
+ switch (propid) {
+ case PROP_ELEMENT:
+ media->priv->element = g_value_get_object (value);
+ gst_object_ref_sink (media->priv->element);
+ break;
+ case PROP_SHARED:
+ gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
+ break;
+ case PROP_SUSPEND_MODE:
+ gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
+ break;
+ case PROP_REUSABLE:
+ gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
+ break;
+ case PROP_PROFILES:
+ gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
+ break;
+ case PROP_PROTOCOLS:
+ gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
+ break;
+ case PROP_EOS_SHUTDOWN:
+ gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
+ break;
+ case PROP_BUFFER_SIZE:
+ gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
+ break;
+ case PROP_TIME_PROVIDER:
+ gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
+ break;
+ case PROP_LATENCY:
+ gst_rtsp_media_set_latency (media, g_value_get_uint (value));
+ break;
+ case PROP_TRANSPORT_MODE:
+ gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
+ break;
+ case PROP_STOP_ON_DISCONNECT:
+ gst_rtsp_media_set_stop_on_disconnect (media,
+ g_value_get_boolean (value));
+ break;
+ case PROP_CLOCK:
+ gst_rtsp_media_set_clock (media, g_value_get_object (value));
+ break;
+ case PROP_MAX_MCAST_TTL:
+ gst_rtsp_media_set_max_mcast_ttl (media, g_value_get_uint (value));
+ break;
+ case PROP_BIND_MCAST_ADDRESS:
+ gst_rtsp_media_set_bind_mcast_address (media,
+ g_value_get_boolean (value));
+ break;
+ case PROP_DSCP_QOS:
+ gst_rtsp_media_set_dscp_qos (media, g_value_get_int (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+typedef struct
+{
+ gint64 position;
+ gboolean complete_streams_only;
+ gboolean ret;
+} DoQueryPositionData;
+
+static void
+do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
+{
+ gint64 tmp;
+
+ if (!gst_rtsp_stream_is_sender (stream))
+ return;
+
+ if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream)) {
+ GST_DEBUG_OBJECT (stream, "stream not complete, do not query position");
+ return;
+ }
+
+ if (gst_rtsp_stream_query_position (stream, &tmp)) {
+ data->position = MIN (data->position, tmp);
+ data->ret = TRUE;
+ }
+
+ GST_INFO_OBJECT (stream, "media position: %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (data->position));
+}
+
+static gboolean
+default_query_position (GstRTSPMedia * media, gint64 * position)
+{
+ GstRTSPMediaPrivate *priv;
+ DoQueryPositionData data;
+
+ priv = media->priv;
+
+ data.position = G_MAXINT64;
+ data.ret = FALSE;
+
+ /* if the media is complete, i.e. one or more streams have been configured
+ * with sinks, then we want to query the position on those streams only.
+ * a query on an incmplete stream may return a position that originates from
+ * an earlier preroll */
+ if (check_complete (media))
+ data.complete_streams_only = TRUE;
+ else
+ data.complete_streams_only = FALSE;
+
+ g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
+
+ if (!data.ret)
+ *position = GST_CLOCK_TIME_NONE;
+ else
+ *position = data.position;
+
+ return data.ret;
+}
+
+typedef struct
+{
+ gint64 stop;
+ gboolean ret;
+} DoQueryStopData;
+
+static void
+do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
+{
+ gint64 tmp = 0;
+
+ if (gst_rtsp_stream_query_stop (stream, &tmp)) {
+ data->stop = MAX (data->stop, tmp);
+ data->ret = TRUE;
+ }
+}
+
+static gboolean
+default_query_stop (GstRTSPMedia * media, gint64 * stop)
+{
+ GstRTSPMediaPrivate *priv;
+ DoQueryStopData data;
+
+ priv = media->priv;
+
+ data.stop = -1;
+ data.ret = FALSE;
+
+ g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
+
+ *stop = data.stop;
+
+ return data.ret;
+}
+
+static GstElement *
+default_create_rtpbin (GstRTSPMedia * media)
+{
+ GstElement *rtpbin;
+
+ rtpbin = gst_element_factory_make ("rtpbin", NULL);
+
+ return rtpbin;
+}
+
+/* Must be called with priv->lock */
+static gboolean
+is_receive_only (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gboolean receive_only = TRUE;
+ guint i;
+
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ if (gst_rtsp_stream_is_sender (stream) ||
+ !gst_rtsp_stream_is_receiver (stream)) {
+ receive_only = FALSE;
+ break;
+ }
+ }
+
+ return receive_only;
+}
+
+/* must be called with state lock */
+static void
+check_seekable (GstRTSPMedia * media)
+{
+ GstQuery *query;
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* Update the seekable state of the pipeline in case it changed */
+ if (is_receive_only (media)) {
+ /* TODO: Seeking for "receive-only"? */
+ priv->seekable = -1;
+ } else {
+ guint i, n = priv->streams->len;
+
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+
+ if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
+ GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
+ priv->seekable = -1;
+ g_mutex_unlock (&priv->lock);
+ return;
+ }
+ }
+ }
+
+ query = gst_query_new_seeking (GST_FORMAT_TIME);
+ if (gst_element_query (priv->pipeline, query)) {
+ GstFormat format;
+ gboolean seekable;
+ gint64 start, end;
+
+ gst_query_parse_seeking (query, &format, &seekable, &start, &end);
+ priv->seekable = seekable ? G_MAXINT64 : 0;
+ } else if (priv->streams->len) {
+ gboolean seekable = TRUE;
+ guint i, n = priv->streams->len;
+
+ GST_DEBUG_OBJECT (media, "Checking %d streams", n);
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ seekable &= gst_rtsp_stream_seekable (stream);
+ }
+ priv->seekable = seekable ? G_MAXINT64 : -1;
+ }
+
+ GST_DEBUG_OBJECT (media, "seekable:%" G_GINT64_FORMAT, priv->seekable);
+ g_mutex_unlock (&priv->lock);
+ gst_query_unref (query);
+}
+
+/* must be called with state lock */
+static gboolean
+check_complete (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ guint i, n = priv->streams->len;
+
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+
+ if (gst_rtsp_stream_is_complete (stream))
+ return TRUE;
+ }
+
+ return FALSE;
+}
+
+/* must be called with state lock and private lock */
+static void
+collect_media_stats (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gint64 position = 0, stop = -1;
+
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+ priv->status != GST_RTSP_MEDIA_STATUS_PREPARING) {
+ return;
+ }
+
+ priv->range.unit = GST_RTSP_RANGE_NPT;
+
+ GST_INFO ("collect media stats");
+
+ if (priv->is_live) {
+ priv->range.min.type = GST_RTSP_TIME_NOW;
+ priv->range.min.seconds = -1;
+ priv->range_start = -1;
+ priv->range.max.type = GST_RTSP_TIME_END;
+ priv->range.max.seconds = -1;
+ priv->range_stop = -1;
+ } else {
+ GstRTSPMediaClass *klass;
+ gboolean ret;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ /* get the position */
+ ret = FALSE;
+ if (klass->query_position)
+ ret = klass->query_position (media, &position);
+
+ if (!ret) {
+ GST_INFO ("position query failed");
+ position = 0;
+ }
+
+ /* get the current segment stop */
+ ret = FALSE;
+ if (klass->query_stop)
+ ret = klass->query_stop (media, &stop);
+
+ if (!ret) {
+ GST_INFO ("stop query failed");
+ stop = -1;
+ }
+
+ GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
+
+ if (position == -1) {
+ priv->range.min.type = GST_RTSP_TIME_NOW;
+ priv->range.min.seconds = -1;
+ priv->range_start = -1;
+ } else {
+ priv->range.min.type = GST_RTSP_TIME_SECONDS;
+ priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
+ priv->range_start = position;
+ }
+ if (stop == -1) {
+ priv->range.max.type = GST_RTSP_TIME_END;
+ priv->range.max.seconds = -1;
+ priv->range_stop = -1;
+ } else {
+ priv->range.max.type = GST_RTSP_TIME_SECONDS;
+ priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
+ priv->range_stop = stop;
+ }
+ g_mutex_unlock (&priv->lock);
+ check_seekable (media);
+ g_mutex_lock (&priv->lock);
+ }
+}
+
+/**
+ * gst_rtsp_media_new:
+ * @element: (transfer full): a #GstElement
+ *
+ * Create a new #GstRTSPMedia instance. @element is the bin element that
+ * provides the different streams. The #GstRTSPMedia object contains the
+ * element to produce RTP data for one or more related (audio/video/..)
+ * streams.
+ *
+ * Ownership is taken of @element.
+ *
+ * Returns: (transfer full): a new #GstRTSPMedia object.
+ */
+GstRTSPMedia *
+gst_rtsp_media_new (GstElement * element)
+{
+ GstRTSPMedia *result;
+
+ g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
+
+ result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_get_element:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the element that was used when constructing @media.
+ *
+ * Returns: (transfer full): a #GstElement. Unref after usage.
+ */
+GstElement *
+gst_rtsp_media_get_element (GstRTSPMedia * media)
+{
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ return gst_object_ref (media->priv->element);
+}
+
+/**
+ * gst_rtsp_media_take_pipeline:
+ * @media: a #GstRTSPMedia
+ * @pipeline: (transfer floating): a #GstPipeline
+ *
+ * Set @pipeline as the #GstPipeline for @media. Ownership is
+ * taken of @pipeline.
+ */
+void
+gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
+{
+ GstRTSPMediaPrivate *priv;
+ GstElement *old;
+ GstNetTimeProvider *nettime;
+ GList *l;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+ g_return_if_fail (GST_IS_PIPELINE (pipeline));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ old = priv->pipeline;
+ priv->pipeline = gst_object_ref_sink (GST_ELEMENT_CAST (pipeline));
+ nettime = priv->nettime;
+ priv->nettime = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ gst_object_unref (old);
+
+ if (nettime)
+ gst_object_unref (nettime);
+
+ gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
+
+ for (l = priv->pending_pipeline_elements; l; l = l->next) {
+ gst_bin_add (GST_BIN_CAST (pipeline), l->data);
+ }
+ g_list_free (priv->pending_pipeline_elements);
+ priv->pending_pipeline_elements = NULL;
+}
+
+/**
+ * gst_rtsp_media_set_permissions:
+ * @media: a #GstRTSPMedia
+ * @permissions: (transfer none) (nullable): a #GstRTSPPermissions
+ *
+ * Set @permissions on @media.
+ */
+void
+gst_rtsp_media_set_permissions (GstRTSPMedia * media,
+ GstRTSPPermissions * permissions)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+ if ((priv->permissions = permissions))
+ gst_rtsp_permissions_ref (permissions);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_permissions:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the permissions object from @media.
+ *
+ * Returns: (transfer full) (nullable): a #GstRTSPPermissions object, unref after usage.
+ */
+GstRTSPPermissions *
+gst_rtsp_media_get_permissions (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPPermissions *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->permissions))
+ gst_rtsp_permissions_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_set_suspend_mode:
+ * @media: a #GstRTSPMedia
+ * @mode: the new #GstRTSPSuspendMode
+ *
+ * Control how @ media will be suspended after the SDP has been generated and
+ * after a PAUSE request has been performed.
+ *
+ * Media must be unprepared when setting the suspend mode.
+ */
+void
+gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto was_prepared;
+ priv->suspend_mode = mode;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return;
+
+ /* ERRORS */
+was_prepared:
+ {
+ GST_WARNING ("media %p was prepared", media);
+ g_rec_mutex_unlock (&priv->state_lock);
+ }
+}
+
+/**
+ * gst_rtsp_media_get_suspend_mode:
+ * @media: a #GstRTSPMedia
+ *
+ * Get how @media will be suspended.
+ *
+ * Returns: #GstRTSPSuspendMode.
+ */
+GstRTSPSuspendMode
+gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPSuspendMode res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ res = priv->suspend_mode;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_shared:
+ * @media: a #GstRTSPMedia
+ * @shared: the new value
+ *
+ * Set or unset if the pipeline for @media can be shared will multiple clients.
+ * When @shared is %TRUE, client requests for this media will share the media
+ * pipeline.
+ */
+void
+gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->shared = shared;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_shared:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media can be shared between multiple clients.
+ *
+ * Returns: %TRUE if the media can be shared between clients.
+ */
+gboolean
+gst_rtsp_media_is_shared (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->shared;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_reusable:
+ * @media: a #GstRTSPMedia
+ * @reusable: the new value
+ *
+ * Set or unset if the pipeline for @media can be reused after the pipeline has
+ * been unprepared.
+ */
+void
+gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->reusable = reusable;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_reusable:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media can be reused after an unprepare.
+ *
+ * Returns: %TRUE if the media can be reused
+ */
+gboolean
+gst_rtsp_media_is_reusable (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->reusable;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+static void
+do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
+{
+ gst_rtsp_stream_set_profiles (stream, *profiles);
+}
+
+/**
+ * gst_rtsp_media_set_profiles:
+ * @media: a #GstRTSPMedia
+ * @profiles: the new flags
+ *
+ * Configure the allowed lower transport for @media.
+ */
+void
+gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->profiles = profiles;
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_profiles:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the allowed profiles of @media.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_media_get_profiles (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->profiles;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+static void
+do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
+{
+ gst_rtsp_stream_set_protocols (stream, *protocols);
+}
+
+/**
+ * gst_rtsp_media_set_protocols:
+ * @media: a #GstRTSPMedia
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @media.
+ */
+void
+gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->protocols = protocols;
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_protocols:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the allowed protocols of @media.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_media_get_protocols (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPLowerTrans res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
+ GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->protocols;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_eos_shutdown:
+ * @media: a #GstRTSPMedia
+ * @eos_shutdown: the new value
+ *
+ * Set or unset if an EOS event will be sent to the pipeline for @media before
+ * it is unprepared.
+ */
+void
+gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->eos_shutdown = eos_shutdown;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_eos_shutdown:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media will send an EOS down the pipeline before
+ * unpreparing.
+ *
+ * Returns: %TRUE if the media will send EOS before unpreparing.
+ */
+gboolean
+gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->eos_shutdown;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_buffer_size:
+ * @media: a #GstRTSPMedia
+ * @size: the new value
+ *
+ * Set the kernel UDP buffer size.
+ */
+void
+gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ GST_LOG_OBJECT (media, "set buffer size %u", size);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->buffer_size = size;
+
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ gst_rtsp_stream_set_buffer_size (stream, size);
+ }
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_buffer_size:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the kernel UDP buffer size.
+ *
+ * Returns: the kernel UDP buffer size.
+ */
+guint
+gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->buffer_size;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+static void
+do_set_dscp_qos (GstRTSPStream * stream, gint * dscp_qos)
+{
+ gst_rtsp_stream_set_dscp_qos (stream, *dscp_qos);
+}
+
+/**
+ * gst_rtsp_media_set_dscp_qos:
+ * @media: a #GstRTSPMedia
+ * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
+ *
+ * Configure the dscp qos of attached streams to @dscp_qos.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_media_set_dscp_qos (GstRTSPMedia * media, gint dscp_qos)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ GST_LOG_OBJECT (media, "set DSCP QoS %d", dscp_qos);
+
+ if (dscp_qos < -1 || dscp_qos > 63) {
+ GST_WARNING_OBJECT (media, "trying to set illegal dscp qos %d", dscp_qos);
+ return;
+ }
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->dscp_qos = dscp_qos;
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_dscp_qos, &dscp_qos);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_dscp_qos:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the configured DSCP QoS of attached media.
+ *
+ * Returns: the DSCP QoS value of attached streams or -1 if disabled.
+ *
+ * Since: 1.18
+ */
+gint
+gst_rtsp_media_get_dscp_qos (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_unlock (&priv->lock);
+ res = priv->dscp_qos;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_stop_on_disconnect:
+ * @media: a #GstRTSPMedia
+ * @stop_on_disconnect: the new value
+ *
+ * Set or unset if the pipeline for @media should be stopped when a
+ * client disconnects without sending TEARDOWN.
+ */
+void
+gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
+ gboolean stop_on_disconnect)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->stop_on_disconnect = stop_on_disconnect;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_stop_on_disconnect:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media will be stopped when a client disconnects
+ * without sending TEARDOWN.
+ *
+ * Returns: %TRUE if the media will be stopped when a client disconnects
+ * without sending TEARDOWN.
+ */
+gboolean
+gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), TRUE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->stop_on_disconnect;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_retransmission_time:
+ * @media: a #GstRTSPMedia
+ * @time: the new value
+ *
+ * Set the amount of time to store retransmission packets.
+ */
+void
+gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->rtx_time = time;
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+
+ gst_rtsp_stream_set_retransmission_time (stream, time);
+ }
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_retransmission_time:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the amount of time to store retransmission data.
+ *
+ * Returns: the amount of time to store retransmission data.
+ */
+GstClockTime
+gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstClockTime res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->rtx_time;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_do_retransmission:
+ *
+ * Set whether retransmission requests will be sent
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
+ gboolean do_retransmission)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->do_retransmission = do_retransmission;
+
+ if (priv->rtpbin)
+ g_object_set (priv->rtpbin, "do-retransmission", do_retransmission, NULL);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_do_retransmission:
+ *
+ * Returns: Whether retransmission requests will be sent
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->do_retransmission;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+static void
+update_stream_storage_size (GstRTSPMedia * media, GstRTSPStream * stream,
+ guint sessid)
+{
+ GObject *storage = NULL;
+
+ g_signal_emit_by_name (G_OBJECT (media->priv->rtpbin), "get-storage",
+ sessid, &storage);
+
+ if (storage) {
+ guint64 size_time = 0;
+
+ if (!gst_rtsp_stream_is_tcp_receiver (stream))
+ size_time = (media->priv->latency + 50) * GST_MSECOND;
+
+ g_object_set (storage, "size-time", size_time, NULL);
+
+ g_object_unref (storage);
+ }
+}
+
+/**
+ * gst_rtsp_media_set_latency:
+ * @media: a #GstRTSPMedia
+ * @latency: latency in milliseconds
+ *
+ * Configure the latency used for receiving media.
+ */
+void
+gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ GST_LOG_OBJECT (media, "set latency %ums", latency);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->latency = latency;
+ if (priv->rtpbin) {
+ g_object_set (priv->rtpbin, "latency", latency, NULL);
+
+ for (i = 0; i < media->priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
+ update_stream_storage_size (media, stream, i);
+ }
+ }
+
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_latency:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the latency that is used for receiving media.
+ *
+ * Returns: latency in milliseconds
+ */
+guint
+gst_rtsp_media_get_latency (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->latency;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_use_time_provider:
+ * @media: a #GstRTSPMedia
+ * @time_provider: if a #GstNetTimeProvider should be used
+ *
+ * Set @media to provide a #GstNetTimeProvider.
+ */
+void
+gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->time_provider = time_provider;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_time_provider:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
+ *
+ * Use gst_rtsp_media_get_time_provider() to get the network clock.
+ *
+ * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
+ */
+gboolean
+gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->time_provider;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_clock:
+ * @media: a #GstRTSPMedia
+ * @clock: (nullable): #GstClock to be used
+ *
+ * Configure the clock used for the media.
+ */
+void
+gst_rtsp_media_set_clock (GstRTSPMedia * media, GstClock * clock)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+ g_return_if_fail (GST_IS_CLOCK (clock) || clock == NULL);
+
+ GST_LOG_OBJECT (media, "setting clock %" GST_PTR_FORMAT, clock);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->clock)
+ gst_object_unref (priv->clock);
+ priv->clock = clock ? gst_object_ref (clock) : NULL;
+ if (priv->pipeline) {
+ if (clock)
+ gst_pipeline_use_clock (GST_PIPELINE_CAST (priv->pipeline), clock);
+ else
+ gst_pipeline_auto_clock (GST_PIPELINE_CAST (priv->pipeline));
+ }
+
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_set_publish_clock_mode:
+ * @media: a #GstRTSPMedia
+ * @mode: the clock publish mode
+ *
+ * Sets if and how the media clock should be published according to RFC7273.
+ *
+ * Since: 1.8
+ */
+void
+gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media,
+ GstRTSPPublishClockMode mode)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i, n;
+
+ priv = media->priv;
+ g_mutex_lock (&priv->lock);
+ priv->publish_clock_mode = mode;
+
+ n = priv->streams->len;
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+
+ gst_rtsp_stream_set_publish_clock_mode (stream, mode);
+ }
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_publish_clock_mode:
+ * @media: a #GstRTSPMedia
+ *
+ * Gets if and how the media clock should be published according to RFC7273.
+ *
+ * Returns: The GstRTSPPublishClockMode
+ *
+ * Since: 1.8
+ */
+GstRTSPPublishClockMode
+gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPPublishClockMode ret;
+
+ priv = media->priv;
+ g_mutex_lock (&priv->lock);
+ ret = priv->publish_clock_mode;
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_media_set_address_pool:
+ * @media: a #GstRTSPMedia
+ * @pool: (transfer none) (nullable): a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @media.
+ */
+void
+gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
+ GstRTSPAddressPool * pool)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPAddressPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ GST_LOG_OBJECT (media, "set address pool %p", pool);
+
+ g_mutex_lock (&priv->lock);
+ if ((old = priv->pool) != pool)
+ priv->pool = pool ? g_object_ref (pool) : NULL;
+ else
+ old = NULL;
+ g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
+ pool);
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_media_get_address_pool:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @media.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @media.
+ * g_object_unref() after usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPAddressPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->pool))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_set_multicast_iface:
+ * @media: a #GstRTSPMedia
+ * @multicast_iface: (transfer none) (nullable): a multicast interface name
+ *
+ * configure @multicast_iface to be used for @media.
+ */
+void
+gst_rtsp_media_set_multicast_iface (GstRTSPMedia * media,
+ const gchar * multicast_iface)
+{
+ GstRTSPMediaPrivate *priv;
+ gchar *old;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ GST_LOG_OBJECT (media, "set multicast interface %s", multicast_iface);
+
+ g_mutex_lock (&priv->lock);
+ if ((old = priv->multicast_iface) != multicast_iface)
+ priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
+ else
+ old = NULL;
+ g_ptr_array_foreach (priv->streams,
+ (GFunc) gst_rtsp_stream_set_multicast_iface, (gchar *) multicast_iface);
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_free (old);
+}
+
+/**
+ * gst_rtsp_media_get_multicast_iface:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the multicast interface used for @media.
+ *
+ * Returns: (transfer full) (nullable): the multicast interface for @media.
+ * g_free() after usage.
+ */
+gchar *
+gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->multicast_iface))
+ result = g_strdup (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_set_max_mcast_ttl:
+ * @media: a #GstRTSPMedia
+ * @ttl: the new multicast ttl value
+ *
+ * Set the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: %TRUE if the requested ttl has been set successfully.
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia * media, guint ttl)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ GST_LOG_OBJECT (media, "set max mcast ttl %u", ttl);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+
+ if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
+ GST_WARNING_OBJECT (media, "The reqested mcast TTL value is not valid.");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+ priv->max_mcast_ttl = ttl;
+
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ gst_rtsp_stream_set_max_mcast_ttl (stream, ttl);
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_media_get_max_mcast_ttl:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Since: 1.16
+ */
+guint
+gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->max_mcast_ttl;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_bind_mcast_address:
+ * @media: a #GstRTSPMedia
+ * @bind_mcast_addr: the new value
+ *
+ * Decide whether the multicast socket should be bound to a multicast address or
+ * INADDR_ANY.
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia * media,
+ gboolean bind_mcast_addr)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->bind_mcast_address = bind_mcast_addr;
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ gst_rtsp_stream_set_bind_mcast_address (stream, bind_mcast_addr);
+ }
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_bind_mcast_address:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if multicast sockets are configured to be bound to multicast addresses.
+ *
+ * Returns: %TRUE if multicast sockets are configured to be bound to multicast addresses.
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = priv->bind_mcast_address;
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+void
+gst_rtsp_media_set_enable_rtcp (GstRTSPMedia * media, gboolean enable)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->enable_rtcp = enable;
+ g_mutex_unlock (&priv->lock);
+}
+
+static GList *
+_find_payload_types (GstRTSPMedia * media)
+{
+ gint i, n;
+ GQueue queue = G_QUEUE_INIT;
+
+ n = media->priv->streams->len;
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
+ guint pt = gst_rtsp_stream_get_pt (stream);
+
+ g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
+ }
+
+ return queue.head;
+}
+
+static guint
+_next_available_pt (GList * payloads)
+{
+ guint i;
+
+ for (i = 96; i <= 127; i++) {
+ GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
+ if (!iter)
+ return GPOINTER_TO_UINT (i);
+ }
+
+ return 0;
+}
+
+/**
+ * gst_rtsp_media_collect_streams:
+ * @media: a #GstRTSPMedia
+ *
+ * Find all payloader elements, they should be named pay\%d in the
+ * element of @media, and create #GstRTSPStreams for them.
+ *
+ * Collect all dynamic elements, named dynpay\%d, and add them to
+ * the list of dynamic elements.
+ *
+ * Find all depayloader elements, they should be named depay\%d in the
+ * element of @media, and create #GstRTSPStreams for them.
+ */
+void
+gst_rtsp_media_collect_streams (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstElement *element, *elem;
+ GstPad *pad;
+ gint i;
+ gboolean have_elem;
+ gboolean more_elem_remaining = TRUE;
+ GstRTSPTransportMode mode = 0;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+ element = priv->element;
+
+ have_elem = FALSE;
+ for (i = 0; more_elem_remaining; i++) {
+ gchar *name;
+
+ more_elem_remaining = FALSE;
+
+ name = g_strdup_printf ("pay%d", i);
+ if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+ GstElement *pay;
+ GST_INFO ("found stream %d with payloader %p", i, elem);
+
+ /* take the pad of the payloader */
+ pad = gst_element_get_static_pad (elem, "src");
+
+ /* find the real payload element in case elem is a GstBin */
+ pay = find_payload_element (elem, pad);
+
+ /* create the stream */
+ if (pay == NULL) {
+ GST_WARNING ("could not find real payloader, using bin");
+ gst_rtsp_media_create_stream (media, elem, pad);
+ } else {
+ gst_rtsp_media_create_stream (media, pay, pad);
+ gst_object_unref (pay);
+ }
+
+ gst_object_unref (pad);
+ gst_object_unref (elem);
+
+ have_elem = TRUE;
+ more_elem_remaining = TRUE;
+ mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
+ }
+ g_free (name);
+
+ name = g_strdup_printf ("dynpay%d", i);
+ if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+ /* a stream that will dynamically create pads to provide RTP packets */
+ GST_INFO ("found dynamic element %d, %p", i, elem);
+
+ g_mutex_lock (&priv->lock);
+ priv->dynamic = g_list_prepend (priv->dynamic, elem);
+ g_mutex_unlock (&priv->lock);
+
+ priv->nb_dynamic_elements++;
+
+ have_elem = TRUE;
+ more_elem_remaining = TRUE;
+ mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
+ }
+ g_free (name);
+
+ name = g_strdup_printf ("depay%d", i);
+ if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+ GST_INFO ("found stream %d with depayloader %p", i, elem);
+
+ /* take the pad of the payloader */
+ pad = gst_element_get_static_pad (elem, "sink");
+ /* create the stream */
+ gst_rtsp_media_create_stream (media, elem, pad);
+ gst_object_unref (pad);
+ gst_object_unref (elem);
+
+ have_elem = TRUE;
+ more_elem_remaining = TRUE;
+ mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
+ }
+ g_free (name);
+ }
+
+ if (have_elem) {
+ if (priv->transport_mode != mode)
+ GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
+ priv->transport_mode, mode);
+ }
+}
+
+typedef struct
+{
+ GstElement *appsink, *appsrc;
+ GstRTSPStream *stream;
+} AppSinkSrcData;
+
+static GstFlowReturn
+appsink_new_sample (GstAppSink * appsink, gpointer user_data)
+{
+ AppSinkSrcData *data = user_data;
+ GstSample *sample;
+ GstFlowReturn ret;
+
+ sample = gst_app_sink_pull_sample (appsink);
+ if (!sample)
+ return GST_FLOW_FLUSHING;
+
+
+ ret = gst_app_src_push_sample (GST_APP_SRC (data->appsrc), sample);
+ gst_sample_unref (sample);
+ return ret;
+}
+
+static GstAppSinkCallbacks appsink_callbacks = {
+ NULL,
+ NULL,
+ appsink_new_sample,
+};
+
+static GstPadProbeReturn
+appsink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+ AppSinkSrcData *data = user_data;
+
+ if (GST_IS_EVENT (info->data)
+ && GST_EVENT_TYPE (info->data) == GST_EVENT_LATENCY) {
+ GstClockTime min, max;
+
+ if (gst_base_sink_query_latency (GST_BASE_SINK (data->appsink), NULL, NULL,
+ &min, &max)) {
+ g_object_set (data->appsrc, "min-latency", min, "max-latency", max, NULL);
+ GST_DEBUG ("setting latency to min %" GST_TIME_FORMAT " max %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max));
+ }
+ } else if (GST_IS_QUERY (info->data)) {
+ GstPad *srcpad = gst_element_get_static_pad (data->appsrc, "src");
+ if (gst_pad_peer_query (srcpad, GST_QUERY_CAST (info->data))) {
+ gst_object_unref (srcpad);
+ return GST_PAD_PROBE_HANDLED;
+ }
+ gst_object_unref (srcpad);
+ }
+
+ return GST_PAD_PROBE_OK;
+}
+
+static GstPadProbeReturn
+appsrc_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+ AppSinkSrcData *data = user_data;
+
+ if (GST_IS_QUERY (info->data)) {
+ GstPad *sinkpad = gst_element_get_static_pad (data->appsink, "sink");
+ if (gst_pad_peer_query (sinkpad, GST_QUERY_CAST (info->data))) {
+ gst_object_unref (sinkpad);
+ return GST_PAD_PROBE_HANDLED;
+ }
+ gst_object_unref (sinkpad);
+ }
+
+ return GST_PAD_PROBE_OK;
+}
+
+/**
+ * gst_rtsp_media_create_stream:
+ * @media: a #GstRTSPMedia
+ * @payloader: a #GstElement
+ * @pad: a #GstPad
+ *
+ * Create a new stream in @media that provides RTP data on @pad.
+ * @pad should be a pad of an element inside @media->element.
+ *
+ * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
+ * as @media exists.
+ */
+GstRTSPStream *
+gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
+ GstPad * pad)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPStream *stream;
+ GstPad *streampad;
+ gchar *name;
+ gint idx;
+ AppSinkSrcData *data = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
+ g_return_val_if_fail (GST_IS_PAD (pad), NULL);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ idx = priv->streams->len;
+
+ GST_DEBUG ("media %p: creating stream with index %d and payloader %"
+ GST_PTR_FORMAT, media, idx, payloader);
+
+ if (GST_PAD_IS_SRC (pad))
+ name = g_strdup_printf ("src_%u", idx);
+ else
+ name = g_strdup_printf ("sink_%u", idx);
+
+ if ((GST_PAD_IS_SRC (pad) && priv->element->numsinkpads > 0) ||
+ (GST_PAD_IS_SINK (pad) && priv->element->numsrcpads > 0)) {
+ GstElement *appsink, *appsrc;
+ GstPad *sinkpad, *srcpad;
+
+ appsink = gst_element_factory_make ("appsink", NULL);
+ appsrc = gst_element_factory_make ("appsrc", NULL);
+
+ if (GST_PAD_IS_SINK (pad)) {
+ srcpad = gst_element_get_static_pad (appsrc, "src");
+
+ gst_bin_add (GST_BIN (priv->element), appsrc);
+
+ gst_pad_link (srcpad, pad);
+ gst_object_unref (srcpad);
+
+ streampad = gst_element_get_static_pad (appsink, "sink");
+
+ priv->pending_pipeline_elements =
+ g_list_prepend (priv->pending_pipeline_elements, appsink);
+ } else {
+ sinkpad = gst_element_get_static_pad (appsink, "sink");
+
+ gst_pad_link (pad, sinkpad);
+ gst_object_unref (sinkpad);
+
+ streampad = gst_element_get_static_pad (appsrc, "src");
+
+ priv->pending_pipeline_elements =
+ g_list_prepend (priv->pending_pipeline_elements, appsrc);
+ }
+
+ g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
+ TRUE, "emit-signals", FALSE, NULL);
+ g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
+ FALSE, "buffer-list", TRUE, NULL);
+
+ data = g_new0 (AppSinkSrcData, 1);
+ data->appsink = appsink;
+ data->appsrc = appsrc;
+
+ sinkpad = gst_element_get_static_pad (appsink, "sink");
+ gst_pad_add_probe (sinkpad,
+ GST_PAD_PROBE_TYPE_EVENT_UPSTREAM | GST_PAD_PROBE_TYPE_QUERY_DOWNSTREAM,
+ appsink_pad_probe, data, NULL);
+ gst_object_unref (sinkpad);
+
+ srcpad = gst_element_get_static_pad (appsrc, "src");
+ gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_QUERY_UPSTREAM,
+ appsrc_pad_probe, data, NULL);
+ gst_object_unref (srcpad);
+
+ gst_app_sink_set_callbacks (GST_APP_SINK (appsink), &appsink_callbacks,
+ data, NULL);
+ g_object_set_data_full (G_OBJECT (streampad), "media-appsink-appsrc", data,
+ g_free);
+ } else {
+ streampad = gst_ghost_pad_new (name, pad);
+ gst_pad_set_active (streampad, TRUE);
+ gst_element_add_pad (priv->element, streampad);
+ }
+ g_free (name);
+
+ stream = gst_rtsp_stream_new (idx, payloader, streampad);
+ if (data)
+ data->stream = stream;
+ if (priv->pool)
+ gst_rtsp_stream_set_address_pool (stream, priv->pool);
+ gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
+ gst_rtsp_stream_set_max_mcast_ttl (stream, priv->max_mcast_ttl);
+ gst_rtsp_stream_set_bind_mcast_address (stream, priv->bind_mcast_address);
+ gst_rtsp_stream_set_enable_rtcp (stream, priv->enable_rtcp);
+ gst_rtsp_stream_set_profiles (stream, priv->profiles);
+ gst_rtsp_stream_set_protocols (stream, priv->protocols);
+ gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
+ gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
+ gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
+ gst_rtsp_stream_set_rate_control (stream, priv->do_rate_control);
+
+ g_ptr_array_add (priv->streams, stream);
+
+ if (GST_PAD_IS_SRC (pad)) {
+ gint i, n;
+
+ if (priv->payloads)
+ g_list_free (priv->payloads);
+ priv->payloads = _find_payload_types (media);
+
+ n = priv->streams->len;
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ guint rtx_pt = _next_available_pt (priv->payloads);
+
+ if (rtx_pt == 0) {
+ GST_WARNING ("Ran out of space of dynamic payload types");
+ break;
+ }
+
+ gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
+
+ priv->payloads =
+ g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
+ NULL);
+
+ return stream;
+}
+
+static void
+gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
+{
+ GstRTSPMediaPrivate *priv;
+ GstPad *srcpad;
+ AppSinkSrcData *data;
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* remove the ghostpad */
+ srcpad = gst_rtsp_stream_get_srcpad (stream);
+ data = g_object_get_data (G_OBJECT (srcpad), "media-appsink-appsrc");
+ if (data) {
+ if (GST_OBJECT_PARENT (data->appsrc) == GST_OBJECT_CAST (priv->pipeline))
+ gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsrc);
+ else if (GST_OBJECT_PARENT (data->appsrc) ==
+ GST_OBJECT_CAST (priv->element))
+ gst_bin_remove (GST_BIN_CAST (priv->element), data->appsrc);
+ if (GST_OBJECT_PARENT (data->appsink) == GST_OBJECT_CAST (priv->pipeline))
+ gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsink);
+ else if (GST_OBJECT_PARENT (data->appsink) ==
+ GST_OBJECT_CAST (priv->element))
+ gst_bin_remove (GST_BIN_CAST (priv->element), data->appsink);
+ } else {
+ gst_element_remove_pad (priv->element, srcpad);
+ }
+ gst_object_unref (srcpad);
+ /* now remove the stream */
+ g_object_ref (stream);
+ g_ptr_array_remove (priv->streams, stream);
+ g_mutex_unlock (&priv->lock);
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
+ stream, NULL);
+
+ g_object_unref (stream);
+}
+
+/**
+ * gst_rtsp_media_n_streams:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the number of streams in this media.
+ *
+ * Returns: The number of streams.
+ */
+guint
+gst_rtsp_media_n_streams (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->streams->len;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_get_stream:
+ * @media: a #GstRTSPMedia
+ * @idx: the stream index
+ *
+ * Retrieve the stream with index @idx from @media.
+ *
+ * Returns: (nullable) (transfer none): the #GstRTSPStream at index
+ * @idx or %NULL when a stream with that index did not exist.
+ */
+GstRTSPStream *
+gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPStream *res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (idx < priv->streams->len)
+ res = g_ptr_array_index (priv->streams, idx);
+ else
+ res = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_find_stream:
+ * @media: a #GstRTSPMedia
+ * @control: the control of the stream
+ *
+ * Find a stream in @media with @control as the control uri.
+ *
+ * Returns: (nullable) (transfer none): the #GstRTSPStream with
+ * control uri @control or %NULL when a stream with that control did
+ * not exist.
+ */
+GstRTSPStream *
+gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPStream *res;
+ gint i;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ g_return_val_if_fail (control != NULL, NULL);
+
+ priv = media->priv;
+
+ res = NULL;
+
+ g_mutex_lock (&priv->lock);
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *test;
+
+ test = g_ptr_array_index (priv->streams, i);
+ if (gst_rtsp_stream_has_control (test, control)) {
+ res = test;
+ break;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/* called with state-lock */
+static gboolean
+default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
+ GstRTSPRangeUnit unit)
+{
+ return gst_rtsp_range_convert_units (range, unit);
+}
+
+/**
+ * gst_rtsp_media_get_range_string:
+ * @media: a #GstRTSPMedia
+ * @play: for the PLAY request
+ * @unit: the unit to use for the string
+ *
+ * Get the current range as a string. @media must be prepared with
+ * gst_rtsp_media_prepare ().
+ *
+ * Returns: (transfer full) (nullable): The range as a string, g_free() after usage.
+ */
+gchar *
+gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
+ GstRTSPRangeUnit unit)
+{
+ GstRTSPMediaClass *klass;
+ GstRTSPMediaPrivate *priv;
+ gchar *result;
+ GstRTSPTimeRange range;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ g_return_val_if_fail (klass->convert_range != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+ priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto not_prepared;
+
+ /* Update the range value with current position/duration */
+ g_mutex_lock (&priv->lock);
+ collect_media_stats (media);
+
+ /* make copy */
+ range = priv->range;
+
+ if (!play && priv->n_active > 0) {
+ range.min.type = GST_RTSP_TIME_NOW;
+ range.min.seconds = -1;
+ }
+ g_mutex_unlock (&priv->lock);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ if (!klass->convert_range (media, &range, unit))
+ goto conversion_failed;
+
+ result = gst_rtsp_range_to_string (&range);
+
+ return result;
+
+ /* ERRORS */
+not_prepared:
+ {
+ GST_WARNING ("media %p was not prepared", media);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return NULL;
+ }
+conversion_failed:
+ {
+ GST_WARNING ("range conversion to unit %d failed", unit);
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_media_get_rates:
+ * @media: a #GstRTSPMedia
+ * @rate: (optional) (out caller-allocates): the rate of the current segment
+ * @applied_rate: (optional) (out caller-allocates): the applied_rate of the current segment
+ *
+ * Get the rate and applied_rate of the current segment.
+ *
+ * Returns: %FALSE if looking up the rate and applied rate failed. Otherwise
+ * %TRUE is returned and @rate and @applied_rate are set to the rate and
+ * applied_rate of the current segment.
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_get_rates (GstRTSPMedia * media, gdouble * rate,
+ gdouble * applied_rate)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPStream *stream;
+ gdouble save_rate, save_applied_rate;
+ gboolean result = TRUE;
+ gboolean first_stream = TRUE;
+ gint i;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ if (!rate && !applied_rate) {
+ GST_WARNING_OBJECT (media, "rate and applied_rate are both NULL");
+ return FALSE;
+ }
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+
+ g_assert (priv->streams->len > 0);
+ for (i = 0; i < priv->streams->len; i++) {
+ stream = g_ptr_array_index (priv->streams, i);
+ if (gst_rtsp_stream_is_complete (stream)
+ && gst_rtsp_stream_is_sender (stream)) {
+ if (gst_rtsp_stream_get_rates (stream, rate, applied_rate)) {
+ if (first_stream) {
+ save_rate = *rate;
+ save_applied_rate = *applied_rate;
+ first_stream = FALSE;
+ } else {
+ if (save_rate != *rate || save_applied_rate != *applied_rate) {
+ /* diffrent rate or applied_rate, weird */
+ g_assert (FALSE);
+ result = FALSE;
+ break;
+ }
+ }
+ } else {
+ /* complete stream withot rate and applied_rate, weird */
+ g_assert (FALSE);
+ result = FALSE;
+ break;
+ }
+ }
+ }
+
+ if (!result) {
+ GST_WARNING_OBJECT (media,
+ "failed to obtain consistent rate and applied_rate");
+ }
+
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+static void
+stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
+{
+ gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
+}
+
+static void
+media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ GST_DEBUG ("media %p set blocked %d", media, blocked);
+ priv->blocked = blocked;
+ g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
+
+ if (!blocked)
+ priv->blocking_msg_received = 0;
+}
+
+static void
+gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->status = status;
+ GST_DEBUG ("setting new status to %d", status);
+ g_cond_broadcast (&priv->cond);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_status:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the status of @media. When @media is busy preparing, this function waits
+ * until @media is prepared or in error.
+ *
+ * Returns: the status of @media.
+ */
+GstRTSPMediaStatus
+gst_rtsp_media_get_status (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaStatus result;
+ gint64 end_time;
+
+ g_mutex_lock (&priv->lock);
+ end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
+ /* while we are preparing, wait */
+ while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
+ GST_DEBUG ("waiting for status change");
+ if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
+ GST_DEBUG ("timeout, assuming error status");
+ priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
+ }
+ }
+ /* could be success or error */
+ result = priv->status;
+ GST_DEBUG ("got status %d", result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_seek_trickmode:
+ * @media: a #GstRTSPMedia
+ * @range: (transfer none): a #GstRTSPTimeRange
+ * @flags: The minimal set of #GstSeekFlags to use
+ * @rate: the rate to use in the seek
+ * @trickmode_interval: The trickmode interval to use for KEY_UNITS trick mode
+ *
+ * Seek the pipeline of @media to @range with the given @flags and @rate,
+ * and @trickmode_interval.
+ * @media must be prepared with gst_rtsp_media_prepare().
+ * In order to perform the seek operation, the pipeline must contain all
+ * needed transport parts (transport sinks).
+ *
+ * Returns: %TRUE on success.
+ *
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_seek_trickmode (GstRTSPMedia * media,
+ GstRTSPTimeRange * range, GstSeekFlags flags, gdouble rate,
+ GstClockTime trickmode_interval)
+{
+ GstRTSPMediaClass *klass;
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+ GstClockTime start, stop;
+ GstSeekType start_type, stop_type;
+ gint64 current_position;
+ gboolean force_seek;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ /* if there's a range then klass->convert_range must be set */
+ g_return_val_if_fail (range == NULL || klass->convert_range != NULL, FALSE);
+
+ GST_DEBUG ("flags=%x rate=%f", flags, rate);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto not_prepared;
+
+ /* check if the media pipeline is complete in order to perform a
+ * seek operation on it */
+ if (!check_complete (media))
+ goto not_complete;
+
+ /* Update the seekable state of the pipeline in case it changed */
+ check_seekable (media);
+
+ if (priv->seekable == 0) {
+ GST_FIXME_OBJECT (media, "Handle going back to 0 for none live"
+ " not seekable streams.");
+
+ goto not_seekable;
+ } else if (priv->seekable < 0) {
+ goto not_seekable;
+ }
+
+ start_type = stop_type = GST_SEEK_TYPE_NONE;
+ start = stop = GST_CLOCK_TIME_NONE;
+
+ /* if caller provided a range convert it to NPT format
+ * if no range provided the seek is assumed to be the same position but with
+ * e.g. the rate changed */
+ if (range != NULL) {
+ if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
+ goto not_supported;
+ gst_rtsp_range_get_times (range, &start, &stop);
+
+ GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
+ GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
+ }
+
+ current_position = -1;
+ if (klass->query_position)
+ klass->query_position (media, ¤t_position);
+ GST_INFO ("current media position %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (current_position));
+
+ if (start != GST_CLOCK_TIME_NONE)
+ start_type = GST_SEEK_TYPE_SET;
+
+ if (stop != GST_CLOCK_TIME_NONE)
+ stop_type = GST_SEEK_TYPE_SET;
+
+ /* we force a seek if any trickmode flag is set, or if the flush flag is set or
+ * the rate is non-standard, i.e. not 1.0 */
+ force_seek = (flags & TRICKMODE_FLAGS) || (flags & GST_SEEK_FLAG_FLUSH) ||
+ rate != 1.0;
+
+ if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE || force_seek) {
+ GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
+
+ /* depends on the current playing state of the pipeline. We might need to
+ * queue this until we get EOS. */
+ flags |= GST_SEEK_FLAG_FLUSH;
+
+ /* if range start was not supplied we must continue from current position.
+ * but since we're doing a flushing seek, let us query the current position
+ * so we end up at exactly the same position after the seek. */
+ if (range == NULL || range->min.type == GST_RTSP_TIME_END) {
+ if (current_position == -1) {
+ GST_WARNING ("current position unknown");
+ } else {
+ GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (current_position));
+ start = current_position;
+ start_type = GST_SEEK_TYPE_SET;
+ }
+ }
+
+ if (!force_seek &&
+ (start_type == GST_SEEK_TYPE_NONE || start == current_position) &&
+ (stop_type == GST_SEEK_TYPE_NONE || stop == priv->range_stop)) {
+ GST_DEBUG ("no position change, no flags set by caller, so not seeking");
+ res = TRUE;
+ } else {
+ GstEvent *seek_event;
+ gboolean unblock = FALSE;
+
+ /* Handle expected async-done before waiting on next async-done.
+ *
+ * Since the seek further down in code will cause a preroll and
+ * a async-done will be generated it's important to wait on async-done
+ * if that is expected. Otherwise there is the risk that the waiting
+ * for async-done after the seek is detecting the expected async-done
+ * instead of the one that corresponds to the seek. Then execution
+ * continue and act as if the pipeline is prerolled, but it's not.
+ *
+ * During wait_preroll message GST_MESSAGE_ASYNC_DONE will come
+ * and then the state will change from preparing to prepared */
+ if (priv->expected_async_done) {
+ GST_DEBUG (" expected to get async-done, waiting ");
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ /* wait until pipeline is prerolled */
+ if (!wait_preroll (media))
+ goto preroll_failed_expected_async_done;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ GST_DEBUG (" got expected async-done");
+ }
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+
+ if (rate < 0.0) {
+ GstClockTime temp_time = start;
+ GstSeekType temp_type = start_type;
+
+ start = stop;
+ start_type = stop_type;
+ stop = temp_time;
+ stop_type = temp_type;
+ }
+
+ seek_event = gst_event_new_seek (rate, GST_FORMAT_TIME, flags, start_type,
+ start, stop_type, stop);
+
+ gst_event_set_seek_trickmode_interval (seek_event, trickmode_interval);
+
+ if (!media->priv->blocked) {
+ /* Prevent a race condition with multiple streams,
+ * where one stream may have time to preroll before others
+ * have even started flushing, causing async-done to be
+ * posted too early.
+ */
+ media_streams_set_blocked (media, TRUE);
+ unblock = TRUE;
+ }
+
+ res = gst_element_send_event (priv->pipeline, seek_event);
+
+ if (unblock)
+ media_streams_set_blocked (media, FALSE);
+
+ /* and block for the seek to complete */
+ GST_INFO ("done seeking %d", res);
+ if (!res)
+ goto seek_failed;
+
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ /* wait until pipeline is prerolled again, this will also collect stats */
+ if (!wait_preroll (media))
+ goto preroll_failed;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ GST_INFO ("prerolled again");
+ }
+ } else {
+ GST_INFO ("no seek needed");
+ res = TRUE;
+ }
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+
+ /* ERRORS */
+not_prepared:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("media %p is not prepared", media);
+ return FALSE;
+ }
+not_complete:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("pipeline is not complete");
+ return FALSE;
+ }
+not_seekable:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("pipeline is not seekable");
+ return FALSE;
+ }
+not_supported:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("conversion to npt not supported");
+ return FALSE;
+ }
+seek_failed:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("seeking failed");
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll after seek");
+ return FALSE;
+ }
+preroll_failed_expected_async_done:
+ {
+ GST_WARNING ("failed to preroll");
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_seek_full:
+ * @media: a #GstRTSPMedia
+ * @range: (transfer none): a #GstRTSPTimeRange
+ * @flags: The minimal set of #GstSeekFlags to use
+ *
+ * Seek the pipeline of @media to @range with the given @flags.
+ * @media must be prepared with gst_rtsp_media_prepare().
+ *
+ * Returns: %TRUE on success.
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
+ GstSeekFlags flags)
+{
+ return gst_rtsp_media_seek_trickmode (media, range, flags, 1.0, 0);
+}
+
+/**
+ * gst_rtsp_media_seek:
+ * @media: a #GstRTSPMedia
+ * @range: (transfer none): a #GstRTSPTimeRange
+ *
+ * Seek the pipeline of @media to @range. @media must be prepared with
+ * gst_rtsp_media_prepare().
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
+{
+ return gst_rtsp_media_seek_trickmode (media, range, GST_SEEK_FLAG_NONE,
+ 1.0, 0);
+}
+
+static void
+stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
+{
+ *blocked &= gst_rtsp_stream_is_blocking (stream);
+}
+
+static gboolean
+media_streams_blocking (GstRTSPMedia * media)
+{
+ gboolean blocking = TRUE;
+
+ g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
+ &blocking);
+
+ return blocking;
+}
+
+static GstStateChangeReturn
+set_state (GstRTSPMedia * media, GstState state)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
+ media);
+ ret = gst_element_set_state (priv->pipeline, state);
+
+ return ret;
+}
+
+static GstStateChangeReturn
+set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ GST_INFO ("set target state to %s for media %p",
+ gst_element_state_get_name (state), media);
+ priv->target_state = state;
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
+ priv->target_state, NULL);
+
+ if (do_state)
+ ret = set_state (media, state);
+ else
+ ret = GST_STATE_CHANGE_SUCCESS;
+
+ return ret;
+}
+
+static void
+stream_collect_active_sender (GstRTSPStream * stream, guint * active_streams)
+{
+ if (gst_rtsp_stream_is_complete (stream)
+ && gst_rtsp_stream_is_sender (stream))
+ (*active_streams)++;
+}
+
+static guint
+nbr_active_sender_streams (GstRTSPMedia * media)
+{
+ guint ret = 0;
+
+ g_ptr_array_foreach (media->priv->streams,
+ (GFunc) stream_collect_active_sender, &ret);
+
+ return ret;
+}
+
+ /* called with state-lock */
+/* called with state-lock */
+static gboolean
+default_handle_message (GstRTSPMedia * media, GstMessage * message)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstMessageType type;
+
+ type = GST_MESSAGE_TYPE (message);
+
+ switch (type) {
+ case GST_MESSAGE_STATE_CHANGED:
+ {
+ GstState old, new, pending;
+
+ if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
+ break;
+
+ gst_message_parse_state_changed (message, &old, &new, &pending);
+
+ GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
+ gst_element_state_get_name (old), gst_element_state_get_name (new),
+ gst_element_state_get_name (pending));
+ if (priv->no_more_pads_pending == 0
+ && gst_rtsp_media_is_receive_only (media) && old == GST_STATE_READY
+ && new == GST_STATE_PAUSED) {
+ GST_INFO ("%p: went to PAUSED, prepared now", media);
+ g_mutex_lock (&priv->lock);
+ collect_media_stats (media);
+ g_mutex_unlock (&priv->lock);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ }
+
+ break;
+ }
+ case GST_MESSAGE_BUFFERING:
+ {
+ gint percent;
+
+ gst_message_parse_buffering (message, &percent);
+
+ /* no state management needed for live pipelines */
+ if (priv->is_live)
+ break;
+
+ if (percent == 100) {
+ /* a 100% message means buffering is done */
+ priv->buffering = FALSE;
+ /* if the desired state is playing, go back */
+ if (priv->target_state == GST_STATE_PLAYING) {
+ GST_INFO ("Buffering done, setting pipeline to PLAYING");
+ set_state (media, GST_STATE_PLAYING);
+ } else {
+ GST_INFO ("Buffering done");
+ }
+ } else {
+ /* buffering busy */
+ if (priv->buffering == FALSE) {
+ if (priv->target_state == GST_STATE_PLAYING) {
+ /* we were not buffering but PLAYING, PAUSE the pipeline. */
+ GST_INFO ("Buffering, setting pipeline to PAUSED ...");
+ set_state (media, GST_STATE_PAUSED);
+ } else {
+ GST_INFO ("Buffering ...");
+ }
+ }
+ priv->buffering = TRUE;
+ }
+ break;
+ }
+ case GST_MESSAGE_LATENCY:
+ {
+ gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
+ break;
+ }
+ case GST_MESSAGE_ERROR:
+ {
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_error (message, &gerror, &debug);
+ GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
+ g_error_free (gerror);
+ g_free (debug);
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ break;
+ }
+ case GST_MESSAGE_WARNING:
+ {
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_warning (message, &gerror, &debug);
+ GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
+ g_error_free (gerror);
+ g_free (debug);
+ break;
+ }
+ case GST_MESSAGE_ELEMENT:
+ {
+ const GstStructure *s;
+
+ s = gst_message_get_structure (message);
+ if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
+ gboolean is_complete = FALSE;
+ guint n_active_sender_streams;
+ guint expected_nbr_blocking_msg;
+
+ /* to prevent problems when some streams are complete, some are not,
+ * we will ignore incomplete streams. When there are no complete
+ * streams (during DESCRIBE), we will listen to all streams. */
+
+ gst_structure_get_boolean (s, "is_complete", &is_complete);
+ n_active_sender_streams = nbr_active_sender_streams (media);
+ expected_nbr_blocking_msg = n_active_sender_streams;
+ GST_DEBUG_OBJECT (media, "media received blocking message,"
+ " n_active_sender_streams = %d, is_complete = %d",
+ n_active_sender_streams, is_complete);
+
+ if (n_active_sender_streams == 0 || is_complete)
+ priv->blocking_msg_received++;
+
+ if (n_active_sender_streams == 0)
+ expected_nbr_blocking_msg = priv->streams->len;
+
+ if (priv->blocked && media_streams_blocking (media) &&
+ priv->no_more_pads_pending == 0 &&
+ priv->blocking_msg_received == expected_nbr_blocking_msg) {
+ GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "media is blocking");
+ g_mutex_lock (&priv->lock);
+ collect_media_stats (media);
+ g_mutex_unlock (&priv->lock);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+
+ priv->blocking_msg_received = 0;
+ }
+ }
+ break;
+ }
+ case GST_MESSAGE_STREAM_STATUS:
+ break;
+ case GST_MESSAGE_ASYNC_DONE:
+ if (priv->expected_async_done)
+ priv->expected_async_done = FALSE;
+ if (priv->complete) {
+ /* receive the final ASYNC_DONE, that is posted by the media pipeline
+ * after all the transport parts have been successfully added to
+ * the media streams. */
+ GST_DEBUG_OBJECT (media, "got async-done");
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ }
+ break;
+ case GST_MESSAGE_EOS:
+ GST_INFO ("%p: got EOS", media);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
+ GST_DEBUG ("shutting down after EOS");
+ finish_unprepare (media);
+ }
+ break;
+ default:
+ GST_INFO ("%p: got message type %d (%s)", media, type,
+ gst_message_type_get_name (type));
+ break;
+ }
+ return TRUE;
+}
+
+static gboolean
+bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaClass *klass;
+ gboolean ret;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (klass->handle_message)
+ ret = klass->handle_message (media, message);
+ else
+ ret = FALSE;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return ret;
+}
+
+static void
+watch_destroyed (GstRTSPMedia * media)
+{
+ GST_DEBUG_OBJECT (media, "source destroyed");
+ g_object_unref (media);
+}
+
+static gboolean
+is_payloader (GstElement * element)
+{
+ GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
+ const gchar *klass;
+
+ klass = gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
+ if (klass == NULL)
+ return FALSE;
+
+ if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
+ return TRUE;
+ }
+
+ return FALSE;
+}
+
+static GstElement *
+find_payload_element (GstElement * payloader, GstPad * pad)
+{
+ GstElement *pay = NULL;
+
+ if (GST_IS_BIN (payloader)) {
+ GstIterator *iter;
+ GValue item = { 0 };
+ gchar *pad_name, *payloader_name;
+ GstElement *element;
+
+ if ((element = gst_bin_get_by_name (GST_BIN (payloader), "pay"))) {
+ if (is_payloader (element))
+ return element;
+ gst_object_unref (element);
+ }
+
+ pad_name = gst_object_get_name (GST_OBJECT (pad));
+ payloader_name = g_strdup_printf ("pay_%s", pad_name);
+ g_free (pad_name);
+ if ((element = gst_bin_get_by_name (GST_BIN (payloader), payloader_name))) {
+ g_free (payloader_name);
+ if (is_payloader (element))
+ return element;
+ gst_object_unref (element);
+ } else {
+ g_free (payloader_name);
+ }
+
+ iter = gst_bin_iterate_recurse (GST_BIN (payloader));
+ while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
+ element = (GstElement *) g_value_get_object (&item);
+
+ if (is_payloader (element)) {
+ pay = gst_object_ref (element);
+ g_value_unset (&item);
+ break;
+ }
+ g_value_unset (&item);
+ }
+ gst_iterator_free (iter);
+ } else {
+ pay = g_object_ref (payloader);
+ }
+
+ return pay;
+}
+
+/* called from streaming threads */
+static void
+pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPStream *stream;
+ GstElement *pay;
+
+ /* find the real payload element */
+ pay = find_payload_element (element, pad);
+ stream = gst_rtsp_media_create_stream (media, pay, pad);
+ gst_object_unref (pay);
+
+ GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
+ goto not_preparing;
+
+ g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
+
+ /* join the element in the PAUSED state because this callback is
+ * called from the streaming thread and it is PAUSED */
+ if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+ priv->rtpbin, GST_STATE_PAUSED)) {
+ GST_WARNING ("failed to join bin element");
+ }
+
+ if (priv->blocked)
+ gst_rtsp_stream_set_blocked (stream, TRUE);
+
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return;
+
+ /* ERRORS */
+not_preparing:
+ {
+ gst_rtsp_media_remove_stream (media, stream);
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("ignore pad because we are not preparing");
+ return;
+ }
+}
+
+static void
+pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPStream *stream;
+
+ stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
+ if (stream == NULL)
+ return;
+
+ GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
+
+ g_rec_mutex_lock (&priv->state_lock);
+ gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ gst_rtsp_media_remove_stream (media, stream);
+}
+
+static void
+no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ GST_INFO_OBJECT (element, "no more pads");
+ g_mutex_lock (&priv->lock);
+ priv->no_more_pads_pending--;
+ g_mutex_unlock (&priv->lock);
+}
+
+typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
+
+struct _DynPaySignalHandlers
+{
+ gulong pad_added_handler;
+ gulong pad_removed_handler;
+ gulong no_more_pads_handler;
+};
+
+static gboolean
+start_preroll (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ GST_INFO ("setting pipeline to PAUSED for media %p", media);
+
+ /* start blocked since it is possible that there are no sink elements yet */
+ media_streams_set_blocked (media, TRUE);
+ ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+
+ switch (ret) {
+ case GST_STATE_CHANGE_SUCCESS:
+ GST_INFO ("SUCCESS state change for media %p", media);
+ break;
+ case GST_STATE_CHANGE_ASYNC:
+ GST_INFO ("ASYNC state change for media %p", media);
+ break;
+ case GST_STATE_CHANGE_NO_PREROLL:
+ /* we need to go to PLAYING */
+ GST_INFO ("NO_PREROLL state change: live media %p", media);
+ /* FIXME we disable seeking for live streams for now. We should perform a
+ * seeking query in preroll instead */
+ priv->seekable = -1;
+ priv->is_live = TRUE;
+
+ ret = set_state (media, GST_STATE_PLAYING);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto state_failed;
+ break;
+ case GST_STATE_CHANGE_FAILURE:
+ goto state_failed;
+ }
+
+ return TRUE;
+
+state_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ return FALSE;
+ }
+}
+
+static gboolean
+wait_preroll (GstRTSPMedia * media)
+{
+ GstRTSPMediaStatus status;
+
+ GST_DEBUG ("wait to preroll pipeline");
+
+ /* wait until pipeline is prerolled */
+ status = gst_rtsp_media_get_status (media);
+ if (status == GST_RTSP_MEDIA_STATUS_ERROR)
+ goto preroll_failed;
+
+ return TRUE;
+
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ return FALSE;
+ }
+}
+
+static GstElement *
+request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPStream *stream = NULL;
+ guint i;
+ GstElement *res = NULL;
+
+ g_mutex_lock (&priv->lock);
+ for (i = 0; i < priv->streams->len; i++) {
+ stream = g_ptr_array_index (priv->streams, i);
+
+ if (sessid == gst_rtsp_stream_get_index (stream))
+ break;
+
+ stream = NULL;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (stream)
+ res = gst_rtsp_stream_request_aux_sender (stream, sessid);
+
+ return res;
+}
+
+static GstElement *
+request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPStream *stream = NULL;
+ guint i;
+ GstElement *res = NULL;
+
+ g_mutex_lock (&priv->lock);
+ for (i = 0; i < priv->streams->len; i++) {
+ stream = g_ptr_array_index (priv->streams, i);
+
+ if (sessid == gst_rtsp_stream_get_index (stream))
+ break;
+
+ stream = NULL;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (stream)
+ res = gst_rtsp_stream_request_aux_receiver (stream, sessid);
+
+ return res;
+}
+
+static GstElement *
+request_fec_decoder (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPStream *stream = NULL;
+ guint i;
+ GstElement *res = NULL;
+
+ g_mutex_lock (&priv->lock);
+ for (i = 0; i < priv->streams->len; i++) {
+ stream = g_ptr_array_index (priv->streams, i);
+
+ if (sessid == gst_rtsp_stream_get_index (stream))
+ break;
+
+ stream = NULL;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (stream) {
+ res = gst_rtsp_stream_request_ulpfec_decoder (stream, rtpbin, sessid);
+ }
+
+ return res;
+}
+
+static gboolean
+start_prepare (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ guint i;
+ GList *walk;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
+ goto no_longer_preparing;
+
+ g_signal_connect (priv->rtpbin, "request-fec-decoder",
+ G_CALLBACK (request_fec_decoder), media);
+
+ /* link streams we already have, other streams might appear when we have
+ * dynamic elements */
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream;
+
+ stream = g_ptr_array_index (priv->streams, i);
+
+ if (priv->rtx_time > 0) {
+ /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
+ g_signal_connect (priv->rtpbin, "request-aux-sender",
+ (GCallback) request_aux_sender, media);
+ }
+
+ if (priv->do_retransmission) {
+ g_signal_connect (priv->rtpbin, "request-aux-receiver",
+ (GCallback) request_aux_receiver, media);
+ }
+
+ if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+ priv->rtpbin, GST_STATE_NULL)) {
+ goto join_bin_failed;
+ }
+ }
+
+ if (priv->rtpbin)
+ g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission,
+ "do-lost", TRUE, NULL);
+
+ for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
+ GstElement *elem = walk->data;
+ DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
+
+ GST_INFO ("adding callbacks for dynamic element %p", elem);
+
+ handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
+ (GCallback) pad_added_cb, media);
+ handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
+ (GCallback) pad_removed_cb, media);
+ handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
+ (GCallback) no_more_pads_cb, media);
+
+ g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
+ }
+
+ if (priv->nb_dynamic_elements == 0 && gst_rtsp_media_is_receive_only (media)) {
+ /* If we are receive_only (RECORD), do not try to preroll, to avoid
+ * a second ASYNC state change failing */
+ priv->is_live = TRUE;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ } else if (!start_preroll (media)) {
+ goto preroll_failed;
+ }
+
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return FALSE;
+
+no_longer_preparing:
+ {
+ GST_INFO ("media is no longer preparing");
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+join_bin_failed:
+ {
+ GST_WARNING ("failed to join bin element");
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+}
+
+static gboolean
+default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+ GstBus *bus;
+ GMainContext *context;
+ GSource *source;
+
+ priv = media->priv;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ if (!klass->create_rtpbin)
+ goto no_create_rtpbin;
+
+ priv->rtpbin = klass->create_rtpbin (media);
+ if (priv->rtpbin != NULL) {
+ gboolean success = TRUE;
+
+ g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
+
+ if (klass->setup_rtpbin)
+ success = klass->setup_rtpbin (media, priv->rtpbin);
+
+ if (success == FALSE) {
+ gst_object_unref (priv->rtpbin);
+ priv->rtpbin = NULL;
+ }
+ }
+ if (priv->rtpbin == NULL)
+ goto no_rtpbin;
+
+ priv->thread = thread;
+ context = (thread != NULL) ? (thread->context) : NULL;
+
+ bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
+
+ /* add the pipeline bus to our custom mainloop */
+ priv->source = gst_bus_create_watch (bus);
+ gst_object_unref (bus);
+
+ g_source_set_callback (priv->source, (GSourceFunc) bus_message,
+ g_object_ref (media), (GDestroyNotify) watch_destroyed);
+
+ g_source_attach (priv->source, context);
+
+ /* add stuff to the bin */
+ gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
+
+ /* do remainder in context */
+ source = g_idle_source_new ();
+ g_source_set_callback (source, (GSourceFunc) start_prepare,
+ g_object_ref (media), (GDestroyNotify) g_object_unref);
+ g_source_attach (source, context);
+ g_source_unref (source);
+
+ return TRUE;
+
+ /* ERRORS */
+no_create_rtpbin:
+ {
+ GST_ERROR ("no create_rtpbin function");
+ g_critical ("no create_rtpbin vmethod function set");
+ return FALSE;
+ }
+no_rtpbin:
+ {
+ GST_WARNING ("no rtpbin element");
+ g_warning ("failed to create element 'rtpbin', check your installation");
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_prepare:
+ * @media: a #GstRTSPMedia
+ * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
+ * bus handler or %NULL
+ *
+ * Prepare @media for streaming. This function will create the objects
+ * to manage the streaming. A pipeline must have been set on @media with
+ * gst_rtsp_media_take_pipeline().
+ *
+ * It will preroll the pipeline and collect vital information about the streams
+ * such as the duration.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ priv->prepare_count++;
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
+ priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto was_prepared;
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ goto is_preparing;
+
+ if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
+ goto not_unprepared;
+
+ if (!priv->reusable && priv->reused)
+ goto is_reused;
+
+ GST_INFO ("preparing media %p", media);
+
+ /* reset some variables */
+ priv->is_live = FALSE;
+ priv->seekable = -1;
+ priv->buffering = FALSE;
+ priv->no_more_pads_pending = priv->nb_dynamic_elements;
+
+ /* we're preparing now */
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->prepare) {
+ if (!klass->prepare (media, thread))
+ goto prepare_failed;
+ }
+
+wait_status:
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ /* now wait for all pads to be prerolled, FIXME, we should somehow be
+ * able to do this async so that we don't block the server thread. */
+ if (!wait_preroll (media))
+ goto preroll_failed;
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
+
+ GST_INFO ("object %p is prerolled", media);
+
+ return TRUE;
+
+ /* OK */
+is_preparing:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ goto wait_status;
+ }
+was_prepared:
+ {
+ GST_LOG ("media %p was prepared", media);
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return TRUE;
+ }
+ /* ERRORS */
+not_unprepared:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ GST_WARNING ("media %p was not unprepared", media);
+ priv->prepare_count--;
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+is_reused:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ priv->prepare_count--;
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("can not reuse media %p", media);
+ return FALSE;
+ }
+prepare_failed:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ priv->prepare_count--;
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_ERROR ("failed to prepare media");
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ gst_rtsp_media_unprepare (media);
+ return FALSE;
+ }
+}
+
+/* must be called with state-lock */
+static void
+finish_unprepare (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gint i;
+ GList *walk;
+
+ if (priv->finishing_unprepare)
+ return;
+ priv->finishing_unprepare = TRUE;
+
+ GST_DEBUG ("shutting down");
+
+ /* release the lock on shutdown, otherwise pad_added_cb might try to
+ * acquire the lock and then we deadlock */
+ g_rec_mutex_unlock (&priv->state_lock);
+ set_state (media, GST_STATE_NULL);
+ g_rec_mutex_lock (&priv->state_lock);
+
+ media_streams_set_blocked (media, FALSE);
+
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream;
+
+ GST_INFO ("Removing elements of stream %d from pipeline", i);
+
+ stream = g_ptr_array_index (priv->streams, i);
+
+ gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
+ }
+
+ /* remove the pad signal handlers */
+ for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
+ GstElement *elem = walk->data;
+ DynPaySignalHandlers *handlers;
+
+ handlers =
+ g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
+ g_assert (handlers != NULL);
+
+ g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
+ g_signal_handler_disconnect (G_OBJECT (elem),
+ handlers->pad_removed_handler);
+ g_signal_handler_disconnect (G_OBJECT (elem),
+ handlers->no_more_pads_handler);
+
+ g_slice_free (DynPaySignalHandlers, handlers);
+ }
+
+ gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
+ priv->rtpbin = NULL;
+
+ if (priv->nettime)
+ gst_object_unref (priv->nettime);
+ priv->nettime = NULL;
+
+ priv->reused = TRUE;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
+
+ /* when the media is not reusable, this will effectively unref the media and
+ * recreate it */
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
+
+ /* the source has the last ref to the media */
+ if (priv->source) {
+ GstBus *bus;
+
+ GST_DEBUG ("removing bus watch");
+ bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
+ gst_bus_remove_watch (bus);
+ gst_object_unref (bus);
+
+ GST_DEBUG ("destroy source");
+ g_source_destroy (priv->source);
+ g_source_unref (priv->source);
+ priv->source = NULL;
+ }
+ if (priv->thread) {
+ GST_DEBUG ("stop thread");
+ gst_rtsp_thread_stop (priv->thread);
+ }
+
+ priv->finishing_unprepare = FALSE;
+}
+
+/* called with state-lock */
+static gboolean
+default_unprepare (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
+
+ if (priv->eos_shutdown) {
+ GST_DEBUG ("sending EOS for shutdown");
+ /* ref so that we don't disappear */
+ gst_element_send_event (priv->pipeline, gst_event_new_eos ());
+ /* we need to go to playing again for the EOS to propagate, normally in this
+ * state, nothing is receiving data from us anymore so this is ok. */
+ set_state (media, GST_STATE_PLAYING);
+ } else {
+ finish_unprepare (media);
+ }
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_media_unprepare:
+ * @media: a #GstRTSPMedia
+ *
+ * Unprepare @media. After this call, the media should be prepared again before
+ * it can be used again. If the media is set to be non-reusable, a new instance
+ * must be created.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_unprepare (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean success;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
+ goto was_unprepared;
+
+ priv->prepare_count--;
+ if (priv->prepare_count > 0)
+ goto is_busy;
+
+ GST_INFO ("unprepare media %p", media);
+ set_target_state (media, GST_STATE_NULL, FALSE);
+ success = TRUE;
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
+ GstRTSPMediaClass *klass;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->unprepare)
+ success = klass->unprepare (media);
+ } else {
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
+ finish_unprepare (media);
+ }
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return success;
+
+was_unprepared:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("media %p was already unprepared", media);
+ return TRUE;
+ }
+is_busy:
+ {
+ GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return TRUE;
+ }
+}
+
+/* should be called with state-lock */
+static GstClock *
+get_clock_unlocked (GstRTSPMedia * media)
+{
+ if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
+ GST_DEBUG_OBJECT (media, "media was not prepared");
+ return NULL;
+ }
+ return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
+}
+
+/**
+ * gst_rtsp_media_lock:
+ * @media: a #GstRTSPMedia
+ *
+ * Lock the entire media. This is needed by callers such as rtsp_client to
+ * protect the media when it is shared by many clients.
+ * The lock prevents that concurrent clients alters the shared media,
+ * while one client already is working with it.
+ * Typically the lock is taken in external RTSP API calls that uses shared media
+ * such as DESCRIBE, SETUP, ANNOUNCE, TEARDOWN, PLAY, PAUSE.
+ *
+ * As best practice take the lock as soon as the function get hold of a shared
+ * media object. Release the lock right before the function returns.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_media_lock (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->global_lock);
+}
+
+/**
+ * gst_rtsp_media_unlock:
+ * @media: a #GstRTSPMedia
+ *
+ * Unlock the media.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_media_unlock (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_unlock (&priv->global_lock);
+}
+
+/**
+ * gst_rtsp_media_get_clock:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the clock that is used by the pipeline in @media.
+ *
+ * @media must be prepared before this method returns a valid clock object.
+ *
+ * Returns: (transfer full) (nullable): the #GstClock used by @media. unref after usage.
+ */
+GstClock *
+gst_rtsp_media_get_clock (GstRTSPMedia * media)
+{
+ GstClock *clock;
+ GstRTSPMediaPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ clock = get_clock_unlocked (media);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return clock;
+}
+
+/**
+ * gst_rtsp_media_get_base_time:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the base_time that is used by the pipeline in @media.
+ *
+ * @media must be prepared before this method returns a valid base_time.
+ *
+ * Returns: the base_time used by @media.
+ */
+GstClockTime
+gst_rtsp_media_get_base_time (GstRTSPMedia * media)
+{
+ GstClockTime result;
+ GstRTSPMediaPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto not_prepared;
+
+ result = gst_element_get_base_time (media->priv->pipeline);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return result;
+
+ /* ERRORS */
+not_prepared:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_DEBUG_OBJECT (media, "media was not prepared");
+ return GST_CLOCK_TIME_NONE;
+ }
+}
+
+/**
+ * gst_rtsp_media_get_time_provider:
+ * @media: a #GstRTSPMedia
+ * @address: (allow-none): an address or %NULL
+ * @port: a port or 0
+ *
+ * Get the #GstNetTimeProvider for the clock used by @media. The time provider
+ * will listen on @address and @port for client time requests.
+ *
+ * Returns: (transfer full): the #GstNetTimeProvider of @media.
+ */
+GstNetTimeProvider *
+gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
+ guint16 port)
+{
+ GstRTSPMediaPrivate *priv;
+ GstNetTimeProvider *provider = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->time_provider) {
+ if ((provider = priv->nettime) == NULL) {
+ GstClock *clock;
+
+ if (priv->time_provider && (clock = get_clock_unlocked (media))) {
+ provider = gst_net_time_provider_new (clock, address, port);
+ gst_object_unref (clock);
+
+ priv->nettime = provider;
+ }
+ }
+ }
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ if (provider)
+ gst_object_ref (provider);
+
+ return provider;
+}
+
+static gboolean
+default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
+{
+ return gst_rtsp_sdp_from_media (sdp, info, media);
+}
+
+/**
+ * gst_rtsp_media_setup_sdp:
+ * @media: a #GstRTSPMedia
+ * @sdp: (transfer none): a #GstSDPMessage
+ * @info: (transfer none): a #GstSDPInfo
+ *
+ * Add @media specific info to @sdp. @info is used to configure the connection
+ * information in the SDP.
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (sdp != NULL, FALSE);
+ g_return_val_if_fail (info != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ if (!klass->setup_sdp)
+ goto no_setup_sdp;
+
+ res = klass->setup_sdp (media, sdp, info);
+
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+
+ /* ERRORS */
+no_setup_sdp:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_ERROR ("no setup_sdp function");
+ g_critical ("no setup_sdp vmethod function set");
+ return FALSE;
+ }
+}
+
+static gboolean
+default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gint i, medias_len;
+
+ medias_len = gst_sdp_message_medias_len (sdp);
+ if (medias_len != priv->streams->len) {
+ GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
+ priv->streams->len, medias_len);
+ return FALSE;
+ }
+
+ for (i = 0; i < medias_len; i++) {
+ const gchar *proto;
+ const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
+ GstRTSPStream *stream;
+ gint j, formats_len;
+ const gchar *control;
+ GstRTSPProfile profile, profiles;
+
+ stream = g_ptr_array_index (priv->streams, i);
+
+ /* TODO: Should we do something with the other SDP information? */
+
+ /* get proto */
+ proto = gst_sdp_media_get_proto (sdp_media);
+ if (proto == NULL) {
+ GST_ERROR ("%p: SDP media %d has no proto", media, i);
+ return FALSE;
+ }
+
+ if (g_str_equal (proto, "RTP/AVP")) {
+ profile = GST_RTSP_PROFILE_AVP;
+ } else if (g_str_equal (proto, "RTP/SAVP")) {
+ profile = GST_RTSP_PROFILE_SAVP;
+ } else if (g_str_equal (proto, "RTP/AVPF")) {
+ profile = GST_RTSP_PROFILE_AVPF;
+ } else if (g_str_equal (proto, "RTP/SAVPF")) {
+ profile = GST_RTSP_PROFILE_SAVPF;
+ } else {
+ GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
+ return FALSE;
+ }
+
+ profiles = gst_rtsp_stream_get_profiles (stream);
+ if ((profiles & profile) == 0) {
+ GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
+ return FALSE;
+ }
+
+ formats_len = gst_sdp_media_formats_len (sdp_media);
+ for (j = 0; j < formats_len; j++) {
+ gint pt;
+ GstCaps *caps;
+ GstStructure *s;
+
+ pt = atoi (gst_sdp_media_get_format (sdp_media, j));
+
+ GST_DEBUG (" looking at %d pt: %d", j, pt);
+
+ /* convert caps */
+ caps = gst_sdp_media_get_caps_from_media (sdp_media, pt);
+ if (caps == NULL) {
+ GST_WARNING (" skipping pt %d without caps", pt);
+ continue;
+ }
+
+ /* do some tweaks */
+ GST_DEBUG ("mapping sdp session level attributes to caps");
+ gst_sdp_message_attributes_to_caps (sdp, caps);
+ GST_DEBUG ("mapping sdp media level attributes to caps");
+ gst_sdp_media_attributes_to_caps (sdp_media, caps);
+
+ s = gst_caps_get_structure (caps, 0);
+ gst_structure_set_name (s, "application/x-rtp");
+
+ if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
+ gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ gst_rtsp_stream_set_pt_map (stream, pt, caps);
+ gst_caps_unref (caps);
+ }
+
+ control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ if (control)
+ gst_rtsp_stream_set_control (stream, control);
+
+ }
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_media_handle_sdp:
+ * @media: a #GstRTSPMedia
+ * @sdp: (transfer none): a #GstSDPMessage
+ *
+ * Configure an SDP on @media for receiving streams
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (sdp != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ if (!klass->handle_sdp)
+ goto no_handle_sdp;
+
+ res = klass->handle_sdp (media, sdp);
+
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+
+ /* ERRORS */
+no_handle_sdp:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_ERROR ("no handle_sdp function");
+ g_critical ("no handle_sdp vmethod function set");
+ return FALSE;
+ }
+}
+
+static void
+do_set_seqnum (GstRTSPStream * stream)
+{
+ guint16 seq_num;
+
+ if (gst_rtsp_stream_is_sender (stream)) {
+ seq_num = gst_rtsp_stream_get_current_seqnum (stream);
+ gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
+ }
+}
+
+/* call with state_lock */
+static gboolean
+default_suspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret = GST_STATE_CHANGE_FAILURE;
+
+ switch (priv->suspend_mode) {
+ case GST_RTSP_SUSPEND_MODE_NONE:
+ GST_DEBUG ("media %p no suspend", media);
+ break;
+ case GST_RTSP_SUSPEND_MODE_PAUSE:
+ GST_DEBUG ("media %p suspend to PAUSED", media);
+ ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto state_failed;
+ break;
+ case GST_RTSP_SUSPEND_MODE_RESET:
+ GST_DEBUG ("media %p suspend to NULL", media);
+ ret = set_target_state (media, GST_STATE_NULL, TRUE);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto state_failed;
+ /* Because payloader needs to set the sequence number as
+ * monotonic, we need to preserve the sequence number
+ * after pause. (otherwise going from pause to play, which
+ * is actually from NULL to PLAY will create a new sequence
+ * number. */
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
+ break;
+ default:
+ break;
+ }
+
+ /* If we use any suspend mode that changes the state then we must update
+ * expected_async_done, since we might not be doing an asyncronous state
+ * change anymore. */
+ if (ret != GST_STATE_CHANGE_FAILURE && ret != GST_STATE_CHANGE_ASYNC)
+ priv->expected_async_done = FALSE;
+
+ return TRUE;
+
+ /* ERRORS */
+state_failed:
+ {
+ GST_WARNING ("failed changing pipeline's state for media %p", media);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_suspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Suspend @media. The state of the pipeline managed by @media is set to
+ * GST_STATE_NULL but all streams are kept. @media can be prepared again
+ * with gst_rtsp_media_unsuspend()
+ *
+ * @media must be prepared with gst_rtsp_media_prepare();
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_suspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ GST_FIXME ("suspend for dynamic pipelines needs fixing");
+
+ /* this typically can happen for shared media. */
+ if (priv->prepare_count > 1 &&
+ priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED) {
+ goto done;
+ } else if (priv->prepare_count > 1) {
+ goto prepared_by_other_client;
+ }
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto not_prepared;
+
+ /* don't attempt to suspend when something is busy */
+ if (priv->n_active > 0)
+ goto done;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->suspend) {
+ if (!klass->suspend (media))
+ goto suspend_failed;
+ }
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
+done:
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+prepared_by_other_client:
+ {
+ GST_WARNING ("media %p was prepared by other client", media);
+ return FALSE;
+ }
+not_prepared:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("media %p was not prepared", media);
+ return FALSE;
+ }
+suspend_failed:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ GST_WARNING ("failed to suspend media %p", media);
+ return FALSE;
+ }
+}
+
+/* call with state_lock */
+static gboolean
+default_unsuspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gboolean preroll_ok;
+
+ switch (priv->suspend_mode) {
+ case GST_RTSP_SUSPEND_MODE_NONE:
+ if (gst_rtsp_media_is_receive_only (media))
+ break;
+ if (media_streams_blocking (media)) {
+ g_rec_mutex_unlock (&priv->state_lock);
+ if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
+ g_rec_mutex_lock (&priv->state_lock);
+ goto preroll_failed;
+ }
+ g_rec_mutex_lock (&priv->state_lock);
+ }
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ break;
+ case GST_RTSP_SUSPEND_MODE_PAUSE:
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ break;
+ case GST_RTSP_SUSPEND_MODE_RESET:
+ {
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+ /* at this point the media pipeline has been updated and contain all
+ * specific transport parts: all active streams contain at least one sink
+ * element and it's safe to unblock all blocked streams */
+ media_streams_set_blocked (media, FALSE);
+ if (!start_preroll (media))
+ goto start_failed;
+
+ g_rec_mutex_unlock (&priv->state_lock);
+ preroll_ok = wait_preroll (media);
+ g_rec_mutex_lock (&priv->state_lock);
+
+ if (!preroll_ok)
+ goto preroll_failed;
+ }
+ default:
+ break;
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+start_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ return FALSE;
+ }
+}
+
+static void
+gst_rtsp_media_unblock_rtcp (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ priv = media->priv;
+ g_mutex_lock (&priv->lock);
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ gst_rtsp_stream_unblock_rtcp (stream);
+ }
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_unsuspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Unsuspend @media if it was in a suspended state. This method does nothing
+ * when the media was not in the suspended state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_unsuspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto done;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->unsuspend) {
+ if (!klass->unsuspend (media))
+ goto unsuspend_failed;
+ }
+
+done:
+ gst_rtsp_media_unblock_rtcp (media);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+unsuspend_failed:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("failed to unsuspend media %p", media);
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ return FALSE;
+ }
+}
+
+/* must be called with state-lock */
+static void
+media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn set_state_ret;
+ priv->expected_async_done = FALSE;
+
+ if (state == GST_STATE_NULL) {
+ gst_rtsp_media_unprepare (media);
+ } else {
+ GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
+ set_target_state (media, state, FALSE);
+
+ if (state == GST_STATE_PLAYING) {
+ /* make sure pads are not blocking anymore when going to PLAYING */
+ media_streams_set_blocked (media, FALSE);
+ }
+
+ /* when we are buffering, don't update the state yet, this will be done
+ * when buffering finishes */
+ if (priv->buffering) {
+ GST_INFO ("Buffering busy, delay state change");
+ } else {
+ if (state == GST_STATE_PAUSED) {
+ set_state_ret = set_state (media, state);
+ if (set_state_ret == GST_STATE_CHANGE_ASYNC)
+ priv->expected_async_done = TRUE;
+ /* and suspend after pause */
+ gst_rtsp_media_suspend (media);
+ } else {
+ set_state (media, state);
+ }
+ }
+ }
+}
+
+/**
+ * gst_rtsp_media_set_pipeline_state:
+ * @media: a #GstRTSPMedia
+ * @state: the target state of the pipeline
+ *
+ * Set the state of the pipeline managed by @media to @state
+ */
+void
+gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
+{
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ g_rec_mutex_lock (&media->priv->state_lock);
+ media_set_pipeline_state_locked (media, state);
+ g_rec_mutex_unlock (&media->priv->state_lock);
+}
+
+/**
+ * gst_rtsp_media_set_state:
+ * @media: a #GstRTSPMedia
+ * @state: the target state of the media
+ * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
+ * a #GPtrArray of #GstRTSPStreamTransport pointers
+ *
+ * Set the state of @media to @state and for the transports in @transports.
+ *
+ * @media must be prepared with gst_rtsp_media_prepare();
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
+ GPtrArray * transports)
+{
+ GstRTSPMediaPrivate *priv;
+ gint i;
+ gboolean activate, deactivate, do_state;
+ gint old_active;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (transports != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING
+ && gst_rtsp_media_is_shared (media)) {
+ g_rec_mutex_unlock (&priv->state_lock);
+ gst_rtsp_media_get_status (media);
+ g_rec_mutex_lock (&priv->state_lock);
+ }
+ if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
+ goto error_status;
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+ priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto not_prepared;
+
+ /* NULL and READY are the same */
+ if (state == GST_STATE_READY)
+ state = GST_STATE_NULL;
+
+ activate = deactivate = FALSE;
+
+ GST_INFO ("going to state %s media %p, target state %s",
+ gst_element_state_get_name (state), media,
+ gst_element_state_get_name (priv->target_state));
+
+ switch (state) {
+ case GST_STATE_NULL:
+ /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
+ if (priv->target_state >= GST_STATE_PAUSED)
+ deactivate = TRUE;
+ break;
+ case GST_STATE_PAUSED:
+ /* we're going from PLAYING to PAUSED, deactivate */
+ if (priv->target_state == GST_STATE_PLAYING)
+ deactivate = TRUE;
+ break;
+ case GST_STATE_PLAYING:
+ /* we're going to PLAYING, activate */
+ activate = TRUE;
+ break;
+ default:
+ break;
+ }
+ old_active = priv->n_active;
+
+ GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
+ activate, deactivate);
+ for (i = 0; i < transports->len; i++) {
+ GstRTSPStreamTransport *trans;
+
+ /* we need a non-NULL entry in the array */
+ trans = g_ptr_array_index (transports, i);
+ if (trans == NULL)
+ continue;
+
+ if (activate) {
+ if (gst_rtsp_stream_transport_set_active (trans, TRUE))
+ priv->n_active++;
+ } else if (deactivate) {
+ if (gst_rtsp_stream_transport_set_active (trans, FALSE))
+ priv->n_active--;
+ }
+ }
+
+ if (activate)
+ media_streams_set_blocked (media, FALSE);
+
+ /* we just activated the first media, do the playing state change */
+ if (old_active == 0 && activate)
+ do_state = TRUE;
+ /* if we have no more active media and prepare count is not indicate
+ * that there are new session/sessions ongoing,
+ * do the downward state changes */
+ else if (priv->n_active == 0 && priv->prepare_count <= 1)
+ do_state = TRUE;
+ else
+ do_state = FALSE;
+
+ GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
+ media, do_state);
+
+ if (priv->target_state != state) {
+ if (do_state) {
+ media_set_pipeline_state_locked (media, state);
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
+ NULL);
+ }
+ }
+
+ /* remember where we are */
+ if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
+ old_active != priv->n_active)) {
+ g_mutex_lock (&priv->lock);
+ collect_media_stats (media);
+ g_mutex_unlock (&priv->lock);
+ }
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+not_prepared:
+ {
+ GST_WARNING ("media %p was not prepared", media);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+error_status:
+ {
+ GST_WARNING ("media %p in error status while changing to state %d",
+ media, state);
+ if (state == GST_STATE_NULL) {
+ for (i = 0; i < transports->len; i++) {
+ GstRTSPStreamTransport *trans;
+
+ /* we need a non-NULL entry in the array */
+ trans = g_ptr_array_index (transports, i);
+ if (trans == NULL)
+ continue;
+
+ gst_rtsp_stream_transport_set_active (trans, FALSE);
+ }
+ priv->n_active = 0;
+ }
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_set_transport_mode:
+ * @media: a #GstRTSPMedia
+ * @mode: the new value
+ *
+ * Sets if the media pipeline can work in PLAY or RECORD mode
+ */
+void
+gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
+ GstRTSPTransportMode mode)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->transport_mode = mode;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_transport_mode:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media can be used for PLAY or RECORD methods.
+ *
+ * Returns: The transport mode.
+ */
+GstRTSPTransportMode
+gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPTransportMode res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->transport_mode;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_seekable:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media seek and up to what point in time,
+ * it can seek.
+ *
+ * Returns: -1 if the stream is not seekable, 0 if seekable only to the beginning
+ * and > 0 to indicate the longest duration between any two random access points.
+ * %G_MAXINT64 means any value is possible.
+ *
+ * Since: 1.14
+ */
+GstClockTimeDiff
+gst_rtsp_media_seekable (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstClockTimeDiff res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ /* Currently we are not able to seek on live streams,
+ * and no stream is seekable only to the beginning */
+ g_mutex_lock (&priv->lock);
+ res = priv->seekable;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_complete_pipeline:
+ * @media: a #GstRTSPMedia
+ * @transports: (element-type GstRTSPTransport): a list of #GstRTSPTransport
+ *
+ * Add a receiver and sender parts to the pipeline based on the transport from
+ * SETUP.
+ *
+ * Returns: %TRUE if the media pipeline has been sucessfully updated.
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (transports, FALSE);
+
+ GST_DEBUG_OBJECT (media, "complete pipeline");
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStreamTransport *transport;
+ GstRTSPStream *stream;
+ const GstRTSPTransport *rtsp_transport;
+
+ transport = g_ptr_array_index (transports, i);
+ if (!transport)
+ continue;
+
+ stream = gst_rtsp_stream_transport_get_stream (transport);
+ if (!stream)
+ continue;
+
+ rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
+
+ if (!gst_rtsp_stream_complete_stream (stream, rtsp_transport)) {
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+
+ if (!gst_rtsp_stream_add_transport (stream, transport)) {
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+
+ update_stream_storage_size (media, stream, i);
+ }
+
+ priv->complete = TRUE;
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_media_is_receive_only:
+ *
+ * Returns: %TRUE if @media is receive-only, %FALSE otherwise.
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_is_receive_only (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gboolean receive_only;
+
+ g_mutex_lock (&priv->lock);
+ receive_only = is_receive_only (media);
+ g_mutex_unlock (&priv->lock);
+
+ return receive_only;
+}
+
+/**
+ * gst_rtsp_media_has_completed_sender:
+ *
+ * See gst_rtsp_stream_is_complete(), gst_rtsp_stream_is_sender().
+ *
+ * Returns: whether @media has at least one complete sender stream.
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_has_completed_sender (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gboolean sender = FALSE;
+ guint i;
+
+ g_mutex_lock (&priv->lock);
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ if (gst_rtsp_stream_is_complete (stream))
+ if (gst_rtsp_stream_is_sender (stream) ||
+ !gst_rtsp_stream_is_receiver (stream)) {
+ sender = TRUE;
+ break;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return sender;
+}
+
+/**
+ * gst_rtsp_media_set_rate_control:
+ *
+ * Define whether @media will follow the Rate-Control=no behaviour as specified
+ * in the ONVIF replay spec.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ GST_LOG_OBJECT (media, "%s rate control", enabled ? "Enabling" : "Disabling");
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->do_rate_control = enabled;
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+
+ gst_rtsp_stream_set_rate_control (stream, enabled);
+
+ }
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_rate_control:
+ *
+ * Returns: whether @media will follow the Rate-Control=no behaviour as specified
+ * in the ONVIF replay spec.
+ *
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_get_rate_control (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->do_rate_control;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/rtsp.h>
+#include <gst/net/gstnet.h>
+
+#ifndef __GST_RTSP_MEDIA_H__
+#define __GST_RTSP_MEDIA_H__
+
+#include "rtsp-server-prelude.h"
+
+G_BEGIN_DECLS
+
+/* types for the media */
+#define GST_TYPE_RTSP_MEDIA (gst_rtsp_media_get_type ())
+#define GST_IS_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA))
+#define GST_IS_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA))
+#define GST_RTSP_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
+#define GST_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia))
+#define GST_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
+#define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj))
+#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
+
+typedef struct _GstRTSPMedia GstRTSPMedia;
+typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
+typedef struct _GstRTSPMediaPrivate GstRTSPMediaPrivate;
+
+/**
+ * GstRTSPMediaStatus:
+ * @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
+ * @GST_RTSP_MEDIA_STATUS_UNPREPARING: media pipeline is busy doing a clean
+ * shutdown.
+ * @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
+ * @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
+ * @GST_RTSP_MEDIA_STATUS_SUSPENDED: media is suspended
+ * @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
+ *
+ * The state of the media pipeline.
+ */
+typedef enum {
+ GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
+ GST_RTSP_MEDIA_STATUS_UNPREPARING = 1,
+ GST_RTSP_MEDIA_STATUS_PREPARING = 2,
+ GST_RTSP_MEDIA_STATUS_PREPARED = 3,
+ GST_RTSP_MEDIA_STATUS_SUSPENDED = 4,
+ GST_RTSP_MEDIA_STATUS_ERROR = 5
+} GstRTSPMediaStatus;
+
+/**
+ * GstRTSPSuspendMode:
+ * @GST_RTSP_SUSPEND_MODE_NONE: Media is not suspended
+ * @GST_RTSP_SUSPEND_MODE_PAUSE: Media is PAUSED in suspend
+ * @GST_RTSP_SUSPEND_MODE_RESET: The media is set to NULL when suspended
+ *
+ * The suspend mode of the media pipeline. A media pipeline is suspended right
+ * after creating the SDP and when the client performs a PAUSED request.
+ */
+typedef enum {
+ GST_RTSP_SUSPEND_MODE_NONE = 0,
+ GST_RTSP_SUSPEND_MODE_PAUSE = 1,
+ GST_RTSP_SUSPEND_MODE_RESET = 2
+} GstRTSPSuspendMode;
+
+/**
+ * GstRTSPTransportMode:
+ * @GST_RTSP_TRANSPORT_MODE_PLAY: Transport supports PLAY mode
+ * @GST_RTSP_TRANSPORT_MODE_RECORD: Transport supports RECORD mode
+ *
+ * The supported modes of the media.
+ */
+typedef enum {
+ GST_RTSP_TRANSPORT_MODE_PLAY = 1,
+ GST_RTSP_TRANSPORT_MODE_RECORD = 2,
+} GstRTSPTransportMode;
+
+/**
+ * GstRTSPPublishClockMode:
+ * @GST_RTSP_PUBLISH_CLOCK_MODE_NONE: Publish nothing
+ * @GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK: Publish the clock but not the offset
+ * @GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET: Publish the clock and offset
+ *
+ * Whether the clock and possibly RTP/clock offset should be published according to RFC7273.
+ */
+typedef enum {
+ GST_RTSP_PUBLISH_CLOCK_MODE_NONE,
+ GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK,
+ GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET
+} GstRTSPPublishClockMode;
+
+#define GST_TYPE_RTSP_TRANSPORT_MODE (gst_rtsp_transport_mode_get_type())
+GST_RTSP_SERVER_API
+GType gst_rtsp_transport_mode_get_type (void);
+
+#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
+GST_RTSP_SERVER_API
+GType gst_rtsp_suspend_mode_get_type (void);
+
+#define GST_TYPE_RTSP_PUBLISH_CLOCK_MODE (gst_rtsp_publish_clock_mode_get_type())
+GST_RTSP_SERVER_API
+GType gst_rtsp_publish_clock_mode_get_type (void);
+
+#include "rtsp-stream.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-permissions.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-sdp.h"
+
+/**
+ * GstRTSPMedia:
+ *
+ * A class that contains the GStreamer element along with a list of
+ * #GstRTSPStream objects that can produce data.
+ *
+ * This object is usually created from a #GstRTSPMediaFactory.
+ */
+struct _GstRTSPMedia {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPMediaPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaClass:
+ * @handle_message: handle a message
+ * @prepare: the default implementation adds all elements and sets the
+ * pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING
+ * in case of NO_PREROLL elements).
+ * @unprepare: the default implementation sets the pipeline's state
+ * to GST_STATE_NULL and removes all elements.
+ * @suspend: the default implementation sets the pipeline's state to
+ * GST_STATE_NULL GST_STATE_PAUSED depending on the selected
+ * suspend mode.
+ * @unsuspend: the default implementation reverts the suspend operation.
+ * The pipeline will be prerolled again if it's state was
+ * set to GST_STATE_NULL in suspend.
+ * @convert_range: convert a range to the given unit
+ * @query_position: query the current position in the pipeline
+ * @query_stop: query when playback will stop
+ *
+ * The RTSP media class
+ */
+struct _GstRTSPMediaClass {
+ GObjectClass parent_class;
+
+ /* vmethods */
+ gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
+ gboolean (*prepare) (GstRTSPMedia *media, GstRTSPThread *thread);
+ gboolean (*unprepare) (GstRTSPMedia *media);
+ gboolean (*suspend) (GstRTSPMedia *media);
+ gboolean (*unsuspend) (GstRTSPMedia *media);
+ gboolean (*convert_range) (GstRTSPMedia *media, GstRTSPTimeRange *range,
+ GstRTSPRangeUnit unit);
+ gboolean (*query_position) (GstRTSPMedia *media, gint64 *position);
+ gboolean (*query_stop) (GstRTSPMedia *media, gint64 *stop);
+ GstElement * (*create_rtpbin) (GstRTSPMedia *media);
+ gboolean (*setup_rtpbin) (GstRTSPMedia *media, GstElement *rtpbin);
+ gboolean (*setup_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);
+
+ /* signals */
+ void (*new_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
+ void (*removed_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
+
+ void (*prepared) (GstRTSPMedia *media);
+ void (*unprepared) (GstRTSPMedia *media);
+
+ void (*target_state) (GstRTSPMedia *media, GstState state);
+ void (*new_state) (GstRTSPMedia *media, GstState state);
+
+ gboolean (*handle_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE-1];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_media_get_type (void);
+
+/* creating the media */
+
+GST_RTSP_SERVER_API
+GstRTSPMedia * gst_rtsp_media_new (GstElement *element);
+
+GST_RTSP_SERVER_API
+GstElement * gst_rtsp_media_get_element (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_take_pipeline (GstRTSPMedia *media, GstPipeline *pipeline);
+
+GST_RTSP_SERVER_API
+GstRTSPMediaStatus gst_rtsp_media_get_status (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_permissions (GstRTSPMedia *media,
+ GstRTSPPermissions *permissions);
+
+GST_RTSP_SERVER_API
+GstRTSPPermissions * gst_rtsp_media_get_permissions (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia *media, gboolean stop_on_disconnect);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_transport_mode (GstRTSPMedia *media, GstRTSPTransportMode mode);
+
+GST_RTSP_SERVER_API
+GstRTSPTransportMode gst_rtsp_media_get_transport_mode (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_profiles (GstRTSPMedia *media, GstRTSPProfile profiles);
+
+GST_RTSP_SERVER_API
+GstRTSPProfile gst_rtsp_media_get_profiles (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
+
+GST_RTSP_SERVER_API
+GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool);
+
+GST_RTSP_SERVER_API
+GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_multicast_iface (GstRTSPMedia *media, const gchar *multicast_iface);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_media_get_multicast_iface (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_retransmission_time (GstRTSPMedia *media, GstClockTime time);
+
+GST_RTSP_SERVER_API
+GstClockTime gst_rtsp_media_get_retransmission_time (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
+ gboolean do_retransmission);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_latency (GstRTSPMedia *media, guint latency);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_media_get_latency (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_use_time_provider (GstRTSPMedia *media, gboolean time_provider);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_is_time_provider (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+GstNetTimeProvider * gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
+ const gchar *address, guint16 port);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_clock (GstRTSPMedia *media, GstClock * clock);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media, GstRTSPPublishClockMode mode);
+
+GST_RTSP_SERVER_API
+GstRTSPPublishClockMode gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia *media, guint ttl);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia *media, gboolean bind_mcast_addr);
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_dscp_qos (GstRTSPMedia * media, gint dscp_qos);
+GST_RTSP_SERVER_API
+gint gst_rtsp_media_get_dscp_qos (GstRTSPMedia * media);
+
+/* prepare the media for playback */
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_prepare (GstRTSPMedia *media, GstRTSPThread *thread);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media, GstRTSPSuspendMode mode);
+
+GST_RTSP_SERVER_API
+GstRTSPSuspendMode gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_suspend (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_unsuspend (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
+
+/* creating streams */
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media,
+ GstElement *payloader,
+ GstPad *pad);
+
+/* dealing with the media */
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_lock (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_unlock (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+GstClock * gst_rtsp_media_get_clock (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+GstClockTime gst_rtsp_media_get_base_time (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+GstRTSPStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
+
+GST_RTSP_SERVER_API
+GstRTSPStream * gst_rtsp_media_find_stream (GstRTSPMedia *media, const gchar * control);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_seek_full (GstRTSPMedia *media,
+ GstRTSPTimeRange *range,
+ GstSeekFlags flags);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_seek_trickmode (GstRTSPMedia *media,
+ GstRTSPTimeRange *range,
+ GstSeekFlags flags,
+ gdouble rate,
+ GstClockTime trickmode_interval);
+
+GST_RTSP_SERVER_API
+GstClockTimeDiff gst_rtsp_media_seekable (GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media,
+ gboolean play,
+ GstRTSPRangeUnit unit);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_get_rates (GstRTSPMedia * media,
+ gdouble * rate,
+ gdouble * applied_rate);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state,
+ GPtrArray *transports);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media,
+ GstState state);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_is_receive_only (GstRTSPMedia * media);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_has_completed_sender (GstRTSPMedia * media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_get_rate_control (GstRTSPMedia * media);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPMedia, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MEDIA_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-mount-points
+ * @short_description: Map a path to media
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPClient
+ *
+ * A #GstRTSPMountPoints object maintains a relation between paths
+ * and #GstRTSPMediaFactory objects. This object is usually given to
+ * #GstRTSPClient and used to find the media attached to a path.
+ *
+ * With gst_rtsp_mount_points_add_factory () and
+ * gst_rtsp_mount_points_remove_factory(), factories can be added and
+ * removed.
+ *
+ * With gst_rtsp_mount_points_match() you can find the #GstRTSPMediaFactory
+ * object that completely matches the given path.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-mount-points.h"
+
+typedef struct
+{
+ gchar *path;
+ gint len;
+ GstRTSPMediaFactory *factory;
+} DataItem;
+
+static DataItem *
+data_item_new (gchar * path, gint len, GstRTSPMediaFactory * factory)
+{
+ DataItem *item;
+
+ item = g_slice_alloc (sizeof (DataItem));
+ item->path = path;
+ item->len = len;
+ item->factory = factory;
+
+ return item;
+}
+
+static void
+data_item_free (gpointer data)
+{
+ DataItem *item = data;
+
+ g_free (item->path);
+ g_object_unref (item->factory);
+ g_slice_free1 (sizeof (DataItem), item);
+}
+
+static void
+data_item_dump (gconstpointer a, gconstpointer prefix)
+{
+ const DataItem *item = a;
+
+ GST_DEBUG ("%s%s %p", (gchar *) prefix, item->path, item->factory);
+}
+
+static gint
+data_item_compare (gconstpointer a, gconstpointer b, gpointer user_data)
+{
+ const DataItem *item1 = a, *item2 = b;
+ gint res;
+
+ res = g_strcmp0 (item1->path, item2->path);
+
+ return res;
+}
+
+struct _GstRTSPMountPointsPrivate
+{
+ GMutex lock;
+ GSequence *mounts; /* protected by lock */
+ gboolean dirty;
+};
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPMountPoints, gst_rtsp_mount_points,
+ G_TYPE_OBJECT);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
+#define GST_CAT_DEFAULT rtsp_media_debug
+
+static gchar *default_make_path (GstRTSPMountPoints * mounts,
+ const GstRTSPUrl * url);
+static void gst_rtsp_mount_points_finalize (GObject * obj);
+
+static void
+gst_rtsp_mount_points_class_init (GstRTSPMountPointsClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->finalize = gst_rtsp_mount_points_finalize;
+
+ klass->make_path = default_make_path;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmountpoints", 0,
+ "GstRTSPMountPoints");
+}
+
+static void
+gst_rtsp_mount_points_init (GstRTSPMountPoints * mounts)
+{
+ GstRTSPMountPointsPrivate *priv;
+
+ GST_DEBUG_OBJECT (mounts, "created");
+
+ mounts->priv = priv = gst_rtsp_mount_points_get_instance_private (mounts);
+
+ g_mutex_init (&priv->lock);
+ priv->mounts = g_sequence_new (data_item_free);
+ priv->dirty = FALSE;
+}
+
+static void
+gst_rtsp_mount_points_finalize (GObject * obj)
+{
+ GstRTSPMountPoints *mounts = GST_RTSP_MOUNT_POINTS (obj);
+ GstRTSPMountPointsPrivate *priv = mounts->priv;
+
+ GST_DEBUG_OBJECT (mounts, "finalized");
+
+ g_sequence_free (priv->mounts);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_mount_points_parent_class)->finalize (obj);
+}
+
+/**
+ * gst_rtsp_mount_points_new:
+ *
+ * Make a new mount points object.
+ *
+ * Returns: (transfer full): a new #GstRTSPMountPoints
+ */
+GstRTSPMountPoints *
+gst_rtsp_mount_points_new (void)
+{
+ GstRTSPMountPoints *result;
+
+ result = g_object_new (GST_TYPE_RTSP_MOUNT_POINTS, NULL);
+
+ return result;
+}
+
+static gchar *
+default_make_path (GstRTSPMountPoints * mounts, const GstRTSPUrl * url)
+{
+ /* normalize rtsp://<IP>:<PORT> to rtsp://<IP>:<PORT>/ */
+ return g_strdup (url->abspath[0] ? url->abspath : "/");
+}
+
+/**
+ * gst_rtsp_mount_points_make_path:
+ * @mounts: a #GstRTSPMountPoints
+ * @url: a #GstRTSPUrl
+ *
+ * Make a path string from @url.
+ *
+ * Returns: (transfer full) (nullable): a path string for @url, g_free() after usage.
+ */
+gchar *
+gst_rtsp_mount_points_make_path (GstRTSPMountPoints * mounts,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMountPointsClass *klass;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts), NULL);
+ g_return_val_if_fail (url != NULL, NULL);
+
+ klass = GST_RTSP_MOUNT_POINTS_GET_CLASS (mounts);
+
+ if (klass->make_path)
+ result = klass->make_path (mounts, url);
+ else
+ result = NULL;
+
+ return result;
+}
+
+static gboolean
+has_prefix (DataItem * str, DataItem * prefix)
+{
+ /* prefix needs to be smaller than str */
+ if (str->len < prefix->len)
+ return FALSE;
+
+ /* special case when "/" is the entire prefix */
+ if (prefix->len == 1 && prefix->path[0] == '/' && str->path[0] == '/')
+ return TRUE;
+
+ /* if str is larger, it there should be a / following the prefix */
+ if (str->len > prefix->len && str->path[prefix->len] != '/')
+ return FALSE;
+
+ return strncmp (str->path, prefix->path, prefix->len) == 0;
+}
+
+/**
+ * gst_rtsp_mount_points_match:
+ * @mounts: a #GstRTSPMountPoints
+ * @path: a mount point
+ * @matched: (out) (allow-none): the amount of @path matched
+ *
+ * Find the factory in @mounts that has the longest match with @path.
+ *
+ * If @matched is %NULL, @path will match the factory exactly otherwise
+ * the amount of characters that matched is returned in @matched.
+ *
+ * Returns: (transfer full): the #GstRTSPMediaFactory for @path.
+ * g_object_unref() after usage.
+ */
+GstRTSPMediaFactory *
+gst_rtsp_mount_points_match (GstRTSPMountPoints * mounts,
+ const gchar * path, gint * matched)
+{
+ GstRTSPMountPointsPrivate *priv;
+ GstRTSPMediaFactory *result = NULL;
+ GSequenceIter *iter, *best;
+ DataItem item, *ritem;
+
+ g_return_val_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts), NULL);
+ g_return_val_if_fail (path != NULL, NULL);
+
+ priv = mounts->priv;
+
+ item.path = (gchar *) path;
+ item.len = strlen (path);
+
+ GST_LOG ("Looking for mount point path %s", path);
+
+ g_mutex_lock (&priv->lock);
+ if (priv->dirty) {
+ g_sequence_sort (priv->mounts, data_item_compare, mounts);
+ g_sequence_foreach (priv->mounts, (GFunc) data_item_dump,
+ (gpointer) "sort :");
+ priv->dirty = FALSE;
+ }
+
+ /* find the location of the media in the hashtable we only use the absolute
+ * path of the uri to find a media factory. If the factory depends on other
+ * properties found in the url, this method should be overridden. */
+ iter = g_sequence_get_begin_iter (priv->mounts);
+ best = NULL;
+ while (!g_sequence_iter_is_end (iter)) {
+ ritem = g_sequence_get (iter);
+
+ data_item_dump (ritem, "inspect: ");
+
+ /* The sequence is sorted, so any prefix match is an improvement upon
+ * the previous best match, as '/abc' will always be before '/abcd' */
+ if (has_prefix (&item, ritem)) {
+ if (best == NULL) {
+ data_item_dump (ritem, "prefix: ");
+ } else {
+ data_item_dump (ritem, "new best: ");
+ }
+ best = iter;
+ } else {
+ /* if have a match and the current item doesn't prefix match the best we
+ * found so far then we're moving away and can bail out of the loop */
+ if (best != NULL && !has_prefix (ritem, g_sequence_get (best)))
+ break;
+ }
+
+ iter = g_sequence_iter_next (iter);
+ }
+ if (best) {
+ ritem = g_sequence_get (best);
+ data_item_dump (ritem, "result: ");
+ if (matched || ritem->len == item.len) {
+ result = g_object_ref (ritem->factory);
+ if (matched)
+ *matched = ritem->len;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ GST_INFO ("found media factory %p for path %s", result, path);
+
+ return result;
+}
+
+static void
+gst_rtsp_mount_points_remove_factory_unlocked (GstRTSPMountPoints * mounts,
+ const gchar * path)
+{
+ GstRTSPMountPointsPrivate *priv = mounts->priv;
+ DataItem item;
+ GSequenceIter *iter;
+
+ item.path = (gchar *) path;
+
+ if (priv->dirty) {
+ g_sequence_sort (priv->mounts, data_item_compare, mounts);
+ priv->dirty = FALSE;
+ }
+ iter = g_sequence_lookup (priv->mounts, &item, data_item_compare, mounts);
+ if (iter) {
+ g_sequence_remove (iter);
+ priv->dirty = TRUE;
+ }
+}
+
+/**
+ * gst_rtsp_mount_points_add_factory:
+ * @mounts: a #GstRTSPMountPoints
+ * @path: a mount point
+ * @factory: (transfer full): a #GstRTSPMediaFactory
+ *
+ * Attach @factory to the mount point @path in @mounts.
+ *
+ * @path is either of the form (/node)+ or the root path '/'. (An empty path is
+ * not allowed.) Any previous mount point will be freed.
+ *
+ * Ownership is taken of the reference on @factory so that @factory should not be
+ * used after calling this function.
+ */
+void
+gst_rtsp_mount_points_add_factory (GstRTSPMountPoints * mounts,
+ const gchar * path, GstRTSPMediaFactory * factory)
+{
+ GstRTSPMountPointsPrivate *priv;
+ DataItem *item;
+
+ g_return_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts));
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+ g_return_if_fail (path != NULL && path[0] == '/');
+
+ priv = mounts->priv;
+
+ item = data_item_new (g_strdup (path), strlen (path), factory);
+
+ GST_INFO ("adding media factory %p for path %s", factory, path);
+
+ g_mutex_lock (&priv->lock);
+ gst_rtsp_mount_points_remove_factory_unlocked (mounts, path);
+ g_sequence_append (priv->mounts, item);
+ priv->dirty = TRUE;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_mount_points_remove_factory:
+ * @mounts: a #GstRTSPMountPoints
+ * @path: a mount point
+ *
+ * Remove the #GstRTSPMediaFactory associated with @path in @mounts.
+ */
+void
+gst_rtsp_mount_points_remove_factory (GstRTSPMountPoints * mounts,
+ const gchar * path)
+{
+ GstRTSPMountPointsPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts));
+ g_return_if_fail (path != NULL && path[0] == '/');
+
+ priv = mounts->priv;
+
+ GST_INFO ("removing media factory for path %s", path);
+
+ g_mutex_lock (&priv->lock);
+ gst_rtsp_mount_points_remove_factory_unlocked (mounts, path);
+ g_mutex_unlock (&priv->lock);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include "rtsp-media-factory.h"
+
+#ifndef __GST_RTSP_MOUNT_POINTS_H__
+#define __GST_RTSP_MOUNT_POINTS_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_MOUNT_POINTS (gst_rtsp_mount_points_get_type ())
+#define GST_IS_RTSP_MOUNT_POINTS(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MOUNT_POINTS))
+#define GST_IS_RTSP_MOUNT_POINTS_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MOUNT_POINTS))
+#define GST_RTSP_MOUNT_POINTS_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPointsClass))
+#define GST_RTSP_MOUNT_POINTS(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPoints))
+#define GST_RTSP_MOUNT_POINTS_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPointsClass))
+#define GST_RTSP_MOUNT_POINTS_CAST(obj) ((GstRTSPMountPoints*)(obj))
+#define GST_RTSP_MOUNT_POINTS_CLASS_CAST(klass) ((GstRTSPMountPointsClass*)(klass))
+
+typedef struct _GstRTSPMountPoints GstRTSPMountPoints;
+typedef struct _GstRTSPMountPointsClass GstRTSPMountPointsClass;
+typedef struct _GstRTSPMountPointsPrivate GstRTSPMountPointsPrivate;
+
+/**
+ * GstRTSPMountPoints:
+ *
+ * Creates a #GstRTSPMediaFactory object for a given url.
+ */
+struct _GstRTSPMountPoints {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPMountPointsPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMountPointsClass:
+ * @make_path: make a path from the given url.
+ *
+ * The class for the media mounts object.
+ */
+struct _GstRTSPMountPointsClass {
+ GObjectClass parent_class;
+
+ gchar * (*make_path) (GstRTSPMountPoints *mounts,
+ const GstRTSPUrl *url);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_mount_points_get_type (void);
+
+/* creating a mount points */
+
+GST_RTSP_SERVER_API
+GstRTSPMountPoints * gst_rtsp_mount_points_new (void);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_mount_points_make_path (GstRTSPMountPoints *mounts,
+ const GstRTSPUrl * url);
+/* finding a media factory */
+
+GST_RTSP_SERVER_API
+GstRTSPMediaFactory * gst_rtsp_mount_points_match (GstRTSPMountPoints *mounts,
+ const gchar *path,
+ gint * matched);
+/* managing media to a mount point */
+
+GST_RTSP_SERVER_API
+void gst_rtsp_mount_points_add_factory (GstRTSPMountPoints *mounts,
+ const gchar *path,
+ GstRTSPMediaFactory *factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_mount_points_remove_factory (GstRTSPMountPoints *mounts,
+ const gchar *path);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPMountPoints, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MOUNT_POINTS_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-onvif-client.h"
+#include "rtsp-onvif-server.h"
+#include "rtsp-onvif-media-factory.h"
+
+G_DEFINE_TYPE (GstRTSPOnvifClient, gst_rtsp_onvif_client, GST_TYPE_RTSP_CLIENT);
+
+static gchar *
+gst_rtsp_onvif_client_check_requirements (GstRTSPClient * client,
+ GstRTSPContext * ctx, gchar ** requirements)
+{
+ GstRTSPMountPoints *mount_points = NULL;
+ GstRTSPMediaFactory *factory = NULL;
+ gchar *path = NULL;
+ gboolean has_backchannel = FALSE;
+ gboolean has_replay = FALSE;
+ GString *unsupported = g_string_new ("");
+
+ while (*requirements) {
+ if (strcmp (*requirements, GST_RTSP_ONVIF_BACKCHANNEL_REQUIREMENT) == 0) {
+ has_backchannel = TRUE;
+ } else if (strcmp (*requirements, GST_RTSP_ONVIF_REPLAY_REQUIREMENT) == 0) {
+ has_replay = TRUE;
+ } else {
+ if (unsupported->len)
+ g_string_append (unsupported, ", ");
+ g_string_append (unsupported, *requirements);
+ }
+ requirements++;
+ }
+
+ if (unsupported->len)
+ goto out;
+
+ mount_points = gst_rtsp_client_get_mount_points (client);
+ if (!(path = gst_rtsp_mount_points_make_path (mount_points, ctx->uri)))
+ goto out;
+
+ if (!(factory = gst_rtsp_mount_points_match (mount_points, path, NULL)))
+ goto out;
+
+ if (has_backchannel && !GST_IS_RTSP_ONVIF_MEDIA_FACTORY (factory)) {
+ if (unsupported->len)
+ g_string_append (unsupported, ", ");
+ g_string_append (unsupported, GST_RTSP_ONVIF_BACKCHANNEL_REQUIREMENT);
+ } else if (has_backchannel) {
+ GstRTSPOnvifMediaFactory *onvif_factory =
+ GST_RTSP_ONVIF_MEDIA_FACTORY (factory);
+
+ if (!gst_rtsp_onvif_media_factory_has_backchannel_support (onvif_factory)) {
+ if (unsupported->len)
+ g_string_append (unsupported, ", ");
+ g_string_append (unsupported, GST_RTSP_ONVIF_BACKCHANNEL_REQUIREMENT);
+ }
+ }
+
+ if (has_replay && !GST_IS_RTSP_ONVIF_MEDIA_FACTORY (factory)) {
+ if (unsupported->len)
+ g_string_append (unsupported, ", ");
+ g_string_append (unsupported, GST_RTSP_ONVIF_REPLAY_REQUIREMENT);
+ } else if (has_replay) {
+ GstRTSPOnvifMediaFactory *onvif_factory =
+ GST_RTSP_ONVIF_MEDIA_FACTORY (factory);
+
+ if (!gst_rtsp_onvif_media_factory_has_replay_support (onvif_factory)) {
+ if (unsupported->len)
+ g_string_append (unsupported, ", ");
+ g_string_append (unsupported, GST_RTSP_ONVIF_REPLAY_REQUIREMENT);
+ }
+ }
+
+
+out:
+ if (path)
+ g_free (path);
+ if (factory)
+ g_object_unref (factory);
+ if (mount_points)
+ g_object_unref (mount_points);
+
+ return g_string_free (unsupported, FALSE);
+}
+
+static GstRTSPStatusCode
+gst_rtsp_onvif_client_adjust_play_mode (GstRTSPClient * client,
+ GstRTSPContext * ctx, GstRTSPTimeRange ** range, GstSeekFlags * flags,
+ gdouble * rate, GstClockTime * trickmode_interval,
+ gboolean * enable_rate_control)
+{
+ GstRTSPStatusCode ret = GST_RTSP_STS_BAD_REQUEST;
+ gchar **split = NULL;
+ gchar *str;
+
+ if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_FRAMES,
+ &str, 0) == GST_RTSP_OK) {
+
+ split = g_strsplit (str, "/", 2);
+
+ if (!g_strcmp0 (split[0], "intra")) {
+ if (split[1]) {
+ guint64 interval;
+ gchar *end;
+
+ interval = g_ascii_strtoull (split[1], &end, 10);
+
+ if (!end || *end != '\0') {
+ GST_ERROR ("Unexpected interval value %s", split[1]);
+ goto done;
+ }
+
+ *trickmode_interval = interval * GST_MSECOND;
+ }
+ *flags |= GST_SEEK_FLAG_TRICKMODE_KEY_UNITS;
+ } else if (!g_strcmp0 (split[0], "predicted")) {
+ if (split[1]) {
+ GST_ERROR ("Predicted frames mode does not allow an interval (%s)",
+ str);
+ goto done;
+ }
+ *flags |= GST_SEEK_FLAG_TRICKMODE_FORWARD_PREDICTED;
+ } else {
+ GST_ERROR ("Invalid frames mode (%s)", str);
+ goto done;
+ }
+ }
+
+ if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RATE_CONTROL,
+ &str, 0) == GST_RTSP_OK) {
+ if (!g_strcmp0 (str, "no")) {
+ *enable_rate_control = FALSE;
+ } else if (!g_strcmp0 (str, "yes")) {
+ *enable_rate_control = TRUE;
+ } else {
+ GST_ERROR ("Invalid rate control header: %s", str);
+ goto done;
+ }
+ }
+
+ ret = GST_RTSP_STS_OK;
+
+done:
+ if (split)
+ g_strfreev (split);
+ return ret;
+}
+
+static GstRTSPStatusCode
+gst_rtsp_onvif_client_adjust_play_response (GstRTSPClient * client,
+ GstRTSPContext * ctx)
+{
+ GstRTSPStatusCode ret = GST_RTSP_STS_OK;
+ gchar *str;
+
+ if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RATE_CONTROL,
+ &str, 0) == GST_RTSP_OK) {
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_RATE_CONTROL,
+ gst_rtsp_media_get_rate_control (ctx->media) ? "yes" : "no");
+ }
+
+ return ret;
+}
+
+static void
+gst_rtsp_onvif_client_class_init (GstRTSPOnvifClientClass * klass)
+{
+ GstRTSPClientClass *client_klass = (GstRTSPClientClass *) klass;
+
+ client_klass->check_requirements = gst_rtsp_onvif_client_check_requirements;
+ client_klass->adjust_play_mode = gst_rtsp_onvif_client_adjust_play_mode;
+ client_klass->adjust_play_response =
+ gst_rtsp_onvif_client_adjust_play_response;
+}
+
+static void
+gst_rtsp_onvif_client_init (GstRTSPOnvifClient * client)
+{
+}
+
+/**
+ * gst_rtsp_onvif_client_new:
+ *
+ * Create a new #GstRTSPOnvifClient instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPOnvifClient
+ * Since: 1.18
+ */
+GstRTSPClient *
+gst_rtsp_onvif_client_new (void)
+{
+ GstRTSPClient *result;
+
+ result = g_object_new (GST_TYPE_RTSP_ONVIF_CLIENT, NULL);
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_ONVIF_CLIENT_H__
+#define __GST_RTSP_ONVIF_CLIENT_H__
+
+#include <gst/gst.h>
+#include "rtsp-client.h"
+
+#define GST_TYPE_RTSP_ONVIF_CLIENT (gst_rtsp_onvif_client_get_type ())
+#define GST_IS_RTSP_ONVIF_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_ONVIF_CLIENT))
+#define GST_IS_RTSP_ONVIF_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_ONVIF_CLIENT))
+#define GST_RTSP_ONVIF_CLIENT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_ONVIF_CLIENT, GstRTSPOnvifClientClass))
+#define GST_RTSP_ONVIF_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_ONVIF_CLIENT, GstRTSPOnvifClient))
+#define GST_RTSP_ONVIF_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_ONVIF_CLIENT, GstRTSPOnvifClientClass))
+#define GST_RTSP_ONVIF_CLIENT_CAST(obj) ((GstRTSPOnvifClient*)(obj))
+#define GST_RTSP_ONVIF_CLIENT_CLASS_CAST(klass) ((GstRTSPOnvifClientClass*)(klass))
+
+typedef struct GstRTSPOnvifClientClass GstRTSPOnvifClientClass;
+typedef struct GstRTSPOnvifClient GstRTSPOnvifClient;
+
+/**
+ * GstRTSPOnvifClient:
+ *
+ * Since: 1.14
+ */
+struct GstRTSPOnvifClientClass
+{
+ GstRTSPClientClass parent;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+struct GstRTSPOnvifClient
+{
+ GstRTSPClient parent;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_onvif_client_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPClient * gst_rtsp_onvif_client_new (void);
+
+#endif /* __GST_RTSP_ONVIF_CLIENT_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:rtsp-onvif-media-factory
+ * @short_description: A factory for ONVIF media pipelines
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPOnvifMedia
+ *
+ * The #GstRTSPOnvifMediaFactory is responsible for creating or recycling
+ * #GstRTSPMedia objects based on the passed URL. Different to
+ * #GstRTSPMediaFactory, this supports special ONVIF features and can create
+ * #GstRTSPOnvifMedia in addition to normal #GstRTSPMedia.
+ *
+ * Special ONVIF features that are currently supported is a backchannel for
+ * the client to send back media to the server in a normal PLAY media, see
+ * gst_rtsp_onvif_media_factory_set_backchannel_launch() and
+ * gst_rtsp_onvif_media_factory_set_backchannel_bandwidth().
+ *
+ * Since: 1.14
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-onvif-media-factory.h"
+#include "rtsp-onvif-media.h"
+#include "rtsp-onvif-server.h"
+
+struct GstRTSPOnvifMediaFactoryPrivate
+{
+ GMutex lock;
+ gchar *backchannel_launch;
+ guint backchannel_bandwidth;
+ gboolean has_replay_support;
+};
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPOnvifMediaFactory,
+ gst_rtsp_onvif_media_factory, GST_TYPE_RTSP_MEDIA_FACTORY);
+
+/**
+ * gst_rtsp_onvif_media_factory_requires_backchannel:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Checks whether the client request requires backchannel.
+ *
+ * Returns: %TRUE if the client request requires backchannel.
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_onvif_media_factory_requires_backchannel (GstRTSPMediaFactory *
+ factory, GstRTSPContext * ctx)
+{
+ GstRTSPMessage *msg = ctx->request;
+ GstRTSPResult res;
+ gint i;
+ gchar *reqs = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_ONVIF_MEDIA_FACTORY (factory), FALSE);
+
+ i = 0;
+ do {
+ res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
+
+ if (res == GST_RTSP_ENOTIMPL)
+ break;
+
+ if (strcmp (reqs, GST_RTSP_ONVIF_BACKCHANNEL_REQUIREMENT) == 0)
+ return TRUE;
+ } while (TRUE);
+
+ return FALSE;
+}
+
+static gchar *
+gst_rtsp_onvif_media_factory_gen_key (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPContext *ctx = gst_rtsp_context_get_current ();
+
+ /* Only medias where no backchannel was requested can be shared */
+ if (gst_rtsp_onvif_media_factory_requires_backchannel (factory, ctx))
+ return NULL;
+
+ return
+ GST_RTSP_MEDIA_FACTORY_CLASS
+ (gst_rtsp_onvif_media_factory_parent_class)->gen_key (factory, url);
+}
+
+static GstRTSPMedia *
+gst_rtsp_onvif_media_factory_construct (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMedia *media;
+ GstElement *element, *pipeline;
+ GstRTSPMediaFactoryClass *klass;
+ GType media_gtype;
+ gboolean got_backchannel_stream;
+ GstRTSPContext *ctx = gst_rtsp_context_get_current ();
+
+ /* Mostly a copy of the default implementation but with backchannel support below,
+ * unfortunately we can't re-use the default one because of how the virtual
+ * method is define */
+
+ /* Everything but play is unsupported */
+ if (gst_rtsp_media_factory_get_transport_mode (factory) !=
+ GST_RTSP_TRANSPORT_MODE_PLAY)
+ return NULL;
+
+ /* we only support onvif media here: otherwise a plain GstRTSPMediaFactory
+ * could've been used as well */
+ media_gtype = gst_rtsp_media_factory_get_media_gtype (factory);
+ if (!g_type_is_a (media_gtype, GST_TYPE_RTSP_ONVIF_MEDIA))
+ return NULL;
+
+ klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+ if (!klass->create_pipeline)
+ goto no_create;
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ if (element == NULL)
+ goto no_element;
+
+ /* create a new empty media */
+ media =
+ g_object_new (media_gtype, "element", element,
+ "transport-mode", GST_RTSP_TRANSPORT_MODE_PLAY, NULL);
+
+ /* this adds the non-backchannel streams */
+ gst_rtsp_media_collect_streams (media);
+
+ /* this adds the backchannel stream */
+ got_backchannel_stream =
+ gst_rtsp_onvif_media_collect_backchannel (GST_RTSP_ONVIF_MEDIA (media));
+ /* FIXME: This should not happen! We checked for that before */
+ if (gst_rtsp_onvif_media_factory_requires_backchannel (factory, ctx) &&
+ !got_backchannel_stream) {
+ g_object_unref (media);
+ return NULL;
+ }
+
+ pipeline = klass->create_pipeline (factory, media);
+ if (pipeline == NULL)
+ goto no_pipeline;
+
+ gst_rtsp_onvif_media_set_backchannel_bandwidth (GST_RTSP_ONVIF_MEDIA (media),
+ GST_RTSP_ONVIF_MEDIA_FACTORY (factory)->priv->backchannel_bandwidth);
+
+ return media;
+
+ /* ERRORS */
+no_create:
+ {
+ g_critical ("no create_pipeline function");
+ return NULL;
+ }
+no_element:
+ {
+ g_critical ("could not create element");
+ return NULL;
+ }
+no_pipeline:
+ {
+ g_critical ("can't create pipeline");
+ g_object_unref (media);
+ return NULL;
+ }
+}
+
+static GstElement *
+gst_rtsp_onvif_media_factory_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstElement *element;
+ GError *error = NULL;
+ gchar *launch;
+ GstRTSPContext *ctx = gst_rtsp_context_get_current ();
+
+ /* Mostly a copy of the default implementation but with backchannel support below,
+ * unfortunately we can't re-use the default one because of how the virtual
+ * method is define */
+
+ launch = gst_rtsp_media_factory_get_launch (factory);
+
+ /* we need a parse syntax */
+ if (launch == NULL)
+ goto no_launch;
+
+ /* parse the user provided launch line */
+ element =
+ gst_parse_launch_full (launch, NULL, GST_PARSE_FLAG_PLACE_IN_BIN, &error);
+ if (element == NULL)
+ goto parse_error;
+
+ g_free (launch);
+
+ if (error != NULL) {
+ /* a recoverable error was encountered */
+ GST_WARNING ("recoverable parsing error: %s", error->message);
+ g_error_free (error);
+ }
+
+ /* add backchannel pipeline part, if requested */
+ if (gst_rtsp_onvif_media_factory_requires_backchannel (factory, ctx)) {
+ GstRTSPOnvifMediaFactory *onvif_factory =
+ GST_RTSP_ONVIF_MEDIA_FACTORY (factory);
+ GstElement *backchannel_bin;
+ GstElement *backchannel_depay;
+ GstPad *depay_pad, *depay_ghostpad;
+
+ launch =
+ gst_rtsp_onvif_media_factory_get_backchannel_launch (onvif_factory);
+ if (launch == NULL)
+ goto no_launch_backchannel;
+
+ backchannel_bin =
+ gst_parse_bin_from_description_full (launch, FALSE, NULL,
+ GST_PARSE_FLAG_PLACE_IN_BIN, &error);
+ if (backchannel_bin == NULL)
+ goto parse_error_backchannel;
+
+ g_free (launch);
+
+ if (error != NULL) {
+ /* a recoverable error was encountered */
+ GST_WARNING ("recoverable parsing error: %s", error->message);
+ g_error_free (error);
+ }
+
+ gst_object_set_name (GST_OBJECT (backchannel_bin), "onvif-backchannel");
+
+ backchannel_depay =
+ gst_bin_get_by_name (GST_BIN (backchannel_bin), "depay_backchannel");
+ if (!backchannel_depay) {
+ gst_object_unref (backchannel_bin);
+ goto wrongly_formatted_backchannel_bin;
+ }
+
+ depay_pad = gst_element_get_static_pad (backchannel_depay, "sink");
+ if (!depay_pad) {
+ gst_object_unref (backchannel_depay);
+ gst_object_unref (backchannel_bin);
+ goto wrongly_formatted_backchannel_bin;
+ }
+
+ depay_ghostpad = gst_ghost_pad_new ("sink", depay_pad);
+ gst_element_add_pad (backchannel_bin, depay_ghostpad);
+
+ gst_bin_add (GST_BIN (element), backchannel_bin);
+ }
+
+ return element;
+
+ /* ERRORS */
+no_launch:
+ {
+ g_critical ("no launch line specified");
+ g_free (launch);
+ return NULL;
+ }
+parse_error:
+ {
+ g_critical ("could not parse launch syntax (%s): %s", launch,
+ (error ? error->message : "unknown reason"));
+ if (error)
+ g_error_free (error);
+ g_free (launch);
+ return NULL;
+ }
+no_launch_backchannel:
+ {
+ g_critical ("no backchannel launch line specified");
+ gst_object_unref (element);
+ return NULL;
+ }
+parse_error_backchannel:
+ {
+ g_critical ("could not parse backchannel launch syntax (%s): %s", launch,
+ (error ? error->message : "unknown reason"));
+ if (error)
+ g_error_free (error);
+ g_free (launch);
+ gst_object_unref (element);
+ return NULL;
+ }
+
+wrongly_formatted_backchannel_bin:
+ {
+ g_critical ("invalidly formatted backchannel bin");
+
+ gst_object_unref (element);
+ return NULL;
+ }
+}
+
+static gboolean
+ gst_rtsp_onvif_media_factory_has_backchannel_support_default
+ (GstRTSPOnvifMediaFactory * factory)
+{
+ /* No locking here, we just check if it's non-NULL */
+ return factory->priv->backchannel_launch != NULL;
+}
+
+static void
+gst_rtsp_onvif_media_factory_finalize (GObject * object)
+{
+ GstRTSPOnvifMediaFactory *factory = GST_RTSP_ONVIF_MEDIA_FACTORY (object);
+
+ g_free (factory->priv->backchannel_launch);
+ factory->priv->backchannel_launch = NULL;
+
+ g_mutex_clear (&factory->priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_onvif_media_factory_parent_class)->finalize (object);
+}
+
+static void
+gst_rtsp_onvif_media_factory_class_init (GstRTSPOnvifMediaFactoryClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstRTSPMediaFactoryClass *factory_klass = (GstRTSPMediaFactoryClass *) klass;
+
+ gobject_class->finalize = gst_rtsp_onvif_media_factory_finalize;
+
+ factory_klass->gen_key = gst_rtsp_onvif_media_factory_gen_key;
+ factory_klass->construct = gst_rtsp_onvif_media_factory_construct;
+ factory_klass->create_element = gst_rtsp_onvif_media_factory_create_element;
+
+ klass->has_backchannel_support =
+ gst_rtsp_onvif_media_factory_has_backchannel_support_default;
+}
+
+static void
+gst_rtsp_onvif_media_factory_init (GstRTSPOnvifMediaFactory * factory)
+{
+ factory->priv = gst_rtsp_onvif_media_factory_get_instance_private (factory);
+ g_mutex_init (&factory->priv->lock);
+}
+
+/**
+ * gst_rtsp_onvif_media_factory_set_backchannel_launch:
+ * @factory: a #GstRTSPMediaFactory
+ * @launch: the launch description
+ *
+ * The gst_parse_launch() line to use for constructing the ONVIF backchannel
+ * pipeline in the default prepare vmethod if requested by the client.
+ *
+ * The pipeline description should return a GstBin as the toplevel element
+ * which can be accomplished by enclosing the description with brackets '('
+ * ')'.
+ *
+ * The description should return a pipeline with a single depayloader named
+ * depay_backchannel. A caps query on the depayloader's sinkpad should return
+ * all possible, complete RTP caps that are going to be supported. At least
+ * the payload type, clock-rate and encoding-name need to be specified.
+ *
+ * Note: The pipeline part passed here must end in sinks that are not waiting
+ * until pre-rolling before reaching the PAUSED state, i.e. setting
+ * async=false on #GstBaseSink. Otherwise the whole media will not be able to
+ * prepare.
+ *
+ * Since: 1.14
+ */
+void
+gst_rtsp_onvif_media_factory_set_backchannel_launch (GstRTSPOnvifMediaFactory *
+ factory, const gchar * launch)
+{
+ g_return_if_fail (GST_IS_RTSP_ONVIF_MEDIA_FACTORY (factory));
+
+ g_mutex_lock (&factory->priv->lock);
+ g_free (factory->priv->backchannel_launch);
+ factory->priv->backchannel_launch = g_strdup (launch);
+ g_mutex_unlock (&factory->priv->lock);
+}
+
+/**
+ * gst_rtsp_onvif_media_factory_get_backchannel_launch:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the gst_parse_launch() pipeline description that will be used in the
+ * default prepare vmethod for generating the ONVIF backchannel pipeline.
+ *
+ * Returns: (transfer full): the configured backchannel launch description. g_free() after
+ * usage.
+ *
+ * Since: 1.14
+ */
+gchar *
+gst_rtsp_onvif_media_factory_get_backchannel_launch (GstRTSPOnvifMediaFactory *
+ factory)
+{
+ gchar *launch;
+
+ g_return_val_if_fail (GST_IS_RTSP_ONVIF_MEDIA_FACTORY (factory), NULL);
+
+ g_mutex_lock (&factory->priv->lock);
+ launch = g_strdup (factory->priv->backchannel_launch);
+ g_mutex_unlock (&factory->priv->lock);
+
+ return launch;
+}
+
+/**
+ * gst_rtsp_onvif_media_factory_has_backchannel_support:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Returns %TRUE if an ONVIF backchannel is supported by the media factory.
+ *
+ * Returns: %TRUE if an ONVIF backchannel is supported by the media factory.
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_onvif_media_factory_has_backchannel_support (GstRTSPOnvifMediaFactory *
+ factory)
+{
+ GstRTSPOnvifMediaFactoryClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_ONVIF_MEDIA_FACTORY (factory), FALSE);
+
+ klass = GST_RTSP_ONVIF_MEDIA_FACTORY_GET_CLASS (factory);
+
+ if (klass->has_backchannel_support)
+ return klass->has_backchannel_support (factory);
+
+ return FALSE;
+}
+
+/**
+ * gst_rtsp_onvif_media_factory_has_replay_support:
+ *
+ * Returns: %TRUE if ONVIF replay is supported by the media factory.
+ *
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_onvif_media_factory_has_replay_support (GstRTSPOnvifMediaFactory *
+ factory)
+{
+ gboolean has_replay_support;
+
+ g_mutex_lock (&factory->priv->lock);
+ has_replay_support = factory->priv->has_replay_support;
+ g_mutex_unlock (&factory->priv->lock);
+
+ return has_replay_support;
+}
+
+/**
+ * gst_rtsp_onvif_media_factory_set_replay_support:
+ *
+ * Set to %TRUE if ONVIF replay is supported by the media factory.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_onvif_media_factory_set_replay_support (GstRTSPOnvifMediaFactory *
+ factory, gboolean has_replay_support)
+{
+ g_mutex_lock (&factory->priv->lock);
+ factory->priv->has_replay_support = has_replay_support;
+ g_mutex_unlock (&factory->priv->lock);
+}
+
+/**
+ * gst_rtsp_onvif_media_factory_set_backchannel_bandwidth:
+ * @factory: a #GstRTSPMediaFactory
+ * @bandwidth: the bandwidth in bits per second
+ *
+ * Set the configured/supported bandwidth of the ONVIF backchannel pipeline in
+ * bits per second.
+ *
+ * Since: 1.14
+ */
+void
+gst_rtsp_onvif_media_factory_set_backchannel_bandwidth (GstRTSPOnvifMediaFactory
+ * factory, guint bandwidth)
+{
+ g_return_if_fail (GST_IS_RTSP_ONVIF_MEDIA_FACTORY (factory));
+
+ g_mutex_lock (&factory->priv->lock);
+ factory->priv->backchannel_bandwidth = bandwidth;
+ g_mutex_unlock (&factory->priv->lock);
+}
+
+/**
+ * gst_rtsp_onvif_media_factory_get_backchannel_bandwidth:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the configured/supported bandwidth of the ONVIF backchannel pipeline in
+ * bits per second.
+ *
+ * Returns: the configured/supported backchannel bandwidth.
+ *
+ * Since: 1.14
+ */
+guint
+gst_rtsp_onvif_media_factory_get_backchannel_bandwidth (GstRTSPOnvifMediaFactory
+ * factory)
+{
+ guint bandwidth;
+
+ g_return_val_if_fail (GST_IS_RTSP_ONVIF_MEDIA_FACTORY (factory), 0);
+
+ g_mutex_lock (&factory->priv->lock);
+ bandwidth = factory->priv->backchannel_bandwidth;
+ g_mutex_unlock (&factory->priv->lock);
+
+ return bandwidth;
+}
+
+/**
+ * gst_rtsp_onvif_media_factory_new:
+ *
+ * Create a new #GstRTSPOnvifMediaFactory
+ *
+ * Returns: A new #GstRTSPOnvifMediaFactory
+ *
+ * Since: 1.14
+ */
+GstRTSPMediaFactory *
+gst_rtsp_onvif_media_factory_new (void)
+{
+ return g_object_new (GST_TYPE_RTSP_ONVIF_MEDIA_FACTORY, NULL);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_ONVIF_MEDIA_FACTORY_H__
+#define __GST_RTSP_ONVIF_MEDIA_FACTORY_H__
+
+#include <gst/gst.h>
+#include "rtsp-media-factory.h"
+
+#define GST_TYPE_RTSP_ONVIF_MEDIA_FACTORY (gst_rtsp_onvif_media_factory_get_type ())
+#define GST_IS_RTSP_ONVIF_MEDIA_FACTORY(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_ONVIF_MEDIA_FACTORY))
+#define GST_IS_RTSP_ONVIF_MEDIA_FACTORY_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_ONVIF_MEDIA_FACTORY))
+#define GST_RTSP_ONVIF_MEDIA_FACTORY_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_ONVIF_MEDIA_FACTORY, GstRTSPOnvifMediaFactoryClass))
+#define GST_RTSP_ONVIF_MEDIA_FACTORY(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_ONVIF_MEDIA_FACTORY, GstRTSPOnvifMediaFactory))
+#define GST_RTSP_ONVIF_MEDIA_FACTORY_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_ONVIF_MEDIA_FACTORY, GstRTSPOnvifMediaFactoryClass))
+#define GST_RTSP_ONVIF_MEDIA_FACTORY_CAST(obj) ((GstRTSPOnvifMediaFactory*)(obj))
+#define GST_RTSP_ONVIF_MEDIA_FACTORY_CLASS_CAST(klass) ((GstRTSPOnvifMediaFactoryClass*)(klass))
+
+typedef struct GstRTSPOnvifMediaFactoryClass GstRTSPOnvifMediaFactoryClass;
+typedef struct GstRTSPOnvifMediaFactory GstRTSPOnvifMediaFactory;
+typedef struct GstRTSPOnvifMediaFactoryPrivate GstRTSPOnvifMediaFactoryPrivate;
+
+/**
+ * GstRTSPOnvifMediaFactory:
+ *
+ * Since: 1.14
+ */
+struct GstRTSPOnvifMediaFactoryClass
+{
+ GstRTSPMediaFactoryClass parent;
+ gboolean (*has_backchannel_support) (GstRTSPOnvifMediaFactory * factory);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+struct GstRTSPOnvifMediaFactory
+{
+ GstRTSPMediaFactory parent;
+ GstRTSPOnvifMediaFactoryPrivate *priv;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_onvif_media_factory_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPMediaFactory *gst_rtsp_onvif_media_factory_new (void);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_onvif_media_factory_set_backchannel_launch (GstRTSPOnvifMediaFactory *
+ factory, const gchar * launch);
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_onvif_media_factory_get_backchannel_launch (GstRTSPOnvifMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_onvif_media_factory_has_backchannel_support (GstRTSPOnvifMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_onvif_media_factory_has_replay_support (GstRTSPOnvifMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_onvif_media_factory_set_replay_support (GstRTSPOnvifMediaFactory * factory, gboolean has_replay_support);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_onvif_media_factory_set_backchannel_bandwidth (GstRTSPOnvifMediaFactory * factory, guint bandwidth);
+GST_RTSP_SERVER_API
+guint gst_rtsp_onvif_media_factory_get_backchannel_bandwidth (GstRTSPOnvifMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_onvif_media_factory_requires_backchannel (GstRTSPMediaFactory * factory, GstRTSPContext * ctx);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPOnvifMediaFactory, gst_object_unref)
+#endif
+
+#endif /* __GST_RTSP_ONVIF_MEDIA_FACTORY_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:rtsp-onvif-media
+ * @short_description: The ONVIF media pipeline
+ * @see_also: #GstRTSPMedia, #GstRTSPOnvifMediaFactory, #GstRTSPStream, #GstRTSPSession,
+ * #GstRTSPSessionMedia
+ *
+ * a #GstRTSPOnvifMedia contains the complete GStreamer pipeline to manage the
+ * streaming to the clients. The actual data transfer is done by the
+ * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
+ *
+ * On top of #GstRTSPMedia this subclass adds special ONVIF features.
+ * Special ONVIF features that are currently supported is a backchannel for
+ * the client to send back media to the server in a normal PLAY media. To
+ * handle the ONVIF backchannel, a #GstRTSPOnvifMediaFactory and
+ * #GstRTSPOnvifServer has to be used.
+ *
+ * Since: 1.14
+ *
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "rtsp-onvif-media.h"
+#include "rtsp-latency-bin.h"
+
+struct GstRTSPOnvifMediaPrivate
+{
+ GMutex lock;
+ guint backchannel_bandwidth;
+};
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPOnvifMedia, gst_rtsp_onvif_media,
+ GST_TYPE_RTSP_MEDIA);
+
+static gboolean
+gst_rtsp_onvif_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info)
+{
+ guint i, n_streams;
+ gchar *rangestr;
+ gboolean res;
+
+ /* Mostly a copy of gst_rtsp_sdp_from_media() which handles the backchannel
+ * stream separately and adds sendonly/recvonly attributes to each media
+ */
+
+ n_streams = gst_rtsp_media_n_streams (media);
+
+ rangestr = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ if (rangestr == NULL)
+ goto not_prepared;
+
+ gst_sdp_message_add_attribute (sdp, "range", rangestr);
+ g_free (rangestr);
+
+ res = TRUE;
+ for (i = 0; res && (i < n_streams); i++) {
+ GstRTSPStream *stream;
+ GstCaps *caps = NULL;
+ GstRTSPProfile profiles;
+ guint mask;
+ GstPad *sinkpad = NULL;
+ guint n_caps, j;
+
+ /* Mostly a copy of gst_rtsp_sdp_from_stream() which handles the
+ * backchannel stream separately */
+
+ stream = gst_rtsp_media_get_stream (media, i);
+
+ if ((sinkpad = gst_rtsp_stream_get_sinkpad (stream))) {
+ caps = gst_pad_query_caps (sinkpad, NULL);
+ } else {
+ caps = gst_rtsp_stream_get_caps (stream);
+ }
+
+ if (caps == NULL) {
+ GST_ERROR ("stream %p has no caps", stream);
+ res = FALSE;
+ if (sinkpad)
+ gst_object_unref (sinkpad);
+ break;
+ } else if (!sinkpad && !gst_caps_is_fixed (caps)) {
+ GST_ERROR ("stream %p has unfixed caps", stream);
+ res = FALSE;
+ gst_caps_unref (caps);
+ break;
+ }
+
+ n_caps = gst_caps_get_size (caps);
+ for (j = 0; res && j < n_caps; j++) {
+ GstStructure *s = gst_caps_get_structure (caps, j);
+ GstCaps *media_caps = gst_caps_new_full (gst_structure_copy (s), NULL);
+
+ if (!gst_caps_is_fixed (media_caps)) {
+ GST_ERROR ("media caps for stream %p are not all fixed", stream);
+ res = FALSE;
+ gst_caps_unref (media_caps);
+ break;
+ }
+
+ /* make a new media for each profile */
+ profiles = gst_rtsp_stream_get_profiles (stream);
+ mask = 1;
+ res = TRUE;
+ while (res && (profiles >= mask)) {
+ GstRTSPProfile prof = profiles & mask;
+
+ if (prof) {
+ res = gst_rtsp_sdp_make_media (sdp, info, stream, media_caps, prof);
+ if (res) {
+ GstSDPMedia *smedia =
+ &g_array_index (sdp->medias, GstSDPMedia, sdp->medias->len - 1);
+ gchar *x_onvif_track, *media_str;
+
+ media_str =
+ g_ascii_strup (gst_structure_get_string (s, "media"), -1);
+ x_onvif_track =
+ g_strdup_printf ("%s%03d", media_str, sdp->medias->len - 1);
+ gst_sdp_media_add_attribute (smedia, "x-onvif-track",
+ x_onvif_track);
+ g_free (x_onvif_track);
+ g_free (media_str);
+
+ if (sinkpad) {
+ GstRTSPOnvifMedia *onvif_media = GST_RTSP_ONVIF_MEDIA (media);
+
+ gst_sdp_media_add_attribute (smedia, "sendonly", "");
+ if (onvif_media->priv->backchannel_bandwidth > 0)
+ gst_sdp_media_add_bandwidth (smedia, GST_SDP_BWTYPE_AS,
+ onvif_media->priv->backchannel_bandwidth);
+ } else {
+ gst_sdp_media_add_attribute (smedia, "recvonly", "");
+ }
+ }
+ }
+
+ mask <<= 1;
+ }
+
+ if (sinkpad) {
+ GstStructure *s = gst_caps_get_structure (media_caps, 0);
+ gint pt = -1;
+
+ if (!gst_structure_get_int (s, "payload", &pt) || pt < 0) {
+ GST_ERROR ("stream %p has no payload type", stream);
+ res = FALSE;
+ gst_caps_unref (media_caps);
+ gst_object_unref (sinkpad);
+ break;
+ }
+
+ gst_rtsp_stream_set_pt_map (stream, pt, media_caps);
+ }
+
+ gst_caps_unref (media_caps);
+ }
+
+ gst_caps_unref (caps);
+ if (sinkpad)
+ gst_object_unref (sinkpad);
+ }
+
+ {
+ GstNetTimeProvider *provider;
+
+ if ((provider =
+ gst_rtsp_media_get_time_provider (media, info->server_ip, 0))) {
+ GstClock *clock;
+ gchar *address, *str;
+ gint port;
+
+ g_object_get (provider, "clock", &clock, "address", &address, "port",
+ &port, NULL);
+
+ str = g_strdup_printf ("GstNetTimeProvider %s %s:%d %" G_GUINT64_FORMAT,
+ g_type_name (G_TYPE_FROM_INSTANCE (clock)), address, port,
+ gst_clock_get_time (clock));
+
+ gst_sdp_message_add_attribute (sdp, "x-gst-clock", str);
+ g_free (str);
+ gst_object_unref (clock);
+ g_free (address);
+ gst_object_unref (provider);
+ }
+ }
+
+ return res;
+
+ /* ERRORS */
+not_prepared:
+ {
+ GST_ERROR ("media %p is not prepared", media);
+ return FALSE;
+ }
+}
+
+static void
+gst_rtsp_onvif_media_finalize (GObject * object)
+{
+ GstRTSPOnvifMedia *media = GST_RTSP_ONVIF_MEDIA (object);
+
+ g_mutex_clear (&media->priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_onvif_media_parent_class)->finalize (object);
+}
+
+static void
+gst_rtsp_onvif_media_class_init (GstRTSPOnvifMediaClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstRTSPMediaClass *media_class = (GstRTSPMediaClass *) klass;
+
+ gobject_class->finalize = gst_rtsp_onvif_media_finalize;
+
+ media_class->setup_sdp = gst_rtsp_onvif_media_setup_sdp;
+}
+
+static void
+gst_rtsp_onvif_media_init (GstRTSPOnvifMedia * media)
+{
+ media->priv = gst_rtsp_onvif_media_get_instance_private (media);
+ g_mutex_init (&media->priv->lock);
+}
+
+/**
+ * gst_rtsp_onvif_media_collect_backchannel:
+ * @media: a #GstRTSPOnvifMedia
+ *
+ * Find the ONVIF backchannel depayloader element. It should be named
+ * 'depay_backchannel', be placed in a bin called 'onvif-backchannel'
+ * and return all supported RTP caps on a caps query. Complete RTP caps with
+ * at least the payload type, clock-rate and encoding-name are required.
+ *
+ * A new #GstRTSPStream is created for the backchannel if found.
+ *
+ * Returns: %TRUE if a backchannel stream could be found and created
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_onvif_media_collect_backchannel (GstRTSPOnvifMedia * media)
+{
+ GstElement *element, *backchannel_bin = NULL;
+ GstElement *latency_bin;
+ GstPad *pad = NULL;
+ gboolean ret = FALSE;
+
+ g_return_val_if_fail (GST_IS_RTSP_ONVIF_MEDIA (media), FALSE);
+
+ element = gst_rtsp_media_get_element (GST_RTSP_MEDIA (media));
+ if (!element)
+ return ret;
+
+ backchannel_bin =
+ gst_bin_get_by_name (GST_BIN (element), "onvif-backchannel");
+ if (!backchannel_bin)
+ goto out;
+
+ /* We don't want the backchannel element, which is a receiver, to affect
+ * latency on the complete pipeline. That's why we remove it from the
+ * pipeline and add it to a @GstRTSPLatencyBin which will prevent it from
+ * messing up pipelines latency. The extra reference is needed so that it
+ * is not freed in case the pipeline holds the the only ref to it.
+ *
+ * TODO: a more generic solution should be implemented in
+ * gst_rtsp_media_collect_streams() where all receivers are encapsulated
+ * in a @GstRTSPLatencyBin in cases when there are senders too. */
+ gst_object_ref (backchannel_bin);
+ gst_bin_remove (GST_BIN (element), backchannel_bin);
+
+ latency_bin = gst_rtsp_latency_bin_new (backchannel_bin);
+ g_assert (latency_bin);
+
+ gst_bin_add (GST_BIN (element), latency_bin);
+
+ pad = gst_element_get_static_pad (latency_bin, "sink");
+ if (!pad)
+ goto out;
+
+ gst_rtsp_media_create_stream (GST_RTSP_MEDIA (media), latency_bin, pad);
+ ret = TRUE;
+
+out:
+ if (pad)
+ gst_object_unref (pad);
+ if (backchannel_bin)
+ gst_object_unref (backchannel_bin);
+ gst_object_unref (element);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_onvif_media_set_backchannel_bandwidth:
+ * @media: a #GstRTSPMedia
+ * @bandwidth: the bandwidth in bits per second
+ *
+ * Set the configured/supported bandwidth of the ONVIF backchannel pipeline in
+ * bits per second.
+ *
+ * Since: 1.14
+ */
+void
+gst_rtsp_onvif_media_set_backchannel_bandwidth (GstRTSPOnvifMedia * media,
+ guint bandwidth)
+{
+ g_return_if_fail (GST_IS_RTSP_ONVIF_MEDIA (media));
+
+ g_mutex_lock (&media->priv->lock);
+ media->priv->backchannel_bandwidth = bandwidth;
+ g_mutex_unlock (&media->priv->lock);
+}
+
+/**
+ * gst_rtsp_onvif_media_get_backchannel_bandwidth:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the configured/supported bandwidth of the ONVIF backchannel pipeline in
+ * bits per second.
+ *
+ * Returns: the configured/supported backchannel bandwidth.
+ *
+ * Since: 1.14
+ */
+guint
+gst_rtsp_onvif_media_get_backchannel_bandwidth (GstRTSPOnvifMedia * media)
+{
+ guint bandwidth;
+
+ g_return_val_if_fail (GST_IS_RTSP_ONVIF_MEDIA (media), 0);
+
+ g_mutex_lock (&media->priv->lock);
+ bandwidth = media->priv->backchannel_bandwidth;
+ g_mutex_unlock (&media->priv->lock);
+
+ return bandwidth;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_ONVIF_MEDIA_H__
+#define __GST_RTSP_ONVIF_MEDIA_H__
+
+#include <gst/gst.h>
+#include "rtsp-media.h"
+
+#define GST_TYPE_RTSP_ONVIF_MEDIA (gst_rtsp_onvif_media_get_type ())
+#define GST_IS_RTSP_ONVIF_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_ONVIF_MEDIA))
+#define GST_IS_RTSP_ONVIF_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_ONVIF_MEDIA))
+#define GST_RTSP_ONVIF_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_ONVIF_MEDIA, GstRTSPOnvifMediaClass))
+#define GST_RTSP_ONVIF_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_ONVIF_MEDIA, GstRTSPOnvifMedia))
+#define GST_RTSP_ONVIF_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_ONVIF_MEDIA, GstRTSPOnvifMediaClass))
+#define GST_RTSP_ONVIF_MEDIA_CAST(obj) ((GstRTSPOnvifMedia*)(obj))
+#define GST_RTSP_ONVIF_MEDIA_CLASS_CAST(klass) ((GstRTSPOnvifMediaClass*)(klass))
+
+typedef struct GstRTSPOnvifMediaClass GstRTSPOnvifMediaClass;
+typedef struct GstRTSPOnvifMedia GstRTSPOnvifMedia;
+typedef struct GstRTSPOnvifMediaPrivate GstRTSPOnvifMediaPrivate;
+
+/**
+ * GstRTSPOnvifMedia:
+ *
+ * Since: 1.14
+ */
+struct GstRTSPOnvifMediaClass
+{
+ GstRTSPMediaClass parent;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+struct GstRTSPOnvifMedia
+{
+ GstRTSPMedia parent;
+ GstRTSPOnvifMediaPrivate *priv;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_onvif_media_get_type (void);
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_onvif_media_collect_backchannel (GstRTSPOnvifMedia * media);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_onvif_media_set_backchannel_bandwidth (GstRTSPOnvifMedia * media, guint bandwidth);
+GST_RTSP_SERVER_API
+guint gst_rtsp_onvif_media_get_backchannel_bandwidth (GstRTSPOnvifMedia * media);
+
+#endif /* __GST_RTSP_ONVIF_MEDIA_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-onvif-server
+ * @short_description: The main server object
+ * @see_also: #GstRTSPOnvifMediaFactory, #GstRTSPClient
+ *
+ * The server object is the object listening for connections on a port and
+ * creating #GstRTSPOnvifClient objects to handle those connections.
+ *
+ * The only different to #GstRTSPServer is that #GstRTSPOnvifServer creates
+ * #GstRTSPOnvifClient that have special handling for ONVIF specific features,
+ * like a backchannel that allows clients to send back media to the server.
+ *
+ * Since: 1.14
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "rtsp-context.h"
+#include "rtsp-onvif-server.h"
+#include "rtsp-onvif-client.h"
+
+G_DEFINE_TYPE (GstRTSPOnvifServer, gst_rtsp_onvif_server, GST_TYPE_RTSP_SERVER);
+
+static GstRTSPClient *
+gst_rtsp_onvif_server_create_client (GstRTSPServer * server)
+{
+ GstRTSPClient *client;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPMountPoints *mount_points;
+ GstRTSPAuth *auth;
+ GstRTSPThreadPool *thread_pool;
+
+ /* a new client connected, create a session to handle the client. */
+ client = g_object_new (GST_TYPE_RTSP_ONVIF_CLIENT, NULL);
+
+ /* set the session pool that this client should use */
+ session_pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_client_set_session_pool (client, session_pool);
+ g_object_unref (session_pool);
+ /* set the mount points that this client should use */
+ mount_points = gst_rtsp_server_get_mount_points (server);
+ gst_rtsp_client_set_mount_points (client, mount_points);
+ g_object_unref (mount_points);
+ /* set authentication manager */
+ auth = gst_rtsp_server_get_auth (server);
+ gst_rtsp_client_set_auth (client, auth);
+ if (auth)
+ g_object_unref (auth);
+ /* set threadpool */
+ thread_pool = gst_rtsp_server_get_thread_pool (server);
+ gst_rtsp_client_set_thread_pool (client, thread_pool);
+ g_object_unref (thread_pool);
+
+ return client;
+}
+
+/**
+ * gst_rtsp_onvif_server_new:
+ *
+ * Create a new #GstRTSPOnvifServer instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPOnvifServer
+ */
+GstRTSPServer *
+gst_rtsp_onvif_server_new (void)
+{
+ return g_object_new (GST_TYPE_RTSP_ONVIF_SERVER, NULL);
+}
+
+static void
+gst_rtsp_onvif_server_class_init (GstRTSPOnvifServerClass * klass)
+{
+ GstRTSPServerClass *server_klass = (GstRTSPServerClass *) klass;
+
+ server_klass->create_client = gst_rtsp_onvif_server_create_client;
+}
+
+static void
+gst_rtsp_onvif_server_init (GstRTSPOnvifServer * server)
+{
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_ONVIF_SERVER_H__
+#define __GST_RTSP_ONVIF_SERVER_H__
+
+#include <gst/gst.h>
+#include "rtsp-server-object.h"
+
+#define GST_TYPE_RTSP_ONVIF_SERVER (gst_rtsp_onvif_server_get_type ())
+#define GST_IS_RTSP_ONVIF_SERVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_ONVIF_SERVER))
+#define GST_IS_RTSP_ONVIF_SERVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_ONVIF_SERVER))
+#define GST_RTSP_ONVIF_SERVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_ONVIF_SERVER, GstRTSPOnvifServerClass))
+#define GST_RTSP_ONVIF_SERVER(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_ONVIF_SERVER, GstRTSPOnvifServer))
+#define GST_RTSP_ONVIF_SERVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_ONVIF_SERVER, GstRTSPOnvifServerClass))
+#define GST_RTSP_ONVIF_SERVER_CAST(obj) ((GstRTSPOnvifServer*)(obj))
+#define GST_RTSP_ONVIF_SERVER_CLASS_CAST(klass) ((GstRTSPOnvifServerClass*)(klass))
+
+typedef struct GstRTSPOnvifServerClass GstRTSPOnvifServerClass;
+typedef struct GstRTSPOnvifServer GstRTSPOnvifServer;
+
+/**
+ * GstRTSPOnvifServer:
+ *
+ * Since: 1.14
+ */
+struct GstRTSPOnvifServerClass
+{
+ GstRTSPServerClass parent;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+struct GstRTSPOnvifServer
+{
+ GstRTSPServer parent;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_onvif_server_get_type (void);
+GST_RTSP_SERVER_API
+GstRTSPServer *gst_rtsp_onvif_server_new (void);
+
+#define GST_RTSP_ONVIF_BACKCHANNEL_REQUIREMENT "www.onvif.org/ver20/backchannel"
+#define GST_RTSP_ONVIF_REPLAY_REQUIREMENT "onvif-replay"
+
+#include "rtsp-onvif-client.h"
+#include "rtsp-onvif-media-factory.h"
+#include "rtsp-onvif-media.h"
+
+#endif /* __GST_RTSP_ONVIF_SERVER_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-params
+ * @short_description: Param get and set implementation
+ * @see_also: #GstRTSPClient
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-params.h"
+
+/**
+ * gst_rtsp_params_set:
+ * @client: a #GstRTSPClient
+ * @ctx: (transfer none): a #GstRTSPContext
+ *
+ * Set parameters (not implemented yet)
+ *
+ * Returns: a #GstRTSPResult
+ */
+GstRTSPResult
+gst_rtsp_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPStatusCode code;
+
+ /* FIXME, actually parse the request based on the mime type and try to repond
+ * with a list of the parameters */
+ code = GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD;
+
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_params_get:
+ * @client: a #GstRTSPClient
+ * @ctx: (transfer none): a #GstRTSPContext
+ *
+ * Get parameters (not implemented yet)
+ *
+ * Returns: a #GstRTSPResult
+ */
+GstRTSPResult
+gst_rtsp_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPStatusCode code;
+
+ /* FIXME, actually parse the request based on the mime type and try to repond
+ * with a list of the parameters */
+ code = GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD;
+
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ return GST_RTSP_OK;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp/gstrtspurl.h>
+#include <gst/rtsp/gstrtspmessage.h>
+
+#ifndef __GST_RTSP_PARAMS_H__
+#define __GST_RTSP_PARAMS_H__
+
+#include "rtsp-client.h"
+#include "rtsp-session.h"
+
+G_BEGIN_DECLS
+
+GST_RTSP_SERVER_API
+GstRTSPResult gst_rtsp_params_set (GstRTSPClient * client, GstRTSPContext * ctx);
+
+GST_RTSP_SERVER_API
+GstRTSPResult gst_rtsp_params_get (GstRTSPClient * client, GstRTSPContext * ctx);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_PARAMS_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-permissions
+ * @short_description: Roles and associated permissions
+ * @see_also: #GstRTSPToken, #GstRTSPAuth
+ *
+ * The #GstRTSPPermissions object contains an array of roles and associated
+ * permissions. The roles are represented with a string and the permissions with
+ * a generic #GstStructure.
+ *
+ * The permissions are deliberately kept generic. The possible values of the
+ * roles and #GstStructure keys and values are only determined by the #GstRTSPAuth
+ * object that performs the checks on the permissions and the current
+ * #GstRTSPToken.
+ *
+ * As a convenience function, gst_rtsp_permissions_is_allowed() can be used to
+ * check if the permissions contains a role that contains the boolean value
+ * %TRUE for the the given key.
+ *
+ * Last reviewed on 2013-07-15 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-permissions.h"
+
+typedef struct _GstRTSPPermissionsImpl
+{
+ GstRTSPPermissions permissions;
+
+ /* Roles, array of GstStructure */
+ GPtrArray *roles;
+} GstRTSPPermissionsImpl;
+
+static void
+free_structure (GstStructure * structure)
+{
+ gst_structure_set_parent_refcount (structure, NULL);
+ gst_structure_free (structure);
+}
+
+//GST_DEBUG_CATEGORY_STATIC (rtsp_permissions_debug);
+//#define GST_CAT_DEFAULT rtsp_permissions_debug
+
+GST_DEFINE_MINI_OBJECT_TYPE (GstRTSPPermissions, gst_rtsp_permissions);
+
+static void gst_rtsp_permissions_init (GstRTSPPermissionsImpl * permissions);
+
+static void
+_gst_rtsp_permissions_free (GstRTSPPermissions * permissions)
+{
+ GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+
+ g_ptr_array_free (impl->roles, TRUE);
+
+ g_slice_free1 (sizeof (GstRTSPPermissionsImpl), permissions);
+}
+
+static GstRTSPPermissions *
+_gst_rtsp_permissions_copy (GstRTSPPermissionsImpl * permissions)
+{
+ GstRTSPPermissionsImpl *copy;
+ guint i;
+
+ copy = (GstRTSPPermissionsImpl *) gst_rtsp_permissions_new ();
+
+ for (i = 0; i < permissions->roles->len; i++) {
+ GstStructure *entry = g_ptr_array_index (permissions->roles, i);
+ GstStructure *entry_copy = gst_structure_copy (entry);
+
+ gst_structure_set_parent_refcount (entry_copy,
+ ©->permissions.mini_object.refcount);
+ g_ptr_array_add (copy->roles, entry_copy);
+ }
+
+ return GST_RTSP_PERMISSIONS (copy);
+}
+
+static void
+gst_rtsp_permissions_init (GstRTSPPermissionsImpl * permissions)
+{
+ gst_mini_object_init (GST_MINI_OBJECT_CAST (permissions), 0,
+ GST_TYPE_RTSP_PERMISSIONS,
+ (GstMiniObjectCopyFunction) _gst_rtsp_permissions_copy, NULL,
+ (GstMiniObjectFreeFunction) _gst_rtsp_permissions_free);
+
+ permissions->roles =
+ g_ptr_array_new_with_free_func ((GDestroyNotify) free_structure);
+}
+
+static void
+add_role_from_structure (GstRTSPPermissionsImpl * impl,
+ GstStructure * structure)
+{
+ guint i, len;
+ const gchar *role = gst_structure_get_name (structure);
+
+ len = impl->roles->len;
+ for (i = 0; i < len; i++) {
+ GstStructure *entry = g_ptr_array_index (impl->roles, i);
+
+ if (gst_structure_has_name (entry, role)) {
+ g_ptr_array_remove_index_fast (impl->roles, i);
+ break;
+ }
+ }
+
+ gst_structure_set_parent_refcount (structure,
+ &impl->permissions.mini_object.refcount);
+ g_ptr_array_add (impl->roles, structure);
+}
+
+/**
+ * gst_rtsp_permissions_new:
+ *
+ * Create a new empty Authorization permissions.
+ *
+ * Returns: (transfer full): a new empty authorization permissions.
+ */
+GstRTSPPermissions *
+gst_rtsp_permissions_new (void)
+{
+ GstRTSPPermissionsImpl *permissions;
+
+ permissions = g_slice_new0 (GstRTSPPermissionsImpl);
+ gst_rtsp_permissions_init (permissions);
+
+ return GST_RTSP_PERMISSIONS (permissions);
+}
+
+/**
+ * gst_rtsp_permissions_add_permission_for_role:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ * @permission: the permission
+ * @allowed: whether the role has this permission or not
+ *
+ * Add a new @permission for @role to @permissions with the access in @allowed.
+ *
+ * Since: 1.14
+ */
+void
+gst_rtsp_permissions_add_permission_for_role (GstRTSPPermissions * permissions,
+ const gchar * role, const gchar * permission, gboolean allowed)
+{
+ GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+ guint i, len;
+
+ g_return_if_fail (GST_IS_RTSP_PERMISSIONS (permissions));
+ g_return_if_fail (gst_mini_object_is_writable (&permissions->mini_object));
+ g_return_if_fail (role != NULL);
+ g_return_if_fail (permission != NULL);
+
+ len = impl->roles->len;
+ for (i = 0; i < len; i++) {
+ GstStructure *entry = g_ptr_array_index (impl->roles, i);
+
+ if (gst_structure_has_name (entry, role)) {
+ gst_structure_set (entry, permission, G_TYPE_BOOLEAN, allowed, NULL);
+ return;
+ }
+ }
+
+ gst_rtsp_permissions_add_role (permissions, role,
+ permission, G_TYPE_BOOLEAN, allowed, NULL);
+}
+
+/**
+ * gst_rtsp_permissions_add_role_empty: (rename-to gst_rtsp_permissions_add_role)
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ *
+ * Add a new @role to @permissions without any permissions. You can add
+ * permissions for the role with gst_rtsp_permissions_add_permission_for_role().
+ *
+ * Since: 1.14
+ */
+void
+gst_rtsp_permissions_add_role_empty (GstRTSPPermissions * permissions,
+ const gchar * role)
+{
+ gst_rtsp_permissions_add_role (permissions, role, NULL);
+}
+
+/**
+ * gst_rtsp_permissions_add_role:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ * @fieldname: the first field name
+ * @...: additional arguments
+ *
+ * Add a new @role to @permissions with the given variables. The fields
+ * are the same layout as gst_structure_new().
+ */
+void
+gst_rtsp_permissions_add_role (GstRTSPPermissions * permissions,
+ const gchar * role, const gchar * fieldname, ...)
+{
+ va_list var_args;
+
+ va_start (var_args, fieldname);
+ gst_rtsp_permissions_add_role_valist (permissions, role, fieldname, var_args);
+ va_end (var_args);
+}
+
+/**
+ * gst_rtsp_permissions_add_role_valist:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ * @fieldname: the first field name
+ * @var_args: additional fields to add
+ *
+ * Add a new @role to @permissions with the given variables. Structure fields
+ * are set according to the varargs in a manner similar to gst_structure_new().
+ */
+void
+gst_rtsp_permissions_add_role_valist (GstRTSPPermissions * permissions,
+ const gchar * role, const gchar * fieldname, va_list var_args)
+{
+ GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+ GstStructure *structure;
+
+ g_return_if_fail (GST_IS_RTSP_PERMISSIONS (permissions));
+ g_return_if_fail (gst_mini_object_is_writable (&permissions->mini_object));
+ g_return_if_fail (role != NULL);
+
+ structure = gst_structure_new_valist (role, fieldname, var_args);
+ g_return_if_fail (structure != NULL);
+
+ add_role_from_structure (impl, structure);
+}
+
+/**
+ * gst_rtsp_permissions_add_role_from_structure:
+ *
+ * Add a new role to @permissions based on @structure, for example
+ * given a role named `tester`, which should be granted a permission named
+ * `permission1`, the structure could be created with:
+ *
+ * ```
+ * gst_structure_new ("tester", "permission1", G_TYPE_BOOLEAN, TRUE, NULL);
+ * ```
+ *
+ * Since: 1.14
+ */
+void
+gst_rtsp_permissions_add_role_from_structure (GstRTSPPermissions * permissions,
+ GstStructure * structure)
+{
+ GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+ GstStructure *copy;
+
+ g_return_if_fail (GST_IS_RTSP_PERMISSIONS (permissions));
+ g_return_if_fail (GST_IS_STRUCTURE (structure));
+
+ copy = gst_structure_copy (structure);
+
+ add_role_from_structure (impl, copy);
+}
+
+/**
+ * gst_rtsp_permissions_remove_role:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ *
+ * Remove all permissions for @role in @permissions.
+ */
+void
+gst_rtsp_permissions_remove_role (GstRTSPPermissions * permissions,
+ const gchar * role)
+{
+ GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+ guint i, len;
+
+ g_return_if_fail (GST_IS_RTSP_PERMISSIONS (permissions));
+ g_return_if_fail (gst_mini_object_is_writable (&permissions->mini_object));
+ g_return_if_fail (role != NULL);
+
+ len = impl->roles->len;
+ for (i = 0; i < len; i++) {
+ GstStructure *entry = g_ptr_array_index (impl->roles, i);
+
+ if (gst_structure_has_name (entry, role)) {
+ g_ptr_array_remove_index_fast (impl->roles, i);
+ break;
+ }
+ }
+}
+
+/**
+ * gst_rtsp_permissions_get_role:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ *
+ * Get all permissions for @role in @permissions.
+ *
+ * Returns: (transfer none): the structure with permissions for @role. It
+ * remains valid for as long as @permissions is valid.
+ */
+const GstStructure *
+gst_rtsp_permissions_get_role (GstRTSPPermissions * permissions,
+ const gchar * role)
+{
+ GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+ guint i, len;
+
+ g_return_val_if_fail (GST_IS_RTSP_PERMISSIONS (permissions), NULL);
+ g_return_val_if_fail (role != NULL, NULL);
+
+ len = impl->roles->len;
+ for (i = 0; i < len; i++) {
+ GstStructure *entry = g_ptr_array_index (impl->roles, i);
+
+ if (gst_structure_has_name (entry, role))
+ return entry;
+ }
+ return NULL;
+}
+
+/**
+ * gst_rtsp_permissions_is_allowed:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ * @permission: a permission
+ *
+ * Check if @role in @permissions is given permission for @permission.
+ *
+ * Returns: %TRUE if @role is allowed @permission.
+ */
+gboolean
+gst_rtsp_permissions_is_allowed (GstRTSPPermissions * permissions,
+ const gchar * role, const gchar * permission)
+{
+ const GstStructure *str;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_PERMISSIONS (permissions), FALSE);
+ g_return_val_if_fail (role != NULL, FALSE);
+ g_return_val_if_fail (permission != NULL, FALSE);
+
+ str = gst_rtsp_permissions_get_role (permissions, role);
+ if (str == NULL)
+ return FALSE;
+
+ if (!gst_structure_get_boolean (str, permission, &result))
+ result = FALSE;
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_PERMISSIONS_H__
+#define __GST_RTSP_PERMISSIONS_H__
+
+#include "rtsp-server-prelude.h"
+
+typedef struct _GstRTSPPermissions GstRTSPPermissions;
+
+G_BEGIN_DECLS
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_permissions_get_type (void);
+
+#define GST_TYPE_RTSP_PERMISSIONS (gst_rtsp_permissions_get_type ())
+#define GST_IS_RTSP_PERMISSIONS(obj) (GST_IS_MINI_OBJECT_TYPE (obj, GST_TYPE_RTSP_PERMISSIONS))
+#define GST_RTSP_PERMISSIONS_CAST(obj) ((GstRTSPPermissions*)(obj))
+#define GST_RTSP_PERMISSIONS(obj) (GST_RTSP_PERMISSIONS_CAST(obj))
+
+/**
+ * GstRTSPPermissions:
+ *
+ * The opaque permissions structure. It is used to define the permissions
+ * of objects in different roles.
+ */
+struct _GstRTSPPermissions {
+ GstMiniObject mini_object;
+};
+
+/* refcounting */
+/**
+ * gst_rtsp_permissions_ref:
+ * @permissions: The permissions to refcount
+ *
+ * Increase the refcount of this permissions.
+ *
+ * Returns: (transfer full): @permissions (for convenience when doing assignments)
+ */
+static inline GstRTSPPermissions *
+gst_rtsp_permissions_ref (GstRTSPPermissions * permissions)
+{
+ return (GstRTSPPermissions *) gst_mini_object_ref (GST_MINI_OBJECT_CAST (permissions));
+}
+
+/**
+ * gst_rtsp_permissions_unref:
+ * @permissions: (transfer full): the permissions to refcount
+ *
+ * Decrease the refcount of an permissions, freeing it if the refcount reaches 0.
+ */
+static inline void
+gst_rtsp_permissions_unref (GstRTSPPermissions * permissions)
+{
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (permissions));
+}
+
+
+GST_RTSP_SERVER_API
+GstRTSPPermissions * gst_rtsp_permissions_new (void);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_permissions_add_role (GstRTSPPermissions *permissions,
+ const gchar *role,
+ const gchar *fieldname, ...);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_permissions_add_role_valist (GstRTSPPermissions *permissions,
+ const gchar *role,
+ const gchar *fieldname,
+ va_list var_args);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_permissions_add_role_empty (GstRTSPPermissions * permissions,
+ const gchar * role);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_permissions_add_role_from_structure (GstRTSPPermissions * permissions,
+ GstStructure *structure);
+GST_RTSP_SERVER_API
+void gst_rtsp_permissions_add_permission_for_role (GstRTSPPermissions * permissions,
+ const gchar * role,
+ const gchar * permission,
+ gboolean allowed);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_permissions_remove_role (GstRTSPPermissions *permissions,
+ const gchar *role);
+
+GST_RTSP_SERVER_API
+const GstStructure * gst_rtsp_permissions_get_role (GstRTSPPermissions *permissions,
+ const gchar *role);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_permissions_is_allowed (GstRTSPPermissions *permissions,
+ const gchar *role, const gchar *permission);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPPermissions, gst_rtsp_permissions_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_PERMISSIONS_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
+/**
+ * SECTION:rtsp-sdp
+ * @short_description: Make SDP messages
+ * @see_also: #GstRTSPMedia
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/net/net.h>
+#include <gst/sdp/gstmikey.h>
+
+#include "rtsp-sdp.h"
+
+static gboolean
+get_info_from_tags (GstPad * pad, GstEvent ** event, gpointer user_data)
+{
+ GstSDPMedia *media = (GstSDPMedia *) user_data;
+
+ if (GST_EVENT_TYPE (*event) == GST_EVENT_TAG) {
+ GstTagList *tags;
+ guint bitrate = 0;
+
+ gst_event_parse_tag (*event, &tags);
+
+ if (gst_tag_list_get_scope (tags) != GST_TAG_SCOPE_STREAM)
+ return TRUE;
+
+ if (!gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE,
+ &bitrate) || bitrate == 0)
+ if (!gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &bitrate) ||
+ bitrate == 0)
+ return TRUE;
+
+ /* set bandwidth (kbits/s) */
+ gst_sdp_media_add_bandwidth (media, GST_SDP_BWTYPE_AS, bitrate / 1000);
+
+ return FALSE;
+
+ }
+
+ return TRUE;
+}
+
+static void
+update_sdp_from_tags (GstRTSPStream * stream, GstSDPMedia * stream_media)
+{
+ GstPad *src_pad;
+
+ src_pad = gst_rtsp_stream_get_srcpad (stream);
+ if (!src_pad)
+ return;
+
+ gst_pad_sticky_events_foreach (src_pad, get_info_from_tags, stream_media);
+
+ gst_object_unref (src_pad);
+}
+
+static guint
+get_roc_from_stats (GstStructure * stats, guint ssrc)
+{
+ const GValue *va, *v;
+ guint i, len;
+ /* initialize roc to something different than 0, so if we don't get
+ the proper ROC from the encoder, streaming should fail initially. */
+ guint roc = -1;
+
+ va = gst_structure_get_value (stats, "streams");
+ if (!va || !G_VALUE_HOLDS (va, GST_TYPE_ARRAY)) {
+ GST_WARNING ("stats doesn't have a valid 'streams' field");
+ return 0;
+ }
+
+ len = gst_value_array_get_size (va);
+
+ /* look if there's any SSRC that matches. */
+ for (i = 0; i < len; i++) {
+ GstStructure *stream;
+ v = gst_value_array_get_value (va, i);
+ if (v && (stream = g_value_get_boxed (v))) {
+ guint stream_ssrc;
+ gst_structure_get_uint (stream, "ssrc", &stream_ssrc);
+ if (stream_ssrc == ssrc) {
+ gst_structure_get_uint (stream, "roc", &roc);
+ break;
+ }
+ }
+ }
+
+ return roc;
+}
+
+static gboolean
+mikey_add_crypto_sessions (GstRTSPStream * stream, GstMIKEYMessage * msg)
+{
+ guint i;
+ GObject *session;
+ GstElement *encoder;
+ GValueArray *sources;
+ gboolean roc_found;
+
+ encoder = gst_rtsp_stream_get_srtp_encoder (stream);
+ if (encoder == NULL) {
+ GST_ERROR ("unable to get SRTP encoder from stream %p", stream);
+ return FALSE;
+ }
+
+ session = gst_rtsp_stream_get_rtpsession (stream);
+ if (session == NULL) {
+ GST_ERROR ("unable to get RTP session from stream %p", stream);
+ gst_object_unref (encoder);
+ return FALSE;
+ }
+
+ roc_found = FALSE;
+ g_object_get (session, "sources", &sources, NULL);
+ for (i = 0; sources && (i < sources->n_values); i++) {
+ GValue *val;
+ GObject *source;
+ guint32 ssrc;
+ gboolean is_sender;
+
+ val = g_value_array_get_nth (sources, i);
+ source = (GObject *) g_value_get_object (val);
+
+ g_object_get (source, "ssrc", &ssrc, "is-sender", &is_sender, NULL);
+
+ if (is_sender) {
+ guint32 roc = -1;
+ GstStructure *stats;
+
+ g_object_get (encoder, "stats", &stats, NULL);
+
+ if (stats) {
+ roc = get_roc_from_stats (stats, ssrc);
+ gst_structure_free (stats);
+ }
+
+ roc_found = ! !(roc != -1);
+ if (!roc_found) {
+ GST_ERROR ("unable to obtain ROC for stream %p with SSRC %u",
+ stream, ssrc);
+ goto cleanup;
+ }
+
+ GST_INFO ("stream %p with SSRC %u has a ROC of %u", stream, ssrc, roc);
+
+ gst_mikey_message_add_cs_srtp (msg, 0, ssrc, roc);
+ }
+ }
+
+cleanup:
+ {
+ g_value_array_free (sources);
+
+ gst_object_unref (encoder);
+ g_object_unref (session);
+ return roc_found;
+ }
+}
+
+/**
+ * gst_rtsp_sdp_make_media:
+ * @sdp: a #GstRTSPMessage
+ * @info: a #GstSDPInfo
+ * @stream: a #GstRTSPStream
+ * @caps: a #GstCaps
+ * @profile: a #GstRTSPProfile
+ *
+ * Creates a #GstSDPMedia from the parameters and stores it in @sdp.
+ *
+ * Returns: %TRUE on success
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_sdp_make_media (GstSDPMessage * sdp, GstSDPInfo * info,
+ GstRTSPStream * stream, GstCaps * caps, GstRTSPProfile profile)
+{
+ GstSDPMedia *smedia;
+ gchar *tmp;
+ GstRTSPLowerTrans ltrans;
+ GSocketFamily family;
+ const gchar *addrtype, *proto;
+ gchar *address;
+ guint ttl;
+ GstClockTime rtx_time;
+ gchar *base64;
+ GstMIKEYMessage *mikey_msg;
+
+ gst_sdp_media_new (&smedia);
+
+ if (gst_sdp_media_set_media_from_caps (caps, smedia) != GST_SDP_OK) {
+ goto caps_error;
+ }
+
+ gst_sdp_media_set_port_info (smedia, 0, 1);
+
+ switch (profile) {
+ case GST_RTSP_PROFILE_AVP:
+ proto = "RTP/AVP";
+ break;
+ case GST_RTSP_PROFILE_AVPF:
+ proto = "RTP/AVPF";
+ break;
+ case GST_RTSP_PROFILE_SAVP:
+ proto = "RTP/SAVP";
+ break;
+ case GST_RTSP_PROFILE_SAVPF:
+ proto = "RTP/SAVPF";
+ break;
+ default:
+ proto = "udp";
+ break;
+ }
+ gst_sdp_media_set_proto (smedia, proto);
+
+ if (info->is_ipv6) {
+ addrtype = "IP6";
+ family = G_SOCKET_FAMILY_IPV6;
+ } else {
+ addrtype = "IP4";
+ family = G_SOCKET_FAMILY_IPV4;
+ }
+
+ ltrans = gst_rtsp_stream_get_protocols (stream);
+ if (ltrans == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ GstRTSPAddress *addr;
+
+ addr = gst_rtsp_stream_get_multicast_address (stream, family);
+ if (addr == NULL)
+ goto no_multicast;
+
+ address = g_strdup (addr->address);
+ ttl = addr->ttl;
+ gst_rtsp_address_free (addr);
+ } else {
+ ttl = 16;
+ if (info->is_ipv6)
+ address = g_strdup ("::");
+ else
+ address = g_strdup ("0.0.0.0");
+ }
+
+ /* for the c= line */
+ gst_sdp_media_add_connection (smedia, "IN", addrtype, address, ttl, 1);
+ g_free (address);
+
+ /* the config uri */
+ tmp = gst_rtsp_stream_get_control (stream);
+ gst_sdp_media_add_attribute (smedia, "control", tmp);
+ g_free (tmp);
+
+ /* check for srtp */
+ mikey_msg = gst_mikey_message_new_from_caps (caps);
+ if (mikey_msg) {
+ /* add policy '0' for all sending SSRC */
+ if (!mikey_add_crypto_sessions (stream, mikey_msg)) {
+ gst_mikey_message_unref (mikey_msg);
+ goto crypto_sessions_error;
+ }
+
+ base64 = gst_mikey_message_base64_encode (mikey_msg);
+ if (base64) {
+ tmp = g_strdup_printf ("mikey %s", base64);
+ g_free (base64);
+ gst_sdp_media_add_attribute (smedia, "key-mgmt", tmp);
+ g_free (tmp);
+ }
+
+ gst_mikey_message_unref (mikey_msg);
+ }
+
+ /* RFC 7273 clock signalling */
+ if (gst_rtsp_stream_is_sender (stream)) {
+ GstBin *joined_bin = gst_rtsp_stream_get_joined_bin (stream);
+ GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (joined_bin));
+ gchar *ts_refclk = NULL;
+ gchar *mediaclk = NULL;
+ guint rtptime, clock_rate;
+ GstClockTime running_time, base_time, clock_time;
+ GstRTSPPublishClockMode publish_clock_mode =
+ gst_rtsp_stream_get_publish_clock_mode (stream);
+
+ if (!gst_rtsp_stream_get_rtpinfo (stream, &rtptime, NULL, &clock_rate,
+ &running_time))
+ goto clock_signalling_cleanup;
+ base_time = gst_element_get_base_time (GST_ELEMENT_CAST (joined_bin));
+ g_assert (base_time != GST_CLOCK_TIME_NONE);
+ clock_time = running_time + base_time;
+
+ if (publish_clock_mode != GST_RTSP_PUBLISH_CLOCK_MODE_NONE && clock) {
+ if (GST_IS_NTP_CLOCK (clock) || GST_IS_PTP_CLOCK (clock)) {
+ if (publish_clock_mode == GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
+ guint32 mediaclk_offset;
+
+ /* Calculate RTP time at the clock's epoch. That's the direct offset */
+ clock_time =
+ gst_util_uint64_scale (clock_time, clock_rate, GST_SECOND);
+
+ clock_time &= 0xffffffff;
+ mediaclk_offset =
+ G_GUINT64_CONSTANT (0xffffffff) + rtptime - clock_time;
+ mediaclk = g_strdup_printf ("direct=%u", (guint32) mediaclk_offset);
+ }
+
+ if (GST_IS_NTP_CLOCK (clock)) {
+ gchar *ntp_address;
+ guint ntp_port;
+
+ g_object_get (clock, "address", &ntp_address, "port", &ntp_port,
+ NULL);
+
+ if (ntp_port == 123)
+ ts_refclk = g_strdup_printf ("ntp=%s", ntp_address);
+ else
+ ts_refclk = g_strdup_printf ("ntp=%s:%u", ntp_address, ntp_port);
+
+ g_free (ntp_address);
+ } else {
+ guint64 ptp_clock_id;
+ guint ptp_domain;
+
+ g_object_get (clock, "grandmaster-clock-id", &ptp_clock_id, "domain",
+ &ptp_domain, NULL);
+
+ if (ptp_domain != 0)
+ ts_refclk =
+ g_strdup_printf
+ ("ptp=IEEE1588-2008:%02X-%02X-%02X-%02X-%02X-%02X-%02X-%02X:%u",
+ (guint) (ptp_clock_id >> 56) & 0xff,
+ (guint) (ptp_clock_id >> 48) & 0xff,
+ (guint) (ptp_clock_id >> 40) & 0xff,
+ (guint) (ptp_clock_id >> 32) & 0xff,
+ (guint) (ptp_clock_id >> 24) & 0xff,
+ (guint) (ptp_clock_id >> 16) & 0xff,
+ (guint) (ptp_clock_id >> 8) & 0xff,
+ (guint) (ptp_clock_id >> 0) & 0xff, ptp_domain);
+ else
+ ts_refclk =
+ g_strdup_printf
+ ("ptp=IEEE1588-2008:%02X-%02X-%02X-%02X-%02X-%02X-%02X-%02X",
+ (guint) (ptp_clock_id >> 56) & 0xff,
+ (guint) (ptp_clock_id >> 48) & 0xff,
+ (guint) (ptp_clock_id >> 40) & 0xff,
+ (guint) (ptp_clock_id >> 32) & 0xff,
+ (guint) (ptp_clock_id >> 24) & 0xff,
+ (guint) (ptp_clock_id >> 16) & 0xff,
+ (guint) (ptp_clock_id >> 8) & 0xff,
+ (guint) (ptp_clock_id >> 0) & 0xff);
+ }
+ }
+ }
+ clock_signalling_cleanup:
+ if (clock)
+ gst_object_unref (clock);
+
+ if (!ts_refclk)
+ ts_refclk = g_strdup ("local");
+ if (!mediaclk)
+ mediaclk = g_strdup ("sender");
+
+ gst_sdp_media_add_attribute (smedia, "ts-refclk", ts_refclk);
+ gst_sdp_media_add_attribute (smedia, "mediaclk", mediaclk);
+ g_free (ts_refclk);
+ g_free (mediaclk);
+ gst_object_unref (joined_bin);
+ }
+
+ update_sdp_from_tags (stream, smedia);
+
+ if (profile == GST_RTSP_PROFILE_AVPF || profile == GST_RTSP_PROFILE_SAVPF) {
+ if ((rtx_time = gst_rtsp_stream_get_retransmission_time (stream))) {
+ /* ssrc multiplexed retransmit functionality */
+ guint rtx_pt = gst_rtsp_stream_get_retransmission_pt (stream);
+
+ if (rtx_pt == 0) {
+ g_warning ("failed to find an available dynamic payload type. "
+ "Not adding retransmission");
+ } else {
+ gchar *tmp;
+ GstStructure *s;
+ gint caps_pt, caps_rate;
+
+ s = gst_caps_get_structure (caps, 0);
+ if (s == NULL)
+ goto no_caps_info;
+
+ /* get payload type and clock rate */
+ gst_structure_get_int (s, "payload", &caps_pt);
+ gst_structure_get_int (s, "clock-rate", &caps_rate);
+
+ tmp = g_strdup_printf ("%d", rtx_pt);
+ gst_sdp_media_add_format (smedia, tmp);
+ g_free (tmp);
+
+ tmp = g_strdup_printf ("%d rtx/%d", rtx_pt, caps_rate);
+ gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
+ g_free (tmp);
+
+ tmp =
+ g_strdup_printf ("%d apt=%d;rtx-time=%" G_GINT64_FORMAT, rtx_pt,
+ caps_pt, GST_TIME_AS_MSECONDS (rtx_time));
+ gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
+ g_free (tmp);
+ }
+ }
+
+ if (gst_rtsp_stream_get_ulpfec_percentage (stream)) {
+ guint ulpfec_pt = gst_rtsp_stream_get_ulpfec_pt (stream);
+
+ if (ulpfec_pt == 0) {
+ g_warning ("failed to find an available dynamic payload type. "
+ "Not adding ulpfec");
+ } else {
+ gchar *tmp;
+ GstStructure *s;
+ gint caps_pt, caps_rate;
+
+ s = gst_caps_get_structure (caps, 0);
+ if (s == NULL)
+ goto no_caps_info;
+
+ /* get payload type and clock rate */
+ gst_structure_get_int (s, "payload", &caps_pt);
+ gst_structure_get_int (s, "clock-rate", &caps_rate);
+
+ tmp = g_strdup_printf ("%d", ulpfec_pt);
+ gst_sdp_media_add_format (smedia, tmp);
+ g_free (tmp);
+
+ tmp = g_strdup_printf ("%d ulpfec/%d", ulpfec_pt, caps_rate);
+ gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
+ g_free (tmp);
+
+ tmp = g_strdup_printf ("%d apt=%d", ulpfec_pt, caps_pt);
+ gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
+ g_free (tmp);
+ }
+ }
+ }
+
+ gst_sdp_message_add_media (sdp, smedia);
+ gst_sdp_media_free (smedia);
+
+ return TRUE;
+
+ /* ERRORS */
+caps_error:
+ {
+ gst_sdp_media_free (smedia);
+ GST_ERROR ("unable to set media from caps for stream %d",
+ gst_rtsp_stream_get_index (stream));
+ return FALSE;
+ }
+no_multicast:
+ {
+ gst_sdp_media_free (smedia);
+ GST_ERROR ("stream %d has no multicast address",
+ gst_rtsp_stream_get_index (stream));
+ return FALSE;
+ }
+no_caps_info:
+ {
+ gst_sdp_media_free (smedia);
+ GST_ERROR ("caps for stream %d have no structure",
+ gst_rtsp_stream_get_index (stream));
+ return FALSE;
+ }
+crypto_sessions_error:
+ {
+ gst_sdp_media_free (smedia);
+ GST_ERROR ("unable to add MIKEY crypto sessions for stream %d",
+ gst_rtsp_stream_get_index (stream));
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_sdp_from_media:
+ * @sdp: a #GstSDPMessage
+ * @info: (transfer none): a #GstSDPInfo
+ * @media: (transfer none): a #GstRTSPMedia
+ *
+ * Add @media specific info to @sdp. @info is used to configure the connection
+ * information in the SDP.
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
+ GstRTSPMedia * media)
+{
+ guint i, n_streams;
+ gchar *rangestr;
+ gboolean res;
+
+ n_streams = gst_rtsp_media_n_streams (media);
+
+ rangestr = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ if (rangestr == NULL)
+ goto not_prepared;
+
+ gst_sdp_message_add_attribute (sdp, "range", rangestr);
+ g_free (rangestr);
+
+ res = TRUE;
+ for (i = 0; res && (i < n_streams); i++) {
+ GstRTSPStream *stream;
+
+ stream = gst_rtsp_media_get_stream (media, i);
+ res = gst_rtsp_sdp_from_stream (sdp, info, stream);
+ if (!res) {
+ GST_ERROR ("could not get SDP from stream %p", stream);
+ goto sdp_error;
+ }
+ }
+
+ {
+ GstNetTimeProvider *provider;
+
+ if ((provider =
+ gst_rtsp_media_get_time_provider (media, info->server_ip, 0))) {
+ GstClock *clock;
+ gchar *address, *str;
+ gint port;
+
+ g_object_get (provider, "clock", &clock, "address", &address, "port",
+ &port, NULL);
+
+ str = g_strdup_printf ("GstNetTimeProvider %s %s:%d %" G_GUINT64_FORMAT,
+ g_type_name (G_TYPE_FROM_INSTANCE (clock)), address, port,
+ gst_clock_get_time (clock));
+
+ gst_sdp_message_add_attribute (sdp, "x-gst-clock", str);
+ g_free (str);
+ gst_object_unref (clock);
+ g_free (address);
+ gst_object_unref (provider);
+ }
+ }
+
+ return res;
+
+ /* ERRORS */
+not_prepared:
+ {
+ GST_ERROR ("media %p is not prepared", media);
+ return FALSE;
+ }
+sdp_error:
+ {
+ GST_ERROR ("could not get SDP from media %p", media);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_sdp_from_stream:
+ * @sdp: a #GstSDPMessage
+ * @info: (transfer none): a #GstSDPInfo
+ * @stream: (transfer none): a #GstRTSPStream
+ *
+ * Add info from @stream to @sdp.
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_sdp_from_stream (GstSDPMessage * sdp, GstSDPInfo * info,
+ GstRTSPStream * stream)
+{
+ GstCaps *caps;
+ GstRTSPProfile profiles;
+ guint mask;
+ gboolean res;
+
+ caps = gst_rtsp_stream_get_caps (stream);
+
+ if (caps == NULL) {
+ GST_ERROR ("stream %p has no caps", stream);
+ return FALSE;
+ }
+
+ /* make a new media for each profile */
+ profiles = gst_rtsp_stream_get_profiles (stream);
+ mask = 1;
+ res = TRUE;
+ while (res && (profiles >= mask)) {
+ GstRTSPProfile prof = profiles & mask;
+
+ if (prof)
+ res = gst_rtsp_sdp_make_media (sdp, info, stream, caps, prof);
+
+ mask <<= 1;
+ }
+ gst_caps_unref (caps);
+
+ return res;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/sdp/gstsdpmessage.h>
+
+#include "rtsp-media.h"
+
+#ifndef __GST_RTSP_SDP_H__
+#define __GST_RTSP_SDP_H__
+
+G_BEGIN_DECLS
+
+typedef struct {
+ gboolean is_ipv6;
+ const gchar *server_ip;
+} GstSDPInfo;
+
+/* creating SDP */
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_sdp_from_media (GstSDPMessage *sdp, GstSDPInfo *info, GstRTSPMedia * media);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_sdp_from_stream (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+gboolean
+gst_rtsp_sdp_make_media (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPStream * stream, GstCaps * caps, GstRTSPProfile profile);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SDP_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2019 Mathieu Duponchelle <mathieu@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_SERVER_INTERNAL_H__
+#define __GST_RTSP_SERVER_INTERNAL_H__
+
+#include <glib.h>
+
+G_BEGIN_DECLS
+
+#include "rtsp-stream-transport.h"
+
+/* Internal GstRTSPStreamTransport interface */
+
+typedef gboolean (*GstRTSPBackPressureFunc) (guint8 channel, gpointer user_data);
+
+gboolean gst_rtsp_stream_transport_backlog_push (GstRTSPStreamTransport *trans,
+ GstBuffer *buffer,
+ GstBufferList *buffer_list,
+ gboolean is_rtp);
+
+gboolean gst_rtsp_stream_transport_backlog_pop (GstRTSPStreamTransport *trans,
+ GstBuffer **buffer,
+ GstBufferList **buffer_list,
+ gboolean *is_rtp);
+
+gboolean gst_rtsp_stream_transport_backlog_is_empty (GstRTSPStreamTransport *trans);
+
+void gst_rtsp_stream_transport_clear_backlog (GstRTSPStreamTransport * trans);
+
+void gst_rtsp_stream_transport_lock_backlog (GstRTSPStreamTransport * trans);
+
+void gst_rtsp_stream_transport_unlock_backlog (GstRTSPStreamTransport * trans);
+
+void gst_rtsp_stream_transport_set_back_pressure_callback (GstRTSPStreamTransport *trans,
+ GstRTSPBackPressureFunc back_pressure_func,
+ gpointer user_data,
+ GDestroyNotify notify);
+
+gboolean gst_rtsp_stream_transport_check_back_pressure (GstRTSPStreamTransport *trans,
+ gboolean is_rtp);
+
+gboolean gst_rtsp_stream_is_tcp_receiver (GstRTSPStream * stream);
+
+void gst_rtsp_media_set_enable_rtcp (GstRTSPMedia *media, gboolean enable);
+void gst_rtsp_stream_set_enable_rtcp (GstRTSPStream *stream, gboolean enable);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SERVER_INTERNAL_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_SERVER_OBJECT_H__
+#define __GST_RTSP_SERVER_OBJECT_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPServer GstRTSPServer;
+typedef struct _GstRTSPServerClass GstRTSPServerClass;
+typedef struct _GstRTSPServerPrivate GstRTSPServerPrivate;
+
+#include "rtsp-server-prelude.h"
+#include "rtsp-session-pool.h"
+#include "rtsp-session.h"
+#include "rtsp-media.h"
+#include "rtsp-stream.h"
+#include "rtsp-stream-transport.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-client.h"
+#include "rtsp-context.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-media-factory.h"
+#include "rtsp-permissions.h"
+#include "rtsp-auth.h"
+#include "rtsp-token.h"
+#include "rtsp-session-media.h"
+#include "rtsp-sdp.h"
+#include "rtsp-media-factory-uri.h"
+#include "rtsp-params.h"
+
+#define GST_TYPE_RTSP_SERVER (gst_rtsp_server_get_type ())
+#define GST_IS_RTSP_SERVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SERVER))
+#define GST_IS_RTSP_SERVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SERVER))
+#define GST_RTSP_SERVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SERVER, GstRTSPServerClass))
+#define GST_RTSP_SERVER(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SERVER, GstRTSPServer))
+#define GST_RTSP_SERVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SERVER, GstRTSPServerClass))
+#define GST_RTSP_SERVER_CAST(obj) ((GstRTSPServer*)(obj))
+#define GST_RTSP_SERVER_CLASS_CAST(klass) ((GstRTSPServerClass*)(klass))
+
+/**
+ * GstRTSPServer:
+ *
+ * This object listens on a port, creates and manages the clients connected to
+ * it.
+ */
+struct _GstRTSPServer {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPServerPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPServerClass:
+ * @create_client: Create, configure a new GstRTSPClient
+ * object that handles the new connection on @socket. The default
+ * implementation will create a GstRTSPClient and will configure the
+ * mount-points, auth, session-pool and thread-pool on the client.
+ * @client_connected: emitted when a new client connected.
+ *
+ * The RTSP server class structure
+ */
+struct _GstRTSPServerClass {
+ GObjectClass parent_class;
+
+ GstRTSPClient * (*create_client) (GstRTSPServer *server);
+
+ /* signals */
+ void (*client_connected) (GstRTSPServer *server, GstRTSPClient *client);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_server_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPServer * gst_rtsp_server_new (void);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_server_set_address (GstRTSPServer *server, const gchar *address);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_server_get_address (GstRTSPServer *server);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_server_set_service (GstRTSPServer *server, const gchar *service);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_server_get_service (GstRTSPServer *server);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_server_set_backlog (GstRTSPServer *server, gint backlog);
+
+GST_RTSP_SERVER_API
+gint gst_rtsp_server_get_backlog (GstRTSPServer *server);
+
+GST_RTSP_SERVER_API
+int gst_rtsp_server_get_bound_port (GstRTSPServer *server);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_server_set_session_pool (GstRTSPServer *server, GstRTSPSessionPool *pool);
+
+GST_RTSP_SERVER_API
+GstRTSPSessionPool * gst_rtsp_server_get_session_pool (GstRTSPServer *server);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_server_set_mount_points (GstRTSPServer *server, GstRTSPMountPoints *mounts);
+
+GST_RTSP_SERVER_API
+GstRTSPMountPoints * gst_rtsp_server_get_mount_points (GstRTSPServer *server);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_server_set_content_length_limit (GstRTSPServer * server, guint limit);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_server_get_content_length_limit (GstRTSPServer * server);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_server_set_auth (GstRTSPServer *server, GstRTSPAuth *auth);
+
+GST_RTSP_SERVER_API
+GstRTSPAuth * gst_rtsp_server_get_auth (GstRTSPServer *server);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_server_set_thread_pool (GstRTSPServer *server, GstRTSPThreadPool *pool);
+
+GST_RTSP_SERVER_API
+GstRTSPThreadPool * gst_rtsp_server_get_thread_pool (GstRTSPServer *server);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_server_transfer_connection (GstRTSPServer * server, GSocket *socket,
+ const gchar * ip, gint port,
+ const gchar *initial_buffer);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_server_io_func (GSocket *socket, GIOCondition condition,
+ GstRTSPServer *server);
+
+GST_RTSP_SERVER_API
+GSocket * gst_rtsp_server_create_socket (GstRTSPServer *server,
+ GCancellable *cancellable,
+ GError **error);
+
+GST_RTSP_SERVER_API
+GSource * gst_rtsp_server_create_source (GstRTSPServer *server,
+ GCancellable * cancellable,
+ GError **error);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_server_attach (GstRTSPServer *server,
+ GMainContext *context);
+
+/**
+ * GstRTSPServerClientFilterFunc:
+ * @server: a #GstRTSPServer object
+ * @client: a #GstRTSPClient in @server
+ * @user_data: user data that has been given to gst_rtsp_server_client_filter()
+ *
+ * This function will be called by the gst_rtsp_server_client_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @client will be removed
+ * from @server.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @client untouched in
+ * @server.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @client to the result #GList of
+ * gst_rtsp_server_client_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPServerClientFilterFunc) (GstRTSPServer *server,
+ GstRTSPClient *client,
+ gpointer user_data);
+
+GST_RTSP_SERVER_API
+GList * gst_rtsp_server_client_filter (GstRTSPServer *server,
+ GstRTSPServerClientFilterFunc func,
+ gpointer user_data);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPServer, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SERVER_OBJECT_H__ */
--- /dev/null
+/* GStreamer RtspServer Library
+ * Copyright (C) 2018 GStreamer developers
+ *
+ * rtspserver-prelude.h: prelude include header for gst-rtspserver library
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_SERVER_PRELUDE_H__
+#define __GST_RTSP_SERVER_PRELUDE_H__
+
+#include <gst/gst.h>
+
+#ifndef GST_RTSP_SERVER_API
+# ifdef BUILDING_GST_RTSP_SERVER
+# define GST_RTSP_SERVER_API GST_API_EXPORT /* from config.h */
+# else
+# define GST_RTSP_SERVER_API GST_API_IMPORT
+# endif
+#endif
+
+/* Do *not* use these defines outside of rtsp-server. Use G_DEPRECATED instead. */
+#ifdef GST_DISABLE_DEPRECATED
+#define GST_RTSP_SERVER_DEPRECATED GST_RTSP_SERVER_API
+#define GST_RTSP_SERVER_DEPRECATED_FOR(f) GST_RTSP_SERVER_API
+#else
+#define GST_RTSP_SERVER_DEPRECATED G_DEPRECATED GST_RTSP_SERVER_API
+#define GST_RTSP_SERVER_DEPRECATED_FOR(f) G_DEPRECATED_FOR(f) GST_RTSP_SERVER_API
+#endif
+
+#endif /* __GST_RTSP_SERVER_PRELUDE_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-server
+ * @short_description: The main server object
+ * @see_also: #GstRTSPClient, #GstRTSPThreadPool
+ *
+ * The server object is the object listening for connections on a port and
+ * creating #GstRTSPClient objects to handle those connections.
+ *
+ * The server will listen on the address set with gst_rtsp_server_set_address()
+ * and the port or service configured with gst_rtsp_server_set_service().
+ * Use gst_rtsp_server_set_backlog() to configure the amount of pending requests
+ * that the server will keep. By default the server listens on the current
+ * network (0.0.0.0) and port 8554.
+ *
+ * The server will require an SSL connection when a TLS certificate has been
+ * set in the auth object with gst_rtsp_auth_set_tls_certificate().
+ *
+ * To start the server, use gst_rtsp_server_attach() to attach it to a
+ * #GMainContext. For more control, gst_rtsp_server_create_source() and
+ * gst_rtsp_server_create_socket() can be used to get a #GSource and #GSocket
+ * respectively.
+ *
+ * gst_rtsp_server_transfer_connection() can be used to transfer an existing
+ * socket to the RTSP server, for example from an HTTP server.
+ *
+ * Once the server socket is attached to a mainloop, it will start accepting
+ * connections. When a new connection is received, a new #GstRTSPClient object
+ * is created to handle the connection. The new client will be configured with
+ * the server #GstRTSPAuth, #GstRTSPMountPoints, #GstRTSPSessionPool and
+ * #GstRTSPThreadPool.
+ *
+ * The server uses the configured #GstRTSPThreadPool object to handle the
+ * remainder of the communication with this client.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "rtsp-context.h"
+#include "rtsp-server-object.h"
+#include "rtsp-client.h"
+
+#define GST_RTSP_SERVER_GET_LOCK(server) (&(GST_RTSP_SERVER_CAST(server)->priv->lock))
+#define GST_RTSP_SERVER_LOCK(server) (g_mutex_lock(GST_RTSP_SERVER_GET_LOCK(server)))
+#define GST_RTSP_SERVER_UNLOCK(server) (g_mutex_unlock(GST_RTSP_SERVER_GET_LOCK(server)))
+
+struct _GstRTSPServerPrivate
+{
+ GMutex lock; /* protects everything in this struct */
+
+ /* server information */
+ gchar *address;
+ gchar *service;
+ gint backlog;
+
+ GSocket *socket;
+
+ /* sessions on this server */
+ GstRTSPSessionPool *session_pool;
+
+ /* mount points for this server */
+ GstRTSPMountPoints *mount_points;
+
+ /* request size limit */
+ guint content_length_limit;
+
+ /* authentication manager */
+ GstRTSPAuth *auth;
+
+ /* resource manager */
+ GstRTSPThreadPool *thread_pool;
+
+ /* the clients that are connected */
+ GList *clients;
+ guint clients_cookie;
+};
+
+#define DEFAULT_ADDRESS "0.0.0.0"
+#define DEFAULT_BOUND_PORT -1
+/* #define DEFAULT_ADDRESS "::0" */
+#define DEFAULT_SERVICE "8554"
+#define DEFAULT_BACKLOG 5
+
+/* Define to use the SO_LINGER option so that the server sockets can be resused
+ * sooner. Disabled for now because it is not very well implemented by various
+ * OSes and it causes clients to fail to read the TEARDOWN response. */
+#undef USE_SOLINGER
+
+enum
+{
+ PROP_0,
+ PROP_ADDRESS,
+ PROP_SERVICE,
+ PROP_BOUND_PORT,
+ PROP_BACKLOG,
+
+ PROP_SESSION_POOL,
+ PROP_MOUNT_POINTS,
+ PROP_CONTENT_LENGTH_LIMIT,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_CLIENT_CONNECTED,
+ SIGNAL_LAST
+};
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_server_debug);
+#define GST_CAT_DEFAULT rtsp_server_debug
+
+typedef struct _ClientContext ClientContext;
+
+static guint gst_rtsp_server_signals[SIGNAL_LAST] = { 0 };
+
+static void gst_rtsp_server_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_server_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_server_finalize (GObject * object);
+
+static GstRTSPClient *default_create_client (GstRTSPServer * server);
+
+static void
+gst_rtsp_server_class_init (GstRTSPServerClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_server_get_property;
+ gobject_class->set_property = gst_rtsp_server_set_property;
+ gobject_class->finalize = gst_rtsp_server_finalize;
+
+ /**
+ * GstRTSPServer::address:
+ *
+ * The address of the server. This is the address where the server will
+ * listen on.
+ */
+ g_object_class_install_property (gobject_class, PROP_ADDRESS,
+ g_param_spec_string ("address", "Address",
+ "The address the server uses to listen on", DEFAULT_ADDRESS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPServer::service:
+ *
+ * The service of the server. This is either a string with the service name or
+ * a port number (as a string) the server will listen on.
+ */
+ g_object_class_install_property (gobject_class, PROP_SERVICE,
+ g_param_spec_string ("service", "Service",
+ "The service or port number the server uses to listen on",
+ DEFAULT_SERVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPServer::bound-port:
+ *
+ * The actual port the server is listening on. Can be used to retrieve the
+ * port number when the server is started on port 0, which means bind to a
+ * random port. Set to -1 if the server has not been bound yet.
+ */
+ g_object_class_install_property (gobject_class, PROP_BOUND_PORT,
+ g_param_spec_int ("bound-port", "Bound port",
+ "The port number the server is listening on",
+ -1, G_MAXUINT16, DEFAULT_BOUND_PORT,
+ G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPServer::backlog:
+ *
+ * The backlog argument defines the maximum length to which the queue of
+ * pending connections for the server may grow. If a connection request arrives
+ * when the queue is full, the client may receive an error with an indication of
+ * ECONNREFUSED or, if the underlying protocol supports retransmission, the
+ * request may be ignored so that a later reattempt at connection succeeds.
+ */
+ g_object_class_install_property (gobject_class, PROP_BACKLOG,
+ g_param_spec_int ("backlog", "Backlog",
+ "The maximum length to which the queue "
+ "of pending connections may grow", 0, G_MAXINT, DEFAULT_BACKLOG,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPServer::session-pool:
+ *
+ * The session pool of the server. By default each server has a separate
+ * session pool but sessions can be shared between servers by setting the same
+ * session pool on multiple servers.
+ */
+ g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
+ g_param_spec_object ("session-pool", "Session Pool",
+ "The session pool to use for client session",
+ GST_TYPE_RTSP_SESSION_POOL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPServer::mount-points:
+ *
+ * The mount points to use for this server. By default the server has no
+ * mount points and thus cannot map urls to media streams.
+ */
+ g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
+ g_param_spec_object ("mount-points", "Mount Points",
+ "The mount points to use for client session",
+ GST_TYPE_RTSP_MOUNT_POINTS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * RTSPServer::content-length-limit:
+ *
+ * Define an appropriate request size limit and reject requests exceeding the
+ * limit.
+ *
+ * Since: 1.18
+ */
+ g_object_class_install_property (gobject_class, PROP_CONTENT_LENGTH_LIMIT,
+ g_param_spec_uint ("content-length-limit", "Limitation of Content-Length",
+ "Limitation of Content-Length",
+ 0, G_MAXUINT, G_MAXUINT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED] =
+ g_signal_new ("client-connected", G_TYPE_FROM_CLASS (gobject_class),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPServerClass, client_connected),
+ NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CLIENT);
+
+ klass->create_client = default_create_client;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
+}
+
+static void
+gst_rtsp_server_init (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv = gst_rtsp_server_get_instance_private (server);
+
+ server->priv = priv;
+
+ g_mutex_init (&priv->lock);
+ priv->address = g_strdup (DEFAULT_ADDRESS);
+ priv->service = g_strdup (DEFAULT_SERVICE);
+ priv->socket = NULL;
+ priv->backlog = DEFAULT_BACKLOG;
+ priv->session_pool = gst_rtsp_session_pool_new ();
+ priv->mount_points = gst_rtsp_mount_points_new ();
+ priv->content_length_limit = G_MAXUINT;
+ priv->thread_pool = gst_rtsp_thread_pool_new ();
+}
+
+static void
+gst_rtsp_server_finalize (GObject * object)
+{
+ GstRTSPServer *server = GST_RTSP_SERVER (object);
+ GstRTSPServerPrivate *priv = server->priv;
+
+ GST_DEBUG_OBJECT (server, "finalize server");
+
+ g_free (priv->address);
+ g_free (priv->service);
+
+ if (priv->socket)
+ g_object_unref (priv->socket);
+
+ if (priv->session_pool)
+ g_object_unref (priv->session_pool);
+ if (priv->mount_points)
+ g_object_unref (priv->mount_points);
+ if (priv->thread_pool)
+ g_object_unref (priv->thread_pool);
+
+ if (priv->auth)
+ g_object_unref (priv->auth);
+
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_server_parent_class)->finalize (object);
+}
+
+/**
+ * gst_rtsp_server_new:
+ *
+ * Create a new #GstRTSPServer instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPServer
+ */
+GstRTSPServer *
+gst_rtsp_server_new (void)
+{
+ GstRTSPServer *result;
+
+ result = g_object_new (GST_TYPE_RTSP_SERVER, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_address:
+ * @server: a #GstRTSPServer
+ * @address: the address
+ *
+ * Configure @server to accept connections on the given address.
+ *
+ * This function must be called before the server is bound.
+ */
+void
+gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
+{
+ GstRTSPServerPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+ g_return_if_fail (address != NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ g_free (priv->address);
+ priv->address = g_strdup (address);
+ GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_address:
+ * @server: a #GstRTSPServer
+ *
+ * Get the address on which the server will accept connections.
+ *
+ * Returns: (transfer full) (nullable): the server address. g_free() after usage.
+ */
+gchar *
+gst_rtsp_server_get_address (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ result = g_strdup (priv->address);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_get_bound_port:
+ * @server: a #GstRTSPServer
+ *
+ * Get the port number where the server was bound to.
+ *
+ * Returns: the port number
+ */
+int
+gst_rtsp_server_get_bound_port (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ GSocketAddress *address;
+ int result = -1;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), result);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ if (priv->socket == NULL)
+ goto out;
+
+ address = g_socket_get_local_address (priv->socket, NULL);
+ result = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (address));
+ g_object_unref (address);
+
+out:
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_service:
+ * @server: a #GstRTSPServer
+ * @service: the service
+ *
+ * Configure @server to accept connections on the given service.
+ * @service should be a string containing the service name (see services(5)) or
+ * a string containing a port number between 1 and 65535.
+ *
+ * When @service is set to "0", the server will listen on a random free
+ * port. The actual used port can be retrieved with
+ * gst_rtsp_server_get_bound_port().
+ *
+ * This function must be called before the server is bound.
+ */
+void
+gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
+{
+ GstRTSPServerPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+ g_return_if_fail (service != NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ g_free (priv->service);
+ priv->service = g_strdup (service);
+ GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_service:
+ * @server: a #GstRTSPServer
+ *
+ * Get the service on which the server will accept connections.
+ *
+ * Returns: (transfer full) (nullable): the service. use g_free() after usage.
+ */
+gchar *
+gst_rtsp_server_get_service (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ result = g_strdup (priv->service);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_backlog:
+ * @server: a #GstRTSPServer
+ * @backlog: the backlog
+ *
+ * configure the maximum amount of requests that may be queued for the
+ * server.
+ *
+ * This function must be called before the server is bound.
+ */
+void
+gst_rtsp_server_set_backlog (GstRTSPServer * server, gint backlog)
+{
+ GstRTSPServerPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ priv->backlog = backlog;
+ GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_backlog:
+ * @server: a #GstRTSPServer
+ *
+ * The maximum amount of queued requests for the server.
+ *
+ * Returns: the server backlog.
+ */
+gint
+gst_rtsp_server_get_backlog (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ gint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ result = priv->backlog;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_session_pool:
+ * @server: a #GstRTSPServer
+ * @pool: (transfer none) (nullable): a #GstRTSPSessionPool
+ *
+ * configure @pool to be used as the session pool of @server.
+ */
+void
+gst_rtsp_server_set_session_pool (GstRTSPServer * server,
+ GstRTSPSessionPool * pool)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPSessionPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ if (pool)
+ g_object_ref (pool);
+
+ GST_RTSP_SERVER_LOCK (server);
+ old = priv->session_pool;
+ priv->session_pool = pool;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_server_get_session_pool:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPSessionPool used as the session pool of @server.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPSessionPool used for sessions. g_object_unref() after
+ * usage.
+ */
+GstRTSPSessionPool *
+gst_rtsp_server_get_session_pool (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPSessionPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ if ((result = priv->session_pool))
+ g_object_ref (result);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_mount_points:
+ * @server: a #GstRTSPServer
+ * @mounts: (transfer none) (nullable): a #GstRTSPMountPoints
+ *
+ * configure @mounts to be used as the mount points of @server.
+ */
+void
+gst_rtsp_server_set_mount_points (GstRTSPServer * server,
+ GstRTSPMountPoints * mounts)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPMountPoints *old;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ if (mounts)
+ g_object_ref (mounts);
+
+ GST_RTSP_SERVER_LOCK (server);
+ old = priv->mount_points;
+ priv->mount_points = mounts;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (old)
+ g_object_unref (old);
+}
+
+
+/**
+ * gst_rtsp_server_get_mount_points:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPMountPoints used as the mount points of @server.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPMountPoints of @server. g_object_unref() after
+ * usage.
+ */
+GstRTSPMountPoints *
+gst_rtsp_server_get_mount_points (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPMountPoints *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ if ((result = priv->mount_points))
+ g_object_ref (result);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_content_length_limit
+ * @server: a #GstRTSPServer
+ * Configure @server to use the specified Content-Length limit.
+ *
+ * Define an appropriate request size limit and reject requests exceeding the
+ * limit.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_server_set_content_length_limit (GstRTSPServer * server, guint limit)
+{
+ GstRTSPServerPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ priv->content_length_limit = limit;
+ GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_content_length_limit:
+ * @server: a #GstRTSPServer
+ *
+ * Get the Content-Length limit of @server.
+ *
+ * Returns: the Content-Length limit.
+ *
+ * Since: 1.18
+ */
+guint
+gst_rtsp_server_get_content_length_limit (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ guint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), G_MAXUINT);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ result = priv->content_length_limit;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_auth:
+ * @server: a #GstRTSPServer
+ * @auth: (transfer none) (nullable): a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @server.
+ */
+void
+gst_rtsp_server_set_auth (GstRTSPServer * server, GstRTSPAuth * auth)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPAuth *old;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ if (auth)
+ g_object_ref (auth);
+
+ GST_RTSP_SERVER_LOCK (server);
+ old = priv->auth;
+ priv->auth = auth;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (old)
+ g_object_unref (old);
+}
+
+
+/**
+ * gst_rtsp_server_get_auth:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @server.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPAuth of @server. g_object_unref() after
+ * usage.
+ */
+GstRTSPAuth *
+gst_rtsp_server_get_auth (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPAuth *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ if ((result = priv->auth))
+ g_object_ref (result);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_thread_pool:
+ * @server: a #GstRTSPServer
+ * @pool: (transfer none) (nullable): a #GstRTSPThreadPool
+ *
+ * configure @pool to be used as the thread pool of @server.
+ */
+void
+gst_rtsp_server_set_thread_pool (GstRTSPServer * server,
+ GstRTSPThreadPool * pool)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPThreadPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ if (pool)
+ g_object_ref (pool);
+
+ GST_RTSP_SERVER_LOCK (server);
+ old = priv->thread_pool;
+ priv->thread_pool = pool;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_server_get_thread_pool:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPThreadPool used as the thread pool of @server.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPThreadPool of @server. g_object_unref() after
+ * usage.
+ */
+GstRTSPThreadPool *
+gst_rtsp_server_get_thread_pool (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPThreadPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ if ((result = priv->thread_pool))
+ g_object_ref (result);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+static void
+gst_rtsp_server_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPServer *server = GST_RTSP_SERVER (object);
+
+ switch (propid) {
+ case PROP_ADDRESS:
+ g_value_take_string (value, gst_rtsp_server_get_address (server));
+ break;
+ case PROP_SERVICE:
+ g_value_take_string (value, gst_rtsp_server_get_service (server));
+ break;
+ case PROP_BOUND_PORT:
+ g_value_set_int (value, gst_rtsp_server_get_bound_port (server));
+ break;
+ case PROP_BACKLOG:
+ g_value_set_int (value, gst_rtsp_server_get_backlog (server));
+ break;
+ case PROP_SESSION_POOL:
+ g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
+ break;
+ case PROP_MOUNT_POINTS:
+ g_value_take_object (value, gst_rtsp_server_get_mount_points (server));
+ break;
+ case PROP_CONTENT_LENGTH_LIMIT:
+ g_value_set_uint (value,
+ gst_rtsp_server_get_content_length_limit (server));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_server_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPServer *server = GST_RTSP_SERVER (object);
+
+ switch (propid) {
+ case PROP_ADDRESS:
+ gst_rtsp_server_set_address (server, g_value_get_string (value));
+ break;
+ case PROP_SERVICE:
+ gst_rtsp_server_set_service (server, g_value_get_string (value));
+ break;
+ case PROP_BACKLOG:
+ gst_rtsp_server_set_backlog (server, g_value_get_int (value));
+ break;
+ case PROP_SESSION_POOL:
+ gst_rtsp_server_set_session_pool (server, g_value_get_object (value));
+ break;
+ case PROP_MOUNT_POINTS:
+ gst_rtsp_server_set_mount_points (server, g_value_get_object (value));
+ break;
+ case PROP_CONTENT_LENGTH_LIMIT:
+ gst_rtsp_server_set_content_length_limit (server,
+ g_value_get_uint (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_server_create_socket:
+ * @server: a #GstRTSPServer
+ * @cancellable: (allow-none): a #GCancellable
+ * @error: (out): a #GError
+ *
+ * Create a #GSocket for @server. The socket will listen on the
+ * configured service.
+ *
+ * Returns: (transfer full): the #GSocket for @server or %NULL when an error
+ * occurred.
+ */
+GSocket *
+gst_rtsp_server_create_socket (GstRTSPServer * server,
+ GCancellable * cancellable, GError ** error)
+{
+ GstRTSPServerPrivate *priv;
+ GSocketConnectable *conn;
+ GSocketAddressEnumerator *enumerator;
+ GSocket *socket = NULL;
+#ifdef USE_SOLINGER
+ struct linger linger;
+#endif
+ GError *sock_error = NULL;
+ GError *bind_error = NULL;
+ guint16 port;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ GST_DEBUG_OBJECT (server, "getting address info of %s/%s", priv->address,
+ priv->service);
+
+ /* resolve the server IP address */
+ port = atoi (priv->service);
+ if (port != 0 || !strcmp (priv->service, "0"))
+ conn = g_network_address_new (priv->address, port);
+ else
+ conn = g_network_service_new (priv->service, "tcp", priv->address);
+
+ enumerator = g_socket_connectable_enumerate (conn);
+ g_object_unref (conn);
+
+ /* create server socket, we loop through all the addresses until we manage to
+ * create a socket and bind. */
+ while (TRUE) {
+ GSocketAddress *sockaddr;
+
+ sockaddr =
+ g_socket_address_enumerator_next (enumerator, cancellable, error);
+ if (!sockaddr) {
+ if (!*error)
+ GST_DEBUG_OBJECT (server, "no more addresses %s",
+ *error ? (*error)->message : "");
+ else
+ GST_DEBUG_OBJECT (server, "failed to retrieve next address %s",
+ (*error)->message);
+ break;
+ }
+
+ /* only keep the first error */
+ socket = g_socket_new (g_socket_address_get_family (sockaddr),
+ G_SOCKET_TYPE_STREAM, G_SOCKET_PROTOCOL_TCP,
+ sock_error ? NULL : &sock_error);
+
+ if (socket == NULL) {
+ GST_DEBUG_OBJECT (server, "failed to make socket (%s), try next",
+ sock_error->message);
+ g_object_unref (sockaddr);
+ continue;
+ }
+
+ if (g_socket_bind (socket, sockaddr, TRUE, bind_error ? NULL : &bind_error)) {
+ /* ask what port the socket has been bound to */
+ if (port == 0 || !strcmp (priv->service, "0")) {
+ GError *addr_error = NULL;
+
+ g_object_unref (sockaddr);
+ sockaddr = g_socket_get_local_address (socket, &addr_error);
+
+ if (addr_error != NULL) {
+ GST_DEBUG_OBJECT (server,
+ "failed to get the local address of a bound socket %s",
+ addr_error->message);
+ g_clear_error (&addr_error);
+ break;
+ }
+ port =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
+
+ if (port != 0) {
+ g_free (priv->service);
+ priv->service = g_strdup_printf ("%d", port);
+ } else {
+ GST_DEBUG_OBJECT (server, "failed to get the port of a bound socket");
+ }
+ }
+ g_object_unref (sockaddr);
+ break;
+ }
+
+ GST_DEBUG_OBJECT (server, "failed to bind socket (%s), try next",
+ bind_error->message);
+ g_object_unref (sockaddr);
+ g_object_unref (socket);
+ socket = NULL;
+ }
+ g_object_unref (enumerator);
+
+ if (socket == NULL)
+ goto no_socket;
+
+ g_clear_error (&sock_error);
+ g_clear_error (&bind_error);
+
+ GST_DEBUG_OBJECT (server, "opened sending server socket");
+
+ /* keep connection alive; avoids SIGPIPE during write */
+ g_socket_set_keepalive (socket, TRUE);
+
+#if 0
+#ifdef USE_SOLINGER
+ /* make sure socket is reset 5 seconds after close. This ensure that we can
+ * reuse the socket quickly while still having a chance to send data to the
+ * client. */
+ linger.l_onoff = 1;
+ linger.l_linger = 5;
+ if (setsockopt (sockfd, SOL_SOCKET, SO_LINGER,
+ (void *) &linger, sizeof (linger)) < 0)
+ goto linger_failed;
+#endif
+#endif
+
+ /* set the server socket to nonblocking */
+ g_socket_set_blocking (socket, FALSE);
+
+ /* set listen backlog */
+ g_socket_set_listen_backlog (socket, priv->backlog);
+
+ if (!g_socket_listen (socket, error))
+ goto listen_failed;
+
+ GST_DEBUG_OBJECT (server, "listening on server socket %p with queue of %d",
+ socket, priv->backlog);
+
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return socket;
+
+ /* ERRORS */
+no_socket:
+ {
+ GST_ERROR_OBJECT (server, "failed to create socket");
+ goto close_error;
+ }
+#if 0
+#ifdef USE_SOLINGER
+linger_failed:
+ {
+ GST_ERROR_OBJECT (server, "failed to no linger socket: %s",
+ g_strerror (errno));
+ goto close_error;
+ }
+#endif
+#endif
+listen_failed:
+ {
+ GST_ERROR_OBJECT (server, "failed to listen on socket: %s",
+ (*error)->message);
+ goto close_error;
+ }
+close_error:
+ {
+ if (socket)
+ g_object_unref (socket);
+
+ if (sock_error) {
+ if (error == NULL)
+ g_propagate_error (error, sock_error);
+ else
+ g_error_free (sock_error);
+ }
+ if (bind_error) {
+ if ((error == NULL) || (*error == NULL))
+ g_propagate_error (error, bind_error);
+ else
+ g_error_free (bind_error);
+ }
+ GST_RTSP_SERVER_UNLOCK (server);
+ return NULL;
+ }
+}
+
+struct _ClientContext
+{
+ GstRTSPServer *server;
+ GstRTSPThread *thread;
+ GstRTSPClient *client;
+};
+
+static gboolean
+free_client_context (ClientContext * ctx)
+{
+ GST_DEBUG ("free context %p", ctx);
+
+ GST_RTSP_SERVER_LOCK (ctx->server);
+ if (ctx->thread)
+ gst_rtsp_thread_stop (ctx->thread);
+ GST_RTSP_SERVER_UNLOCK (ctx->server);
+
+ g_object_unref (ctx->client);
+ g_object_unref (ctx->server);
+ g_slice_free (ClientContext, ctx);
+
+ return G_SOURCE_REMOVE;
+}
+
+static void
+unmanage_client (GstRTSPClient * client, ClientContext * ctx)
+{
+ GstRTSPServer *server = ctx->server;
+ GstRTSPServerPrivate *priv = server->priv;
+
+ GST_DEBUG_OBJECT (server, "unmanage client %p", client);
+
+ GST_RTSP_SERVER_LOCK (server);
+ priv->clients = g_list_remove (priv->clients, ctx);
+ priv->clients_cookie++;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (ctx->thread) {
+ GSource *src;
+
+ src = g_idle_source_new ();
+ g_source_set_callback (src, (GSourceFunc) free_client_context, ctx, NULL);
+ g_source_attach (src, ctx->thread->context);
+ g_source_unref (src);
+ } else {
+ free_client_context (ctx);
+ }
+}
+
+/* add the client context to the active list of clients, takes ownership
+ * of client */
+static void
+manage_client (GstRTSPServer * server, GstRTSPClient * client)
+{
+ ClientContext *cctx;
+ GstRTSPServerPrivate *priv = server->priv;
+ GMainContext *mainctx = NULL;
+ GstRTSPContext ctx = { NULL };
+
+ GST_DEBUG_OBJECT (server, "manage client %p", client);
+
+ g_signal_emit (server, gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED], 0,
+ client);
+
+ cctx = g_slice_new0 (ClientContext);
+ cctx->server = g_object_ref (server);
+ cctx->client = client;
+
+ GST_RTSP_SERVER_LOCK (server);
+
+ ctx.server = server;
+ ctx.client = client;
+
+ cctx->thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+ GST_RTSP_THREAD_TYPE_CLIENT, &ctx);
+ if (cctx->thread)
+ mainctx = cctx->thread->context;
+ else {
+ GSource *source;
+ /* find the context to add the watch */
+ if ((source = g_main_current_source ()))
+ mainctx = g_source_get_context (source);
+ }
+
+ g_signal_connect (client, "closed", (GCallback) unmanage_client, cctx);
+ priv->clients = g_list_prepend (priv->clients, cctx);
+ priv->clients_cookie++;
+
+ gst_rtsp_client_attach (client, mainctx);
+
+ GST_RTSP_SERVER_UNLOCK (server);
+}
+
+static GstRTSPClient *
+default_create_client (GstRTSPServer * server)
+{
+ GstRTSPClient *client;
+ GstRTSPServerPrivate *priv = server->priv;
+
+ /* a new client connected, create a session to handle the client. */
+ client = gst_rtsp_client_new ();
+
+ /* set the session pool that this client should use */
+ GST_RTSP_SERVER_LOCK (server);
+ gst_rtsp_client_set_session_pool (client, priv->session_pool);
+ /* set the mount points that this client should use */
+ gst_rtsp_client_set_mount_points (client, priv->mount_points);
+ /* Set content-length limit */
+ gst_rtsp_client_set_content_length_limit (GST_RTSP_CLIENT (client),
+ priv->content_length_limit);
+ /* set authentication manager */
+ gst_rtsp_client_set_auth (client, priv->auth);
+ /* set threadpool */
+ gst_rtsp_client_set_thread_pool (client, priv->thread_pool);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return client;
+}
+
+/**
+ * gst_rtsp_server_transfer_connection:
+ * @server: a #GstRTSPServer
+ * @socket: (transfer full): a network socket
+ * @ip: the IP address of the remote client
+ * @port: the port used by the other end
+ * @initial_buffer: (nullable): any initial data that was already read from the socket
+ *
+ * Take an existing network socket and use it for an RTSP connection. This
+ * is used when transferring a socket from an HTTP server which should be used
+ * as an RTSP over HTTP tunnel. The @initial_buffer contains any remaining data
+ * that the HTTP server read from the socket while parsing the HTTP header.
+ *
+ * Returns: TRUE if all was ok, FALSE if an error occurred.
+ */
+gboolean
+gst_rtsp_server_transfer_connection (GstRTSPServer * server, GSocket * socket,
+ const gchar * ip, gint port, const gchar * initial_buffer)
+{
+ GstRTSPClient *client = NULL;
+ GstRTSPServerClass *klass;
+ GstRTSPConnection *conn;
+ GstRTSPResult res;
+
+ klass = GST_RTSP_SERVER_GET_CLASS (server);
+
+ if (klass->create_client)
+ client = klass->create_client (server);
+ if (client == NULL)
+ goto client_failed;
+
+ GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
+ initial_buffer, &conn), no_connection);
+ g_object_unref (socket);
+
+ /* set connection on the client now */
+ gst_rtsp_client_set_connection (client, conn);
+
+ /* manage the client connection */
+ manage_client (server, client);
+
+ return TRUE;
+
+ /* ERRORS */
+client_failed:
+ {
+ GST_ERROR_OBJECT (server, "failed to create a client");
+ g_object_unref (socket);
+ return FALSE;
+ }
+no_connection:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+ GST_ERROR ("could not create connection from socket %p: %s", socket, str);
+ g_free (str);
+ g_object_unref (socket);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_server_io_func:
+ * @socket: a #GSocket
+ * @condition: the condition on @source
+ * @server: (transfer none): a #GstRTSPServer
+ *
+ * A default #GSocketSourceFunc that creates a new #GstRTSPClient to accept and handle a
+ * new connection on @socket or @server.
+ *
+ * Returns: TRUE if the source could be connected, FALSE if an error occurred.
+ */
+gboolean
+gst_rtsp_server_io_func (GSocket * socket, GIOCondition condition,
+ GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv = server->priv;
+ GstRTSPClient *client = NULL;
+ GstRTSPServerClass *klass;
+ GstRTSPResult res;
+ GstRTSPConnection *conn = NULL;
+ GstRTSPContext ctx = { NULL };
+
+ if (condition & G_IO_IN) {
+ /* a new client connected. */
+ GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, NULL),
+ accept_failed);
+
+ ctx.server = server;
+ ctx.conn = conn;
+ ctx.auth = priv->auth;
+ gst_rtsp_context_push_current (&ctx);
+
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_CONNECT))
+ goto connection_refused;
+
+ klass = GST_RTSP_SERVER_GET_CLASS (server);
+ /* a new client connected, create a client object to handle the client. */
+ if (klass->create_client)
+ client = klass->create_client (server);
+ if (client == NULL)
+ goto client_failed;
+
+ /* set connection on the client now */
+ gst_rtsp_client_set_connection (client, conn);
+
+ /* manage the client connection */
+ manage_client (server, client);
+ } else {
+ GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
+ goto exit_no_ctx;
+ }
+exit:
+ gst_rtsp_context_pop_current (&ctx);
+exit_no_ctx:
+
+ return G_SOURCE_CONTINUE;
+
+ /* ERRORS */
+accept_failed:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+ GST_ERROR_OBJECT (server, "Could not accept client on socket %p: %s",
+ socket, str);
+ g_free (str);
+ /* We haven't pushed the context yet, so just return */
+ goto exit_no_ctx;
+ }
+connection_refused:
+ {
+ GST_ERROR_OBJECT (server, "connection refused");
+ gst_rtsp_connection_free (conn);
+ goto exit;
+ }
+client_failed:
+ {
+ GST_ERROR_OBJECT (server, "failed to create a client");
+ gst_rtsp_connection_free (conn);
+ goto exit;
+ }
+}
+
+static void
+watch_destroyed (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv = server->priv;
+
+ GST_DEBUG_OBJECT (server, "source destroyed");
+
+ g_object_unref (priv->socket);
+ priv->socket = NULL;
+ g_object_unref (server);
+}
+
+/**
+ * gst_rtsp_server_create_source:
+ * @server: a #GstRTSPServer
+ * @cancellable: (allow-none): a #GCancellable or %NULL.
+ * @error: (out): a #GError
+ *
+ * Create a #GSource for @server. The new source will have a default
+ * #GSocketSourceFunc of gst_rtsp_server_io_func().
+ *
+ * @cancellable if not %NULL can be used to cancel the source, which will cause
+ * the source to trigger, reporting the current condition (which is likely 0
+ * unless cancellation happened at the same time as a condition change). You can
+ * check for this in the callback using g_cancellable_is_cancelled().
+ *
+ * This takes a reference on @server until @source is destroyed.
+ *
+ * Returns: (transfer full): the #GSource for @server or %NULL when an error
+ * occurred. Free with g_source_unref ()
+ */
+GSource *
+gst_rtsp_server_create_source (GstRTSPServer * server,
+ GCancellable * cancellable, GError ** error)
+{
+ GstRTSPServerPrivate *priv;
+ GSocket *socket, *old;
+ GSource *source;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ socket = gst_rtsp_server_create_socket (server, NULL, error);
+ if (socket == NULL)
+ goto no_socket;
+
+ GST_RTSP_SERVER_LOCK (server);
+ old = priv->socket;
+ priv->socket = g_object_ref (socket);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (old)
+ g_object_unref (old);
+
+ /* create a watch for reads (new connections) and possible errors */
+ source = g_socket_create_source (socket, G_IO_IN |
+ G_IO_ERR | G_IO_HUP | G_IO_NVAL, cancellable);
+ g_object_unref (socket);
+
+ /* configure the callback */
+ g_source_set_callback (source,
+ (GSourceFunc) gst_rtsp_server_io_func, g_object_ref (server),
+ (GDestroyNotify) watch_destroyed);
+
+ return source;
+
+no_socket:
+ {
+ GST_ERROR_OBJECT (server, "failed to create socket");
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_server_attach:
+ * @server: a #GstRTSPServer
+ * @context: (allow-none): a #GMainContext
+ *
+ * Attaches @server to @context. When the mainloop for @context is run, the
+ * server will be dispatched. When @context is %NULL, the default context will be
+ * used).
+ *
+ * This function should be called when the server properties and urls are fully
+ * configured and the server is ready to start.
+ *
+ * This takes a reference on @server until the source is destroyed. Note that
+ * if @context is not the default main context as returned by
+ * g_main_context_default() (or %NULL), g_source_remove() cannot be used to
+ * destroy the source. In that case it is recommended to use
+ * gst_rtsp_server_create_source() and attach it to @context manually.
+ *
+ * Returns: the ID (greater than 0) for the source within the GMainContext.
+ */
+guint
+gst_rtsp_server_attach (GstRTSPServer * server, GMainContext * context)
+{
+ guint res;
+ GSource *source;
+ GError *error = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
+
+ source = gst_rtsp_server_create_source (server, NULL, &error);
+ if (source == NULL)
+ goto no_source;
+
+ res = g_source_attach (source, context);
+ g_source_unref (source);
+
+ return res;
+
+ /* ERRORS */
+no_source:
+ {
+ GST_ERROR_OBJECT (server, "failed to create watch: %s", error->message);
+ g_error_free (error);
+ return 0;
+ }
+}
+
+/**
+ * gst_rtsp_server_client_filter:
+ * @server: a #GstRTSPServer
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: user data passed to @func
+ *
+ * Call @func for each client managed by @server. The result value of @func
+ * determines what happens to the client. @func will be called with @server
+ * locked so no further actions on @server can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the client will be removed from
+ * @server.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the client will remain in @server.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the client will remain in @server but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each client.
+ *
+ * Returns: (element-type GstRTSPClient) (transfer full): a #GList with all
+ * clients for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_server_client_filter (GstRTSPServer * server,
+ GstRTSPServerClientFilterFunc func, gpointer user_data)
+{
+ GstRTSPServerPrivate *priv;
+ GList *result, *walk, *next;
+ GHashTable *visited;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ GST_RTSP_SERVER_LOCK (server);
+restart:
+ cookie = priv->clients_cookie;
+ for (walk = priv->clients; walk; walk = next) {
+ ClientContext *cctx = walk->data;
+ GstRTSPClient *client = cctx->client;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ next = g_list_next (walk);
+
+ if (func) {
+ /* only visit each media once */
+ if (g_hash_table_contains (visited, client))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (client));
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ res = func (server, client, user_data);
+
+ GST_RTSP_SERVER_LOCK (server);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->clients_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ gst_rtsp_client_close (client);
+
+ GST_RTSP_SERVER_LOCK (server);
+ changed |= (cookie != priv->clients_cookie);
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (client));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_SERVER_H__
+#define __GST_RTSP_SERVER_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#include "rtsp-server-prelude.h"
+#include "rtsp-server-object.h"
+#include "rtsp-session-pool.h"
+#include "rtsp-session.h"
+#include "rtsp-media.h"
+#include "rtsp-stream.h"
+#include "rtsp-stream-transport.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-client.h"
+#include "rtsp-context.h"
+#include "rtsp-server.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-media-factory.h"
+#include "rtsp-permissions.h"
+#include "rtsp-auth.h"
+#include "rtsp-token.h"
+#include "rtsp-session-media.h"
+#include "rtsp-sdp.h"
+#include "rtsp-media-factory-uri.h"
+#include "rtsp-params.h"
+
+#include "rtsp-onvif-client.h"
+#include "rtsp-onvif-media-factory.h"
+#include "rtsp-onvif-media.h"
+#include "rtsp-onvif-server.h"
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SERVER_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2015 Centricular Ltd
+ * Author: Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-session-media
+ * @short_description: Media managed in a session
+ * @see_also: #GstRTSPMedia, #GstRTSPSession
+ *
+ * The #GstRTSPSessionMedia object manages a #GstRTSPMedia with a given path.
+ *
+ * With gst_rtsp_session_media_get_transport() and
+ * gst_rtsp_session_media_set_transport() the transports of a #GstRTSPStream of
+ * the managed #GstRTSPMedia can be retrieved and configured.
+ *
+ * Use gst_rtsp_session_media_set_state() to control the media state and
+ * transports.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-session.h"
+
+struct _GstRTSPSessionMediaPrivate
+{
+ GMutex lock;
+ gchar *path; /* unmutable */
+ gint path_len; /* unmutable */
+ GstRTSPMedia *media; /* unmutable */
+ GstRTSPState state; /* protected by lock */
+ guint counter; /* protected by lock */
+
+ GPtrArray *transports; /* protected by lock */
+};
+
+enum
+{
+ PROP_0,
+ PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_media_debug);
+#define GST_CAT_DEFAULT rtsp_session_media_debug
+
+static void gst_rtsp_session_media_finalize (GObject * obj);
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPSessionMedia, gst_rtsp_session_media,
+ G_TYPE_OBJECT);
+
+static void
+gst_rtsp_session_media_class_init (GstRTSPSessionMediaClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->finalize = gst_rtsp_session_media_finalize;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_session_media_debug, "rtspsessionmedia", 0,
+ "GstRTSPSessionMedia");
+}
+
+static void
+gst_rtsp_session_media_init (GstRTSPSessionMedia * media)
+{
+ GstRTSPSessionMediaPrivate *priv;
+
+ media->priv = priv = gst_rtsp_session_media_get_instance_private (media);
+
+ g_mutex_init (&priv->lock);
+ priv->state = GST_RTSP_STATE_INIT;
+}
+
+static void
+gst_rtsp_session_media_finalize (GObject * obj)
+{
+ GstRTSPSessionMedia *media;
+ GstRTSPSessionMediaPrivate *priv;
+
+ media = GST_RTSP_SESSION_MEDIA (obj);
+ priv = media->priv;
+
+ GST_INFO ("free session media %p", media);
+
+ gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
+
+ gst_rtsp_media_unprepare (priv->media);
+
+ g_ptr_array_unref (priv->transports);
+
+ g_free (priv->path);
+ g_object_unref (priv->media);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_session_media_parent_class)->finalize (obj);
+}
+
+static void
+free_session_media (gpointer data)
+{
+ if (data)
+ g_object_unref (data);
+}
+
+/**
+ * gst_rtsp_session_media_new:
+ * @path: the path
+ * @media: (transfer full): the #GstRTSPMedia
+ *
+ * Create a new #GstRTSPSessionMedia that manages the streams
+ * in @media for @path. @media should be prepared.
+ *
+ * Ownership is taken of @media.
+ *
+ * Returns: (transfer full): a new #GstRTSPSessionMedia.
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_media_new (const gchar * path, GstRTSPMedia * media)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GstRTSPSessionMedia *result;
+ guint n_streams;
+ GstRTSPMediaStatus status;
+
+ g_return_val_if_fail (path != NULL, NULL);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ status = gst_rtsp_media_get_status (media);
+ g_return_val_if_fail (status == GST_RTSP_MEDIA_STATUS_PREPARED || status ==
+ GST_RTSP_MEDIA_STATUS_SUSPENDED, NULL);
+
+ result = g_object_new (GST_TYPE_RTSP_SESSION_MEDIA, NULL);
+ priv = result->priv;
+
+ priv->path = g_strdup (path);
+ priv->path_len = strlen (path);
+ priv->media = media;
+
+ /* prealloc the streams now, filled with NULL */
+ n_streams = gst_rtsp_media_n_streams (media);
+ priv->transports = g_ptr_array_new_full (n_streams, free_session_media);
+ g_ptr_array_set_size (priv->transports, n_streams);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_media_matches:
+ * @media: a #GstRTSPSessionMedia
+ * @path: a path
+ * @matched: (out): the amount of matched characters of @path
+ *
+ * Check if the path of @media matches @path. It @path matches, the amount of
+ * matched characters is returned in @matched.
+ *
+ * Returns: %TRUE when @path matches the path of @media.
+ */
+gboolean
+gst_rtsp_session_media_matches (GstRTSPSessionMedia * media,
+ const gchar * path, gint * matched)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ gint len;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
+ g_return_val_if_fail (path != NULL, FALSE);
+ g_return_val_if_fail (matched != NULL, FALSE);
+
+ priv = media->priv;
+ len = strlen (path);
+
+ /* path needs to be smaller than the media path */
+ if (len < priv->path_len)
+ return FALSE;
+
+ /* special case when "/" is the entire path */
+ if (priv->path_len == 1 && priv->path[0] == '/' && path[0] == '/') {
+ *matched = 1;
+ return TRUE;
+ }
+
+ /* if media path is larger, it there should be a / following the path */
+ if (len > priv->path_len && path[priv->path_len] != '/')
+ return FALSE;
+
+ *matched = priv->path_len;
+
+ return strncmp (path, priv->path, priv->path_len) == 0;
+}
+
+/**
+ * gst_rtsp_session_media_get_media:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Get the #GstRTSPMedia that was used when constructing @media
+ *
+ * Returns: (transfer none) (nullable): the #GstRTSPMedia of @media.
+ * Remains valid as long as @media is valid.
+ */
+GstRTSPMedia *
+gst_rtsp_session_media_get_media (GstRTSPSessionMedia * media)
+{
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+
+ return media->priv->media;
+}
+
+/**
+ * gst_rtsp_session_media_get_base_time:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Get the base_time of the #GstRTSPMedia in @media
+ *
+ * Returns: the base_time of the media.
+ */
+GstClockTime
+gst_rtsp_session_media_get_base_time (GstRTSPSessionMedia * media)
+{
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), GST_CLOCK_TIME_NONE);
+
+ return gst_rtsp_media_get_base_time (media->priv->media);
+}
+
+/**
+ * gst_rtsp_session_media_get_rtpinfo:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Retrieve the RTP-Info header string for all streams in @media
+ * with configured transports.
+ *
+ * Returns: (transfer full) (nullable): The RTP-Info as a string or
+ * %NULL when no RTP-Info could be generated, g_free() after usage.
+ */
+gchar *
+gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia * media)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GString *rtpinfo = NULL;
+ GstRTSPStreamTransport *transport;
+ GstRTSPStream *stream;
+ guint i, n_streams;
+ GstClockTime earliest = GST_CLOCK_TIME_NONE;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+
+ priv = media->priv;
+ g_mutex_lock (&priv->lock);
+
+ if (gst_rtsp_media_get_status (priv->media) != GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto not_prepared;
+
+ n_streams = priv->transports->len;
+
+ /* first step, take lowest running-time from all streams */
+ GST_LOG_OBJECT (media, "determining start time among %d transports",
+ n_streams);
+
+ for (i = 0; i < n_streams; i++) {
+ GstClockTime running_time;
+
+ transport = g_ptr_array_index (priv->transports, i);
+ if (transport == NULL) {
+ GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i);
+ continue;
+ }
+
+ stream = gst_rtsp_stream_transport_get_stream (transport);
+ if (!gst_rtsp_stream_is_sender (stream))
+ continue;
+ if (!gst_rtsp_stream_get_rtpinfo (stream, NULL, NULL, NULL, &running_time))
+ continue;
+
+ GST_LOG_OBJECT (media, "running time of %d stream: %" GST_TIME_FORMAT, i,
+ GST_TIME_ARGS (running_time));
+
+ if (!GST_CLOCK_TIME_IS_VALID (earliest)) {
+ earliest = running_time;
+ } else {
+ earliest = MIN (earliest, running_time);
+ }
+ }
+
+ GST_LOG_OBJECT (media, "media start time: %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (earliest));
+
+ /* next step, scale all rtptime of all streams to lowest running-time */
+ GST_LOG_OBJECT (media, "collecting RTP info for %d transports", n_streams);
+
+ for (i = 0; i < n_streams; i++) {
+ gchar *stream_rtpinfo;
+
+ transport = g_ptr_array_index (priv->transports, i);
+ if (transport == NULL) {
+ GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i);
+ continue;
+ }
+
+ stream_rtpinfo =
+ gst_rtsp_stream_transport_get_rtpinfo (transport, earliest);
+ if (stream_rtpinfo == NULL) {
+ GST_DEBUG_OBJECT (media, "ignoring unknown RTPInfo %d", i);
+ continue;
+ }
+
+ if (rtpinfo == NULL)
+ rtpinfo = g_string_new ("");
+ else
+ g_string_append (rtpinfo, ", ");
+
+ g_string_append (rtpinfo, stream_rtpinfo);
+ g_free (stream_rtpinfo);
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (rtpinfo == NULL) {
+ GST_WARNING_OBJECT (media, "RTP info is empty");
+ return NULL;
+ }
+ return g_string_free (rtpinfo, FALSE);
+
+ /* ERRORS */
+not_prepared:
+ {
+ g_mutex_unlock (&priv->lock);
+ GST_ERROR_OBJECT (media, "media was not prepared");
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_session_media_set_transport:
+ * @media: a #GstRTSPSessionMedia
+ * @stream: a #GstRTSPStream
+ * @tr: (transfer full): a #GstRTSPTransport
+ *
+ * Configure the transport for @stream to @tr in @media.
+ *
+ * Returns: (transfer none): the new or updated #GstRTSPStreamTransport for @stream.
+ */
+GstRTSPStreamTransport *
+gst_rtsp_session_media_set_transport (GstRTSPSessionMedia * media,
+ GstRTSPStream * stream, GstRTSPTransport * tr)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GstRTSPStreamTransport *result;
+ guint idx;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (tr != NULL, NULL);
+ priv = media->priv;
+ idx = gst_rtsp_stream_get_index (stream);
+ g_return_val_if_fail (idx < priv->transports->len, NULL);
+
+ g_mutex_lock (&priv->lock);
+ result = g_ptr_array_index (priv->transports, idx);
+ if (result == NULL) {
+ result = gst_rtsp_stream_transport_new (stream, tr);
+ g_ptr_array_index (priv->transports, idx) = result;
+ g_mutex_unlock (&priv->lock);
+ } else {
+ gst_rtsp_stream_transport_set_transport (result, tr);
+ g_mutex_unlock (&priv->lock);
+ }
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_media_get_transport:
+ * @media: a #GstRTSPSessionMedia
+ * @idx: the stream index
+ *
+ * Get a previously created #GstRTSPStreamTransport for the stream at @idx.
+ *
+ * Returns: (transfer none) (nullable): a #GstRTSPStreamTransport that is
+ * valid until the session of @media is unreffed.
+ */
+GstRTSPStreamTransport *
+gst_rtsp_session_media_get_transport (GstRTSPSessionMedia * media, guint idx)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GstRTSPStreamTransport *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+ priv = media->priv;
+ g_return_val_if_fail (idx < priv->transports->len, NULL);
+
+ g_mutex_lock (&priv->lock);
+ result = g_ptr_array_index (priv->transports, idx);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_media_get_transports:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Get a list of all available #GstRTSPStreamTransport in this session.
+ *
+ * Returns: (transfer full) (element-type GstRTSPStreamTransport): a
+ * list of #GstRTSPStreamTransport, g_ptr_array_unref () after usage.
+ *
+ * Since: 1.14
+ */
+GPtrArray *
+gst_rtsp_session_media_get_transports (GstRTSPSessionMedia * media)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GPtrArray *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = g_ptr_array_ref (priv->transports);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_media_alloc_channels:
+ * @media: a #GstRTSPSessionMedia
+ * @range: (out): a #GstRTSPRange
+ *
+ * Fill @range with the next available min and max channels for
+ * interleaved transport.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia * media,
+ GstRTSPRange * range)
+{
+ GstRTSPSessionMediaPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ range->min = priv->counter++;
+ range->max = priv->counter++;
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_session_media_set_state:
+ * @media: a #GstRTSPSessionMedia
+ * @state: the new state
+ *
+ * Tell the media object @media to change to @state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_session_media_set_state (GstRTSPSessionMedia * media, GstState state)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ gboolean ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ ret = gst_rtsp_media_set_state (priv->media, state, priv->transports);
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_session_media_set_rtsp_state:
+ * @media: a #GstRTSPSessionMedia
+ * @state: a #GstRTSPState
+ *
+ * Set the RTSP state of @media to @state.
+ */
+void
+gst_rtsp_session_media_set_rtsp_state (GstRTSPSessionMedia * media,
+ GstRTSPState state)
+{
+ GstRTSPSessionMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SESSION_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->state = state;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_media_get_rtsp_state:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Get the current RTSP state of @media.
+ *
+ * Returns: the current RTSP state of @media.
+ */
+GstRTSPState
+gst_rtsp_session_media_get_rtsp_state (GstRTSPSessionMedia * media)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GstRTSPState ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media),
+ GST_RTSP_STATE_INVALID);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ ret = priv->state;
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp/gstrtsptransport.h>
+
+#ifndef __GST_RTSP_SESSION_MEDIA_H__
+#define __GST_RTSP_SESSION_MEDIA_H__
+
+#include "rtsp-server-prelude.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_SESSION_MEDIA (gst_rtsp_session_media_get_type ())
+#define GST_IS_RTSP_SESSION_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION_MEDIA))
+#define GST_IS_RTSP_SESSION_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION_MEDIA))
+#define GST_RTSP_SESSION_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaClass))
+#define GST_RTSP_SESSION_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMedia))
+#define GST_RTSP_SESSION_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaClass))
+#define GST_RTSP_SESSION_MEDIA_CAST(obj) ((GstRTSPSessionMedia*)(obj))
+#define GST_RTSP_SESSION_MEDIA_CLASS_CAST(klass) ((GstRTSPSessionMediaClass*)(klass))
+
+typedef struct _GstRTSPSessionMedia GstRTSPSessionMedia;
+typedef struct _GstRTSPSessionMediaClass GstRTSPSessionMediaClass;
+typedef struct _GstRTSPSessionMediaPrivate GstRTSPSessionMediaPrivate;
+
+/**
+ * GstRTSPSessionMedia:
+ *
+ * State of a client session regarding a specific media identified by path.
+ */
+struct _GstRTSPSessionMedia
+{
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPSessionMediaPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstRTSPSessionMediaClass
+{
+ GObjectClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_session_media_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPSessionMedia * gst_rtsp_session_media_new (const gchar *path,
+ GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_session_media_matches (GstRTSPSessionMedia *media,
+ const gchar *path,
+ gint * matched);
+
+GST_RTSP_SERVER_API
+GstRTSPMedia * gst_rtsp_session_media_get_media (GstRTSPSessionMedia *media);
+
+GST_RTSP_SERVER_API
+GstClockTime gst_rtsp_session_media_get_base_time (GstRTSPSessionMedia *media);
+/* control media */
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_session_media_set_state (GstRTSPSessionMedia *media,
+ GstState state);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_session_media_set_rtsp_state (GstRTSPSessionMedia *media,
+ GstRTSPState state);
+
+GST_RTSP_SERVER_API
+GstRTSPState gst_rtsp_session_media_get_rtsp_state (GstRTSPSessionMedia *media);
+
+/* get stream transport config */
+
+GST_RTSP_SERVER_API
+GstRTSPStreamTransport * gst_rtsp_session_media_set_transport (GstRTSPSessionMedia *media,
+ GstRTSPStream *stream,
+ GstRTSPTransport *tr);
+
+GST_RTSP_SERVER_API
+GstRTSPStreamTransport * gst_rtsp_session_media_get_transport (GstRTSPSessionMedia *media,
+ guint idx);
+
+GST_RTSP_SERVER_API
+GPtrArray * gst_rtsp_session_media_get_transports (GstRTSPSessionMedia *media);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia *media,
+ GstRTSPRange *range);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia * media);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPSessionMedia, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SESSION_MEDIA_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-session-pool
+ * @short_description: An object for managing sessions
+ * @see_also: #GstRTSPSession
+ *
+ * The #GstRTSPSessionPool object manages a list of #GstRTSPSession objects.
+ *
+ * The maximum number of sessions can be configured with
+ * gst_rtsp_session_pool_set_max_sessions(). The current number of sessions can
+ * be retrieved with gst_rtsp_session_pool_get_n_sessions().
+ *
+ * Use gst_rtsp_session_pool_create() to create a new #GstRTSPSession object.
+ * The session object can be found again with its id and
+ * gst_rtsp_session_pool_find().
+ *
+ * All sessions can be iterated with gst_rtsp_session_pool_filter().
+ *
+ * Run gst_rtsp_session_pool_cleanup() periodically to remove timed out sessions
+ * or use gst_rtsp_session_pool_create_watch() to be notified when session
+ * cleanup should be performed.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "rtsp-session-pool.h"
+
+struct _GstRTSPSessionPoolPrivate
+{
+ GMutex lock; /* protects everything in this struct */
+ guint max_sessions;
+ GHashTable *sessions;
+ guint sessions_cookie;
+};
+
+#define DEFAULT_MAX_SESSIONS 0
+
+enum
+{
+ PROP_0,
+ PROP_MAX_SESSIONS,
+ PROP_LAST
+};
+
+static const gchar session_id_charset[] =
+ { 'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm', 'n', 'o',
+ 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z', 'A', 'B', 'C', 'D',
+ 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M', 'N', 'O', 'P', 'Q', 'R', 'S',
+ 'T', 'U', 'V', 'W', 'X', 'Y', 'Z', '0', '1', '2', '3', '4', '5', '6', '7',
+ '8', '9', '-', '_', '.', '+' /* '$' Live555 in VLC strips off $ chars */
+};
+
+enum
+{
+ SIGNAL_SESSION_REMOVED,
+ SIGNAL_LAST
+};
+
+static guint gst_rtsp_session_pool_signals[SIGNAL_LAST] = { 0 };
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_debug);
+#define GST_CAT_DEFAULT rtsp_session_debug
+
+static void gst_rtsp_session_pool_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_pool_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_pool_finalize (GObject * object);
+
+static gchar *create_session_id (GstRTSPSessionPool * pool);
+static GstRTSPSession *create_session (GstRTSPSessionPool * pool,
+ const gchar * id);
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPSessionPool, gst_rtsp_session_pool,
+ G_TYPE_OBJECT);
+
+static void
+gst_rtsp_session_pool_class_init (GstRTSPSessionPoolClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_session_pool_get_property;
+ gobject_class->set_property = gst_rtsp_session_pool_set_property;
+ gobject_class->finalize = gst_rtsp_session_pool_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_MAX_SESSIONS,
+ g_param_spec_uint ("max-sessions", "Max Sessions",
+ "the maximum amount of sessions (0 = unlimited)",
+ 0, G_MAXUINT, DEFAULT_MAX_SESSIONS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED] =
+ g_signal_new ("session-removed", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPSessionPoolClass,
+ session_removed), NULL, NULL, NULL, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_SESSION);
+
+ klass->create_session_id = create_session_id;
+ klass->create_session = create_session;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsessionpool", 0,
+ "GstRTSPSessionPool");
+}
+
+static void
+gst_rtsp_session_pool_init (GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPoolPrivate *priv;
+
+ pool->priv = priv = gst_rtsp_session_pool_get_instance_private (pool);
+
+ g_mutex_init (&priv->lock);
+ priv->sessions = g_hash_table_new_full (g_str_hash, g_str_equal,
+ NULL, g_object_unref);
+ priv->max_sessions = DEFAULT_MAX_SESSIONS;
+}
+
+static GstRTSPFilterResult
+remove_sessions_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
+ gpointer user_data)
+{
+ return GST_RTSP_FILTER_REMOVE;
+}
+
+static void
+gst_rtsp_session_pool_finalize (GObject * object)
+{
+ GstRTSPSessionPool *pool = GST_RTSP_SESSION_POOL (object);
+ GstRTSPSessionPoolPrivate *priv = pool->priv;
+
+ gst_rtsp_session_pool_filter (pool, remove_sessions_func, NULL);
+ g_hash_table_unref (priv->sessions);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_session_pool_parent_class)->finalize (object);
+}
+
+static void
+gst_rtsp_session_pool_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPSessionPool *pool = GST_RTSP_SESSION_POOL (object);
+
+ switch (propid) {
+ case PROP_MAX_SESSIONS:
+ g_value_set_uint (value, gst_rtsp_session_pool_get_max_sessions (pool));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtsp_session_pool_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPSessionPool *pool = GST_RTSP_SESSION_POOL (object);
+
+ switch (propid) {
+ case PROP_MAX_SESSIONS:
+ gst_rtsp_session_pool_set_max_sessions (pool, g_value_get_uint (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ break;
+ }
+}
+
+/**
+ * gst_rtsp_session_pool_new:
+ *
+ * Create a new #GstRTSPSessionPool instance.
+ *
+ * Returns: (transfer full): A new #GstRTSPSessionPool. g_object_unref() after
+ * usage.
+ */
+GstRTSPSessionPool *
+gst_rtsp_session_pool_new (void)
+{
+ GstRTSPSessionPool *result;
+
+ result = g_object_new (GST_TYPE_RTSP_SESSION_POOL, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_pool_set_max_sessions:
+ * @pool: a #GstRTSPSessionPool
+ * @max: the maximum number of sessions
+ *
+ * Configure the maximum allowed number of sessions in @pool to @max.
+ * A value of 0 means an unlimited amount of sessions.
+ */
+void
+gst_rtsp_session_pool_set_max_sessions (GstRTSPSessionPool * pool, guint max)
+{
+ GstRTSPSessionPoolPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SESSION_POOL (pool));
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->max_sessions = max;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_pool_get_max_sessions:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Get the maximum allowed number of sessions in @pool. 0 means an unlimited
+ * amount of sessions.
+ *
+ * Returns: the maximum allowed number of sessions.
+ */
+guint
+gst_rtsp_session_pool_get_max_sessions (GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ guint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), 0);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = priv->max_sessions;
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_pool_get_n_sessions:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Get the amount of active sessions in @pool.
+ *
+ * Returns: the amount of active sessions in @pool.
+ */
+guint
+gst_rtsp_session_pool_get_n_sessions (GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ guint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), 0);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = g_hash_table_size (priv->sessions);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_pool_find:
+ * @pool: the pool to search
+ * @sessionid: the session id
+ *
+ * Find the session with @sessionid in @pool. The access time of the session
+ * will be updated with gst_rtsp_session_touch().
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPSession with @sessionid
+ * or %NULL when the session did not exist. g_object_unref() after usage.
+ */
+GstRTSPSession *
+gst_rtsp_session_pool_find (GstRTSPSessionPool * pool, const gchar * sessionid)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ GstRTSPSession *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+ g_return_val_if_fail (sessionid != NULL, NULL);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = g_hash_table_lookup (priv->sessions, sessionid);
+ if (result) {
+ g_object_ref (result);
+ gst_rtsp_session_touch (result);
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+static gchar *
+create_session_id (GstRTSPSessionPool * pool)
+{
+ gchar id[16];
+ gint i;
+
+ for (i = 0; i < 16; i++) {
+ id[i] =
+ session_id_charset[g_random_int_range (0,
+ G_N_ELEMENTS (session_id_charset))];
+ }
+
+ return g_strndup (id, 16);
+}
+
+static GstRTSPSession *
+create_session (GstRTSPSessionPool * pool, const gchar * id)
+{
+ return gst_rtsp_session_new (id);
+}
+
+/**
+ * gst_rtsp_session_pool_create:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Create a new #GstRTSPSession object in @pool.
+ *
+ * Returns: (transfer full) (nullable): a new #GstRTSPSession.
+ */
+GstRTSPSession *
+gst_rtsp_session_pool_create (GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ GstRTSPSession *result = NULL;
+ GstRTSPSessionPoolClass *klass;
+ gchar *id = NULL;
+ guint retry;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+
+ priv = pool->priv;
+
+ klass = GST_RTSP_SESSION_POOL_GET_CLASS (pool);
+
+ retry = 0;
+ do {
+ /* start by creating a new random session id, we assume that this is random
+ * enough to not cause a collision, which we will check later */
+ if (klass->create_session_id)
+ id = klass->create_session_id (pool);
+ else
+ goto no_function;
+
+ if (id == NULL)
+ goto no_session;
+
+ g_mutex_lock (&priv->lock);
+ /* check session limit */
+ if (priv->max_sessions > 0) {
+ if (g_hash_table_size (priv->sessions) >= priv->max_sessions)
+ goto too_many_sessions;
+ }
+ /* check if the sessionid existed */
+ result = g_hash_table_lookup (priv->sessions, id);
+ if (result) {
+ /* found, retry with a different session id */
+ result = NULL;
+ retry++;
+ if (retry > 100)
+ goto collision;
+ } else {
+ /* not found, create session and insert it in the pool */
+ if (klass->create_session)
+ result = klass->create_session (pool, id);
+ if (result == NULL)
+ goto too_many_sessions;
+ /* take additional ref for the pool */
+ g_object_ref (result);
+ g_hash_table_insert (priv->sessions,
+ (gchar *) gst_rtsp_session_get_sessionid (result), result);
+ priv->sessions_cookie++;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ g_free (id);
+ } while (result == NULL);
+
+ return result;
+
+ /* ERRORS */
+no_function:
+ {
+ GST_WARNING ("no create_session_id vmethod in GstRTSPSessionPool %p", pool);
+ return NULL;
+ }
+no_session:
+ {
+ GST_WARNING ("can't create session id with GstRTSPSessionPool %p", pool);
+ return NULL;
+ }
+collision:
+ {
+ GST_WARNING ("can't find unique sessionid for GstRTSPSessionPool %p", pool);
+ g_mutex_unlock (&priv->lock);
+ g_free (id);
+ return NULL;
+ }
+too_many_sessions:
+ {
+ GST_WARNING ("session pool reached max sessions of %d", priv->max_sessions);
+ g_mutex_unlock (&priv->lock);
+ g_free (id);
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_session_pool_remove:
+ * @pool: a #GstRTSPSessionPool
+ * @sess: (transfer none): a #GstRTSPSession
+ *
+ * Remove @sess from @pool, releasing the ref that the pool has on @sess.
+ *
+ * Returns: %TRUE if the session was found and removed.
+ */
+gboolean
+gst_rtsp_session_pool_remove (GstRTSPSessionPool * pool, GstRTSPSession * sess)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ gboolean found;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), FALSE);
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), FALSE);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_object_ref (sess);
+ found =
+ g_hash_table_remove (priv->sessions,
+ gst_rtsp_session_get_sessionid (sess));
+ if (found)
+ priv->sessions_cookie++;
+ g_mutex_unlock (&priv->lock);
+
+ if (found)
+ g_signal_emit (pool, gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED],
+ 0, sess);
+
+ g_object_unref (sess);
+
+ return found;
+}
+
+typedef struct
+{
+ gint64 now_monotonic_time;
+ GstRTSPSessionPool *pool;
+ GList *removed;
+} CleanupData;
+
+static gboolean
+cleanup_func (gchar * sessionid, GstRTSPSession * sess, CleanupData * data)
+{
+ gboolean expired;
+
+ expired = gst_rtsp_session_is_expired_usec (sess, data->now_monotonic_time);
+
+ if (expired) {
+ GST_DEBUG ("session expired");
+ data->removed = g_list_prepend (data->removed, g_object_ref (sess));
+ }
+
+ return expired;
+}
+
+/**
+ * gst_rtsp_session_pool_cleanup:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Inspect all the sessions in @pool and remove the sessions that are inactive
+ * for more than their timeout.
+ *
+ * Returns: the amount of sessions that got removed.
+ */
+guint
+gst_rtsp_session_pool_cleanup (GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ guint result;
+ CleanupData data;
+ GList *walk;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), 0);
+
+ priv = pool->priv;
+
+ data.now_monotonic_time = g_get_monotonic_time ();
+
+ data.pool = pool;
+ data.removed = NULL;
+
+ g_mutex_lock (&priv->lock);
+ result =
+ g_hash_table_foreach_remove (priv->sessions, (GHRFunc) cleanup_func,
+ &data);
+ if (result > 0)
+ priv->sessions_cookie++;
+ g_mutex_unlock (&priv->lock);
+
+ for (walk = data.removed; walk; walk = walk->next) {
+ GstRTSPSession *sess = walk->data;
+
+ g_signal_emit (pool,
+ gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED], 0, sess);
+
+ g_object_unref (sess);
+ }
+ g_list_free (data.removed);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_pool_filter:
+ * @pool: a #GstRTSPSessionPool
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each session in @pool. The result value of @func determines
+ * what happens to the session. @func will be called with the session pool
+ * locked so no further actions on @pool can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be set to the
+ * expired state and removed from @pool.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @pool.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @pool but
+ * will also be added with an additional ref to the result GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for all sessions.
+ *
+ * Returns: (element-type GstRTSPSession) (transfer full): a GList with all
+ * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_session_pool_filter (GstRTSPSessionPool * pool,
+ GstRTSPSessionPoolFilterFunc func, gpointer user_data)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ GHashTableIter iter;
+ gpointer key, value;
+ GList *result;
+ GHashTable *visited;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+
+ priv = pool->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ g_mutex_lock (&priv->lock);
+restart:
+ g_hash_table_iter_init (&iter, priv->sessions);
+ cookie = priv->sessions_cookie;
+ while (g_hash_table_iter_next (&iter, &key, &value)) {
+ GstRTSPSession *session = value;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ if (func) {
+ /* only visit each session once */
+ if (g_hash_table_contains (visited, session))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (session));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (pool, session, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->sessions_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ {
+ gboolean removed = TRUE;
+
+ if (changed)
+ /* something changed, check if we still have the session */
+ removed = g_hash_table_remove (priv->sessions, key);
+ else
+ g_hash_table_iter_remove (&iter);
+
+ if (removed) {
+ /* if we managed to remove the session, update the cookie and
+ * signal */
+ cookie = ++priv->sessions_cookie;
+ g_mutex_unlock (&priv->lock);
+
+ g_signal_emit (pool,
+ gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED], 0,
+ session);
+
+ g_mutex_lock (&priv->lock);
+ /* cookie could have changed again, make sure we restart */
+ changed |= (cookie != priv->sessions_cookie);
+ }
+ break;
+ }
+ case GST_RTSP_FILTER_REF:
+ /* keep ref */
+ result = g_list_prepend (result, g_object_ref (session));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
+
+typedef struct
+{
+ GSource source;
+ GstRTSPSessionPool *pool;
+ gint timeout;
+} GstPoolSource;
+
+static void
+collect_timeout (gchar * sessionid, GstRTSPSession * sess, GstPoolSource * psrc)
+{
+ gint timeout;
+ gint64 now_monotonic_time;
+
+ now_monotonic_time = g_get_monotonic_time ();
+
+ timeout = gst_rtsp_session_next_timeout_usec (sess, now_monotonic_time);
+
+ GST_INFO ("%p: next timeout: %d", sess, timeout);
+ if (psrc->timeout == -1 || timeout < psrc->timeout)
+ psrc->timeout = timeout;
+}
+
+static gboolean
+gst_pool_source_prepare (GSource * source, gint * timeout)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ GstPoolSource *psrc;
+ gboolean result;
+
+ psrc = (GstPoolSource *) source;
+ psrc->timeout = -1;
+ priv = psrc->pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_hash_table_foreach (priv->sessions, (GHFunc) collect_timeout, psrc);
+ g_mutex_unlock (&priv->lock);
+
+ if (timeout)
+ *timeout = psrc->timeout;
+
+ result = psrc->timeout == 0;
+
+ GST_INFO ("prepare %d, %d", psrc->timeout, result);
+
+ return result;
+}
+
+static gboolean
+gst_pool_source_check (GSource * source)
+{
+ GST_INFO ("check");
+
+ return gst_pool_source_prepare (source, NULL);
+}
+
+static gboolean
+gst_pool_source_dispatch (GSource * source, GSourceFunc callback,
+ gpointer user_data)
+{
+ gboolean res;
+ GstPoolSource *psrc = (GstPoolSource *) source;
+ GstRTSPSessionPoolFunc func = (GstRTSPSessionPoolFunc) callback;
+
+ GST_INFO ("dispatch");
+
+ if (func)
+ res = func (psrc->pool, user_data);
+ else
+ res = FALSE;
+
+ return res;
+}
+
+static void
+gst_pool_source_finalize (GSource * source)
+{
+ GstPoolSource *psrc = (GstPoolSource *) source;
+
+ GST_INFO ("finalize %p", psrc);
+
+ g_object_unref (psrc->pool);
+ psrc->pool = NULL;
+}
+
+static GSourceFuncs gst_pool_source_funcs = {
+ gst_pool_source_prepare,
+ gst_pool_source_check,
+ gst_pool_source_dispatch,
+ gst_pool_source_finalize
+};
+
+/**
+ * gst_rtsp_session_pool_create_watch:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Create a #GSource that will be dispatched when the session should be cleaned
+ * up.
+ *
+ * Returns: (transfer full): a #GSource
+ */
+GSource *
+gst_rtsp_session_pool_create_watch (GstRTSPSessionPool * pool)
+{
+ GstPoolSource *source;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+
+ source = (GstPoolSource *) g_source_new (&gst_pool_source_funcs,
+ sizeof (GstPoolSource));
+ source->pool = g_object_ref (pool);
+
+ return (GSource *) source;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_SESSION_POOL_H__
+#define __GST_RTSP_SESSION_POOL_H__
+
+#include "rtsp-server-prelude.h"
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPSessionPool GstRTSPSessionPool;
+typedef struct _GstRTSPSessionPoolClass GstRTSPSessionPoolClass;
+typedef struct _GstRTSPSessionPoolPrivate GstRTSPSessionPoolPrivate;
+
+#include "rtsp-session.h"
+
+#define GST_TYPE_RTSP_SESSION_POOL (gst_rtsp_session_pool_get_type ())
+#define GST_IS_RTSP_SESSION_POOL(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION_POOL))
+#define GST_IS_RTSP_SESSION_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION_POOL))
+#define GST_RTSP_SESSION_POOL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPoolClass))
+#define GST_RTSP_SESSION_POOL(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPool))
+#define GST_RTSP_SESSION_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPoolClass))
+#define GST_RTSP_SESSION_POOL_CAST(obj) ((GstRTSPSessionPool*)(obj))
+#define GST_RTSP_SESSION_POOL_CLASS_CAST(klass) ((GstRTSPSessionPoolClass*)(klass))
+
+/**
+ * GstRTSPSessionPool:
+ *
+ * An object that keeps track of the active sessions. This object is usually
+ * attached to a #GstRTSPServer object to manage the sessions in that server.
+ */
+struct _GstRTSPSessionPool {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPSessionPoolPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPSessionPoolClass:
+ * @create_session_id: create a new random session id. Subclasses can create
+ * custom session ids and should not check if the session exists.
+ * @create_session: make a new session object.
+ * @session_removed: a session was removed from the pool
+ */
+struct _GstRTSPSessionPoolClass {
+ GObjectClass parent_class;
+
+ gchar * (*create_session_id) (GstRTSPSessionPool *pool);
+ GstRTSPSession * (*create_session) (GstRTSPSessionPool *pool, const gchar *id);
+
+ /* signals */
+ void (*session_removed) (GstRTSPSessionPool *pool,
+ GstRTSPSession *session);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE - 1];
+};
+
+/**
+ * GstRTSPSessionPoolFunc:
+ * @pool: a #GstRTSPSessionPool object
+ * @user_data: user data that has been given when registering the handler
+ *
+ * The function that will be called from the GSource watch on the session pool.
+ *
+ * The function will be called when the pool must be cleaned up because one or
+ * more sessions timed out.
+ *
+ * Returns: %FALSE if the source should be removed.
+ */
+typedef gboolean (*GstRTSPSessionPoolFunc) (GstRTSPSessionPool *pool, gpointer user_data);
+
+/**
+ * GstRTSPSessionPoolFilterFunc:
+ * @pool: a #GstRTSPSessionPool object
+ * @session: a #GstRTSPSession in @pool
+ * @user_data: user data that has been given to gst_rtsp_session_pool_filter()
+ *
+ * This function will be called by the gst_rtsp_session_pool_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @session will be removed
+ * from @pool.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @session untouched in
+ * @pool.
+ *
+ * A value of GST_RTSP_FILTER_REF will add @session to the result #GList of
+ * gst_rtsp_session_pool_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPSessionPoolFilterFunc) (GstRTSPSessionPool *pool,
+ GstRTSPSession *session,
+ gpointer user_data);
+
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_session_pool_get_type (void);
+
+/* creating a session pool */
+
+GST_RTSP_SERVER_API
+GstRTSPSessionPool * gst_rtsp_session_pool_new (void);
+
+/* counting sessions */
+
+GST_RTSP_SERVER_API
+void gst_rtsp_session_pool_set_max_sessions (GstRTSPSessionPool *pool, guint max);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_session_pool_get_max_sessions (GstRTSPSessionPool *pool);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_session_pool_get_n_sessions (GstRTSPSessionPool *pool);
+
+/* managing sessions */
+
+GST_RTSP_SERVER_API
+GstRTSPSession * gst_rtsp_session_pool_create (GstRTSPSessionPool *pool);
+
+GST_RTSP_SERVER_API
+GstRTSPSession * gst_rtsp_session_pool_find (GstRTSPSessionPool *pool,
+ const gchar *sessionid);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_session_pool_remove (GstRTSPSessionPool *pool,
+ GstRTSPSession *sess);
+
+/* perform session maintenance */
+
+GST_RTSP_SERVER_API
+GList * gst_rtsp_session_pool_filter (GstRTSPSessionPool *pool,
+ GstRTSPSessionPoolFilterFunc func,
+ gpointer user_data);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_session_pool_cleanup (GstRTSPSessionPool *pool);
+
+GST_RTSP_SERVER_API
+GSource * gst_rtsp_session_pool_create_watch (GstRTSPSessionPool *pool);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPSessionPool, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SESSION_POOL_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-session
+ * @short_description: An object to manage media
+ * @see_also: #GstRTSPSessionPool, #GstRTSPSessionMedia, #GstRTSPMedia
+ *
+ * The #GstRTSPSession is identified by an id, unique in the
+ * #GstRTSPSessionPool that created the session and manages media and its
+ * configuration.
+ *
+ * A #GstRTSPSession has a timeout that can be retrieved with
+ * gst_rtsp_session_get_timeout(). You can check if the sessions is expired with
+ * gst_rtsp_session_is_expired(). gst_rtsp_session_touch() will reset the
+ * expiration counter of the session.
+ *
+ * When a client configures a media with SETUP, a session will be created to
+ * keep track of the configuration of that media. With
+ * gst_rtsp_session_manage_media(), the media is added to the managed media
+ * in the session. With gst_rtsp_session_release_media() the media can be
+ * released again from the session. Managed media is identified in the sessions
+ * with a url. Use gst_rtsp_session_get_media() to get the media that matches
+ * (part of) the given url.
+ *
+ * The media in a session can be iterated with gst_rtsp_session_filter().
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-session.h"
+
+struct _GstRTSPSessionPrivate
+{
+ GMutex lock; /* protects everything but sessionid and create_time */
+ gchar *sessionid;
+
+ guint timeout;
+ gboolean timeout_always_visible;
+ GMutex last_access_lock;
+ gint64 last_access_monotonic_time;
+ gint64 last_access_real_time;
+ gint expire_count;
+
+ GList *medias;
+ guint medias_cookie;
+ guint extra_time_timeout;
+};
+
+#undef DEBUG
+
+#define DEFAULT_TIMEOUT 60
+#define NO_TIMEOUT -1
+#define DEFAULT_ALWAYS_VISIBLE FALSE
+#define DEFAULT_EXTRA_TIMEOUT 5
+
+enum
+{
+ PROP_0,
+ PROP_SESSIONID,
+ PROP_TIMEOUT,
+ PROP_TIMEOUT_ALWAYS_VISIBLE,
+ PROP_EXTRA_TIME_TIMEOUT,
+ PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_debug);
+#define GST_CAT_DEFAULT rtsp_session_debug
+
+static void gst_rtsp_session_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_finalize (GObject * obj);
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPSession, gst_rtsp_session, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_session_class_init (GstRTSPSessionClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_session_get_property;
+ gobject_class->set_property = gst_rtsp_session_set_property;
+ gobject_class->finalize = gst_rtsp_session_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_SESSIONID,
+ g_param_spec_string ("sessionid", "Sessionid", "the session id",
+ NULL, G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY |
+ G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TIMEOUT,
+ g_param_spec_uint ("timeout", "timeout",
+ "the timeout of the session (0 = never)", 0, G_MAXUINT,
+ DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TIMEOUT_ALWAYS_VISIBLE,
+ g_param_spec_boolean ("timeout-always-visible", "Timeout Always Visible ",
+ "timeout always visible in header",
+ DEFAULT_ALWAYS_VISIBLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPSession::extra-timeout:
+ *
+ * Extra time to add to the timeout, in seconds. This only affects the
+ * time until a session is considered timed out and is not signalled
+ * in the RTSP request responses. Only the value of the timeout
+ * property is signalled in the request responses.
+ *
+ * Default value is 5 seconds.
+ * If the application is using a buffer that is configured to hold
+ * amount of data equal to the sessiontimeout, extra-timeout can be
+ * set to zero to prevent loss of data
+ *
+ * Since: 1.18
+ */
+ g_object_class_install_property (gobject_class, PROP_EXTRA_TIME_TIMEOUT,
+ g_param_spec_uint ("extra-timeout",
+ "Add extra time to timeout ", "Add extra time to timeout", 0,
+ G_MAXUINT, DEFAULT_EXTRA_TIMEOUT,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsession", 0,
+ "GstRTSPSession");
+}
+
+static void
+gst_rtsp_session_init (GstRTSPSession * session)
+{
+ GstRTSPSessionPrivate *priv;
+
+ session->priv = priv = gst_rtsp_session_get_instance_private (session);
+
+ GST_INFO ("init session %p", session);
+
+ g_mutex_init (&priv->lock);
+ g_mutex_init (&priv->last_access_lock);
+ priv->timeout = DEFAULT_TIMEOUT;
+ priv->extra_time_timeout = DEFAULT_EXTRA_TIMEOUT;
+
+ gst_rtsp_session_touch (session);
+}
+
+static void
+gst_rtsp_session_finalize (GObject * obj)
+{
+ GstRTSPSession *session;
+ GstRTSPSessionPrivate *priv;
+
+ session = GST_RTSP_SESSION (obj);
+ priv = session->priv;
+
+ GST_INFO ("finalize session %p", session);
+
+ /* free all media */
+ g_list_free_full (priv->medias, g_object_unref);
+
+ /* free session id */
+ g_free (priv->sessionid);
+ g_mutex_clear (&priv->last_access_lock);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_session_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_session_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPSession *session = GST_RTSP_SESSION (object);
+ GstRTSPSessionPrivate *priv = session->priv;
+
+ switch (propid) {
+ case PROP_SESSIONID:
+ g_value_set_string (value, priv->sessionid);
+ break;
+ case PROP_TIMEOUT:
+ g_value_set_uint (value, gst_rtsp_session_get_timeout (session));
+ break;
+ case PROP_TIMEOUT_ALWAYS_VISIBLE:
+ g_value_set_boolean (value, priv->timeout_always_visible);
+ break;
+ case PROP_EXTRA_TIME_TIMEOUT:
+ g_value_set_uint (value, priv->extra_time_timeout);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_session_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPSession *session = GST_RTSP_SESSION (object);
+ GstRTSPSessionPrivate *priv = session->priv;
+
+ switch (propid) {
+ case PROP_SESSIONID:
+ g_free (priv->sessionid);
+ priv->sessionid = g_value_dup_string (value);
+ break;
+ case PROP_TIMEOUT:
+ gst_rtsp_session_set_timeout (session, g_value_get_uint (value));
+ break;
+ case PROP_TIMEOUT_ALWAYS_VISIBLE:
+ g_mutex_lock (&priv->lock);
+ priv->timeout_always_visible = g_value_get_boolean (value);
+ g_mutex_unlock (&priv->lock);
+ break;
+ case PROP_EXTRA_TIME_TIMEOUT:
+ g_mutex_lock (&priv->lock);
+ priv->extra_time_timeout = g_value_get_uint (value);
+ g_mutex_unlock (&priv->lock);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_session_manage_media:
+ * @sess: a #GstRTSPSession
+ * @path: the path for the media
+ * @media: (transfer full): a #GstRTSPMedia
+ *
+ * Manage the media object @obj in @sess. @path will be used to retrieve this
+ * media from the session with gst_rtsp_session_get_media().
+ *
+ * Ownership is taken from @media.
+ *
+ * Returns: (transfer none): a new @GstRTSPSessionMedia object.
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_manage_media (GstRTSPSession * sess, const gchar * path,
+ GstRTSPMedia * media)
+{
+ GstRTSPSessionPrivate *priv;
+ GstRTSPSessionMedia *result;
+ GstRTSPMediaStatus status;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
+ g_return_val_if_fail (path != NULL, NULL);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ status = gst_rtsp_media_get_status (media);
+ g_return_val_if_fail (status == GST_RTSP_MEDIA_STATUS_PREPARED || status ==
+ GST_RTSP_MEDIA_STATUS_SUSPENDED, NULL);
+
+ priv = sess->priv;
+
+ result = gst_rtsp_session_media_new (path, media);
+
+ g_mutex_lock (&priv->lock);
+ priv->medias = g_list_prepend (priv->medias, result);
+ priv->medias_cookie++;
+ g_mutex_unlock (&priv->lock);
+
+ GST_INFO ("manage new media %p in session %p", media, result);
+
+ return result;
+}
+
+static void
+gst_rtsp_session_unset_transport_keepalive (GstRTSPSessionMedia * sessmedia)
+{
+ GstRTSPMedia *media;
+ guint i, n_streams;
+
+ media = gst_rtsp_session_media_get_media (sessmedia);
+ n_streams = gst_rtsp_media_n_streams (media);
+
+ for (i = 0; i < n_streams; i++) {
+ GstRTSPStreamTransport *transport =
+ gst_rtsp_session_media_get_transport (sessmedia, i);
+
+ if (!transport)
+ continue;
+
+ gst_rtsp_stream_transport_set_keepalive (transport, NULL, NULL, NULL);
+ }
+}
+
+/**
+ * gst_rtsp_session_release_media:
+ * @sess: a #GstRTSPSession
+ * @media: (transfer none): a #GstRTSPMedia
+ *
+ * Release the managed @media in @sess, freeing the memory allocated by it.
+ *
+ * Returns: %TRUE if there are more media session left in @sess.
+ */
+gboolean
+gst_rtsp_session_release_media (GstRTSPSession * sess,
+ GstRTSPSessionMedia * media)
+{
+ GstRTSPSessionPrivate *priv;
+ GList *find;
+ gboolean more;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), FALSE);
+ g_return_val_if_fail (media != NULL, FALSE);
+
+ priv = sess->priv;
+
+ g_mutex_lock (&priv->lock);
+ find = g_list_find (priv->medias, media);
+ if (find) {
+ priv->medias = g_list_delete_link (priv->medias, find);
+ priv->medias_cookie++;
+ }
+ more = (priv->medias != NULL);
+ g_mutex_unlock (&priv->lock);
+
+ if (find && !more)
+ gst_rtsp_session_unset_transport_keepalive (media);
+
+ if (find)
+ g_object_unref (media);
+
+ return more;
+}
+
+/**
+ * gst_rtsp_session_get_media:
+ * @sess: a #GstRTSPSession
+ * @path: the path for the media
+ * @matched: (out): the amount of matched characters
+ *
+ * Get the session media for @path. @matched will contain the number of matched
+ * characters of @path.
+ *
+ * Returns: (transfer none) (nullable): the configuration for @path in @sess.
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_get_media (GstRTSPSession * sess, const gchar * path,
+ gint * matched)
+{
+ GstRTSPSessionPrivate *priv;
+ GstRTSPSessionMedia *result;
+ GList *walk;
+ gint best;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
+ g_return_val_if_fail (path != NULL, NULL);
+
+ priv = sess->priv;
+ result = NULL;
+ best = 0;
+
+ g_mutex_lock (&priv->lock);
+ for (walk = priv->medias; walk; walk = g_list_next (walk)) {
+ GstRTSPSessionMedia *test;
+
+ test = (GstRTSPSessionMedia *) walk->data;
+
+ /* find largest match */
+ if (gst_rtsp_session_media_matches (test, path, matched)) {
+ if (best < *matched) {
+ result = test;
+ best = *matched;
+ }
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ *matched = best;
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_filter:
+ * @sess: a #GstRTSPSession
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each media in @sess. The result value of @func determines
+ * what happens to the media. @func will be called with @sess
+ * locked so no further actions on @sess can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the media will be removed from
+ * @sess.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the media will remain in @sess.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the media will remain in @sess but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for all media.
+ *
+ * Returns: (element-type GstRTSPSessionMedia) (transfer full): a GList with all
+ * media for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_session_filter (GstRTSPSession * sess,
+ GstRTSPSessionFilterFunc func, gpointer user_data)
+{
+ GstRTSPSessionPrivate *priv;
+ GList *result, *walk, *next;
+ GHashTable *visited;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
+
+ priv = sess->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ g_mutex_lock (&priv->lock);
+restart:
+ cookie = priv->medias_cookie;
+ for (walk = priv->medias; walk; walk = next) {
+ GstRTSPSessionMedia *media = walk->data;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ next = g_list_next (walk);
+
+ if (func) {
+ /* only visit each media once */
+ if (g_hash_table_contains (visited, media))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (media));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (sess, media, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->medias_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ if (changed)
+ priv->medias = g_list_remove (priv->medias, media);
+ else
+ priv->medias = g_list_delete_link (priv->medias, walk);
+ cookie = ++priv->medias_cookie;
+ g_object_unref (media);
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (media));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_new:
+ * @sessionid: a session id
+ *
+ * Create a new #GstRTSPSession instance with @sessionid.
+ *
+ * Returns: (transfer full): a new #GstRTSPSession
+ */
+GstRTSPSession *
+gst_rtsp_session_new (const gchar * sessionid)
+{
+ GstRTSPSession *result;
+
+ g_return_val_if_fail (sessionid != NULL, NULL);
+
+ result = g_object_new (GST_TYPE_RTSP_SESSION, "sessionid", sessionid, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_get_sessionid:
+ * @session: a #GstRTSPSession
+ *
+ * Get the sessionid of @session.
+ *
+ * Returns: (transfer none) (nullable): the sessionid of @session.
+ * The value remains valid as long as @session is alive.
+ */
+const gchar *
+gst_rtsp_session_get_sessionid (GstRTSPSession * session)
+{
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (session), NULL);
+
+ return session->priv->sessionid;
+}
+
+/**
+ * gst_rtsp_session_get_header:
+ * @session: a #GstRTSPSession
+ *
+ * Get the string that can be placed in the Session header field.
+ *
+ * Returns: (transfer full) (nullable): the Session header of @session.
+ * g_free() after usage.
+ */
+gchar *
+gst_rtsp_session_get_header (GstRTSPSession * session)
+{
+ GstRTSPSessionPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (session), NULL);
+
+ priv = session->priv;
+
+
+ g_mutex_lock (&priv->lock);
+ if (priv->timeout_always_visible || priv->timeout != 60)
+ result = g_strdup_printf ("%s;timeout=%d", priv->sessionid, priv->timeout);
+ else
+ result = g_strdup (priv->sessionid);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_set_timeout:
+ * @session: a #GstRTSPSession
+ * @timeout: the new timeout
+ *
+ * Configure @session for a timeout of @timeout seconds. The session will be
+ * cleaned up when there is no activity for @timeout seconds.
+ */
+void
+gst_rtsp_session_set_timeout (GstRTSPSession * session, guint timeout)
+{
+ GstRTSPSessionPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SESSION (session));
+
+ priv = session->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->timeout = timeout;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_get_timeout:
+ * @session: a #GstRTSPSession
+ *
+ * Get the timeout value of @session.
+ *
+ * Returns: the timeout of @session in seconds.
+ */
+guint
+gst_rtsp_session_get_timeout (GstRTSPSession * session)
+{
+ GstRTSPSessionPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (session), 0);
+
+ priv = session->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->timeout;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_session_touch:
+ * @session: a #GstRTSPSession
+ *
+ * Update the last_access time of the session to the current time.
+ */
+void
+gst_rtsp_session_touch (GstRTSPSession * session)
+{
+ GstRTSPSessionPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SESSION (session));
+
+ priv = session->priv;
+
+ g_mutex_lock (&priv->last_access_lock);
+ priv->last_access_monotonic_time = g_get_monotonic_time ();
+ priv->last_access_real_time = g_get_real_time ();
+ g_mutex_unlock (&priv->last_access_lock);
+}
+
+/**
+ * gst_rtsp_session_prevent_expire:
+ * @session: a #GstRTSPSession
+ *
+ * Prevent @session from expiring.
+ */
+void
+gst_rtsp_session_prevent_expire (GstRTSPSession * session)
+{
+ g_return_if_fail (GST_IS_RTSP_SESSION (session));
+
+ g_atomic_int_add (&session->priv->expire_count, 1);
+}
+
+/**
+ * gst_rtsp_session_allow_expire:
+ * @session: a #GstRTSPSession
+ *
+ * Allow @session to expire. This method must be called an equal
+ * amount of time as gst_rtsp_session_prevent_expire().
+ */
+void
+gst_rtsp_session_allow_expire (GstRTSPSession * session)
+{
+ g_atomic_int_add (&session->priv->expire_count, -1);
+}
+
+/**
+ * gst_rtsp_session_next_timeout_usec:
+ * @session: a #GstRTSPSession
+ * @now: the current monotonic time
+ *
+ * Get the amount of milliseconds till the session will expire.
+ *
+ * Returns: the amount of milliseconds since the session will time out.
+ */
+gint
+gst_rtsp_session_next_timeout_usec (GstRTSPSession * session, gint64 now)
+{
+ GstRTSPSessionPrivate *priv;
+ gint res;
+ GstClockTime last_access, now_ns;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (session), -1);
+
+ priv = session->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* If timeout is set to 0, we never timeout */
+ if (priv->timeout == 0) {
+ g_mutex_unlock (&priv->lock);
+ return NO_TIMEOUT;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ g_mutex_lock (&priv->last_access_lock);
+ if (g_atomic_int_get (&priv->expire_count) != 0) {
+ /* touch session when the expire count is not 0 */
+ priv->last_access_monotonic_time = g_get_monotonic_time ();
+ priv->last_access_real_time = g_get_real_time ();
+ }
+
+ last_access = GST_USECOND * (priv->last_access_monotonic_time);
+
+ /* add timeout allow for priv->extra_time_timeout
+ * seconds of extra time */
+ last_access += priv->timeout * GST_SECOND +
+ (priv->extra_time_timeout * GST_SECOND);
+
+ g_mutex_unlock (&priv->last_access_lock);
+
+ now_ns = GST_USECOND * now;
+
+ if (last_access > now_ns) {
+ res = GST_TIME_AS_MSECONDS (last_access - now_ns);
+ } else {
+ res = 0;
+ }
+
+ return res;
+}
+
+/****** Deprecated API *******/
+
+/**
+ * gst_rtsp_session_next_timeout:
+ * @session: a #GstRTSPSession
+ * @now: (transfer none): the current system time
+ *
+ * Get the amount of milliseconds till the session will expire.
+ *
+ * Returns: the amount of milliseconds since the session will time out.
+ *
+ * Deprecated: Use gst_rtsp_session_next_timeout_usec() instead.
+ */
+#ifndef GST_REMOVE_DEPRECATED
+G_GNUC_BEGIN_IGNORE_DEPRECATIONS gint
+gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
+{
+ GstRTSPSessionPrivate *priv;
+ gint res;
+ GstClockTime last_access, now_ns;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (session), -1);
+ g_return_val_if_fail (now != NULL, -1);
+
+ priv = session->priv;
+
+ g_mutex_lock (&priv->last_access_lock);
+ if (g_atomic_int_get (&priv->expire_count) != 0) {
+ /* touch session when the expire count is not 0 */
+ priv->last_access_monotonic_time = g_get_monotonic_time ();
+ priv->last_access_real_time = g_get_real_time ();
+ }
+
+ last_access = GST_USECOND * (priv->last_access_real_time);
+
+ /* add timeout allow for priv->extra_time_timeout
+ * seconds of extra time */
+ last_access += priv->timeout * GST_SECOND +
+ (priv->extra_time_timeout * GST_SECOND);
+
+ g_mutex_unlock (&priv->last_access_lock);
+
+ now_ns = GST_TIMEVAL_TO_TIME (*now);
+
+ if (last_access > now_ns) {
+ res = GST_TIME_AS_MSECONDS (last_access - now_ns);
+ } else {
+ res = 0;
+ }
+
+ return res;
+}
+
+G_GNUC_END_IGNORE_DEPRECATIONS
+#endif
+/**
+ * gst_rtsp_session_is_expired_usec:
+ * @session: a #GstRTSPSession
+ * @now: the current monotonic time
+ *
+ * Check if @session timeout out.
+ *
+ * Returns: %TRUE if @session timed out
+ */
+ gboolean
+gst_rtsp_session_is_expired_usec (GstRTSPSession * session, gint64 now)
+{
+ gboolean res;
+
+ res = (gst_rtsp_session_next_timeout_usec (session, now) == 0);
+
+ return res;
+}
+
+
+/****** Deprecated API *******/
+
+/**
+ * gst_rtsp_session_is_expired:
+ * @session: a #GstRTSPSession
+ * @now: (transfer none): the current system time
+ *
+ * Check if @session timeout out.
+ *
+ * Returns: %TRUE if @session timed out
+ *
+ * Deprecated: Use gst_rtsp_session_is_expired_usec() instead.
+ */
+#ifndef GST_REMOVE_DEPRECATED
+G_GNUC_BEGIN_IGNORE_DEPRECATIONS gboolean
+gst_rtsp_session_is_expired (GstRTSPSession * session, GTimeVal * now)
+{
+ gboolean res;
+
+ res = gst_rtsp_session_next_timeout_usec (session,
+ (now->tv_sec * G_USEC_PER_SEC) + (now->tv_usec));
+
+ return res;
+}
+
+G_GNUC_END_IGNORE_DEPRECATIONS
+#endif
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp/gstrtsptransport.h>
+#include "rtsp-server-prelude.h" /* for GST_RTSP_SERVER_DEPRECATED_FOR */
+
+#ifndef __GST_RTSP_SESSION_H__
+#define __GST_RTSP_SESSION_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_SESSION (gst_rtsp_session_get_type ())
+#define GST_IS_RTSP_SESSION(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION))
+#define GST_IS_RTSP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION))
+#define GST_RTSP_SESSION_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION, GstRTSPSessionClass))
+#define GST_RTSP_SESSION(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION, GstRTSPSession))
+#define GST_RTSP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION, GstRTSPSessionClass))
+#define GST_RTSP_SESSION_CAST(obj) ((GstRTSPSession*)(obj))
+#define GST_RTSP_SESSION_CLASS_CAST(klass) ((GstRTSPSessionClass*)(klass))
+
+typedef struct _GstRTSPSession GstRTSPSession;
+typedef struct _GstRTSPSessionClass GstRTSPSessionClass;
+typedef struct _GstRTSPSessionPrivate GstRTSPSessionPrivate;
+
+/**
+ * GstRTSPFilterResult:
+ * @GST_RTSP_FILTER_REMOVE: Remove session
+ * @GST_RTSP_FILTER_KEEP: Keep session in the pool
+ * @GST_RTSP_FILTER_REF: Ref session in the result list
+ *
+ * Possible return values for gst_rtsp_session_pool_filter().
+ */
+typedef enum
+{
+ GST_RTSP_FILTER_REMOVE,
+ GST_RTSP_FILTER_KEEP,
+ GST_RTSP_FILTER_REF,
+} GstRTSPFilterResult;
+
+#include "rtsp-media.h"
+#include "rtsp-session-media.h"
+
+/**
+ * GstRTSPSession:
+ *
+ * Session information kept by the server for a specific client.
+ * One client session, identified with a session id, can handle multiple medias
+ * identified with the url of a media.
+ */
+struct _GstRTSPSession {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPSessionPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstRTSPSessionClass {
+ GObjectClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_session_get_type (void);
+
+/* create a new session */
+
+GST_RTSP_SERVER_API
+GstRTSPSession * gst_rtsp_session_new (const gchar *sessionid);
+
+GST_RTSP_SERVER_API
+const gchar * gst_rtsp_session_get_sessionid (GstRTSPSession *session);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_session_get_header (GstRTSPSession *session);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_session_set_timeout (GstRTSPSession *session, guint timeout);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_session_get_timeout (GstRTSPSession *session);
+
+/* session timeout stuff */
+
+GST_RTSP_SERVER_API
+void gst_rtsp_session_touch (GstRTSPSession *session);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_session_prevent_expire (GstRTSPSession *session);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_session_allow_expire (GstRTSPSession *session);
+
+GST_RTSP_SERVER_API
+gint gst_rtsp_session_next_timeout_usec (GstRTSPSession *session, gint64 now);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_session_is_expired_usec (GstRTSPSession *session, gint64 now);
+
+G_GNUC_BEGIN_IGNORE_DEPRECATIONS
+GST_RTSP_SERVER_DEPRECATED_FOR(gst_rtsp_session_next_timeout_usec)
+gint gst_rtsp_session_next_timeout (GstRTSPSession *session, GTimeVal *now);
+
+GST_RTSP_SERVER_DEPRECATED_FOR(gst_rtsp_session_is_expired_usec)
+gboolean gst_rtsp_session_is_expired (GstRTSPSession *session, GTimeVal *now);
+G_GNUC_END_IGNORE_DEPRECATIONS
+
+/* handle media in a session */
+
+GST_RTSP_SERVER_API
+GstRTSPSessionMedia * gst_rtsp_session_manage_media (GstRTSPSession *sess,
+ const gchar *path,
+ GstRTSPMedia *media);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_session_release_media (GstRTSPSession *sess,
+ GstRTSPSessionMedia *media);
+/* get media in a session */
+
+GST_RTSP_SERVER_API
+GstRTSPSessionMedia * gst_rtsp_session_get_media (GstRTSPSession *sess,
+ const gchar *path,
+ gint * matched);
+
+/**
+ * GstRTSPSessionFilterFunc:
+ * @sess: a #GstRTSPSession object
+ * @media: a #GstRTSPSessionMedia in @sess
+ * @user_data: user data that has been given to gst_rtsp_session_filter()
+ *
+ * This function will be called by the gst_rtsp_session_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @media will be removed
+ * from @sess.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @media untouched in
+ * @sess.
+ *
+ * A value of GST_RTSP_FILTER_REF will add @media to the result #GList of
+ * gst_rtsp_session_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPSessionFilterFunc) (GstRTSPSession *sess,
+ GstRTSPSessionMedia *media,
+ gpointer user_data);
+
+GST_RTSP_SERVER_API
+GList * gst_rtsp_session_filter (GstRTSPSession *sess,
+ GstRTSPSessionFilterFunc func,
+ gpointer user_data);
+
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPSession, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SESSION_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-stream-transport
+ * @short_description: A media stream transport configuration
+ * @see_also: #GstRTSPStream, #GstRTSPSessionMedia
+ *
+ * The #GstRTSPStreamTransport configures the transport used by a
+ * #GstRTSPStream. It is usually manages by a #GstRTSPSessionMedia object.
+ *
+ * With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured
+ * to handle the RTP and RTCP packets from the stream, for example when they
+ * need to be sent over TCP.
+ *
+ * With gst_rtsp_stream_transport_set_active() the transports are added and
+ * removed from the stream.
+ *
+ * A #GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP
+ * is received from the client. It will also call
+ * gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out.
+ *
+ * A #GstRTSPClient will call gst_rtsp_stream_transport_message_sent() when it
+ * has sent a data message for the transport.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include <stdlib.h>
+
+#include "rtsp-stream-transport.h"
+#include "rtsp-server-internal.h"
+
+struct _GstRTSPStreamTransportPrivate
+{
+ GstRTSPStream *stream;
+
+ GstRTSPSendFunc send_rtp;
+ GstRTSPSendFunc send_rtcp;
+ gpointer user_data;
+ GDestroyNotify notify;
+
+ GstRTSPSendListFunc send_rtp_list;
+ GstRTSPSendListFunc send_rtcp_list;
+ gpointer list_user_data;
+ GDestroyNotify list_notify;
+
+ GstRTSPBackPressureFunc back_pressure_func;
+ gpointer back_pressure_func_data;
+ GDestroyNotify back_pressure_func_notify;
+
+ GstRTSPKeepAliveFunc keep_alive;
+ gpointer ka_user_data;
+ GDestroyNotify ka_notify;
+ gboolean timed_out;
+
+ GstRTSPMessageSentFunc message_sent;
+ gpointer ms_user_data;
+ GDestroyNotify ms_notify;
+
+ GstRTSPMessageSentFuncFull message_sent_full;
+ gpointer msf_user_data;
+ GDestroyNotify msf_notify;
+
+ GstRTSPTransport *transport;
+ GstRTSPUrl *url;
+
+ GObject *rtpsource;
+
+ /* TCP backlog */
+ GstClockTime first_rtp_timestamp;
+ GstQueueArray *items;
+ GRecMutex backlog_lock;
+};
+
+#define MAX_BACKLOG_DURATION (10 * GST_SECOND)
+#define MAX_BACKLOG_SIZE 100
+
+typedef struct
+{
+ GstBuffer *buffer;
+ GstBufferList *buffer_list;
+ gboolean is_rtp;
+} BackLogItem;
+
+
+enum
+{
+ PROP_0,
+ PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_stream_transport_debug);
+#define GST_CAT_DEFAULT rtsp_stream_transport_debug
+
+static void gst_rtsp_stream_transport_finalize (GObject * obj);
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPStreamTransport, gst_rtsp_stream_transport,
+ G_TYPE_OBJECT);
+
+static void
+gst_rtsp_stream_transport_class_init (GstRTSPStreamTransportClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->finalize = gst_rtsp_stream_transport_finalize;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_stream_transport_debug, "rtspmediatransport",
+ 0, "GstRTSPStreamTransport");
+}
+
+static void
+clear_backlog_item (BackLogItem * item)
+{
+ gst_clear_buffer (&item->buffer);
+ gst_clear_buffer_list (&item->buffer_list);
+}
+
+static void
+gst_rtsp_stream_transport_init (GstRTSPStreamTransport * trans)
+{
+ trans->priv = gst_rtsp_stream_transport_get_instance_private (trans);
+ trans->priv->items = gst_queue_array_new_for_struct (sizeof (BackLogItem), 0);
+ trans->priv->first_rtp_timestamp = GST_CLOCK_TIME_NONE;
+ gst_queue_array_set_clear_func (trans->priv->items,
+ (GDestroyNotify) clear_backlog_item);
+ g_rec_mutex_init (&trans->priv->backlog_lock);
+}
+
+static void
+gst_rtsp_stream_transport_finalize (GObject * obj)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ GstRTSPStreamTransport *trans;
+
+ trans = GST_RTSP_STREAM_TRANSPORT (obj);
+ priv = trans->priv;
+
+ /* remove callbacks now */
+ gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
+ gst_rtsp_stream_transport_set_keepalive (trans, NULL, NULL, NULL);
+ gst_rtsp_stream_transport_set_message_sent (trans, NULL, NULL, NULL);
+
+ if (priv->stream)
+ g_object_unref (priv->stream);
+
+ if (priv->transport)
+ gst_rtsp_transport_free (priv->transport);
+
+ if (priv->url)
+ gst_rtsp_url_free (priv->url);
+
+ gst_queue_array_free (priv->items);
+
+ g_rec_mutex_clear (&priv->backlog_lock);
+
+ G_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
+}
+
+/**
+ * gst_rtsp_stream_transport_new:
+ * @stream: a #GstRTSPStream
+ * @tr: (transfer full): a GstRTSPTransport
+ *
+ * Create a new #GstRTSPStreamTransport that can be used to manage
+ * @stream with transport @tr.
+ *
+ * Returns: (transfer full): a new #GstRTSPStreamTransport
+ */
+GstRTSPStreamTransport *
+gst_rtsp_stream_transport_new (GstRTSPStream * stream, GstRTSPTransport * tr)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ GstRTSPStreamTransport *trans;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (tr != NULL, NULL);
+
+ trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
+ priv = trans->priv;
+ priv->stream = stream;
+ priv->stream = g_object_ref (priv->stream);
+ priv->transport = tr;
+
+ return trans;
+}
+
+/**
+ * gst_rtsp_stream_transport_get_stream:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the #GstRTSPStream used when constructing @trans.
+ *
+ * Returns: (transfer none) (nullable): the stream used when constructing @trans.
+ */
+GstRTSPStream *
+gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ return trans->priv->stream;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_callbacks:
+ * @trans: a #GstRTSPStreamTransport
+ * @send_rtp: (scope notified): a callback called when RTP should be sent
+ * @send_rtcp: (scope notified): a callback called when RTCP should be sent
+ * @user_data: (closure): user data passed to callbacks
+ * @notify: (allow-none): called with the user_data when no longer needed.
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
+ */
+void
+gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
+ GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
+ gpointer user_data, GDestroyNotify notify)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ priv->send_rtp = send_rtp;
+ priv->send_rtcp = send_rtcp;
+ if (priv->notify)
+ priv->notify (priv->user_data);
+ priv->user_data = user_data;
+ priv->notify = notify;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_list_callbacks:
+ * @trans: a #GstRTSPStreamTransport
+ * @send_rtp_list: (scope notified): a callback called when RTP should be sent
+ * @send_rtcp_list: (scope notified): a callback called when RTCP should be sent
+ * @user_data: (closure): user data passed to callbacks
+ * @notify: (allow-none): called with the user_data when no longer needed.
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_stream_transport_set_list_callbacks (GstRTSPStreamTransport * trans,
+ GstRTSPSendListFunc send_rtp_list, GstRTSPSendListFunc send_rtcp_list,
+ gpointer user_data, GDestroyNotify notify)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ priv->send_rtp_list = send_rtp_list;
+ priv->send_rtcp_list = send_rtcp_list;
+ if (priv->list_notify)
+ priv->list_notify (priv->list_user_data);
+ priv->list_user_data = user_data;
+ priv->list_notify = notify;
+}
+
+void
+gst_rtsp_stream_transport_set_back_pressure_callback (GstRTSPStreamTransport *
+ trans, GstRTSPBackPressureFunc back_pressure_func, gpointer user_data,
+ GDestroyNotify notify)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ priv->back_pressure_func = back_pressure_func;
+ if (priv->back_pressure_func_notify)
+ priv->back_pressure_func_notify (priv->back_pressure_func_data);
+ priv->back_pressure_func_data = user_data;
+ priv->back_pressure_func_notify = notify;
+}
+
+gboolean
+gst_rtsp_stream_transport_check_back_pressure (GstRTSPStreamTransport * trans,
+ gboolean is_rtp)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gboolean ret = FALSE;
+ guint8 channel;
+
+ priv = trans->priv;
+
+ if (is_rtp)
+ channel = priv->transport->interleaved.min;
+ else
+ channel = priv->transport->interleaved.max;
+
+ if (priv->back_pressure_func)
+ ret = priv->back_pressure_func (channel, priv->back_pressure_func_data);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_keepalive:
+ * @trans: a #GstRTSPStreamTransport
+ * @keep_alive: (scope notified): a callback called when the receiver is active
+ * @user_data: (closure): user data passed to callback
+ * @notify: (allow-none): called with the user_data when no longer needed.
+ *
+ * Install callbacks that will be called when RTCP packets are received from the
+ * receiver of @trans.
+ */
+void
+gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
+ GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ priv->keep_alive = keep_alive;
+ if (priv->ka_notify)
+ priv->ka_notify (priv->ka_user_data);
+ priv->ka_user_data = user_data;
+ priv->ka_notify = notify;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_message_sent:
+ * @trans: a #GstRTSPStreamTransport
+ * @message_sent: (scope notified): a callback called when a message has been sent
+ * @user_data: (closure): user data passed to callback
+ * @notify: (allow-none): called with the user_data when no longer needed
+ *
+ * Install a callback that will be called when a message has been sent on @trans.
+ */
+void
+gst_rtsp_stream_transport_set_message_sent (GstRTSPStreamTransport * trans,
+ GstRTSPMessageSentFunc message_sent, gpointer user_data,
+ GDestroyNotify notify)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ priv->message_sent = message_sent;
+ if (priv->ms_notify)
+ priv->ms_notify (priv->ms_user_data);
+ priv->ms_user_data = user_data;
+ priv->ms_notify = notify;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_message_sent_full:
+ * @trans: a #GstRTSPStreamTransport
+ * @message_sent: (scope notified): a callback called when a message has been sent
+ * @user_data: (closure): user data passed to callback
+ * @notify: (allow-none): called with the user_data when no longer needed
+ *
+ * Install a callback that will be called when a message has been sent on @trans.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_stream_transport_set_message_sent_full (GstRTSPStreamTransport * trans,
+ GstRTSPMessageSentFuncFull message_sent, gpointer user_data,
+ GDestroyNotify notify)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ priv->message_sent_full = message_sent;
+ if (priv->msf_notify)
+ priv->msf_notify (priv->msf_user_data);
+ priv->msf_user_data = user_data;
+ priv->msf_notify = notify;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_transport:
+ * @trans: a #GstRTSPStreamTransport
+ * @tr: (transfer full): a client #GstRTSPTransport
+ *
+ * Set @tr as the client transport. This function takes ownership of the
+ * passed @tr.
+ */
+void
+gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
+ GstRTSPTransport * tr)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+ g_return_if_fail (tr != NULL);
+
+ priv = trans->priv;
+
+ /* keep track of the transports in the stream. */
+ if (priv->transport)
+ gst_rtsp_transport_free (priv->transport);
+ priv->transport = tr;
+}
+
+/**
+ * gst_rtsp_stream_transport_get_transport:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the transport configured in @trans.
+ *
+ * Returns: (transfer none) (nullable): the transport configured in @trans. It remains
+ * valid for as long as @trans is valid.
+ */
+const GstRTSPTransport *
+gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ return trans->priv->transport;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_url:
+ * @trans: a #GstRTSPStreamTransport
+ * @url: (transfer none) (nullable): a client #GstRTSPUrl
+ *
+ * Set @url as the client url.
+ */
+void
+gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans,
+ const GstRTSPUrl * url)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ /* keep track of the transports in the stream. */
+ if (priv->url)
+ gst_rtsp_url_free (priv->url);
+ priv->url = (url ? gst_rtsp_url_copy (url) : NULL);
+}
+
+/**
+ * gst_rtsp_stream_transport_get_url:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the url configured in @trans.
+ *
+ * Returns: (transfer none) (nullable): the url configured in @trans.
+ * It remains valid for as long as @trans is valid.
+ */
+const GstRTSPUrl *
+gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ return trans->priv->url;
+}
+
+ /**
+ * gst_rtsp_stream_transport_get_rtpinfo:
+ * @trans: a #GstRTSPStreamTransport
+ * @start_time: a star time
+ *
+ * Get the RTP-Info string for @trans and @start_time.
+ *
+ * Returns: (transfer full) (nullable): the RTPInfo string for @trans
+ * and @start_time or %NULL when the RTP-Info could not be
+ * determined. g_free() after usage.
+ */
+gchar *
+gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport * trans,
+ GstClockTime start_time)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gchar *url_str;
+ GString *rtpinfo;
+ guint rtptime, seq, clock_rate;
+ GstClockTime running_time = GST_CLOCK_TIME_NONE;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ priv = trans->priv;
+
+ if (!gst_rtsp_stream_is_sender (priv->stream))
+ return NULL;
+ if (!gst_rtsp_stream_get_rtpinfo (priv->stream, &rtptime, &seq, &clock_rate,
+ &running_time))
+ return NULL;
+
+ GST_DEBUG ("RTP time %u, seq %u, rate %u, running-time %" GST_TIME_FORMAT,
+ rtptime, seq, clock_rate, GST_TIME_ARGS (running_time));
+
+ if (GST_CLOCK_TIME_IS_VALID (running_time)
+ && GST_CLOCK_TIME_IS_VALID (start_time)) {
+ if (running_time > start_time) {
+ rtptime -=
+ gst_util_uint64_scale_int (running_time - start_time, clock_rate,
+ GST_SECOND);
+ } else {
+ rtptime +=
+ gst_util_uint64_scale_int (start_time - running_time, clock_rate,
+ GST_SECOND);
+ }
+ }
+ GST_DEBUG ("RTP time %u, for start-time %" GST_TIME_FORMAT,
+ rtptime, GST_TIME_ARGS (start_time));
+
+ rtpinfo = g_string_new ("");
+
+ url_str = gst_rtsp_url_get_request_uri (trans->priv->url);
+ g_string_append_printf (rtpinfo, "url=%s;seq=%u;rtptime=%u",
+ url_str, seq, rtptime);
+ g_free (url_str);
+
+ return g_string_free (rtpinfo, FALSE);
+}
+
+/**
+ * gst_rtsp_stream_transport_set_active:
+ * @trans: a #GstRTSPStreamTransport
+ * @active: new state of @trans
+ *
+ * Activate or deactivate datatransfer configured in @trans.
+ *
+ * Returns: %TRUE when the state was changed.
+ */
+gboolean
+gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport * trans,
+ gboolean active)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+
+ priv = trans->priv;
+
+ if (active)
+ res = gst_rtsp_stream_add_transport (priv->stream, trans);
+ else
+ res = gst_rtsp_stream_remove_transport (priv->stream, trans);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_timed_out:
+ * @trans: a #GstRTSPStreamTransport
+ * @timedout: timed out value
+ *
+ * Set the timed out state of @trans to @timedout
+ */
+void
+gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport * trans,
+ gboolean timedout)
+{
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ trans->priv->timed_out = timedout;
+}
+
+/**
+ * gst_rtsp_stream_transport_is_timed_out:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Check if @trans is timed out.
+ *
+ * Returns: %TRUE if @trans timed out.
+ */
+gboolean
+gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+
+ return trans->priv->timed_out;
+}
+
+/**
+ * gst_rtsp_stream_transport_send_rtp:
+ * @trans: a #GstRTSPStreamTransport
+ * @buffer: (transfer none): a #GstBuffer
+ *
+ * Send @buffer to the installed RTP callback for @trans.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
+ GstBuffer * buffer)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gboolean res = FALSE;
+
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
+
+ priv = trans->priv;
+
+ if (priv->send_rtp)
+ res =
+ priv->send_rtp (buffer, priv->transport->interleaved.min,
+ priv->user_data);
+
+ if (res)
+ gst_rtsp_stream_transport_keep_alive (trans);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_send_rtcp:
+ * @trans: a #GstRTSPStreamTransport
+ * @buffer: (transfer none): a #GstBuffer
+ *
+ * Send @buffer to the installed RTCP callback for @trans.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
+ GstBuffer * buffer)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gboolean res = FALSE;
+
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
+
+ priv = trans->priv;
+
+ if (priv->send_rtcp)
+ res =
+ priv->send_rtcp (buffer, priv->transport->interleaved.max,
+ priv->user_data);
+
+ if (res)
+ gst_rtsp_stream_transport_keep_alive (trans);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_send_rtp_list:
+ * @trans: a #GstRTSPStreamTransport
+ * @buffer_list: (transfer none): a #GstBufferList
+ *
+ * Send @buffer_list to the installed RTP callback for @trans.
+ *
+ * Returns: %TRUE on success
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_stream_transport_send_rtp_list (GstRTSPStreamTransport * trans,
+ GstBufferList * buffer_list)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gboolean res = FALSE;
+
+ g_return_val_if_fail (GST_IS_BUFFER_LIST (buffer_list), FALSE);
+
+ priv = trans->priv;
+
+ if (priv->send_rtp_list) {
+ res =
+ priv->send_rtp_list (buffer_list, priv->transport->interleaved.min,
+ priv->list_user_data);
+ } else if (priv->send_rtp) {
+ guint n = gst_buffer_list_length (buffer_list), i;
+
+ for (i = 0; i < n; i++) {
+ GstBuffer *buffer = gst_buffer_list_get (buffer_list, i);
+
+ res =
+ priv->send_rtp (buffer, priv->transport->interleaved.min,
+ priv->user_data);
+ if (!res)
+ break;
+ }
+ }
+
+ if (res)
+ gst_rtsp_stream_transport_keep_alive (trans);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_send_rtcp_list:
+ * @trans: a #GstRTSPStreamTransport
+ * @buffer_list: (transfer none): a #GstBuffer
+ *
+ * Send @buffer_list to the installed RTCP callback for @trans.
+ *
+ * Returns: %TRUE on success
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_stream_transport_send_rtcp_list (GstRTSPStreamTransport * trans,
+ GstBufferList * buffer_list)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gboolean res = FALSE;
+
+ g_return_val_if_fail (GST_IS_BUFFER_LIST (buffer_list), FALSE);
+
+ priv = trans->priv;
+
+ if (priv->send_rtcp_list) {
+ res =
+ priv->send_rtcp_list (buffer_list, priv->transport->interleaved.max,
+ priv->list_user_data);
+ } else if (priv->send_rtcp) {
+ guint n = gst_buffer_list_length (buffer_list), i;
+
+ for (i = 0; i < n; i++) {
+ GstBuffer *buffer = gst_buffer_list_get (buffer_list, i);
+
+ res =
+ priv->send_rtcp (buffer, priv->transport->interleaved.max,
+ priv->user_data);
+ if (!res)
+ break;
+ }
+ }
+
+ if (res)
+ gst_rtsp_stream_transport_keep_alive (trans);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_keep_alive:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Signal the installed keep_alive callback for @trans.
+ */
+void
+gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ priv = trans->priv;
+
+ if (priv->keep_alive)
+ priv->keep_alive (priv->ka_user_data);
+}
+
+/**
+ * gst_rtsp_stream_transport_message_sent:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Signal the installed message_sent / message_sent_full callback for @trans.
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_stream_transport_message_sent (GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ priv = trans->priv;
+
+ if (priv->message_sent_full)
+ priv->message_sent_full (trans, priv->msf_user_data);
+ if (priv->message_sent)
+ priv->message_sent (priv->ms_user_data);
+}
+
+/**
+ * gst_rtsp_stream_transport_recv_data:
+ * @trans: a #GstRTSPStreamTransport
+ * @channel: a channel
+ * @buffer: (transfer full): a #GstBuffer
+ *
+ * Receive @buffer on @channel @trans.
+ *
+ * Returns: a #GstFlowReturn. Returns GST_FLOW_NOT_LINKED when @channel is not
+ * configured in the transport of @trans.
+ */
+GstFlowReturn
+gst_rtsp_stream_transport_recv_data (GstRTSPStreamTransport * trans,
+ guint channel, GstBuffer * buffer)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ const GstRTSPTransport *tr;
+ GstFlowReturn res;
+
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+
+ priv = trans->priv;
+ tr = priv->transport;
+
+ if (tr->interleaved.min == channel) {
+ res = gst_rtsp_stream_recv_rtp (priv->stream, buffer);
+ } else if (tr->interleaved.max == channel) {
+ res = gst_rtsp_stream_recv_rtcp (priv->stream, buffer);
+ } else {
+ res = GST_FLOW_NOT_LINKED;
+ }
+ return res;
+}
+
+static GstClockTime
+get_backlog_item_timestamp (BackLogItem * item)
+{
+ GstClockTime ret = GST_CLOCK_TIME_NONE;
+
+ if (item->buffer) {
+ ret = GST_BUFFER_DTS_OR_PTS (item->buffer);
+ } else if (item->buffer_list) {
+ g_assert (gst_buffer_list_length (item->buffer_list) > 0);
+ ret = GST_BUFFER_DTS_OR_PTS (gst_buffer_list_get (item->buffer_list, 0));
+ }
+
+ return ret;
+}
+
+static GstClockTime
+get_first_backlog_timestamp (GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamTransportPrivate *priv = trans->priv;
+ GstClockTime ret = GST_CLOCK_TIME_NONE;
+ guint i, l;
+
+ l = gst_queue_array_get_length (priv->items);
+
+ for (i = 0; i < l; i++) {
+ BackLogItem *item = (BackLogItem *)
+ gst_queue_array_peek_nth_struct (priv->items, i);
+
+ if (item->is_rtp) {
+ ret = get_backlog_item_timestamp (item);
+ break;
+ }
+ }
+
+ return ret;
+}
+
+/* Not MT-safe, caller should ensure consistent locking (see
+ * gst_rtsp_stream_transport_lock_backlog()). Ownership
+ * of @buffer and @buffer_list is transfered to the transport */
+gboolean
+gst_rtsp_stream_transport_backlog_push (GstRTSPStreamTransport * trans,
+ GstBuffer * buffer, GstBufferList * buffer_list, gboolean is_rtp)
+{
+ gboolean ret = TRUE;
+ BackLogItem item = { 0, };
+ GstClockTime item_timestamp;
+ GstRTSPStreamTransportPrivate *priv;
+
+ priv = trans->priv;
+
+ if (buffer)
+ item.buffer = buffer;
+ if (buffer_list)
+ item.buffer_list = buffer_list;
+ item.is_rtp = is_rtp;
+
+ gst_queue_array_push_tail_struct (priv->items, &item);
+
+ item_timestamp = get_backlog_item_timestamp (&item);
+
+ if (is_rtp && priv->first_rtp_timestamp != GST_CLOCK_TIME_NONE) {
+ GstClockTimeDiff queue_duration;
+
+ g_assert (GST_CLOCK_TIME_IS_VALID (item_timestamp));
+
+ queue_duration = GST_CLOCK_DIFF (priv->first_rtp_timestamp, item_timestamp);
+
+ g_assert (queue_duration >= 0);
+
+ if (queue_duration > MAX_BACKLOG_DURATION &&
+ gst_queue_array_get_length (priv->items) > MAX_BACKLOG_SIZE) {
+ ret = FALSE;
+ }
+ } else if (is_rtp) {
+ priv->first_rtp_timestamp = item_timestamp;
+ }
+
+ return ret;
+}
+
+/* Not MT-safe, caller should ensure consistent locking (see
+ * gst_rtsp_stream_transport_lock_backlog()). Ownership
+ * of @buffer and @buffer_list is transfered back to the caller,
+ * if either of those is NULL the underlying object is unreffed */
+gboolean
+gst_rtsp_stream_transport_backlog_pop (GstRTSPStreamTransport * trans,
+ GstBuffer ** buffer, GstBufferList ** buffer_list, gboolean * is_rtp)
+{
+ BackLogItem *item;
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_val_if_fail (!gst_rtsp_stream_transport_backlog_is_empty (trans),
+ FALSE);
+
+ priv = trans->priv;
+
+ item = (BackLogItem *) gst_queue_array_pop_head_struct (priv->items);
+
+ priv->first_rtp_timestamp = get_first_backlog_timestamp (trans);
+
+ if (buffer)
+ *buffer = item->buffer;
+ else if (item->buffer)
+ gst_buffer_unref (item->buffer);
+
+ if (buffer_list)
+ *buffer_list = item->buffer_list;
+ else if (item->buffer_list)
+ gst_buffer_list_unref (item->buffer_list);
+
+ if (is_rtp)
+ *is_rtp = item->is_rtp;
+
+ return TRUE;
+}
+
+/* Not MT-safe, caller should ensure consistent locking.
+ * See gst_rtsp_stream_transport_lock_backlog() */
+gboolean
+gst_rtsp_stream_transport_backlog_is_empty (GstRTSPStreamTransport * trans)
+{
+ return gst_queue_array_is_empty (trans->priv->items);
+}
+
+/* Not MT-safe, caller should ensure consistent locking.
+ * See gst_rtsp_stream_transport_lock_backlog() */
+void
+gst_rtsp_stream_transport_clear_backlog (GstRTSPStreamTransport * trans)
+{
+ while (!gst_rtsp_stream_transport_backlog_is_empty (trans)) {
+ gst_rtsp_stream_transport_backlog_pop (trans, NULL, NULL, NULL);
+ }
+}
+
+/* Internal API, protects access to the TCP backlog. Safe to
+ * call recursively */
+void
+gst_rtsp_stream_transport_lock_backlog (GstRTSPStreamTransport * trans)
+{
+ g_rec_mutex_lock (&trans->priv->backlog_lock);
+}
+
+/* See gst_rtsp_stream_transport_lock_backlog() */
+void
+gst_rtsp_stream_transport_unlock_backlog (GstRTSPStreamTransport * trans)
+{
+ g_rec_mutex_unlock (&trans->priv->backlog_lock);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/base/base.h>
+#include <gst/rtsp/gstrtsprange.h>
+#include <gst/rtsp/gstrtspurl.h>
+
+#ifndef __GST_RTSP_STREAM_TRANSPORT_H__
+#define __GST_RTSP_STREAM_TRANSPORT_H__
+
+#include "rtsp-server-prelude.h"
+
+G_BEGIN_DECLS
+
+/* types for the media */
+#define GST_TYPE_RTSP_STREAM_TRANSPORT (gst_rtsp_stream_transport_get_type ())
+#define GST_IS_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT))
+#define GST_IS_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT))
+#define GST_RTSP_STREAM_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
+#define GST_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransport))
+#define GST_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
+#define GST_RTSP_STREAM_TRANSPORT_CAST(obj) ((GstRTSPStreamTransport*)(obj))
+#define GST_RTSP_STREAM_TRANSPORT_CLASS_CAST(klass) ((GstRTSPStreamTransportClass*)(klass))
+
+typedef struct _GstRTSPStreamTransport GstRTSPStreamTransport;
+typedef struct _GstRTSPStreamTransportClass GstRTSPStreamTransportClass;
+typedef struct _GstRTSPStreamTransportPrivate GstRTSPStreamTransportPrivate;
+
+#include "rtsp-stream.h"
+
+/**
+ * GstRTSPSendFunc:
+ * @buffer: a #GstBuffer
+ * @channel: a channel
+ * @user_data: user data
+ *
+ * Function registered with gst_rtsp_stream_transport_set_callbacks() and
+ * called when @buffer must be sent on @channel.
+ *
+ * Returns: %TRUE on success
+ */
+typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
+
+/**
+ * GstRTSPSendListFunc:
+ * @buffer_list: a #GstBufferList
+ * @channel: a channel
+ * @user_data: user data
+ *
+ * Function registered with gst_rtsp_stream_transport_set_callbacks() and
+ * called when @buffer_list must be sent on @channel.
+ *
+ * Returns: %TRUE on success
+ *
+ * Since: 1.16
+ */
+typedef gboolean (*GstRTSPSendListFunc) (GstBufferList *buffer_list, guint8 channel, gpointer user_data);
+
+/**
+ * GstRTSPKeepAliveFunc:
+ * @user_data: user data
+ *
+ * Function registered with gst_rtsp_stream_transport_set_keepalive() and called
+ * when the stream is active.
+ */
+typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
+
+/**
+ * GstRTSPMessageSentFunc:
+ * @user_data: user data
+ *
+ * Function registered with gst_rtsp_stream_transport_set_message_sent()
+ * and called when a message has been sent on the transport.
+ */
+typedef void (*GstRTSPMessageSentFunc) (gpointer user_data);
+
+/**
+ * GstRTSPMessageSentFuncFull:
+ * @user_data: user data
+ *
+ * Function registered with gst_rtsp_stream_transport_set_message_sent_full()
+ * and called when a message has been sent on the transport.
+ *
+ * Since: 1.18
+ */
+typedef void (*GstRTSPMessageSentFuncFull) (GstRTSPStreamTransport *trans, gpointer user_data);
+
+/**
+ * GstRTSPStreamTransport:
+ * @parent: parent instance
+ *
+ * A Transport description for a stream
+ */
+struct _GstRTSPStreamTransport {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPStreamTransportPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstRTSPStreamTransportClass {
+ GObjectClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_stream_transport_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPStreamTransport * gst_rtsp_stream_transport_new (GstRTSPStream *stream,
+ GstRTSPTransport *tr);
+
+GST_RTSP_SERVER_API
+GstRTSPStream * gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport *trans);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport *trans,
+ GstRTSPTransport * tr);
+
+GST_RTSP_SERVER_API
+const GstRTSPTransport * gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport *trans);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport *trans,
+ const GstRTSPUrl * url);
+
+GST_RTSP_SERVER_API
+const GstRTSPUrl * gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport *trans);
+
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport *trans,
+ GstClockTime start_time);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport *trans,
+ GstRTSPSendFunc send_rtp,
+ GstRTSPSendFunc send_rtcp,
+ gpointer user_data,
+ GDestroyNotify notify);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_transport_set_list_callbacks (GstRTSPStreamTransport *trans,
+ GstRTSPSendListFunc send_rtp_list,
+ GstRTSPSendListFunc send_rtcp_list,
+ gpointer user_data,
+ GDestroyNotify notify);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport *trans,
+ GstRTSPKeepAliveFunc keep_alive,
+ gpointer user_data,
+ GDestroyNotify notify);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_transport_set_message_sent (GstRTSPStreamTransport *trans,
+ GstRTSPMessageSentFunc message_sent,
+ gpointer user_data,
+ GDestroyNotify notify);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_transport_set_message_sent_full (GstRTSPStreamTransport *trans,
+ GstRTSPMessageSentFuncFull message_sent,
+ gpointer user_data,
+ GDestroyNotify notify);
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport *trans);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_transport_message_sent (GstRTSPStreamTransport *trans);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport *trans,
+ gboolean active);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport *trans,
+ gboolean timedout);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport *trans);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport *trans,
+ GstBuffer *buffer);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport *trans,
+ GstBuffer *buffer);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_transport_send_rtp_list (GstRTSPStreamTransport *trans,
+ GstBufferList *buffer_list);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_transport_send_rtcp_list(GstRTSPStreamTransport *trans,
+ GstBufferList *buffer_list);
+
+GST_RTSP_SERVER_API
+GstFlowReturn gst_rtsp_stream_transport_recv_data (GstRTSPStreamTransport *trans,
+ guint channel, GstBuffer *buffer);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPStreamTransport, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_STREAM_TRANSPORT_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2015 Centricular Ltd
+ * Author: Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-stream
+ * @short_description: A media stream
+ * @see_also: #GstRTSPMedia
+ *
+ * The #GstRTSPStream object manages the data transport for one stream. It
+ * is created from a payloader element and a source pad that produce the RTP
+ * packets for the stream.
+ *
+ * With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
+ * and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
+ *
+ * The #GstRTSPStream will use the configured addresspool, as set with
+ * gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
+ * stream. With gst_rtsp_stream_get_multicast_address() you can get the
+ * configured address.
+ *
+ * With gst_rtsp_stream_get_server_port () you can get the port that the server
+ * will use to receive RTCP. This is the part that the clients will use to send
+ * RTCP to.
+ *
+ * With gst_rtsp_stream_add_transport() destinations can be added where the
+ * stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
+ * the destination again.
+ *
+ * Each #GstRTSPStreamTransport spawns one queue that will serve as a backlog of a
+ * controllable maximum size when the reflux from the TCP connection's backpressure
+ * starts spilling all over.
+ *
+ * Unlike the backlog in rtspconnection, which we have decided should only contain
+ * at most one RTP and one RTCP data message in order to allow control messages to
+ * go through unobstructed, this backlog only consists of data messages, allowing
+ * us to fill it up without concern.
+ *
+ * When multiple TCP transports exist, for example in the context of a shared media,
+ * we only pop samples from our appsinks when at least one of the transports doesn't
+ * experience back pressure: this allows us to pace our sample popping to the speed
+ * of the fastest client.
+ *
+ * When a sample is popped, it is either sent directly on transports that don't
+ * experience backpressure, or queued on the transport's backlog otherwise. Samples
+ * are then popped from that backlog when the transport reports it has sent the message.
+ *
+ * Once the backlog reaches an overly large duration, the transport is dropped as
+ * the client was deemed too slow.
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include <gio/gio.h>
+
+#include <gst/app/gstappsrc.h>
+#include <gst/app/gstappsink.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "rtsp-stream.h"
+#include "rtsp-server-internal.h"
+
+struct _GstRTSPStreamPrivate
+{
+ GMutex lock;
+ guint idx;
+ /* Only one pad is ever set */
+ GstPad *srcpad, *sinkpad;
+ GstElement *payloader;
+ guint buffer_size;
+ GstBin *joined_bin;
+
+ /* TRUE if this stream is running on
+ * the client side of an RTSP link (for RECORD) */
+ gboolean client_side;
+ gchar *control;
+
+ /* TRUE if stream is complete. This means that the receiver and the sender
+ * parts are present in the stream. */
+ gboolean is_complete;
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans allowed_protocols;
+ GstRTSPLowerTrans configured_protocols;
+
+ /* pads on the rtpbin */
+ GstPad *send_rtp_sink;
+ GstPad *recv_rtp_src;
+ GstPad *recv_sink[2];
+ GstPad *send_src[2];
+
+ /* the RTPSession object */
+ GObject *session;
+
+ /* SRTP encoder/decoder */
+ GstElement *srtpenc;
+ GstElement *srtpdec;
+ GHashTable *keys;
+
+ /* for UDP unicast */
+ GstElement *udpsrc_v4[2];
+ GstElement *udpsrc_v6[2];
+ GstElement *udpqueue[2];
+ GstElement *udpsink[2];
+ GSocket *socket_v4[2];
+ GSocket *socket_v6[2];
+
+ /* for UDP multicast */
+ GstElement *mcast_udpsrc_v4[2];
+ GstElement *mcast_udpsrc_v6[2];
+ GstElement *mcast_udpqueue[2];
+ GstElement *mcast_udpsink[2];
+ GSocket *mcast_socket_v4[2];
+ GSocket *mcast_socket_v6[2];
+ GList *mcast_clients;
+
+ /* for TCP transport */
+ GstElement *appsrc[2];
+ GstClockTime appsrc_base_time[2];
+ GstElement *appqueue[2];
+ GstElement *appsink[2];
+
+ GstElement *tee[2];
+ GstElement *funnel[2];
+
+ /* retransmission */
+ GstElement *rtxsend;
+ GstElement *rtxreceive;
+ guint rtx_pt;
+ GstClockTime rtx_time;
+
+ /* rate control */
+ gboolean do_rate_control;
+
+ /* Forward Error Correction with RFC 5109 */
+ GstElement *ulpfec_decoder;
+ GstElement *ulpfec_encoder;
+ guint ulpfec_pt;
+ gboolean ulpfec_enabled;
+ guint ulpfec_percentage;
+
+ /* pool used to manage unicast and multicast addresses */
+ GstRTSPAddressPool *pool;
+
+ /* unicast server addr/port */
+ GstRTSPAddress *server_addr_v4;
+ GstRTSPAddress *server_addr_v6;
+
+ /* multicast addresses */
+ GstRTSPAddress *mcast_addr_v4;
+ GstRTSPAddress *mcast_addr_v6;
+
+ gchar *multicast_iface;
+ guint max_mcast_ttl;
+ gboolean bind_mcast_address;
+
+ /* the caps of the stream */
+ gulong caps_sig;
+ GstCaps *caps;
+
+ /* transports we stream to */
+ guint n_active;
+ GList *transports;
+ guint transports_cookie;
+ GPtrArray *tr_cache;
+ guint tr_cache_cookie;
+ guint n_tcp_transports;
+ gboolean have_buffer[2];
+
+ gint dscp_qos;
+
+ /* Sending logic for TCP */
+ GThread *send_thread;
+ GCond send_cond;
+ GMutex send_lock;
+ /* @send_lock is released when pushing data out, we use
+ * a cookie to decide whether we should wait on @send_cond
+ * before checking the transports' backlogs again
+ */
+ guint send_cookie;
+ /* Used to control shutdown of @send_thread */
+ gboolean continue_sending;
+
+ /* stream blocking */
+ gulong blocked_id[2];
+ gboolean blocking;
+
+ /* current stream postion */
+ GstClockTime position;
+
+ /* pt->caps map for RECORD streams */
+ GHashTable *ptmap;
+
+ GstRTSPPublishClockMode publish_clock_mode;
+ GThreadPool *send_pool;
+
+ /* Used to provide accurate rtpinfo when the stream is blocking */
+ gboolean blocked_buffer;
+ guint32 blocked_seqnum;
+ guint32 blocked_rtptime;
+ GstClockTime blocked_running_time;
+ gint blocked_clock_rate;
+
+ /* Whether we should send and receive RTCP */
+ gboolean enable_rtcp;
+
+ /* blocking early rtcp packets */
+ GstPad *block_early_rtcp_pad;
+ gulong block_early_rtcp_probe;
+ GstPad *block_early_rtcp_pad_ipv6;
+ gulong block_early_rtcp_probe_ipv6;
+};
+
+#define DEFAULT_CONTROL NULL
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+ GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_MAX_MCAST_TTL 255
+#define DEFAULT_BIND_MCAST_ADDRESS FALSE
+#define DEFAULT_DO_RATE_CONTROL TRUE
+#define DEFAULT_ENABLE_RTCP TRUE
+
+enum
+{
+ PROP_0,
+ PROP_CONTROL,
+ PROP_PROFILES,
+ PROP_PROTOCOLS,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_NEW_RTP_ENCODER,
+ SIGNAL_NEW_RTCP_ENCODER,
+ SIGNAL_NEW_RTP_RTCP_DECODER,
+ SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
+#define GST_CAT_DEFAULT rtsp_stream_debug
+
+static GQuark ssrc_stream_map_key;
+
+static void gst_rtsp_stream_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_stream_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+
+static void gst_rtsp_stream_finalize (GObject * obj);
+
+static gboolean
+update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
+ gboolean add);
+
+static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_stream_get_property;
+ gobject_class->set_property = gst_rtsp_stream_set_property;
+ gobject_class->finalize = gst_rtsp_stream_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_CONTROL,
+ g_param_spec_string ("control", "Control",
+ "The control string for this stream", DEFAULT_CONTROL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
+ g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
+ g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTP_RTCP_DECODER] =
+ g_signal_new ("new-rtp-rtcp-decoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
+
+ ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
+}
+
+static void
+gst_rtsp_stream_init (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = gst_rtsp_stream_get_instance_private (stream);
+
+ GST_DEBUG ("new stream %p", stream);
+
+ stream->priv = priv;
+
+ priv->dscp_qos = -1;
+ priv->control = g_strdup (DEFAULT_CONTROL);
+ priv->profiles = DEFAULT_PROFILES;
+ priv->allowed_protocols = DEFAULT_PROTOCOLS;
+ priv->configured_protocols = 0;
+ priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
+ priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
+ priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
+ priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
+ priv->enable_rtcp = DEFAULT_ENABLE_RTCP;
+
+ g_mutex_init (&priv->lock);
+
+ priv->continue_sending = TRUE;
+ priv->send_cookie = 0;
+ g_cond_init (&priv->send_cond);
+ g_mutex_init (&priv->send_lock);
+
+ priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
+ NULL, (GDestroyNotify) gst_caps_unref);
+ priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
+ (GDestroyNotify) gst_caps_unref);
+ priv->send_pool = NULL;
+ priv->block_early_rtcp_pad = NULL;
+ priv->block_early_rtcp_probe = 0;
+ priv->block_early_rtcp_pad_ipv6 = NULL;
+ priv->block_early_rtcp_probe_ipv6 = 0;
+}
+
+typedef struct _UdpClientAddrInfo UdpClientAddrInfo;
+
+struct _UdpClientAddrInfo
+{
+ gchar *address;
+ guint rtp_port;
+ guint add_count; /* how often this address has been added */
+};
+
+static void
+free_mcast_client (gpointer data)
+{
+ UdpClientAddrInfo *client = data;
+
+ g_free (client->address);
+ g_free (client);
+}
+
+static void
+gst_rtsp_stream_finalize (GObject * obj)
+{
+ GstRTSPStream *stream;
+ GstRTSPStreamPrivate *priv;
+ guint i;
+
+ stream = GST_RTSP_STREAM (obj);
+ priv = stream->priv;
+
+ GST_DEBUG ("finalize stream %p", stream);
+
+ /* we really need to be unjoined now */
+ g_return_if_fail (priv->joined_bin == NULL);
+
+ if (priv->send_pool)
+ g_thread_pool_free (priv->send_pool, TRUE, TRUE);
+ if (priv->mcast_addr_v4)
+ gst_rtsp_address_free (priv->mcast_addr_v4);
+ if (priv->mcast_addr_v6)
+ gst_rtsp_address_free (priv->mcast_addr_v6);
+ if (priv->server_addr_v4)
+ gst_rtsp_address_free (priv->server_addr_v4);
+ if (priv->server_addr_v6)
+ gst_rtsp_address_free (priv->server_addr_v6);
+ if (priv->pool)
+ g_object_unref (priv->pool);
+ if (priv->rtxsend)
+ g_object_unref (priv->rtxsend);
+ if (priv->rtxreceive)
+ g_object_unref (priv->rtxreceive);
+ if (priv->ulpfec_encoder)
+ gst_object_unref (priv->ulpfec_encoder);
+ if (priv->ulpfec_decoder)
+ gst_object_unref (priv->ulpfec_decoder);
+
+ for (i = 0; i < 2; i++) {
+ if (priv->socket_v4[i])
+ g_object_unref (priv->socket_v4[i]);
+ if (priv->socket_v6[i])
+ g_object_unref (priv->socket_v6[i]);
+ if (priv->mcast_socket_v4[i])
+ g_object_unref (priv->mcast_socket_v4[i]);
+ if (priv->mcast_socket_v6[i])
+ g_object_unref (priv->mcast_socket_v6[i]);
+ }
+
+ g_free (priv->multicast_iface);
+ g_list_free_full (priv->mcast_clients, (GDestroyNotify) free_mcast_client);
+
+ gst_object_unref (priv->payloader);
+ if (priv->srcpad)
+ gst_object_unref (priv->srcpad);
+ if (priv->sinkpad)
+ gst_object_unref (priv->sinkpad);
+ g_free (priv->control);
+ g_mutex_clear (&priv->lock);
+
+ g_hash_table_unref (priv->keys);
+ g_hash_table_destroy (priv->ptmap);
+
+ g_mutex_clear (&priv->send_lock);
+ g_cond_clear (&priv->send_cond);
+
+ if (priv->block_early_rtcp_probe != 0) {
+ gst_pad_remove_probe
+ (priv->block_early_rtcp_pad, priv->block_early_rtcp_probe);
+ gst_object_unref (priv->block_early_rtcp_pad);
+ }
+
+ if (priv->block_early_rtcp_probe_ipv6 != 0) {
+ gst_pad_remove_probe
+ (priv->block_early_rtcp_pad_ipv6, priv->block_early_rtcp_probe_ipv6);
+ gst_object_unref (priv->block_early_rtcp_pad_ipv6);
+ }
+
+ G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_stream_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPStream *stream = GST_RTSP_STREAM (object);
+
+ switch (propid) {
+ case PROP_CONTROL:
+ g_value_take_string (value, gst_rtsp_stream_get_control (stream));
+ break;
+ case PROP_PROFILES:
+ g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
+ break;
+ case PROP_PROTOCOLS:
+ g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_stream_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPStream *stream = GST_RTSP_STREAM (object);
+
+ switch (propid) {
+ case PROP_CONTROL:
+ gst_rtsp_stream_set_control (stream, g_value_get_string (value));
+ break;
+ case PROP_PROFILES:
+ gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
+ break;
+ case PROP_PROTOCOLS:
+ gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_stream_new:
+ * @idx: an index
+ * @pad: a #GstPad
+ * @payloader: a #GstElement
+ *
+ * Create a new media stream with index @idx that handles RTP data on
+ * @pad and has a payloader element @payloader if @pad is a source pad
+ * or a depayloader element @payloader if @pad is a sink pad.
+ *
+ * Returns: (transfer full): a new #GstRTSPStream
+ */
+GstRTSPStream *
+gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * pad)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPStream *stream;
+
+ g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
+ g_return_val_if_fail (GST_IS_PAD (pad), NULL);
+
+ stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
+ priv = stream->priv;
+ priv->idx = idx;
+ priv->payloader = gst_object_ref (payloader);
+ if (GST_PAD_IS_SRC (pad))
+ priv->srcpad = gst_object_ref (pad);
+ else
+ priv->sinkpad = gst_object_ref (pad);
+
+ return stream;
+}
+
+/**
+ * gst_rtsp_stream_get_index:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the stream index.
+ *
+ * Return: the stream index.
+ */
+guint
+gst_rtsp_stream_get_index (GstRTSPStream * stream)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ return stream->priv->idx;
+}
+
+/**
+ * gst_rtsp_stream_get_pt:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the stream payload type.
+ *
+ * Return: the stream payload type.
+ */
+guint
+gst_rtsp_stream_get_pt (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint pt;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
+
+ return pt;
+}
+
+/**
+ * gst_rtsp_stream_get_srcpad:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the srcpad associated with @stream.
+ *
+ * Returns: (transfer full) (nullable): the srcpad. Unref after usage.
+ */
+GstPad *
+gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ if (!stream->priv->srcpad)
+ return NULL;
+
+ return gst_object_ref (stream->priv->srcpad);
+}
+
+/**
+ * gst_rtsp_stream_get_sinkpad:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the sinkpad associated with @stream.
+ *
+ * Returns: (transfer full) (nullable): the sinkpad. Unref after usage.
+ */
+GstPad *
+gst_rtsp_stream_get_sinkpad (GstRTSPStream * stream)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ if (!stream->priv->sinkpad)
+ return NULL;
+
+ return gst_object_ref (stream->priv->sinkpad);
+}
+
+/**
+ * gst_rtsp_stream_get_control:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the control string to identify this stream.
+ *
+ * Returns: (transfer full) (nullable): the control string. g_free() after usage.
+ */
+gchar *
+gst_rtsp_stream_get_control (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = g_strdup (priv->control)) == NULL)
+ result = g_strdup_printf ("stream=%u", priv->idx);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_set_control:
+ * @stream: a #GstRTSPStream
+ * @control: (nullable): a control string
+ *
+ * Set the control string in @stream.
+ */
+void
+gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_free (priv->control);
+ priv->control = g_strdup (control);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_has_control:
+ * @stream: a #GstRTSPStream
+ * @control: (nullable): a control string
+ *
+ * Check if @stream has the control string @control.
+ *
+ * Returns: %TRUE is @stream has @control as the control string
+ */
+gboolean
+gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->control)
+ res = (g_strcmp0 (priv->control, control) == 0);
+ else {
+ guint streamid;
+
+ if (sscanf (control, "stream=%u", &streamid) > 0)
+ res = (streamid == priv->idx);
+ else
+ res = FALSE;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_mtu:
+ * @stream: a #GstRTSPStream
+ * @mtu: a new MTU
+ *
+ * Configure the mtu in the payloader of @stream to @mtu.
+ */
+void
+gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ GST_LOG_OBJECT (stream, "set MTU %u", mtu);
+
+ g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
+}
+
+/**
+ * gst_rtsp_stream_get_mtu:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured MTU in the payloader of @stream.
+ *
+ * Returns: the MTU of the payloader.
+ */
+guint
+gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint mtu;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
+
+ return mtu;
+}
+
+/* Update the dscp qos property on the udp sinks */
+static void
+update_dscp_qos (GstRTSPStream * stream, GstElement ** udpsink)
+{
+ GstRTSPStreamPrivate *priv;
+
+ priv = stream->priv;
+
+ if (*udpsink) {
+ g_object_set (G_OBJECT (*udpsink), "qos-dscp", priv->dscp_qos, NULL);
+ }
+}
+
+/**
+ * gst_rtsp_stream_set_dscp_qos:
+ * @stream: a #GstRTSPStream
+ * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
+ *
+ * Configure the dscp qos of the outgoing sockets to @dscp_qos.
+ */
+void
+gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
+
+ if (dscp_qos < -1 || dscp_qos > 63) {
+ GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
+ return;
+ }
+
+ priv->dscp_qos = dscp_qos;
+
+ update_dscp_qos (stream, priv->udpsink);
+}
+
+/**
+ * gst_rtsp_stream_get_dscp_qos:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured DSCP QoS in of the outgoing sockets.
+ *
+ * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
+ */
+gint
+gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ priv = stream->priv;
+
+ return priv->dscp_qos;
+}
+
+/**
+ * gst_rtsp_stream_is_transport_supported:
+ * @stream: a #GstRTSPStream
+ * @transport: (transfer none): a #GstRTSPTransport
+ *
+ * Check if @transport can be handled by stream
+ *
+ * Returns: %TRUE if @transport can be handled by @stream.
+ */
+gboolean
+gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
+ GstRTSPTransport * transport)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (transport != NULL, FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (transport->trans != GST_RTSP_TRANS_RTP)
+ goto unsupported_transmode;
+
+ if (!(transport->profile & priv->profiles))
+ goto unsupported_profile;
+
+ if (!(transport->lower_transport & priv->allowed_protocols))
+ goto unsupported_ltrans;
+
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+ /* ERRORS */
+unsupported_transmode:
+ {
+ GST_DEBUG ("unsupported transport mode %d", transport->trans);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+unsupported_profile:
+ {
+ GST_DEBUG ("unsupported profile %d", transport->profile);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+unsupported_ltrans:
+ {
+ GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_set_profiles:
+ * @stream: a #GstRTSPStream
+ * @profiles: the new profiles
+ *
+ * Configure the allowed profiles for @stream.
+ */
+void
+gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->profiles = profiles;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_profiles:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed profiles of @stream.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->profiles;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_protocols:
+ * @stream: a #GstRTSPStream
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @stream.
+ */
+void
+gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
+ GstRTSPLowerTrans protocols)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->allowed_protocols = protocols;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_protocols:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed protocols of @stream.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPLowerTrans res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
+ GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->allowed_protocols;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_address_pool:
+ * @stream: a #GstRTSPStream
+ * @pool: (transfer none) (nullable): a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @stream.
+ */
+void
+gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
+ GstRTSPAddressPool * pool)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddressPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ GST_LOG_OBJECT (stream, "set address pool %p", pool);
+
+ g_mutex_lock (&priv->lock);
+ if ((old = priv->pool) != pool)
+ priv->pool = pool ? g_object_ref (pool) : NULL;
+ else
+ old = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_stream_get_address_pool:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @stream.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @stream.
+ * g_object_unref() after usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddressPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->pool))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_set_multicast_iface:
+ * @stream: a #GstRTSPStream
+ * @multicast_iface: (transfer none) (nullable): a multicast interface name
+ *
+ * configure @multicast_iface to be used for @stream.
+ */
+void
+gst_rtsp_stream_set_multicast_iface (GstRTSPStream * stream,
+ const gchar * multicast_iface)
+{
+ GstRTSPStreamPrivate *priv;
+ gchar *old;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ GST_LOG_OBJECT (stream, "set multicast iface %s",
+ GST_STR_NULL (multicast_iface));
+
+ g_mutex_lock (&priv->lock);
+ if ((old = priv->multicast_iface) != multicast_iface)
+ priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
+ else
+ old = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_free (old);
+}
+
+/**
+ * gst_rtsp_stream_get_multicast_iface:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the multicast interface used for @stream.
+ *
+ * Returns: (transfer full) (nullable): the multicast interface for @stream.
+ * g_free() after usage.
+ */
+gchar *
+gst_rtsp_stream_get_multicast_iface (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->multicast_iface))
+ result = g_strdup (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_get_multicast_address:
+ * @stream: a #GstRTSPStream
+ * @family: the #GSocketFamily
+ *
+ * Get the multicast address of @stream for @family. The original
+ * #GstRTSPAddress is cached and copy is returned, so freeing the return value
+ * won't release the address from the pool.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
+ * or %NULL when no address could be allocated. gst_rtsp_address_free()
+ * after usage.
+ */
+GstRTSPAddress *
+gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
+ GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddress *result;
+ GstRTSPAddress **addrp;
+ GstRTSPAddressFlags flags;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&stream->priv->lock);
+
+ if (family == G_SOCKET_FAMILY_IPV6) {
+ flags = GST_RTSP_ADDRESS_FLAG_IPV6;
+ addrp = &priv->mcast_addr_v6;
+ } else {
+ flags = GST_RTSP_ADDRESS_FLAG_IPV4;
+ addrp = &priv->mcast_addr_v4;
+ }
+
+ if (*addrp == NULL) {
+ if (priv->pool == NULL)
+ goto no_pool;
+
+ flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
+
+ *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
+ if (*addrp == NULL)
+ goto no_address;
+
+ /* FIXME: Also reserve the same port with unicast ANY address, since that's
+ * where we are going to bind our socket. Probably loop until we find a port
+ * available in both mcast and unicast pools. Maybe GstRTSPAddressPool
+ * should do it for us when both GST_RTSP_ADDRESS_FLAG_MULTICAST and
+ * GST_RTSP_ADDRESS_FLAG_UNICAST are givent. */
+ }
+ result = gst_rtsp_address_copy (*addrp);
+
+ g_mutex_unlock (&stream->priv->lock);
+
+ return result;
+
+ /* ERRORS */
+no_pool:
+ {
+ GST_ERROR_OBJECT (stream, "no address pool specified");
+ g_mutex_unlock (&stream->priv->lock);
+ return NULL;
+ }
+no_address:
+ {
+ GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
+ g_mutex_unlock (&stream->priv->lock);
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_stream_reserve_address:
+ * @stream: a #GstRTSPStream
+ * @address: an address
+ * @port: a port
+ * @n_ports: n_ports
+ * @ttl: a TTL
+ *
+ * Reserve @address and @port as the address and port of @stream. The original
+ * #GstRTSPAddress is cached and copy is returned, so freeing the return value
+ * won't release the address from the pool.
+ *
+ * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
+ * the address could not be reserved. gst_rtsp_address_free() after
+ * usage.
+ */
+GstRTSPAddress *
+gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
+ const gchar * address, guint port, guint n_ports, guint ttl)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddress *result;
+ GInetAddress *addr;
+ GSocketFamily family;
+ GstRTSPAddress **addrp;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (address != NULL, NULL);
+ g_return_val_if_fail (port > 0, NULL);
+ g_return_val_if_fail (n_ports > 0, NULL);
+ g_return_val_if_fail (ttl > 0, NULL);
+
+ priv = stream->priv;
+
+ addr = g_inet_address_new_from_string (address);
+ if (!addr) {
+ GST_ERROR ("failed to get inet addr from %s", address);
+ family = G_SOCKET_FAMILY_IPV4;
+ } else {
+ family = g_inet_address_get_family (addr);
+ g_object_unref (addr);
+ }
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ addrp = &priv->mcast_addr_v6;
+ else
+ addrp = &priv->mcast_addr_v4;
+
+ g_mutex_lock (&priv->lock);
+ if (*addrp == NULL) {
+ GstRTSPAddressPoolResult res;
+
+ if (priv->pool == NULL)
+ goto no_pool;
+
+ res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
+ port, n_ports, ttl, addrp);
+ if (res != GST_RTSP_ADDRESS_POOL_OK)
+ goto no_address;
+
+ /* FIXME: Also reserve the same port with unicast ANY address, since that's
+ * where we are going to bind our socket. */
+ } else {
+ if (g_ascii_strcasecmp ((*addrp)->address, address) ||
+ (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
+ (*addrp)->ttl != ttl)
+ goto different_address;
+ }
+ result = gst_rtsp_address_copy (*addrp);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+
+ /* ERRORS */
+no_pool:
+ {
+ GST_ERROR_OBJECT (stream, "no address pool specified");
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+no_address:
+ {
+ GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
+ address);
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+different_address:
+ {
+ GST_ERROR_OBJECT (stream,
+ "address %s is not the same as %s that was already reserved",
+ address, (*addrp)->address);
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+}
+
+/* must be called with lock */
+static void
+set_socket_for_udpsink (GstElement * udpsink, GSocket * socket,
+ GSocketFamily family)
+{
+ const gchar *multisink_socket;
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ multisink_socket = "socket-v6";
+ else
+ multisink_socket = "socket";
+
+ g_object_set (G_OBJECT (udpsink), multisink_socket, socket, NULL);
+}
+
+/* must be called with lock */
+static void
+set_multicast_socket_for_udpsink (GstElement * udpsink, GSocket * socket,
+ GSocketFamily family, const gchar * multicast_iface,
+ const gchar * addr_str, gint port, gint mcast_ttl)
+{
+ set_socket_for_udpsink (udpsink, socket, family);
+
+ if (multicast_iface) {
+ GST_INFO ("setting multicast-iface %s", multicast_iface);
+ g_object_set (G_OBJECT (udpsink), "multicast-iface", multicast_iface, NULL);
+ }
+
+ if (mcast_ttl > 0) {
+ GST_INFO ("setting ttl-mc %d", mcast_ttl);
+ g_object_set (G_OBJECT (udpsink), "ttl-mc", mcast_ttl, NULL);
+ }
+}
+
+
+/* must be called with lock */
+static void
+set_unicast_socket_for_udpsink (GstElement * udpsink, GSocket * socket,
+ GSocketFamily family)
+{
+ set_socket_for_udpsink (udpsink, socket, family);
+}
+
+static guint16
+get_port_from_socket (GSocket * socket)
+{
+ guint16 port;
+ GSocketAddress *sockaddr;
+ GError *err;
+
+ GST_DEBUG ("socket: %p", socket);
+ sockaddr = g_socket_get_local_address (socket, &err);
+ if (sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (sockaddr)) {
+ g_clear_object (&sockaddr);
+ GST_ERROR ("failed to get sockaddr: %s", err->message);
+ g_error_free (err);
+ return 0;
+ }
+
+ port = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
+ g_object_unref (sockaddr);
+
+ return port;
+}
+
+
+static gboolean
+create_and_configure_udpsink (GstRTSPStream * stream, GstElement ** udpsink,
+ GSocket * socket_v4, GSocket * socket_v6, gboolean multicast,
+ gboolean is_rtp, gint mcast_ttl)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ *udpsink = gst_element_factory_make ("multiudpsink", NULL);
+
+ if (!*udpsink)
+ goto no_udp_protocol;
+
+ /* configure sinks */
+
+ g_object_set (G_OBJECT (*udpsink), "close-socket", FALSE, NULL);
+
+ g_object_set (G_OBJECT (*udpsink), "send-duplicates", FALSE, NULL);
+
+ if (is_rtp)
+ g_object_set (G_OBJECT (*udpsink), "buffer-size", priv->buffer_size, NULL);
+ else
+ g_object_set (G_OBJECT (*udpsink), "sync", FALSE, NULL);
+
+ /* Needs to be async for RECORD streams, otherwise we will never go to
+ * PLAYING because the sinks will wait for data while the udpsrc can't
+ * provide data with timestamps in PAUSED. */
+ if (!is_rtp || priv->sinkpad)
+ g_object_set (G_OBJECT (*udpsink), "async", FALSE, NULL);
+
+ if (multicast) {
+ /* join multicast group when adding clients, so we'll start receiving from it.
+ * We cannot rely on the udpsrc to join the group since its socket is always a
+ * local unicast one. */
+ g_object_set (G_OBJECT (*udpsink), "auto-multicast", TRUE, NULL);
+
+ g_object_set (G_OBJECT (*udpsink), "loop", FALSE, NULL);
+ }
+
+ /* update the dscp qos field in the sinks */
+ update_dscp_qos (stream, udpsink);
+
+ if (priv->server_addr_v4) {
+ GST_DEBUG_OBJECT (stream, "udp IPv4, configure udpsinks");
+ set_unicast_socket_for_udpsink (*udpsink, socket_v4, G_SOCKET_FAMILY_IPV4);
+ }
+
+ if (priv->server_addr_v6) {
+ GST_DEBUG_OBJECT (stream, "udp IPv6, configure udpsinks");
+ set_unicast_socket_for_udpsink (*udpsink, socket_v6, G_SOCKET_FAMILY_IPV6);
+ }
+
+ if (multicast) {
+ gint port;
+ if (priv->mcast_addr_v4) {
+ GST_DEBUG_OBJECT (stream, "mcast IPv4, configure udpsinks");
+ port = get_port_from_socket (socket_v4);
+ if (!port)
+ goto get_port_failed;
+ set_multicast_socket_for_udpsink (*udpsink, socket_v4,
+ G_SOCKET_FAMILY_IPV4, priv->multicast_iface,
+ priv->mcast_addr_v4->address, port, mcast_ttl);
+ }
+
+ if (priv->mcast_addr_v6) {
+ GST_DEBUG_OBJECT (stream, "mcast IPv6, configure udpsinks");
+ port = get_port_from_socket (socket_v6);
+ if (!port)
+ goto get_port_failed;
+ set_multicast_socket_for_udpsink (*udpsink, socket_v6,
+ G_SOCKET_FAMILY_IPV6, priv->multicast_iface,
+ priv->mcast_addr_v6->address, port, mcast_ttl);
+ }
+
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+no_udp_protocol:
+ {
+ GST_ERROR_OBJECT (stream, "failed to create udpsink element");
+ return FALSE;
+ }
+get_port_failed:
+ {
+ GST_ERROR_OBJECT (stream, "failed to get udp port");
+ return FALSE;
+ }
+}
+
+/* must be called with lock */
+static gboolean
+create_and_configure_udpsource (GstElement ** udpsrc, GSocket * socket)
+{
+ GstStateChangeReturn ret;
+
+ g_assert (socket != NULL);
+
+ *udpsrc = gst_element_factory_make ("udpsrc", NULL);
+ if (*udpsrc == NULL)
+ goto error;
+
+ g_object_set (G_OBJECT (*udpsrc), "socket", socket, NULL);
+
+ /* The udpsrc cannot do the join because its socket is always a local unicast
+ * one. The udpsink sharing the same socket will do it for us. */
+ g_object_set (G_OBJECT (*udpsrc), "auto-multicast", FALSE, NULL);
+
+ g_object_set (G_OBJECT (*udpsrc), "loop", FALSE, NULL);
+
+ g_object_set (G_OBJECT (*udpsrc), "close-socket", FALSE, NULL);
+
+ ret = gst_element_set_state (*udpsrc, GST_STATE_READY);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto error;
+
+ return TRUE;
+
+ /* ERRORS */
+error:
+ {
+ if (*udpsrc) {
+ gst_element_set_state (*udpsrc, GST_STATE_NULL);
+ g_clear_object (udpsrc);
+ }
+ return FALSE;
+ }
+}
+
+static gboolean
+alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
+ GSocket * socket_out[2], GstRTSPAddress ** server_addr_out,
+ gboolean multicast, GstRTSPTransport * ct, gboolean use_transport_settings)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GSocket *rtp_socket = NULL;
+ GSocket *rtcp_socket = NULL;
+ gint tmp_rtp, tmp_rtcp;
+ guint count;
+ GList *rejected_addresses = NULL;
+ GstRTSPAddress *addr = NULL;
+ GInetAddress *inetaddr = NULL;
+ GSocketAddress *rtp_sockaddr = NULL;
+ GSocketAddress *rtcp_sockaddr = NULL;
+ GstRTSPAddressPool *pool;
+ gboolean transport_settings_defined = FALSE;
+
+ pool = priv->pool;
+ count = 0;
+
+ /* Start with random port */
+ tmp_rtp = 0;
+ tmp_rtcp = 0;
+
+ if (use_transport_settings) {
+ if (!multicast)
+ goto no_mcast;
+
+ if (ct == NULL)
+ goto no_transport;
+
+ /* multicast and transport specific case */
+ if (ct->destination != NULL) {
+ tmp_rtp = ct->port.min;
+ tmp_rtcp = ct->port.max;
+
+ /* check if the provided address is a multicast address */
+ inetaddr = g_inet_address_new_from_string (ct->destination);
+ if (inetaddr == NULL)
+ goto destination_error;
+ if (!g_inet_address_get_is_multicast (inetaddr))
+ goto destination_no_mcast;
+
+
+ if (!priv->bind_mcast_address) {
+ g_clear_object (&inetaddr);
+ inetaddr = g_inet_address_new_any (family);
+ }
+
+ GST_DEBUG_OBJECT (stream, "use transport settings");
+ transport_settings_defined = TRUE;
+ }
+ }
+
+ if (priv->enable_rtcp) {
+ rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ if (!rtcp_socket)
+ goto no_udp_protocol;
+ g_socket_set_multicast_loopback (rtcp_socket, FALSE);
+ }
+
+ /* try to allocate UDP ports, the RTP port should be an even
+ * number and the RTCP port (if enabled) should be the next (uneven) port */
+again:
+
+ if (rtp_socket == NULL) {
+ rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ if (!rtp_socket)
+ goto no_udp_protocol;
+ g_socket_set_multicast_loopback (rtp_socket, FALSE);
+ }
+
+ if (!transport_settings_defined) {
+ if ((pool && gst_rtsp_address_pool_has_unicast_addresses (pool))
+ || multicast) {
+ GstRTSPAddressFlags flags;
+
+ if (addr)
+ rejected_addresses = g_list_prepend (rejected_addresses, addr);
+
+ if (!pool)
+ goto no_pool;
+
+ flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT;
+ if (multicast)
+ flags |= GST_RTSP_ADDRESS_FLAG_MULTICAST;
+ else
+ flags |= GST_RTSP_ADDRESS_FLAG_UNICAST;
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
+ else
+ flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
+
+ if (*server_addr_out)
+ addr = *server_addr_out;
+ else
+ addr = gst_rtsp_address_pool_acquire_address (pool, flags,
+ priv->enable_rtcp ? 2 : 1);
+
+ if (addr == NULL)
+ goto no_address;
+
+ tmp_rtp = addr->port;
+
+ g_clear_object (&inetaddr);
+ /* FIXME: Does it really work with the IP_MULTICAST_ALL socket option and
+ * socket control message set in udpsrc? */
+ if (priv->bind_mcast_address || !multicast)
+ inetaddr = g_inet_address_new_from_string (addr->address);
+ else
+ inetaddr = g_inet_address_new_any (family);
+ } else {
+ if (tmp_rtp != 0) {
+ tmp_rtp += 2;
+ if (++count > 20)
+ goto no_ports;
+ }
+
+ if (inetaddr == NULL)
+ inetaddr = g_inet_address_new_any (family);
+ }
+ }
+
+ rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
+ if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
+ GST_DEBUG_OBJECT (stream, "rtp bind() failed, will try again");
+ g_object_unref (rtp_sockaddr);
+ if (transport_settings_defined)
+ goto transport_settings_error;
+ goto again;
+ }
+ g_object_unref (rtp_sockaddr);
+
+ rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
+ if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
+ g_clear_object (&rtp_sockaddr);
+ goto socket_error;
+ }
+
+ if (!transport_settings_defined) {
+ tmp_rtp =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
+
+ /* check if port is even. RFC 3550 encorages the use of an even/odd port
+ * pair, however it's not a strict requirement so this check is not done
+ * for the client selected ports. */
+ if ((tmp_rtp & 1) != 0) {
+ /* port not even, close and allocate another */
+ tmp_rtp++;
+ g_object_unref (rtp_sockaddr);
+ g_clear_object (&rtp_socket);
+ goto again;
+ }
+ }
+ g_object_unref (rtp_sockaddr);
+
+ /* set port */
+ if (priv->enable_rtcp) {
+ tmp_rtcp = tmp_rtp + 1;
+
+ rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
+ if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
+ GST_DEBUG_OBJECT (stream, "rctp bind() failed, will try again");
+ g_object_unref (rtcp_sockaddr);
+ g_clear_object (&rtp_socket);
+ if (transport_settings_defined)
+ goto transport_settings_error;
+ goto again;
+ }
+ g_object_unref (rtcp_sockaddr);
+ }
+
+ if (!addr) {
+ addr = g_slice_new0 (GstRTSPAddress);
+ addr->port = tmp_rtp;
+ addr->n_ports = 2;
+ if (transport_settings_defined)
+ addr->address = g_strdup (ct->destination);
+ else
+ addr->address = g_inet_address_to_string (inetaddr);
+ addr->ttl = ct->ttl;
+ }
+
+ g_clear_object (&inetaddr);
+
+ if (multicast && (ct->ttl > 0) && (ct->ttl <= priv->max_mcast_ttl)) {
+ GST_DEBUG ("setting mcast ttl to %d", ct->ttl);
+ g_socket_set_multicast_ttl (rtp_socket, ct->ttl);
+ if (rtcp_socket)
+ g_socket_set_multicast_ttl (rtcp_socket, ct->ttl);
+ }
+
+ socket_out[0] = rtp_socket;
+ socket_out[1] = rtcp_socket;
+ *server_addr_out = addr;
+
+ if (priv->enable_rtcp) {
+ GST_DEBUG_OBJECT (stream, "allocated address: %s and ports: %d, %d",
+ addr->address, tmp_rtp, tmp_rtcp);
+ } else {
+ GST_DEBUG_OBJECT (stream, "allocated address: %s and port: %d",
+ addr->address, tmp_rtp);
+ }
+
+ g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
+
+ return TRUE;
+
+ /* ERRORS */
+no_mcast:
+ {
+ GST_ERROR_OBJECT (stream, "failed to allocate UDP ports: wrong transport");
+ goto cleanup;
+ }
+no_transport:
+ {
+ GST_ERROR_OBJECT (stream, "failed to allocate UDP ports: no transport");
+ goto cleanup;
+ }
+destination_error:
+ {
+ GST_ERROR_OBJECT (stream,
+ "failed to allocate UDP ports: destination error");
+ goto cleanup;
+ }
+destination_no_mcast:
+ {
+ GST_ERROR_OBJECT (stream,
+ "failed to allocate UDP ports: destination not multicast address");
+ goto cleanup;
+ }
+no_udp_protocol:
+ {
+ GST_WARNING_OBJECT (stream, "failed to allocate UDP ports: protocol error");
+ goto cleanup;
+ }
+no_pool:
+ {
+ GST_WARNING_OBJECT (stream,
+ "failed to allocate UDP ports: no address pool specified");
+ goto cleanup;
+ }
+no_address:
+ {
+ GST_WARNING_OBJECT (stream, "failed to acquire address from pool");
+ goto cleanup;
+ }
+no_ports:
+ {
+ GST_WARNING_OBJECT (stream, "failed to allocate UDP ports: no ports");
+ goto cleanup;
+ }
+transport_settings_error:
+ {
+ GST_ERROR_OBJECT (stream,
+ "failed to allocate UDP ports with requested transport settings");
+ goto cleanup;
+ }
+socket_error:
+ {
+ GST_WARNING_OBJECT (stream, "failed to allocate UDP ports: socket error");
+ goto cleanup;
+ }
+cleanup:
+ {
+ if (inetaddr)
+ g_object_unref (inetaddr);
+ g_list_free_full (rejected_addresses,
+ (GDestroyNotify) gst_rtsp_address_free);
+ if (addr)
+ gst_rtsp_address_free (addr);
+ if (rtp_socket)
+ g_object_unref (rtp_socket);
+ if (rtcp_socket)
+ g_object_unref (rtcp_socket);
+ return FALSE;
+ }
+}
+
+/* must be called with lock */
+static gboolean
+add_mcast_client_addr (GstRTSPStream * stream, const gchar * destination,
+ guint rtp_port, guint rtcp_port)
+{
+ GstRTSPStreamPrivate *priv;
+ GList *walk;
+ UdpClientAddrInfo *client;
+ GInetAddress *inet;
+
+ priv = stream->priv;
+
+ if (destination == NULL)
+ return FALSE;
+
+ inet = g_inet_address_new_from_string (destination);
+ if (inet == NULL)
+ goto invalid_address;
+
+ if (!g_inet_address_get_is_multicast (inet)) {
+ g_object_unref (inet);
+ goto invalid_address;
+ }
+ g_object_unref (inet);
+
+ for (walk = priv->mcast_clients; walk; walk = g_list_next (walk)) {
+ UdpClientAddrInfo *cli = walk->data;
+
+ if ((g_strcmp0 (cli->address, destination) == 0) &&
+ (cli->rtp_port == rtp_port)) {
+ GST_DEBUG ("requested destination already exists: %s:%u-%u",
+ destination, rtp_port, rtcp_port);
+ cli->add_count++;
+ return TRUE;
+ }
+ }
+
+ client = g_new0 (UdpClientAddrInfo, 1);
+ client->address = g_strdup (destination);
+ client->rtp_port = rtp_port;
+ client->add_count = 1;
+ priv->mcast_clients = g_list_prepend (priv->mcast_clients, client);
+
+ GST_DEBUG ("added mcast client %s:%u-%u", destination, rtp_port, rtcp_port);
+
+ return TRUE;
+
+invalid_address:
+ {
+ GST_WARNING_OBJECT (stream, "Multicast address is invalid: %s",
+ destination);
+ return FALSE;
+ }
+}
+
+/* must be called with lock */
+static gboolean
+remove_mcast_client_addr (GstRTSPStream * stream, const gchar * destination,
+ guint rtp_port, guint rtcp_port)
+{
+ GstRTSPStreamPrivate *priv;
+ GList *walk;
+
+ priv = stream->priv;
+
+ if (destination == NULL)
+ goto no_destination;
+
+ for (walk = priv->mcast_clients; walk; walk = g_list_next (walk)) {
+ UdpClientAddrInfo *cli = walk->data;
+
+ if ((g_strcmp0 (cli->address, destination) == 0) &&
+ (cli->rtp_port == rtp_port)) {
+ cli->add_count--;
+
+ if (!cli->add_count) {
+ priv->mcast_clients = g_list_remove (priv->mcast_clients, cli);
+ free_mcast_client (cli);
+ }
+ return TRUE;
+ }
+ }
+
+ GST_WARNING_OBJECT (stream, "Address not found");
+ return FALSE;
+
+no_destination:
+ {
+ GST_WARNING_OBJECT (stream, "No destination has been provided");
+ return FALSE;
+ }
+}
+
+
+/**
+ * gst_rtsp_stream_allocate_udp_sockets:
+ * @stream: a #GstRTSPStream
+ * @family: protocol family
+ * @transport: transport method
+ * @use_client_settings: Whether to use client settings or not
+ *
+ * Allocates RTP and RTCP ports.
+ *
+ * Returns: %TRUE if the RTP and RTCP sockets have been succeccully allocated.
+ */
+gboolean
+gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream,
+ GSocketFamily family, GstRTSPTransport * ct,
+ gboolean use_transport_settings)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean ret = FALSE;
+ GstRTSPLowerTrans transport;
+ gboolean allocated = FALSE;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (ct != NULL, FALSE);
+ priv = stream->priv;
+
+ transport = ct->lower_transport;
+
+ g_mutex_lock (&priv->lock);
+
+ if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ if (family == G_SOCKET_FAMILY_IPV4 && priv->mcast_socket_v4[0])
+ allocated = TRUE;
+ else if (family == G_SOCKET_FAMILY_IPV6 && priv->mcast_socket_v6[0])
+ allocated = TRUE;
+ } else if (transport == GST_RTSP_LOWER_TRANS_UDP) {
+ if (family == G_SOCKET_FAMILY_IPV4 && priv->socket_v4[0])
+ allocated = TRUE;
+ else if (family == G_SOCKET_FAMILY_IPV6 && priv->socket_v6[0])
+ allocated = TRUE;
+ }
+
+ if (allocated) {
+ GST_DEBUG_OBJECT (stream, "Allocated already");
+ g_mutex_unlock (&priv->lock);
+ return TRUE;
+ }
+
+ if (family == G_SOCKET_FAMILY_IPV4) {
+ /* IPv4 */
+ if (transport == GST_RTSP_LOWER_TRANS_UDP) {
+ /* UDP unicast */
+ GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_UDP, ipv4");
+ ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
+ priv->socket_v4, &priv->server_addr_v4, FALSE, ct, FALSE);
+ } else {
+ /* multicast */
+ GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_MCAST_UDP, ipv4");
+ ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
+ priv->mcast_socket_v4, &priv->mcast_addr_v4, TRUE, ct,
+ use_transport_settings);
+ }
+ } else {
+ /* IPv6 */
+ if (transport == GST_RTSP_LOWER_TRANS_UDP) {
+ /* unicast */
+ GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_UDP, ipv6");
+ ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
+ priv->socket_v6, &priv->server_addr_v6, FALSE, ct, FALSE);
+
+ } else {
+ /* multicast */
+ GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_MCAST_UDP, ipv6");
+ ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
+ priv->mcast_socket_v6, &priv->mcast_addr_v6, TRUE, ct,
+ use_transport_settings);
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_set_client_side:
+ * @stream: a #GstRTSPStream
+ * @client_side: TRUE if this #GstRTSPStream is running on the 'client' side of
+ * an RTSP connection.
+ *
+ * Sets the #GstRTSPStream as a 'client side' stream - used for sending
+ * streams to an RTSP server via RECORD. This has the practical effect
+ * of changing which UDP port numbers are used when setting up the local
+ * side of the stream sending to be either the 'server' or 'client' pair
+ * of a configured UDP transport.
+ */
+void
+gst_rtsp_stream_set_client_side (GstRTSPStream * stream, gboolean client_side)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ priv->client_side = client_side;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_is_client_side:
+ * @stream: a #GstRTSPStream
+ *
+ * See gst_rtsp_stream_set_client_side()
+ *
+ * Returns: TRUE if this #GstRTSPStream is client-side.
+ */
+gboolean
+gst_rtsp_stream_is_client_side (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ ret = priv->client_side;
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_get_server_port:
+ * @stream: a #GstRTSPStream
+ * @server_port: (out): result server port
+ * @family: the port family to get
+ *
+ * Fill @server_port with the port pair used by the server. This function can
+ * only be called when @stream has been joined.
+ */
+void
+gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
+ GstRTSPRange * server_port, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ priv = stream->priv;
+ g_return_if_fail (priv->joined_bin != NULL);
+
+ if (server_port) {
+ server_port->min = 0;
+ server_port->max = 0;
+ }
+
+ g_mutex_lock (&priv->lock);
+ if (family == G_SOCKET_FAMILY_IPV4) {
+ if (server_port && priv->server_addr_v4) {
+ server_port->min = priv->server_addr_v4->port;
+ if (priv->enable_rtcp) {
+ server_port->max =
+ priv->server_addr_v4->port + priv->server_addr_v4->n_ports - 1;
+ }
+ }
+ } else {
+ if (server_port && priv->server_addr_v6) {
+ server_port->min = priv->server_addr_v6->port;
+ if (priv->enable_rtcp) {
+ server_port->max =
+ priv->server_addr_v6->port + priv->server_addr_v6->n_ports - 1;
+ }
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_rtpsession:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the RTP session of this stream.
+ *
+ * Returns: (transfer full): The RTP session of this stream. Unref after usage.
+ */
+GObject *
+gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GObject *session;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((session = priv->session))
+ g_object_ref (session);
+ g_mutex_unlock (&priv->lock);
+
+ return session;
+}
+
+/**
+ * gst_rtsp_stream_get_srtp_encoder:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the SRTP encoder for this stream.
+ *
+ * Returns: (transfer full): The SRTP encoder for this stream. Unref after usage.
+ */
+GstElement *
+gst_rtsp_stream_get_srtp_encoder (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstElement *encoder;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((encoder = priv->srtpenc))
+ g_object_ref (encoder);
+ g_mutex_unlock (&priv->lock);
+
+ return encoder;
+}
+
+/**
+ * gst_rtsp_stream_get_ssrc:
+ * @stream: a #GstRTSPStream
+ * @ssrc: (out): result ssrc
+ *
+ * Get the SSRC used by the RTP session of this stream. This function can only
+ * be called when @stream has been joined.
+ */
+void
+gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ priv = stream->priv;
+ g_return_if_fail (priv->joined_bin != NULL);
+
+ g_mutex_lock (&priv->lock);
+ if (ssrc && priv->session)
+ g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_set_retransmission_time:
+ * @stream: a #GstRTSPStream
+ * @time: a #GstClockTime
+ *
+ * Set the amount of time to store retransmission packets.
+ */
+void
+gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
+ GstClockTime time)
+{
+ GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
+
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->rtx_time = time;
+ if (stream->priv->rtxsend)
+ g_object_set (stream->priv->rtxsend, "max-size-time",
+ GST_TIME_AS_MSECONDS (time), NULL);
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_retransmission_time:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the amount of time to store retransmission data.
+ *
+ * Returns: the amount of time to store retransmission data.
+ */
+GstClockTime
+gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
+{
+ GstClockTime ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+ g_mutex_lock (&stream->priv->lock);
+ ret = stream->priv->rtx_time;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_set_retransmission_pt:
+ * @stream: a #GstRTSPStream
+ * @rtx_pt: a #guint
+ *
+ * Set the payload type (pt) for retransmission of this stream.
+ */
+void
+gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
+{
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
+
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->rtx_pt = rtx_pt;
+ if (stream->priv->rtxsend) {
+ guint pt = gst_rtsp_stream_get_pt (stream);
+ gchar *pt_s = g_strdup_printf ("%d", pt);
+ GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
+ pt_s, G_TYPE_UINT, rtx_pt, NULL);
+ g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
+ g_free (pt_s);
+ gst_structure_free (rtx_pt_map);
+ }
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_retransmission_pt:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the payload-type used for retransmission of this stream
+ *
+ * Returns: The retransmission PT.
+ */
+guint
+gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
+{
+ guint rtx_pt;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+ g_mutex_lock (&stream->priv->lock);
+ rtx_pt = stream->priv->rtx_pt;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return rtx_pt;
+}
+
+/**
+ * gst_rtsp_stream_set_buffer_size:
+ * @stream: a #GstRTSPStream
+ * @size: the buffer size
+ *
+ * Set the size of the UDP transmission buffer (in bytes)
+ * Needs to be set before the stream is joined to a bin.
+ *
+ * Since: 1.6
+ */
+void
+gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
+{
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->buffer_size = size;
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_buffer_size:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the size of the UDP transmission buffer (in bytes)
+ *
+ * Returns: the size of the UDP TX buffer
+ *
+ * Since: 1.6
+ */
+guint
+gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
+{
+ guint buffer_size;
+
+ g_mutex_lock (&stream->priv->lock);
+ buffer_size = stream->priv->buffer_size;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return buffer_size;
+}
+
+/**
+ * gst_rtsp_stream_set_max_mcast_ttl:
+ * @stream: a #GstRTSPStream
+ * @ttl: the new multicast ttl value
+ *
+ * Set the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: %TRUE if the requested ttl has been set successfully.
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_stream_set_max_mcast_ttl (GstRTSPStream * stream, guint ttl)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ g_mutex_lock (&stream->priv->lock);
+ if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
+ GST_WARNING_OBJECT (stream, "The reqested mcast TTL value is not valid.");
+ g_mutex_unlock (&stream->priv->lock);
+ return FALSE;
+ }
+ stream->priv->max_mcast_ttl = ttl;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_get_max_mcast_ttl:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Since: 1.16
+ */
+guint
+gst_rtsp_stream_get_max_mcast_ttl (GstRTSPStream * stream)
+{
+ guint ttl;
+
+ g_mutex_lock (&stream->priv->lock);
+ ttl = stream->priv->max_mcast_ttl;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return ttl;
+}
+
+/**
+ * gst_rtsp_stream_verify_mcast_ttl:
+ * @stream: a #GstRTSPStream
+ * @ttl: a requested multicast ttl
+ *
+ * Check if the requested multicast ttl value is allowed.
+ *
+ * Returns: TRUE if the requested ttl value is allowed.
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_stream_verify_mcast_ttl (GstRTSPStream * stream, guint ttl)
+{
+ gboolean res = FALSE;
+
+ g_mutex_lock (&stream->priv->lock);
+ if ((ttl > 0) && (ttl <= stream->priv->max_mcast_ttl))
+ res = TRUE;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_bind_mcast_address:
+ * @stream: a #GstRTSPStream,
+ * @bind_mcast_addr: the new value
+ *
+ * Decide whether the multicast socket should be bound to a multicast address or
+ * INADDR_ANY.
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_stream_set_bind_mcast_address (GstRTSPStream * stream,
+ gboolean bind_mcast_addr)
+{
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->bind_mcast_address = bind_mcast_addr;
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_is_bind_mcast_address:
+ * @stream: a #GstRTSPStream
+ *
+ * Check if multicast sockets are configured to be bound to multicast addresses.
+ *
+ * Returns: %TRUE if multicast sockets are configured to be bound to multicast addresses.
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_stream_is_bind_mcast_address (GstRTSPStream * stream)
+{
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ g_mutex_lock (&stream->priv->lock);
+ result = stream->priv->bind_mcast_address;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return result;
+}
+
+void
+gst_rtsp_stream_set_enable_rtcp (GstRTSPStream * stream, gboolean enable)
+{
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->enable_rtcp = enable;
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/* executed from streaming thread */
+static void
+caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstCaps *newcaps, *oldcaps;
+
+ newcaps = gst_pad_get_current_caps (pad);
+
+ GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
+ newcaps);
+
+ g_mutex_lock (&priv->lock);
+ oldcaps = priv->caps;
+ priv->caps = newcaps;
+ g_mutex_unlock (&priv->lock);
+
+ if (oldcaps)
+ gst_caps_unref (oldcaps);
+}
+
+static void
+dump_structure (const GstStructure * s)
+{
+ gchar *sstr;
+
+ sstr = gst_structure_to_string (s);
+ GST_INFO ("structure: %s", sstr);
+ g_free (sstr);
+}
+
+static GstRTSPStreamTransport *
+find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GList *walk;
+ GstRTSPStreamTransport *result = NULL;
+ const gchar *tmp;
+ gchar *dest;
+ guint port;
+
+ if (rtcp_from == NULL)
+ return NULL;
+
+ tmp = g_strrstr (rtcp_from, ":");
+ if (tmp == NULL)
+ return NULL;
+
+ port = atoi (tmp + 1);
+ dest = g_strndup (rtcp_from, tmp - rtcp_from);
+
+ g_mutex_lock (&priv->lock);
+ GST_INFO ("finding %s:%d in %d transports", dest, port,
+ g_list_length (priv->transports));
+
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *trans = walk->data;
+ const GstRTSPTransport *tr;
+ gint min, max;
+
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+
+ if (priv->client_side) {
+ /* In client side mode the 'destination' is the RTSP server, so send
+ * to those ports */
+ min = tr->server_port.min;
+ max = tr->server_port.max;
+ } else {
+ min = tr->client_port.min;
+ max = tr->client_port.max;
+ }
+
+ if ((g_ascii_strcasecmp (tr->destination, dest) == 0) &&
+ (min == port || max == port)) {
+ result = trans;
+ break;
+ }
+ }
+ if (result)
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ g_free (dest);
+
+ return result;
+}
+
+static GstRTSPStreamTransport *
+check_transport (GObject * source, GstRTSPStream * stream)
+{
+ GstStructure *stats;
+ GstRTSPStreamTransport *trans;
+
+ /* see if we have a stream to match with the origin of the RTCP packet */
+ trans = g_object_get_qdata (source, ssrc_stream_map_key);
+ if (trans == NULL) {
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ const gchar *rtcp_from;
+
+ dump_structure (stats);
+
+ rtcp_from = gst_structure_get_string (stats, "rtcp-from");
+ if ((trans = find_transport (stream, rtcp_from))) {
+ GST_INFO ("%p: found transport %p for source %p", stream, trans,
+ source);
+ g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
+ g_object_unref);
+ }
+ gst_structure_free (stats);
+ }
+ }
+ return trans;
+}
+
+
+static void
+on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ GST_INFO ("%p: new source %p", stream, source);
+
+ trans = check_transport (source, stream);
+
+ if (trans)
+ GST_INFO ("%p: source %p for transport %p", stream, source, trans);
+}
+
+static void
+on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GST_INFO ("%p: new SDES %p", stream, source);
+}
+
+static void
+on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ trans = check_transport (source, stream);
+
+ if (trans) {
+ GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
+ gst_rtsp_stream_transport_keep_alive (trans);
+ }
+#ifdef DUMP_STATS
+ {
+ GstStructure *stats;
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ dump_structure (stats);
+ gst_structure_free (stats);
+ }
+ }
+#endif
+}
+
+static void
+on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GST_INFO ("%p: source %p bye", stream, source);
+}
+
+static void
+on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ GST_INFO ("%p: source %p bye timeout", stream, source);
+
+ if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
+ gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+ g_object_set_qdata (source, ssrc_stream_map_key, NULL);
+ }
+}
+
+static void
+on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ GST_INFO ("%p: source %p timeout", stream, source);
+
+ if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
+ gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+ g_object_set_qdata (source, ssrc_stream_map_key, NULL);
+ }
+}
+
+static void
+on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GST_INFO ("%p: new sender source %p", stream, source);
+#ifndef DUMP_STATS
+ {
+ GstStructure *stats;
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ dump_structure (stats);
+ gst_structure_free (stats);
+ }
+ }
+#endif
+}
+
+static void
+on_sender_ssrc_active (GObject * session, GObject * source,
+ GstRTSPStream * stream)
+{
+#ifndef DUMP_STATS
+ {
+ GstStructure *stats;
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ dump_structure (stats);
+ gst_structure_free (stats);
+ }
+ }
+#endif
+}
+
+static void
+clear_tr_cache (GstRTSPStreamPrivate * priv)
+{
+ if (priv->tr_cache)
+ g_ptr_array_unref (priv->tr_cache);
+ priv->tr_cache = NULL;
+}
+
+/* With lock taken */
+static gboolean
+any_transport_ready (GstRTSPStream * stream, gboolean is_rtp)
+{
+ gboolean ret = TRUE;
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GPtrArray *transports;
+ gint index;
+
+ transports = priv->tr_cache;
+
+ if (!transports)
+ goto done;
+
+ for (index = 0; index < transports->len; index++) {
+ GstRTSPStreamTransport *tr = g_ptr_array_index (transports, index);
+ if (!gst_rtsp_stream_transport_check_back_pressure (tr, is_rtp)) {
+ ret = TRUE;
+ break;
+ } else {
+ ret = FALSE;
+ }
+ }
+
+done:
+ return ret;
+}
+
+/* Must be called *without* priv->lock */
+static gboolean
+push_data (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
+ GstBuffer * buffer, GstBufferList * buffer_list, gboolean is_rtp)
+{
+ gboolean send_ret = TRUE;
+
+ if (is_rtp) {
+ if (buffer)
+ send_ret = gst_rtsp_stream_transport_send_rtp (trans, buffer);
+ if (buffer_list)
+ send_ret = gst_rtsp_stream_transport_send_rtp_list (trans, buffer_list);
+ } else {
+ if (buffer)
+ send_ret = gst_rtsp_stream_transport_send_rtcp (trans, buffer);
+ if (buffer_list)
+ send_ret = gst_rtsp_stream_transport_send_rtcp_list (trans, buffer_list);
+ }
+
+ return send_ret;
+}
+
+/* With priv->lock */
+static void
+ensure_cached_transports (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GList *walk;
+
+ if (priv->tr_cache_cookie != priv->transports_cookie) {
+ clear_tr_cache (priv);
+ priv->tr_cache =
+ g_ptr_array_new_full (priv->n_tcp_transports, g_object_unref);
+
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+ const GstRTSPTransport *t = gst_rtsp_stream_transport_get_transport (tr);
+
+ if (t->lower_transport != GST_RTSP_LOWER_TRANS_TCP)
+ continue;
+
+ g_ptr_array_add (priv->tr_cache, g_object_ref (tr));
+ }
+ priv->tr_cache_cookie = priv->transports_cookie;
+ }
+}
+
+/* Must be called *without* priv->lock */
+static void
+check_transport_backlog (GstRTSPStream * stream, GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ gboolean send_ret = TRUE;
+
+ gst_rtsp_stream_transport_lock_backlog (trans);
+
+ if (!gst_rtsp_stream_transport_backlog_is_empty (trans)) {
+ GstBuffer *buffer;
+ GstBufferList *buffer_list;
+ gboolean is_rtp;
+ gboolean popped;
+
+ popped =
+ gst_rtsp_stream_transport_backlog_pop (trans, &buffer, &buffer_list,
+ &is_rtp);
+
+ g_assert (popped == TRUE);
+
+ send_ret = push_data (stream, trans, buffer, buffer_list, is_rtp);
+
+ gst_clear_buffer (&buffer);
+ gst_clear_buffer_list (&buffer_list);
+ }
+
+ gst_rtsp_stream_transport_unlock_backlog (trans);
+
+ if (!send_ret) {
+ /* remove transport on send error */
+ g_mutex_lock (&priv->lock);
+ update_transport (stream, trans, FALSE);
+ g_mutex_unlock (&priv->lock);
+ }
+}
+
+/* Must be called with priv->lock */
+static void
+send_tcp_message (GstRTSPStream * stream, gint idx)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstAppSink *sink;
+ GstSample *sample;
+ GstBuffer *buffer;
+ GstBufferList *buffer_list;
+ guint n_messages = 0;
+ gboolean is_rtp;
+ GPtrArray *transports;
+
+ if (!priv->have_buffer[idx])
+ return;
+
+ ensure_cached_transports (stream);
+
+ is_rtp = (idx == 0);
+
+ if (!any_transport_ready (stream, is_rtp))
+ return;
+
+ priv->have_buffer[idx] = FALSE;
+
+ if (priv->appsink[idx] == NULL) {
+ /* session expired */
+ return;
+ }
+
+ sink = GST_APP_SINK (priv->appsink[idx]);
+ sample = gst_app_sink_pull_sample (sink);
+ if (!sample) {
+ return;
+ }
+
+ buffer = gst_sample_get_buffer (sample);
+ buffer_list = gst_sample_get_buffer_list (sample);
+
+ /* We will get one message-sent notification per buffer or
+ * complete buffer-list. We handle each buffer-list as a unit */
+ if (buffer)
+ n_messages += 1;
+ if (buffer_list)
+ n_messages += 1;
+
+ transports = priv->tr_cache;
+ if (transports)
+ g_ptr_array_ref (transports);
+
+ if (transports) {
+ gint index;
+
+ for (index = 0; index < transports->len; index++) {
+ GstRTSPStreamTransport *tr = g_ptr_array_index (transports, index);
+ GstBuffer *buf_ref = NULL;
+ GstBufferList *buflist_ref = NULL;
+
+ gst_rtsp_stream_transport_lock_backlog (tr);
+
+ if (buffer)
+ buf_ref = gst_buffer_ref (buffer);
+ if (buffer_list)
+ buflist_ref = gst_buffer_list_ref (buffer_list);
+
+ if (!gst_rtsp_stream_transport_backlog_push (tr,
+ buf_ref, buflist_ref, is_rtp)) {
+ GST_ERROR_OBJECT (stream,
+ "Dropping slow transport %" GST_PTR_FORMAT, tr);
+ update_transport (stream, tr, FALSE);
+ }
+
+ gst_rtsp_stream_transport_unlock_backlog (tr);
+ }
+ }
+ gst_sample_unref (sample);
+
+ g_mutex_unlock (&priv->lock);
+
+ if (transports) {
+ gint index;
+
+ for (index = 0; index < transports->len; index++) {
+ GstRTSPStreamTransport *tr = g_ptr_array_index (transports, index);
+
+ check_transport_backlog (stream, tr);
+ }
+ g_ptr_array_unref (transports);
+ }
+
+ g_mutex_lock (&priv->lock);
+}
+
+static gpointer
+send_func (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ g_mutex_lock (&priv->send_lock);
+
+ while (priv->continue_sending) {
+ int i;
+ int idx = -1;
+ guint cookie;
+
+ cookie = priv->send_cookie;
+ g_mutex_unlock (&priv->send_lock);
+
+ g_mutex_lock (&priv->lock);
+
+ /* iterate from 1 and down, so we prioritize RTCP over RTP */
+ for (i = 1; i >= 0; i--) {
+ if (priv->have_buffer[i]) {
+ /* send message */
+ idx = i;
+ break;
+ }
+ }
+
+ if (idx != -1) {
+ send_tcp_message (stream, idx);
+ }
+
+ g_mutex_unlock (&priv->lock);
+
+ g_mutex_lock (&priv->send_lock);
+ while (cookie == priv->send_cookie && priv->continue_sending) {
+ g_cond_wait (&priv->send_cond, &priv->send_lock);
+ }
+ }
+
+ g_mutex_unlock (&priv->send_lock);
+
+ return NULL;
+}
+
+static GstFlowReturn
+handle_new_sample (GstAppSink * sink, gpointer user_data)
+{
+ GstRTSPStream *stream = user_data;
+ GstRTSPStreamPrivate *priv = stream->priv;
+ int i;
+
+ g_mutex_lock (&priv->lock);
+
+ for (i = 0; i < 2; i++) {
+ if (GST_ELEMENT_CAST (sink) == priv->appsink[i]) {
+ priv->have_buffer[i] = TRUE;
+ break;
+ }
+ }
+
+ if (priv->send_thread == NULL) {
+ priv->send_thread = g_thread_new (NULL, (GThreadFunc) send_func, user_data);
+ }
+
+ g_mutex_unlock (&priv->lock);
+
+ g_mutex_lock (&priv->send_lock);
+ priv->send_cookie++;
+ g_cond_signal (&priv->send_cond);
+ g_mutex_unlock (&priv->send_lock);
+
+ return GST_FLOW_OK;
+}
+
+static GstAppSinkCallbacks sink_cb = {
+ NULL, /* not interested in EOS */
+ NULL, /* not interested in preroll samples */
+ handle_new_sample,
+};
+
+static GstElement *
+get_rtp_encoder (GstRTSPStream * stream, guint session)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ if (priv->srtpenc == NULL) {
+ gchar *name;
+
+ name = g_strdup_printf ("srtpenc_%u", session);
+ priv->srtpenc = gst_element_factory_make ("srtpenc", name);
+ g_free (name);
+
+ g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
+ }
+ return gst_object_ref (priv->srtpenc);
+}
+
+static GstElement *
+request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstElement *oldenc, *enc;
+ GstPad *pad;
+ gchar *name;
+
+ if (priv->idx != session)
+ return NULL;
+
+ GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
+
+ oldenc = priv->srtpenc;
+ enc = get_rtp_encoder (stream, session);
+ name = g_strdup_printf ("rtp_sink_%d", session);
+ pad = gst_element_request_pad_simple (enc, name);
+ g_free (name);
+ gst_object_unref (pad);
+
+ if (oldenc == NULL)
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
+ enc);
+
+ return enc;
+}
+
+static GstElement *
+request_rtcp_encoder (GstElement * rtpbin, guint session,
+ GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstElement *oldenc, *enc;
+ GstPad *pad;
+ gchar *name;
+
+ if (priv->idx != session)
+ return NULL;
+
+ GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
+
+ oldenc = priv->srtpenc;
+ enc = get_rtp_encoder (stream, session);
+ name = g_strdup_printf ("rtcp_sink_%d", session);
+ pad = gst_element_request_pad_simple (enc, name);
+ g_free (name);
+ gst_object_unref (pad);
+
+ if (oldenc == NULL)
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
+ enc);
+
+ return enc;
+}
+
+static GstCaps *
+request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstCaps *caps;
+
+ GST_DEBUG ("request key %08x", ssrc);
+
+ g_mutex_lock (&priv->lock);
+ if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
+ gst_caps_ref (caps);
+ g_mutex_unlock (&priv->lock);
+
+ return caps;
+}
+
+static GstElement *
+request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
+ GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ if (priv->idx != session)
+ return NULL;
+
+ if (priv->srtpdec == NULL) {
+ gchar *name;
+
+ name = g_strdup_printf ("srtpdec_%u", session);
+ priv->srtpdec = gst_element_factory_make ("srtpdec", name);
+ g_free (name);
+
+ g_signal_connect (priv->srtpdec, "request-key",
+ (GCallback) request_key, stream);
+
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_RTCP_DECODER],
+ 0, priv->srtpdec);
+
+ }
+ return gst_object_ref (priv->srtpdec);
+}
+
+/**
+ * gst_rtsp_stream_request_aux_sender:
+ * @stream: a #GstRTSPStream
+ * @sessid: the session id
+ *
+ * Creating a rtxsend bin
+ *
+ * Returns: (transfer full) (nullable): a #GstElement.
+ *
+ * Since: 1.6
+ */
+GstElement *
+gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
+{
+ GstElement *bin;
+ GstPad *pad;
+ GstStructure *pt_map;
+ gchar *name;
+ guint pt, rtx_pt;
+ gchar *pt_s;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ pt = gst_rtsp_stream_get_pt (stream);
+ pt_s = g_strdup_printf ("%u", pt);
+ rtx_pt = stream->priv->rtx_pt;
+
+ GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
+
+ bin = gst_bin_new (NULL);
+ stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
+ pt_map = gst_structure_new ("application/x-rtp-pt-map",
+ pt_s, G_TYPE_UINT, rtx_pt, NULL);
+ g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
+ "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
+ g_free (pt_s);
+ gst_structure_free (pt_map);
+ gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
+
+ pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
+ name = g_strdup_printf ("src_%u", sessid);
+ gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
+ g_free (name);
+ gst_object_unref (pad);
+
+ pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
+ name = g_strdup_printf ("sink_%u", sessid);
+ gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
+ g_free (name);
+ gst_object_unref (pad);
+
+ return bin;
+}
+
+static void
+add_rtx_pt (gpointer key, GstCaps * caps, GstStructure * pt_map)
+{
+ guint pt = GPOINTER_TO_INT (key);
+ const GstStructure *s = gst_caps_get_structure (caps, 0);
+ const gchar *apt;
+
+ if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "RTX") &&
+ (apt = gst_structure_get_string (s, "apt"))) {
+ gst_structure_set (pt_map, apt, G_TYPE_UINT, pt, NULL);
+ }
+}
+
+/* Call with priv->lock taken */
+static void
+update_rtx_receive_pt_map (GstRTSPStream * stream)
+{
+ GstStructure *pt_map;
+
+ if (!stream->priv->rtxreceive)
+ goto done;
+
+ pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
+ g_hash_table_foreach (stream->priv->ptmap, (GHFunc) add_rtx_pt, pt_map);
+ g_object_set (stream->priv->rtxreceive, "payload-type-map", pt_map, NULL);
+ gst_structure_free (pt_map);
+
+done:
+ return;
+}
+
+static void
+retrieve_ulpfec_pt (gpointer key, GstCaps * caps, GstElement * ulpfec_decoder)
+{
+ guint pt = GPOINTER_TO_INT (key);
+ const GstStructure *s = gst_caps_get_structure (caps, 0);
+
+ if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
+ g_object_set (ulpfec_decoder, "pt", pt, NULL);
+}
+
+static void
+update_ulpfec_decoder_pt (GstRTSPStream * stream)
+{
+ if (!stream->priv->ulpfec_decoder)
+ goto done;
+
+ g_hash_table_foreach (stream->priv->ptmap, (GHFunc) retrieve_ulpfec_pt,
+ stream->priv->ulpfec_decoder);
+
+done:
+ return;
+}
+
+/**
+ * gst_rtsp_stream_request_aux_receiver:
+ * @stream: a #GstRTSPStream
+ * @sessid: the session id
+ *
+ * Creating a rtxreceive bin
+ *
+ * Returns: (transfer full) (nullable): a #GstElement.
+ *
+ * Since: 1.16
+ */
+GstElement *
+gst_rtsp_stream_request_aux_receiver (GstRTSPStream * stream, guint sessid)
+{
+ GstElement *bin;
+ GstPad *pad;
+ gchar *name;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ bin = gst_bin_new (NULL);
+ stream->priv->rtxreceive = gst_element_factory_make ("rtprtxreceive", NULL);
+ update_rtx_receive_pt_map (stream);
+ update_ulpfec_decoder_pt (stream);
+ gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxreceive));
+
+ pad = gst_element_get_static_pad (stream->priv->rtxreceive, "src");
+ name = g_strdup_printf ("src_%u", sessid);
+ gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
+ g_free (name);
+ gst_object_unref (pad);
+
+ pad = gst_element_get_static_pad (stream->priv->rtxreceive, "sink");
+ name = g_strdup_printf ("sink_%u", sessid);
+ gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
+ g_free (name);
+ gst_object_unref (pad);
+
+ return bin;
+}
+
+/**
+ * gst_rtsp_stream_set_pt_map:
+ * @stream: a #GstRTSPStream
+ * @pt: the pt
+ * @caps: a #GstCaps
+ *
+ * Configure a pt map between @pt and @caps.
+ */
+void
+gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ if (!GST_IS_CAPS (caps))
+ return;
+
+ g_mutex_lock (&priv->lock);
+ g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
+ update_rtx_receive_pt_map (stream);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_set_publish_clock_mode:
+ * @stream: a #GstRTSPStream
+ * @mode: the clock publish mode
+ *
+ * Sets if and how the stream clock should be published according to RFC7273.
+ *
+ * Since: 1.8
+ */
+void
+gst_rtsp_stream_set_publish_clock_mode (GstRTSPStream * stream,
+ GstRTSPPublishClockMode mode)
+{
+ GstRTSPStreamPrivate *priv;
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ priv->publish_clock_mode = mode;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_publish_clock_mode:
+ * @stream: a #GstRTSPStream
+ *
+ * Gets if and how the stream clock should be published according to RFC7273.
+ *
+ * Returns: The GstRTSPPublishClockMode
+ *
+ * Since: 1.8
+ */
+GstRTSPPublishClockMode
+gst_rtsp_stream_get_publish_clock_mode (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPPublishClockMode ret;
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ ret = priv->publish_clock_mode;
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
+
+static GstCaps *
+request_pt_map (GstElement * rtpbin, guint session, guint pt,
+ GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstCaps *caps = NULL;
+
+ g_mutex_lock (&priv->lock);
+
+ if (priv->idx == session) {
+ caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
+ if (caps) {
+ GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
+ gst_caps_ref (caps);
+ } else {
+ GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
+ }
+ }
+
+ g_mutex_unlock (&priv->lock);
+
+ return caps;
+}
+
+static void
+pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ gchar *name;
+ GstPadLinkReturn ret;
+ guint sessid;
+
+ GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
+ GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
+
+ name = gst_pad_get_name (pad);
+ if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
+ g_free (name);
+ return;
+ }
+ g_free (name);
+
+ if (priv->idx != sessid)
+ return;
+
+ if (gst_pad_is_linked (priv->sinkpad)) {
+ GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
+ GST_DEBUG_PAD_NAME (priv->sinkpad));
+ return;
+ }
+
+ /* link the RTP pad to the session manager, it should not really fail unless
+ * this is not really an RTP pad */
+ ret = gst_pad_link (pad, priv->sinkpad);
+ if (ret != GST_PAD_LINK_OK)
+ goto link_failed;
+ priv->recv_rtp_src = gst_object_ref (pad);
+
+ return;
+
+/* ERRORS */
+link_failed:
+ {
+ GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
+ GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
+ }
+}
+
+static void
+on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
+ GstRTSPStream * stream)
+{
+ /* TODO: What to do here other than this? */
+ GST_DEBUG ("Stream %p: Got EOS", stream);
+ gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
+}
+
+typedef struct _ProbeData ProbeData;
+
+struct _ProbeData
+{
+ GstRTSPStream *stream;
+ /* existing sink, already linked to tee */
+ GstElement *sink1;
+ /* new sink, about to be linked */
+ GstElement *sink2;
+ /* new queue element, that will be linked to tee and sink1 */
+ GstElement **queue1;
+ /* new queue element, that will be linked to tee and sink2 */
+ GstElement **queue2;
+ GstPad *sink_pad;
+ GstPad *tee_pad;
+ guint index;
+};
+
+static void
+free_cb_data (gpointer user_data)
+{
+ ProbeData *data = user_data;
+
+ gst_object_unref (data->stream);
+ gst_object_unref (data->sink1);
+ gst_object_unref (data->sink2);
+ gst_object_unref (data->sink_pad);
+ gst_object_unref (data->tee_pad);
+ g_free (data);
+}
+
+
+static void
+create_and_plug_queue_to_unlinked_stream (GstRTSPStream * stream,
+ GstElement * tee, GstElement * sink, GstElement ** queue)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstPad *tee_pad;
+ GstPad *queue_pad;
+ GstPad *sink_pad;
+
+ /* create queue for the new stream */
+ *queue = gst_element_factory_make ("queue", NULL);
+ g_object_set (*queue, "max-size-buffers", 1, "max-size-bytes", 0,
+ "max-size-time", G_GINT64_CONSTANT (0), NULL);
+ gst_bin_add (priv->joined_bin, *queue);
+
+ /* link tee to queue */
+ tee_pad = gst_element_request_pad_simple (tee, "src_%u");
+ queue_pad = gst_element_get_static_pad (*queue, "sink");
+ gst_pad_link (tee_pad, queue_pad);
+ gst_object_unref (queue_pad);
+ gst_object_unref (tee_pad);
+
+ /* link queue to sink */
+ queue_pad = gst_element_get_static_pad (*queue, "src");
+ sink_pad = gst_element_get_static_pad (sink, "sink");
+ gst_pad_link (queue_pad, sink_pad);
+ gst_object_unref (queue_pad);
+ gst_object_unref (sink_pad);
+
+ gst_element_sync_state_with_parent (sink);
+ gst_element_sync_state_with_parent (*queue);
+}
+
+static GstPadProbeReturn
+create_and_plug_queue_to_linked_stream_probe_cb (GstPad * inpad,
+ GstPadProbeInfo * info, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ ProbeData *data = user_data;
+ GstRTSPStream *stream;
+ GstElement **queue1;
+ GstElement **queue2;
+ GstPad *sink_pad;
+ GstPad *tee_pad;
+ GstPad *queue_pad;
+ guint index;
+
+ stream = data->stream;
+ priv = stream->priv;
+ queue1 = data->queue1;
+ queue2 = data->queue2;
+ sink_pad = data->sink_pad;
+ tee_pad = data->tee_pad;
+ index = data->index;
+
+ /* unlink tee and the existing sink:
+ * .-----. .---------.
+ * | tee | | sink1 |
+ * sink src->sink |
+ * '-----' '---------'
+ */
+ g_assert (gst_pad_unlink (tee_pad, sink_pad));
+
+ /* add queue to the already existing stream */
+ *queue1 = gst_element_factory_make ("queue", NULL);
+ g_object_set (*queue1, "max-size-buffers", 1, "max-size-bytes", 0,
+ "max-size-time", G_GINT64_CONSTANT (0), NULL);
+ gst_bin_add (priv->joined_bin, *queue1);
+
+ /* link tee, queue and sink:
+ * .-----. .---------. .---------.
+ * | tee | | queue1 | | sink1 |
+ * sink src->sink src->sink |
+ * '-----' '---------' '---------'
+ */
+ queue_pad = gst_element_get_static_pad (*queue1, "sink");
+ gst_pad_link (tee_pad, queue_pad);
+ gst_object_unref (queue_pad);
+ queue_pad = gst_element_get_static_pad (*queue1, "src");
+ gst_pad_link (queue_pad, sink_pad);
+ gst_object_unref (queue_pad);
+
+ gst_element_sync_state_with_parent (*queue1);
+
+ /* create queue and link it to tee and the new sink */
+ create_and_plug_queue_to_unlinked_stream (stream,
+ priv->tee[index], data->sink2, queue2);
+
+ /* the final stream:
+ *
+ * .-----. .---------. .---------.
+ * | tee | | queue1 | | sink1 |
+ * sink src->sink src->sink |
+ * | | '---------' '---------'
+ * | | .---------. .---------.
+ * | | | queue2 | | sink2 |
+ * | src->sink src->sink |
+ * '-----' '---------' '---------'
+ */
+
+ return GST_PAD_PROBE_REMOVE;
+}
+
+static void
+create_and_plug_queue_to_linked_stream (GstRTSPStream * stream,
+ GstElement * sink1, GstElement * sink2, guint index, GstElement ** queue1,
+ GstElement ** queue2)
+{
+ ProbeData *data;
+
+ data = g_new0 (ProbeData, 1);
+ data->stream = gst_object_ref (stream);
+ data->sink1 = gst_object_ref (sink1);
+ data->sink2 = gst_object_ref (sink2);
+ data->queue1 = queue1;
+ data->queue2 = queue2;
+ data->index = index;
+
+ data->sink_pad = gst_element_get_static_pad (sink1, "sink");
+ g_assert (data->sink_pad);
+ data->tee_pad = gst_pad_get_peer (data->sink_pad);
+ g_assert (data->tee_pad);
+
+ gst_pad_add_probe (data->tee_pad, GST_PAD_PROBE_TYPE_IDLE,
+ create_and_plug_queue_to_linked_stream_probe_cb, data, free_cb_data);
+}
+
+static void
+plug_udp_sink (GstRTSPStream * stream, GstElement * sink_to_plug,
+ GstElement ** queue_to_plug, guint index, gboolean is_mcast)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstElement *existing_sink;
+
+ if (is_mcast)
+ existing_sink = priv->udpsink[index];
+ else
+ existing_sink = priv->mcast_udpsink[index];
+
+ GST_DEBUG_OBJECT (stream, "plug %s sink", is_mcast ? "mcast" : "udp");
+
+ /* add sink to the bin */
+ gst_bin_add (priv->joined_bin, sink_to_plug);
+
+ if (priv->appsink[index] && existing_sink) {
+
+ /* queues are already added for the existing stream, add one for
+ the newly added udp stream */
+ create_and_plug_queue_to_unlinked_stream (stream, priv->tee[index],
+ sink_to_plug, queue_to_plug);
+
+ } else if (priv->appsink[index] || existing_sink) {
+ GstElement **queue;
+ GstElement *element;
+
+ /* add queue to the already existing stream plus the newly created udp
+ stream */
+ if (priv->appsink[index]) {
+ element = priv->appsink[index];
+ queue = &priv->appqueue[index];
+ } else {
+ element = existing_sink;
+ if (is_mcast)
+ queue = &priv->udpqueue[index];
+ else
+ queue = &priv->mcast_udpqueue[index];
+ }
+
+ create_and_plug_queue_to_linked_stream (stream, element, sink_to_plug,
+ index, queue, queue_to_plug);
+
+ } else {
+ GstPad *tee_pad;
+ GstPad *sink_pad;
+
+ GST_DEBUG_OBJECT (stream, "creating first stream");
+
+ /* no need to add queues */
+ tee_pad = gst_element_request_pad_simple (priv->tee[index], "src_%u");
+ sink_pad = gst_element_get_static_pad (sink_to_plug, "sink");
+ gst_pad_link (tee_pad, sink_pad);
+ gst_object_unref (tee_pad);
+ gst_object_unref (sink_pad);
+ }
+
+ gst_element_sync_state_with_parent (sink_to_plug);
+}
+
+static void
+plug_tcp_sink (GstRTSPStream * stream, guint index)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ GST_DEBUG_OBJECT (stream, "plug tcp sink");
+
+ /* add sink to the bin */
+ gst_bin_add (priv->joined_bin, priv->appsink[index]);
+
+ if (priv->mcast_udpsink[index] && priv->udpsink[index]) {
+
+ /* queues are already added for the existing stream, add one for
+ the newly added tcp stream */
+ create_and_plug_queue_to_unlinked_stream (stream,
+ priv->tee[index], priv->appsink[index], &priv->appqueue[index]);
+
+ } else if (priv->mcast_udpsink[index] || priv->udpsink[index]) {
+ GstElement **queue;
+ GstElement *element;
+
+ /* add queue to the already existing stream plus the newly created tcp
+ stream */
+ if (priv->mcast_udpsink[index]) {
+ element = priv->mcast_udpsink[index];
+ queue = &priv->mcast_udpqueue[index];
+ } else {
+ element = priv->udpsink[index];
+ queue = &priv->udpqueue[index];
+ }
+
+ create_and_plug_queue_to_linked_stream (stream, element,
+ priv->appsink[index], index, queue, &priv->appqueue[index]);
+
+ } else {
+ GstPad *tee_pad;
+ GstPad *sink_pad;
+
+ /* no need to add queues */
+ tee_pad = gst_element_request_pad_simple (priv->tee[index], "src_%u");
+ sink_pad = gst_element_get_static_pad (priv->appsink[index], "sink");
+ gst_pad_link (tee_pad, sink_pad);
+ gst_object_unref (tee_pad);
+ gst_object_unref (sink_pad);
+ }
+
+ gst_element_sync_state_with_parent (priv->appsink[index]);
+}
+
+static void
+plug_sink (GstRTSPStream * stream, const GstRTSPTransport * transport,
+ guint index)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean is_tcp, is_udp, is_mcast;
+ priv = stream->priv;
+
+ is_tcp = transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP;
+ is_udp = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP;
+ is_mcast = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST;
+
+ if (is_udp)
+ plug_udp_sink (stream, priv->udpsink[index],
+ &priv->udpqueue[index], index, FALSE);
+
+ else if (is_mcast)
+ plug_udp_sink (stream, priv->mcast_udpsink[index],
+ &priv->mcast_udpqueue[index], index, TRUE);
+
+ else if (is_tcp)
+ plug_tcp_sink (stream, index);
+}
+
+/* must be called with lock */
+static gboolean
+create_sender_part (GstRTSPStream * stream, const GstRTSPTransport * transport)
+{
+ GstRTSPStreamPrivate *priv;
+ GstPad *pad;
+ GstBin *bin;
+ gboolean is_tcp, is_udp, is_mcast;
+ gint mcast_ttl = 0;
+ gint i;
+
+ GST_DEBUG_OBJECT (stream, "create sender part");
+ priv = stream->priv;
+ bin = priv->joined_bin;
+
+ is_tcp = transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP;
+ is_udp = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP;
+ is_mcast = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST;
+
+ if (is_mcast)
+ mcast_ttl = transport->ttl;
+
+ GST_DEBUG_OBJECT (stream, "tcp: %d, udp: %d, mcast: %d (ttl: %d)", is_tcp,
+ is_udp, is_mcast, mcast_ttl);
+
+ if (is_udp && !priv->server_addr_v4 && !priv->server_addr_v6) {
+ GST_WARNING_OBJECT (stream, "no sockets assigned for UDP");
+ return FALSE;
+ }
+
+ if (is_mcast && !priv->mcast_addr_v4 && !priv->mcast_addr_v6) {
+ GST_WARNING_OBJECT (stream, "no sockets assigned for UDP multicast");
+ return FALSE;
+ }
+
+ if (g_object_class_find_property (G_OBJECT_GET_CLASS (priv->payloader),
+ "onvif-no-rate-control"))
+ g_object_set (priv->payloader, "onvif-no-rate-control",
+ !priv->do_rate_control, NULL);
+
+ for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
+ gboolean link_tee = FALSE;
+ /* For the sender we create this bit of pipeline for both
+ * RTP and RTCP (when enabled).
+ * Initially there will be only one active transport for
+ * the stream, so the pipeline will look like this:
+ *
+ * .--------. .-----. .---------.
+ * | rtpbin | | tee | | sink |
+ * | send->sink src->sink |
+ * '--------' '-----' '---------'
+ *
+ * For each new transport, the already existing branch will
+ * be reconfigured by adding a queue element:
+ *
+ * .--------. .-----. .---------. .---------.
+ * | rtpbin | | tee | | queue | | udpsink |
+ * | send->sink src->sink src->sink |
+ * '--------' | | '---------' '---------'
+ * | | .---------. .---------.
+ * | | | queue | | udpsink |
+ * | src->sink src->sink |
+ * | | '---------' '---------'
+ * | | .---------. .---------.
+ * | | | queue | | appsink |
+ * | src->sink src->sink |
+ * '-----' '---------' '---------'
+ */
+
+ /* Only link the RTP send src if we're going to send RTP, link
+ * the RTCP send src always */
+ if (!priv->srcpad && i == 0)
+ continue;
+
+ if (!priv->tee[i]) {
+ /* make tee for RTP/RTCP */
+ priv->tee[i] = gst_element_factory_make ("tee", NULL);
+ gst_bin_add (bin, priv->tee[i]);
+ link_tee = TRUE;
+ }
+
+ if (is_udp && !priv->udpsink[i]) {
+ /* we create only one pair of udpsinks for IPv4 and IPv6 */
+ create_and_configure_udpsink (stream, &priv->udpsink[i],
+ priv->socket_v4[i], priv->socket_v6[i], FALSE, (i == 0), mcast_ttl);
+ plug_sink (stream, transport, i);
+ } else if (is_mcast && !priv->mcast_udpsink[i]) {
+ /* we create only one pair of mcast-udpsinks for IPv4 and IPv6 */
+ create_and_configure_udpsink (stream, &priv->mcast_udpsink[i],
+ priv->mcast_socket_v4[i], priv->mcast_socket_v6[i], TRUE, (i == 0),
+ mcast_ttl);
+ plug_sink (stream, transport, i);
+ } else if (is_tcp && !priv->appsink[i]) {
+ /* make appsink */
+ priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
+ TRUE, "max-buffers", 1, NULL);
+
+ if (i == 0)
+ g_object_set (priv->appsink[i], "sync", priv->do_rate_control, NULL);
+
+ /* we need to set sync and preroll to FALSE for the sink to avoid
+ * deadlock. This is only needed for sink sending RTCP data. */
+ if (i == 1)
+ g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
+
+ gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
+ &sink_cb, stream, NULL);
+ plug_sink (stream, transport, i);
+ }
+
+ if (link_tee) {
+ /* and link to rtpbin send pad */
+ gst_element_sync_state_with_parent (priv->tee[i]);
+ pad = gst_element_get_static_pad (priv->tee[i], "sink");
+ gst_pad_link (priv->send_src[i], pad);
+ gst_object_unref (pad);
+ }
+ }
+
+ return TRUE;
+}
+
+/* must be called with lock */
+static void
+plug_src (GstRTSPStream * stream, GstBin * bin, GstElement * src,
+ GstElement * funnel)
+{
+ GstRTSPStreamPrivate *priv;
+ GstPad *pad, *selpad;
+ gulong id = 0;
+
+ priv = stream->priv;
+
+ /* add src */
+ gst_bin_add (bin, src);
+
+ pad = gst_element_get_static_pad (src, "src");
+ if (priv->srcpad) {
+ /* block pad so src can't push data while it's not yet linked */
+ id = gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BLOCK |
+ GST_PAD_PROBE_TYPE_BUFFER, NULL, NULL, NULL);
+ /* we set and keep these to playing so that they don't cause NO_PREROLL return
+ * values. This is only relevant for PLAY pipelines */
+ gst_element_set_state (src, GST_STATE_PLAYING);
+ gst_element_set_locked_state (src, TRUE);
+ }
+
+ /* and link to the funnel */
+ selpad = gst_element_request_pad_simple (funnel, "sink_%u");
+ gst_pad_link (pad, selpad);
+ if (id != 0)
+ gst_pad_remove_probe (pad, id);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+}
+
+/* must be called with lock */
+static gboolean
+create_receiver_part (GstRTSPStream * stream, const GstRTSPTransport *
+ transport)
+{
+ gboolean ret = FALSE;
+ GstRTSPStreamPrivate *priv;
+ GstPad *pad;
+ GstBin *bin;
+ gboolean tcp;
+ gboolean udp;
+ gboolean mcast;
+ gboolean secure;
+ gint i;
+ GstCaps *rtp_caps;
+ GstCaps *rtcp_caps;
+
+ GST_DEBUG_OBJECT (stream, "create receiver part");
+ priv = stream->priv;
+ bin = priv->joined_bin;
+
+ tcp = transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP;
+ udp = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP;
+ mcast = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST;
+ secure = (priv->profiles & GST_RTSP_PROFILE_SAVP)
+ || (priv->profiles & GST_RTSP_PROFILE_SAVPF);
+
+ if (secure) {
+ rtp_caps = gst_caps_new_empty_simple ("application/x-srtp");
+ rtcp_caps = gst_caps_new_empty_simple ("application/x-srtcp");
+ } else {
+ rtp_caps = gst_caps_new_empty_simple ("application/x-rtp");
+ rtcp_caps = gst_caps_new_empty_simple ("application/x-rtcp");
+ }
+
+ GST_DEBUG_OBJECT (stream,
+ "RTP caps: %" GST_PTR_FORMAT " RTCP caps: %" GST_PTR_FORMAT, rtp_caps,
+ rtcp_caps);
+
+ for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
+ /* For the receiver we create this bit of pipeline for both
+ * RTP and RTCP (when enabled). We receive RTP/RTCP on appsrc and udpsrc
+ * and it is all funneled into the rtpbin receive pad.
+ *
+ *
+ * .--------. .--------. .--------.
+ * | udpsrc | | funnel | | rtpbin |
+ * | RTP src->sink src->sink |
+ * '--------' | | | |
+ * .--------. | | | |
+ * | appsrc | | | | |
+ * | RTP src->sink | | |
+ * '--------' '--------' | |
+ * | |
+ * .--------. .--------. | |
+ * | udpsrc | | funnel | | |
+ * | RTCP src->sink src->sink |
+ * '--------' | | '--------'
+ * .--------. | |
+ * | appsrc | | |
+ * | RTCP src->sink |
+ * '--------' '--------'
+ */
+
+ if (!priv->sinkpad && i == 0) {
+ /* Only connect recv RTP sink if we expect to receive RTP. Connect recv
+ * RTCP sink always */
+ continue;
+ }
+
+ /* make funnel for the RTP/RTCP receivers */
+ if (!priv->funnel[i]) {
+ priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
+ gst_bin_add (bin, priv->funnel[i]);
+
+ pad = gst_element_get_static_pad (priv->funnel[i], "src");
+ gst_pad_link (pad, priv->recv_sink[i]);
+ gst_object_unref (pad);
+ }
+
+ if (udp && !priv->udpsrc_v4[i] && priv->server_addr_v4) {
+ GST_DEBUG_OBJECT (stream, "udp IPv4, create and configure udpsources");
+ if (!create_and_configure_udpsource (&priv->udpsrc_v4[i],
+ priv->socket_v4[i]))
+ goto done;
+
+ if (i == 0) {
+ g_object_set (priv->udpsrc_v4[i], "caps", rtp_caps, NULL);
+ } else {
+ g_object_set (priv->udpsrc_v4[i], "caps", rtcp_caps, NULL);
+
+ /* block early rtcp packets, pipeline not ready */
+ g_assert (priv->block_early_rtcp_pad == NULL);
+ priv->block_early_rtcp_pad = gst_element_get_static_pad
+ (priv->udpsrc_v4[i], "src");
+ priv->block_early_rtcp_probe = gst_pad_add_probe
+ (priv->block_early_rtcp_pad,
+ GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER, NULL, NULL,
+ NULL);
+ }
+
+ plug_src (stream, bin, priv->udpsrc_v4[i], priv->funnel[i]);
+ }
+
+ if (udp && !priv->udpsrc_v6[i] && priv->server_addr_v6) {
+ GST_DEBUG_OBJECT (stream, "udp IPv6, create and configure udpsources");
+ if (!create_and_configure_udpsource (&priv->udpsrc_v6[i],
+ priv->socket_v6[i]))
+ goto done;
+
+ if (i == 0) {
+ g_object_set (priv->udpsrc_v6[i], "caps", rtp_caps, NULL);
+ } else {
+ g_object_set (priv->udpsrc_v6[i], "caps", rtcp_caps, NULL);
+
+ /* block early rtcp packets, pipeline not ready */
+ g_assert (priv->block_early_rtcp_pad_ipv6 == NULL);
+ priv->block_early_rtcp_pad_ipv6 = gst_element_get_static_pad
+ (priv->udpsrc_v6[i], "src");
+ priv->block_early_rtcp_probe_ipv6 = gst_pad_add_probe
+ (priv->block_early_rtcp_pad_ipv6,
+ GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER, NULL, NULL,
+ NULL);
+ }
+
+ plug_src (stream, bin, priv->udpsrc_v6[i], priv->funnel[i]);
+ }
+
+ if (mcast && !priv->mcast_udpsrc_v4[i] && priv->mcast_addr_v4) {
+ GST_DEBUG_OBJECT (stream, "mcast IPv4, create and configure udpsources");
+ if (!create_and_configure_udpsource (&priv->mcast_udpsrc_v4[i],
+ priv->mcast_socket_v4[i]))
+ goto done;
+
+ if (i == 0) {
+ g_object_set (priv->mcast_udpsrc_v4[i], "caps", rtp_caps, NULL);
+ } else {
+ g_object_set (priv->mcast_udpsrc_v4[i], "caps", rtcp_caps, NULL);
+ }
+
+ plug_src (stream, bin, priv->mcast_udpsrc_v4[i], priv->funnel[i]);
+ }
+
+ if (mcast && !priv->mcast_udpsrc_v6[i] && priv->mcast_addr_v6) {
+ GST_DEBUG_OBJECT (stream, "mcast IPv6, create and configure udpsources");
+ if (!create_and_configure_udpsource (&priv->mcast_udpsrc_v6[i],
+ priv->mcast_socket_v6[i]))
+ goto done;
+
+ if (i == 0) {
+ g_object_set (priv->mcast_udpsrc_v6[i], "caps", rtp_caps, NULL);
+ } else {
+ g_object_set (priv->mcast_udpsrc_v6[i], "caps", rtcp_caps, NULL);
+ }
+
+ plug_src (stream, bin, priv->mcast_udpsrc_v6[i], priv->funnel[i]);
+ }
+
+ if (tcp && !priv->appsrc[i]) {
+ /* make and add appsrc */
+ priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
+ priv->appsrc_base_time[i] = -1;
+ g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, "is-live",
+ TRUE, NULL);
+ plug_src (stream, bin, priv->appsrc[i], priv->funnel[i]);
+ }
+
+ gst_element_sync_state_with_parent (priv->funnel[i]);
+ }
+
+ ret = TRUE;
+
+done:
+ gst_caps_unref (rtp_caps);
+ gst_caps_unref (rtcp_caps);
+ return ret;
+}
+
+gboolean
+gst_rtsp_stream_is_tcp_receiver (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean ret = FALSE;
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ ret = (priv->sinkpad != NULL && priv->appsrc[0] != NULL);
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
+
+static gboolean
+check_mcast_client_addr (GstRTSPStream * stream, const GstRTSPTransport * tr)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GList *walk;
+
+ if (priv->mcast_clients == NULL)
+ goto no_addr;
+
+ if (tr == NULL)
+ goto no_transport;
+
+ if (tr->destination == NULL)
+ goto no_destination;
+
+ for (walk = priv->mcast_clients; walk; walk = g_list_next (walk)) {
+ UdpClientAddrInfo *cli = walk->data;
+
+ if ((g_strcmp0 (cli->address, tr->destination) == 0) &&
+ (cli->rtp_port == tr->port.min))
+ return TRUE;
+ }
+
+ return FALSE;
+
+no_addr:
+ {
+ GST_WARNING_OBJECT (stream, "Adding mcast transport, but no mcast address "
+ "has been reserved");
+ return FALSE;
+ }
+no_transport:
+ {
+ GST_WARNING_OBJECT (stream, "Adding mcast transport, but no transport "
+ "has been provided");
+ return FALSE;
+ }
+no_destination:
+ {
+ GST_WARNING_OBJECT (stream, "Adding mcast transport, but it doesn't match "
+ "the reserved address");
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_join_bin:
+ * @stream: a #GstRTSPStream
+ * @bin: (transfer none): a #GstBin to join
+ * @rtpbin: (transfer none): a rtpbin element in @bin
+ * @state: the target state of the new elements
+ *
+ * Join the #GstBin @bin that contains the element @rtpbin.
+ *
+ * @stream will link to @rtpbin, which must be inside @bin. The elements
+ * added to @bin will be set to the state given in @state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
+ GstElement * rtpbin, GstState state)
+{
+ GstRTSPStreamPrivate *priv;
+ guint idx;
+ gchar *name;
+ GstPadLinkReturn ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
+ g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->joined_bin != NULL)
+ goto was_joined;
+
+ /* create a session with the same index as the stream */
+ idx = priv->idx;
+
+ GST_INFO ("stream %p joining bin as session %u", stream, idx);
+
+ if (priv->profiles & GST_RTSP_PROFILE_SAVP
+ || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
+ /* For SRTP */
+ g_signal_connect (rtpbin, "request-rtp-encoder",
+ (GCallback) request_rtp_encoder, stream);
+ g_signal_connect (rtpbin, "request-rtcp-encoder",
+ (GCallback) request_rtcp_encoder, stream);
+ g_signal_connect (rtpbin, "request-rtp-decoder",
+ (GCallback) request_rtp_rtcp_decoder, stream);
+ g_signal_connect (rtpbin, "request-rtcp-decoder",
+ (GCallback) request_rtp_rtcp_decoder, stream);
+ }
+
+ if (priv->sinkpad) {
+ g_signal_connect (rtpbin, "request-pt-map",
+ (GCallback) request_pt_map, stream);
+ }
+
+ /* get pads from the RTP session element for sending and receiving
+ * RTP/RTCP*/
+ if (priv->srcpad) {
+ /* get a pad for sending RTP */
+ name = g_strdup_printf ("send_rtp_sink_%u", idx);
+ priv->send_rtp_sink = gst_element_request_pad_simple (rtpbin, name);
+ g_free (name);
+
+ /* link the RTP pad to the session manager, it should not really fail unless
+ * this is not really an RTP pad */
+ ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
+ if (ret != GST_PAD_LINK_OK)
+ goto link_failed;
+
+ name = g_strdup_printf ("send_rtp_src_%u", idx);
+ priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
+ g_free (name);
+ } else {
+ /* RECORD case: need to connect our sinkpad from here */
+ g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
+ /* EOS */
+ g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
+
+ name = g_strdup_printf ("recv_rtp_sink_%u", idx);
+ priv->recv_sink[0] = gst_element_request_pad_simple (rtpbin, name);
+ g_free (name);
+ }
+
+ if (priv->enable_rtcp) {
+ name = g_strdup_printf ("send_rtcp_src_%u", idx);
+ priv->send_src[1] = gst_element_request_pad_simple (rtpbin, name);
+ g_free (name);
+
+ name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
+ priv->recv_sink[1] = gst_element_request_pad_simple (rtpbin, name);
+ g_free (name);
+ }
+
+ /* get the session */
+ g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
+
+ g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
+ stream);
+ g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
+ stream);
+ g_signal_connect (priv->session, "on-ssrc-active",
+ (GCallback) on_ssrc_active, stream);
+ g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
+ stream);
+ g_signal_connect (priv->session, "on-bye-timeout",
+ (GCallback) on_bye_timeout, stream);
+ g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
+ stream);
+
+ /* signal for sender ssrc */
+ g_signal_connect (priv->session, "on-new-sender-ssrc",
+ (GCallback) on_new_sender_ssrc, stream);
+ g_signal_connect (priv->session, "on-sender-ssrc-active",
+ (GCallback) on_sender_ssrc_active, stream);
+
+ g_object_set (priv->session, "disable-sr-timestamp", !priv->do_rate_control,
+ NULL);
+
+ if (priv->srcpad) {
+ /* be notified of caps changes */
+ priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
+ (GCallback) caps_notify, stream);
+ priv->caps = gst_pad_get_current_caps (priv->send_src[0]);
+ }
+
+ priv->joined_bin = bin;
+ GST_DEBUG_OBJECT (stream, "successfully joined bin");
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+ /* ERRORS */
+was_joined:
+ {
+ g_mutex_unlock (&priv->lock);
+ return TRUE;
+ }
+link_failed:
+ {
+ GST_WARNING ("failed to link stream %u", idx);
+ gst_object_unref (priv->send_rtp_sink);
+ priv->send_rtp_sink = NULL;
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+static void
+clear_element (GstBin * bin, GstElement ** elementptr)
+{
+ if (*elementptr) {
+ gst_element_set_locked_state (*elementptr, FALSE);
+ gst_element_set_state (*elementptr, GST_STATE_NULL);
+ if (GST_ELEMENT_PARENT (*elementptr))
+ gst_bin_remove (bin, *elementptr);
+ else
+ gst_object_unref (*elementptr);
+ *elementptr = NULL;
+ }
+}
+
+/**
+ * gst_rtsp_stream_leave_bin:
+ * @stream: a #GstRTSPStream
+ * @bin: (transfer none): a #GstBin
+ * @rtpbin: (transfer none): a rtpbin #GstElement
+ *
+ * Remove the elements of @stream from @bin.
+ *
+ * Return: %TRUE on success.
+ */
+gboolean
+gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
+ GstElement * rtpbin)
+{
+ GstRTSPStreamPrivate *priv;
+ gint i;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
+ g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->send_lock);
+ priv->continue_sending = FALSE;
+ priv->send_cookie++;
+ g_cond_signal (&priv->send_cond);
+ g_mutex_unlock (&priv->send_lock);
+
+ if (priv->send_thread) {
+ g_thread_join (priv->send_thread);
+ }
+
+ g_mutex_lock (&priv->lock);
+ if (priv->joined_bin == NULL)
+ goto was_not_joined;
+ if (priv->joined_bin != bin)
+ goto wrong_bin;
+
+ priv->joined_bin = NULL;
+
+ /* all transports must be removed by now */
+ if (priv->transports != NULL)
+ goto transports_not_removed;
+
+ if (priv->send_pool) {
+ GThreadPool *slask;
+
+ slask = priv->send_pool;
+ priv->send_pool = NULL;
+ g_mutex_unlock (&priv->lock);
+ g_thread_pool_free (slask, TRUE, TRUE);
+ g_mutex_lock (&priv->lock);
+ }
+
+ clear_tr_cache (priv);
+
+ GST_INFO ("stream %p leaving bin", stream);
+
+ if (priv->srcpad) {
+ gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
+
+ g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
+ gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
+ gst_object_unref (priv->send_rtp_sink);
+ priv->send_rtp_sink = NULL;
+ } else if (priv->recv_rtp_src) {
+ gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
+ gst_object_unref (priv->recv_rtp_src);
+ priv->recv_rtp_src = NULL;
+ }
+
+ for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
+ clear_element (bin, &priv->udpsrc_v4[i]);
+ clear_element (bin, &priv->udpsrc_v6[i]);
+ clear_element (bin, &priv->udpqueue[i]);
+ clear_element (bin, &priv->udpsink[i]);
+
+ clear_element (bin, &priv->mcast_udpsrc_v4[i]);
+ clear_element (bin, &priv->mcast_udpsrc_v6[i]);
+ clear_element (bin, &priv->mcast_udpqueue[i]);
+ clear_element (bin, &priv->mcast_udpsink[i]);
+
+ clear_element (bin, &priv->appsrc[i]);
+ clear_element (bin, &priv->appqueue[i]);
+ clear_element (bin, &priv->appsink[i]);
+
+ clear_element (bin, &priv->tee[i]);
+ clear_element (bin, &priv->funnel[i]);
+
+ if (priv->sinkpad || i == 1) {
+ gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
+ gst_object_unref (priv->recv_sink[i]);
+ priv->recv_sink[i] = NULL;
+ }
+ }
+
+ if (priv->srcpad) {
+ gst_object_unref (priv->send_src[0]);
+ priv->send_src[0] = NULL;
+ }
+
+ if (priv->enable_rtcp) {
+ gst_element_release_request_pad (rtpbin, priv->send_src[1]);
+ gst_object_unref (priv->send_src[1]);
+ priv->send_src[1] = NULL;
+ }
+
+ g_object_unref (priv->session);
+ priv->session = NULL;
+ if (priv->caps)
+ gst_caps_unref (priv->caps);
+ priv->caps = NULL;
+
+ if (priv->srtpenc)
+ gst_object_unref (priv->srtpenc);
+ if (priv->srtpdec)
+ gst_object_unref (priv->srtpdec);
+
+ if (priv->mcast_addr_v4)
+ gst_rtsp_address_free (priv->mcast_addr_v4);
+ priv->mcast_addr_v4 = NULL;
+ if (priv->mcast_addr_v6)
+ gst_rtsp_address_free (priv->mcast_addr_v6);
+ priv->mcast_addr_v6 = NULL;
+ if (priv->server_addr_v4)
+ gst_rtsp_address_free (priv->server_addr_v4);
+ priv->server_addr_v4 = NULL;
+ if (priv->server_addr_v6)
+ gst_rtsp_address_free (priv->server_addr_v6);
+ priv->server_addr_v6 = NULL;
+
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+was_not_joined:
+ {
+ g_mutex_unlock (&priv->lock);
+ return TRUE;
+ }
+transports_not_removed:
+ {
+ GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+wrong_bin:
+ {
+ GST_ERROR_OBJECT (stream, "leaving the wrong bin");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_get_joined_bin:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the previous joined bin with gst_rtsp_stream_join_bin() or NULL.
+ *
+ * Return: (transfer full) (nullable): the joined bin or NULL.
+ */
+GstBin *
+gst_rtsp_stream_get_joined_bin (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstBin *bin = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ bin = priv->joined_bin ? gst_object_ref (priv->joined_bin) : NULL;
+ g_mutex_unlock (&priv->lock);
+
+ return bin;
+}
+
+/**
+ * gst_rtsp_stream_get_rtpinfo:
+ * @stream: a #GstRTSPStream
+ * @rtptime: (allow-none) (out caller-allocates): result RTP timestamp
+ * @seq: (allow-none) (out caller-allocates): result RTP seqnum
+ * @clock_rate: (allow-none) (out caller-allocates): the clock rate
+ * @running_time: (out caller-allocates): result running-time
+ *
+ * Retrieve the current rtptime, seq and running-time. This is used to
+ * construct a RTPInfo reply header.
+ *
+ * Returns: %TRUE when rtptime, seq and running-time could be determined.
+ */
+gboolean
+gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
+ guint * rtptime, guint * seq, guint * clock_rate,
+ GstClockTime * running_time)
+{
+ GstRTSPStreamPrivate *priv;
+ GstStructure *stats;
+ GObjectClass *payobjclass;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
+
+ g_mutex_lock (&priv->lock);
+
+ /* First try to extract the information from the last buffer on the sinks.
+ * This will have a more accurate sequence number and timestamp, as between
+ * the payloader and the sink there can be some queues
+ */
+ if (priv->udpsink[0] || priv->mcast_udpsink[0] || priv->appsink[0]) {
+ GstSample *last_sample;
+
+ if (priv->udpsink[0])
+ g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
+ else if (priv->mcast_udpsink[0])
+ g_object_get (priv->mcast_udpsink[0], "last-sample", &last_sample, NULL);
+ else
+ g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
+
+ if (last_sample) {
+ GstCaps *caps;
+ GstBuffer *buffer;
+ GstSegment *segment;
+ GstStructure *s;
+ GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
+
+ caps = gst_sample_get_caps (last_sample);
+ buffer = gst_sample_get_buffer (last_sample);
+ segment = gst_sample_get_segment (last_sample);
+ s = gst_caps_get_structure (caps, 0);
+
+ if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
+ guint ssrc_buf = gst_rtp_buffer_get_ssrc (&rtp_buffer);
+ guint ssrc_stream = 0;
+ if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT) &&
+ gst_structure_get_uint (s, "ssrc", &ssrc_stream) &&
+ ssrc_buf != ssrc_stream) {
+ /* Skip buffers from auxiliary streams. */
+ GST_DEBUG_OBJECT (stream,
+ "not a buffer from the payloader, SSRC: %08x", ssrc_buf);
+
+ gst_rtp_buffer_unmap (&rtp_buffer);
+ gst_sample_unref (last_sample);
+ goto stats;
+ }
+
+ if (seq) {
+ *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
+ }
+
+ if (rtptime) {
+ *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
+ }
+
+ gst_rtp_buffer_unmap (&rtp_buffer);
+
+ if (running_time) {
+ *running_time =
+ gst_segment_to_running_time (segment, GST_FORMAT_TIME,
+ GST_BUFFER_TIMESTAMP (buffer));
+ }
+
+ if (clock_rate) {
+ gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
+
+ if (*clock_rate == 0 && running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ gst_sample_unref (last_sample);
+
+ goto done;
+ } else {
+ gst_sample_unref (last_sample);
+ }
+ } else if (priv->blocking) {
+ if (seq) {
+ if (!priv->blocked_buffer)
+ goto stats;
+ *seq = priv->blocked_seqnum;
+ }
+
+ if (rtptime) {
+ if (!priv->blocked_buffer)
+ goto stats;
+ *rtptime = priv->blocked_rtptime;
+ }
+
+ if (running_time) {
+ if (!GST_CLOCK_TIME_IS_VALID (priv->blocked_running_time))
+ goto stats;
+ *running_time = priv->blocked_running_time;
+ }
+
+ if (clock_rate) {
+ *clock_rate = priv->blocked_clock_rate;
+
+ if (*clock_rate == 0 && running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+
+ goto done;
+ }
+ }
+
+stats:
+ if (g_object_class_find_property (payobjclass, "stats")) {
+ g_object_get (priv->payloader, "stats", &stats, NULL);
+ if (stats == NULL)
+ goto no_stats;
+
+ if (seq)
+ gst_structure_get_uint (stats, "seqnum-offset", seq);
+
+ if (rtptime)
+ gst_structure_get_uint (stats, "timestamp", rtptime);
+
+ if (running_time)
+ gst_structure_get_clock_time (stats, "running-time", running_time);
+
+ if (clock_rate) {
+ gst_structure_get_uint (stats, "clock-rate", clock_rate);
+ if (*clock_rate == 0 && running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ gst_structure_free (stats);
+ } else {
+ if (!g_object_class_find_property (payobjclass, "seqnum") ||
+ !g_object_class_find_property (payobjclass, "timestamp"))
+ goto no_stats;
+
+ if (seq)
+ g_object_get (priv->payloader, "seqnum", seq, NULL);
+
+ if (rtptime)
+ g_object_get (priv->payloader, "timestamp", rtptime, NULL);
+
+ if (running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+
+done:
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+ /* ERRORS */
+no_stats:
+ {
+ GST_WARNING ("Could not get payloader stats");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_get_rates:
+ * @stream: a #GstRTSPStream
+ * @rate: (optional) (out caller-allocates): the configured rate
+ * @applied_rate: (optional) (out caller-allocates): the configured applied_rate
+ *
+ * Retrieve the current rate and/or applied_rate.
+ *
+ * Returns: %TRUE if rate and/or applied_rate could be determined.
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_stream_get_rates (GstRTSPStream * stream, gdouble * rate,
+ gdouble * applied_rate)
+{
+ GstRTSPStreamPrivate *priv;
+ GstEvent *event;
+ const GstSegment *segment;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ if (!rate && !applied_rate) {
+ GST_WARNING_OBJECT (stream, "rate and applied_rate are both NULL");
+ return FALSE;
+ }
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+
+ if (!priv->send_rtp_sink)
+ goto no_rtp_sink_pad;
+
+ event = gst_pad_get_sticky_event (priv->send_rtp_sink, GST_EVENT_SEGMENT, 0);
+ if (!event)
+ goto no_sticky_event;
+
+ gst_event_parse_segment (event, &segment);
+ if (rate)
+ *rate = segment->rate;
+ if (applied_rate)
+ *applied_rate = segment->applied_rate;
+
+ gst_event_unref (event);
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+/* ERRORS */
+no_rtp_sink_pad:
+ {
+ GST_WARNING_OBJECT (stream, "no send_rtp_sink pad yet");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+no_sticky_event:
+ {
+ GST_WARNING_OBJECT (stream, "no segment event on send_rtp_sink pad");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+
+}
+
+/**
+ * gst_rtsp_stream_get_caps:
+ * @stream: a #GstRTSPStream
+ *
+ * Retrieve the current caps of @stream.
+ *
+ * Returns: (transfer full) (nullable): the #GstCaps of @stream.
+ * use gst_caps_unref() after usage.
+ */
+GstCaps *
+gst_rtsp_stream_get_caps (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstCaps *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->caps))
+ gst_caps_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_recv_rtp:
+ * @stream: a #GstRTSPStream
+ * @buffer: (transfer full): a #GstBuffer
+ *
+ * Handle an RTP buffer for the stream. This method is usually called when a
+ * message has been received from a client using the TCP transport.
+ *
+ * This function takes ownership of @buffer.
+ *
+ * Returns: a GstFlowReturn.
+ */
+GstFlowReturn
+gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
+{
+ GstRTSPStreamPrivate *priv;
+ GstFlowReturn ret;
+ GstElement *element;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
+ priv = stream->priv;
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+ g_return_val_if_fail (priv->joined_bin != NULL, FALSE);
+
+ g_mutex_lock (&priv->lock);
+ if (priv->appsrc[0])
+ element = gst_object_ref (priv->appsrc[0]);
+ else
+ element = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (element) {
+ if (priv->appsrc_base_time[0] == -1) {
+ /* Take current running_time. This timestamp will be put on
+ * the first buffer of each stream because we are a live source and so we
+ * timestamp with the running_time. When we are dealing with TCP, we also
+ * only timestamp the first buffer (using the DISCONT flag) because a server
+ * typically bursts data, for which we don't want to compensate by speeding
+ * up the media. The other timestamps will be interpollated from this one
+ * using the RTP timestamps. */
+ GST_OBJECT_LOCK (element);
+ if (GST_ELEMENT_CLOCK (element)) {
+ GstClockTime now;
+ GstClockTime base_time;
+
+ now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
+ base_time = GST_ELEMENT_CAST (element)->base_time;
+
+ priv->appsrc_base_time[0] = now - base_time;
+ GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
+ GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
+ ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
+ GST_TIME_ARGS (base_time));
+ }
+ GST_OBJECT_UNLOCK (element);
+ }
+
+ ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
+ gst_object_unref (element);
+ } else {
+ ret = GST_FLOW_OK;
+ }
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_recv_rtcp:
+ * @stream: a #GstRTSPStream
+ * @buffer: (transfer full): a #GstBuffer
+ *
+ * Handle an RTCP buffer for the stream. This method is usually called when a
+ * message has been received from a client using the TCP transport.
+ *
+ * This function takes ownership of @buffer.
+ *
+ * Returns: a GstFlowReturn.
+ */
+GstFlowReturn
+gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
+{
+ GstRTSPStreamPrivate *priv;
+ GstFlowReturn ret;
+ GstElement *element;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
+ priv = stream->priv;
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+
+ if (priv->joined_bin == NULL) {
+ gst_buffer_unref (buffer);
+ return GST_FLOW_NOT_LINKED;
+ }
+ g_mutex_lock (&priv->lock);
+ if (priv->appsrc[1])
+ element = gst_object_ref (priv->appsrc[1]);
+ else
+ element = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (element) {
+ if (priv->appsrc_base_time[1] == -1) {
+ /* Take current running_time. This timestamp will be put on
+ * the first buffer of each stream because we are a live source and so we
+ * timestamp with the running_time. When we are dealing with TCP, we also
+ * only timestamp the first buffer (using the DISCONT flag) because a server
+ * typically bursts data, for which we don't want to compensate by speeding
+ * up the media. The other timestamps will be interpollated from this one
+ * using the RTP timestamps. */
+ GST_OBJECT_LOCK (element);
+ if (GST_ELEMENT_CLOCK (element)) {
+ GstClockTime now;
+ GstClockTime base_time;
+
+ now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
+ base_time = GST_ELEMENT_CAST (element)->base_time;
+
+ priv->appsrc_base_time[1] = now - base_time;
+ GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
+ GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
+ ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
+ GST_TIME_ARGS (base_time));
+ }
+ GST_OBJECT_UNLOCK (element);
+ }
+
+ ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
+ gst_object_unref (element);
+ } else {
+ ret = GST_FLOW_OK;
+ gst_buffer_unref (buffer);
+ }
+ return ret;
+}
+
+/* must be called with lock */
+static inline void
+add_client (GstElement * rtp_sink, GstElement * rtcp_sink, const gchar * host,
+ gint rtp_port, gint rtcp_port)
+{
+ if (rtp_sink != NULL)
+ g_signal_emit_by_name (rtp_sink, "add", host, rtp_port, NULL);
+ if (rtcp_sink != NULL)
+ g_signal_emit_by_name (rtcp_sink, "add", host, rtcp_port, NULL);
+}
+
+/* must be called with lock */
+static void
+remove_client (GstElement * rtp_sink, GstElement * rtcp_sink,
+ const gchar * host, gint rtp_port, gint rtcp_port)
+{
+ if (rtp_sink != NULL)
+ g_signal_emit_by_name (rtp_sink, "remove", host, rtp_port, NULL);
+ if (rtcp_sink != NULL)
+ g_signal_emit_by_name (rtcp_sink, "remove", host, rtcp_port, NULL);
+}
+
+/* must be called with lock */
+static gboolean
+update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
+ gboolean add)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ const GstRTSPTransport *tr;
+ gchar *dest;
+ gint min, max;
+ GList *tr_element;
+
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+ dest = tr->destination;
+
+ tr_element = g_list_find (priv->transports, trans);
+
+ if (add && tr_element)
+ return TRUE;
+ else if (!add && !tr_element)
+ return FALSE;
+
+ switch (tr->lower_transport) {
+ case GST_RTSP_LOWER_TRANS_UDP_MCAST:
+ {
+ min = tr->port.min;
+ max = tr->port.max;
+
+ if (add) {
+ GST_INFO ("adding %s:%d-%d", dest, min, max);
+ if (!check_mcast_client_addr (stream, tr))
+ goto mcast_error;
+ add_client (priv->mcast_udpsink[0], priv->mcast_udpsink[1], dest, min,
+ max);
+
+ if (tr->ttl > 0) {
+ GST_INFO ("setting ttl-mc %d", tr->ttl);
+ if (priv->mcast_udpsink[0])
+ g_object_set (G_OBJECT (priv->mcast_udpsink[0]), "ttl-mc", tr->ttl,
+ NULL);
+ if (priv->mcast_udpsink[1])
+ g_object_set (G_OBJECT (priv->mcast_udpsink[1]), "ttl-mc", tr->ttl,
+ NULL);
+ }
+ priv->transports = g_list_prepend (priv->transports, trans);
+ } else {
+ GST_INFO ("removing %s:%d-%d", dest, min, max);
+ if (!remove_mcast_client_addr (stream, dest, min, max))
+ GST_WARNING_OBJECT (stream,
+ "Failed to remove multicast address: %s:%d-%d", dest, min, max);
+ priv->transports = g_list_delete_link (priv->transports, tr_element);
+ remove_client (priv->mcast_udpsink[0], priv->mcast_udpsink[1], dest,
+ min, max);
+ }
+ break;
+ }
+ case GST_RTSP_LOWER_TRANS_UDP:
+ {
+ if (priv->client_side) {
+ /* In client side mode the 'destination' is the RTSP server, so send
+ * to those ports */
+ min = tr->server_port.min;
+ max = tr->server_port.max;
+ } else {
+ min = tr->client_port.min;
+ max = tr->client_port.max;
+ }
+
+ if (add) {
+ GST_INFO ("adding %s:%d-%d", dest, min, max);
+ add_client (priv->udpsink[0], priv->udpsink[1], dest, min, max);
+ priv->transports = g_list_prepend (priv->transports, trans);
+ } else {
+ GST_INFO ("removing %s:%d-%d", dest, min, max);
+ priv->transports = g_list_delete_link (priv->transports, tr_element);
+ remove_client (priv->udpsink[0], priv->udpsink[1], dest, min, max);
+ }
+ priv->transports_cookie++;
+ break;
+ }
+ case GST_RTSP_LOWER_TRANS_TCP:
+ if (add) {
+ GST_INFO ("adding TCP %s", tr->destination);
+ priv->transports = g_list_prepend (priv->transports, trans);
+ priv->n_tcp_transports++;
+ } else {
+ GST_INFO ("removing TCP %s", tr->destination);
+ priv->transports = g_list_delete_link (priv->transports, tr_element);
+
+ gst_rtsp_stream_transport_lock_backlog (trans);
+ gst_rtsp_stream_transport_clear_backlog (trans);
+ gst_rtsp_stream_transport_unlock_backlog (trans);
+
+ priv->n_tcp_transports--;
+ }
+ priv->transports_cookie++;
+ break;
+ default:
+ goto unknown_transport;
+ }
+ return TRUE;
+
+ /* ERRORS */
+unknown_transport:
+ {
+ GST_INFO ("Unknown transport %d", tr->lower_transport);
+ return FALSE;
+ }
+mcast_error:
+ {
+ return FALSE;
+ }
+}
+
+static void
+on_message_sent (GstRTSPStreamTransport * trans, gpointer user_data)
+{
+ GstRTSPStream *stream = GST_RTSP_STREAM (user_data);
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ GST_DEBUG_OBJECT (stream, "message send complete");
+
+ check_transport_backlog (stream, trans);
+
+ g_mutex_lock (&priv->send_lock);
+ priv->send_cookie++;
+ g_cond_signal (&priv->send_cond);
+ g_mutex_unlock (&priv->send_lock);
+}
+
+/**
+ * gst_rtsp_stream_add_transport:
+ * @stream: a #GstRTSPStream
+ * @trans: (transfer none): a #GstRTSPStreamTransport
+ *
+ * Add the transport in @trans to @stream. The media of @stream will
+ * then also be send to the values configured in @trans. Adding the
+ * same transport twice will not add it a second time.
+ *
+ * @stream must be joined to a bin.
+ *
+ * @trans must contain a valid #GstRTSPTransport.
+ *
+ * Returns: %TRUE if @trans was added
+ */
+gboolean
+gst_rtsp_stream_add_transport (GstRTSPStream * stream,
+ GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ priv = stream->priv;
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+ g_return_val_if_fail (priv->joined_bin != NULL, FALSE);
+
+ g_mutex_lock (&priv->lock);
+ res = update_transport (stream, trans, TRUE);
+ if (res)
+ gst_rtsp_stream_transport_set_message_sent_full (trans, on_message_sent,
+ stream, NULL);
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_remove_transport:
+ * @stream: a #GstRTSPStream
+ * @trans: (transfer none): a #GstRTSPStreamTransport
+ *
+ * Remove the transport in @trans from @stream. The media of @stream will
+ * not be sent to the values configured in @trans.
+ *
+ * @stream must be joined to a bin.
+ *
+ * @trans must contain a valid #GstRTSPTransport.
+ *
+ * Returns: %TRUE if @trans was removed
+ */
+gboolean
+gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
+ GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ priv = stream->priv;
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+ g_return_val_if_fail (priv->joined_bin != NULL, FALSE);
+
+ g_mutex_lock (&priv->lock);
+ res = update_transport (stream, trans, FALSE);
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_update_crypto:
+ * @stream: a #GstRTSPStream
+ * @ssrc: the SSRC
+ * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
+ *
+ * Update the new crypto information for @ssrc in @stream. If information
+ * for @ssrc did not exist, it will be added. If information
+ * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
+ * be removed from @stream.
+ *
+ * Returns: %TRUE if @crypto could be updated
+ */
+gboolean
+gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
+ guint ssrc, GstCaps * crypto)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
+
+ priv = stream->priv;
+
+ GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
+
+ g_mutex_lock (&priv->lock);
+ if (crypto)
+ g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
+ gst_caps_ref (crypto));
+ else
+ g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_get_rtp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GSocket *socket;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+
+ g_mutex_lock (&priv->lock);
+ if (family == G_SOCKET_FAMILY_IPV6)
+ socket = priv->socket_v6[0];
+ else
+ socket = priv->socket_v4[0];
+
+ if (socket != NULL)
+ socket = g_object_ref (socket);
+ g_mutex_unlock (&priv->lock);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_get_rtcp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTCP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GSocket *socket;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+
+ g_mutex_lock (&priv->lock);
+ if (family == G_SOCKET_FAMILY_IPV6)
+ socket = priv->socket_v6[1];
+ else
+ socket = priv->socket_v4[1];
+
+ if (socket != NULL)
+ socket = g_object_ref (socket);
+ g_mutex_unlock (&priv->lock);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_get_rtp_multicast_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the multicast RTP socket from @stream for a @family.
+ *
+ * Returns: (transfer full) (nullable): the multicast RTP socket or %NULL if no
+ *
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtp_multicast_socket (GstRTSPStream * stream,
+ GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GSocket *socket;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+
+ g_mutex_lock (&priv->lock);
+ if (family == G_SOCKET_FAMILY_IPV6)
+ socket = priv->mcast_socket_v6[0];
+ else
+ socket = priv->mcast_socket_v4[0];
+
+ if (socket != NULL)
+ socket = g_object_ref (socket);
+ g_mutex_unlock (&priv->lock);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_get_rtcp_multicast_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the multicast RTCP socket from @stream for a @family.
+ *
+ * Returns: (transfer full) (nullable): the multicast RTCP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ *
+ * Since: 1.14
+ */
+GSocket *
+gst_rtsp_stream_get_rtcp_multicast_socket (GstRTSPStream * stream,
+ GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GSocket *socket;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+
+ g_mutex_lock (&priv->lock);
+ if (family == G_SOCKET_FAMILY_IPV6)
+ socket = priv->mcast_socket_v6[1];
+ else
+ socket = priv->mcast_socket_v4[1];
+
+ if (socket != NULL)
+ socket = g_object_ref (socket);
+ g_mutex_unlock (&priv->lock);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_add_multicast_client_address:
+ * @stream: a #GstRTSPStream
+ * @destination: (transfer none): a multicast address to add
+ * @rtp_port: RTP port
+ * @rtcp_port: RTCP port
+ * @family: socket family
+ *
+ * Add multicast client address to stream. At this point, the sockets that
+ * will stream RTP and RTCP data to @destination are supposed to be
+ * allocated.
+ *
+ * Returns: %TRUE if @destination can be addedd and handled by @stream.
+ *
+ * Since: 1.16
+ */
+gboolean
+gst_rtsp_stream_add_multicast_client_address (GstRTSPStream * stream,
+ const gchar * destination, guint rtp_port, guint rtcp_port,
+ GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (destination != NULL, FALSE);
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ if ((family == G_SOCKET_FAMILY_IPV4) && (priv->mcast_socket_v4[0] == NULL))
+ goto socket_error;
+ else if ((family == G_SOCKET_FAMILY_IPV6) &&
+ (priv->mcast_socket_v6[0] == NULL))
+ goto socket_error;
+
+ if (!add_mcast_client_addr (stream, destination, rtp_port, rtcp_port))
+ goto add_addr_error;
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+socket_error:
+ {
+ GST_WARNING_OBJECT (stream,
+ "Failed to add multicast address: no udp socket");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+add_addr_error:
+ {
+ GST_WARNING_OBJECT (stream,
+ "Failed to add multicast address: invalid address");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_get_multicast_client_addresses
+ * @stream: a #GstRTSPStream
+ *
+ * Get all multicast client addresses that RTP data will be sent to
+ *
+ * Returns: A comma separated list of host:port pairs with destinations
+ *
+ * Since: 1.16
+ */
+gchar *
+gst_rtsp_stream_get_multicast_client_addresses (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GString *str;
+ GList *clients;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+ str = g_string_new ("");
+
+ g_mutex_lock (&priv->lock);
+ clients = priv->mcast_clients;
+ while (clients != NULL) {
+ UdpClientAddrInfo *client;
+
+ client = (UdpClientAddrInfo *) clients->data;
+ clients = g_list_next (clients);
+ g_string_append_printf (str, "%s:%d%s", client->address, client->rtp_port,
+ (clients != NULL ? "," : ""));
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return g_string_free (str, FALSE);
+}
+
+/**
+ * gst_rtsp_stream_set_seqnum:
+ * @stream: a #GstRTSPStream
+ * @seqnum: a new sequence number
+ *
+ * Configure the sequence number in the payloader of @stream to @seqnum.
+ */
+void
+gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
+}
+
+/**
+ * gst_rtsp_stream_get_seqnum:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured sequence number in the payloader of @stream.
+ *
+ * Returns: the sequence number of the payloader.
+ */
+guint16
+gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint seqnum;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
+
+ return seqnum;
+}
+
+/**
+ * gst_rtsp_stream_transport_filter:
+ * @stream: a #GstRTSPStream
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each transport managed by @stream. The result value of @func
+ * determines what happens to the transport. @func will be called with @stream
+ * locked so no further actions on @stream can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
+ * @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
+ *
+ * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
+ * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
+ GstRTSPStreamTransportFilterFunc func, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GList *result, *walk, *next;
+ GHashTable *visited = NULL;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ g_mutex_lock (&priv->lock);
+restart:
+ cookie = priv->transports_cookie;
+ for (walk = priv->transports; walk; walk = next) {
+ GstRTSPStreamTransport *trans = walk->data;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ next = g_list_next (walk);
+
+ if (func) {
+ /* only visit each transport once */
+ if (g_hash_table_contains (visited, trans))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (trans));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (stream, trans, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->transports_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ update_transport (stream, trans, FALSE);
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (trans));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
+
+static GstPadProbeReturn
+pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPStream *stream;
+ GstBuffer *buffer = NULL;
+ GstPadProbeReturn ret = GST_PAD_PROBE_OK;
+ GstEvent *event;
+
+ stream = user_data;
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+
+ if ((info->type & GST_PAD_PROBE_TYPE_BUFFER)) {
+ GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+
+ buffer = gst_pad_probe_info_get_buffer (info);
+ if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)) {
+ priv->blocked_buffer = TRUE;
+ priv->blocked_seqnum = gst_rtp_buffer_get_seq (&rtp);
+ priv->blocked_rtptime = gst_rtp_buffer_get_timestamp (&rtp);
+ gst_rtp_buffer_unmap (&rtp);
+ }
+ priv->position = GST_BUFFER_TIMESTAMP (buffer);
+ } else if ((info->type & GST_PAD_PROBE_TYPE_BUFFER_LIST)) {
+ GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+
+ GstBufferList *list = gst_pad_probe_info_get_buffer_list (info);
+ buffer = gst_buffer_list_get (list, 0);
+ if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)) {
+ priv->blocked_buffer = TRUE;
+ priv->blocked_seqnum = gst_rtp_buffer_get_seq (&rtp);
+ priv->blocked_rtptime = gst_rtp_buffer_get_timestamp (&rtp);
+ gst_rtp_buffer_unmap (&rtp);
+ }
+ priv->position = GST_BUFFER_TIMESTAMP (buffer);
+ } else if ((info->type & GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM)) {
+ if (GST_EVENT_TYPE (info->data) == GST_EVENT_GAP) {
+ gst_event_parse_gap (info->data, &priv->position, NULL);
+ } else {
+ ret = GST_PAD_PROBE_PASS;
+ g_mutex_unlock (&priv->lock);
+ goto done;
+ }
+ } else {
+ g_assert_not_reached ();
+ }
+
+ event = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
+ if (event) {
+ const GstSegment *segment;
+
+ gst_event_parse_segment (event, &segment);
+ priv->blocked_running_time =
+ gst_segment_to_stream_time (segment, GST_FORMAT_TIME, priv->position);
+ gst_event_unref (event);
+ }
+
+ event = gst_pad_get_sticky_event (pad, GST_EVENT_CAPS, 0);
+ if (event) {
+ GstCaps *caps;
+ GstStructure *s;
+
+ gst_event_parse_caps (event, &caps);
+ s = gst_caps_get_structure (caps, 0);
+ gst_structure_get_int (s, "clock-rate", &priv->blocked_clock_rate);
+ gst_event_unref (event);
+ }
+
+ priv->blocking = TRUE;
+
+ GST_DEBUG_OBJECT (pad, "Now blocking");
+
+ GST_DEBUG_OBJECT (stream, "position: %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (priv->position));
+
+ g_mutex_unlock (&priv->lock);
+
+ gst_element_post_message (priv->payloader,
+ gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
+ gst_structure_new ("GstRTSPStreamBlocking", "is_complete",
+ G_TYPE_BOOLEAN, priv->is_complete, NULL)));
+
+done:
+ return ret;
+}
+
+static void
+set_blocked (GstRTSPStream * stream, gboolean blocked)
+{
+ GstRTSPStreamPrivate *priv;
+ int i;
+
+ GST_DEBUG_OBJECT (stream, "blocked: %d", blocked);
+
+ priv = stream->priv;
+
+ if (blocked) {
+ /* if receiver */
+ if (priv->sinkpad) {
+ priv->blocking = TRUE;
+ return;
+ }
+ for (i = 0; i < 2; i++) {
+ if (priv->blocked_id[i] != 0)
+ continue;
+ if (priv->send_src[i]) {
+ priv->blocking = FALSE;
+ priv->blocked_buffer = FALSE;
+ priv->blocked_running_time = GST_CLOCK_TIME_NONE;
+ priv->blocked_clock_rate = 0;
+ priv->blocked_id[i] = gst_pad_add_probe (priv->send_src[i],
+ GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
+ GST_PAD_PROBE_TYPE_BUFFER_LIST |
+ GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, pad_blocking,
+ g_object_ref (stream), g_object_unref);
+ }
+ }
+ } else {
+ for (i = 0; i < 2; i++) {
+ if (priv->blocked_id[i] != 0) {
+ gst_pad_remove_probe (priv->send_src[i], priv->blocked_id[i]);
+ priv->blocked_id[i] = 0;
+ }
+ }
+ priv->blocking = FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_set_blocked:
+ * @stream: a #GstRTSPStream
+ * @blocked: boolean indicating we should block or unblock
+ *
+ * Blocks or unblocks the dataflow on @stream.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ set_blocked (stream, blocked);
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_ublock_linked:
+ * @stream: a #GstRTSPStream
+ *
+ * Unblocks the dataflow on @stream if it is linked.
+ *
+ * Returns: %TRUE on success
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_stream_unblock_linked (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ if (priv->send_src[0] && gst_pad_is_linked (priv->send_src[0]))
+ set_blocked (stream, FALSE);
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_is_blocking:
+ * @stream: a #GstRTSPStream
+ *
+ * Check if @stream is blocking on a #GstBuffer.
+ *
+ * Returns: %TRUE if @stream is blocking
+ */
+gboolean
+gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = priv->blocking;
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_query_position:
+ * @stream: a #GstRTSPStream
+ * @position: (out): current position of a #GstRTSPStream
+ *
+ * Query the position of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the position could be queried
+ */
+gboolean
+gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
+{
+ GstRTSPStreamPrivate *priv;
+ GstElement *sink;
+ GstPad *pad = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ /* query position: if no sinks have been added yet,
+ * we obtain the position from the pad otherwise we query the sinks */
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+
+ if (priv->blocking && GST_CLOCK_TIME_IS_VALID (priv->blocked_running_time)) {
+ *position = priv->blocked_running_time;
+ g_mutex_unlock (&priv->lock);
+ return TRUE;
+ }
+
+ /* depending on the transport type, it should query corresponding sink */
+ if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP)
+ sink = priv->udpsink[0];
+ else if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
+ sink = priv->mcast_udpsink[0];
+ else
+ sink = priv->appsink[0];
+
+ if (sink) {
+ gst_object_ref (sink);
+ } else if (priv->send_src[0]) {
+ pad = gst_object_ref (priv->send_src[0]);
+ } else {
+ g_mutex_unlock (&priv->lock);
+ GST_WARNING_OBJECT (stream, "Couldn't obtain postion: erroneous pipeline");
+ return FALSE;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (sink) {
+ if (!gst_element_query_position (sink, GST_FORMAT_TIME, position)) {
+ GST_WARNING_OBJECT (stream,
+ "Couldn't obtain postion: position query failed");
+ gst_object_unref (sink);
+ return FALSE;
+ }
+ gst_object_unref (sink);
+ } else if (pad) {
+ GstEvent *event;
+ const GstSegment *segment;
+
+ event = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
+ if (!event) {
+ GST_WARNING_OBJECT (stream, "Couldn't obtain postion: no segment event");
+ gst_object_unref (pad);
+ return FALSE;
+ }
+
+ gst_event_parse_segment (event, &segment);
+ if (segment->format != GST_FORMAT_TIME) {
+ *position = -1;
+ } else {
+ g_mutex_lock (&priv->lock);
+ *position = priv->position;
+ g_mutex_unlock (&priv->lock);
+ *position =
+ gst_segment_to_stream_time (segment, GST_FORMAT_TIME, *position);
+ }
+ gst_event_unref (event);
+ gst_object_unref (pad);
+ }
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_query_stop:
+ * @stream: a #GstRTSPStream
+ * @stop: (out): current stop of a #GstRTSPStream
+ *
+ * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the stop could be queried
+ */
+gboolean
+gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
+{
+ GstRTSPStreamPrivate *priv;
+ GstElement *sink;
+ GstPad *pad = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ /* query stop position: if no sinks have been added yet,
+ * we obtain the stop position from the pad otherwise we query the sinks */
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* depending on the transport type, it should query corresponding sink */
+ if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP)
+ sink = priv->udpsink[0];
+ else if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
+ sink = priv->mcast_udpsink[0];
+ else
+ sink = priv->appsink[0];
+
+ if (sink) {
+ gst_object_ref (sink);
+ } else if (priv->send_src[0]) {
+ pad = gst_object_ref (priv->send_src[0]);
+ } else {
+ g_mutex_unlock (&priv->lock);
+ GST_WARNING_OBJECT (stream, "Couldn't obtain stop: erroneous pipeline");
+ return FALSE;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (sink) {
+ GstQuery *query;
+ GstFormat format;
+ gdouble rate;
+ gint64 start_value;
+ gint64 stop_value;
+
+ query = gst_query_new_segment (GST_FORMAT_TIME);
+ if (!gst_element_query (sink, query)) {
+ GST_WARNING_OBJECT (stream, "Couldn't obtain stop: element query failed");
+ gst_query_unref (query);
+ gst_object_unref (sink);
+ return FALSE;
+ }
+ gst_query_parse_segment (query, &rate, &format, &start_value, &stop_value);
+ if (format != GST_FORMAT_TIME)
+ *stop = -1;
+ else
+ *stop = rate > 0.0 ? stop_value : start_value;
+ gst_query_unref (query);
+ gst_object_unref (sink);
+ } else if (pad) {
+ GstEvent *event;
+ const GstSegment *segment;
+
+ event = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
+ if (!event) {
+ GST_WARNING_OBJECT (stream, "Couldn't obtain stop: no segment event");
+ gst_object_unref (pad);
+ return FALSE;
+ }
+ gst_event_parse_segment (event, &segment);
+ if (segment->format != GST_FORMAT_TIME) {
+ *stop = -1;
+ } else {
+ *stop = segment->stop;
+ if (*stop == -1)
+ *stop = segment->duration;
+ else
+ *stop = gst_segment_to_stream_time (segment, GST_FORMAT_TIME, *stop);
+ }
+ gst_event_unref (event);
+ gst_object_unref (pad);
+ }
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_seekable:
+ * @stream: a #GstRTSPStream
+ *
+ * Checks whether the individual @stream is seekable.
+ *
+ * Returns: %TRUE if @stream is seekable, else %FALSE.
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_stream_seekable (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstPad *pad = NULL;
+ GstQuery *query = NULL;
+ gboolean seekable = FALSE;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ /* query stop position: if no sinks have been added yet,
+ * we obtain the stop position from the pad otherwise we query the sinks */
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* depending on the transport type, it should query corresponding sink */
+ if (priv->srcpad) {
+ pad = gst_object_ref (priv->srcpad);
+ } else {
+ g_mutex_unlock (&priv->lock);
+ GST_WARNING_OBJECT (stream, "Pad not available, can't query seekability");
+ goto beach;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ query = gst_query_new_seeking (GST_FORMAT_TIME);
+ if (!gst_pad_query (pad, query)) {
+ GST_WARNING_OBJECT (stream, "seeking query failed");
+ goto beach;
+ }
+ gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL);
+
+beach:
+ if (pad)
+ gst_object_unref (pad);
+ if (query)
+ gst_query_unref (query);
+
+ GST_DEBUG_OBJECT (stream, "Returning %d", seekable);
+
+ return seekable;
+}
+
+/**
+ * gst_rtsp_stream_complete_stream:
+ * @stream: a #GstRTSPStream
+ * @transport: a #GstRTSPTransport
+ *
+ * Add a receiver and sender part to the pipeline based on the transport from
+ * SETUP.
+ *
+ * Returns: %TRUE if the stream has been sucessfully updated.
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_stream_complete_stream (GstRTSPStream * stream,
+ const GstRTSPTransport * transport)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+ GST_DEBUG_OBJECT (stream, "complete stream");
+
+ g_mutex_lock (&priv->lock);
+
+ if (!(priv->allowed_protocols & transport->lower_transport))
+ goto unallowed_transport;
+
+ if (!create_receiver_part (stream, transport))
+ goto create_receiver_error;
+
+ /* in the RECORD case, we only add RTCP sender part */
+ if (!create_sender_part (stream, transport))
+ goto create_sender_error;
+
+ priv->configured_protocols |= transport->lower_transport;
+
+ priv->is_complete = TRUE;
+ g_mutex_unlock (&priv->lock);
+
+ GST_DEBUG_OBJECT (stream, "pipeline sucsessfully updated");
+ return TRUE;
+
+create_receiver_error:
+create_sender_error:
+unallowed_transport:
+ {
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_is_complete:
+ * @stream: a #GstRTSPStream
+ *
+ * Checks whether the stream is complete, contains the receiver and the sender
+ * parts. As the stream contains sink(s) element(s), it's possible to perform
+ * seek operations on it.
+ *
+ * Returns: %TRUE if the stream contains at least one sink element.
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_stream_is_complete (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean ret = FALSE;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ ret = priv->is_complete;
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_is_sender:
+ * @stream: a #GstRTSPStream
+ *
+ * Checks whether the stream is a sender.
+ *
+ * Returns: %TRUE if the stream is a sender and %FALSE otherwise.
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_stream_is_sender (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean ret = FALSE;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ ret = (priv->srcpad != NULL);
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_is_receiver:
+ * @stream: a #GstRTSPStream
+ *
+ * Checks whether the stream is a receiver.
+ *
+ * Returns: %TRUE if the stream is a receiver and %FALSE otherwise.
+ *
+ * Since: 1.14
+ */
+gboolean
+gst_rtsp_stream_is_receiver (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean ret = FALSE;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ ret = (priv->sinkpad != NULL);
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
+
+#define AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
+static gboolean
+mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
+{
+ const gchar *srtp_cipher;
+ const gchar *srtp_auth;
+ const GstMIKEYPayload *sp;
+ guint i;
+
+ /* loop over Security policy until we find one containing policy */
+ for (i = 0;; i++) {
+ if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
+ break;
+
+ if (((GstMIKEYPayloadSP *) sp)->policy == policy)
+ break;
+ }
+
+ /* the default ciphers */
+ srtp_cipher = "aes-128-icm";
+ srtp_auth = "hmac-sha1-80";
+
+ /* now override the defaults with what is in the Security Policy */
+ if (sp != NULL) {
+ guint len;
+ guint enc_alg = GST_MIKEY_ENC_AES_CM_128;
+
+ /* collect all the params and go over them */
+ len = gst_mikey_payload_sp_get_n_params (sp);
+ for (i = 0; i < len; i++) {
+ const GstMIKEYPayloadSPParam *param =
+ gst_mikey_payload_sp_get_param (sp, i);
+
+ switch (param->type) {
+ case GST_MIKEY_SP_SRTP_ENC_ALG:
+ enc_alg = param->val[0];
+ switch (param->val[0]) {
+ case GST_MIKEY_ENC_NULL:
+ srtp_cipher = "null";
+ break;
+ case GST_MIKEY_ENC_AES_CM_128:
+ case GST_MIKEY_ENC_AES_KW_128:
+ srtp_cipher = "aes-128-icm";
+ break;
+ case GST_MIKEY_ENC_AES_GCM_128:
+ srtp_cipher = "aes-128-gcm";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
+ switch (param->val[0]) {
+ case AES_128_KEY_LEN:
+ if (enc_alg == GST_MIKEY_ENC_AES_CM_128 ||
+ enc_alg == GST_MIKEY_ENC_AES_KW_128) {
+ srtp_cipher = "aes-128-icm";
+ } else if (enc_alg == GST_MIKEY_ENC_AES_GCM_128) {
+ srtp_cipher = "aes-128-gcm";
+ }
+ break;
+ case AES_256_KEY_LEN:
+ if (enc_alg == GST_MIKEY_ENC_AES_CM_128 ||
+ enc_alg == GST_MIKEY_ENC_AES_KW_128) {
+ srtp_cipher = "aes-256-icm";
+ } else if (enc_alg == GST_MIKEY_ENC_AES_GCM_128) {
+ srtp_cipher = "aes-256-gcm";
+ }
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_AUTH_ALG:
+ switch (param->val[0]) {
+ case GST_MIKEY_MAC_NULL:
+ srtp_auth = "null";
+ break;
+ case GST_MIKEY_MAC_HMAC_SHA_1_160:
+ srtp_auth = "hmac-sha1-80";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
+ switch (param->val[0]) {
+ case HMAC_32_KEY_LEN:
+ srtp_auth = "hmac-sha1-32";
+ break;
+ case HMAC_80_KEY_LEN:
+ srtp_auth = "hmac-sha1-80";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_SRTP_ENC:
+ break;
+ case GST_MIKEY_SP_SRTP_SRTCP_ENC:
+ break;
+ default:
+ break;
+ }
+ }
+ }
+ /* now configure the SRTP parameters */
+ gst_caps_set_simple (caps,
+ "srtp-cipher", G_TYPE_STRING, srtp_cipher,
+ "srtp-auth", G_TYPE_STRING, srtp_auth,
+ "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
+ "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
+
+ return TRUE;
+}
+
+static gboolean
+handle_mikey_data (GstRTSPStream * stream, guint8 * data, gsize size)
+{
+ GstMIKEYMessage *msg;
+ guint i, n_cs;
+ GstCaps *caps = NULL;
+ GstMIKEYPayloadKEMAC *kemac;
+ const GstMIKEYPayloadKeyData *pkd;
+ GstBuffer *key;
+
+ /* the MIKEY message contains a CSB or crypto session bundle. It is a
+ * set of Crypto Sessions protected with the same master key.
+ * In the context of SRTP, an RTP and its RTCP stream is part of a
+ * crypto session */
+ if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
+ goto parse_failed;
+
+ /* we can only handle SRTP crypto sessions for now */
+ if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
+ goto invalid_map_type;
+
+ /* get the number of crypto sessions. This maps SSRC to its
+ * security parameters */
+ n_cs = gst_mikey_message_get_n_cs (msg);
+ if (n_cs == 0)
+ goto no_crypto_sessions;
+
+ /* we also need keys */
+ if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
+ (msg, GST_MIKEY_PT_KEMAC, 0)))
+ goto no_keys;
+
+ /* we don't support encrypted keys */
+ if (kemac->enc_alg != GST_MIKEY_ENC_NULL
+ || kemac->mac_alg != GST_MIKEY_MAC_NULL)
+ goto unsupported_encryption;
+
+ /* get Key data sub-payload */
+ pkd = (const GstMIKEYPayloadKeyData *)
+ gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
+
+ key = gst_buffer_new_memdup (pkd->key_data, pkd->key_len);
+
+ /* go over all crypto sessions and create the security policy for each
+ * SSRC */
+ for (i = 0; i < n_cs; i++) {
+ const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
+
+ caps = gst_caps_new_simple ("application/x-srtp",
+ "ssrc", G_TYPE_UINT, map->ssrc,
+ "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
+ mikey_apply_policy (caps, msg, map->policy);
+
+ gst_rtsp_stream_update_crypto (stream, map->ssrc, caps);
+ gst_caps_unref (caps);
+ }
+ gst_mikey_message_unref (msg);
+ gst_buffer_unref (key);
+
+ return TRUE;
+
+ /* ERRORS */
+parse_failed:
+ {
+ GST_DEBUG_OBJECT (stream, "failed to parse MIKEY message");
+ return FALSE;
+ }
+invalid_map_type:
+ {
+ GST_DEBUG_OBJECT (stream, "invalid map type %d", msg->map_type);
+ goto cleanup_message;
+ }
+no_crypto_sessions:
+ {
+ GST_DEBUG_OBJECT (stream, "no crypto sessions");
+ goto cleanup_message;
+ }
+no_keys:
+ {
+ GST_DEBUG_OBJECT (stream, "no keys found");
+ goto cleanup_message;
+ }
+unsupported_encryption:
+ {
+ GST_DEBUG_OBJECT (stream, "unsupported key encryption");
+ goto cleanup_message;
+ }
+cleanup_message:
+ {
+ gst_mikey_message_unref (msg);
+ return FALSE;
+ }
+}
+
+#define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
+
+static void
+strip_chars (gchar * str)
+{
+ gchar *s;
+ gsize len;
+
+ len = strlen (str);
+ while (len--) {
+ if (!IS_STRIP_CHAR (str[len]))
+ break;
+ str[len] = '\0';
+ }
+ for (s = str; *s && IS_STRIP_CHAR (*s); s++);
+ memmove (str, s, len + 1);
+}
+
+/**
+ * gst_rtsp_stream_handle_keymgmt:
+ * @stream: a #GstRTSPStream
+ * @keymgmt: a keymgmt header
+ *
+ * Parse and handle a KeyMgmt header.
+ *
+ * Since: 1.16
+ */
+/* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
+ * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
+ */
+gboolean
+gst_rtsp_stream_handle_keymgmt (GstRTSPStream * stream, const gchar * keymgmt)
+{
+ gchar **specs;
+ gint i, j;
+
+ specs = g_strsplit (keymgmt, ",", 0);
+ for (i = 0; specs[i]; i++) {
+ gchar **split;
+
+ split = g_strsplit (specs[i], ";", 0);
+ for (j = 0; split[j]; j++) {
+ g_strstrip (split[j]);
+ if (g_str_has_prefix (split[j], "prot=")) {
+ g_strstrip (split[j] + 5);
+ if (!g_str_equal (split[j] + 5, "mikey"))
+ break;
+ GST_DEBUG ("found mikey");
+ } else if (g_str_has_prefix (split[j], "uri=")) {
+ strip_chars (split[j] + 4);
+ GST_DEBUG ("found uri '%s'", split[j] + 4);
+ } else if (g_str_has_prefix (split[j], "data=")) {
+ guchar *data;
+ gsize size;
+ strip_chars (split[j] + 5);
+ GST_DEBUG ("found data '%s'", split[j] + 5);
+ data = g_base64_decode_inplace (split[j] + 5, &size);
+ handle_mikey_data (stream, data, size);
+ }
+ }
+ g_strfreev (split);
+ }
+ g_strfreev (specs);
+ return TRUE;
+}
+
+
+/**
+ * gst_rtsp_stream_get_ulpfec_pt:
+ *
+ * Returns: the payload type used for ULPFEC protection packets
+ *
+ * Since: 1.16
+ */
+guint
+gst_rtsp_stream_get_ulpfec_pt (GstRTSPStream * stream)
+{
+ guint res;
+
+ g_mutex_lock (&stream->priv->lock);
+ res = stream->priv->ulpfec_pt;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_ulpfec_pt:
+ *
+ * Set the payload type to be used for ULPFEC protection packets
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_stream_set_ulpfec_pt (GstRTSPStream * stream, guint pt)
+{
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->ulpfec_pt = pt;
+ if (stream->priv->ulpfec_encoder) {
+ g_object_set (stream->priv->ulpfec_encoder, "pt", pt, NULL);
+ }
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_request_ulpfec_decoder:
+ *
+ * Creating a rtpulpfecdec element
+ *
+ * Returns: (transfer full) (nullable): a #GstElement.
+ *
+ * Since: 1.16
+ */
+GstElement *
+gst_rtsp_stream_request_ulpfec_decoder (GstRTSPStream * stream,
+ GstElement * rtpbin, guint sessid)
+{
+ GObject *internal_storage = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ stream->priv->ulpfec_decoder =
+ gst_object_ref (gst_element_factory_make ("rtpulpfecdec", NULL));
+
+ g_signal_emit_by_name (G_OBJECT (rtpbin), "get-internal-storage", sessid,
+ &internal_storage);
+ g_object_set (stream->priv->ulpfec_decoder, "storage", internal_storage,
+ NULL);
+ g_object_unref (internal_storage);
+ update_ulpfec_decoder_pt (stream);
+
+ return stream->priv->ulpfec_decoder;
+}
+
+/**
+ * gst_rtsp_stream_request_ulpfec_encoder:
+ *
+ * Creating a rtpulpfecenc element
+ *
+ * Returns: (transfer full) (nullable): a #GstElement.
+ *
+ * Since: 1.16
+ */
+GstElement *
+gst_rtsp_stream_request_ulpfec_encoder (GstRTSPStream * stream, guint sessid)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ if (!stream->priv->ulpfec_percentage)
+ return NULL;
+
+ stream->priv->ulpfec_encoder =
+ gst_object_ref (gst_element_factory_make ("rtpulpfecenc", NULL));
+
+ g_object_set (stream->priv->ulpfec_encoder, "pt", stream->priv->ulpfec_pt,
+ "percentage", stream->priv->ulpfec_percentage, NULL);
+
+ return stream->priv->ulpfec_encoder;
+}
+
+/**
+ * gst_rtsp_stream_set_ulpfec_percentage:
+ *
+ * Sets the amount of redundancy to apply when creating ULPFEC
+ * protection packets.
+ *
+ * Since: 1.16
+ */
+void
+gst_rtsp_stream_set_ulpfec_percentage (GstRTSPStream * stream, guint percentage)
+{
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->ulpfec_percentage = percentage;
+ if (stream->priv->ulpfec_encoder) {
+ g_object_set (stream->priv->ulpfec_encoder, "percentage", percentage, NULL);
+ }
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_ulpfec_percentage:
+ *
+ * Returns: the amount of redundancy applied when creating ULPFEC
+ * protection packets.
+ *
+ * Since: 1.16
+ */
+guint
+gst_rtsp_stream_get_ulpfec_percentage (GstRTSPStream * stream)
+{
+ guint res;
+
+ g_mutex_lock (&stream->priv->lock);
+ res = stream->priv->ulpfec_percentage;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_rate_control:
+ *
+ * Define whether @stream will follow the Rate-Control=no behaviour as specified
+ * in the ONVIF replay spec.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_stream_set_rate_control (GstRTSPStream * stream, gboolean enabled)
+{
+ GST_DEBUG_OBJECT (stream, "%s rate control",
+ enabled ? "Enabling" : "Disabling");
+
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->do_rate_control = enabled;
+ if (stream->priv->appsink[0])
+ g_object_set (stream->priv->appsink[0], "sync", enabled, NULL);
+ if (stream->priv->payloader
+ && g_object_class_find_property (G_OBJECT_GET_CLASS (stream->
+ priv->payloader), "onvif-no-rate-control"))
+ g_object_set (stream->priv->payloader, "onvif-no-rate-control", !enabled,
+ NULL);
+ if (stream->priv->session) {
+ g_object_set (stream->priv->session, "disable-sr-timestamp", !enabled,
+ NULL);
+ }
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_rate_control:
+ *
+ * Returns: whether @stream will follow the Rate-Control=no behaviour as specified
+ * in the ONVIF replay spec.
+ *
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_stream_get_rate_control (GstRTSPStream * stream)
+{
+ gboolean ret;
+
+ g_mutex_lock (&stream->priv->lock);
+ ret = stream->priv->do_rate_control;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_unblock_rtcp:
+ *
+ * Remove blocking probe from the RTCP source. When creating an UDP source for
+ * RTCP it is initially blocked until this function is called.
+ * This functions should be called once the pipeline is ready for handling RTCP
+ * packets.
+ *
+ * Since: 1.20
+ */
+void
+gst_rtsp_stream_unblock_rtcp (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+
+ priv = stream->priv;
+ g_mutex_lock (&priv->lock);
+ if (priv->block_early_rtcp_probe != 0) {
+ gst_pad_remove_probe
+ (priv->block_early_rtcp_pad, priv->block_early_rtcp_probe);
+ priv->block_early_rtcp_probe = 0;
+ gst_object_unref (priv->block_early_rtcp_pad);
+ priv->block_early_rtcp_pad = NULL;
+ }
+ if (priv->block_early_rtcp_probe_ipv6 != 0) {
+ gst_pad_remove_probe
+ (priv->block_early_rtcp_pad_ipv6, priv->block_early_rtcp_probe_ipv6);
+ priv->block_early_rtcp_probe_ipv6 = 0;
+ gst_object_unref (priv->block_early_rtcp_pad_ipv6);
+ priv->block_early_rtcp_pad_ipv6 = NULL;
+ }
+ g_mutex_unlock (&priv->lock);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/rtsp.h>
+#include <gio/gio.h>
+
+#ifndef __GST_RTSP_STREAM_H__
+#define __GST_RTSP_STREAM_H__
+
+#include "rtsp-server-prelude.h"
+
+G_BEGIN_DECLS
+
+/* types for the media stream */
+#define GST_TYPE_RTSP_STREAM (gst_rtsp_stream_get_type ())
+#define GST_IS_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM))
+#define GST_IS_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM))
+#define GST_RTSP_STREAM_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
+#define GST_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStream))
+#define GST_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
+#define GST_RTSP_STREAM_CAST(obj) ((GstRTSPStream*)(obj))
+#define GST_RTSP_STREAM_CLASS_CAST(klass) ((GstRTSPStreamClass*)(klass))
+
+typedef struct _GstRTSPStream GstRTSPStream;
+typedef struct _GstRTSPStreamClass GstRTSPStreamClass;
+typedef struct _GstRTSPStreamPrivate GstRTSPStreamPrivate;
+
+#include "rtsp-stream-transport.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-session.h"
+#include "rtsp-media.h"
+
+/**
+ * GstRTSPStream:
+ *
+ * The definition of a media stream.
+ */
+struct _GstRTSPStream {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPStreamPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstRTSPStreamClass {
+ GObjectClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_stream_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPStream * gst_rtsp_stream_new (guint idx, GstElement *payloader,
+ GstPad *pad);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_stream_get_index (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_stream_get_pt (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+GstPad * gst_rtsp_stream_get_srcpad (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+GstPad * gst_rtsp_stream_get_sinkpad (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_control (GstRTSPStream *stream, const gchar *control);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_stream_get_control (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_has_control (GstRTSPStream *stream, const gchar *control);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_mtu (GstRTSPStream *stream, guint mtu);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_stream_get_mtu (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_dscp_qos (GstRTSPStream *stream, gint dscp_qos);
+
+GST_RTSP_SERVER_API
+gint gst_rtsp_stream_get_dscp_qos (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_is_transport_supported (GstRTSPStream *stream,
+ GstRTSPTransport *transport);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_profiles (GstRTSPStream *stream, GstRTSPProfile profiles);
+
+GST_RTSP_SERVER_API
+GstRTSPProfile gst_rtsp_stream_get_profiles (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_protocols (GstRTSPStream *stream, GstRTSPLowerTrans protocols);
+
+GST_RTSP_SERVER_API
+GstRTSPLowerTrans gst_rtsp_stream_get_protocols (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_address_pool (GstRTSPStream *stream, GstRTSPAddressPool *pool);
+
+GST_RTSP_SERVER_API
+GstRTSPAddressPool *
+ gst_rtsp_stream_get_address_pool (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_multicast_iface (GstRTSPStream *stream, const gchar * multicast_iface);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_stream_get_multicast_iface (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+GstRTSPAddress * gst_rtsp_stream_reserve_address (GstRTSPStream *stream,
+ const gchar * address,
+ guint port,
+ guint n_ports,
+ guint ttl);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_join_bin (GstRTSPStream *stream,
+ GstBin *bin, GstElement *rtpbin,
+ GstState state);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_leave_bin (GstRTSPStream *stream,
+ GstBin *bin, GstElement *rtpbin);
+
+GST_RTSP_SERVER_API
+GstBin * gst_rtsp_stream_get_joined_bin (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_set_blocked (GstRTSPStream * stream,
+ gboolean blocked);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_is_blocking (GstRTSPStream * stream);
+
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_unblock_linked (GstRTSPStream * stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_client_side (GstRTSPStream *stream, gboolean client_side);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_is_client_side (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_get_server_port (GstRTSPStream *stream,
+ GstRTSPRange *server_port,
+ GSocketFamily family);
+
+GST_RTSP_SERVER_API
+GstRTSPAddress * gst_rtsp_stream_get_multicast_address (GstRTSPStream *stream,
+ GSocketFamily family);
+
+
+GST_RTSP_SERVER_API
+GObject * gst_rtsp_stream_get_rtpsession (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+GstElement * gst_rtsp_stream_get_srtp_encoder (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_get_ssrc (GstRTSPStream *stream,
+ guint *ssrc);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_get_rtpinfo (GstRTSPStream *stream,
+ guint *rtptime, guint *seq,
+ guint *clock_rate,
+ GstClockTime *running_time);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_get_rates (GstRTSPStream * stream,
+ gdouble * rate,
+ gdouble * applied_rate);
+
+GST_RTSP_SERVER_API
+GstCaps * gst_rtsp_stream_get_caps (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+GstFlowReturn gst_rtsp_stream_recv_rtp (GstRTSPStream *stream,
+ GstBuffer *buffer);
+
+GST_RTSP_SERVER_API
+GstFlowReturn gst_rtsp_stream_recv_rtcp (GstRTSPStream *stream,
+ GstBuffer *buffer);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_add_transport (GstRTSPStream *stream,
+ GstRTSPStreamTransport *trans);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
+ GstRTSPStreamTransport *trans);
+
+GST_RTSP_SERVER_API
+GSocket * gst_rtsp_stream_get_rtp_socket (GstRTSPStream *stream,
+ GSocketFamily family);
+
+GST_RTSP_SERVER_API
+GSocket * gst_rtsp_stream_get_rtcp_socket (GstRTSPStream *stream,
+ GSocketFamily family);
+
+GST_RTSP_SERVER_API
+GSocket * gst_rtsp_stream_get_rtp_multicast_socket (GstRTSPStream *stream,
+ GSocketFamily family);
+
+GST_RTSP_SERVER_API
+GSocket * gst_rtsp_stream_get_rtcp_multicast_socket (GstRTSPStream *stream,
+ GSocketFamily family);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_add_multicast_client_address (GstRTSPStream * stream,
+ const gchar * destination,
+ guint rtp_port,
+ guint rtcp_port,
+ GSocketFamily family);
+
+GST_RTSP_SERVER_API
+gchar * gst_rtsp_stream_get_multicast_client_addresses (GstRTSPStream * stream);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
+ guint ssrc, GstCaps * crypto);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_query_position (GstRTSPStream * stream,
+ gint64 * position);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_query_stop (GstRTSPStream * stream,
+ gint64 * stop);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_seekable (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_seqnum_offset (GstRTSPStream *stream, guint16 seqnum);
+
+GST_RTSP_SERVER_API
+guint16 gst_rtsp_stream_get_current_seqnum (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_retransmission_time (GstRTSPStream *stream, GstClockTime time);
+
+GST_RTSP_SERVER_API
+GstClockTime gst_rtsp_stream_get_retransmission_time (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream,
+ guint rtx_pt);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_buffer_size (GstRTSPStream *stream, guint size);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_stream_get_buffer_size (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps);
+
+GST_RTSP_SERVER_API
+GstElement * gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid);
+
+GST_RTSP_SERVER_API
+GstElement * gst_rtsp_stream_request_aux_receiver (GstRTSPStream * stream, guint sessid);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream, GSocketFamily family,
+ GstRTSPTransport *transport, gboolean use_client_settings);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_publish_clock_mode (GstRTSPStream * stream, GstRTSPPublishClockMode mode);
+
+GST_RTSP_SERVER_API
+GstRTSPPublishClockMode gst_rtsp_stream_get_publish_clock_mode (GstRTSPStream * stream);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_set_max_mcast_ttl (GstRTSPStream *stream, guint ttl);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_stream_get_max_mcast_ttl (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_verify_mcast_ttl (GstRTSPStream *stream, guint ttl);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_bind_mcast_address (GstRTSPStream * stream, gboolean bind_mcast_addr);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_is_bind_mcast_address (GstRTSPStream * stream);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_complete_stream (GstRTSPStream * stream, const GstRTSPTransport * transport);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_is_complete (GstRTSPStream * stream);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_is_sender (GstRTSPStream * stream);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_is_receiver (GstRTSPStream * stream);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_handle_keymgmt (GstRTSPStream *stream, const gchar *keymgmt);
+
+/* ULP Forward Error Correction (RFC 5109) */
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_get_ulpfec_enabled (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_ulpfec_pt (GstRTSPStream *stream, guint pt);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_stream_get_ulpfec_pt (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+GstElement * gst_rtsp_stream_request_ulpfec_decoder (GstRTSPStream *stream, GstElement *rtpbin, guint sessid);
+
+GST_RTSP_SERVER_API
+GstElement * gst_rtsp_stream_request_ulpfec_encoder (GstRTSPStream *stream, guint sessid);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_ulpfec_percentage (GstRTSPStream *stream, guint percentage);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_stream_get_ulpfec_percentage (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_set_rate_control (GstRTSPStream * stream, gboolean enabled);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_get_rate_control (GstRTSPStream * stream);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_stream_unblock_rtcp (GstRTSPStream * stream);
+
+/**
+ * GstRTSPStreamTransportFilterFunc:
+ * @stream: a #GstRTSPStream object
+ * @trans: a #GstRTSPStreamTransport in @stream
+ * @user_data: user data that has been given to gst_rtsp_stream_transport_filter()
+ *
+ * This function will be called by the gst_rtsp_stream_transport_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @trans will be removed
+ * from @stream.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @trans untouched in
+ * @stream.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @trans to the result #GList of
+ * gst_rtsp_stream_transport_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPStreamTransportFilterFunc) (GstRTSPStream *stream,
+ GstRTSPStreamTransport *trans,
+ gpointer user_data);
+
+GST_RTSP_SERVER_API
+GList * gst_rtsp_stream_transport_filter (GstRTSPStream *stream,
+ GstRTSPStreamTransportFilterFunc func,
+ gpointer user_data);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPStream, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_STREAM_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-thread-pool
+ * @short_description: A pool of threads
+ * @see_also: #GstRTSPMedia, #GstRTSPClient
+ *
+ * A #GstRTSPThreadPool manages reusable threads for various server tasks.
+ * Currently the defined thread types can be found in #GstRTSPThreadType.
+ *
+ * Threads of type #GST_RTSP_THREAD_TYPE_CLIENT are used to handle requests from
+ * a connected client. With gst_rtsp_thread_pool_get_max_threads() a maximum
+ * number of threads can be set after which the pool will start to reuse the
+ * same thread for multiple clients.
+ *
+ * Threads of type #GST_RTSP_THREAD_TYPE_MEDIA will be used to perform the state
+ * changes of the media pipelines and handle its bus messages.
+ *
+ * gst_rtsp_thread_pool_get_thread() can be used to create a #GstRTSPThread
+ * object of the right type. The thread object contains a mainloop and context
+ * that run in a seperate thread and can be used to attached sources to.
+ *
+ * gst_rtsp_thread_reuse() can be used to reuse a thread for multiple purposes.
+ * If all gst_rtsp_thread_reuse() calls are matched with a
+ * gst_rtsp_thread_stop() call, the mainloop will be quit and the thread will
+ * stop.
+ *
+ * To configure the threads, a subclass of this object should be made and the
+ * virtual methods should be overriden to implement the desired functionality.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-thread-pool.h"
+
+typedef struct _GstRTSPThreadImpl
+{
+ GstRTSPThread thread;
+
+ gint reused;
+ GSource *source;
+ /* FIXME, the source has to be part of GstRTSPThreadImpl, due to a bug in GLib:
+ * https://bugzilla.gnome.org/show_bug.cgi?id=720186 */
+} GstRTSPThreadImpl;
+
+GST_DEFINE_MINI_OBJECT_TYPE (GstRTSPThread, gst_rtsp_thread);
+
+static void gst_rtsp_thread_init (GstRTSPThreadImpl * impl);
+
+static void
+_gst_rtsp_thread_free (GstRTSPThreadImpl * impl)
+{
+ GST_DEBUG ("free thread %p", impl);
+
+ g_source_unref (impl->source);
+ g_main_loop_unref (impl->thread.loop);
+ g_main_context_unref (impl->thread.context);
+ g_slice_free1 (sizeof (GstRTSPThreadImpl), impl);
+}
+
+static GstRTSPThread *
+_gst_rtsp_thread_copy (GstRTSPThreadImpl * impl)
+{
+ GstRTSPThreadImpl *copy;
+
+ GST_DEBUG ("copy thread %p", impl);
+
+ copy = g_slice_new0 (GstRTSPThreadImpl);
+ gst_rtsp_thread_init (copy);
+ copy->thread.context = g_main_context_ref (impl->thread.context);
+ copy->thread.loop = g_main_loop_ref (impl->thread.loop);
+
+ return GST_RTSP_THREAD (copy);
+}
+
+static void
+gst_rtsp_thread_init (GstRTSPThreadImpl * impl)
+{
+ gst_mini_object_init (GST_MINI_OBJECT_CAST (impl), 0,
+ GST_TYPE_RTSP_THREAD,
+ (GstMiniObjectCopyFunction) _gst_rtsp_thread_copy, NULL,
+ (GstMiniObjectFreeFunction) _gst_rtsp_thread_free);
+
+ g_atomic_int_set (&impl->reused, 1);
+}
+
+/**
+ * gst_rtsp_thread_new:
+ * @type: the thread type
+ *
+ * Create a new thread object that can run a mainloop.
+ *
+ * Returns: (transfer full): a #GstRTSPThread.
+ */
+GstRTSPThread *
+gst_rtsp_thread_new (GstRTSPThreadType type)
+{
+ GstRTSPThreadImpl *impl;
+
+ impl = g_slice_new0 (GstRTSPThreadImpl);
+
+ gst_rtsp_thread_init (impl);
+ impl->thread.type = type;
+ impl->thread.context = g_main_context_new ();
+ impl->thread.loop = g_main_loop_new (impl->thread.context, TRUE);
+
+ return GST_RTSP_THREAD (impl);
+}
+
+/**
+ * gst_rtsp_thread_reuse:
+ * @thread: (transfer none): a #GstRTSPThread
+ *
+ * Reuse the mainloop of @thread
+ *
+ * Returns: %TRUE if the mainloop could be reused
+ */
+gboolean
+gst_rtsp_thread_reuse (GstRTSPThread * thread)
+{
+ GstRTSPThreadImpl *impl = (GstRTSPThreadImpl *) thread;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_THREAD (thread), FALSE);
+
+ GST_DEBUG ("reuse thread %p", thread);
+
+ res = g_atomic_int_add (&impl->reused, 1) > 0;
+ if (res)
+ gst_rtsp_thread_ref (thread);
+
+ return res;
+}
+
+static gboolean
+do_quit (GstRTSPThread * thread)
+{
+ GST_DEBUG ("stop mainloop of thread %p", thread);
+ g_main_loop_quit (thread->loop);
+ return FALSE;
+}
+
+/**
+ * gst_rtsp_thread_stop:
+ * @thread: (transfer full): a #GstRTSPThread
+ *
+ * Stop and unref @thread. When no threads are using the mainloop, the thread
+ * will be stopped and the final ref to @thread will be released.
+ */
+void
+gst_rtsp_thread_stop (GstRTSPThread * thread)
+{
+ GstRTSPThreadImpl *impl = (GstRTSPThreadImpl *) thread;
+
+ g_return_if_fail (GST_IS_RTSP_THREAD (thread));
+
+ GST_DEBUG ("stop thread %p", thread);
+
+ if (g_atomic_int_dec_and_test (&impl->reused)) {
+ GST_DEBUG ("add idle source to quit mainloop of thread %p", thread);
+ impl->source = g_idle_source_new ();
+ g_source_set_callback (impl->source, (GSourceFunc) do_quit,
+ thread, (GDestroyNotify) gst_rtsp_thread_unref);
+ g_source_attach (impl->source, thread->context);
+ } else
+ gst_rtsp_thread_unref (thread);
+}
+
+struct _GstRTSPThreadPoolPrivate
+{
+ GMutex lock;
+
+ gint max_threads;
+ /* currently used mainloops */
+ GQueue threads;
+};
+
+#define DEFAULT_MAX_THREADS 1
+
+enum
+{
+ PROP_0,
+ PROP_MAX_THREADS,
+ PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_thread_pool_debug);
+#define GST_CAT_DEFAULT rtsp_thread_pool_debug
+
+static GQuark thread_pool;
+
+static void gst_rtsp_thread_pool_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_thread_pool_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_thread_pool_finalize (GObject * obj);
+
+static gpointer do_loop (GstRTSPThread * thread);
+static GstRTSPThread *default_get_thread (GstRTSPThreadPool * pool,
+ GstRTSPThreadType type, GstRTSPContext * ctx);
+
+G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPThreadPool, gst_rtsp_thread_pool,
+ G_TYPE_OBJECT);
+
+static void
+gst_rtsp_thread_pool_class_init (GstRTSPThreadPoolClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_thread_pool_get_property;
+ gobject_class->set_property = gst_rtsp_thread_pool_set_property;
+ gobject_class->finalize = gst_rtsp_thread_pool_finalize;
+
+ /**
+ * GstRTSPThreadPool::max-threads:
+ *
+ * The maximum amount of threads to use for client connections. A value of
+ * 0 means to use only the mainloop, -1 means an unlimited amount of
+ * threads.
+ */
+ g_object_class_install_property (gobject_class, PROP_MAX_THREADS,
+ g_param_spec_int ("max-threads", "Max Threads",
+ "The maximum amount of threads to use for client connections "
+ "(0 = only mainloop, -1 = unlimited)", -1, G_MAXINT,
+ DEFAULT_MAX_THREADS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ klass->get_thread = default_get_thread;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_thread_pool_debug, "rtspthreadpool", 0,
+ "GstRTSPThreadPool");
+
+ thread_pool = g_quark_from_string ("gst.rtsp.thread.pool");
+}
+
+static void
+gst_rtsp_thread_pool_init (GstRTSPThreadPool * pool)
+{
+ GstRTSPThreadPoolPrivate *priv;
+
+ pool->priv = priv = gst_rtsp_thread_pool_get_instance_private (pool);
+
+ g_mutex_init (&priv->lock);
+ priv->max_threads = DEFAULT_MAX_THREADS;
+ g_queue_init (&priv->threads);
+}
+
+static void
+gst_rtsp_thread_pool_finalize (GObject * obj)
+{
+ GstRTSPThreadPool *pool = GST_RTSP_THREAD_POOL (obj);
+ GstRTSPThreadPoolPrivate *priv = pool->priv;
+
+ GST_INFO ("finalize pool %p", pool);
+
+ g_queue_clear (&priv->threads);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_thread_pool_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_thread_pool_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPThreadPool *pool = GST_RTSP_THREAD_POOL (object);
+
+ switch (propid) {
+ case PROP_MAX_THREADS:
+ g_value_set_int (value, gst_rtsp_thread_pool_get_max_threads (pool));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_thread_pool_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPThreadPool *pool = GST_RTSP_THREAD_POOL (object);
+
+ switch (propid) {
+ case PROP_MAX_THREADS:
+ gst_rtsp_thread_pool_set_max_threads (pool, g_value_get_int (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static gpointer
+do_loop (GstRTSPThread * thread)
+{
+ GstRTSPThreadPoolPrivate *priv;
+ GstRTSPThreadPoolClass *klass;
+ GstRTSPThreadPool *pool;
+
+ pool = gst_mini_object_get_qdata (GST_MINI_OBJECT (thread), thread_pool);
+ priv = pool->priv;
+
+ klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+ if (klass->thread_enter)
+ klass->thread_enter (pool, thread);
+
+ GST_INFO ("enter mainloop of thread %p", thread);
+ g_main_loop_run (thread->loop);
+ GST_INFO ("exit mainloop of thread %p", thread);
+
+ if (klass->thread_leave)
+ klass->thread_leave (pool, thread);
+
+ g_mutex_lock (&priv->lock);
+ g_queue_remove (&priv->threads, thread);
+ g_mutex_unlock (&priv->lock);
+
+ gst_rtsp_thread_unref (thread);
+
+ return NULL;
+}
+
+/**
+ * gst_rtsp_thread_pool_new:
+ *
+ * Create a new #GstRTSPThreadPool instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPThreadPool
+ */
+GstRTSPThreadPool *
+gst_rtsp_thread_pool_new (void)
+{
+ GstRTSPThreadPool *result;
+
+ result = g_object_new (GST_TYPE_RTSP_THREAD_POOL, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_thread_pool_set_max_threads:
+ * @pool: a #GstRTSPThreadPool
+ * @max_threads: maximum threads
+ *
+ * Set the maximum threads used by the pool to handle client requests.
+ * A value of 0 will use the pool mainloop, a value of -1 will use an
+ * unlimited number of threads.
+ */
+void
+gst_rtsp_thread_pool_set_max_threads (GstRTSPThreadPool * pool,
+ gint max_threads)
+{
+ GstRTSPThreadPoolPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_THREAD_POOL (pool));
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->max_threads = max_threads;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_thread_pool_get_max_threads:
+ * @pool: a #GstRTSPThreadPool
+ *
+ * Get the maximum number of threads used for client connections.
+ * See gst_rtsp_thread_pool_set_max_threads().
+ *
+ * Returns: the maximum number of threads.
+ */
+gint
+gst_rtsp_thread_pool_get_max_threads (GstRTSPThreadPool * pool)
+{
+ GstRTSPThreadPoolPrivate *priv;
+ gint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_THREAD_POOL (pool), -1);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->max_threads;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+static GstRTSPThread *
+make_thread (GstRTSPThreadPool * pool, GstRTSPThreadType type,
+ GstRTSPContext * ctx)
+{
+ GstRTSPThreadPoolClass *klass;
+ GstRTSPThread *thread;
+
+ klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+ thread = gst_rtsp_thread_new (type);
+ gst_mini_object_set_qdata (GST_MINI_OBJECT (thread), thread_pool,
+ g_object_ref (pool), g_object_unref);
+
+ GST_DEBUG_OBJECT (pool, "new thread %p", thread);
+
+ if (klass->configure_thread)
+ klass->configure_thread (pool, thread, ctx);
+
+ return thread;
+}
+
+static GstRTSPThread *
+default_get_thread (GstRTSPThreadPool * pool,
+ GstRTSPThreadType type, GstRTSPContext * ctx)
+{
+ GstRTSPThreadPoolPrivate *priv = pool->priv;
+ GstRTSPThreadPoolClass *klass;
+ GstRTSPThread *thread;
+ GError *error = NULL;
+
+ klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+ switch (type) {
+ case GST_RTSP_THREAD_TYPE_CLIENT:
+ if (priv->max_threads == 0) {
+ /* no threads allowed */
+ GST_DEBUG_OBJECT (pool, "no client threads allowed");
+ thread = NULL;
+ } else {
+ g_mutex_lock (&priv->lock);
+ retry:
+ if (priv->max_threads > 0 &&
+ g_queue_get_length (&priv->threads) >= priv->max_threads) {
+ /* max threads reached, recycle from queue */
+ thread = g_queue_pop_head (&priv->threads);
+ GST_DEBUG_OBJECT (pool, "recycle client thread %p", thread);
+ if (!gst_rtsp_thread_reuse (thread)) {
+ GST_DEBUG_OBJECT (pool, "thread %p stopping, retry", thread);
+ /* this can happen if we just decremented the reuse counter of the
+ * thread and signaled the mainloop that it should stop. We leave
+ * the thread out of the queue now, there is no point to add it
+ * again, it will be removed from the mainloop otherwise after it
+ * stops. */
+ goto retry;
+ }
+ } else {
+ /* make more threads */
+ GST_DEBUG_OBJECT (pool, "make new client thread");
+ thread = make_thread (pool, type, ctx);
+
+ if (!g_thread_pool_push (klass->pool, gst_rtsp_thread_ref (thread),
+ &error))
+ goto thread_error;
+ }
+ g_queue_push_tail (&priv->threads, thread);
+ g_mutex_unlock (&priv->lock);
+ }
+ break;
+ case GST_RTSP_THREAD_TYPE_MEDIA:
+ GST_DEBUG_OBJECT (pool, "make new media thread");
+ thread = make_thread (pool, type, ctx);
+
+ if (!g_thread_pool_push (klass->pool, gst_rtsp_thread_ref (thread),
+ &error))
+ goto thread_error;
+ break;
+ default:
+ thread = NULL;
+ break;
+ }
+ return thread;
+
+ /* ERRORS */
+thread_error:
+ {
+ GST_ERROR_OBJECT (pool, "failed to push thread %s", error->message);
+ gst_rtsp_thread_unref (thread);
+ /* drop also the ref dedicated for the pool */
+ gst_rtsp_thread_unref (thread);
+ g_clear_error (&error);
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_thread_pool_get_thread:
+ * @pool: a #GstRTSPThreadPool
+ * @type: the #GstRTSPThreadType
+ * @ctx: (transfer none): a #GstRTSPContext
+ *
+ * Get a new #GstRTSPThread for @type and @ctx.
+ *
+ * Returns: (transfer full) (nullable): a new #GstRTSPThread,
+ * gst_rtsp_thread_stop() after usage
+ */
+GstRTSPThread *
+gst_rtsp_thread_pool_get_thread (GstRTSPThreadPool * pool,
+ GstRTSPThreadType type, GstRTSPContext * ctx)
+{
+ GstRTSPThreadPoolClass *klass;
+ GstRTSPThread *result = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_THREAD_POOL (pool), NULL);
+
+ klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+ /* We want to be thread safe as there might be 2 threads wanting to get new
+ * #GstRTSPThread at the same time
+ */
+ if (G_UNLIKELY (!g_atomic_pointer_get (&klass->pool))) {
+ GThreadPool *t_pool;
+ t_pool = g_thread_pool_new ((GFunc) do_loop, klass, -1, FALSE, NULL);
+ if (!g_atomic_pointer_compare_and_exchange (&klass->pool,
+ (GThreadPool *) NULL, t_pool))
+ g_thread_pool_free (t_pool, FALSE, TRUE);
+ }
+
+ if (klass->get_thread)
+ result = klass->get_thread (pool, type, ctx);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_thread_pool_cleanup:
+ *
+ * Wait for all tasks to be stopped and free all allocated resources. This is
+ * mainly used in test suites to ensure proper cleanup of internal data
+ * structures.
+ */
+void
+gst_rtsp_thread_pool_cleanup (void)
+{
+ GstRTSPThreadPoolClass *klass;
+
+ klass =
+ GST_RTSP_THREAD_POOL_CLASS (g_type_class_ref
+ (gst_rtsp_thread_pool_get_type ()));
+ if (klass->pool != NULL) {
+ g_thread_pool_free (klass->pool, FALSE, TRUE);
+ klass->pool = NULL;
+ }
+ g_type_class_unref (klass);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_THREAD_POOL_H__
+#define __GST_RTSP_THREAD_POOL_H__
+
+typedef struct _GstRTSPThread GstRTSPThread;
+typedef struct _GstRTSPThreadPool GstRTSPThreadPool;
+typedef struct _GstRTSPThreadPoolClass GstRTSPThreadPoolClass;
+typedef struct _GstRTSPThreadPoolPrivate GstRTSPThreadPoolPrivate;
+
+#include "rtsp-client.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_THREAD_POOL (gst_rtsp_thread_pool_get_type ())
+#define GST_IS_RTSP_THREAD_POOL(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_THREAD_POOL))
+#define GST_IS_RTSP_THREAD_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_THREAD_POOL))
+#define GST_RTSP_THREAD_POOL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPoolClass))
+#define GST_RTSP_THREAD_POOL(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPool))
+#define GST_RTSP_THREAD_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPoolClass))
+#define GST_RTSP_THREAD_POOL_CAST(obj) ((GstRTSPThreadPool*)(obj))
+#define GST_RTSP_THREAD_POOL_CLASS_CAST(klass) ((GstRTSPThreadPoolClass*)(klass))
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_thread_get_type (void);
+
+#define GST_TYPE_RTSP_THREAD (gst_rtsp_thread_get_type ())
+#define GST_IS_RTSP_THREAD(obj) (GST_IS_MINI_OBJECT_TYPE (obj, GST_TYPE_RTSP_THREAD))
+#define GST_RTSP_THREAD_CAST(obj) ((GstRTSPThread*)(obj))
+#define GST_RTSP_THREAD(obj) (GST_RTSP_THREAD_CAST(obj))
+
+/**
+ * GstRTSPThreadType:
+ * @GST_RTSP_THREAD_TYPE_CLIENT: a thread to handle the client communication
+ * @GST_RTSP_THREAD_TYPE_MEDIA: a thread to handle media
+ *
+ * Different thread types
+ */
+typedef enum
+{
+ GST_RTSP_THREAD_TYPE_CLIENT,
+ GST_RTSP_THREAD_TYPE_MEDIA
+} GstRTSPThreadType;
+
+/**
+ * GstRTSPThread:
+ * @mini_object: parent #GstMiniObject
+ * @type: the thread type
+ * @context: a #GMainContext
+ * @loop: a #GMainLoop
+ *
+ * Structure holding info about a mainloop running in a thread
+ */
+struct _GstRTSPThread {
+ GstMiniObject mini_object;
+
+ GstRTSPThreadType type;
+ GMainContext *context;
+ GMainLoop *loop;
+};
+
+GST_RTSP_SERVER_API
+GstRTSPThread * gst_rtsp_thread_new (GstRTSPThreadType type);
+
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_thread_reuse (GstRTSPThread * thread);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_thread_stop (GstRTSPThread * thread);
+
+/**
+ * gst_rtsp_thread_ref:
+ * @thread: The thread to refcount
+ *
+ * Increase the refcount of this thread.
+ *
+ * Returns: (transfer full): @thread (for convenience when doing assignments)
+ */
+static inline GstRTSPThread *
+gst_rtsp_thread_ref (GstRTSPThread * thread)
+{
+ return (GstRTSPThread *) gst_mini_object_ref (GST_MINI_OBJECT_CAST (thread));
+}
+
+/**
+ * gst_rtsp_thread_unref:
+ * @thread: (transfer full): the thread to refcount
+ *
+ * Decrease the refcount of an thread, freeing it if the refcount reaches 0.
+ */
+static inline void
+gst_rtsp_thread_unref (GstRTSPThread * thread)
+{
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (thread));
+}
+
+/**
+ * GstRTSPThreadPool:
+ *
+ * The thread pool structure.
+ */
+struct _GstRTSPThreadPool {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPThreadPoolPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPThreadPoolClass:
+ * @pool: a #GThreadPool used internally
+ * @get_thread: this function should make or reuse an existing thread that runs
+ * a mainloop.
+ * @configure_thread: configure a thread object. this vmethod is called when
+ * a new thread has been created and should be configured.
+ * @thread_enter: called from the thread when it is entered
+ * @thread_leave: called from the thread when it is left
+ *
+ * Class for managing threads.
+ */
+struct _GstRTSPThreadPoolClass {
+ GObjectClass parent_class;
+
+ GThreadPool *pool;
+
+ GstRTSPThread * (*get_thread) (GstRTSPThreadPool *pool,
+ GstRTSPThreadType type,
+ GstRTSPContext *ctx);
+ void (*configure_thread) (GstRTSPThreadPool *pool,
+ GstRTSPThread * thread,
+ GstRTSPContext *ctx);
+
+ void (*thread_enter) (GstRTSPThreadPool *pool,
+ GstRTSPThread *thread);
+ void (*thread_leave) (GstRTSPThreadPool *pool,
+ GstRTSPThread *thread);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_thread_pool_get_type (void);
+
+GST_RTSP_SERVER_API
+GstRTSPThreadPool * gst_rtsp_thread_pool_new (void);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_thread_pool_set_max_threads (GstRTSPThreadPool * pool, gint max_threads);
+
+GST_RTSP_SERVER_API
+gint gst_rtsp_thread_pool_get_max_threads (GstRTSPThreadPool * pool);
+
+GST_RTSP_SERVER_API
+GstRTSPThread * gst_rtsp_thread_pool_get_thread (GstRTSPThreadPool *pool,
+ GstRTSPThreadType type,
+ GstRTSPContext *ctx);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_thread_pool_cleanup (void);
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPThread, gst_rtsp_thread_unref)
+#endif
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPThreadPool, gst_object_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_THREAD_POOL_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-token
+ * @short_description: Roles and permissions for a client
+ * @see_also: #GstRTSPClient, #GstRTSPPermissions, #GstRTSPAuth
+ *
+ * A #GstRTSPToken contains the permissions and roles of the user
+ * performing the current request. A token is usually created when a user is
+ * authenticated by the #GstRTSPAuth object and is then placed as the current
+ * token for the current request.
+ *
+ * #GstRTSPAuth can use the token and its contents to check authorization for
+ * various operations by comparing the token to the #GstRTSPPermissions of the
+ * object.
+ *
+ * The accepted values of the token are entirely defined by the #GstRTSPAuth
+ * object that implements the security policy.
+ *
+ * Last reviewed on 2013-07-15 (1.0.0)
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "rtsp-token.h"
+
+typedef struct _GstRTSPTokenImpl
+{
+ GstRTSPToken token;
+
+ GstStructure *structure;
+} GstRTSPTokenImpl;
+
+#define GST_RTSP_TOKEN_STRUCTURE(t) (((GstRTSPTokenImpl *)(t))->structure)
+
+//GST_DEBUG_CATEGORY_STATIC (rtsp_token_debug);
+//#define GST_CAT_DEFAULT rtsp_token_debug
+
+GST_DEFINE_MINI_OBJECT_TYPE (GstRTSPToken, gst_rtsp_token);
+
+static void gst_rtsp_token_init (GstRTSPTokenImpl * token,
+ GstStructure * structure);
+
+static void
+_gst_rtsp_token_free (GstRTSPToken * token)
+{
+ GstRTSPTokenImpl *impl = (GstRTSPTokenImpl *) token;
+
+ gst_structure_set_parent_refcount (impl->structure, NULL);
+ gst_structure_free (impl->structure);
+
+ g_slice_free1 (sizeof (GstRTSPTokenImpl), token);
+}
+
+static GstRTSPToken *
+_gst_rtsp_token_copy (GstRTSPTokenImpl * token)
+{
+ GstRTSPTokenImpl *copy;
+ GstStructure *structure;
+
+ structure = gst_structure_copy (token->structure);
+
+ copy = g_slice_new0 (GstRTSPTokenImpl);
+ gst_rtsp_token_init (copy, structure);
+
+ return (GstRTSPToken *) copy;
+}
+
+static void
+gst_rtsp_token_init (GstRTSPTokenImpl * token, GstStructure * structure)
+{
+ gst_mini_object_init (GST_MINI_OBJECT_CAST (token), 0,
+ GST_TYPE_RTSP_TOKEN,
+ (GstMiniObjectCopyFunction) _gst_rtsp_token_copy, NULL,
+ (GstMiniObjectFreeFunction) _gst_rtsp_token_free);
+
+ token->structure = structure;
+ gst_structure_set_parent_refcount (token->structure,
+ &token->token.mini_object.refcount);
+}
+
+/**
+ * gst_rtsp_token_new_empty: (rename-to gst_rtsp_token_new)
+ *
+ * Create a new empty Authorization token.
+ *
+ * Returns: (transfer full): a new empty authorization token.
+ */
+GstRTSPToken *
+gst_rtsp_token_new_empty (void)
+{
+ GstRTSPTokenImpl *token;
+ GstStructure *s;
+
+ s = gst_structure_new_empty ("GstRTSPToken");
+ g_return_val_if_fail (s != NULL, NULL);
+
+ token = g_slice_new0 (GstRTSPTokenImpl);
+ gst_rtsp_token_init (token, s);
+
+ return (GstRTSPToken *) token;
+}
+
+/**
+ * gst_rtsp_token_new: (skip)
+ * @firstfield: the first fieldname
+ * @...: additional arguments
+ *
+ * Create a new Authorization token with the given fieldnames and values.
+ * Arguments are given similar to gst_structure_new().
+ *
+ * Returns: (transfer full): a new authorization token.
+ */
+GstRTSPToken *
+gst_rtsp_token_new (const gchar * firstfield, ...)
+{
+ GstRTSPToken *result;
+ va_list var_args;
+
+ va_start (var_args, firstfield);
+ result = gst_rtsp_token_new_valist (firstfield, var_args);
+ va_end (var_args);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_token_new_valist: (skip)
+ * @firstfield: the first fieldname
+ * @var_args: additional arguments
+ *
+ * Create a new Authorization token with the given fieldnames and values.
+ * Arguments are given similar to gst_structure_new_valist().
+ *
+ * Returns: (transfer full): a new authorization token.
+ */
+GstRTSPToken *
+gst_rtsp_token_new_valist (const gchar * firstfield, va_list var_args)
+{
+ GstRTSPToken *token;
+ GstStructure *s;
+
+ g_return_val_if_fail (firstfield != NULL, NULL);
+
+ token = gst_rtsp_token_new_empty ();
+ s = GST_RTSP_TOKEN_STRUCTURE (token);
+ gst_structure_set_valist (s, firstfield, var_args);
+
+ return token;
+}
+
+/**
+ * gst_rtsp_token_set_string:
+ * @token: The #GstRTSPToken.
+ * @field: field to set
+ * @string_value: string value to set
+ *
+ * Sets a string value on @token.
+ *
+ * Since: 1.14
+ */
+void
+gst_rtsp_token_set_string (GstRTSPToken * token, const gchar * field,
+ const gchar * string_value)
+{
+ GstStructure *s;
+
+ g_return_if_fail (token != NULL);
+ g_return_if_fail (field != NULL);
+ g_return_if_fail (string_value != NULL);
+
+ s = gst_rtsp_token_writable_structure (token);
+ if (s != NULL)
+ gst_structure_set (s, field, G_TYPE_STRING, string_value, NULL);
+}
+
+/**
+ * gst_rtsp_token_set_bool:
+ * @token: The #GstRTSPToken.
+ * @field: field to set
+ * @bool_value: boolean value to set
+ *
+ * Sets a boolean value on @token.
+ *
+ * Since: 1.14
+ */
+void
+gst_rtsp_token_set_bool (GstRTSPToken * token, const gchar * field,
+ gboolean bool_value)
+{
+ GstStructure *s;
+
+ g_return_if_fail (token != NULL);
+ g_return_if_fail (field != NULL);
+
+ s = gst_rtsp_token_writable_structure (token);
+ if (s != NULL)
+ gst_structure_set (s, field, G_TYPE_BOOLEAN, bool_value, NULL);
+}
+
+/**
+ * gst_rtsp_token_get_structure:
+ * @token: The #GstRTSPToken.
+ *
+ * Access the structure of the token.
+ *
+ * Returns: (transfer none): The structure of the token. The structure is still
+ * owned by the token, which means that you should not free it and that the
+ * pointer becomes invalid when you free the token.
+ *
+ * MT safe.
+ */
+const GstStructure *
+gst_rtsp_token_get_structure (GstRTSPToken * token)
+{
+ g_return_val_if_fail (GST_IS_RTSP_TOKEN (token), NULL);
+
+ return GST_RTSP_TOKEN_STRUCTURE (token);
+}
+
+/**
+ * gst_rtsp_token_writable_structure:
+ * @token: The #GstRTSPToken.
+ *
+ * Get a writable version of the structure.
+ *
+ * Returns: (transfer none): The structure of the token. The structure is still
+ * owned by the token, which means that you should not free it and that the
+ * pointer becomes invalid when you free the token. This function checks if
+ * @token is writable and will never return %NULL.
+ *
+ * MT safe.
+ */
+GstStructure *
+gst_rtsp_token_writable_structure (GstRTSPToken * token)
+{
+ g_return_val_if_fail (GST_IS_RTSP_TOKEN (token), NULL);
+ g_return_val_if_fail (gst_mini_object_is_writable (GST_MINI_OBJECT_CAST
+ (token)), NULL);
+
+ return GST_RTSP_TOKEN_STRUCTURE (token);
+}
+
+/**
+ * gst_rtsp_token_get_string:
+ * @token: a #GstRTSPToken
+ * @field: a field name
+ *
+ * Get the string value of @field in @token.
+ *
+ * Returns: (transfer none) (nullable): the string value of @field in
+ * @token or %NULL when @field is not defined in @token. The string
+ * becomes invalid when you free @token.
+ */
+const gchar *
+gst_rtsp_token_get_string (GstRTSPToken * token, const gchar * field)
+{
+ return gst_structure_get_string (GST_RTSP_TOKEN_STRUCTURE (token), field);
+}
+
+/**
+ * gst_rtsp_token_is_allowed:
+ * @token: a #GstRTSPToken
+ * @field: a field name
+ *
+ * Check if @token has a boolean @field and if it is set to %TRUE.
+ *
+ * Returns: %TRUE if @token has a boolean field named @field set to %TRUE.
+ */
+gboolean
+gst_rtsp_token_is_allowed (GstRTSPToken * token, const gchar * field)
+{
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_TOKEN (token), FALSE);
+ g_return_val_if_fail (field != NULL, FALSE);
+
+ if (!gst_structure_get_boolean (GST_RTSP_TOKEN_STRUCTURE (token), field,
+ &result))
+ result = FALSE;
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_TOKEN_H__
+#define __GST_RTSP_TOKEN_H__
+
+typedef struct _GstRTSPToken GstRTSPToken;
+
+#include "rtsp-auth.h"
+
+G_BEGIN_DECLS
+
+GST_RTSP_SERVER_API
+GType gst_rtsp_token_get_type(void);
+
+#define GST_TYPE_RTSP_TOKEN (gst_rtsp_token_get_type())
+#define GST_IS_RTSP_TOKEN(obj) (GST_IS_MINI_OBJECT_TYPE (obj, GST_TYPE_RTSP_TOKEN))
+#define GST_RTSP_TOKEN_CAST(obj) ((GstRTSPToken*)(obj))
+#define GST_RTSP_TOKEN(obj) (GST_RTSP_TOKEN_CAST(obj))
+
+/**
+ * GstRTSPToken:
+ *
+ * An opaque object used for checking authorisations.
+ * It is generated after successful authentication.
+ */
+struct _GstRTSPToken {
+ GstMiniObject mini_object;
+};
+
+/* refcounting */
+/**
+ * gst_rtsp_token_ref:
+ * @token: The token to refcount
+ *
+ * Increase the refcount of this token.
+ *
+ * Returns: (transfer full): @token (for convenience when doing assignments)
+ */
+static inline GstRTSPToken *
+gst_rtsp_token_ref (GstRTSPToken * token)
+{
+ return (GstRTSPToken *) gst_mini_object_ref (GST_MINI_OBJECT_CAST (token));
+}
+
+/**
+ * gst_rtsp_token_unref:
+ * @token: (transfer full): the token to refcount
+ *
+ * Decrease the refcount of an token, freeing it if the refcount reaches 0.
+ */
+static inline void
+gst_rtsp_token_unref (GstRTSPToken * token)
+{
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (token));
+}
+
+
+GST_RTSP_SERVER_API
+GstRTSPToken * gst_rtsp_token_new_empty (void);
+
+GST_RTSP_SERVER_API
+GstRTSPToken * gst_rtsp_token_new (const gchar * firstfield, ...);
+
+GST_RTSP_SERVER_API
+GstRTSPToken * gst_rtsp_token_new_valist (const gchar * firstfield, va_list var_args);
+
+GST_RTSP_SERVER_API
+const GstStructure * gst_rtsp_token_get_structure (GstRTSPToken *token);
+
+GST_RTSP_SERVER_API
+GstStructure * gst_rtsp_token_writable_structure (GstRTSPToken *token);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_token_set_string (GstRTSPToken * token,
+ const gchar * field,
+ const gchar * string_value);
+GST_RTSP_SERVER_API
+const gchar * gst_rtsp_token_get_string (GstRTSPToken *token,
+ const gchar *field);
+GST_RTSP_SERVER_API
+void gst_rtsp_token_set_bool (GstRTSPToken * token,
+ const gchar * field,
+ gboolean bool_value);
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_token_is_allowed (GstRTSPToken *token,
+ const gchar *field);
+
+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPToken, gst_rtsp_token_unref)
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_TOKEN_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
+ * <2006> Lutz Mueller <lutz at topfrose dot de>
+ * <2015> Jan Schmidt <jan at centricular dot com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+/**
+ * SECTION:element-rtspclientsink
+ *
+ * Makes a connection to an RTSP server and send data via RTSP RECORD.
+ * rtspclientsink strictly follows RFC 2326
+ *
+ * RTSP supports transport over TCP or UDP in unicast or multicast mode. By
+ * default rtspclientsink will negotiate a connection in the following order:
+ * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
+ * protocols can be controlled with the #GstRTSPClientSink:protocols property.
+ *
+ * rtspclientsink will internally instantiate an RTP session manager element
+ * that will handle the RTCP messages to and from the server, jitter removal,
+ * and packet reordering.
+ * This feature is implemented using the gstrtpbin element.
+ *
+ * rtspclientsink accepts any stream for which there is an installed payloader,
+ * creates the payloader and manages payload-types, as well as RTX setup.
+ * The new-payloader signal is fired when a payloader is created, in case
+ * an app wants to do custom configuration (such as for MTU).
+ *
+ * ## Example launch line
+ *
+ * |[
+ * gst-launch-1.0 videotestsrc ! jpegenc ! rtspclientsink location=rtsp://some.server/url
+ * ]| Establish a connection to an RTSP server and send JPEG encoded video packets
+ */
+
+/* FIXMEs
+ * - Handle EOS properly and shutdown. The problem with EOS is we don't know
+ * when the server has received all data, so we don't know when to do teardown.
+ * At the moment, we forward EOS to the app as soon as we stop sending. Is there
+ * a way to know from the receiver that it's got all data? Some session timeout?
+ * - Implement extension support for Real / WMS if they support RECORD?
+ * - Add support for network clock synchronised streaming?
+ * - Fix crypto key nego so SAVP/SAVPF profiles work.
+ * - Test (&fix?) HTTP tunnel support
+ * - Add an address pool object for GstRTSPStreams to use for multicast
+ * - Test multicast UDP transport
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#ifdef HAVE_UNISTD_H
+#include <unistd.h>
+#endif /* HAVE_UNISTD_H */
+#include <stdlib.h>
+#include <string.h>
+#include <stdio.h>
+#include <stdarg.h>
+
+#include <gst/net/gstnet.h>
+#include <gst/sdp/gstsdpmessage.h>
+#include <gst/sdp/gstmikey.h>
+#include <gst/rtp/rtp.h>
+
+#include "gstrtspclientsink.h"
+
+typedef struct _GstRtspClientSinkPad GstRtspClientSinkPad;
+typedef GstGhostPadClass GstRtspClientSinkPadClass;
+
+struct _GstRtspClientSinkPad
+{
+ GstGhostPad parent;
+ GstElement *custom_payloader;
+ guint ulpfec_percentage;
+};
+
+enum
+{
+ PROP_PAD_0,
+ PROP_PAD_PAYLOADER,
+ PROP_PAD_ULPFEC_PERCENTAGE
+};
+
+#define DEFAULT_PAD_ULPFEC_PERCENTAGE 0
+
+static GType gst_rtsp_client_sink_pad_get_type (void);
+G_DEFINE_TYPE (GstRtspClientSinkPad, gst_rtsp_client_sink_pad,
+ GST_TYPE_GHOST_PAD);
+#define GST_TYPE_RTSP_CLIENT_SINK_PAD (gst_rtsp_client_sink_pad_get_type ())
+#define GST_RTSP_CLIENT_SINK_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSP_CLIENT_SINK_PAD,GstRtspClientSinkPad))
+
+static void
+gst_rtsp_client_sink_pad_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRtspClientSinkPad *pad;
+
+ pad = GST_RTSP_CLIENT_SINK_PAD (object);
+
+ switch (prop_id) {
+ case PROP_PAD_PAYLOADER:
+ GST_OBJECT_LOCK (pad);
+ if (pad->custom_payloader)
+ gst_object_unref (pad->custom_payloader);
+ pad->custom_payloader = g_value_get_object (value);
+ gst_object_ref_sink (pad->custom_payloader);
+ GST_OBJECT_UNLOCK (pad);
+ break;
+ case PROP_PAD_ULPFEC_PERCENTAGE:
+ GST_OBJECT_LOCK (pad);
+ pad->ulpfec_percentage = g_value_get_uint (value);
+ GST_OBJECT_UNLOCK (pad);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtsp_client_sink_pad_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRtspClientSinkPad *pad;
+
+ pad = GST_RTSP_CLIENT_SINK_PAD (object);
+
+ switch (prop_id) {
+ case PROP_PAD_PAYLOADER:
+ GST_OBJECT_LOCK (pad);
+ g_value_set_object (value, pad->custom_payloader);
+ GST_OBJECT_UNLOCK (pad);
+ break;
+ case PROP_PAD_ULPFEC_PERCENTAGE:
+ GST_OBJECT_LOCK (pad);
+ g_value_set_uint (value, pad->ulpfec_percentage);
+ GST_OBJECT_UNLOCK (pad);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtsp_client_sink_pad_dispose (GObject * object)
+{
+ GstRtspClientSinkPad *pad = GST_RTSP_CLIENT_SINK_PAD (object);
+
+ if (pad->custom_payloader)
+ gst_object_unref (pad->custom_payloader);
+
+ G_OBJECT_CLASS (gst_rtsp_client_sink_pad_parent_class)->dispose (object);
+}
+
+static void
+gst_rtsp_client_sink_pad_class_init (GstRtspClientSinkPadClass * klass)
+{
+ GObjectClass *gobject_klass;
+
+ gobject_klass = (GObjectClass *) klass;
+
+ gobject_klass->set_property = gst_rtsp_client_sink_pad_set_property;
+ gobject_klass->get_property = gst_rtsp_client_sink_pad_get_property;
+ gobject_klass->dispose = gst_rtsp_client_sink_pad_dispose;
+
+ g_object_class_install_property (gobject_klass, PROP_PAD_PAYLOADER,
+ g_param_spec_object ("payloader", "Payloader",
+ "The payloader element to use (NULL = default automatically selected)",
+ GST_TYPE_ELEMENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_klass, PROP_PAD_ULPFEC_PERCENTAGE,
+ g_param_spec_uint ("ulpfec-percentage", "ULPFEC percentage",
+ "The percentage of ULP redundancy to apply", 0, 100,
+ DEFAULT_PAD_ULPFEC_PERCENTAGE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_rtsp_client_sink_pad_init (GstRtspClientSinkPad * pad)
+{
+}
+
+static GstPad *
+gst_rtsp_client_sink_pad_new (const GstPadTemplate * pad_tmpl,
+ const gchar * name)
+{
+ GstRtspClientSinkPad *ret;
+
+ ret =
+ g_object_new (GST_TYPE_RTSP_CLIENT_SINK_PAD, "direction", GST_PAD_SINK,
+ "template", pad_tmpl, "name", name, NULL);
+
+ return GST_PAD (ret);
+}
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_client_sink_debug);
+#define GST_CAT_DEFAULT (rtsp_client_sink_debug)
+
+static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("sink_%u",
+ GST_PAD_SINK,
+ GST_PAD_REQUEST,
+ GST_STATIC_CAPS_ANY); /* Actual caps come from available set of payloaders */
+
+enum
+{
+ SIGNAL_HANDLE_REQUEST,
+ SIGNAL_NEW_MANAGER,
+ SIGNAL_NEW_PAYLOADER,
+ SIGNAL_REQUEST_RTCP_KEY,
+ SIGNAL_ACCEPT_CERTIFICATE,
+ SIGNAL_UPDATE_SDP,
+ LAST_SIGNAL
+};
+
+enum _GstRTSPClientSinkNtpTimeSource
+{
+ NTP_TIME_SOURCE_NTP,
+ NTP_TIME_SOURCE_UNIX,
+ NTP_TIME_SOURCE_RUNNING_TIME,
+ NTP_TIME_SOURCE_CLOCK_TIME
+};
+
+#define GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE (gst_rtsp_client_sink_ntp_time_source_get_type())
+static GType
+gst_rtsp_client_sink_ntp_time_source_get_type (void)
+{
+ static GType ntp_time_source_type = 0;
+ static const GEnumValue ntp_time_source_values[] = {
+ {NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"},
+ {NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"},
+ {NTP_TIME_SOURCE_RUNNING_TIME,
+ "Running time based on pipeline clock",
+ "running-time"},
+ {NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"},
+ {0, NULL, NULL},
+ };
+
+ if (!ntp_time_source_type) {
+ ntp_time_source_type =
+ g_enum_register_static ("GstRTSPClientSinkNtpTimeSource",
+ ntp_time_source_values);
+ }
+ return ntp_time_source_type;
+}
+
+#define DEFAULT_LOCATION NULL
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_DEBUG FALSE
+#define DEFAULT_RETRY 20
+#define DEFAULT_TIMEOUT 5000000
+#define DEFAULT_UDP_BUFFER_SIZE 0x80000
+#define DEFAULT_TCP_TIMEOUT 20000000
+#define DEFAULT_LATENCY_MS 2000
+#define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE
+#define DEFAULT_PROXY NULL
+#define DEFAULT_RTP_BLOCKSIZE 0
+#define DEFAULT_USER_ID NULL
+#define DEFAULT_USER_PW NULL
+#define DEFAULT_PORT_RANGE NULL
+#define DEFAULT_UDP_RECONNECT TRUE
+#define DEFAULT_MULTICAST_IFACE NULL
+#define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
+#define DEFAULT_TLS_DATABASE NULL
+#define DEFAULT_TLS_INTERACTION NULL
+#define DEFAULT_NTP_TIME_SOURCE NTP_TIME_SOURCE_NTP
+#define DEFAULT_USER_AGENT "GStreamer/" PACKAGE_VERSION
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
+#define DEFAULT_RTX_TIME_MS 500
+
+enum
+{
+ PROP_0,
+ PROP_LOCATION,
+ PROP_PROTOCOLS,
+ PROP_DEBUG,
+ PROP_RETRY,
+ PROP_TIMEOUT,
+ PROP_TCP_TIMEOUT,
+ PROP_LATENCY,
+ PROP_RTX_TIME,
+ PROP_DO_RTSP_KEEP_ALIVE,
+ PROP_PROXY,
+ PROP_PROXY_ID,
+ PROP_PROXY_PW,
+ PROP_RTP_BLOCKSIZE,
+ PROP_USER_ID,
+ PROP_USER_PW,
+ PROP_PORT_RANGE,
+ PROP_UDP_BUFFER_SIZE,
+ PROP_UDP_RECONNECT,
+ PROP_MULTICAST_IFACE,
+ PROP_SDES,
+ PROP_TLS_VALIDATION_FLAGS,
+ PROP_TLS_DATABASE,
+ PROP_TLS_INTERACTION,
+ PROP_NTP_TIME_SOURCE,
+ PROP_USER_AGENT,
+ PROP_PROFILES
+};
+
+static void gst_rtsp_client_sink_finalize (GObject * object);
+
+static void gst_rtsp_client_sink_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_client_sink_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+
+static GstClock *gst_rtsp_client_sink_provide_clock (GstElement * element);
+
+static void gst_rtsp_client_sink_uri_handler_init (gpointer g_iface,
+ gpointer iface_data);
+
+static gboolean gst_rtsp_client_sink_set_proxy (GstRTSPClientSink * rtsp,
+ const gchar * proxy);
+static void gst_rtsp_client_sink_set_tcp_timeout (GstRTSPClientSink *
+ rtsp_client_sink, guint64 timeout);
+
+static GstStateChangeReturn gst_rtsp_client_sink_change_state (GstElement *
+ element, GstStateChange transition);
+static void gst_rtsp_client_sink_handle_message (GstBin * bin,
+ GstMessage * message);
+
+static gboolean gst_rtsp_client_sink_setup_auth (GstRTSPClientSink * sink,
+ GstRTSPMessage * response);
+
+static gboolean gst_rtsp_client_sink_loop_send_cmd (GstRTSPClientSink * sink,
+ gint cmd, gint mask);
+
+static GstRTSPResult gst_rtsp_client_sink_open (GstRTSPClientSink * sink,
+ gboolean async);
+static GstRTSPResult gst_rtsp_client_sink_record (GstRTSPClientSink * sink,
+ gboolean async);
+static GstRTSPResult gst_rtsp_client_sink_pause (GstRTSPClientSink * sink,
+ gboolean async);
+static GstRTSPResult gst_rtsp_client_sink_close (GstRTSPClientSink * sink,
+ gboolean async, gboolean only_close);
+static gboolean gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink);
+
+static gboolean gst_rtsp_client_sink_uri_set_uri (GstURIHandler * handler,
+ const gchar * uri, GError ** error);
+static gchar *gst_rtsp_client_sink_uri_get_uri (GstURIHandler * handler);
+
+static gboolean gst_rtsp_client_sink_loop (GstRTSPClientSink * sink);
+static void gst_rtsp_client_sink_connection_flush (GstRTSPClientSink * sink,
+ gboolean flush);
+
+static GstPad *gst_rtsp_client_sink_request_new_pad (GstElement * element,
+ GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
+static void gst_rtsp_client_sink_release_pad (GstElement * element,
+ GstPad * pad);
+
+/* commands we send to out loop to notify it of events */
+#define CMD_OPEN (1 << 0)
+#define CMD_RECORD (1 << 1)
+#define CMD_PAUSE (1 << 2)
+#define CMD_CLOSE (1 << 3)
+#define CMD_WAIT (1 << 4)
+#define CMD_RECONNECT (1 << 5)
+#define CMD_LOOP (1 << 6)
+
+/* mask for all commands */
+#define CMD_ALL ((CMD_LOOP << 1) - 1)
+
+#define GST_ELEMENT_PROGRESS(el, type, code, text) \
+G_STMT_START { \
+ gchar *__txt = _gst_element_error_printf text; \
+ gst_element_post_message (GST_ELEMENT_CAST (el), \
+ gst_message_new_progress (GST_OBJECT_CAST (el), \
+ GST_PROGRESS_TYPE_ ##type, code, __txt)); \
+ g_free (__txt); \
+} G_STMT_END
+
+static guint gst_rtsp_client_sink_signals[LAST_SIGNAL] = { 0 };
+
+/*********************************
+ * GstChildProxy implementation *
+ *********************************/
+static GObject *
+gst_rtsp_client_sink_child_proxy_get_child_by_index (GstChildProxy *
+ child_proxy, guint index)
+{
+ GObject *obj;
+ GstRTSPClientSink *cs = GST_RTSP_CLIENT_SINK (child_proxy);
+
+ GST_OBJECT_LOCK (cs);
+ if ((obj = g_list_nth_data (GST_ELEMENT (cs)->sinkpads, index)))
+ g_object_ref (obj);
+ GST_OBJECT_UNLOCK (cs);
+
+ return obj;
+}
+
+static guint
+gst_rtsp_client_sink_child_proxy_get_children_count (GstChildProxy *
+ child_proxy)
+{
+ guint count = 0;
+
+ GST_OBJECT_LOCK (child_proxy);
+ count = GST_ELEMENT (child_proxy)->numsinkpads;
+ GST_OBJECT_UNLOCK (child_proxy);
+
+ GST_INFO_OBJECT (child_proxy, "Children Count: %d", count);
+
+ return count;
+}
+
+static void
+gst_rtsp_client_sink_child_proxy_init (gpointer g_iface, gpointer iface_data)
+{
+ GstChildProxyInterface *iface = g_iface;
+
+ GST_INFO ("intializing child proxy interface");
+ iface->get_child_by_index =
+ gst_rtsp_client_sink_child_proxy_get_child_by_index;
+ iface->get_children_count =
+ gst_rtsp_client_sink_child_proxy_get_children_count;
+}
+
+#define gst_rtsp_client_sink_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstRTSPClientSink, gst_rtsp_client_sink, GST_TYPE_BIN,
+ G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
+ gst_rtsp_client_sink_uri_handler_init);
+ G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY,
+ gst_rtsp_client_sink_child_proxy_init);
+ );
+
+#ifndef GST_DISABLE_GST_DEBUG
+static inline const gchar *
+cmd_to_string (guint cmd)
+{
+ switch (cmd) {
+ case CMD_OPEN:
+ return "OPEN";
+ case CMD_RECORD:
+ return "RECORD";
+ case CMD_PAUSE:
+ return "PAUSE";
+ case CMD_CLOSE:
+ return "CLOSE";
+ case CMD_WAIT:
+ return "WAIT";
+ case CMD_RECONNECT:
+ return "RECONNECT";
+ case CMD_LOOP:
+ return "LOOP";
+ }
+
+ return "unknown";
+}
+#endif
+
+static void
+gst_rtsp_client_sink_class_init (GstRTSPClientSinkClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBinClass *gstbin_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbin_class = (GstBinClass *) klass;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_client_sink_debug, "rtspclientsink", 0,
+ "RTSP sink element");
+
+ gobject_class->set_property = gst_rtsp_client_sink_set_property;
+ gobject_class->get_property = gst_rtsp_client_sink_get_property;
+
+ gobject_class->finalize = gst_rtsp_client_sink_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_LOCATION,
+ g_param_spec_string ("location", "RTSP Location",
+ "Location of the RTSP url to read",
+ DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed RTSP profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_DEBUG,
+ g_param_spec_boolean ("debug", "Debug",
+ "Dump request and response messages to stdout",
+ DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_RETRY,
+ g_param_spec_uint ("retry", "Retry",
+ "Max number of retries when allocating RTP ports.",
+ 0, G_MAXUINT16, DEFAULT_RETRY,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TIMEOUT,
+ g_param_spec_uint64 ("timeout", "Timeout",
+ "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
+ 0, G_MAXUINT64, DEFAULT_TIMEOUT,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
+ g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
+ "Fail after timeout microseconds on TCP connections (0 = disabled)",
+ 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_LATENCY,
+ g_param_spec_uint ("latency", "Buffer latency in ms",
+ "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_RTX_TIME,
+ g_param_spec_uint ("rtx-time", "Retransmission buffer in ms",
+ "Amount of ms to buffer for retransmission. 0 disables retransmission",
+ 0, G_MAXUINT, DEFAULT_RTX_TIME_MS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink:do-rtsp-keep-alive:
+ *
+ * Enable RTSP keep alive support. Some old server don't like RTSP
+ * keep alive and then this property needs to be set to FALSE.
+ */
+ g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
+ g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
+ "Send RTSP keep alive packets, disable for old incompatible server.",
+ DEFAULT_DO_RTSP_KEEP_ALIVE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink:proxy:
+ *
+ * Set the proxy parameters. This has to be a string of the format
+ * [http://][user:passwd@]host[:port].
+ */
+ g_object_class_install_property (gobject_class, PROP_PROXY,
+ g_param_spec_string ("proxy", "Proxy",
+ "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
+ DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPClientSink:proxy-id:
+ *
+ * Sets the proxy URI user id for authentication. If the URI set via the
+ * "proxy" property contains a user-id already, that will take precedence.
+ *
+ */
+ g_object_class_install_property (gobject_class, PROP_PROXY_ID,
+ g_param_spec_string ("proxy-id", "proxy-id",
+ "HTTP proxy URI user id for authentication", "",
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPClientSink:proxy-pw:
+ *
+ * Sets the proxy URI password for authentication. If the URI set via the
+ * "proxy" property contains a password already, that will take precedence.
+ *
+ */
+ g_object_class_install_property (gobject_class, PROP_PROXY_PW,
+ g_param_spec_string ("proxy-pw", "proxy-pw",
+ "HTTP proxy URI user password for authentication", "",
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink:rtp-blocksize:
+ *
+ * RTP package size to suggest to server.
+ */
+ g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
+ g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
+ "RTP package size to suggest to server (0 = disabled)",
+ 0, 65536, DEFAULT_RTP_BLOCKSIZE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class,
+ PROP_USER_ID,
+ g_param_spec_string ("user-id", "user-id",
+ "RTSP location URI user id for authentication", DEFAULT_USER_ID,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_USER_PW,
+ g_param_spec_string ("user-pw", "user-pw",
+ "RTSP location URI user password for authentication", DEFAULT_USER_PW,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink:port-range:
+ *
+ * Configure the client port numbers that can be used to receive
+ * RTCP.
+ */
+ g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
+ g_param_spec_string ("port-range", "Port range",
+ "Client port range that can be used to receive RTCP data, "
+ "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink:udp-buffer-size:
+ *
+ * Size of the kernel UDP receive buffer in bytes.
+ */
+ g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
+ g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
+ "Size of the kernel UDP receive buffer in bytes, 0=default",
+ 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
+ g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
+ "Reconnect to the server if RTSP connection is closed when doing UDP",
+ DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
+ g_param_spec_string ("multicast-iface", "Multicast Interface",
+ "The network interface on which to join the multicast group",
+ DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_SDES,
+ g_param_spec_boxed ("sdes", "SDES",
+ "The SDES items of this session",
+ GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink::tls-validation-flags:
+ *
+ * TLS certificate validation flags used to validate server
+ * certificate.
+ *
+ */
+ g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
+ g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
+ "TLS certificate validation flags used to validate the server certificate",
+ G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink::tls-database:
+ *
+ * TLS database with anchor certificate authorities used to validate
+ * the server certificate.
+ *
+ */
+ g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
+ g_param_spec_object ("tls-database", "TLS database",
+ "TLS database with anchor certificate authorities used to validate the server certificate",
+ G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink::tls-interaction:
+ *
+ * A #GTlsInteraction object to be used when the connection or certificate
+ * database need to interact with the user. This will be used to prompt the
+ * user for passwords where necessary.
+ *
+ */
+ g_object_class_install_property (gobject_class, PROP_TLS_INTERACTION,
+ g_param_spec_object ("tls-interaction", "TLS interaction",
+ "A GTlsInteraction object to prompt the user for password or certificate",
+ G_TYPE_TLS_INTERACTION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink::ntp-time-source:
+ *
+ * allows to select the time source that should be used
+ * for the NTP time in outgoing packets
+ *
+ */
+ g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
+ g_param_spec_enum ("ntp-time-source", "NTP Time Source",
+ "NTP time source for RTCP packets",
+ GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE, DEFAULT_NTP_TIME_SOURCE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink::user-agent:
+ *
+ * The string to set in the User-Agent header.
+ *
+ */
+ g_object_class_install_property (gobject_class, PROP_USER_AGENT,
+ g_param_spec_string ("user-agent", "User Agent",
+ "The User-Agent string to send to the server",
+ DEFAULT_USER_AGENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPClientSink::handle-request:
+ * @rtsp_client_sink: a #GstRTSPClientSink
+ * @request: a #GstRTSPMessage
+ * @response: a #GstRTSPMessage
+ *
+ * Handle a server request in @request and prepare @response.
+ *
+ * This signal is called from the streaming thread, you should therefore not
+ * do any state changes on @rtsp_client_sink because this might deadlock. If you want
+ * to modify the state as a result of this signal, post a
+ * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
+ * in some other way.
+ *
+ */
+ gst_rtsp_client_sink_signals[SIGNAL_HANDLE_REQUEST] =
+ g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
+ 0, NULL, NULL, NULL, G_TYPE_NONE, 2,
+ GST_TYPE_RTSP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE,
+ GST_TYPE_RTSP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
+
+ /**
+ * GstRTSPClientSink::new-manager:
+ * @rtsp_client_sink: a #GstRTSPClientSink
+ * @manager: a #GstElement
+ *
+ * Emitted after a new manager (like rtpbin) was created and the default
+ * properties were configured.
+ *
+ */
+ gst_rtsp_client_sink_signals[SIGNAL_NEW_MANAGER] =
+ g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_FIRST, 0, NULL, NULL, NULL,
+ G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+ /**
+ * GstRTSPClientSink::new-payloader:
+ * @rtsp_client_sink: a #GstRTSPClientSink
+ * @payloader: a #GstElement
+ *
+ * Emitted after a new RTP payloader was created and the default
+ * properties were configured.
+ *
+ */
+ gst_rtsp_client_sink_signals[SIGNAL_NEW_PAYLOADER] =
+ g_signal_new_class_handler ("new-payloader", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_FIRST, 0, NULL, NULL, NULL,
+ G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+ /**
+ * GstRTSPClientSink::request-rtcp-key:
+ * @rtsp_client_sink: a #GstRTSPClientSink
+ * @num: the stream number
+ *
+ * Signal emitted to get the crypto parameters relevant to the RTCP
+ * stream. User should provide the key and the RTCP encryption ciphers
+ * and authentication, and return them wrapped in a GstCaps.
+ *
+ */
+ gst_rtsp_client_sink_signals[SIGNAL_REQUEST_RTCP_KEY] =
+ g_signal_new ("request-rtcp-key", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
+
+ /**
+ * GstRTSPClientSink::accept-certificate:
+ * @rtsp_client_sink: a #GstRTSPClientSink
+ * @peer_cert: the peer's #GTlsCertificate
+ * @errors: the problems with @peer_cert
+ * @user_data: user data set when the signal handler was connected.
+ *
+ * This will directly map to #GTlsConnection 's "accept-certificate"
+ * signal and be performed after the default checks of #GstRTSPConnection
+ * (checking against the #GTlsDatabase with the given #GTlsCertificateFlags)
+ * have failed. If no #GTlsDatabase is set on this connection, only this
+ * signal will be emitted.
+ *
+ * Since: 1.14
+ */
+ gst_rtsp_client_sink_signals[SIGNAL_ACCEPT_CERTIFICATE] =
+ g_signal_new ("accept-certificate", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, g_signal_accumulator_true_handled, NULL, NULL,
+ G_TYPE_BOOLEAN, 3, G_TYPE_TLS_CONNECTION, G_TYPE_TLS_CERTIFICATE,
+ G_TYPE_TLS_CERTIFICATE_FLAGS);
+
+ /**
+ * GstRTSPClientSink::update-sdp:
+ * @rtsp_client_sink: a #GstRTSPClientSink
+ * @sdp: a #GstSDPMessage
+ *
+ * Emitted right before the ANNOUNCE request is sent to the server with the
+ * generated SDP. The SDP can be updated from signal handlers but the order
+ * and number of medias must not be changed.
+ *
+ * Since: 1.20
+ */
+ gst_rtsp_client_sink_signals[SIGNAL_UPDATE_SDP] =
+ g_signal_new_class_handler ("update-sdp", G_TYPE_FROM_CLASS (klass),
+ 0, 0, NULL, NULL, NULL,
+ G_TYPE_NONE, 1, GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
+
+ gstelement_class->provide_clock = gst_rtsp_client_sink_provide_clock;
+ gstelement_class->change_state = gst_rtsp_client_sink_change_state;
+ gstelement_class->request_new_pad =
+ GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_request_new_pad);
+ gstelement_class->release_pad =
+ GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_release_pad);
+
+ gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
+ &rtptemplate, GST_TYPE_RTSP_CLIENT_SINK_PAD);
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "RTSP RECORD client", "Sink/Network",
+ "Send data over the network via RTSP RECORD(RFC 2326)",
+ "Jan Schmidt <jan@centricular.com>");
+
+ gstbin_class->handle_message = gst_rtsp_client_sink_handle_message;
+
+ gst_type_mark_as_plugin_api (GST_TYPE_RTSP_CLIENT_SINK_PAD, 0);
+ gst_type_mark_as_plugin_api (GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE, 0);
+}
+
+static void
+gst_rtsp_client_sink_init (GstRTSPClientSink * sink)
+{
+ sink->conninfo.location = g_strdup (DEFAULT_LOCATION);
+ sink->protocols = DEFAULT_PROTOCOLS;
+ sink->debug = DEFAULT_DEBUG;
+ sink->retry = DEFAULT_RETRY;
+ sink->udp_timeout = DEFAULT_TIMEOUT;
+ gst_rtsp_client_sink_set_tcp_timeout (sink, DEFAULT_TCP_TIMEOUT);
+ sink->latency = DEFAULT_LATENCY_MS;
+ sink->rtx_time = DEFAULT_RTX_TIME_MS;
+ sink->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
+ gst_rtsp_client_sink_set_proxy (sink, DEFAULT_PROXY);
+ sink->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
+ sink->user_id = g_strdup (DEFAULT_USER_ID);
+ sink->user_pw = g_strdup (DEFAULT_USER_PW);
+ sink->client_port_range.min = 0;
+ sink->client_port_range.max = 0;
+ sink->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
+ sink->udp_reconnect = DEFAULT_UDP_RECONNECT;
+ sink->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
+ sink->sdes = NULL;
+ sink->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
+ sink->tls_database = DEFAULT_TLS_DATABASE;
+ sink->tls_interaction = DEFAULT_TLS_INTERACTION;
+ sink->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
+ sink->user_agent = g_strdup (DEFAULT_USER_AGENT);
+
+ sink->profiles = DEFAULT_PROFILES;
+
+ /* protects the streaming thread in interleaved mode or the polling
+ * thread in UDP mode. */
+ g_rec_mutex_init (&sink->stream_rec_lock);
+
+ /* protects our state changes from multiple invocations */
+ g_rec_mutex_init (&sink->state_rec_lock);
+
+ g_mutex_init (&sink->send_lock);
+
+ g_mutex_init (&sink->preroll_lock);
+ g_cond_init (&sink->preroll_cond);
+
+ sink->state = GST_RTSP_STATE_INVALID;
+
+ g_mutex_init (&sink->conninfo.send_lock);
+ g_mutex_init (&sink->conninfo.recv_lock);
+
+ g_mutex_init (&sink->block_streams_lock);
+ g_cond_init (&sink->block_streams_cond);
+
+ g_mutex_init (&sink->open_conn_lock);
+ g_cond_init (&sink->open_conn_cond);
+
+ sink->internal_bin = (GstBin *) gst_bin_new ("rtspbin");
+ g_object_set (sink->internal_bin, "async-handling", TRUE, NULL);
+ gst_element_set_locked_state (GST_ELEMENT_CAST (sink->internal_bin), TRUE);
+ gst_bin_add (GST_BIN (sink), GST_ELEMENT_CAST (sink->internal_bin));
+
+ sink->next_dyn_pt = 96;
+
+ gst_sdp_message_init (&sink->cursdp);
+
+ GST_OBJECT_FLAG_SET (sink, GST_ELEMENT_FLAG_SINK);
+}
+
+static void
+gst_rtsp_client_sink_finalize (GObject * object)
+{
+ GstRTSPClientSink *rtsp_client_sink;
+
+ rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
+
+ gst_sdp_message_uninit (&rtsp_client_sink->cursdp);
+
+ g_free (rtsp_client_sink->conninfo.location);
+ gst_rtsp_url_free (rtsp_client_sink->conninfo.url);
+ g_free (rtsp_client_sink->conninfo.url_str);
+ g_free (rtsp_client_sink->user_id);
+ g_free (rtsp_client_sink->user_pw);
+ g_free (rtsp_client_sink->multi_iface);
+ g_free (rtsp_client_sink->user_agent);
+
+ if (rtsp_client_sink->uri_sdp) {
+ gst_sdp_message_free (rtsp_client_sink->uri_sdp);
+ rtsp_client_sink->uri_sdp = NULL;
+ }
+ if (rtsp_client_sink->provided_clock)
+ gst_object_unref (rtsp_client_sink->provided_clock);
+
+ if (rtsp_client_sink->sdes)
+ gst_structure_free (rtsp_client_sink->sdes);
+
+ if (rtsp_client_sink->tls_database)
+ g_object_unref (rtsp_client_sink->tls_database);
+
+ if (rtsp_client_sink->tls_interaction)
+ g_object_unref (rtsp_client_sink->tls_interaction);
+
+ /* free locks */
+ g_rec_mutex_clear (&rtsp_client_sink->stream_rec_lock);
+ g_rec_mutex_clear (&rtsp_client_sink->state_rec_lock);
+
+ g_mutex_clear (&rtsp_client_sink->conninfo.send_lock);
+ g_mutex_clear (&rtsp_client_sink->conninfo.recv_lock);
+
+ g_mutex_clear (&rtsp_client_sink->send_lock);
+
+ g_mutex_clear (&rtsp_client_sink->preroll_lock);
+ g_cond_clear (&rtsp_client_sink->preroll_cond);
+
+ g_mutex_clear (&rtsp_client_sink->block_streams_lock);
+ g_cond_clear (&rtsp_client_sink->block_streams_cond);
+
+ g_mutex_clear (&rtsp_client_sink->open_conn_lock);
+ g_cond_clear (&rtsp_client_sink->open_conn_cond);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_rtp_payloader_filter_func (GstPluginFeature * feature, gpointer user_data)
+{
+ GstElementFactory *factory = NULL;
+ const gchar *klass;
+
+ if (!GST_IS_ELEMENT_FACTORY (feature))
+ return FALSE;
+
+ factory = GST_ELEMENT_FACTORY (feature);
+
+ if (gst_plugin_feature_get_rank (feature) == GST_RANK_NONE)
+ return FALSE;
+
+ if (!gst_element_factory_list_is_type (factory,
+ GST_ELEMENT_FACTORY_TYPE_PAYLOADER))
+ return FALSE;
+
+ klass =
+ gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_KLASS);
+ if (strstr (klass, "Codec") == NULL)
+ return FALSE;
+ if (strstr (klass, "RTP") == NULL)
+ return FALSE;
+
+ return TRUE;
+}
+
+static gint
+compare_ranks (GstPluginFeature * f1, GstPluginFeature * f2)
+{
+ gint diff;
+ const gchar *rname1, *rname2;
+ GstRank rank1, rank2;
+
+ rname1 = gst_plugin_feature_get_name (f1);
+ rname2 = gst_plugin_feature_get_name (f2);
+
+ rank1 = gst_plugin_feature_get_rank (f1);
+ rank2 = gst_plugin_feature_get_rank (f2);
+
+ /* HACK: Prefer rtpmp4apay over rtpmp4gpay */
+ if (g_str_equal (rname1, "rtpmp4apay"))
+ rank1 = GST_RANK_SECONDARY + 1;
+ if (g_str_equal (rname2, "rtpmp4apay"))
+ rank2 = GST_RANK_SECONDARY + 1;
+
+ diff = rank2 - rank1;
+ if (diff != 0)
+ return diff;
+
+ diff = strcmp (rname2, rname1);
+
+ return diff;
+}
+
+static GList *
+gst_rtsp_client_sink_get_factories (void)
+{
+ static GList *payloader_factories = NULL;
+
+ if (g_once_init_enter (&payloader_factories)) {
+ GList *all_factories;
+
+ all_factories =
+ gst_registry_feature_filter (gst_registry_get (),
+ gst_rtp_payloader_filter_func, FALSE, NULL);
+
+ all_factories = g_list_sort (all_factories, (GCompareFunc) compare_ranks);
+
+ g_once_init_leave (&payloader_factories, all_factories);
+ }
+
+ return payloader_factories;
+}
+
+static GstCaps *
+gst_rtsp_client_sink_get_payloader_caps (GstElementFactory * factory)
+{
+ const GList *tmp;
+ GstCaps *caps = gst_caps_new_empty ();
+
+ for (tmp = gst_element_factory_get_static_pad_templates (factory);
+ tmp; tmp = g_list_next (tmp)) {
+ GstStaticPadTemplate *template = tmp->data;
+
+ if (template->direction == GST_PAD_SINK) {
+ GstCaps *static_caps = gst_static_pad_template_get_caps (template);
+
+ GST_LOG ("Found pad template %s on factory %s",
+ template->name_template, gst_plugin_feature_get_name (factory));
+
+ if (static_caps)
+ caps = gst_caps_merge (caps, static_caps);
+
+ /* Early out, any is absorbing */
+ if (gst_caps_is_any (caps))
+ goto out;
+ }
+ }
+
+out:
+ return caps;
+}
+
+static GstCaps *
+gst_rtsp_client_sink_get_all_payloaders_caps (void)
+{
+ /* Cached caps result */
+ static GstCaps *ret;
+
+ if (g_once_init_enter (&ret)) {
+ GList *factories, *cur;
+ GstCaps *caps = gst_caps_new_empty ();
+
+ factories = gst_rtsp_client_sink_get_factories ();
+ for (cur = factories; cur != NULL; cur = g_list_next (cur)) {
+ GstElementFactory *factory = GST_ELEMENT_FACTORY (cur->data);
+ GstCaps *payloader_caps =
+ gst_rtsp_client_sink_get_payloader_caps (factory);
+
+ caps = gst_caps_merge (caps, payloader_caps);
+
+ /* Early out, any is absorbing */
+ if (gst_caps_is_any (caps))
+ goto out;
+ }
+
+ GST_MINI_OBJECT_FLAG_SET (caps, GST_MINI_OBJECT_FLAG_MAY_BE_LEAKED);
+
+ out:
+ g_once_init_leave (&ret, caps);
+ }
+
+ /* Return cached result */
+ return gst_caps_ref (ret);
+}
+
+static GstElement *
+gst_rtsp_client_sink_make_payloader (GstCaps * caps)
+{
+ GList *factories, *cur;
+
+ factories = gst_rtsp_client_sink_get_factories ();
+ for (cur = factories; cur != NULL; cur = g_list_next (cur)) {
+ GstElementFactory *factory = GST_ELEMENT_FACTORY (cur->data);
+ const GList *tmp;
+
+ for (tmp = gst_element_factory_get_static_pad_templates (factory);
+ tmp; tmp = g_list_next (tmp)) {
+ GstStaticPadTemplate *template = tmp->data;
+
+ if (template->direction == GST_PAD_SINK) {
+ GstCaps *static_caps = gst_static_pad_template_get_caps (template);
+ GstElement *payloader = NULL;
+
+ if (gst_caps_can_intersect (static_caps, caps)) {
+ GST_DEBUG ("caps %" GST_PTR_FORMAT " intersects with template %"
+ GST_PTR_FORMAT " for payloader %s", caps, static_caps,
+ gst_plugin_feature_get_name (factory));
+ payloader = gst_element_factory_create (factory, NULL);
+ }
+
+ gst_caps_unref (static_caps);
+
+ if (payloader)
+ return payloader;
+ }
+ }
+ }
+
+ return NULL;
+}
+
+static GstRTSPStream *
+gst_rtsp_client_sink_create_stream (GstRTSPClientSink * sink,
+ GstRTSPStreamContext * context, GstElement * payloader, GstPad * pad)
+{
+ GstRTSPStream *stream = NULL;
+ guint pt, aux_pt, ulpfec_pt;
+
+ GST_OBJECT_LOCK (sink);
+
+ g_object_get (G_OBJECT (payloader), "pt", &pt, NULL);
+ if (pt >= 96 && pt <= sink->next_dyn_pt) {
+ /* Payloader has a dynamic PT, but one that's already used */
+ /* FIXME: Create a caps->ptmap instead? */
+ pt = sink->next_dyn_pt;
+
+ if (pt > 127)
+ goto no_free_pt;
+
+ GST_DEBUG_OBJECT (sink, "Assigning pt %u to stream %d", pt, context->index);
+
+ sink->next_dyn_pt++;
+ } else {
+ GST_DEBUG_OBJECT (sink, "Keeping existing pt %u for stream %d",
+ pt, context->index);
+ }
+
+ aux_pt = sink->next_dyn_pt;
+ if (aux_pt > 127)
+ goto no_free_pt;
+ sink->next_dyn_pt++;
+
+ ulpfec_pt = sink->next_dyn_pt;
+ if (ulpfec_pt > 127)
+ goto no_free_pt;
+ sink->next_dyn_pt++;
+
+ GST_OBJECT_UNLOCK (sink);
+
+
+ g_object_set (G_OBJECT (payloader), "pt", pt, NULL);
+
+ stream = gst_rtsp_stream_new (context->index, payloader, pad);
+
+ gst_rtsp_stream_set_client_side (stream, TRUE);
+ gst_rtsp_stream_set_retransmission_time (stream,
+ (GstClockTime) (sink->rtx_time) * GST_MSECOND);
+ gst_rtsp_stream_set_protocols (stream, sink->protocols);
+ gst_rtsp_stream_set_profiles (stream, sink->profiles);
+ gst_rtsp_stream_set_retransmission_pt (stream, aux_pt);
+ gst_rtsp_stream_set_buffer_size (stream, sink->udp_buffer_size);
+ if (sink->rtp_blocksize > 0)
+ gst_rtsp_stream_set_mtu (stream, sink->rtp_blocksize);
+ gst_rtsp_stream_set_multicast_iface (stream, sink->multi_iface);
+
+ gst_rtsp_stream_set_ulpfec_pt (stream, ulpfec_pt);
+ gst_rtsp_stream_set_ulpfec_percentage (stream, context->ulpfec_percentage);
+
+#if 0
+ if (priv->pool)
+ gst_rtsp_stream_set_address_pool (stream, priv->pool);
+#endif
+
+ return stream;
+no_free_pt:
+ GST_OBJECT_UNLOCK (sink);
+
+ GST_ELEMENT_ERROR (sink, RESOURCE, NO_SPACE_LEFT, (NULL),
+ ("Ran out of dynamic payload types."));
+
+ return NULL;
+}
+
+static GstPadProbeReturn
+handle_payloader_block (GstPad * pad, GstPadProbeInfo * info,
+ GstRTSPStreamContext * context)
+{
+ GstRTSPClientSink *sink = context->parent;
+
+ GST_INFO_OBJECT (sink, "Block on pad %" GST_PTR_FORMAT, pad);
+
+ g_mutex_lock (&sink->preroll_lock);
+ context->prerolled = TRUE;
+ g_cond_broadcast (&sink->preroll_cond);
+ g_mutex_unlock (&sink->preroll_lock);
+
+ GST_INFO_OBJECT (sink, "Announced preroll on pad %" GST_PTR_FORMAT, pad);
+
+ return GST_PAD_PROBE_OK;
+}
+
+static gboolean
+gst_rtsp_client_sink_setup_payloader (GstRTSPClientSink * sink, GstPad * pad,
+ GstCaps * caps)
+{
+ GstRTSPStreamContext *context;
+ GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad);
+
+ GstElement *payloader;
+ GstPad *sinkpad, *srcpad, *ghostsink;
+
+ context = gst_pad_get_element_private (pad);
+
+ if (cspad->custom_payloader) {
+ payloader = cspad->custom_payloader;
+ } else {
+ /* Find the payloader. */
+ payloader = gst_rtsp_client_sink_make_payloader (caps);
+ }
+
+ if (payloader == NULL)
+ return FALSE;
+
+ GST_DEBUG_OBJECT (sink, "Configuring payloader %" GST_PTR_FORMAT
+ " for pad %" GST_PTR_FORMAT, payloader, pad);
+
+ sinkpad = gst_element_get_static_pad (payloader, "sink");
+ if (sinkpad == NULL)
+ goto no_sinkpad;
+
+ srcpad = gst_element_get_static_pad (payloader, "src");
+ if (srcpad == NULL)
+ goto no_srcpad;
+
+ gst_bin_add (GST_BIN (sink->internal_bin), payloader);
+ ghostsink = gst_ghost_pad_new (NULL, sinkpad);
+ gst_pad_set_active (ghostsink, TRUE);
+ gst_element_add_pad (GST_ELEMENT (sink->internal_bin), ghostsink);
+
+ g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_NEW_PAYLOADER], 0,
+ payloader);
+
+ GST_RTSP_STATE_LOCK (sink);
+ context->payloader_block_id =
+ gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM,
+ (GstPadProbeCallback) handle_payloader_block, context, NULL);
+ context->payloader = payloader;
+
+ payloader = gst_object_ref (payloader);
+
+ gst_ghost_pad_set_target (GST_GHOST_PAD (pad), ghostsink);
+ gst_object_unref (GST_OBJECT (sinkpad));
+ GST_RTSP_STATE_UNLOCK (sink);
+
+ context->ulpfec_percentage = cspad->ulpfec_percentage;
+
+ gst_element_sync_state_with_parent (payloader);
+
+ gst_object_unref (payloader);
+ gst_object_unref (GST_OBJECT (srcpad));
+
+ return TRUE;
+
+no_sinkpad:
+ GST_ERROR_OBJECT (sink,
+ "Could not find sink pad on payloader %" GST_PTR_FORMAT, payloader);
+ if (!cspad->custom_payloader)
+ gst_object_unref (payloader);
+ return FALSE;
+
+no_srcpad:
+ GST_ERROR_OBJECT (sink,
+ "Could not find src pad on payloader %" GST_PTR_FORMAT, payloader);
+ gst_object_unref (GST_OBJECT (sinkpad));
+ gst_object_unref (payloader);
+ return TRUE;
+}
+
+static gboolean
+gst_rtsp_client_sink_sinkpad_event (GstPad * pad, GstObject * parent,
+ GstEvent * event)
+{
+ if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
+ GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
+ if (target == NULL) {
+ GstCaps *caps;
+
+ /* No target yet - choose a payloader and configure it */
+ gst_event_parse_caps (event, &caps);
+
+ GST_DEBUG_OBJECT (parent,
+ "Have set caps event on pad %" GST_PTR_FORMAT
+ " caps %" GST_PTR_FORMAT, pad, caps);
+
+ if (!gst_rtsp_client_sink_setup_payloader (GST_RTSP_CLIENT_SINK (parent),
+ pad, caps)) {
+ GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad);
+ GST_ELEMENT_ERROR (parent, CORE, NEGOTIATION,
+ ("Could not create payloader"),
+ ("Custom payloader: %p, caps: %" GST_PTR_FORMAT,
+ cspad->custom_payloader, caps));
+ gst_event_unref (event);
+ return FALSE;
+ }
+ } else {
+ gst_object_unref (target);
+ }
+ }
+
+ return gst_pad_event_default (pad, parent, event);
+}
+
+static gboolean
+gst_rtsp_client_sink_sinkpad_query (GstPad * pad, GstObject * parent,
+ GstQuery * query)
+{
+ if (GST_QUERY_TYPE (query) == GST_QUERY_CAPS) {
+ GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
+ if (target == NULL) {
+ GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad);
+ GstCaps *caps;
+
+ if (cspad->custom_payloader) {
+ GstPad *sinkpad =
+ gst_element_get_static_pad (cspad->custom_payloader, "sink");
+
+ if (sinkpad) {
+ caps = gst_pad_query_caps (sinkpad, NULL);
+ gst_object_unref (sinkpad);
+ } else {
+ GST_ELEMENT_ERROR (parent, CORE, NEGOTIATION, (NULL),
+ ("Custom payloaders are expected to expose a sink pad named 'sink'"));
+ return FALSE;
+ }
+ } else {
+ /* No target yet - return the union of all payloader caps */
+ caps = gst_rtsp_client_sink_get_all_payloaders_caps ();
+ }
+
+ GST_TRACE_OBJECT (parent, "Returning payloader caps %" GST_PTR_FORMAT,
+ caps);
+
+ gst_query_set_caps_result (query, caps);
+ gst_caps_unref (caps);
+
+ return TRUE;
+ }
+ gst_object_unref (target);
+ }
+
+ return gst_pad_query_default (pad, parent, query);
+}
+
+static GstPad *
+gst_rtsp_client_sink_request_new_pad (GstElement * element,
+ GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
+{
+ GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
+ GstPad *pad;
+ GstRTSPStreamContext *context;
+ guint idx = (guint) - 1;
+ gchar *tmpname;
+
+ g_mutex_lock (&sink->preroll_lock);
+ if (sink->streams_collected) {
+ GST_WARNING_OBJECT (element, "Can't add streams to a running session");
+ g_mutex_unlock (&sink->preroll_lock);
+ return NULL;
+ }
+ g_mutex_unlock (&sink->preroll_lock);
+
+ GST_OBJECT_LOCK (sink);
+ if (name) {
+ if (!sscanf (name, "sink_%u", &idx)) {
+ GST_OBJECT_UNLOCK (sink);
+ GST_ERROR_OBJECT (element, "Invalid sink pad name %s", name);
+ return NULL;
+ }
+
+ if (idx >= sink->next_pad_id)
+ sink->next_pad_id = idx + 1;
+ }
+ if (idx == (guint) - 1) {
+ idx = sink->next_pad_id;
+ sink->next_pad_id++;
+ }
+ GST_OBJECT_UNLOCK (sink);
+
+ tmpname = g_strdup_printf ("sink_%u", idx);
+ pad = gst_rtsp_client_sink_pad_new (templ, tmpname);
+ g_free (tmpname);
+
+ GST_DEBUG_OBJECT (element, "Creating request pad %" GST_PTR_FORMAT, pad);
+
+ gst_pad_set_event_function (pad,
+ GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_sinkpad_event));
+ gst_pad_set_query_function (pad,
+ GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_sinkpad_query));
+
+ context = g_new0 (GstRTSPStreamContext, 1);
+ context->parent = sink;
+ context->index = idx;
+
+ gst_pad_set_element_private (pad, context);
+
+ /* The rest of the context is configured on a caps set */
+ gst_pad_set_active (pad, TRUE);
+ gst_element_add_pad (element, pad);
+ gst_child_proxy_child_added (GST_CHILD_PROXY (element), G_OBJECT (pad),
+ GST_PAD_NAME (pad));
+
+ (void) gst_rtsp_client_sink_get_factories ();
+
+ g_mutex_init (&context->conninfo.send_lock);
+ g_mutex_init (&context->conninfo.recv_lock);
+
+ GST_RTSP_STATE_LOCK (sink);
+ sink->contexts = g_list_prepend (sink->contexts, context);
+ GST_RTSP_STATE_UNLOCK (sink);
+
+ return pad;
+}
+
+static void
+gst_rtsp_client_sink_release_pad (GstElement * element, GstPad * pad)
+{
+ GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
+ GstRTSPStreamContext *context;
+
+ context = gst_pad_get_element_private (pad);
+
+ /* FIXME: we may need to change our blocking state waiting for
+ * GstRTSPStreamBlocking messages */
+
+ GST_RTSP_STATE_LOCK (sink);
+ sink->contexts = g_list_remove (sink->contexts, context);
+ GST_RTSP_STATE_UNLOCK (sink);
+
+ /* FIXME: Shut down and clean up streaming on this pad,
+ * do teardown if needed */
+ GST_LOG_OBJECT (sink,
+ "Cleaning up payloader and stream for released pad %" GST_PTR_FORMAT,
+ pad);
+
+ if (context->stream_transport) {
+ gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
+ gst_object_unref (context->stream_transport);
+ context->stream_transport = NULL;
+ }
+ if (context->stream) {
+ if (context->joined) {
+ gst_rtsp_stream_leave_bin (context->stream,
+ GST_BIN (sink->internal_bin), sink->rtpbin);
+ context->joined = FALSE;
+ }
+ gst_object_unref (context->stream);
+ context->stream = NULL;
+ }
+ if (context->srtcpparams)
+ gst_caps_unref (context->srtcpparams);
+
+ g_free (context->conninfo.location);
+ context->conninfo.location = NULL;
+
+ g_mutex_clear (&context->conninfo.send_lock);
+ g_mutex_clear (&context->conninfo.recv_lock);
+
+ g_free (context);
+
+ gst_element_remove_pad (element, pad);
+}
+
+static GstClock *
+gst_rtsp_client_sink_provide_clock (GstElement * element)
+{
+ GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
+ GstClock *clock;
+
+ if ((clock = sink->provided_clock) != NULL)
+ gst_object_ref (clock);
+
+ return clock;
+}
+
+/* a proxy string of the format [user:passwd@]host[:port] */
+static gboolean
+gst_rtsp_client_sink_set_proxy (GstRTSPClientSink * rtsp, const gchar * proxy)
+{
+ gchar *p, *at, *col;
+
+ g_free (rtsp->proxy_user);
+ rtsp->proxy_user = NULL;
+ g_free (rtsp->proxy_passwd);
+ rtsp->proxy_passwd = NULL;
+ g_free (rtsp->proxy_host);
+ rtsp->proxy_host = NULL;
+ rtsp->proxy_port = 0;
+
+ p = (gchar *) proxy;
+
+ if (p == NULL)
+ return TRUE;
+
+ /* we allow http:// in front but ignore it */
+ if (g_str_has_prefix (p, "http://"))
+ p += 7;
+
+ at = strchr (p, '@');
+ if (at) {
+ /* look for user:passwd */
+ col = strchr (proxy, ':');
+ if (col == NULL || col > at)
+ return FALSE;
+
+ rtsp->proxy_user = g_strndup (p, col - p);
+ col++;
+ rtsp->proxy_passwd = g_strndup (col, at - col);
+
+ /* move to host */
+ p = at + 1;
+ } else {
+ if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
+ rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
+ if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
+ rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
+ if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
+ GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
+ GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
+ }
+ }
+ col = strchr (p, ':');
+
+ if (col) {
+ /* everything before the colon is the hostname */
+ rtsp->proxy_host = g_strndup (p, col - p);
+ p = col + 1;
+ rtsp->proxy_port = strtoul (p, (char **) &p, 10);
+ } else {
+ rtsp->proxy_host = g_strdup (p);
+ rtsp->proxy_port = 8080;
+ }
+ return TRUE;
+}
+
+static void
+gst_rtsp_client_sink_set_tcp_timeout (GstRTSPClientSink * rtsp_client_sink,
+ guint64 timeout)
+{
+ rtsp_client_sink->tcp_timeout = timeout;
+}
+
+static void
+gst_rtsp_client_sink_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPClientSink *rtsp_client_sink;
+
+ rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
+
+ switch (prop_id) {
+ case PROP_LOCATION:
+ gst_rtsp_client_sink_uri_set_uri (GST_URI_HANDLER (rtsp_client_sink),
+ g_value_get_string (value), NULL);
+ break;
+ case PROP_PROTOCOLS:
+ rtsp_client_sink->protocols = g_value_get_flags (value);
+ break;
+ case PROP_PROFILES:
+ rtsp_client_sink->profiles = g_value_get_flags (value);
+ break;
+ case PROP_DEBUG:
+ rtsp_client_sink->debug = g_value_get_boolean (value);
+ break;
+ case PROP_RETRY:
+ rtsp_client_sink->retry = g_value_get_uint (value);
+ break;
+ case PROP_TIMEOUT:
+ rtsp_client_sink->udp_timeout = g_value_get_uint64 (value);
+ break;
+ case PROP_TCP_TIMEOUT:
+ gst_rtsp_client_sink_set_tcp_timeout (rtsp_client_sink,
+ g_value_get_uint64 (value));
+ break;
+ case PROP_LATENCY:
+ rtsp_client_sink->latency = g_value_get_uint (value);
+ break;
+ case PROP_RTX_TIME:
+ rtsp_client_sink->rtx_time = g_value_get_uint (value);
+ break;
+ case PROP_DO_RTSP_KEEP_ALIVE:
+ rtsp_client_sink->do_rtsp_keep_alive = g_value_get_boolean (value);
+ break;
+ case PROP_PROXY:
+ gst_rtsp_client_sink_set_proxy (rtsp_client_sink,
+ g_value_get_string (value));
+ break;
+ case PROP_PROXY_ID:
+ if (rtsp_client_sink->prop_proxy_id)
+ g_free (rtsp_client_sink->prop_proxy_id);
+ rtsp_client_sink->prop_proxy_id = g_value_dup_string (value);
+ break;
+ case PROP_PROXY_PW:
+ if (rtsp_client_sink->prop_proxy_pw)
+ g_free (rtsp_client_sink->prop_proxy_pw);
+ rtsp_client_sink->prop_proxy_pw = g_value_dup_string (value);
+ break;
+ case PROP_RTP_BLOCKSIZE:
+ rtsp_client_sink->rtp_blocksize = g_value_get_uint (value);
+ break;
+ case PROP_USER_ID:
+ if (rtsp_client_sink->user_id)
+ g_free (rtsp_client_sink->user_id);
+ rtsp_client_sink->user_id = g_value_dup_string (value);
+ break;
+ case PROP_USER_PW:
+ if (rtsp_client_sink->user_pw)
+ g_free (rtsp_client_sink->user_pw);
+ rtsp_client_sink->user_pw = g_value_dup_string (value);
+ break;
+ case PROP_PORT_RANGE:
+ {
+ const gchar *str;
+
+ str = g_value_get_string (value);
+ if (!str || !sscanf (str, "%u-%u",
+ &rtsp_client_sink->client_port_range.min,
+ &rtsp_client_sink->client_port_range.max)) {
+ rtsp_client_sink->client_port_range.min = 0;
+ rtsp_client_sink->client_port_range.max = 0;
+ }
+ break;
+ }
+ case PROP_UDP_BUFFER_SIZE:
+ rtsp_client_sink->udp_buffer_size = g_value_get_int (value);
+ break;
+ case PROP_UDP_RECONNECT:
+ rtsp_client_sink->udp_reconnect = g_value_get_boolean (value);
+ break;
+ case PROP_MULTICAST_IFACE:
+ g_free (rtsp_client_sink->multi_iface);
+
+ if (g_value_get_string (value) == NULL)
+ rtsp_client_sink->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
+ else
+ rtsp_client_sink->multi_iface = g_value_dup_string (value);
+ break;
+ case PROP_SDES:
+ rtsp_client_sink->sdes = g_value_dup_boxed (value);
+ break;
+ case PROP_TLS_VALIDATION_FLAGS:
+ rtsp_client_sink->tls_validation_flags = g_value_get_flags (value);
+ break;
+ case PROP_TLS_DATABASE:
+ g_clear_object (&rtsp_client_sink->tls_database);
+ rtsp_client_sink->tls_database = g_value_dup_object (value);
+ break;
+ case PROP_TLS_INTERACTION:
+ g_clear_object (&rtsp_client_sink->tls_interaction);
+ rtsp_client_sink->tls_interaction = g_value_dup_object (value);
+ break;
+ case PROP_NTP_TIME_SOURCE:
+ rtsp_client_sink->ntp_time_source = g_value_get_enum (value);
+ break;
+ case PROP_USER_AGENT:
+ g_free (rtsp_client_sink->user_agent);
+ rtsp_client_sink->user_agent = g_value_dup_string (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtsp_client_sink_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPClientSink *rtsp_client_sink;
+
+ rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
+
+ switch (prop_id) {
+ case PROP_LOCATION:
+ g_value_set_string (value, rtsp_client_sink->conninfo.location);
+ break;
+ case PROP_PROTOCOLS:
+ g_value_set_flags (value, rtsp_client_sink->protocols);
+ break;
+ case PROP_PROFILES:
+ g_value_set_flags (value, rtsp_client_sink->profiles);
+ break;
+ case PROP_DEBUG:
+ g_value_set_boolean (value, rtsp_client_sink->debug);
+ break;
+ case PROP_RETRY:
+ g_value_set_uint (value, rtsp_client_sink->retry);
+ break;
+ case PROP_TIMEOUT:
+ g_value_set_uint64 (value, rtsp_client_sink->udp_timeout);
+ break;
+ case PROP_TCP_TIMEOUT:
+ g_value_set_uint64 (value, rtsp_client_sink->tcp_timeout);
+ break;
+ case PROP_LATENCY:
+ g_value_set_uint (value, rtsp_client_sink->latency);
+ break;
+ case PROP_RTX_TIME:
+ g_value_set_uint (value, rtsp_client_sink->rtx_time);
+ break;
+ case PROP_DO_RTSP_KEEP_ALIVE:
+ g_value_set_boolean (value, rtsp_client_sink->do_rtsp_keep_alive);
+ break;
+ case PROP_PROXY:
+ {
+ gchar *str;
+
+ if (rtsp_client_sink->proxy_host) {
+ str =
+ g_strdup_printf ("%s:%d", rtsp_client_sink->proxy_host,
+ rtsp_client_sink->proxy_port);
+ } else {
+ str = NULL;
+ }
+ g_value_take_string (value, str);
+ break;
+ }
+ case PROP_PROXY_ID:
+ g_value_set_string (value, rtsp_client_sink->prop_proxy_id);
+ break;
+ case PROP_PROXY_PW:
+ g_value_set_string (value, rtsp_client_sink->prop_proxy_pw);
+ break;
+ case PROP_RTP_BLOCKSIZE:
+ g_value_set_uint (value, rtsp_client_sink->rtp_blocksize);
+ break;
+ case PROP_USER_ID:
+ g_value_set_string (value, rtsp_client_sink->user_id);
+ break;
+ case PROP_USER_PW:
+ g_value_set_string (value, rtsp_client_sink->user_pw);
+ break;
+ case PROP_PORT_RANGE:
+ {
+ gchar *str;
+
+ if (rtsp_client_sink->client_port_range.min != 0) {
+ str = g_strdup_printf ("%u-%u", rtsp_client_sink->client_port_range.min,
+ rtsp_client_sink->client_port_range.max);
+ } else {
+ str = NULL;
+ }
+ g_value_take_string (value, str);
+ break;
+ }
+ case PROP_UDP_BUFFER_SIZE:
+ g_value_set_int (value, rtsp_client_sink->udp_buffer_size);
+ break;
+ case PROP_UDP_RECONNECT:
+ g_value_set_boolean (value, rtsp_client_sink->udp_reconnect);
+ break;
+ case PROP_MULTICAST_IFACE:
+ g_value_set_string (value, rtsp_client_sink->multi_iface);
+ break;
+ case PROP_SDES:
+ g_value_set_boxed (value, rtsp_client_sink->sdes);
+ break;
+ case PROP_TLS_VALIDATION_FLAGS:
+ g_value_set_flags (value, rtsp_client_sink->tls_validation_flags);
+ break;
+ case PROP_TLS_DATABASE:
+ g_value_set_object (value, rtsp_client_sink->tls_database);
+ break;
+ case PROP_TLS_INTERACTION:
+ g_value_set_object (value, rtsp_client_sink->tls_interaction);
+ break;
+ case PROP_NTP_TIME_SOURCE:
+ g_value_set_enum (value, rtsp_client_sink->ntp_time_source);
+ break;
+ case PROP_USER_AGENT:
+ g_value_set_string (value, rtsp_client_sink->user_agent);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static const gchar *
+get_aggregate_control (GstRTSPClientSink * sink)
+{
+ const gchar *base;
+
+ if (sink->control)
+ base = sink->control;
+ else if (sink->content_base)
+ base = sink->content_base;
+ else if (sink->conninfo.url_str)
+ base = sink->conninfo.url_str;
+ else
+ base = "/";
+
+ return base;
+}
+
+static void
+gst_rtsp_client_sink_cleanup (GstRTSPClientSink * sink)
+{
+ GList *walk;
+
+ GST_DEBUG_OBJECT (sink, "cleanup");
+
+ gst_element_set_state (GST_ELEMENT (sink->internal_bin), GST_STATE_NULL);
+
+ /* Clean up any left over stream objects */
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *context = (GstRTSPStreamContext *) (walk->data);
+ if (context->stream_transport) {
+ gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
+ gst_object_unref (context->stream_transport);
+ context->stream_transport = NULL;
+ }
+
+ if (context->stream) {
+ if (context->joined) {
+ gst_rtsp_stream_leave_bin (context->stream,
+ GST_BIN (sink->internal_bin), sink->rtpbin);
+ context->joined = FALSE;
+ }
+ gst_object_unref (context->stream);
+ context->stream = NULL;
+ }
+
+ if (context->srtcpparams) {
+ gst_caps_unref (context->srtcpparams);
+ context->srtcpparams = NULL;
+ }
+ g_free (context->conninfo.location);
+ context->conninfo.location = NULL;
+ }
+
+ if (sink->rtpbin) {
+ gst_element_set_state (sink->rtpbin, GST_STATE_NULL);
+ gst_bin_remove (GST_BIN_CAST (sink->internal_bin), sink->rtpbin);
+ sink->rtpbin = NULL;
+ }
+
+ g_free (sink->content_base);
+ sink->content_base = NULL;
+
+ g_free (sink->control);
+ sink->control = NULL;
+
+ if (sink->range)
+ gst_rtsp_range_free (sink->range);
+ sink->range = NULL;
+
+ /* don't clear the SDP when it was used in the url */
+ if (sink->uri_sdp && !sink->from_sdp) {
+ gst_sdp_message_free (sink->uri_sdp);
+ sink->uri_sdp = NULL;
+ }
+
+ if (sink->provided_clock) {
+ gst_object_unref (sink->provided_clock);
+ sink->provided_clock = NULL;
+ }
+
+ g_free (sink->server_ip);
+ sink->server_ip = NULL;
+
+ sink->next_pad_id = 0;
+ sink->next_dyn_pt = 96;
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_connection_send (GstRTSPClientSink * sink,
+ GstRTSPConnInfo * conninfo, GstRTSPMessage * message, gint64 timeout)
+{
+ GstRTSPResult ret;
+
+ if (conninfo->connection) {
+ g_mutex_lock (&conninfo->send_lock);
+ ret =
+ gst_rtsp_connection_send_usec (conninfo->connection, message, timeout);
+ g_mutex_unlock (&conninfo->send_lock);
+ } else {
+ ret = GST_RTSP_ERROR;
+ }
+
+ return ret;
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_connection_send_messages (GstRTSPClientSink * sink,
+ GstRTSPConnInfo * conninfo, GstRTSPMessage * messages, guint n_messages,
+ gint64 timeout)
+{
+ GstRTSPResult ret;
+
+ if (conninfo->connection) {
+ g_mutex_lock (&conninfo->send_lock);
+ ret =
+ gst_rtsp_connection_send_messages_usec (conninfo->connection, messages,
+ n_messages, timeout);
+ g_mutex_unlock (&conninfo->send_lock);
+ } else {
+ ret = GST_RTSP_ERROR;
+ }
+
+ return ret;
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_connection_receive (GstRTSPClientSink * sink,
+ GstRTSPConnInfo * conninfo, GstRTSPMessage * message, gint64 timeout)
+{
+ GstRTSPResult ret;
+
+ if (conninfo->connection) {
+ g_mutex_lock (&conninfo->recv_lock);
+ ret = gst_rtsp_connection_receive_usec (conninfo->connection, message,
+ timeout);
+ g_mutex_unlock (&conninfo->recv_lock);
+ } else {
+ ret = GST_RTSP_ERROR;
+ }
+
+ return ret;
+}
+
+static gboolean
+accept_certificate_cb (GTlsConnection * conn, GTlsCertificate * peer_cert,
+ GTlsCertificateFlags errors, gpointer user_data)
+{
+ GstRTSPClientSink *sink = user_data;
+ gboolean accept = FALSE;
+
+ g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_ACCEPT_CERTIFICATE],
+ 0, conn, peer_cert, errors, &accept);
+
+ return accept;
+}
+
+static GstRTSPResult
+gst_rtsp_conninfo_connect (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
+ gboolean async)
+{
+ GstRTSPResult res;
+
+ if (info->connection == NULL) {
+ if (info->url == NULL) {
+ GST_DEBUG_OBJECT (sink, "parsing uri (%s)...", info->location);
+ if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
+ goto parse_error;
+ }
+
+ /* create connection */
+ GST_DEBUG_OBJECT (sink, "creating connection (%s)...", info->location);
+ if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
+ goto could_not_create;
+
+ if (info->url_str)
+ g_free (info->url_str);
+ info->url_str = gst_rtsp_url_get_request_uri (info->url);
+
+ GST_DEBUG_OBJECT (sink, "sanitized uri %s", info->url_str);
+
+ if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
+ if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
+ sink->tls_validation_flags))
+ GST_WARNING_OBJECT (sink, "Unable to set TLS validation flags");
+
+ if (sink->tls_database)
+ gst_rtsp_connection_set_tls_database (info->connection,
+ sink->tls_database);
+
+ if (sink->tls_interaction)
+ gst_rtsp_connection_set_tls_interaction (info->connection,
+ sink->tls_interaction);
+
+ gst_rtsp_connection_set_accept_certificate_func (info->connection,
+ accept_certificate_cb, sink, NULL);
+ }
+
+ if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
+ gst_rtsp_connection_set_tunneled (info->connection, TRUE);
+
+ if (sink->proxy_host) {
+ GST_DEBUG_OBJECT (sink, "setting proxy %s:%d", sink->proxy_host,
+ sink->proxy_port);
+ gst_rtsp_connection_set_proxy (info->connection, sink->proxy_host,
+ sink->proxy_port);
+ }
+ }
+
+ if (!info->connected) {
+ /* connect */
+ if (async)
+ GST_ELEMENT_PROGRESS (sink, CONTINUE, "connect",
+ ("Connecting to %s", info->location));
+ GST_DEBUG_OBJECT (sink, "connecting (%s)...", info->location);
+ if ((res =
+ gst_rtsp_connection_connect_usec (info->connection,
+ sink->tcp_timeout)) < 0)
+ goto could_not_connect;
+
+ info->connected = TRUE;
+ }
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+parse_error:
+ {
+ GST_ERROR_OBJECT (sink, "No valid RTSP URL was provided");
+ return res;
+ }
+could_not_create:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+ GST_ERROR_OBJECT (sink, "Could not create connection. (%s)", str);
+ g_free (str);
+ return res;
+ }
+could_not_connect:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+ GST_ERROR_OBJECT (sink, "Could not connect to server. (%s)", str);
+ g_free (str);
+ return res;
+ }
+}
+
+static GstRTSPResult
+gst_rtsp_conninfo_close (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
+ gboolean free)
+{
+ GST_RTSP_STATE_LOCK (sink);
+ if (info->connected) {
+ GST_DEBUG_OBJECT (sink, "closing connection...");
+ gst_rtsp_connection_close (info->connection);
+ info->connected = FALSE;
+ }
+ if (free && info->connection) {
+ /* free connection */
+ GST_DEBUG_OBJECT (sink, "freeing connection...");
+ gst_rtsp_connection_free (info->connection);
+ g_mutex_lock (&sink->preroll_lock);
+ info->connection = NULL;
+ g_cond_broadcast (&sink->preroll_cond);
+ g_mutex_unlock (&sink->preroll_lock);
+ }
+ GST_RTSP_STATE_UNLOCK (sink);
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+gst_rtsp_conninfo_reconnect (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
+ gboolean async)
+{
+ GstRTSPResult res;
+
+ GST_DEBUG_OBJECT (sink, "reconnecting connection...");
+ gst_rtsp_conninfo_close (sink, info, FALSE);
+ res = gst_rtsp_conninfo_connect (sink, info, async);
+
+ return res;
+}
+
+static void
+gst_rtsp_client_sink_connection_flush (GstRTSPClientSink * sink, gboolean flush)
+{
+ GList *walk;
+
+ GST_DEBUG_OBJECT (sink, "set flushing %d", flush);
+ g_mutex_lock (&sink->preroll_lock);
+ if (sink->conninfo.connection && sink->conninfo.flushing != flush) {
+ GST_DEBUG_OBJECT (sink, "connection flush");
+ gst_rtsp_connection_flush (sink->conninfo.connection, flush);
+ sink->conninfo.flushing = flush;
+ }
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
+ if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
+ GST_DEBUG_OBJECT (sink, "stream %p flush", stream);
+ gst_rtsp_connection_flush (stream->conninfo.connection, flush);
+ stream->conninfo.flushing = flush;
+ }
+ }
+ g_cond_broadcast (&sink->preroll_cond);
+ g_mutex_unlock (&sink->preroll_lock);
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_init_request (GstRTSPClientSink * sink,
+ GstRTSPMessage * msg, GstRTSPMethod method, const gchar * uri)
+{
+ GstRTSPResult res;
+
+ res = gst_rtsp_message_init_request (msg, method, uri);
+ if (res < 0)
+ return res;
+
+ /* set user-agent */
+ if (sink->user_agent)
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_USER_AGENT,
+ sink->user_agent);
+
+ return res;
+}
+
+/* FIXME, handle server request, reply with OK, for now */
+static GstRTSPResult
+gst_rtsp_client_sink_handle_request (GstRTSPClientSink * sink,
+ GstRTSPConnInfo * conninfo, GstRTSPMessage * request)
+{
+ GstRTSPMessage response = { 0 };
+ GstRTSPResult res;
+
+ GST_DEBUG_OBJECT (sink, "got server request message");
+
+ if (sink->debug)
+ gst_rtsp_message_dump (request);
+
+ /* default implementation, send OK */
+ GST_DEBUG_OBJECT (sink, "prepare OK reply");
+ res =
+ gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
+ request);
+ if (res < 0)
+ goto send_error;
+
+ /* let app parse and reply */
+ g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_HANDLE_REQUEST],
+ 0, request, &response);
+
+ if (sink->debug)
+ gst_rtsp_message_dump (&response);
+
+ res = gst_rtsp_client_sink_connection_send (sink, conninfo, &response, 0);
+ if (res < 0)
+ goto send_error;
+
+ gst_rtsp_message_unset (&response);
+
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+send_error:
+ {
+ gst_rtsp_message_unset (&response);
+ return res;
+ }
+}
+
+/* send server keep-alive */
+static GstRTSPResult
+gst_rtsp_client_sink_send_keep_alive (GstRTSPClientSink * sink)
+{
+ GstRTSPMessage request = { 0 };
+ GstRTSPResult res;
+ GstRTSPMethod method;
+ const gchar *control;
+
+ if (sink->do_rtsp_keep_alive == FALSE) {
+ GST_DEBUG_OBJECT (sink, "do-rtsp-keep-alive is FALSE, not sending.");
+ gst_rtsp_connection_reset_timeout (sink->conninfo.connection);
+ return GST_RTSP_OK;
+ }
+
+ GST_DEBUG_OBJECT (sink, "creating server keep-alive");
+
+ /* find a method to use for keep-alive */
+ if (sink->methods & GST_RTSP_GET_PARAMETER)
+ method = GST_RTSP_GET_PARAMETER;
+ else
+ method = GST_RTSP_OPTIONS;
+
+ control = get_aggregate_control (sink);
+ if (control == NULL)
+ goto no_control;
+
+ res = gst_rtsp_client_sink_init_request (sink, &request, method, control);
+ if (res < 0)
+ goto send_error;
+
+ if (sink->debug)
+ gst_rtsp_message_dump (&request);
+
+ res =
+ gst_rtsp_client_sink_connection_send (sink, &sink->conninfo, &request, 0);
+ if (res < 0)
+ goto send_error;
+
+ gst_rtsp_connection_reset_timeout (sink->conninfo.connection);
+ gst_rtsp_message_unset (&request);
+
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+no_control:
+ {
+ GST_WARNING_OBJECT (sink, "no control url to send keepalive");
+ return GST_RTSP_OK;
+ }
+send_error:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ gst_rtsp_message_unset (&request);
+ GST_ELEMENT_WARNING (sink, RESOURCE, WRITE, (NULL),
+ ("Could not send keep-alive. (%s)", str));
+ g_free (str);
+ return res;
+ }
+}
+
+static GstFlowReturn
+gst_rtsp_client_sink_loop_rx (GstRTSPClientSink * sink)
+{
+ GstRTSPResult res;
+ GstRTSPMessage message = { 0 };
+ gint retry = 0;
+
+ while (TRUE) {
+ gint64 timeout;
+
+ /* get the next timeout interval */
+ timeout = gst_rtsp_connection_next_timeout_usec (sink->conninfo.connection);
+
+ GST_DEBUG_OBJECT (sink, "doing receive with timeout %d seconds",
+ (gint) timeout / G_USEC_PER_SEC);
+
+ gst_rtsp_message_unset (&message);
+
+ /* we should continue reading the TCP socket because the server might
+ * send us requests. When the session timeout expires, we need to send a
+ * keep-alive request to keep the session open. */
+ res =
+ gst_rtsp_client_sink_connection_receive (sink,
+ &sink->conninfo, &message, timeout);
+
+ switch (res) {
+ case GST_RTSP_OK:
+ GST_DEBUG_OBJECT (sink, "we received a server message");
+ break;
+ case GST_RTSP_EINTR:
+ /* we got interrupted, see what we have to do */
+ goto interrupt;
+ case GST_RTSP_ETIMEOUT:
+ /* send keep-alive, ignore the result, a warning will be posted. */
+ GST_DEBUG_OBJECT (sink, "timeout, sending keep-alive");
+ if ((res =
+ gst_rtsp_client_sink_send_keep_alive (sink)) == GST_RTSP_EINTR)
+ goto interrupt;
+ continue;
+ case GST_RTSP_EEOF:
+ /* server closed the connection. not very fatal for UDP, reconnect and
+ * see what happens. */
+ GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
+ ("The server closed the connection."));
+ if (sink->udp_reconnect) {
+ if ((res =
+ gst_rtsp_conninfo_reconnect (sink, &sink->conninfo,
+ FALSE)) < 0)
+ goto connect_error;
+ } else {
+ goto server_eof;
+ }
+ continue;
+ break;
+ case GST_RTSP_ENET:
+ GST_DEBUG_OBJECT (sink, "An ethernet problem occured.");
+ default:
+ GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
+ ("Unhandled return value %d.", res));
+ goto receive_error;
+ }
+
+ switch (message.type) {
+ case GST_RTSP_MESSAGE_REQUEST:
+ /* server sends us a request message, handle it */
+ res =
+ gst_rtsp_client_sink_handle_request (sink,
+ &sink->conninfo, &message);
+ if (res == GST_RTSP_EEOF)
+ goto server_eof;
+ else if (res < 0)
+ goto handle_request_failed;
+ break;
+ case GST_RTSP_MESSAGE_RESPONSE:
+ /* we ignore response and data messages */
+ GST_DEBUG_OBJECT (sink, "ignoring response message");
+ if (sink->debug)
+ gst_rtsp_message_dump (&message);
+ if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
+ GST_DEBUG_OBJECT (sink, "but is Unauthorized response ...");
+ if (gst_rtsp_client_sink_setup_auth (sink, &message) && !(retry++)) {
+ GST_DEBUG_OBJECT (sink, "so retrying keep-alive");
+ if ((res =
+ gst_rtsp_client_sink_send_keep_alive (sink)) ==
+ GST_RTSP_EINTR)
+ goto interrupt;
+ }
+ } else {
+ retry = 0;
+ }
+ break;
+ case GST_RTSP_MESSAGE_DATA:
+ /* we ignore response and data messages */
+ GST_DEBUG_OBJECT (sink, "ignoring data message");
+ break;
+ default:
+ GST_WARNING_OBJECT (sink, "ignoring unknown message type %d",
+ message.type);
+ break;
+ }
+ }
+ g_assert_not_reached ();
+
+ /* we get here when the connection got interrupted */
+interrupt:
+ {
+ gst_rtsp_message_unset (&message);
+ GST_DEBUG_OBJECT (sink, "got interrupted");
+ return GST_FLOW_FLUSHING;
+ }
+connect_error:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+ GstFlowReturn ret;
+
+ sink->conninfo.connected = FALSE;
+ if (res != GST_RTSP_EINTR) {
+ GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ_WRITE, (NULL),
+ ("Could not connect to server. (%s)", str));
+ g_free (str);
+ ret = GST_FLOW_ERROR;
+ } else {
+ ret = GST_FLOW_FLUSHING;
+ }
+ return ret;
+ }
+receive_error:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
+ ("Could not receive message. (%s)", str));
+ g_free (str);
+ return GST_FLOW_ERROR;
+ }
+handle_request_failed:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+ GstFlowReturn ret;
+
+ gst_rtsp_message_unset (&message);
+ if (res != GST_RTSP_EINTR) {
+ GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
+ ("Could not handle server message. (%s)", str));
+ g_free (str);
+ ret = GST_FLOW_ERROR;
+ } else {
+ ret = GST_FLOW_FLUSHING;
+ }
+ return ret;
+ }
+server_eof:
+ {
+ GST_DEBUG_OBJECT (sink, "we got an eof from the server");
+ GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
+ ("The server closed the connection."));
+ sink->conninfo.connected = FALSE;
+ gst_rtsp_message_unset (&message);
+ return GST_FLOW_EOS;
+ }
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_reconnect (GstRTSPClientSink * sink, gboolean async)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ gboolean restart = FALSE;
+
+ GST_DEBUG_OBJECT (sink, "doing reconnect");
+
+ GST_FIXME_OBJECT (sink, "Reconnection is not yet implemented");
+
+ /* no need to restart, we're done */
+ if (!restart)
+ goto done;
+
+ /* we can try only TCP now */
+ sink->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
+
+ /* close and cleanup our state */
+ if ((res = gst_rtsp_client_sink_close (sink, async, FALSE)) < 0)
+ goto done;
+
+ /* see if we have TCP left to try. Also don't try TCP when we were configured
+ * with an SDP. */
+ if (!(sink->protocols & GST_RTSP_LOWER_TRANS_TCP) || sink->from_sdp)
+ goto no_protocols;
+
+ /* We post a warning message now to inform the user
+ * that nothing happened. It's most likely a firewall thing. */
+ GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
+ ("Could not receive any UDP packets for %.4f seconds, maybe your "
+ "firewall is blocking it. Retrying using a TCP connection.",
+ gst_guint64_to_gdouble (sink->udp_timeout / 1000000.0)));
+
+ /* open new connection using tcp */
+ if (gst_rtsp_client_sink_open (sink, async) < 0)
+ goto open_failed;
+
+ /* start recording */
+ if (gst_rtsp_client_sink_record (sink, async) < 0)
+ goto play_failed;
+
+done:
+ return res;
+
+ /* ERRORS */
+no_protocols:
+ {
+ sink->cur_protocols = 0;
+ /* no transport possible, post an error and stop */
+ GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
+ ("Could not receive any UDP packets for %.4f seconds, maybe your "
+ "firewall is blocking it. No other protocols to try.",
+ gst_guint64_to_gdouble (sink->udp_timeout / 1000000.0)));
+ return GST_RTSP_ERROR;
+ }
+open_failed:
+ {
+ GST_DEBUG_OBJECT (sink, "open failed");
+ return GST_RTSP_OK;
+ }
+play_failed:
+ {
+ GST_DEBUG_OBJECT (sink, "play failed");
+ return GST_RTSP_OK;
+ }
+}
+
+static void
+gst_rtsp_client_sink_loop_start_cmd (GstRTSPClientSink * sink, gint cmd)
+{
+ switch (cmd) {
+ case CMD_OPEN:
+ GST_ELEMENT_PROGRESS (sink, START, "open", ("Opening Stream"));
+ break;
+ case CMD_RECORD:
+ GST_ELEMENT_PROGRESS (sink, START, "request", ("Sending RECORD request"));
+ break;
+ case CMD_PAUSE:
+ GST_ELEMENT_PROGRESS (sink, START, "request", ("Sending PAUSE request"));
+ break;
+ case CMD_CLOSE:
+ GST_ELEMENT_PROGRESS (sink, START, "close", ("Closing Stream"));
+ break;
+ default:
+ break;
+ }
+}
+
+static void
+gst_rtsp_client_sink_loop_complete_cmd (GstRTSPClientSink * sink, gint cmd)
+{
+ switch (cmd) {
+ case CMD_OPEN:
+ GST_ELEMENT_PROGRESS (sink, COMPLETE, "open", ("Opened Stream"));
+ break;
+ case CMD_RECORD:
+ GST_ELEMENT_PROGRESS (sink, COMPLETE, "request", ("Sent RECORD request"));
+ break;
+ case CMD_PAUSE:
+ GST_ELEMENT_PROGRESS (sink, COMPLETE, "request", ("Sent PAUSE request"));
+ break;
+ case CMD_CLOSE:
+ GST_ELEMENT_PROGRESS (sink, COMPLETE, "close", ("Closed Stream"));
+ break;
+ default:
+ break;
+ }
+}
+
+static void
+gst_rtsp_client_sink_loop_cancel_cmd (GstRTSPClientSink * sink, gint cmd)
+{
+ switch (cmd) {
+ case CMD_OPEN:
+ GST_ELEMENT_PROGRESS (sink, CANCELED, "open", ("Open canceled"));
+ break;
+ case CMD_RECORD:
+ GST_ELEMENT_PROGRESS (sink, CANCELED, "request", ("RECORD canceled"));
+ break;
+ case CMD_PAUSE:
+ GST_ELEMENT_PROGRESS (sink, CANCELED, "request", ("PAUSE canceled"));
+ break;
+ case CMD_CLOSE:
+ GST_ELEMENT_PROGRESS (sink, CANCELED, "close", ("Close canceled"));
+ break;
+ default:
+ break;
+ }
+}
+
+static void
+gst_rtsp_client_sink_loop_error_cmd (GstRTSPClientSink * sink, gint cmd)
+{
+ switch (cmd) {
+ case CMD_OPEN:
+ GST_ELEMENT_PROGRESS (sink, ERROR, "open", ("Open failed"));
+ break;
+ case CMD_RECORD:
+ GST_ELEMENT_PROGRESS (sink, ERROR, "request", ("RECORD failed"));
+ break;
+ case CMD_PAUSE:
+ GST_ELEMENT_PROGRESS (sink, ERROR, "request", ("PAUSE failed"));
+ break;
+ case CMD_CLOSE:
+ GST_ELEMENT_PROGRESS (sink, ERROR, "close", ("Close failed"));
+ break;
+ default:
+ break;
+ }
+}
+
+static void
+gst_rtsp_client_sink_loop_end_cmd (GstRTSPClientSink * sink, gint cmd,
+ GstRTSPResult ret)
+{
+ if (ret == GST_RTSP_OK)
+ gst_rtsp_client_sink_loop_complete_cmd (sink, cmd);
+ else if (ret == GST_RTSP_EINTR)
+ gst_rtsp_client_sink_loop_cancel_cmd (sink, cmd);
+ else
+ gst_rtsp_client_sink_loop_error_cmd (sink, cmd);
+}
+
+static gboolean
+gst_rtsp_client_sink_loop_send_cmd (GstRTSPClientSink * sink, gint cmd,
+ gint mask)
+{
+ gint old;
+ gboolean flushed = FALSE;
+
+ /* start new request */
+ gst_rtsp_client_sink_loop_start_cmd (sink, cmd);
+
+ GST_DEBUG_OBJECT (sink, "sending cmd %s", cmd_to_string (cmd));
+
+ GST_OBJECT_LOCK (sink);
+ old = sink->pending_cmd;
+ if (old == CMD_RECONNECT) {
+ GST_DEBUG_OBJECT (sink, "ignore, we were reconnecting");
+ cmd = CMD_RECONNECT;
+ }
+ if (old != CMD_WAIT) {
+ sink->pending_cmd = CMD_WAIT;
+ GST_OBJECT_UNLOCK (sink);
+ /* cancel previous request */
+ GST_DEBUG_OBJECT (sink, "cancel previous request %s", cmd_to_string (old));
+ gst_rtsp_client_sink_loop_cancel_cmd (sink, old);
+ GST_OBJECT_LOCK (sink);
+ }
+ sink->pending_cmd = cmd;
+ /* interrupt if allowed */
+ if (sink->busy_cmd & mask) {
+ GST_DEBUG_OBJECT (sink, "connection flush busy %s",
+ cmd_to_string (sink->busy_cmd));
+ gst_rtsp_client_sink_connection_flush (sink, TRUE);
+ flushed = TRUE;
+ } else {
+ GST_DEBUG_OBJECT (sink, "not interrupting busy cmd %s",
+ cmd_to_string (sink->busy_cmd));
+ }
+ if (sink->task)
+ gst_task_start (sink->task);
+ GST_OBJECT_UNLOCK (sink);
+
+ return flushed;
+}
+
+static gboolean
+gst_rtsp_client_sink_loop (GstRTSPClientSink * sink)
+{
+ GstFlowReturn ret;
+
+ if (!sink->conninfo.connection || !sink->conninfo.connected)
+ goto no_connection;
+
+ ret = gst_rtsp_client_sink_loop_rx (sink);
+ if (ret != GST_FLOW_OK)
+ goto pause;
+
+ return TRUE;
+
+ /* ERRORS */
+no_connection:
+ {
+ GST_WARNING_OBJECT (sink, "we are not connected");
+ ret = GST_FLOW_FLUSHING;
+ goto pause;
+ }
+pause:
+ {
+ const gchar *reason = gst_flow_get_name (ret);
+
+ GST_DEBUG_OBJECT (sink, "pausing task, reason %s", reason);
+ gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_LOOP);
+ return FALSE;
+ }
+}
+
+#ifndef GST_DISABLE_GST_DEBUG
+static const gchar *
+gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
+{
+ gint index = 0;
+
+ while (method != 0) {
+ index++;
+ method >>= 1;
+ }
+ switch (index) {
+ case 0:
+ return "None";
+ case 1:
+ return "Basic";
+ case 2:
+ return "Digest";
+ }
+
+ return "Unknown";
+}
+#endif
+
+/* Parse a WWW-Authenticate Response header and determine the
+ * available authentication methods
+ *
+ * This code should also cope with the fact that each WWW-Authenticate
+ * header can contain multiple challenge methods + tokens
+ *
+ * At the moment, for Basic auth, we just do a minimal check and don't
+ * even parse out the realm */
+static void
+gst_rtsp_client_sink_parse_auth_hdr (GstRTSPMessage * response,
+ GstRTSPAuthMethod * methods, GstRTSPConnection * conn, gboolean * stale)
+{
+ GstRTSPAuthCredential **credentials, **credential;
+
+ g_return_if_fail (response != NULL);
+ g_return_if_fail (methods != NULL);
+ g_return_if_fail (stale != NULL);
+
+ credentials =
+ gst_rtsp_message_parse_auth_credentials (response,
+ GST_RTSP_HDR_WWW_AUTHENTICATE);
+ if (!credentials)
+ return;
+
+ credential = credentials;
+ while (*credential) {
+ if ((*credential)->scheme == GST_RTSP_AUTH_BASIC) {
+ *methods |= GST_RTSP_AUTH_BASIC;
+ } else if ((*credential)->scheme == GST_RTSP_AUTH_DIGEST) {
+ GstRTSPAuthParam **param = (*credential)->params;
+
+ *methods |= GST_RTSP_AUTH_DIGEST;
+
+ gst_rtsp_connection_clear_auth_params (conn);
+ *stale = FALSE;
+
+ while (*param) {
+ if (strcmp ((*param)->name, "stale") == 0
+ && g_ascii_strcasecmp ((*param)->value, "TRUE") == 0)
+ *stale = TRUE;
+ gst_rtsp_connection_set_auth_param (conn, (*param)->name,
+ (*param)->value);
+ param++;
+ }
+ }
+
+ credential++;
+ }
+
+ gst_rtsp_auth_credentials_free (credentials);
+}
+
+/**
+ * gst_rtsp_client_sink_setup_auth:
+ * @src: the rtsp source
+ *
+ * Configure a username and password and auth method on the
+ * connection object based on a response we received from the
+ * peer.
+ *
+ * Currently, this requires that a username and password were supplied
+ * in the uri. In the future, they may be requested on demand by sending
+ * a message up the bus.
+ *
+ * Returns: TRUE if authentication information could be set up correctly.
+ */
+static gboolean
+gst_rtsp_client_sink_setup_auth (GstRTSPClientSink * sink,
+ GstRTSPMessage * response)
+{
+ gchar *user = NULL;
+ gchar *pass = NULL;
+ GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
+ GstRTSPAuthMethod method;
+ GstRTSPResult auth_result;
+ GstRTSPUrl *url;
+ GstRTSPConnection *conn;
+ gboolean stale = FALSE;
+
+ conn = sink->conninfo.connection;
+
+ /* Identify the available auth methods and see if any are supported */
+ gst_rtsp_client_sink_parse_auth_hdr (response, &avail_methods, conn, &stale);
+
+ if (avail_methods == GST_RTSP_AUTH_NONE)
+ goto no_auth_available;
+
+ /* For digest auth, if the response indicates that the session
+ * data are stale, we just update them in the connection object and
+ * return TRUE to retry the request */
+ if (stale)
+ sink->tried_url_auth = FALSE;
+
+ url = gst_rtsp_connection_get_url (conn);
+
+ /* Do we have username and password available? */
+ if (url != NULL && !sink->tried_url_auth && url->user != NULL
+ && url->passwd != NULL) {
+ user = url->user;
+ pass = url->passwd;
+ sink->tried_url_auth = TRUE;
+ GST_DEBUG_OBJECT (sink,
+ "Attempting authentication using credentials from the URL");
+ } else {
+ user = sink->user_id;
+ pass = sink->user_pw;
+ GST_DEBUG_OBJECT (sink,
+ "Attempting authentication using credentials from the properties");
+ }
+
+ /* FIXME: If the url didn't contain username and password or we tried them
+ * already, request a username and passwd from the application via some kind
+ * of credentials request message */
+
+ /* If we don't have a username and passwd at this point, bail out. */
+ if (user == NULL || pass == NULL)
+ goto no_user_pass;
+
+ /* Try to configure for each available authentication method, strongest to
+ * weakest */
+ for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
+ /* Check if this method is available on the server */
+ if ((method & avail_methods) == 0)
+ continue;
+
+ /* Pass the credentials to the connection to try on the next request */
+ auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
+ /* INVAL indicates an invalid username/passwd were supplied, so we'll just
+ * ignore it and end up retrying later */
+ if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
+ GST_DEBUG_OBJECT (sink, "Attempting %s authentication",
+ gst_rtsp_auth_method_to_string (method));
+ break;
+ }
+ }
+
+ if (method == GST_RTSP_AUTH_NONE)
+ goto no_auth_available;
+
+ return TRUE;
+
+no_auth_available:
+ {
+ /* Output an error indicating that we couldn't connect because there were
+ * no supported authentication protocols */
+ GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ, (NULL),
+ ("No supported authentication protocol was found"));
+ return FALSE;
+ }
+no_user_pass:
+ {
+ /* We don't fire an error message, we just return FALSE and let the
+ * normal NOT_AUTHORIZED error be propagated */
+ return FALSE;
+ }
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_try_send (GstRTSPClientSink * sink,
+ GstRTSPConnInfo * conninfo, GstRTSPMessage * requests,
+ guint n_requests, GstRTSPMessage * response, GstRTSPStatusCode * code)
+{
+ GstRTSPResult res;
+ GstRTSPStatusCode thecode;
+ gchar *content_base = NULL;
+ gint try = 0;
+
+ g_assert (n_requests == 1 || response == NULL);
+
+again:
+ GST_DEBUG_OBJECT (sink, "sending message");
+
+ if (sink->debug && n_requests == 1)
+ gst_rtsp_message_dump (&requests[0]);
+
+ g_mutex_lock (&sink->send_lock);
+
+ res =
+ gst_rtsp_client_sink_connection_send_messages (sink, conninfo, requests,
+ n_requests, sink->tcp_timeout);
+ if (res < 0) {
+ g_mutex_unlock (&sink->send_lock);
+ goto send_error;
+ }
+
+ gst_rtsp_connection_reset_timeout (conninfo->connection);
+
+ /* See if we should handle the response */
+ if (response == NULL) {
+ g_mutex_unlock (&sink->send_lock);
+ return GST_RTSP_OK;
+ }
+next:
+ res =
+ gst_rtsp_client_sink_connection_receive (sink, conninfo, response,
+ sink->tcp_timeout);
+
+ g_mutex_unlock (&sink->send_lock);
+
+ if (res < 0)
+ goto receive_error;
+
+ if (sink->debug)
+ gst_rtsp_message_dump (response);
+
+
+ switch (response->type) {
+ case GST_RTSP_MESSAGE_REQUEST:
+ res = gst_rtsp_client_sink_handle_request (sink, conninfo, response);
+ if (res == GST_RTSP_EEOF)
+ goto server_eof;
+ else if (res < 0)
+ goto handle_request_failed;
+ g_mutex_lock (&sink->send_lock);
+ goto next;
+ case GST_RTSP_MESSAGE_RESPONSE:
+ /* ok, a response is good */
+ GST_DEBUG_OBJECT (sink, "received response message");
+ break;
+ case GST_RTSP_MESSAGE_DATA:
+ /* we ignore data messages */
+ GST_DEBUG_OBJECT (sink, "ignoring data message");
+ g_mutex_lock (&sink->send_lock);
+ goto next;
+ default:
+ GST_WARNING_OBJECT (sink, "ignoring unknown message type %d",
+ response->type);
+ g_mutex_lock (&sink->send_lock);
+ goto next;
+ }
+
+ thecode = response->type_data.response.code;
+
+ GST_DEBUG_OBJECT (sink, "got response message %d", thecode);
+
+ /* if the caller wanted the result code, we store it. */
+ if (code)
+ *code = thecode;
+
+ /* If the request didn't succeed, bail out before doing any more */
+ if (thecode != GST_RTSP_STS_OK)
+ return GST_RTSP_OK;
+
+ /* store new content base if any */
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
+ &content_base, 0);
+ if (content_base) {
+ g_free (sink->content_base);
+ sink->content_base = g_strdup (content_base);
+ }
+
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+send_error:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ if (res != GST_RTSP_EINTR) {
+ GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
+ ("Could not send message. (%s)", str));
+ } else {
+ GST_WARNING_OBJECT (sink, "send interrupted");
+ }
+ g_free (str);
+ return res;
+ }
+receive_error:
+ {
+ switch (res) {
+ case GST_RTSP_EEOF:
+ GST_WARNING_OBJECT (sink, "server closed connection");
+ if ((try == 0) && !sink->interleaved && sink->udp_reconnect) {
+ try++;
+ /* if reconnect succeeds, try again */
+ if ((res =
+ gst_rtsp_conninfo_reconnect (sink, &sink->conninfo,
+ FALSE)) == 0)
+ goto again;
+ }
+ /* only try once after reconnect, then fallthrough and error out */
+ default:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ if (res != GST_RTSP_EINTR) {
+ GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
+ ("Could not receive message. (%s)", str));
+ } else {
+ GST_WARNING_OBJECT (sink, "receive interrupted");
+ }
+ g_free (str);
+ break;
+ }
+ }
+ return res;
+ }
+handle_request_failed:
+ {
+ /* ERROR was posted */
+ gst_rtsp_message_unset (response);
+ return res;
+ }
+server_eof:
+ {
+ GST_DEBUG_OBJECT (sink, "we got an eof from the server");
+ GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
+ ("The server closed the connection."));
+ gst_rtsp_message_unset (response);
+ return res;
+ }
+}
+
+static void
+gst_rtsp_client_sink_set_state (GstRTSPClientSink * sink, GstState state)
+{
+ GST_DEBUG_OBJECT (sink, "Setting internal state to %s",
+ gst_element_state_get_name (state));
+ gst_element_set_state (GST_ELEMENT (sink->internal_bin), state);
+}
+
+/**
+ * gst_rtsp_client_sink_send:
+ * @src: the rtsp source
+ * @conn: the connection to send on
+ * @request: must point to a valid request
+ * @response: must point to an empty #GstRTSPMessage
+ * @code: an optional code result
+ *
+ * send @request and retrieve the response in @response. optionally @code can be
+ * non-NULL in which case it will contain the status code of the response.
+ *
+ * If This function returns #GST_RTSP_OK, @response will contain a valid response
+ * message that should be cleaned with gst_rtsp_message_unset() after usage.
+ *
+ * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
+ * @response message) if the response code was not 200 (OK).
+ *
+ * If the attempt results in an authentication failure, then this will attempt
+ * to retrieve authentication credentials via gst_rtsp_client_sink_setup_auth and retry
+ * the request.
+ *
+ * Returns: #GST_RTSP_OK if the processing was successful.
+ */
+static GstRTSPResult
+gst_rtsp_client_sink_send (GstRTSPClientSink * sink, GstRTSPConnInfo * conninfo,
+ GstRTSPMessage * request, GstRTSPMessage * response,
+ GstRTSPStatusCode * code)
+{
+ GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
+ GstRTSPResult res = GST_RTSP_ERROR;
+ gint count;
+ gboolean retry;
+ GstRTSPMethod method = GST_RTSP_INVALID;
+
+ count = 0;
+ do {
+ retry = FALSE;
+
+ /* make sure we don't loop forever */
+ if (count++ > 8)
+ break;
+
+ /* save method so we can disable it when the server complains */
+ method = request->type_data.request.method;
+
+ if ((res =
+ gst_rtsp_client_sink_try_send (sink, conninfo, request, 1, response,
+ &int_code)) < 0)
+ goto error;
+
+ switch (int_code) {
+ case GST_RTSP_STS_UNAUTHORIZED:
+ if (gst_rtsp_client_sink_setup_auth (sink, response)) {
+ /* Try the request/response again after configuring the auth info
+ * and loop again */
+ retry = TRUE;
+ }
+ break;
+ default:
+ break;
+ }
+ } while (retry == TRUE);
+
+ /* If the user requested the code, let them handle errors, otherwise
+ * post an error below */
+ if (code != NULL)
+ *code = int_code;
+ else if (int_code != GST_RTSP_STS_OK)
+ goto error_response;
+
+ return res;
+
+ /* ERRORS */
+error:
+ {
+ GST_DEBUG_OBJECT (sink, "got error %d", res);
+ return res;
+ }
+error_response:
+ {
+ res = GST_RTSP_ERROR;
+
+ switch (response->type_data.response.code) {
+ case GST_RTSP_STS_NOT_FOUND:
+ GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, (NULL), ("%s",
+ response->type_data.response.reason));
+ break;
+ case GST_RTSP_STS_UNAUTHORIZED:
+ GST_ELEMENT_ERROR (sink, RESOURCE, NOT_AUTHORIZED, (NULL), ("%s",
+ response->type_data.response.reason));
+ break;
+ case GST_RTSP_STS_MOVED_PERMANENTLY:
+ case GST_RTSP_STS_MOVE_TEMPORARILY:
+ {
+ gchar *new_location;
+ GstRTSPLowerTrans transports;
+
+ GST_DEBUG_OBJECT (sink, "got redirection");
+ /* if we don't have a Location Header, we must error */
+ if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
+ &new_location, 0) < 0)
+ break;
+
+ /* When we receive a redirect result, we go back to the INIT state after
+ * parsing the new URI. The caller should do the needed steps to issue
+ * a new setup when it detects this state change. */
+ GST_DEBUG_OBJECT (sink, "redirection to %s", new_location);
+
+ /* save current transports */
+ if (sink->conninfo.url)
+ transports = sink->conninfo.url->transports;
+ else
+ transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
+
+ gst_rtsp_client_sink_uri_set_uri (GST_URI_HANDLER (sink), new_location,
+ NULL);
+
+ /* set old transports */
+ if (sink->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
+ sink->conninfo.url->transports = transports;
+
+ sink->need_redirect = TRUE;
+ sink->state = GST_RTSP_STATE_INIT;
+ res = GST_RTSP_OK;
+ break;
+ }
+ case GST_RTSP_STS_NOT_ACCEPTABLE:
+ case GST_RTSP_STS_NOT_IMPLEMENTED:
+ case GST_RTSP_STS_METHOD_NOT_ALLOWED:
+ GST_WARNING_OBJECT (sink, "got NOT IMPLEMENTED, disable method %s",
+ gst_rtsp_method_as_text (method));
+ sink->methods &= ~method;
+ res = GST_RTSP_OK;
+ break;
+ default:
+ GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
+ ("Got error response: %d (%s).", response->type_data.response.code,
+ response->type_data.response.reason));
+ break;
+ }
+ /* if we return ERROR we should unset the response ourselves */
+ if (res == GST_RTSP_ERROR)
+ gst_rtsp_message_unset (response);
+
+ return res;
+ }
+}
+
+/* parse the response and collect all the supported methods. We need this
+ * information so that we don't try to send an unsupported request to the
+ * server.
+ */
+static gboolean
+gst_rtsp_client_sink_parse_methods (GstRTSPClientSink * sink,
+ GstRTSPMessage * response)
+{
+ GstRTSPHeaderField field;
+ gchar *respoptions;
+ gint indx = 0;
+
+ /* reset supported methods */
+ sink->methods = 0;
+
+ /* Try Allow Header first */
+ field = GST_RTSP_HDR_ALLOW;
+ while (TRUE) {
+ respoptions = NULL;
+ gst_rtsp_message_get_header (response, field, &respoptions, indx);
+ if (indx == 0 && !respoptions) {
+ /* if no Allow header was found then try the Public header... */
+ field = GST_RTSP_HDR_PUBLIC;
+ gst_rtsp_message_get_header (response, field, &respoptions, indx);
+ }
+ if (!respoptions)
+ break;
+
+ sink->methods |= gst_rtsp_options_from_text (respoptions);
+
+ indx++;
+ }
+
+ if (sink->methods == 0) {
+ /* neither Allow nor Public are required, assume the server supports
+ * at least SETUP. */
+ GST_DEBUG_OBJECT (sink, "could not get OPTIONS");
+ sink->methods = GST_RTSP_SETUP;
+ }
+
+ /* Even if the server replied, and didn't say it supports
+ * RECORD|ANNOUNCE, try anyway by assuming it does */
+ sink->methods |= GST_RTSP_ANNOUNCE | GST_RTSP_RECORD;
+
+ if (!(sink->methods & GST_RTSP_SETUP))
+ goto no_setup;
+
+ return TRUE;
+
+ /* ERRORS */
+no_setup:
+ {
+ GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ, (NULL),
+ ("Server does not support SETUP."));
+ return FALSE;
+ }
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_connect_to_server (GstRTSPClientSink * sink,
+ gboolean async)
+{
+ GstRTSPResult res;
+ GstRTSPMessage request = { 0 };
+ GstRTSPMessage response = { 0 };
+ GSocket *conn_socket;
+ GSocketAddress *sa;
+ GInetAddress *ia;
+
+ sink->need_redirect = FALSE;
+
+ /* can't continue without a valid url */
+ if (G_UNLIKELY (sink->conninfo.url == NULL)) {
+ res = GST_RTSP_EINVAL;
+ goto no_url;
+ }
+ sink->tried_url_auth = FALSE;
+
+ if ((res = gst_rtsp_conninfo_connect (sink, &sink->conninfo, async)) < 0)
+ goto connect_failed;
+
+ conn_socket = gst_rtsp_connection_get_read_socket (sink->conninfo.connection);
+ sa = g_socket_get_remote_address (conn_socket, NULL);
+ ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
+
+ sink->server_ip = g_inet_address_to_string (ia);
+
+ g_object_unref (sa);
+
+ /* create OPTIONS */
+ GST_DEBUG_OBJECT (sink, "create options...");
+ res =
+ gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_OPTIONS,
+ sink->conninfo.url_str);
+ if (res < 0)
+ goto create_request_failed;
+
+ /* send OPTIONS */
+ GST_DEBUG_OBJECT (sink, "send options...");
+
+ if (async)
+ GST_ELEMENT_PROGRESS (sink, CONTINUE, "open",
+ ("Retrieving server options"));
+
+ if ((res =
+ gst_rtsp_client_sink_send (sink, &sink->conninfo, &request,
+ &response, NULL)) < 0)
+ goto send_error;
+
+ /* parse OPTIONS */
+ if (!gst_rtsp_client_sink_parse_methods (sink, &response))
+ goto methods_error;
+
+ /* FIXME: Do we need to handle REDIRECT responses for OPTIONS? */
+
+ /* clean up any messages */
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_message_unset (&response);
+
+ return res;
+
+ /* ERRORS */
+no_url:
+ {
+ GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, (NULL),
+ ("No valid RTSP URL was provided"));
+ goto cleanup_error;
+ }
+connect_failed:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ if (res != GST_RTSP_EINTR) {
+ GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ_WRITE, (NULL),
+ ("Failed to connect. (%s)", str));
+ } else {
+ GST_WARNING_OBJECT (sink, "connect interrupted");
+ }
+ g_free (str);
+ goto cleanup_error;
+ }
+create_request_failed:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
+ ("Could not create request. (%s)", str));
+ g_free (str);
+ goto cleanup_error;
+ }
+send_error:
+ {
+ /* Don't post a message - the rtsp_send method will have
+ * taken care of it because we passed NULL for the response code */
+ goto cleanup_error;
+ }
+methods_error:
+ {
+ /* error was posted */
+ res = GST_RTSP_ERROR;
+ goto cleanup_error;
+ }
+cleanup_error:
+ {
+ if (sink->conninfo.connection) {
+ GST_DEBUG_OBJECT (sink, "free connection");
+ gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
+ }
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_message_unset (&response);
+ return res;
+ }
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_open (GstRTSPClientSink * sink, gboolean async)
+{
+ GstRTSPResult ret;
+
+ sink->methods =
+ GST_RTSP_SETUP | GST_RTSP_RECORD | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
+
+ g_mutex_lock (&sink->open_conn_lock);
+ sink->open_conn_start = TRUE;
+ g_cond_broadcast (&sink->open_conn_cond);
+ GST_DEBUG_OBJECT (sink, "connection to server started");
+ g_mutex_unlock (&sink->open_conn_lock);
+
+ if ((ret = gst_rtsp_client_sink_connect_to_server (sink, async)) < 0)
+ goto open_failed;
+
+ if (async)
+ gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret);
+
+ return ret;
+
+ /* ERRORS */
+open_failed:
+ {
+ GST_WARNING_OBJECT (sink, "Failed to connect to server");
+ sink->open_error = TRUE;
+ if (async)
+ gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret);
+ return ret;
+ }
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_close (GstRTSPClientSink * sink, gboolean async,
+ gboolean only_close)
+{
+ GstRTSPMessage request = { 0 };
+ GstRTSPMessage response = { 0 };
+ GstRTSPResult res = GST_RTSP_OK;
+ GList *walk;
+ const gchar *control;
+
+ GST_DEBUG_OBJECT (sink, "TEARDOWN...");
+
+ gst_rtsp_client_sink_set_state (sink, GST_STATE_NULL);
+
+ if (sink->state < GST_RTSP_STATE_READY) {
+ GST_DEBUG_OBJECT (sink, "not ready, doing cleanup");
+ goto close;
+ }
+
+ if (only_close)
+ goto close;
+
+ /* construct a control url */
+ control = get_aggregate_control (sink);
+
+ if (!(sink->methods & (GST_RTSP_RECORD | GST_RTSP_TEARDOWN)))
+ goto not_supported;
+
+ /* stop streaming */
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
+
+ if (context->stream_transport) {
+ gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
+ gst_object_unref (context->stream_transport);
+ context->stream_transport = NULL;
+ }
+
+ if (context->joined) {
+ gst_rtsp_stream_leave_bin (context->stream, GST_BIN (sink->internal_bin),
+ sink->rtpbin);
+ context->joined = FALSE;
+ }
+ }
+
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
+ const gchar *setup_url;
+ GstRTSPConnInfo *info;
+
+ GST_DEBUG_OBJECT (sink, "Looking at stream %p for teardown",
+ context->stream);
+
+ /* try aggregate control first but do non-aggregate control otherwise */
+ if (control)
+ setup_url = control;
+ else if ((setup_url = context->conninfo.location) == NULL) {
+ GST_DEBUG_OBJECT (sink, "Skipping TEARDOWN stream %p - no setup URL",
+ context->stream);
+ continue;
+ }
+
+ if (sink->conninfo.connection) {
+ info = &sink->conninfo;
+ } else if (context->conninfo.connection) {
+ info = &context->conninfo;
+ } else {
+ continue;
+ }
+ if (!info->connected)
+ goto next;
+
+ /* do TEARDOWN */
+ GST_DEBUG_OBJECT (sink, "Sending teardown for stream %p at URL %s",
+ context->stream, setup_url);
+ res =
+ gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_TEARDOWN,
+ setup_url);
+ if (res < 0)
+ goto create_request_failed;
+
+ if (async)
+ GST_ELEMENT_PROGRESS (sink, CONTINUE, "close", ("Closing stream"));
+
+ if ((res =
+ gst_rtsp_client_sink_send (sink, info, &request,
+ &response, NULL)) < 0)
+ goto send_error;
+
+ /* FIXME, parse result? */
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_message_unset (&response);
+
+ next:
+ /* early exit when we did aggregate control */
+ if (control)
+ break;
+ }
+
+close:
+ /* close connections */
+ GST_DEBUG_OBJECT (sink, "closing connection...");
+ gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
+ gst_rtsp_conninfo_close (sink, &stream->conninfo, TRUE);
+ }
+
+ /* cleanup */
+ gst_rtsp_client_sink_cleanup (sink);
+
+ sink->state = GST_RTSP_STATE_INVALID;
+
+ if (async)
+ gst_rtsp_client_sink_loop_end_cmd (sink, CMD_CLOSE, res);
+
+ return res;
+
+ /* ERRORS */
+create_request_failed:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
+ ("Could not create request. (%s)", str));
+ g_free (str);
+ goto close;
+ }
+send_error:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ gst_rtsp_message_unset (&request);
+ if (res != GST_RTSP_EINTR) {
+ GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
+ ("Could not send message. (%s)", str));
+ } else {
+ GST_WARNING_OBJECT (sink, "TEARDOWN interrupted");
+ }
+ g_free (str);
+ goto close;
+ }
+not_supported:
+ {
+ GST_DEBUG_OBJECT (sink,
+ "TEARDOWN and PLAY not supported, can't do TEARDOWN");
+ goto close;
+ }
+}
+
+static gboolean
+gst_rtsp_client_sink_configure_manager (GstRTSPClientSink * sink)
+{
+ GstElement *rtpbin;
+ GstStateChangeReturn ret;
+
+ rtpbin = sink->rtpbin;
+
+ if (rtpbin == NULL) {
+ GObjectClass *klass;
+
+ rtpbin = gst_element_factory_make ("rtpbin", NULL);
+ if (rtpbin == NULL)
+ goto no_rtpbin;
+
+ gst_bin_add (GST_BIN_CAST (sink->internal_bin), rtpbin);
+
+ sink->rtpbin = rtpbin;
+
+ /* Any more settings we should configure on rtpbin here? */
+ g_object_set (sink->rtpbin, "latency", sink->latency, NULL);
+
+ klass = G_OBJECT_GET_CLASS (G_OBJECT (rtpbin));
+
+ if (g_object_class_find_property (klass, "ntp-time-source")) {
+ g_object_set (sink->rtpbin, "ntp-time-source", sink->ntp_time_source,
+ NULL);
+ }
+
+ if (sink->sdes && g_object_class_find_property (klass, "sdes")) {
+ g_object_set (sink->rtpbin, "sdes", sink->sdes, NULL);
+ }
+
+ g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_NEW_MANAGER], 0,
+ sink->rtpbin);
+ }
+
+ ret = gst_element_set_state (rtpbin, GST_STATE_PAUSED);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto start_manager_failure;
+
+ return TRUE;
+
+no_rtpbin:
+ {
+ GST_WARNING ("no rtpbin element");
+ g_warning ("failed to create element 'rtpbin', check your installation");
+ return FALSE;
+ }
+start_manager_failure:
+ {
+ GST_DEBUG_OBJECT (sink, "could not start session manager");
+ gst_bin_remove (GST_BIN_CAST (sink->internal_bin), rtpbin);
+ return FALSE;
+ }
+}
+
+static GstElement *
+request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPClientSink * sink)
+{
+ GstRTSPStream *stream = NULL;
+ GstElement *ret = NULL;
+ GList *walk;
+
+ GST_RTSP_STATE_LOCK (sink);
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
+
+ if (sessid == gst_rtsp_stream_get_index (context->stream)) {
+ stream = context->stream;
+ break;
+ }
+ }
+
+ if (stream != NULL) {
+ GST_DEBUG_OBJECT (sink, "Creating aux sender for stream %u", sessid);
+ ret = gst_rtsp_stream_request_aux_sender (stream, sessid);
+ }
+
+ GST_RTSP_STATE_UNLOCK (sink);
+
+ return ret;
+}
+
+static GstElement *
+request_fec_encoder (GstElement * rtpbin, guint sessid,
+ GstRTSPClientSink * sink)
+{
+ GstRTSPStream *stream = NULL;
+ GstElement *ret = NULL;
+ GList *walk;
+
+ GST_RTSP_STATE_LOCK (sink);
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
+
+ if (sessid == gst_rtsp_stream_get_index (context->stream)) {
+ stream = context->stream;
+ break;
+ }
+ }
+
+ if (stream != NULL) {
+ ret = gst_rtsp_stream_request_ulpfec_encoder (stream, sessid);
+ }
+
+ GST_RTSP_STATE_UNLOCK (sink);
+
+ return ret;
+}
+
+static gboolean
+gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink)
+{
+ GstRTSPStreamContext *context;
+ GList *walk;
+ const gchar *base;
+ gchar *stream_path;
+ GstUri *base_uri, *uri;
+
+ GST_DEBUG_OBJECT (sink, "Collecting stream information");
+
+ if (!gst_rtsp_client_sink_configure_manager (sink))
+ return FALSE;
+
+ base = get_aggregate_control (sink);
+
+ base_uri = gst_uri_from_string (base);
+ if (!base_uri) {
+ GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, (NULL),
+ ("Could not parse uri %s", base));
+ return FALSE;
+ }
+
+ g_mutex_lock (&sink->preroll_lock);
+ while (sink->contexts == NULL && !sink->conninfo.flushing) {
+ g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
+ }
+ g_mutex_unlock (&sink->preroll_lock);
+
+ /* FIXME: Need different locking - need to protect against pad releases
+ * and potential state changes ruining things here */
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstPad *srcpad;
+
+ context = (GstRTSPStreamContext *) walk->data;
+ if (context->stream)
+ continue;
+
+ g_mutex_lock (&sink->preroll_lock);
+ while (!context->prerolled && !sink->conninfo.flushing) {
+ GST_DEBUG_OBJECT (sink, "Waiting for caps on stream %d", context->index);
+ g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
+ }
+ if (sink->conninfo.flushing) {
+ g_mutex_unlock (&sink->preroll_lock);
+ break;
+ }
+ g_mutex_unlock (&sink->preroll_lock);
+
+ if (context->payloader == NULL)
+ continue;
+
+ srcpad = gst_element_get_static_pad (context->payloader, "src");
+
+ GST_DEBUG_OBJECT (sink, "Creating stream object for stream %d",
+ context->index);
+ context->stream =
+ gst_rtsp_client_sink_create_stream (sink, context, context->payloader,
+ srcpad);
+
+ /* append stream index to uri path */
+ g_free (context->conninfo.location);
+
+ stream_path = g_strdup_printf ("stream=%d", context->index);
+ uri = gst_uri_copy (base_uri);
+ gst_uri_append_path (uri, stream_path);
+
+ context->conninfo.location = gst_uri_to_string (uri);
+ gst_uri_unref (uri);
+ g_free (stream_path);
+
+ if (sink->rtx_time > 0) {
+ /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
+ g_signal_connect (sink->rtpbin, "request-aux-sender",
+ (GCallback) request_aux_sender, sink);
+ }
+
+ g_signal_connect (sink->rtpbin, "request-fec-encoder",
+ (GCallback) request_fec_encoder, sink);
+
+ if (!gst_rtsp_stream_join_bin (context->stream,
+ GST_BIN (sink->internal_bin), sink->rtpbin, GST_STATE_PAUSED)) {
+ goto join_bin_failed;
+ }
+ context->joined = TRUE;
+
+ /* Block the stream, as it does not have any transport parts yet */
+ gst_rtsp_stream_set_blocked (context->stream, TRUE);
+
+ /* Let the stream object receive data */
+ gst_pad_remove_probe (srcpad, context->payloader_block_id);
+
+ gst_object_unref (srcpad);
+ }
+
+ /* Now wait for the preroll of the rtp bin */
+ g_mutex_lock (&sink->preroll_lock);
+ while (!sink->prerolled && sink->conninfo.connection
+ && !sink->conninfo.flushing) {
+ GST_LOG_OBJECT (sink, "Waiting for preroll before continuing");
+ g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
+ }
+ GST_LOG_OBJECT (sink, "Marking streams as collected");
+ sink->streams_collected = TRUE;
+ g_mutex_unlock (&sink->preroll_lock);
+
+ gst_uri_unref (base_uri);
+ return TRUE;
+
+join_bin_failed:
+
+ gst_uri_unref (base_uri);
+ GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
+ ("Could not start stream %d", context->index));
+ return FALSE;
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_create_transports_string (GstRTSPClientSink * sink,
+ GstRTSPStreamContext * context, GSocketFamily family,
+ GstRTSPLowerTrans protocols, GstRTSPProfile profiles, gchar ** transports)
+{
+ GString *result;
+ GstRTSPStream *stream = context->stream;
+ gboolean first = TRUE;
+
+ /* the default RTSP transports */
+ result = g_string_new ("RTP");
+
+ while (profiles != 0) {
+ if (!first)
+ g_string_append (result, ",RTP");
+
+ if (profiles & GST_RTSP_PROFILE_SAVPF) {
+ g_string_append (result, "/SAVPF");
+ profiles &= ~GST_RTSP_PROFILE_SAVPF;
+ } else if (profiles & GST_RTSP_PROFILE_SAVP) {
+ g_string_append (result, "/SAVP");
+ profiles &= ~GST_RTSP_PROFILE_SAVP;
+ } else if (profiles & GST_RTSP_PROFILE_AVPF) {
+ g_string_append (result, "/AVPF");
+ profiles &= ~GST_RTSP_PROFILE_AVPF;
+ } else if (profiles & GST_RTSP_PROFILE_AVP) {
+ g_string_append (result, "/AVP");
+ profiles &= ~GST_RTSP_PROFILE_AVP;
+ } else {
+ GST_WARNING_OBJECT (sink, "Unimplemented profile(s) 0x%x", profiles);
+ break;
+ }
+
+ if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
+ GstRTSPRange ports;
+
+ GST_DEBUG_OBJECT (sink, "adding UDP unicast");
+ gst_rtsp_stream_get_server_port (stream, &ports, family);
+
+ g_string_append_printf (result, "/UDP;unicast;client_port=%d-%d",
+ ports.min, ports.max);
+ } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ GstRTSPAddress *addr =
+ gst_rtsp_stream_get_multicast_address (stream, family);
+ if (addr) {
+ GST_DEBUG_OBJECT (sink, "adding UDP multicast");
+ g_string_append_printf (result, "/UDP;multicast;client_port=%d-%d",
+ addr->port, addr->port + addr->n_ports - 1);
+ gst_rtsp_address_free (addr);
+ }
+ } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
+ GST_DEBUG_OBJECT (sink, "adding TCP");
+ g_string_append_printf (result, "/TCP;unicast;interleaved=%d-%d",
+ sink->free_channel, sink->free_channel + 1);
+ }
+
+ g_string_append (result, ";mode=RECORD");
+ /* FIXME: Support appending too:
+ if (sink->append)
+ g_string_append (result, ";append");
+ */
+
+ first = FALSE;
+ }
+
+ if (first) {
+ /* No valid transport could be constructed */
+ GST_ERROR_OBJECT (sink, "No supported profiles configured");
+ goto fail;
+ }
+
+ *transports = g_string_free (result, FALSE);
+
+ GST_DEBUG_OBJECT (sink, "prepared transports %s", GST_STR_NULL (*transports));
+
+ return GST_RTSP_OK;
+fail:
+ g_string_free (result, TRUE);
+ return GST_RTSP_ERROR;
+}
+
+static GstCaps *
+signal_get_srtcp_params (GstRTSPClientSink * sink,
+ GstRTSPStreamContext * context)
+{
+ GstCaps *caps = NULL;
+
+ g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_REQUEST_RTCP_KEY], 0,
+ context->index, &caps);
+
+ if (caps != NULL)
+ GST_DEBUG_OBJECT (sink, "SRTP parameters received");
+
+ return caps;
+}
+
+static gchar *
+gst_rtsp_client_sink_stream_make_keymgmt (GstRTSPClientSink * sink,
+ GstRTSPStreamContext * context)
+{
+ gchar *base64, *result = NULL;
+ GstMIKEYMessage *mikey_msg;
+
+ context->srtcpparams = signal_get_srtcp_params (sink, context);
+ if (context->srtcpparams == NULL)
+ context->srtcpparams = gst_rtsp_stream_get_caps (context->stream);
+
+ mikey_msg = gst_mikey_message_new_from_caps (context->srtcpparams);
+ if (mikey_msg) {
+ guint send_ssrc, send_rtx_ssrc;
+ const GstStructure *s = gst_caps_get_structure (context->srtcpparams, 0);
+
+ /* add policy '0' for our SSRC */
+ gst_rtsp_stream_get_ssrc (context->stream, &send_ssrc);
+ GST_LOG_OBJECT (sink, "Stream %p ssrc %x", context->stream, send_ssrc);
+ gst_mikey_message_add_cs_srtp (mikey_msg, 0, send_ssrc, 0);
+
+ if (gst_structure_get_uint (s, "rtx-ssrc", &send_rtx_ssrc))
+ gst_mikey_message_add_cs_srtp (mikey_msg, 0, send_rtx_ssrc, 0);
+
+ base64 = gst_mikey_message_base64_encode (mikey_msg);
+ gst_mikey_message_unref (mikey_msg);
+
+ if (base64) {
+ result = gst_sdp_make_keymgmt (context->conninfo.location, base64);
+ g_free (base64);
+ }
+ }
+
+ return result;
+}
+
+/* masks to be kept in sync with the hardcoded protocol order of preference
+ * in code below */
+static const guint protocol_masks[] = {
+ GST_RTSP_LOWER_TRANS_UDP,
+ GST_RTSP_LOWER_TRANS_UDP_MCAST,
+ GST_RTSP_LOWER_TRANS_TCP,
+ 0
+};
+
+/* Same for profile_masks */
+static const guint profile_masks[] = {
+ GST_RTSP_PROFILE_SAVPF,
+ GST_RTSP_PROFILE_SAVP,
+ GST_RTSP_PROFILE_AVPF,
+ GST_RTSP_PROFILE_AVP,
+ 0
+};
+
+static gboolean
+do_send_data (GstBuffer * buffer, guint8 channel,
+ GstRTSPStreamContext * context)
+{
+ GstRTSPClientSink *sink = context->parent;
+ GstRTSPMessage message = { 0 };
+ GstRTSPResult res = GST_RTSP_OK;
+
+ gst_rtsp_message_init_data (&message, channel);
+
+ gst_rtsp_message_set_body_buffer (&message, buffer);
+
+ res =
+ gst_rtsp_client_sink_try_send (sink, &sink->conninfo, &message, 1,
+ NULL, NULL);
+
+ gst_rtsp_message_unset (&message);
+
+ gst_rtsp_stream_transport_message_sent (context->stream_transport);
+
+ return res == GST_RTSP_OK;
+}
+
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
+ GstRTSPStreamContext * context)
+{
+ GstRTSPClientSink *sink = context->parent;
+ GstRTSPResult res = GST_RTSP_OK;
+ guint i, n = gst_buffer_list_length (buffer_list);
+ GstRTSPMessage *messages = g_newa (GstRTSPMessage, n);
+
+ memset (messages, 0, n * sizeof (GstRTSPMessage));
+
+ for (i = 0; i < n; i++) {
+ GstBuffer *buffer = gst_buffer_list_get (buffer_list, i);
+
+ gst_rtsp_message_init_data (&messages[i], channel);
+
+ gst_rtsp_message_set_body_buffer (&messages[i], buffer);
+ }
+
+ res =
+ gst_rtsp_client_sink_try_send (sink, &sink->conninfo, messages, n,
+ NULL, NULL);
+
+ for (i = 0; i < n; i++) {
+ gst_rtsp_message_unset (&messages[i]);
+ gst_rtsp_stream_transport_message_sent (context->stream_transport);
+ }
+
+ return res == GST_RTSP_OK;
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_setup_streams (GstRTSPClientSink * sink, gboolean async)
+{
+ GstRTSPResult res = GST_RTSP_ERROR;
+ GstRTSPMessage request = { 0 };
+ GstRTSPMessage response = { 0 };
+ GstRTSPLowerTrans protocols;
+ GstRTSPStatusCode code;
+ GSocketFamily family;
+ GSocketAddress *sa;
+ GSocket *conn_socket;
+ GstRTSPUrl *url;
+ GList *walk;
+ gchar *hval;
+
+ if (sink->conninfo.connection) {
+ url = gst_rtsp_connection_get_url (sink->conninfo.connection);
+ /* we initially allow all configured lower transports. based on the URL
+ * transports and the replies from the server we narrow them down. */
+ protocols = url->transports & sink->cur_protocols;
+ } else {
+ url = NULL;
+ protocols = sink->cur_protocols;
+ }
+
+ if (protocols == 0)
+ goto no_protocols;
+
+ GST_RTSP_STATE_LOCK (sink);
+
+ if (G_UNLIKELY (sink->contexts == NULL))
+ goto no_streams;
+
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
+ GstRTSPStream *stream;
+
+ GstRTSPConnInfo *info;
+ GstRTSPProfile profiles;
+ GstRTSPProfile cur_profile;
+ gchar *transports;
+ gint retry = 0;
+ guint profile_mask = 0;
+ guint mask = 0;
+ GstCaps *caps;
+ const GstSDPMedia *media;
+
+ stream = context->stream;
+ profiles = gst_rtsp_stream_get_profiles (stream);
+
+ caps = gst_rtsp_stream_get_caps (stream);
+ if (caps == NULL) {
+ GST_DEBUG_OBJECT (sink, "skipping stream %p, no caps", stream);
+ continue;
+ }
+ gst_caps_unref (caps);
+ media = gst_sdp_message_get_media (&sink->cursdp, context->sdp_index);
+ if (media == NULL) {
+ GST_DEBUG_OBJECT (sink, "skipping stream %p, no SDP info", stream);
+ continue;
+ }
+
+ /* skip setup if we have no URL for it */
+ if (context->conninfo.location == NULL) {
+ GST_DEBUG_OBJECT (sink, "skipping stream %p, no setup", stream);
+ continue;
+ }
+
+ if (sink->conninfo.connection == NULL) {
+ if (!gst_rtsp_conninfo_connect (sink, &context->conninfo, async)) {
+ GST_DEBUG_OBJECT (sink, "skipping stream %p, failed to connect",
+ stream);
+ continue;
+ }
+ info = &context->conninfo;
+ } else {
+ info = &sink->conninfo;
+ }
+ GST_DEBUG_OBJECT (sink, "doing setup of stream %p with %s", stream,
+ context->conninfo.location);
+
+ conn_socket = gst_rtsp_connection_get_read_socket (info->connection);
+ sa = g_socket_get_local_address (conn_socket, NULL);
+ family = g_socket_address_get_family (sa);
+ g_object_unref (sa);
+
+ next_protocol:
+ /* first selectable profile */
+ while (profile_masks[profile_mask]
+ && !(profiles & profile_masks[profile_mask]))
+ profile_mask++;
+ if (!profile_masks[profile_mask])
+ goto no_profiles;
+
+ /* first selectable protocol */
+ while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
+ mask++;
+ if (!protocol_masks[mask])
+ goto no_protocols;
+
+ retry:
+ GST_DEBUG_OBJECT (sink, "protocols = 0x%x, protocol mask = 0x%x", protocols,
+ protocol_masks[mask]);
+ /* create a string with first transport in line */
+ transports = NULL;
+ cur_profile = profiles & profile_masks[profile_mask];
+ res = gst_rtsp_client_sink_create_transports_string (sink, context, family,
+ protocols & protocol_masks[mask], cur_profile, &transports);
+ if (res < 0 || transports == NULL)
+ goto setup_transport_failed;
+
+ if (strlen (transports) == 0) {
+ g_free (transports);
+ GST_DEBUG_OBJECT (sink, "no transports found");
+ mask++;
+ profile_mask = 0;
+ goto next_protocol;
+ }
+
+ GST_DEBUG_OBJECT (sink, "transport is %s", GST_STR_NULL (transports));
+
+ /* create SETUP request */
+ res =
+ gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_SETUP,
+ context->conninfo.location);
+ if (res < 0) {
+ g_free (transports);
+ goto create_request_failed;
+ }
+
+ /* set up keys */
+ if (cur_profile == GST_RTSP_PROFILE_SAVP ||
+ cur_profile == GST_RTSP_PROFILE_SAVPF) {
+ hval = gst_rtsp_client_sink_stream_make_keymgmt (sink, context);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval);
+ }
+
+ /* if the user wants a non default RTP packet size we add the blocksize
+ * parameter */
+ if (sink->rtp_blocksize > 0) {
+ hval = g_strdup_printf ("%d", sink->rtp_blocksize);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
+ }
+
+ if (async)
+ GST_ELEMENT_PROGRESS (sink, CONTINUE, "request", ("SETUP stream %d",
+ context->index));
+
+ {
+ GstRTSPTransport *transport;
+
+ gst_rtsp_transport_new (&transport);
+ if (gst_rtsp_transport_parse (transports, transport) != GST_RTSP_OK)
+ goto parse_transport_failed;
+ if (transport->lower_transport != GST_RTSP_LOWER_TRANS_TCP) {
+ if (!gst_rtsp_stream_allocate_udp_sockets (stream, family, transport,
+ FALSE)) {
+ gst_rtsp_transport_free (transport);
+ goto allocate_udp_ports_failed;
+ }
+ }
+ if (!gst_rtsp_stream_complete_stream (stream, transport)) {
+ gst_rtsp_transport_free (transport);
+ goto complete_stream_failed;
+ }
+
+ gst_rtsp_transport_free (transport);
+ gst_rtsp_stream_set_blocked (stream, FALSE);
+ }
+
+ /* FIXME:
+ * the creation of the transports string depends on
+ * calling stream_get_server_port, which only starts returning
+ * something meaningful after a call to stream_allocate_udp_sockets
+ * has been made, this function expects a transport that we parse
+ * from the transport string ...
+ *
+ * Significant refactoring is in order, but does not look entirely
+ * trivial, for now we put a band aid on and create a second transport
+ * string after the stream has been completed, to pass it in
+ * the request headers instead of the previous, incomplete one.
+ */
+ g_free (transports);
+ transports = NULL;
+ res = gst_rtsp_client_sink_create_transports_string (sink, context, family,
+ protocols & protocol_masks[mask], cur_profile, &transports);
+
+ if (res < 0 || transports == NULL)
+ goto setup_transport_failed;
+
+ /* select transport */
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
+
+ /* handle the code ourselves */
+ res = gst_rtsp_client_sink_send (sink, info, &request, &response, &code);
+ if (res < 0)
+ goto send_error;
+
+ switch (code) {
+ case GST_RTSP_STS_OK:
+ break;
+ case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_message_unset (&response);
+
+ /* Try another profile. If no more, move to the next protocol */
+ profile_mask++;
+ while (profile_masks[profile_mask]
+ && !(profiles & profile_masks[profile_mask]))
+ profile_mask++;
+ if (profile_masks[profile_mask])
+ goto retry;
+
+ /* select next available protocol, give up on this stream if none */
+ /* Reset profiles to try: */
+ profile_mask = 0;
+
+ mask++;
+ while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
+ mask++;
+ if (!protocol_masks[mask])
+ continue;
+ else
+ goto retry;
+ default:
+ goto response_error;
+ }
+
+ /* parse response transport */
+ {
+ gchar *resptrans = NULL;
+ GstRTSPTransport *transport;
+
+ gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
+ &resptrans, 0);
+ if (!resptrans) {
+ goto no_transport;
+ }
+
+ gst_rtsp_transport_new (&transport);
+
+ /* parse transport, go to next stream on parse error */
+ if (gst_rtsp_transport_parse (resptrans, transport) != GST_RTSP_OK) {
+ GST_WARNING_OBJECT (sink, "failed to parse transport %s", resptrans);
+ goto next;
+ }
+
+ /* update allowed transports for other streams. once the transport of
+ * one stream has been determined, we make sure that all other streams
+ * are configured in the same way */
+ switch (transport->lower_transport) {
+ case GST_RTSP_LOWER_TRANS_TCP:
+ GST_DEBUG_OBJECT (sink, "stream %p as TCP interleaved", stream);
+ protocols = GST_RTSP_LOWER_TRANS_TCP;
+ sink->interleaved = TRUE;
+ /* update free channels */
+ sink->free_channel =
+ MAX (transport->interleaved.min, sink->free_channel);
+ sink->free_channel =
+ MAX (transport->interleaved.max, sink->free_channel);
+ sink->free_channel++;
+ break;
+ case GST_RTSP_LOWER_TRANS_UDP_MCAST:
+ /* only allow multicast for other streams */
+ GST_DEBUG_OBJECT (sink, "stream %p as UDP multicast", stream);
+ protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
+ break;
+ case GST_RTSP_LOWER_TRANS_UDP:
+ /* only allow unicast for other streams */
+ GST_DEBUG_OBJECT (sink, "stream %p as UDP unicast", stream);
+ protocols = GST_RTSP_LOWER_TRANS_UDP;
+ /* Update transport with server destination if not provided by the server */
+ if (transport->destination == NULL) {
+ transport->destination = g_strdup (sink->server_ip);
+ }
+ break;
+ default:
+ GST_DEBUG_OBJECT (sink, "stream %p unknown transport %d", stream,
+ transport->lower_transport);
+ break;
+ }
+
+ if (!retry) {
+ GST_DEBUG ("Configuring the stream transport for stream %d",
+ context->index);
+ if (context->stream_transport == NULL)
+ context->stream_transport =
+ gst_rtsp_stream_transport_new (stream, transport);
+ else
+ gst_rtsp_stream_transport_set_transport (context->stream_transport,
+ transport);
+
+ if (transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+ /* our callbacks to send data on this TCP connection */
+ gst_rtsp_stream_transport_set_callbacks (context->stream_transport,
+ (GstRTSPSendFunc) do_send_data,
+ (GstRTSPSendFunc) do_send_data, context, NULL);
+ gst_rtsp_stream_transport_set_list_callbacks
+ (context->stream_transport,
+ (GstRTSPSendListFunc) do_send_data_list,
+ (GstRTSPSendListFunc) do_send_data_list, context, NULL);
+ }
+
+ /* The stream_transport now owns the transport */
+ transport = NULL;
+
+ gst_rtsp_stream_transport_set_active (context->stream_transport, TRUE);
+ }
+ next:
+ if (transport)
+ gst_rtsp_transport_free (transport);
+ /* clean up used RTSP messages */
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_message_unset (&response);
+ }
+ }
+ GST_RTSP_STATE_UNLOCK (sink);
+
+ /* store the transport protocol that was configured */
+ sink->cur_protocols = protocols;
+
+ return res;
+
+no_streams:
+ {
+ GST_RTSP_STATE_UNLOCK (sink);
+ GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+ ("SDP contains no streams"));
+ return GST_RTSP_ERROR;
+ }
+setup_transport_failed:
+ {
+ GST_RTSP_STATE_UNLOCK (sink);
+ GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+ ("Could not setup transport."));
+ res = GST_RTSP_ERROR;
+ goto cleanup_error;
+ }
+no_profiles:
+ {
+ GST_RTSP_STATE_UNLOCK (sink);
+ /* no transport possible, post an error and stop */
+ GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
+ ("Could not connect to server, no profiles left"));
+ return GST_RTSP_ERROR;
+ }
+no_protocols:
+ {
+ GST_RTSP_STATE_UNLOCK (sink);
+ /* no transport possible, post an error and stop */
+ GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
+ ("Could not connect to server, no protocols left"));
+ return GST_RTSP_ERROR;
+ }
+no_transport:
+ {
+ GST_RTSP_STATE_UNLOCK (sink);
+ GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+ ("Server did not select transport."));
+ res = GST_RTSP_ERROR;
+ goto cleanup_error;
+ }
+create_request_failed:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ GST_RTSP_STATE_UNLOCK (sink);
+ GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
+ ("Could not create request. (%s)", str));
+ g_free (str);
+ goto cleanup_error;
+ }
+parse_transport_failed:
+ {
+ GST_RTSP_STATE_UNLOCK (sink);
+ GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+ ("Could not parse transport."));
+ res = GST_RTSP_ERROR;
+ goto cleanup_error;
+ }
+allocate_udp_ports_failed:
+ {
+ GST_RTSP_STATE_UNLOCK (sink);
+ GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+ ("Could not parse transport."));
+ res = GST_RTSP_ERROR;
+ goto cleanup_error;
+ }
+complete_stream_failed:
+ {
+ GST_RTSP_STATE_UNLOCK (sink);
+ GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+ ("Could not parse transport."));
+ res = GST_RTSP_ERROR;
+ goto cleanup_error;
+ }
+send_error:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ GST_RTSP_STATE_UNLOCK (sink);
+ if (res != GST_RTSP_EINTR) {
+ GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
+ ("Could not send message. (%s)", str));
+ } else {
+ GST_WARNING_OBJECT (sink, "send interrupted");
+ }
+ g_free (str);
+ goto cleanup_error;
+ }
+response_error:
+ {
+ const gchar *str = gst_rtsp_status_as_text (code);
+
+ GST_RTSP_STATE_UNLOCK (sink);
+ GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
+ ("Error (%d): %s", code, GST_STR_NULL (str)));
+ res = GST_RTSP_ERROR;
+ goto cleanup_error;
+ }
+cleanup_error:
+ {
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_message_unset (&response);
+ return res;
+ }
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_ensure_open (GstRTSPClientSink * sink, gboolean async)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+
+ if (sink->state < GST_RTSP_STATE_READY) {
+ res = GST_RTSP_ERROR;
+ if (sink->open_error) {
+ GST_DEBUG_OBJECT (sink, "the stream was in error");
+ goto done;
+ }
+ if (async)
+ gst_rtsp_client_sink_loop_start_cmd (sink, CMD_OPEN);
+
+ if ((res = gst_rtsp_client_sink_open (sink, async)) < 0) {
+ GST_DEBUG_OBJECT (sink, "failed to open stream");
+ goto done;
+ }
+ }
+
+done:
+ return res;
+}
+
+static gboolean
+gst_rtsp_client_sink_is_stopping (GstRTSPClientSink * sink)
+{
+ gboolean is_stopping;
+
+ GST_OBJECT_LOCK (sink);
+ is_stopping = sink->task == NULL;
+ GST_OBJECT_UNLOCK (sink);
+
+ return is_stopping;
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_record (GstRTSPClientSink * sink, gboolean async)
+{
+ GstRTSPMessage request = { 0 };
+ GstRTSPMessage response = { 0 };
+ GstRTSPResult res = GST_RTSP_OK;
+ GstSDPMessage *sdp;
+ guint sdp_index = 0;
+ GstSDPInfo info = { 0, };
+ gchar *keymgmt;
+ guint i;
+
+ const gchar *proto;
+ gchar *sess_id, *client_ip, *str;
+ GSocketAddress *sa;
+ GInetAddress *ia;
+ GSocket *conn_socket;
+ GList *walk;
+
+ g_mutex_lock (&sink->preroll_lock);
+ if (sink->state == GST_RTSP_STATE_PLAYING) {
+ /* Already recording, don't send another request */
+ GST_LOG_OBJECT (sink, "Already in RECORD. Skipping duplicate request.");
+ g_mutex_unlock (&sink->preroll_lock);
+ goto done;
+ }
+ g_mutex_unlock (&sink->preroll_lock);
+
+ /* Collect all our input streams and create
+ * stream objects before actually returning.
+ * The streams are blocked at this point as we do not have any transport
+ * parts yet. */
+ gst_rtsp_client_sink_collect_streams (sink);
+
+ if (gst_rtsp_client_sink_is_stopping (sink)) {
+ GST_INFO_OBJECT (sink, "task stopped, shutting down");
+ return GST_RTSP_EINTR;
+ }
+
+ g_mutex_lock (&sink->block_streams_lock);
+ /* Wait for streams to be blocked */
+ while (sink->n_streams_blocked < g_list_length (sink->contexts)
+ && !gst_rtsp_client_sink_is_stopping (sink)) {
+ GST_DEBUG_OBJECT (sink, "waiting for streams to be blocked");
+ g_cond_wait (&sink->block_streams_cond, &sink->block_streams_lock);
+ }
+ g_mutex_unlock (&sink->block_streams_lock);
+
+ if (gst_rtsp_client_sink_is_stopping (sink)) {
+ GST_INFO_OBJECT (sink, "task stopped, shutting down");
+ return GST_RTSP_EINTR;
+ }
+
+ /* Send announce, then setup for all streams */
+ gst_sdp_message_init (&sink->cursdp);
+ sdp = &sink->cursdp;
+
+ /* some standard things first */
+ gst_sdp_message_set_version (sdp, "0");
+
+ /* session ID doesn't have to be super-unique in this case */
+ sess_id = g_strdup_printf ("%u", g_random_int ());
+
+ if (sink->conninfo.connection == NULL)
+ return GST_RTSP_ERROR;
+
+ conn_socket = gst_rtsp_connection_get_read_socket (sink->conninfo.connection);
+
+ sa = g_socket_get_local_address (conn_socket, NULL);
+ ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
+ client_ip = g_inet_address_to_string (ia);
+ if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV6) {
+ info.is_ipv6 = TRUE;
+ proto = "IP6";
+ } else if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV4)
+ proto = "IP4";
+ else
+ g_assert_not_reached ();
+ g_object_unref (sa);
+
+ /* FIXME: Should this actually be the server's IP or ours? */
+ info.server_ip = sink->server_ip;
+
+ gst_sdp_message_set_origin (sdp, "-", sess_id, "1", "IN", proto, client_ip);
+
+ gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
+ gst_sdp_message_set_information (sdp, "rtspclientsink");
+ gst_sdp_message_add_time (sdp, "0", "0", NULL);
+ gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
+
+ /* add stream */
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
+
+ gst_rtsp_sdp_from_stream (sdp, &info, context->stream);
+ context->sdp_index = sdp_index++;
+ }
+
+ g_free (sess_id);
+ g_free (client_ip);
+
+ /* send ANNOUNCE request */
+ GST_DEBUG_OBJECT (sink, "create ANNOUNCE request...");
+ res =
+ gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_ANNOUNCE,
+ sink->conninfo.url_str);
+ if (res < 0)
+ goto create_request_failed;
+
+ g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_UPDATE_SDP], 0, sdp);
+
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/sdp");
+
+ /* add SDP to the request body */
+ str = gst_sdp_message_as_text (sdp);
+ gst_rtsp_message_take_body (&request, (guint8 *) str, strlen (str));
+
+ /* send ANNOUNCE */
+ GST_DEBUG_OBJECT (sink, "sending announce...");
+
+ if (async)
+ GST_ELEMENT_PROGRESS (sink, CONTINUE, "record",
+ ("Sending server stream info"));
+
+ if ((res =
+ gst_rtsp_client_sink_send (sink, &sink->conninfo, &request,
+ &response, NULL)) < 0)
+ goto send_error;
+
+ /* parse the keymgmt */
+ i = 0;
+ walk = sink->contexts;
+ while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_KEYMGMT,
+ &keymgmt, i++) == GST_RTSP_OK) {
+ GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
+ walk = g_list_next (walk);
+ if (!gst_rtsp_stream_handle_keymgmt (context->stream, keymgmt))
+ goto keymgmt_error;
+ }
+
+ /* send setup for all streams */
+ if ((res = gst_rtsp_client_sink_setup_streams (sink, async)) < 0)
+ goto setup_failed;
+
+ res = gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_RECORD,
+ sink->conninfo.url_str);
+
+ if (res < 0)
+ goto create_request_failed;
+
+#if 0 /* FIXME: Configure a range based on input segments? */
+ if (src->need_range) {
+ hval = gen_range_header (src, segment);
+
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
+ }
+
+ if (segment->rate != 1.0) {
+ gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
+
+ g_ascii_dtostr (hval, sizeof (hval), segment->rate);
+ if (src->skip)
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
+ else
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
+ }
+#endif
+
+ if (async)
+ GST_ELEMENT_PROGRESS (sink, CONTINUE, "record", ("Starting recording"));
+ if ((res =
+ gst_rtsp_client_sink_send (sink, &sink->conninfo, &request,
+ &response, NULL)) < 0)
+ goto send_error;
+
+#if 0 /* FIXME: Check if servers return these for record: */
+ /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
+ * for the RTP packets. If this is not present, we assume all starts from 0...
+ * This is info for the RTP session manager that we pass to it in caps. */
+ hval_idx = 0;
+ while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
+ &hval, hval_idx++) == GST_RTSP_OK)
+ gst_rtspsrc_parse_rtpinfo (src, hval);
+
+ /* some servers indicate RTCP parameters in PLAY response,
+ * rather than properly in SDP */
+ if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
+ &hval, 0) == GST_RTSP_OK)
+ gst_rtspsrc_handle_rtcp_interval (src, hval);
+#endif
+
+ gst_rtsp_client_sink_set_state (sink, GST_STATE_PLAYING);
+ sink->state = GST_RTSP_STATE_PLAYING;
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
+
+ gst_rtsp_stream_unblock_rtcp (context->stream);
+ }
+
+ /* clean up any messages */
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_message_unset (&response);
+
+done:
+ return res;
+
+create_request_failed:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
+ ("Could not create request. (%s)", str));
+ g_free (str);
+ goto cleanup_error;
+ }
+send_error:
+ {
+ /* Don't post a message - the rtsp_send method will have
+ * taken care of it because we passed NULL for the response code */
+ goto cleanup_error;
+ }
+keymgmt_error:
+ {
+ GST_ELEMENT_ERROR (sink, STREAM, DECRYPT_NOKEY, (NULL),
+ ("Could not handle KeyMgmt"));
+ }
+setup_failed:
+ {
+ GST_ERROR_OBJECT (sink, "setup failed");
+ goto cleanup_error;
+ }
+cleanup_error:
+ {
+ if (sink->conninfo.connection) {
+ GST_DEBUG_OBJECT (sink, "free connection");
+ gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
+ }
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_message_unset (&response);
+ return res;
+ }
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_pause (GstRTSPClientSink * sink, gboolean async)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPMessage request = { 0 };
+ GstRTSPMessage response = { 0 };
+ GList *walk;
+ const gchar *control;
+
+ GST_DEBUG_OBJECT (sink, "PAUSE...");
+
+ if ((res = gst_rtsp_client_sink_ensure_open (sink, async)) < 0)
+ goto open_failed;
+
+ if (!(sink->methods & GST_RTSP_PAUSE))
+ goto not_supported;
+
+ if (sink->state == GST_RTSP_STATE_READY)
+ goto was_paused;
+
+ if (!sink->conninfo.connection || !sink->conninfo.connected)
+ goto no_connection;
+
+ /* construct a control url */
+ control = get_aggregate_control (sink);
+
+ /* loop over the streams. We might exit the loop early when we could do an
+ * aggregate control */
+ for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
+ GstRTSPConnInfo *info;
+ const gchar *setup_url;
+
+ /* try aggregate control first but do non-aggregate control otherwise */
+ if (control)
+ setup_url = control;
+ else if ((setup_url = stream->conninfo.location) == NULL)
+ continue;
+
+ if (sink->conninfo.connection) {
+ info = &sink->conninfo;
+ } else if (stream->conninfo.connection) {
+ info = &stream->conninfo;
+ } else {
+ continue;
+ }
+
+ if (async)
+ GST_ELEMENT_PROGRESS (sink, CONTINUE, "request",
+ ("Sending PAUSE request"));
+
+ if ((res =
+ gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_PAUSE,
+ setup_url)) < 0)
+ goto create_request_failed;
+
+ if ((res =
+ gst_rtsp_client_sink_send (sink, info, &request, &response,
+ NULL)) < 0)
+ goto send_error;
+
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_message_unset (&response);
+
+ /* exit early when we did agregate control */
+ if (control)
+ break;
+ }
+
+ /* change element states now */
+ gst_rtsp_client_sink_set_state (sink, GST_STATE_PAUSED);
+
+no_connection:
+ sink->state = GST_RTSP_STATE_READY;
+
+done:
+ if (async)
+ gst_rtsp_client_sink_loop_end_cmd (sink, CMD_PAUSE, res);
+
+ return res;
+
+ /* ERRORS */
+open_failed:
+ {
+ GST_DEBUG_OBJECT (sink, "failed to open stream");
+ goto done;
+ }
+not_supported:
+ {
+ GST_DEBUG_OBJECT (sink, "PAUSE is not supported");
+ goto done;
+ }
+was_paused:
+ {
+ GST_DEBUG_OBJECT (sink, "we were already PAUSED");
+ goto done;
+ }
+create_request_failed:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
+ ("Could not create request. (%s)", str));
+ g_free (str);
+ goto done;
+ }
+send_error:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+
+ gst_rtsp_message_unset (&request);
+ if (res != GST_RTSP_EINTR) {
+ GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
+ ("Could not send message. (%s)", str));
+ } else {
+ GST_WARNING_OBJECT (sink, "PAUSE interrupted");
+ }
+ g_free (str);
+ goto done;
+ }
+}
+
+static void
+gst_rtsp_client_sink_handle_message (GstBin * bin, GstMessage * message)
+{
+ GstRTSPClientSink *rtsp_client_sink;
+
+ rtsp_client_sink = GST_RTSP_CLIENT_SINK (bin);
+
+ switch (GST_MESSAGE_TYPE (message)) {
+ case GST_MESSAGE_ELEMENT:
+ {
+ const GstStructure *s = gst_message_get_structure (message);
+
+ if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
+ gboolean ignore_timeout;
+
+ GST_DEBUG_OBJECT (bin, "timeout on UDP port");
+
+ GST_OBJECT_LOCK (rtsp_client_sink);
+ ignore_timeout = rtsp_client_sink->ignore_timeout;
+ rtsp_client_sink->ignore_timeout = TRUE;
+ GST_OBJECT_UNLOCK (rtsp_client_sink);
+
+ /* we only act on the first udp timeout message, others are irrelevant
+ * and can be ignored. */
+ if (!ignore_timeout)
+ gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_RECONNECT,
+ CMD_LOOP);
+ /* eat and free */
+ gst_message_unref (message);
+ return;
+ } else if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
+ /* An RTSPStream has prerolled */
+ GST_DEBUG_OBJECT (rtsp_client_sink, "received GstRTSPStreamBlocking");
+ g_mutex_lock (&rtsp_client_sink->block_streams_lock);
+ rtsp_client_sink->n_streams_blocked++;
+ g_cond_broadcast (&rtsp_client_sink->block_streams_cond);
+ g_mutex_unlock (&rtsp_client_sink->block_streams_lock);
+ }
+ GST_BIN_CLASS (parent_class)->handle_message (bin, message);
+ break;
+ }
+ case GST_MESSAGE_ASYNC_START:{
+ GstObject *sender;
+
+ sender = GST_MESSAGE_SRC (message);
+
+ GST_LOG_OBJECT (rtsp_client_sink,
+ "Have async-start from %" GST_PTR_FORMAT, sender);
+ if (sender == GST_OBJECT (rtsp_client_sink->internal_bin)) {
+ GST_LOG_OBJECT (rtsp_client_sink, "child bin is now ASYNC");
+ }
+ GST_BIN_CLASS (parent_class)->handle_message (bin, message);
+ break;
+ }
+ case GST_MESSAGE_ASYNC_DONE:
+ {
+ GstObject *sender;
+ gboolean need_async_done;
+
+ sender = GST_MESSAGE_SRC (message);
+ GST_LOG_OBJECT (rtsp_client_sink, "Have async-done from %" GST_PTR_FORMAT,
+ sender);
+
+ g_mutex_lock (&rtsp_client_sink->preroll_lock);
+ if (sender == GST_OBJECT_CAST (rtsp_client_sink->internal_bin)) {
+ GST_LOG_OBJECT (rtsp_client_sink, "child bin is no longer ASYNC");
+ }
+ need_async_done = rtsp_client_sink->in_async;
+ if (rtsp_client_sink->in_async) {
+ rtsp_client_sink->in_async = FALSE;
+ g_cond_broadcast (&rtsp_client_sink->preroll_cond);
+ }
+ g_mutex_unlock (&rtsp_client_sink->preroll_lock);
+
+ GST_BIN_CLASS (parent_class)->handle_message (bin, message);
+
+ if (need_async_done) {
+ GST_DEBUG_OBJECT (rtsp_client_sink, "Posting ASYNC-DONE");
+ gst_element_post_message (GST_ELEMENT_CAST (rtsp_client_sink),
+ gst_message_new_async_done (GST_OBJECT_CAST (rtsp_client_sink),
+ GST_CLOCK_TIME_NONE));
+ }
+ break;
+ }
+ case GST_MESSAGE_ERROR:
+ {
+ GstObject *sender;
+
+ sender = GST_MESSAGE_SRC (message);
+
+ GST_DEBUG_OBJECT (rtsp_client_sink, "got error from %s",
+ GST_ELEMENT_NAME (sender));
+
+ /* FIXME: Ignore errors on RTCP? */
+ /* fatal but not our message, forward */
+ GST_BIN_CLASS (parent_class)->handle_message (bin, message);
+ break;
+ }
+ case GST_MESSAGE_STATE_CHANGED:
+ {
+ if (GST_MESSAGE_SRC (message) ==
+ (GstObject *) rtsp_client_sink->internal_bin) {
+ GstState newstate, pending;
+ gst_message_parse_state_changed (message, NULL, &newstate, &pending);
+ g_mutex_lock (&rtsp_client_sink->preroll_lock);
+ rtsp_client_sink->prerolled = (newstate >= GST_STATE_PAUSED)
+ && pending == GST_STATE_VOID_PENDING;
+ g_cond_broadcast (&rtsp_client_sink->preroll_cond);
+ g_mutex_unlock (&rtsp_client_sink->preroll_lock);
+ GST_DEBUG_OBJECT (bin,
+ "Internal bin changed state to %s (pending %s). Prerolled now %d",
+ gst_element_state_get_name (newstate),
+ gst_element_state_get_name (pending), rtsp_client_sink->prerolled);
+ }
+ /* fallthrough */
+ }
+ default:
+ {
+ GST_BIN_CLASS (parent_class)->handle_message (bin, message);
+ break;
+ }
+ }
+}
+
+/* the thread where everything happens */
+static void
+gst_rtsp_client_sink_thread (GstRTSPClientSink * sink)
+{
+ gint cmd;
+
+ GST_OBJECT_LOCK (sink);
+ cmd = sink->pending_cmd;
+ if (cmd == CMD_RECONNECT || cmd == CMD_RECORD || cmd == CMD_PAUSE
+ || cmd == CMD_LOOP || cmd == CMD_OPEN)
+ sink->pending_cmd = CMD_LOOP;
+ else
+ sink->pending_cmd = CMD_WAIT;
+ GST_DEBUG_OBJECT (sink, "got command %s", cmd_to_string (cmd));
+
+ /* we got the message command, so ensure communication is possible again */
+ gst_rtsp_client_sink_connection_flush (sink, FALSE);
+
+ sink->busy_cmd = cmd;
+ GST_OBJECT_UNLOCK (sink);
+
+ switch (cmd) {
+ case CMD_OPEN:
+ if (gst_rtsp_client_sink_open (sink, TRUE) == GST_RTSP_ERROR)
+ gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT,
+ CMD_ALL & ~CMD_CLOSE);
+ break;
+ case CMD_RECORD:
+ gst_rtsp_client_sink_record (sink, TRUE);
+ break;
+ case CMD_PAUSE:
+ gst_rtsp_client_sink_pause (sink, TRUE);
+ break;
+ case CMD_CLOSE:
+ gst_rtsp_client_sink_close (sink, TRUE, FALSE);
+ break;
+ case CMD_LOOP:
+ gst_rtsp_client_sink_loop (sink);
+ break;
+ case CMD_RECONNECT:
+ gst_rtsp_client_sink_reconnect (sink, FALSE);
+ break;
+ default:
+ break;
+ }
+
+ GST_OBJECT_LOCK (sink);
+ /* and go back to sleep */
+ if (sink->pending_cmd == CMD_WAIT) {
+ if (sink->task)
+ gst_task_pause (sink->task);
+ }
+ /* reset waiting */
+ sink->busy_cmd = CMD_WAIT;
+ GST_OBJECT_UNLOCK (sink);
+}
+
+static gboolean
+gst_rtsp_client_sink_start (GstRTSPClientSink * sink)
+{
+ GST_DEBUG_OBJECT (sink, "starting");
+
+ sink->streams_collected = FALSE;
+ gst_element_set_locked_state (GST_ELEMENT (sink->internal_bin), TRUE);
+
+ gst_rtsp_client_sink_set_state (sink, GST_STATE_READY);
+
+ GST_OBJECT_LOCK (sink);
+ sink->pending_cmd = CMD_WAIT;
+
+ if (sink->task == NULL) {
+ sink->task =
+ gst_task_new ((GstTaskFunction) gst_rtsp_client_sink_thread, sink,
+ NULL);
+ if (sink->task == NULL)
+ goto task_error;
+
+ gst_task_set_lock (sink->task, GST_RTSP_STREAM_GET_LOCK (sink));
+ }
+ GST_OBJECT_UNLOCK (sink);
+
+ return TRUE;
+
+ /* ERRORS */
+task_error:
+ {
+ GST_OBJECT_UNLOCK (sink);
+ GST_ERROR_OBJECT (sink, "failed to create task");
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_rtsp_client_sink_stop (GstRTSPClientSink * sink)
+{
+ GstTask *task;
+
+ GST_DEBUG_OBJECT (sink, "stopping");
+
+ /* also cancels pending task */
+ gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_ALL & ~CMD_CLOSE);
+
+ GST_OBJECT_LOCK (sink);
+ if ((task = sink->task)) {
+ sink->task = NULL;
+ GST_OBJECT_UNLOCK (sink);
+
+ gst_task_stop (task);
+
+ g_mutex_lock (&sink->block_streams_lock);
+ g_cond_broadcast (&sink->block_streams_cond);
+ g_mutex_unlock (&sink->block_streams_lock);
+
+ /* make sure it is not running */
+ GST_RTSP_STREAM_LOCK (sink);
+ GST_RTSP_STREAM_UNLOCK (sink);
+
+ /* now wait for the task to finish */
+ gst_task_join (task);
+
+ /* and free the task */
+ gst_object_unref (GST_OBJECT (task));
+
+ GST_OBJECT_LOCK (sink);
+ }
+ GST_OBJECT_UNLOCK (sink);
+
+ /* ensure synchronously all is closed and clean */
+ gst_rtsp_client_sink_close (sink, FALSE, TRUE);
+
+ return TRUE;
+}
+
+static GstStateChangeReturn
+gst_rtsp_client_sink_change_state (GstElement * element,
+ GstStateChange transition)
+{
+ GstRTSPClientSink *rtsp_client_sink;
+ GstStateChangeReturn ret;
+
+ rtsp_client_sink = GST_RTSP_CLIENT_SINK (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ if (!gst_rtsp_client_sink_start (rtsp_client_sink))
+ goto start_failed;
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ /* init some state */
+ rtsp_client_sink->cur_protocols = rtsp_client_sink->protocols;
+ /* first attempt, don't ignore timeouts */
+ rtsp_client_sink->ignore_timeout = FALSE;
+ rtsp_client_sink->open_error = FALSE;
+
+ gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_PAUSED);
+
+ g_mutex_lock (&rtsp_client_sink->preroll_lock);
+ if (rtsp_client_sink->in_async) {
+ GST_DEBUG_OBJECT (rtsp_client_sink, "Posting ASYNC-START");
+ gst_element_post_message (GST_ELEMENT_CAST (rtsp_client_sink),
+ gst_message_new_async_start (GST_OBJECT_CAST (rtsp_client_sink)));
+ }
+ g_mutex_unlock (&rtsp_client_sink->preroll_lock);
+
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ /* fall-through */
+ case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+ /* unblock the tcp tasks and make the loop waiting */
+ if (gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_WAIT,
+ CMD_LOOP)) {
+ /* make sure it is waiting before we send PLAY below */
+ GST_RTSP_STREAM_LOCK (rtsp_client_sink);
+ GST_RTSP_STREAM_UNLOCK (rtsp_client_sink);
+ }
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_READY);
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto done;
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ ret = GST_STATE_CHANGE_SUCCESS;
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ /* Return ASYNC and preroll input streams */
+ g_mutex_lock (&rtsp_client_sink->preroll_lock);
+ if (rtsp_client_sink->in_async)
+ ret = GST_STATE_CHANGE_ASYNC;
+ g_mutex_unlock (&rtsp_client_sink->preroll_lock);
+ gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_OPEN, 0);
+
+ /* CMD_OPEN has been scheduled. Wait until the sink thread starts
+ * opening connection to the server */
+ g_mutex_lock (&rtsp_client_sink->open_conn_lock);
+ while (!rtsp_client_sink->open_conn_start) {
+ GST_DEBUG_OBJECT (rtsp_client_sink,
+ "wait for connection to be started");
+ g_cond_wait (&rtsp_client_sink->open_conn_cond,
+ &rtsp_client_sink->open_conn_lock);
+ }
+ rtsp_client_sink->open_conn_start = FALSE;
+ g_mutex_unlock (&rtsp_client_sink->open_conn_lock);
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
+ GST_DEBUG_OBJECT (rtsp_client_sink,
+ "Switching to playing -sending RECORD");
+ gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_RECORD, 0);
+ ret = GST_STATE_CHANGE_SUCCESS;
+ break;
+ }
+ case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+ /* send pause request and keep the idle task around */
+ gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_PAUSE,
+ CMD_LOOP);
+ ret = GST_STATE_CHANGE_NO_PREROLL;
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_CLOSE,
+ CMD_PAUSE);
+ ret = GST_STATE_CHANGE_SUCCESS;
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ gst_rtsp_client_sink_stop (rtsp_client_sink);
+ ret = GST_STATE_CHANGE_SUCCESS;
+ break;
+ default:
+ break;
+ }
+
+done:
+ return ret;
+
+start_failed:
+ {
+ GST_DEBUG_OBJECT (rtsp_client_sink, "start failed");
+ return GST_STATE_CHANGE_FAILURE;
+ }
+}
+
+/*** GSTURIHANDLER INTERFACE *************************************************/
+
+static GstURIType
+gst_rtsp_client_sink_uri_get_type (GType type)
+{
+ return GST_URI_SINK;
+}
+
+static const gchar *const *
+gst_rtsp_client_sink_uri_get_protocols (GType type)
+{
+ static const gchar *protocols[] =
+ { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
+ "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
+ };
+
+ return protocols;
+}
+
+static gchar *
+gst_rtsp_client_sink_uri_get_uri (GstURIHandler * handler)
+{
+ GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (handler);
+
+ /* FIXME: make thread-safe */
+ return g_strdup (sink->conninfo.location);
+}
+
+static gboolean
+gst_rtsp_client_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri,
+ GError ** error)
+{
+ GstRTSPClientSink *sink;
+ GstRTSPResult res;
+ GstSDPResult sres;
+ GstRTSPUrl *newurl = NULL;
+ GstSDPMessage *sdp = NULL;
+
+ sink = GST_RTSP_CLIENT_SINK (handler);
+
+ /* same URI, we're fine */
+ if (sink->conninfo.location && uri && !strcmp (uri, sink->conninfo.location))
+ goto was_ok;
+
+ if (g_str_has_prefix (uri, "rtsp-sdp://")) {
+ sres = gst_sdp_message_new (&sdp);
+ if (sres < 0)
+ goto sdp_failed;
+
+ GST_DEBUG_OBJECT (sink, "parsing SDP message");
+ sres = gst_sdp_message_parse_uri (uri, sdp);
+ if (sres < 0)
+ goto invalid_sdp;
+ } else {
+ /* try to parse */
+ GST_DEBUG_OBJECT (sink, "parsing URI");
+ if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
+ goto parse_error;
+ }
+
+ /* if worked, free previous and store new url object along with the original
+ * location. */
+ GST_DEBUG_OBJECT (sink, "configuring URI");
+ g_free (sink->conninfo.location);
+ sink->conninfo.location = g_strdup (uri);
+ gst_rtsp_url_free (sink->conninfo.url);
+ sink->conninfo.url = newurl;
+ g_free (sink->conninfo.url_str);
+ if (newurl)
+ sink->conninfo.url_str = gst_rtsp_url_get_request_uri (sink->conninfo.url);
+ else
+ sink->conninfo.url_str = NULL;
+
+ if (sink->uri_sdp)
+ gst_sdp_message_free (sink->uri_sdp);
+ sink->uri_sdp = sdp;
+ sink->from_sdp = sdp != NULL;
+
+ GST_DEBUG_OBJECT (sink, "set uri: %s", GST_STR_NULL (uri));
+ GST_DEBUG_OBJECT (sink, "request uri is: %s",
+ GST_STR_NULL (sink->conninfo.url_str));
+
+ return TRUE;
+
+ /* Special cases */
+was_ok:
+ {
+ GST_DEBUG_OBJECT (sink, "URI was ok: '%s'", GST_STR_NULL (uri));
+ return TRUE;
+ }
+sdp_failed:
+ {
+ GST_ERROR_OBJECT (sink, "Could not create new SDP (%d)", sres);
+ g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
+ "Could not create SDP");
+ return FALSE;
+ }
+invalid_sdp:
+ {
+ GST_ERROR_OBJECT (sink, "Not a valid SDP (%d) '%s'", sres,
+ GST_STR_NULL (uri));
+ gst_sdp_message_free (sdp);
+ g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
+ "Invalid SDP");
+ return FALSE;
+ }
+parse_error:
+ {
+ GST_ERROR_OBJECT (sink, "Not a valid RTSP url '%s' (%d)",
+ GST_STR_NULL (uri), res);
+ g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
+ "Invalid RTSP URI");
+ return FALSE;
+ }
+}
+
+static void
+gst_rtsp_client_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
+{
+ GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
+
+ iface->get_type = gst_rtsp_client_sink_uri_get_type;
+ iface->get_protocols = gst_rtsp_client_sink_uri_get_protocols;
+ iface->get_uri = gst_rtsp_client_sink_uri_get_uri;
+ iface->set_uri = gst_rtsp_client_sink_uri_set_uri;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * <2006> Wim Taymans <wim@fluendo.com>
+ * <2015> Jan Schmidt <jan at centricular dot com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_CLIENT_SINK_H__
+#define __GST_RTSP_CLIENT_SINK_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#include <gst/rtsp-server/rtsp-stream.h>
+#include <gst/rtsp/rtsp.h>
+#include <gio/gio.h>
+
+#define GST_TYPE_RTSP_CLIENT_SINK \
+ (gst_rtsp_client_sink_get_type())
+#define GST_RTSP_CLIENT_SINK(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSP_CLIENT_SINK,GstRTSPClientSink))
+#define GST_RTSP_CLIENT_SINK_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSP_CLIENT_SINK,GstRTSPClientSinkClass))
+#define GST_IS_RTSP_CLIENT_SINK(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSP_CLIENT_SINK))
+#define GST_IS_RTSP_CLIENT_SINK_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSP_CLIENT_SINK))
+#define GST_RTSP_CLIENT_SINK_CAST(obj) \
+ ((GstRTSPClientSink *)(obj))
+
+typedef struct _GstRTSPClientSink GstRTSPClientSink;
+typedef struct _GstRTSPClientSinkClass GstRTSPClientSinkClass;
+
+#define GST_RTSP_STATE_GET_LOCK(rtsp) (&GST_RTSP_CLIENT_SINK_CAST(rtsp)->state_rec_lock)
+#define GST_RTSP_STATE_LOCK(rtsp) (g_rec_mutex_lock (GST_RTSP_STATE_GET_LOCK(rtsp)))
+#define GST_RTSP_STATE_UNLOCK(rtsp) (g_rec_mutex_unlock (GST_RTSP_STATE_GET_LOCK(rtsp)))
+
+#define GST_RTSP_STREAM_GET_LOCK(rtsp) (&GST_RTSP_CLIENT_SINK_CAST(rtsp)->stream_rec_lock)
+#define GST_RTSP_STREAM_LOCK(rtsp) (g_rec_mutex_lock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
+#define GST_RTSP_STREAM_UNLOCK(rtsp) (g_rec_mutex_unlock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
+
+typedef struct _GstRTSPConnInfo GstRTSPConnInfo;
+
+struct _GstRTSPConnInfo {
+ gchar *location;
+ GstRTSPUrl *url;
+ gchar *url_str;
+ GstRTSPConnection *connection;
+ gboolean connected;
+ gboolean flushing;
+
+ GMutex send_lock;
+ GMutex recv_lock;
+};
+
+typedef struct _GstRTSPStreamInfo GstRTSPStreamInfo;
+typedef struct _GstRTSPStreamContext GstRTSPStreamContext;
+
+struct _GstRTSPStreamContext {
+ GstRTSPClientSink *parent;
+
+ guint index;
+ /* Index of the SDPMedia in the stored SDP */
+ guint sdp_index;
+
+ GstElement *payloader;
+ guint payloader_block_id;
+ gboolean prerolled;
+
+ /* Stream management object */
+ GstRTSPStream *stream;
+ gboolean joined;
+
+ /* Secure profile key mgmt */
+ GstCaps *srtcpparams;
+
+ /* per stream connection */
+ GstRTSPConnInfo conninfo;
+ /* For interleaved mode */
+ guint8 channel[2];
+
+ GstRTSPStreamTransport *stream_transport;
+
+ guint ulpfec_percentage;
+};
+
+/**
+ * GstRTSPNatMethod:
+ * @GST_RTSP_NAT_NONE: none
+ * @GST_RTSP_NAT_DUMMY: send dummy packets
+ *
+ * Different methods for trying to traverse firewalls.
+ */
+typedef enum
+{
+ GST_RTSP_NAT_NONE,
+ GST_RTSP_NAT_DUMMY
+} GstRTSPNatMethod;
+
+struct _GstRTSPClientSink {
+ GstBin parent;
+
+ /* task and mutex for interleaved mode */
+ gboolean interleaved;
+ GstTask *task;
+ GRecMutex stream_rec_lock;
+ GstSegment segment;
+ gint free_channel;
+
+ /* UDP mode loop */
+ gint pending_cmd;
+ gint busy_cmd;
+ gboolean ignore_timeout;
+ gboolean open_error;
+
+ /* mutex for protecting state changes */
+ GRecMutex state_rec_lock;
+
+ GstSDPMessage *uri_sdp;
+ gboolean from_sdp;
+
+ /* properties */
+ GstRTSPLowerTrans protocols;
+ gboolean debug;
+ guint retry;
+ guint64 udp_timeout;
+ gint64 tcp_timeout;
+ guint latency;
+ gboolean do_rtsp_keep_alive;
+ gchar *proxy_host;
+ guint proxy_port;
+ gchar *proxy_user; /* from url or property */
+ gchar *proxy_passwd; /* from url or property */
+ gchar *prop_proxy_id; /* set via property */
+ gchar *prop_proxy_pw; /* set via property */
+ guint rtp_blocksize;
+ gchar *user_id;
+ gchar *user_pw;
+ GstRTSPRange client_port_range;
+ gint udp_buffer_size;
+ gboolean udp_reconnect;
+ gchar *multi_iface;
+ gboolean ntp_sync;
+ gboolean use_pipeline_clock;
+ GstStructure *sdes;
+ GTlsCertificateFlags tls_validation_flags;
+ GTlsDatabase *tls_database;
+ GTlsInteraction *tls_interaction;
+ gint ntp_time_source;
+ gchar *user_agent;
+
+ /* state */
+ GstRTSPState state;
+ gchar *content_base;
+ GstRTSPLowerTrans cur_protocols;
+ gboolean tried_url_auth;
+ gchar *addr;
+ gboolean need_redirect;
+ GstRTSPTimeRange *range;
+ gchar *control;
+ guint next_port_num;
+ GstClock *provided_clock;
+
+ /* supported methods */
+ gint methods;
+
+ /* session management */
+ GstRTSPConnInfo conninfo;
+
+ /* Everything goes in an internal
+ * locked-state bin */
+ GstBin *internal_bin;
+ /* Set to true when internal bin state
+ * >= PAUSED */
+ gboolean prerolled;
+
+ /* TRUE if we posted async-start */
+ gboolean in_async;
+
+ /* TRUE when stream info has been collected */
+ gboolean streams_collected;
+
+ /* TRUE when streams have been blocked */
+ guint n_streams_blocked;
+ GMutex block_streams_lock;
+ GCond block_streams_cond;
+
+ guint next_pad_id;
+ gint next_dyn_pt;
+
+ GstElement *rtpbin;
+
+ GList *contexts;
+ GstSDPMessage cursdp;
+
+ GMutex send_lock;
+
+ GMutex preroll_lock;
+ GCond preroll_cond;
+
+ /* TRUE if connection to server has been scheduled */
+ gboolean open_conn_start;
+ GMutex open_conn_lock;
+ GCond open_conn_cond;
+
+ GstClockTime rtx_time;
+
+ GstRTSPProfile profiles;
+ gchar *server_ip;
+};
+
+struct _GstRTSPClientSinkClass {
+ GstBinClass parent_class;
+};
+
+GType gst_rtsp_client_sink_get_type(void);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_CLIENT_SINK_H__ */
--- /dev/null
+rtspsink_sources = [
+ 'gstrtspclientsink.c',
+ 'plugin.c',
+]
+
+rtspsink = library('gstrtspclientsink',
+ rtspsink_sources,
+ c_args : rtspserver_args,
+ include_directories : rtspserver_incs,
+ dependencies : [gstrtsp_dep, gstsdp_dep, gst_rtsp_server_dep],
+ install : true,
+ install_dir : plugins_install_dir)
+pkgconfig.generate(rtspsink, install_dir : plugins_pkgconfig_install_dir)
+plugins += [rtspsink]
--- /dev/null
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstrtspclientsink.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+#ifdef ENABLE_NLS
+ bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
+ bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
+#endif /* ENABLE_NLS */
+
+ if (!gst_element_register (plugin, "rtspclientsink", GST_RANK_NONE,
+ GST_TYPE_RTSP_CLIENT_SINK))
+ return FALSE;
+
+ return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ rtspclientsink,
+ "RTSP client sink element",
+ plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
--- /dev/null
+#!/bin/sh
+#
+# Check that the code follows a consistent code style
+#
+
+# Check for existence of indent, and error out if not present.
+# On some *bsd systems the binary seems to be called gnunindent,
+# so check for that first.
+
+version=`gnuindent --version 2>/dev/null`
+if test "x$version" = "x"; then
+ version=`gindent --version 2>/dev/null`
+ if test "x$version" = "x"; then
+ version=`indent --version 2>/dev/null`
+ if test "x$version" = "x"; then
+ echo "GStreamer git pre-commit hook:"
+ echo "Did not find GNU indent, please install it before continuing."
+ exit 1
+ else
+ INDENT=indent
+ fi
+ else
+ INDENT=gindent
+ fi
+else
+ INDENT=gnuindent
+fi
+
+case `$INDENT --version` in
+ GNU*)
+ ;;
+ default)
+ echo "GStreamer git pre-commit hook:"
+ echo "Did not find GNU indent, please install it before continuing."
+ echo "(Found $INDENT, but it doesn't seem to be GNU indent)"
+ exit 1
+ ;;
+esac
+
+INDENT_PARAMETERS="--braces-on-if-line \
+ --case-brace-indentation0 \
+ --case-indentation2 \
+ --braces-after-struct-decl-line \
+ --line-length80 \
+ --no-tabs \
+ --cuddle-else \
+ --dont-line-up-parentheses \
+ --continuation-indentation4 \
+ --honour-newlines \
+ --tab-size8 \
+ --indent-level2 \
+ --leave-preprocessor-space"
+
+echo "--Checking style--"
+for file in `git diff-index --cached --name-only HEAD --diff-filter=ACMR| grep "\.c$"` ; do
+ # nf is the temporary checkout. This makes sure we check against the
+ # revision in the index (and not the checked out version).
+ nf=`git checkout-index --temp ${file} | cut -f 1`
+ newfile=`mktemp /tmp/${nf}.XXXXXX` || exit 1
+ $INDENT ${INDENT_PARAMETERS} \
+ $nf -o $newfile 2>> /dev/null
+ # FIXME: Call indent twice as it tends to do line-breaks
+ # different for every second call.
+ $INDENT ${INDENT_PARAMETERS} \
+ $newfile 2>> /dev/null
+ diff -u -p "${nf}" "${newfile}"
+ r=$?
+ rm "${newfile}"
+ rm "${nf}"
+ if [ $r != 0 ] ; then
+echo "================================================================================================="
+echo " Code style error in: $file "
+echo " "
+echo " Please fix before committing. Don't forget to run git add before trying to commit again. "
+echo " If the whole file is to be committed, this should work (run from the top-level directory): "
+echo " "
+echo " gst-indent $file; git add $file; git commit"
+echo " "
+echo "================================================================================================="
+ exit 1
+ fi
+done
+echo "--Checking style pass--"
--- /dev/null
+project('gst-rtsp-server', 'c',
+ version : '1.19.2',
+ meson_version : '>= 0.54',
+ default_options : ['warning_level=1', 'buildtype=debugoptimized'])
+
+gst_version = meson.project_version()
+version_arr = gst_version.split('.')
+gst_version_major = version_arr[0].to_int()
+gst_version_minor = version_arr[1].to_int()
+gst_version_micro = version_arr[2].to_int()
+ if version_arr.length() == 4
+ gst_version_nano = version_arr[3].to_int()
+else
+ gst_version_nano = 0
+endif
+gst_version_is_dev = gst_version_minor % 2 == 1 and gst_version_micro < 90
+
+glib_req = '>= 2.56.0'
+gst_req = '>= @0@.@1@.0'.format(gst_version_major, gst_version_minor)
+
+api_version = '1.0'
+soversion = 0
+# maintaining compatibility with the previous libtool versioning
+# current = minor * 100 + micro
+curversion = gst_version_minor * 100 + gst_version_micro
+libversion = '@0@.@1@.0'.format(soversion, curversion)
+osxversion = curversion + 1
+
+plugins_install_dir = '@0@/gstreamer-1.0'.format(get_option('libdir'))
+
+cc = meson.get_compiler('c')
+
+cdata = configuration_data()
+
+if cc.has_link_argument('-Wl,-Bsymbolic-functions')
+ add_project_link_arguments('-Wl,-Bsymbolic-functions', language : 'c')
+endif
+
+# Symbol visibility
+if cc.get_id() == 'msvc'
+ export_define = '__declspec(dllexport) extern'
+elif cc.has_argument('-fvisibility=hidden')
+ add_project_arguments('-fvisibility=hidden', language: 'c')
+ export_define = 'extern __attribute__ ((visibility ("default")))'
+else
+ export_define = 'extern'
+endif
+
+# Passing this through the command line would be too messy
+cdata.set('GST_API_EXPORT', export_define)
+
+# Disable strict aliasing
+if cc.has_argument('-fno-strict-aliasing')
+ add_project_arguments('-fno-strict-aliasing', language: 'c')
+endif
+
+# Define G_DISABLE_DEPRECATED for development versions
+if gst_version_is_dev
+ message('Disabling deprecated GLib API')
+ add_project_arguments('-DG_DISABLE_DEPRECATED', language: 'c')
+endif
+
+cast_checks = get_option('gobject-cast-checks')
+if cast_checks.disabled() or (cast_checks.auto() and not gst_version_is_dev)
+ message('Disabling GLib cast checks')
+ add_project_arguments('-DG_DISABLE_CAST_CHECKS', language: 'c')
+endif
+
+glib_asserts = get_option('glib-asserts')
+if glib_asserts.disabled() or (glib_asserts.auto() and not gst_version_is_dev)
+ message('Disabling GLib asserts')
+ add_project_arguments('-DG_DISABLE_ASSERT', language: 'c')
+endif
+
+glib_checks = get_option('glib-checks')
+if glib_checks.disabled() or (glib_checks.auto() and not gst_version_is_dev)
+ message('Disabling GLib checks')
+ add_project_arguments('-DG_DISABLE_CHECKS', language: 'c')
+endif
+
+cdata.set_quoted('GETTEXT_PACKAGE', 'gst-rtsp-server-1.0')
+cdata.set_quoted('PACKAGE', 'gst-rtsp-server')
+cdata.set_quoted('VERSION', gst_version)
+cdata.set_quoted('PACKAGE_VERSION', gst_version)
+cdata.set_quoted('GST_API_VERSION', api_version)
+cdata.set_quoted('GST_LICENSE', 'LGPL')
+
+# FIXME: ENABLE_NLS (currently also missing from autotools build)
+# cdata.set('ENABLE_NLS', true)
+# cdata.set_quoted('LOCALEDIR', join_paths(get_option('prefix'), get_option('localedir')))
+
+# GStreamer package name and origin url
+gst_package_name = get_option('package-name')
+if gst_package_name == ''
+ if gst_version_nano == 0
+ gst_package_name = 'GStreamer RTSP Server Library source release'
+ elif gst_version_nano == 1
+ gst_package_name = 'GStreamer RTSP Server Library git'
+ else
+ gst_package_name = 'GStreamer RTSP Server Library prerelease'
+ endif
+endif
+cdata.set_quoted('GST_PACKAGE_NAME', gst_package_name)
+cdata.set_quoted('GST_PACKAGE_ORIGIN', get_option('package-origin'))
+
+rtspserver_args = ['-DHAVE_CONFIG_H']
+
+warning_flags = [
+ '-Wmissing-declarations',
+ '-Wmissing-prototypes',
+ '-Wredundant-decls',
+ '-Wundef',
+ '-Wwrite-strings',
+ '-Wformat',
+ '-Wformat-nonliteral',
+ '-Wformat-security',
+ '-Wold-style-definition',
+ '-Waggregate-return',
+ '-Winit-self',
+ '-Wmissing-include-dirs',
+ '-Waddress',
+ '-Wno-multichar',
+ '-Wdeclaration-after-statement',
+ '-Wvla',
+ '-Wpointer-arith',
+]
+
+foreach extra_arg : warning_flags
+ if cc.has_argument (extra_arg)
+ add_project_arguments([extra_arg], language: 'c')
+ endif
+endforeach
+
+rtspserver_incs = include_directories('gst/rtsp-server', '.')
+
+glib_dep = dependency('glib-2.0', version : glib_req,
+ fallback: ['glib', 'libglib_dep'])
+gst_dep = dependency('gstreamer-1.0', version : gst_req,
+ fallback : ['gstreamer', 'gst_dep'])
+gstrtsp_dep = dependency('gstreamer-rtsp-1.0', version : gst_req,
+ fallback : ['gst-plugins-base', 'rtsp_dep'])
+gstrtp_dep = dependency('gstreamer-rtp-1.0', version : gst_req,
+ fallback : ['gst-plugins-base', 'rtp_dep'])
+gstsdp_dep = dependency('gstreamer-sdp-1.0', version : gst_req,
+ fallback : ['gst-plugins-base', 'sdp_dep'])
+gstapp_dep = dependency('gstreamer-app-1.0', version : gst_req,
+ fallback : ['gst-plugins-base', 'app_dep'])
+gstnet_dep = dependency('gstreamer-net-1.0', version : gst_req,
+ fallback : ['gstreamer', 'gst_net_dep'])
+if host_machine.system() != 'windows'
+ gstcheck_dep = dependency('gstreamer-check-1.0', version : gst_req,
+ required : get_option('tests'),
+ fallback : ['gstreamer', 'gst_check_dep'])
+endif
+
+# Disable compiler warnings for unused variables and args if gst debug system is disabled
+if gst_dep.type_name() == 'internal'
+ gst_debug_disabled = not subproject('gstreamer').get_variable('gst_debug')
+else
+ # We can't check that in the case of subprojects as we won't
+ # be able to build against an internal dependency (which is not built yet)
+ gst_debug_disabled = cc.has_header_symbol('gst/gstconfig.h', 'GST_DISABLE_GST_DEBUG', dependencies: gst_dep)
+endif
+
+if gst_debug_disabled
+ message('GStreamer debug system is disabled')
+ add_project_arguments(cc.get_supported_arguments(['-Wno-unused']), language: 'c')
+else
+ message('GStreamer debug system is enabled')
+endif
+
+gir = find_program('g-ir-scanner', required : get_option('introspection'))
+gnome = import('gnome')
+build_gir = gir.found() and (not meson.is_cross_build() or get_option('introspection').enabled())
+gir_init_section = [ '--add-init-section=extern void gst_init(gint*,gchar**);' + \
+ 'g_setenv("GST_REGISTRY_1.0", "@0@", TRUE);'.format(meson.current_build_dir() + '/gir_empty_registry.reg') + \
+ 'g_setenv("GST_PLUGIN_PATH_1_0", "", TRUE);' + \
+ 'g_setenv("GST_PLUGIN_SYSTEM_PATH_1_0", "", TRUE);' + \
+ 'gst_init(NULL,NULL);', '--quiet']
+
+pkgconfig = import('pkgconfig')
+plugins_pkgconfig_install_dir = join_paths(plugins_install_dir, 'pkgconfig')
+if get_option('default_library') == 'shared'
+ # If we don't build static plugins there is no need to generate pc files
+ plugins_pkgconfig_install_dir = disabler()
+endif
+
+plugins = []
+pkgconfig_subdirs = ['gstreamer-1.0']
+
+subdir('gst')
+if not get_option('tests').disabled()
+ subdir('tests')
+endif
+if not get_option('examples').disabled()
+ subdir('examples')
+endif
+subdir('docs')
+
+# Set release date
+if gst_version_nano == 0
+ extract_release_date = find_program('scripts/extract-release-date-from-doap-file.py')
+ run_result = run_command(extract_release_date, gst_version, files('gst-rtsp-server.doap'))
+ if run_result.returncode() == 0
+ release_date = run_result.stdout().strip()
+ cdata.set_quoted('GST_PACKAGE_RELEASE_DATETIME', release_date)
+ message('Package release date: ' + release_date)
+ else
+ # Error out if our release can't be found in the .doap file
+ error(run_result.stderr())
+ endif
+endif
+
+configure_file(output: 'config.h', configuration: cdata)
+
+python3 = import('python').find_installation()
+run_command(python3, '-c', 'import shutil; shutil.copy("hooks/pre-commit.hook", ".git/hooks/pre-commit")')
--- /dev/null
+# Feature options for plugins with no external deps
+option('rtspclientsink', type : 'feature', value : 'auto')
+
+# Common feature options
+option('examples', type : 'feature', value : 'auto', yield : true,
+ description : 'Build the examples')
+option('tests', type : 'feature', value : 'auto', yield : true,
+ description : 'Build and enable unit tests')
+option('introspection', type : 'feature', value : 'auto', yield : true,
+ description : 'Generate gobject-introspection bindings')
+option('gobject-cast-checks', type : 'feature', value : 'auto', yield : true,
+ description: 'Enable run-time GObject cast checks (auto = enabled for development, disabled for stable releases)')
+option('glib-asserts', type : 'feature', value : 'enabled', yield : true,
+ description: 'Enable GLib assertion (auto = enabled for development, disabled for stable releases)')
+option('glib-checks', type : 'feature', value : 'enabled', yield : true,
+ description: 'Enable GLib checks such as API guards (auto = enabled for development, disabled for stable releases)')
+
+# Common options
+option('package-name', type : 'string', yield : true,
+ description : 'package name to use in plugins')
+option('package-origin', type : 'string',
+ value : 'Unknown package origin', yield : true,
+ description : 'package origin URL to use in plugins')
+option('doc', type : 'feature', value : 'auto', yield: true,
+ description: 'Enable documentation.')
--- /dev/null
+#!/usr/bin/env python3
+#
+# extract-release-date-from-doap-file.py VERSION DOAP-FILE
+#
+# Extract release date for the given release version from a DOAP file
+#
+# Copyright (C) 2020 Tim-Philipp Müller <tim centricular com>
+#
+# This library is free software; you can redistribute it and/or
+# modify it under the terms of the GNU Library General Public
+# License as published by the Free Software Foundation; either
+# version 2 of the License, or (at your option) any later version.
+#
+# This library is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+# Library General Public License for more details.
+#
+# You should have received a copy of the GNU Library General Public
+# License along with this library; if not, write to the
+# Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+# Boston, MA 02110-1301, USA.
+
+import sys
+import xml.etree.ElementTree as ET
+
+if len(sys.argv) != 3:
+ sys.exit('Usage: {} VERSION DOAP-FILE'.format(sys.argv[0]))
+
+release_version = sys.argv[1]
+doap_fn = sys.argv[2]
+
+tree = ET.parse(doap_fn)
+root = tree.getroot()
+
+namespaces = {'doap': 'http://usefulinc.com/ns/doap#'}
+
+for v in root.findall('doap:release/doap:Version', namespaces=namespaces):
+ if v.findtext('doap:revision', namespaces=namespaces) == release_version:
+ release_date = v.findtext('doap:created', namespaces=namespaces)
+ if release_date:
+ print(release_date)
+ sys.exit(0)
+
+sys.exit('Could not find a release with version {} in {}'.format(release_version, doap_fn))
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-address-pool.h>
+
+GST_START_TEST (test_pool)
+{
+ GstRTSPAddressPool *pool;
+ GstRTSPAddress *addr, *addr2, *addr3;
+ GstRTSPAddressPoolResult res;
+
+ pool = gst_rtsp_address_pool_new ();
+
+ fail_if (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "233.252.0.0", 5000, 5010, 1));
+ fail_if (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "::1", 5000, 5010, 1));
+ fail_if (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "ff02::1", 5000, 5010, 1));
+ fail_if (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1.1", "233.252.0.1", 5000, 5010, 1));
+ fail_if (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "233.252.0.1.1", 5000, 5010, 1));
+ ASSERT_CRITICAL (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.0", "233.252.0.1", 5010, 5000, 1));
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.0", "233.252.0.255", 5000, 5010, 1));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.255.0.0", "233.255.0.0", 5000, 5010, 1));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.255.0.0", "233.255.0.0", 5020, 5020, 1));
+
+ /* should fail, we can't allocate a block of 256 ports */
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_MULTICAST, 256);
+ fail_unless (addr == NULL);
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr != NULL);
+
+ addr2 = gst_rtsp_address_copy (addr);
+
+ gst_rtsp_address_free (addr2);
+ gst_rtsp_address_free (addr);
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_MULTICAST, 4);
+ fail_unless (addr != NULL);
+
+ /* Will fail because pool is NULL */
+ ASSERT_CRITICAL (gst_rtsp_address_pool_clear (NULL));
+
+ /* will fail because an address is allocated */
+ ASSERT_CRITICAL (gst_rtsp_address_pool_clear (pool));
+
+ gst_rtsp_address_free (addr);
+
+ gst_rtsp_address_pool_clear (pool);
+
+ /* start with odd port to make sure we are allocated address
+ * starting with even port
+ */
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "FF11:DB8::1", "FF11:DB8::1", 5001, 5003, 1));
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_EVEN_PORT |
+ GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5002);
+ fail_unless (!g_ascii_strcasecmp (addr->address, "FF11:DB8::1"));
+
+ /* Will fail becuse there is only one IPv6 address left */
+ addr2 = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr2 == NULL);
+
+ /* Will fail because the only IPv6 address left has an odd port */
+ addr2 = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_EVEN_PORT |
+ GST_RTSP_ADDRESS_FLAG_MULTICAST, 1);
+ fail_unless (addr2 == NULL);
+
+ addr2 = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_IPV4 | GST_RTSP_ADDRESS_FLAG_MULTICAST, 1);
+ fail_unless (addr2 == NULL);
+
+ gst_rtsp_address_free (addr);
+
+ gst_rtsp_address_pool_clear (pool);
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.0", "233.252.0.255", 5000, 5002, 1));
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5000);
+ fail_unless (!strcmp (addr->address, "233.252.0.0"));
+
+ addr2 = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr2 != NULL);
+ fail_unless (addr2->port == 5000);
+ fail_unless (!strcmp (addr2->address, "233.252.0.1"));
+
+ gst_rtsp_address_free (addr);
+ gst_rtsp_address_free (addr2);
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_MULTICAST, 1);
+ fail_unless (addr == NULL);
+
+ gst_rtsp_address_pool_clear (pool);
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.1.1", "233.252.1.1", 5000, 5001, 1));
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 3,
+ 1, &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.2", 5000, 2,
+ 1, &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 500, 2, 1,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+ 2, &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "2000::1", 5000, 2, 2,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_EINVAL);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "ff02::1", 5000, 2, 2,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "1.1", 5000, 2, 2, &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_EINVAL);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+ 1, &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_OK);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5000);
+ fail_unless (!strcmp (addr->address, "233.252.1.1"));
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+ 1, &addr2);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERESERVED);
+ fail_unless (addr2 == NULL);
+
+ gst_rtsp_address_free (addr);
+ gst_rtsp_address_pool_clear (pool);
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.1.1", "233.252.1.3", 5000, 5001, 1));
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+ 1, &addr);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5000);
+ fail_unless (!strcmp (addr->address, "233.252.1.1"));
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.3", 5000, 2,
+ 1, &addr2);
+ fail_unless (addr2 != NULL);
+ fail_unless (addr2->port == 5000);
+ fail_unless (!strcmp (addr2->address, "233.252.1.3"));
+
+ addr3 = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr3 != NULL);
+ fail_unless (addr3->port == 5000);
+ fail_unless (!strcmp (addr3->address, "233.252.1.2"));
+
+ fail_unless (gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2)
+ == NULL);
+
+ gst_rtsp_address_free (addr);
+ gst_rtsp_address_free (addr2);
+ gst_rtsp_address_free (addr3);
+ gst_rtsp_address_pool_clear (pool);
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.1.1", "233.252.1.1", 5000, 5001, 1));
+ fail_if (gst_rtsp_address_pool_has_unicast_addresses (pool));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "192.168.1.1", "192.168.1.1", 6000, 6001, 0));
+ fail_unless (gst_rtsp_address_pool_has_unicast_addresses (pool));
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5000);
+ fail_unless (!strcmp (addr->address, "233.252.1.1"));
+ gst_rtsp_address_free (addr);
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST, 2);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 6000);
+ fail_unless (!strcmp (addr->address, "192.168.1.1"));
+ gst_rtsp_address_free (addr);
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ GST_RTSP_ADDRESS_POOL_ANY_IPV4, GST_RTSP_ADDRESS_POOL_ANY_IPV4, 5000,
+ 5001, 0));
+ res =
+ gst_rtsp_address_pool_reserve_address (pool, "192.168.0.1", 5000, 1, 0,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ res =
+ gst_rtsp_address_pool_reserve_address (pool, "0.0.0.0", 5000, 1, 0,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_OK);
+ gst_rtsp_address_free (addr);
+ gst_rtsp_address_pool_clear (pool);
+
+ /* Error case 2. Using ANY as min address makes it possible to allocate the
+ * same address twice */
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ GST_RTSP_ADDRESS_POOL_ANY_IPV4, "255.255.255.255", 5000, 5001, 0));
+ res =
+ gst_rtsp_address_pool_reserve_address (pool, "192.168.0.1", 5000, 1, 0,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_OK);
+ res =
+ gst_rtsp_address_pool_reserve_address (pool, "192.168.0.1", 5000, 1, 0,
+ &addr2);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERESERVED);
+ gst_rtsp_address_free (addr);
+ gst_rtsp_address_pool_clear (pool);
+
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspaddresspool_suite (void)
+{
+ Suite *s = suite_create ("rtspaddresspool");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_pool);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspaddresspool);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-client.h>
+
+#define VIDEO_PIPELINE "videotestsrc ! " \
+ "video/x-raw,width=352,height=288 ! " \
+ "rtpgstpay name=pay0 pt=96"
+#define AUDIO_PIPELINE "audiotestsrc ! " \
+ "audio/x-raw,rate=8000 ! " \
+ "rtpgstpay name=pay1 pt=97"
+
+static gchar *session_id;
+static gint cseq;
+static guint expected_session_timeout = 60;
+static const gchar *expected_unsupported_header;
+static const gchar *expected_scale_header;
+static const gchar *expected_speed_header;
+static gdouble fake_rate_value = 0;
+static gdouble fake_applied_rate_value = 0;
+
+static gboolean
+test_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OK);
+ fail_unless (g_str_equal (reason, "OK"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static gboolean
+test_response_play_200 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *str;
+ gchar **session_hdr_params;
+ gchar *pattern;
+
+ fail_unless_equals_int (gst_rtsp_message_get_type (response),
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless_equals_int (code, GST_RTSP_STS_OK);
+ fail_unless_equals_string (reason, "OK");
+ fail_unless_equals_int (version, GST_RTSP_VERSION_1_0);
+
+ /* Verify mandatory headers according to RFC 2326 */
+ /* verify mandatory CSeq header */
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+ 0) == GST_RTSP_OK);
+ fail_unless (atoi (str) == cseq++);
+
+ /* verify mandatory Session header */
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION,
+ &str, 0) == GST_RTSP_OK);
+ session_hdr_params = g_strsplit (str, ";", -1);
+ fail_unless (session_hdr_params[0] != NULL);
+ g_strfreev (session_hdr_params);
+
+ /* verify mandatory RTP-Info header */
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_RTP_INFO,
+ &str, 0) == GST_RTSP_OK);
+ pattern = g_strdup_printf ("^url=rtsp://.+;seq=[0-9]+;rtptime=[0-9]+");
+ fail_unless (g_regex_match_simple (pattern, str, 0, 0),
+ "GST_RTSP_HDR_RTP_INFO '%s' doesn't match pattern '%s'", str, pattern);
+ g_free (pattern);
+
+ return TRUE;
+}
+
+static gboolean
+test_response_400 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_BAD_REQUEST);
+ fail_unless (g_str_equal (reason, "Bad Request"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static gboolean
+test_response_404 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_NOT_FOUND);
+ fail_unless (g_str_equal (reason, "Not Found"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static gboolean
+test_response_454 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_SESSION_NOT_FOUND);
+ fail_unless (g_str_equal (reason, "Session Not Found"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static gboolean
+test_response_551 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *options;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OPTION_NOT_SUPPORTED);
+ fail_unless (g_str_equal (reason, "Option not supported"));
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_UNSUPPORTED,
+ &options, 0) == GST_RTSP_OK);
+ fail_unless (!g_strcmp0 (expected_unsupported_header, options));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static void
+create_connection (GstRTSPConnection ** conn)
+{
+ GSocket *sock;
+ GError *error = NULL;
+
+ sock = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_STREAM,
+ G_SOCKET_PROTOCOL_TCP, &error);
+ g_assert_no_error (error);
+ fail_unless (gst_rtsp_connection_create_from_socket (sock, "127.0.0.1", 444,
+ NULL, conn) == GST_RTSP_OK);
+ g_object_unref (sock);
+}
+
+static GstRTSPClient *
+setup_client (const gchar * launch_line, const gchar * mount_point,
+ gboolean enable_rtcp)
+{
+ GstRTSPClient *client;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPMountPoints *mount_points;
+ GstRTSPMediaFactory *factory;
+ GstRTSPThreadPool *thread_pool;
+
+ client = gst_rtsp_client_new ();
+
+ session_pool = gst_rtsp_session_pool_new ();
+ gst_rtsp_client_set_session_pool (client, session_pool);
+
+ mount_points = gst_rtsp_mount_points_new ();
+ factory = gst_rtsp_media_factory_new ();
+ if (launch_line == NULL)
+ gst_rtsp_media_factory_set_launch (factory,
+ "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
+ else
+ gst_rtsp_media_factory_set_launch (factory, launch_line);
+
+ gst_rtsp_media_factory_set_enable_rtcp (factory, enable_rtcp);
+
+ gst_rtsp_mount_points_add_factory (mount_points, mount_point, factory);
+ gst_rtsp_client_set_mount_points (client, mount_points);
+
+ thread_pool = gst_rtsp_thread_pool_new ();
+ gst_rtsp_client_set_thread_pool (client, thread_pool);
+
+ g_object_unref (mount_points);
+ g_object_unref (session_pool);
+ g_object_unref (thread_pool);
+
+ return client;
+}
+
+static void
+teardown_client (GstRTSPClient * client)
+{
+ gst_rtsp_client_set_thread_pool (client, NULL);
+ g_object_unref (client);
+}
+
+static gchar *
+check_requirements_cb (GstRTSPClient * client, GstRTSPContext * ctx,
+ gchar ** req, gpointer user_data)
+{
+ int index = 0;
+ GString *result = g_string_new ("");
+
+ while (req[index] != NULL) {
+ if (g_strcmp0 (req[index], "test-requirements")) {
+ if (result->len > 0)
+ g_string_append (result, ", ");
+ g_string_append (result, req[index]);
+ }
+ index++;
+ }
+
+ return g_string_free (result, FALSE);
+}
+
+GST_START_TEST (test_require)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = gst_rtsp_client_new ();
+
+ /* require header without handler */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("test-not-supported1");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
+ g_free (str);
+
+ expected_unsupported_header = "test-not-supported1";
+ gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ g_signal_connect (G_OBJECT (client), "check-requirements",
+ G_CALLBACK (check_requirements_cb), NULL);
+
+ /* one supported option */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("test-requirements");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* unsupported option */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("test-not-supported1");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
+ g_free (str);
+
+ expected_unsupported_header = "test-not-supported1";
+ gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* more than one unsupported options */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("test-not-supported1");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
+ g_free (str);
+ str = g_strdup_printf ("test-not-supported2");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
+ g_free (str);
+
+ expected_unsupported_header = "test-not-supported1, test-not-supported2";
+ gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* supported and unsupported together */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("test-not-supported1");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
+ g_free (str);
+ str = g_strdup_printf ("test-requirements");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
+ g_free (str);
+ str = g_strdup_printf ("test-not-supported2");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
+ g_free (str);
+
+ expected_unsupported_header = "test-not-supported1, test-not-supported2";
+ gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ g_object_unref (client);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_request)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPConnection *conn;
+
+ client = gst_rtsp_client_new ();
+
+ /* OPTIONS with invalid url */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "foopy://padoop/") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+
+ gst_rtsp_message_unset (&request);
+
+ /* OPTIONS with unknown session id */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, "foobar");
+
+ gst_rtsp_client_set_send_func (client, test_response_454, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+
+ gst_rtsp_message_unset (&request);
+
+ /* OPTIONS with an absolute path instead of an absolute url */
+ /* set host information */
+ create_connection (&conn);
+ fail_unless (gst_rtsp_client_set_connection (client, conn));
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* OPTIONS with an absolute path instead of an absolute url with invalid
+ * host information */
+ g_object_unref (client);
+ client = gst_rtsp_client_new ();
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ g_object_unref (client);
+}
+
+GST_END_TEST;
+
+static gboolean
+test_option_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *str;
+ GstRTSPMethod methods;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OK);
+ fail_unless (g_str_equal (reason, "OK"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+ 0) == GST_RTSP_OK);
+ fail_unless (atoi (str) == cseq++);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_PUBLIC, &str,
+ 0) == GST_RTSP_OK);
+
+ methods = gst_rtsp_options_from_text (str);
+ fail_if (methods == 0);
+ fail_unless (methods == (GST_RTSP_DESCRIBE |
+ GST_RTSP_ANNOUNCE |
+ GST_RTSP_OPTIONS |
+ GST_RTSP_PAUSE |
+ GST_RTSP_PLAY |
+ GST_RTSP_RECORD |
+ GST_RTSP_SETUP |
+ GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN));
+
+ return TRUE;
+}
+
+GST_START_TEST (test_options)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = gst_rtsp_client_new ();
+
+ /* simple OPTIONS */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_option_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ g_object_unref (client);
+}
+
+GST_END_TEST;
+
+static void
+test_describe_sub (const gchar * mount_point, const gchar * url)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = gst_rtsp_client_new ();
+
+ /* simple DESCRIBE for non-existing url */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+ url) == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_404, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ g_object_unref (client);
+
+ /* simple DESCRIBE for an existing url */
+ client = setup_client (NULL, mount_point, TRUE);
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+ url) == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ teardown_client (client);
+}
+
+GST_START_TEST (test_describe)
+{
+ test_describe_sub ("/test", "rtsp://localhost/test");
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_describe_root_mount_point)
+{
+ test_describe_sub ("/", "rtsp://localhost");
+}
+
+GST_END_TEST;
+
+static const gchar *expected_transport = NULL;
+
+static gboolean
+test_setup_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *str;
+ gchar *pattern;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPSession *session;
+ gchar **session_hdr_params;
+
+ fail_unless (expected_transport != NULL);
+
+ fail_unless_equals_int (gst_rtsp_message_get_type (response),
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless_equals_int (code, GST_RTSP_STS_OK);
+ fail_unless_equals_string (reason, "OK");
+ fail_unless_equals_int (version, GST_RTSP_VERSION_1_0);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+ 0) == GST_RTSP_OK);
+ fail_unless (atoi (str) == cseq++);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT,
+ &str, 0) == GST_RTSP_OK);
+
+ pattern = g_strdup_printf ("^%s$", expected_transport);
+ fail_unless (g_regex_match_simple (pattern, str, 0, 0),
+ "Transport '%s' doesn't match pattern '%s'", str, pattern);
+ g_free (pattern);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION,
+ &str, 0) == GST_RTSP_OK);
+ session_hdr_params = g_strsplit (str, ";", -1);
+
+ /* session-id value */
+ fail_unless (session_hdr_params[0] != NULL);
+
+ if (expected_session_timeout != 60) {
+ /* session timeout param */
+ gchar *timeout_str = g_strdup_printf ("timeout=%u",
+ expected_session_timeout);
+
+ fail_unless (session_hdr_params[1] != NULL);
+ g_strstrip (session_hdr_params[1]);
+ fail_unless (g_strcmp0 (session_hdr_params[1], timeout_str) == 0);
+ g_free (timeout_str);
+ }
+
+ session_pool = gst_rtsp_client_get_session_pool (client);
+ fail_unless (session_pool != NULL);
+
+ session = gst_rtsp_session_pool_find (session_pool, session_hdr_params[0]);
+ g_strfreev (session_hdr_params);
+
+ /* remember session id to be able to send teardown */
+ if (session_id)
+ g_free (session_id);
+ session_id = g_strdup (gst_rtsp_session_get_sessionid (session));
+ fail_unless (session_id != NULL);
+
+ fail_unless (session != NULL);
+ g_object_unref (session);
+
+ g_object_unref (session_pool);
+
+
+ return TRUE;
+}
+
+static gboolean
+test_setup_response_461 (GstRTSPClient * client,
+ GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *str;
+
+ fail_unless (expected_transport == NULL);
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_UNSUPPORTED_TRANSPORT);
+ fail_unless (g_str_equal (reason, "Unsupported transport"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+ 0) == GST_RTSP_OK);
+ fail_unless (atoi (str) == cseq++);
+
+
+ return TRUE;
+}
+
+static gboolean
+test_teardown_response_200 (GstRTSPClient * client,
+ GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OK);
+ fail_unless (g_str_equal (reason, "OK"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static void
+send_teardown (GstRTSPClient * client, const gchar * url)
+{
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ fail_unless (session_id != NULL);
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN,
+ url) == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client, test_teardown_response_200,
+ NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ g_free (session_id);
+ session_id = NULL;
+}
+
+static void
+test_setup_tcp_sub (const gchar * mount_point, const gchar * url1,
+ const gchar * url2)
+{
+ GstRTSPClient *client;
+ GstRTSPConnection *conn;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_client (NULL, mount_point, TRUE);
+ create_connection (&conn);
+ fail_unless (gst_rtsp_client_set_connection (client, conn));
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ url1) == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP/TCP;unicast");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ expected_transport =
+ "RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=.*;mode=\"PLAY\"";
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+
+ gst_rtsp_message_unset (&request);
+
+ send_teardown (client, url2);
+ teardown_client (client);
+}
+
+GST_START_TEST (test_setup_tcp)
+{
+ test_setup_tcp_sub ("/test", "rtsp://localhost/test/stream=0",
+ "rtsp://localhost/test");
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_tcp_root_mount_point)
+{
+ test_setup_tcp_sub ("/", "rtsp://localhost/stream=0", "rtsp://localhost");
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_no_rtcp)
+{
+ GstRTSPClient *client;
+ GstRTSPConnection *conn;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_client (NULL, "/test", FALSE);
+ create_connection (&conn);
+ fail_unless (gst_rtsp_client_set_connection (client, conn));
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;unicast;client_port=5000-5001");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ /* We want to verify that server_port holds a single number, not a range */
+ expected_transport =
+ "RTP/AVP;unicast;client_port=5000-5001;server_port=[0-9]+;ssrc=.*;mode=\"PLAY\"";
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+
+ gst_rtsp_message_unset (&request);
+
+ send_teardown (client, "rtsp://localhost/test");
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+static void
+test_setup_tcp_two_streams_same_channels_sub (const gchar * mount_point,
+ const gchar * url1, const gchar * url2, const gchar * url3)
+{
+ GstRTSPClient *client;
+ GstRTSPConnection *conn;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_client (NULL, mount_point, TRUE);
+ create_connection (&conn);
+ fail_unless (gst_rtsp_client_set_connection (client, conn));
+
+ /* test SETUP of a video stream with 0-1 as interleaved channels */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ url1) == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP/TCP;unicast;interleaved=0-1");
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ expected_transport =
+ "RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=.*;mode=\"PLAY\"";
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* test SETUP of an audio stream with *the same* interleaved channels.
+ * we expect the server to allocate new channel numbers */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ url2) == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP/TCP;unicast;interleaved=0-1");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ expected_transport =
+ "RTP/AVP/TCP;unicast;interleaved=2-3;ssrc=.*;mode=\"PLAY\"";
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ send_teardown (client, url3);
+ teardown_client (client);
+}
+
+GST_START_TEST (test_setup_tcp_two_streams_same_channels)
+{
+ test_setup_tcp_two_streams_same_channels_sub ("/test",
+ "rtsp://localhost/test/stream=0", "rtsp://localhost/test/stream=1",
+ "rtsp://localhost/test");
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_tcp_two_streams_same_channels_root_mount_point)
+{
+ test_setup_tcp_two_streams_same_channels_sub ("/",
+ "rtsp://localhost/stream=0", "rtsp://localhost/stream=1",
+ "rtsp://localhost");
+}
+
+GST_END_TEST;
+
+static GstRTSPClient *
+setup_multicast_client (guint max_ttl, const gchar * mount_point)
+{
+ GstRTSPClient *client;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPMountPoints *mount_points;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAddressPool *address_pool;
+ GstRTSPThreadPool *thread_pool;
+
+ client = gst_rtsp_client_new ();
+
+ session_pool = gst_rtsp_session_pool_new ();
+ gst_rtsp_client_set_session_pool (client, session_pool);
+
+ mount_points = gst_rtsp_mount_points_new ();
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
+ address_pool = gst_rtsp_address_pool_new ();
+ fail_unless (gst_rtsp_address_pool_add_range (address_pool,
+ "233.252.0.1", "233.252.0.1", 5000, 5010, 1));
+ gst_rtsp_media_factory_set_address_pool (factory, address_pool);
+ gst_rtsp_media_factory_add_role (factory, "user",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_mount_points_add_factory (mount_points, mount_point, factory);
+ gst_rtsp_client_set_mount_points (client, mount_points);
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory, max_ttl);
+
+ thread_pool = gst_rtsp_thread_pool_new ();
+ gst_rtsp_client_set_thread_pool (client, thread_pool);
+
+ g_object_unref (mount_points);
+ g_object_unref (session_pool);
+ g_object_unref (address_pool);
+ g_object_unref (thread_pool);
+
+ return client;
+}
+
+GST_START_TEST (test_client_multicast_transport_404)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_multicast_client (1, "/test");
+
+ /* simple SETUP for non-existing url */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test2/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast");
+
+ gst_rtsp_client_set_send_func (client, test_response_404, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+static void
+new_session_cb (GObject * client, GstRTSPSession * session, gpointer user_data)
+{
+ GST_DEBUG ("%p: new session %p", client, session);
+ gst_rtsp_session_set_timeout (session, expected_session_timeout);
+}
+
+GST_START_TEST (test_client_multicast_transport)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_multicast_client (1, "/test");
+
+ expected_session_timeout = 20;
+ g_signal_connect (G_OBJECT (client), "new-session",
+ G_CALLBACK (new_session_cb), NULL);
+
+ /* simple SETUP with a valid URI and multicast */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast");
+
+ expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+ expected_session_timeout = 60;
+
+ send_teardown (client, "rtsp://localhost/test");
+
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_multicast_ignore_transport_specific)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_multicast_client (1, "/test");
+
+ /* simple SETUP with a valid URI and multicast and a specific dest,
+ * but ignore it */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast;destination=233.252.0.2;ttl=2;port=5001-5006;");
+
+ expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+
+ send_teardown (client, "rtsp://localhost/test");
+
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+static void
+multicast_transport_specific (void)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPContext ctx = { NULL };
+
+ client = setup_multicast_client (1, "/test");
+
+ ctx.client = client;
+ ctx.auth = gst_rtsp_auth_new ();
+ ctx.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+ G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx);
+
+ /* simple SETUP with a valid URI */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ expected_transport);
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ session_pool = gst_rtsp_client_get_session_pool (client);
+ fail_unless (session_pool != NULL);
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1);
+ g_object_unref (session_pool);
+
+ /* send PLAY request */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ send_teardown (client, "rtsp://localhost/test");
+ teardown_client (client);
+ g_object_unref (ctx.auth);
+ gst_rtsp_token_unref (ctx.token);
+ gst_rtsp_context_pop_current (&ctx);
+}
+
+/* CASE: multicast address requested by the client exists in the address pool */
+GST_START_TEST (test_client_multicast_transport_specific)
+{
+ expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ multicast_transport_specific ();
+ expected_transport = NULL;
+}
+
+GST_END_TEST;
+
+/* CASE: multicast address requested by the client does not exist in the address pool */
+GST_START_TEST (test_client_multicast_transport_specific_no_address_in_pool)
+{
+ expected_transport = "RTP/AVP;multicast;destination=234.252.0.3;"
+ "ttl=1;port=10002-10004;mode=\"PLAY\"";
+ multicast_transport_specific ();
+ expected_transport = NULL;
+}
+
+GST_END_TEST;
+
+static gboolean
+test_response_sdp (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ guint8 *data;
+ guint size;
+ GstSDPMessage *sdp_msg;
+ const GstSDPMedia *sdp_media;
+ const GstSDPBandwidth *bw;
+ gint bandwidth_val = GPOINTER_TO_INT (user_data);
+
+ fail_unless (gst_rtsp_message_get_body (response, &data, &size)
+ == GST_RTSP_OK);
+ gst_sdp_message_new (&sdp_msg);
+ fail_unless (gst_sdp_message_parse_buffer (data, size, sdp_msg)
+ == GST_SDP_OK);
+
+ /* session description */
+ /* v= */
+ fail_unless (gst_sdp_message_get_version (sdp_msg) != NULL);
+ /* o= */
+ fail_unless (gst_sdp_message_get_origin (sdp_msg) != NULL);
+ /* s= */
+ fail_unless (gst_sdp_message_get_session_name (sdp_msg) != NULL);
+ /* t=0 0 */
+ fail_unless (gst_sdp_message_times_len (sdp_msg) == 0);
+
+ /* verify number of medias */
+ fail_unless (gst_sdp_message_medias_len (sdp_msg) == 1);
+
+ /* media description */
+ sdp_media = gst_sdp_message_get_media (sdp_msg, 0);
+ fail_unless (sdp_media != NULL);
+
+ /* m= */
+ fail_unless (gst_sdp_media_get_media (sdp_media) != NULL);
+
+ /* media bandwidth */
+ if (bandwidth_val) {
+ fail_unless (gst_sdp_media_bandwidths_len (sdp_media) == 1);
+ bw = gst_sdp_media_get_bandwidth (sdp_media, 0);
+ fail_unless (bw != NULL);
+ fail_unless (g_strcmp0 (bw->bwtype, "AS") == 0);
+ fail_unless (bw->bandwidth == bandwidth_val);
+ } else {
+ fail_unless (gst_sdp_media_bandwidths_len (sdp_media) == 0);
+ }
+
+ gst_sdp_message_free (sdp_msg);
+
+ return TRUE;
+}
+
+static void
+test_client_sdp (const gchar * launch_line, guint * bandwidth_val)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ /* simple DESCRIBE for an existing url */
+ client = setup_client (launch_line, "/test", TRUE);
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_sdp,
+ (gpointer) bandwidth_val, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ teardown_client (client);
+}
+
+GST_START_TEST (test_client_sdp_with_max_bitrate_tag)
+{
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"maximum-bitrate=(uint)50000000\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (50000));
+
+
+ /* max-bitrate=0: no bandwidth line */
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"maximum-bitrate=(uint)0\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (0));
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_sdp_with_bitrate_tag)
+{
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"bitrate=(uint)7000000\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (7000));
+
+ /* bitrate=0: no bandwdith line */
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"bitrate=(uint)0\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (0));
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_sdp_with_max_bitrate_and_bitrate_tags)
+{
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"bitrate=(uint)7000000,maximum-bitrate=(uint)50000000\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (50000));
+
+ /* max-bitrate is zero: fallback to bitrate */
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"bitrate=(uint)7000000,maximum-bitrate=(uint)0\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (7000));
+
+ /* max-bitrate=bitrate=0o: no bandwidth line */
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"bitrate=(uint)0,maximum-bitrate=(uint)0\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (0));
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_sdp_with_no_bitrate_tags)
+{
+ test_client_sdp ("videotestsrc "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", NULL);
+}
+
+GST_END_TEST;
+
+static void
+mcast_transport_two_clients (gboolean shared, const gchar * transport1,
+ const gchar * expected_transport1, const gchar * addr1,
+ const gchar * transport2, const gchar * expected_transport2,
+ const gchar * addr2, gboolean bind_mcast_address)
+{
+ GstRTSPClient *client1, *client2;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPContext ctx = { NULL };
+ GstRTSPContext ctx2 = { NULL };
+ GstRTSPMountPoints *mount_points;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAddressPool *address_pool;
+ GstRTSPThreadPool *thread_pool;
+ gchar *session_id1;
+ gchar *client_addr = NULL;
+
+ mount_points = gst_rtsp_mount_points_new ();
+ factory = gst_rtsp_media_factory_new ();
+ if (shared)
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory, 5);
+ gst_rtsp_media_factory_set_bind_mcast_address (factory, bind_mcast_address);
+ gst_rtsp_media_factory_set_launch (factory,
+ "audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
+ address_pool = gst_rtsp_address_pool_new ();
+ fail_unless (gst_rtsp_address_pool_add_range (address_pool,
+ "233.252.0.1", "233.252.0.1", 5000, 5001, 1));
+ gst_rtsp_media_factory_set_address_pool (factory, address_pool);
+ gst_rtsp_media_factory_add_role (factory, "user",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
+ session_pool = gst_rtsp_session_pool_new ();
+ thread_pool = gst_rtsp_thread_pool_new ();
+
+ /* first multicast client with transport specific request */
+ client1 = gst_rtsp_client_new ();
+ gst_rtsp_client_set_session_pool (client1, session_pool);
+ gst_rtsp_client_set_mount_points (client1, mount_points);
+ gst_rtsp_client_set_thread_pool (client1, thread_pool);
+
+ ctx.client = client1;
+ ctx.auth = gst_rtsp_auth_new ();
+ ctx.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+ G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx);
+
+ expected_transport = expected_transport1;
+
+ /* send SETUP request */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport1);
+
+ gst_rtsp_client_set_send_func (client1, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client1,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+
+ /* send PLAY request */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client1, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client1,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* check address */
+ client_addr = gst_rtsp_stream_get_multicast_client_addresses (ctx.stream);
+ fail_if (client_addr == NULL);
+ fail_unless (g_str_equal (client_addr, addr1));
+ g_free (client_addr);
+
+ gst_rtsp_context_pop_current (&ctx);
+ session_id1 = g_strdup (session_id);
+
+ /* second multicast client with transport specific request */
+ cseq = 0;
+ client2 = gst_rtsp_client_new ();
+ gst_rtsp_client_set_session_pool (client2, session_pool);
+ gst_rtsp_client_set_mount_points (client2, mount_points);
+ gst_rtsp_client_set_thread_pool (client2, thread_pool);
+
+ ctx2.client = client2;
+ ctx2.auth = gst_rtsp_auth_new ();
+ ctx2.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+ G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx2);
+
+ expected_transport = expected_transport2;
+
+ /* send SETUP request */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport2);
+
+ gst_rtsp_client_set_send_func (client2, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client2,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+
+ /* send PLAY request */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client2, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client2,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* check addresses */
+ client_addr = gst_rtsp_stream_get_multicast_client_addresses (ctx2.stream);
+ fail_if (client_addr == NULL);
+ if (shared) {
+ if (g_str_equal (addr1, addr2)) {
+ fail_unless (g_str_equal (client_addr, addr1));
+ } else {
+ gchar *addr_str = g_strdup_printf ("%s,%s", addr2, addr1);
+ fail_unless (g_str_equal (client_addr, addr_str));
+ g_free (addr_str);
+ }
+ } else {
+ fail_unless (g_str_equal (client_addr, addr2));
+ }
+ g_free (client_addr);
+
+ send_teardown (client2, "rtsp://localhost/test");
+ gst_rtsp_context_pop_current (&ctx2);
+
+ gst_rtsp_context_push_current (&ctx);
+ session_id = session_id1;
+ send_teardown (client1, "rtsp://localhost/test");
+ gst_rtsp_context_pop_current (&ctx);
+
+ teardown_client (client1);
+ teardown_client (client2);
+ g_object_unref (ctx.auth);
+ g_object_unref (ctx2.auth);
+ gst_rtsp_token_unref (ctx.token);
+ gst_rtsp_token_unref (ctx2.token);
+ g_object_unref (mount_points);
+ g_object_unref (session_pool);
+ g_object_unref (address_pool);
+ g_object_unref (thread_pool);
+}
+
+/* CASE: media is shared.
+ * client 1: SETUP --->
+ * client 1: PLAY --->
+ * client 2: SETUP --->
+ * client 1: TEARDOWN --->
+ * client 2: PLAY --->
+ * client 2: TEARDOWN --->
+ */
+static void
+mcast_transport_two_clients_teardown_play (const gchar * transport1,
+ const gchar * expected_transport1, const gchar * transport2,
+ const gchar * expected_transport2, gboolean bind_mcast_address,
+ gboolean is_shared)
+{
+ GstRTSPClient *client1, *client2;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPContext ctx = { NULL };
+ GstRTSPContext ctx2 = { NULL };
+ GstRTSPMountPoints *mount_points;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAddressPool *address_pool;
+ GstRTSPThreadPool *thread_pool;
+ gchar *session_id1, *session_id2;
+
+ mount_points = gst_rtsp_mount_points_new ();
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_shared (factory, is_shared);
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory, 5);
+ gst_rtsp_media_factory_set_bind_mcast_address (factory, bind_mcast_address);
+ gst_rtsp_media_factory_set_launch (factory,
+ "audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
+ address_pool = gst_rtsp_address_pool_new ();
+ if (is_shared)
+ fail_unless (gst_rtsp_address_pool_add_range (address_pool,
+ "233.252.0.1", "233.252.0.1", 5000, 5001, 1));
+ else
+ fail_unless (gst_rtsp_address_pool_add_range (address_pool,
+ "233.252.0.1", "233.252.0.1", 5000, 5003, 1));
+ gst_rtsp_media_factory_set_address_pool (factory, address_pool);
+ gst_rtsp_media_factory_add_role (factory, "user",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
+ session_pool = gst_rtsp_session_pool_new ();
+ thread_pool = gst_rtsp_thread_pool_new ();
+
+ /* client 1 configuration */
+ client1 = gst_rtsp_client_new ();
+ gst_rtsp_client_set_session_pool (client1, session_pool);
+ gst_rtsp_client_set_mount_points (client1, mount_points);
+ gst_rtsp_client_set_thread_pool (client1, thread_pool);
+
+ ctx.client = client1;
+ ctx.auth = gst_rtsp_auth_new ();
+ ctx.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+ G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx);
+
+ expected_transport = expected_transport1;
+
+ /* client 1 sends SETUP request */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport1);
+
+ gst_rtsp_client_set_send_func (client1, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client1,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+
+
+ /* client 1 sends PLAY request */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client1, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client1,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ gst_rtsp_context_pop_current (&ctx);
+ session_id1 = g_strdup (session_id);
+
+ /* client 2 configuration */
+ cseq = 0;
+ client2 = gst_rtsp_client_new ();
+ gst_rtsp_client_set_session_pool (client2, session_pool);
+ gst_rtsp_client_set_mount_points (client2, mount_points);
+ gst_rtsp_client_set_thread_pool (client2, thread_pool);
+
+ ctx2.client = client2;
+ ctx2.auth = gst_rtsp_auth_new ();
+ ctx2.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+ G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx2);
+
+ expected_transport = expected_transport2;
+
+ /* client 2 sends SETUP request */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport2);
+
+ gst_rtsp_client_set_send_func (client2, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client2,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+
+ session_id2 = g_strdup (session_id);
+ g_free (session_id);
+ gst_rtsp_context_pop_current (&ctx2);
+
+ /* the first client sends TEARDOWN request */
+ gst_rtsp_context_push_current (&ctx);
+ session_id = session_id1;
+ send_teardown (client1, "rtsp://localhost/test");
+ gst_rtsp_context_pop_current (&ctx);
+ teardown_client (client1);
+
+ /* the second client sends PLAY request */
+ gst_rtsp_context_push_current (&ctx2);
+ session_id = session_id2;
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client2, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client2,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* client 2 sends TEARDOWN request */
+ send_teardown (client2, "rtsp://localhost/test");
+ gst_rtsp_context_pop_current (&ctx2);
+
+ teardown_client (client2);
+ g_object_unref (ctx.auth);
+ g_object_unref (ctx2.auth);
+ gst_rtsp_token_unref (ctx.token);
+ gst_rtsp_token_unref (ctx2.token);
+ g_object_unref (mount_points);
+ g_object_unref (session_pool);
+ g_object_unref (address_pool);
+ g_object_unref (thread_pool);
+}
+
+/* test if two multicast clients can choose different transport settings
+ * CASE: media is shared */
+GST_START_TEST
+ (test_client_multicast_transport_specific_two_clients_shared_media) {
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=1;port=5002-5003;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = transport_client_2;
+ const gchar *addr_client_2 = "233.252.0.2:5002";
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2, FALSE);
+}
+
+GST_END_TEST;
+
+/* test if two multicast clients can choose different transport settings
+ * CASE: media is not shared */
+GST_START_TEST (test_client_multicast_transport_specific_two_clients)
+{
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=1;port=5002-5003;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = transport_client_2;
+ const gchar *addr_client_2 = "233.252.0.2:5002";
+
+ mcast_transport_two_clients (FALSE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2, FALSE);
+}
+
+GST_END_TEST;
+
+/* test if two multicast clients can choose the same ports but different
+ * multicast destinations
+ * CASE: media is not shared */
+GST_START_TEST (test_client_multicast_transport_specific_two_clients_same_ports)
+{
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=9000-9001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:9000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=1;port=9000-9001;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = transport_client_2;
+ const gchar *addr_client_2 = "233.252.0.2:9000";
+
+ /* configure the multicast socket to be bound to the requested multicast address instead of INADDR_ANY.
+ * The clients request the same rtp/rtcp borts and having the socket that are bound to ANY would result
+ * in bind() failure */
+ gboolean allow_bind_mcast_address = TRUE;
+
+ mcast_transport_two_clients (FALSE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2, allow_bind_mcast_address);
+}
+
+GST_END_TEST;
+
+/* test if two multicast clients can choose the same multicast destination but different
+ * ports
+ * CASE: media is not shared */
+GST_START_TEST
+ (test_client_multicast_transport_specific_two_clients_same_destination) {
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=1;port=9002-9003;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.2:9002";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=1;port=9004-9005;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = transport_client_2;
+ const gchar *addr_client_2 = "233.252.0.2:9004";
+
+ mcast_transport_two_clients (FALSE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2, FALSE);
+}
+
+GST_END_TEST;
+/* test if two multicast clients can choose the same transport settings.
+ * CASE: media is shared */
+GST_START_TEST
+ (test_client_multicast_transport_specific_two_clients_shared_media_same_transport)
+{
+
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = transport_client_1;
+ const gchar *expected_transport_2 = expected_transport_1;
+ const gchar *addr_client_2 = addr_client_1;
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2, FALSE);
+}
+
+GST_END_TEST;
+
+/* test if two multicast clients get the same transport settings without
+ * requesting specific transport.
+ * CASE: media is shared */
+GST_START_TEST (test_client_multicast_two_clients_shared_media)
+{
+ const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\"";
+ const gchar *expected_transport_1 =
+ "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = transport_client_1;
+ const gchar *expected_transport_2 = expected_transport_1;
+ const gchar *addr_client_2 = addr_client_1;
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2, FALSE);
+}
+
+GST_END_TEST;
+
+/* test if it's possible to play the shared media, after one of the clients
+ * has terminated its session.
+ */
+GST_START_TEST (test_client_multicast_two_clients_shared_media_teardown_play)
+{
+ const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\"";
+ const gchar *expected_transport_1 =
+ "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+
+ const gchar *transport_client_2 = transport_client_1;
+ const gchar *expected_transport_2 = expected_transport_1;
+
+ mcast_transport_two_clients_teardown_play (transport_client_1,
+ expected_transport_1, transport_client_2, expected_transport_2, FALSE,
+ TRUE);
+}
+
+GST_END_TEST;
+
+/* test if it's possible to play the shared media, after one of the clients
+ * has terminated its session.
+ */
+GST_START_TEST
+ (test_client_multicast_two_clients_not_shared_media_teardown_play) {
+ const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\"";
+ const gchar *expected_transport_1 =
+ "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+
+ const gchar *transport_client_2 = transport_client_1;
+ const gchar *expected_transport_2 =
+ "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5002-5003;mode=\"PLAY\"";
+
+ mcast_transport_two_clients_teardown_play (transport_client_1,
+ expected_transport_1, transport_client_2, expected_transport_2, FALSE,
+ FALSE);
+}
+
+GST_END_TEST;
+
+/* test if two multicast clients get the different transport settings: the first client
+ * requests the specific transport configuration while the second client lets
+ * the server select the multicast address and the ports.
+ * CASE: media is shared */
+GST_START_TEST
+ (test_client_multicast_two_clients_first_specific_transport_shared_media) {
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = expected_transport_1;
+ const gchar *addr_client_2 = addr_client_1;
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2, FALSE);
+}
+
+GST_END_TEST;
+/* test if two multicast clients get the different transport settings: the first client lets
+ * the server select the multicast address and the ports while the second client requests
+ * the specific transport configuration.
+ * CASE: media is shared */
+GST_START_TEST
+ (test_client_multicast_two_clients_second_specific_transport_shared_media) {
+ const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\"";
+ const gchar *expected_transport_1 =
+ "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=2;port=5004-5005;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = transport_client_2;
+ const gchar *addr_client_2 = "233.252.0.2:5004";
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2, FALSE);
+}
+
+GST_END_TEST;
+
+/* test if the maximum ttl multicast value is chosen by the server
+ * CASE: the first client provides the highest ttl value */
+GST_START_TEST (test_client_multicast_max_ttl_first_client)
+{
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=3;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=1;port=5002-5003;mode=\"PLAY\"";
+ const gchar *expected_transport_2 =
+ "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=3;port=5002-5003;mode=\"PLAY\"";
+ const gchar *addr_client_2 = "233.252.0.2:5002";
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2, FALSE);
+}
+
+GST_END_TEST;
+
+/* test if the maximum ttl multicast value is chosen by the server
+ * CASE: the second client provides the highest ttl value */
+GST_START_TEST (test_client_multicast_max_ttl_second_client)
+{
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=2;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=4;port=5002-5003;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = transport_client_2;
+ const gchar *addr_client_2 = "233.252.0.2:5002";
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2, FALSE);
+}
+
+GST_END_TEST;
+GST_START_TEST (test_client_multicast_invalid_ttl)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPContext ctx = { NULL };
+
+ client = setup_multicast_client (3, "/test");
+
+ ctx.client = client;
+ ctx.auth = gst_rtsp_auth_new ();
+ ctx.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+ G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx);
+
+ /* simple SETUP with an invalid ttl=0 */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast;destination=233.252.0.1;ttl=0;port=5000-5001;");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ session_pool = gst_rtsp_client_get_session_pool (client);
+ fail_unless (session_pool != NULL);
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
+ g_object_unref (session_pool);
+
+ teardown_client (client);
+ g_object_unref (ctx.auth);
+ gst_rtsp_token_unref (ctx.token);
+ gst_rtsp_context_pop_current (&ctx);
+}
+
+GST_END_TEST;
+
+static gboolean
+test_response_scale_speed (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *header_value;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OK);
+ fail_unless (g_str_equal (reason, "OK"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_RANGE,
+ &header_value, 0) == GST_RTSP_OK);
+
+ if (expected_scale_header != NULL) {
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SCALE,
+ &header_value, 0) == GST_RTSP_OK);
+ ck_assert_str_eq (header_value, expected_scale_header);
+ } else {
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SCALE,
+ &header_value, 0) == GST_RTSP_ENOTIMPL);
+ }
+
+ if (expected_speed_header != NULL) {
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SPEED,
+ &header_value, 0) == GST_RTSP_OK);
+ ck_assert_str_eq (header_value, expected_speed_header);
+ } else {
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SPEED,
+ &header_value, 0) == GST_RTSP_ENOTIMPL);
+ }
+
+ return TRUE;
+}
+
+/* Probe that tweaks segment events according to the values of the
+ * fake_rate_value and fake_applied_rate_value variables. Used to simulate
+ * seek results with different combinations of rate and applied rate.
+ */
+static GstPadProbeReturn
+rate_tweaking_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+ GstEvent *event = GST_PAD_PROBE_INFO_EVENT (info);
+ GstSegment segment;
+
+ if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) {
+ GST_DEBUG ("got segment event %" GST_PTR_FORMAT, event);
+ gst_event_copy_segment (event, &segment);
+ if (fake_applied_rate_value)
+ segment.applied_rate = fake_applied_rate_value;
+ if (fake_rate_value)
+ segment.rate = fake_rate_value;
+ gst_event_unref (event);
+ info->data = gst_event_new_segment (&segment);
+ GST_DEBUG ("forwarding segment event %" GST_PTR_FORMAT,
+ GST_EVENT (info->data));
+ }
+
+ return GST_PAD_PROBE_OK;
+}
+
+static void
+attach_rate_tweaking_probe (void)
+{
+ GstRTSPContext *ctx;
+ GstRTSPMedia *media;
+ GstRTSPStream *stream;
+ GstPad *srcpad;
+
+ fail_unless ((ctx = gst_rtsp_context_get_current ()) != NULL);
+
+ media = ctx->media;
+ fail_unless (media != NULL);
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream != NULL);
+
+ srcpad = gst_rtsp_stream_get_srcpad (stream);
+ fail_unless (srcpad != NULL);
+
+ GST_DEBUG ("adding rate_tweaking_probe");
+
+ gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
+ rate_tweaking_probe, NULL, NULL);
+ gst_object_unref (srcpad);
+}
+
+static void
+do_test_scale_and_speed (const gchar * scale, const gchar * speed,
+ GstRTSPStatusCode expected_response_code)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPContext ctx = { NULL };
+
+ client = setup_multicast_client (1, "/test");
+
+ ctx.client = client;
+ ctx.auth = gst_rtsp_auth_new ();
+ ctx.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+ G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx);
+
+ expected_session_timeout = 20;
+ g_signal_connect (G_OBJECT (client), "new-session",
+ G_CALLBACK (new_session_cb), NULL);
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast");
+ expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=.*;mode=\"PLAY\"";
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+ expected_session_timeout = 60;
+
+ if (fake_applied_rate_value || fake_rate_value)
+ attach_rate_tweaking_probe ();
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+
+ if (scale != NULL)
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, scale);
+ if (speed != NULL)
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, speed);
+
+ if (expected_response_code == GST_RTSP_STS_BAD_REQUEST)
+ gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL);
+ else
+ gst_rtsp_client_set_send_func (client, test_response_scale_speed, NULL,
+ NULL);
+
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ send_teardown (client, "rtsp://localhost/test");
+ teardown_client (client);
+ g_object_unref (ctx.auth);
+ gst_rtsp_token_unref (ctx.token);
+ gst_rtsp_context_pop_current (&ctx);
+
+}
+
+GST_START_TEST (test_scale_and_speed)
+{
+ /* no scale/speed requested, no scale/speed should be received */
+ expected_scale_header = NULL;
+ expected_speed_header = NULL;
+ do_test_scale_and_speed (NULL, NULL, GST_RTSP_STS_OK);
+
+ /* scale requested, scale should be received */
+ fake_applied_rate_value = 2;
+ fake_rate_value = 1;
+ expected_scale_header = "2.000";
+ expected_speed_header = NULL;
+ do_test_scale_and_speed ("2.000", NULL, GST_RTSP_STS_OK);
+
+ /* speed requested, speed should be received */
+ fake_applied_rate_value = 0;
+ fake_rate_value = 0;
+ expected_scale_header = NULL;
+ expected_speed_header = "2.000";
+ do_test_scale_and_speed (NULL, "2.000", GST_RTSP_STS_OK);
+
+ /* both requested, both should be received */
+ fake_applied_rate_value = 2;
+ fake_rate_value = 2;
+ expected_scale_header = "2.000";
+ expected_speed_header = "2.000";
+ do_test_scale_and_speed ("2", "2", GST_RTSP_STS_OK);
+
+ /* scale requested but media doesn't handle scaling so both should be
+ * received, with scale set to 1.000 and speed set to (requested scale
+ * requested speed) */
+ fake_applied_rate_value = 0;
+ fake_rate_value = 5;
+ expected_scale_header = "1.000";
+ expected_speed_header = "5.000";
+ do_test_scale_and_speed ("5", NULL, GST_RTSP_STS_OK);
+
+ /* both requested but media only handles scaling so both should be received,
+ * with scale set to (requested scale * requested speed) and speed set to 1.00
+ */
+ fake_rate_value = 1.000;
+ fake_applied_rate_value = 4.000;
+ expected_scale_header = "4.000";
+ expected_speed_header = "1.000";
+ do_test_scale_and_speed ("2", "2", GST_RTSP_STS_OK);
+
+ /* test invalid values */
+ fake_applied_rate_value = 0;
+ fake_rate_value = 0;
+ expected_scale_header = NULL;
+ expected_speed_header = NULL;
+
+ /* scale or speed not decimal values */
+ do_test_scale_and_speed ("x", NULL, GST_RTSP_STS_BAD_REQUEST);
+ do_test_scale_and_speed (NULL, "y", GST_RTSP_STS_BAD_REQUEST);
+
+ /* scale or speed illegal decimal values */
+ do_test_scale_and_speed ("0", NULL, GST_RTSP_STS_BAD_REQUEST);
+ do_test_scale_and_speed (NULL, "0", GST_RTSP_STS_BAD_REQUEST);
+ do_test_scale_and_speed (NULL, "-2", GST_RTSP_STS_BAD_REQUEST);
+}
+
+GST_END_TEST static void
+test_client_play_sub (const gchar * mount_point, const gchar * url1,
+ const gchar * url2)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPContext ctx = { NULL };
+
+ client = setup_multicast_client (1, mount_point);
+
+ ctx.client = client;
+ ctx.auth = gst_rtsp_auth_new ();
+ ctx.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx);
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ url1) == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast");
+ /* destination is from adress pool */
+ expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=.*;mode=\"PLAY\"";
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
+ url2) == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client, test_response_play_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ send_teardown (client, url2);
+ teardown_client (client);
+ g_object_unref (ctx.auth);
+ gst_rtsp_token_unref (ctx.token);
+ gst_rtsp_context_pop_current (&ctx);
+}
+
+GST_START_TEST (test_client_play)
+{
+ test_client_play_sub ("/test", "rtsp://localhost/test/stream=0",
+ "rtsp://localhost/test");
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_play_root_mount_point)
+{
+ test_client_play_sub ("/", "rtsp://localhost/stream=0", "rtsp://localhost");
+}
+
+GST_END_TEST static Suite *
+rtspclient_suite (void)
+{
+ Suite *s = suite_create ("rtspclient");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_require);
+ tcase_add_test (tc, test_request);
+ tcase_add_test (tc, test_options);
+ tcase_add_test (tc, test_describe);
+ tcase_add_test (tc, test_describe_root_mount_point);
+ tcase_add_test (tc, test_setup_tcp);
+ tcase_add_test (tc, test_setup_tcp_root_mount_point);
+ tcase_add_test (tc, test_setup_no_rtcp);
+ tcase_add_test (tc, test_setup_tcp_two_streams_same_channels);
+ tcase_add_test (tc,
+ test_setup_tcp_two_streams_same_channels_root_mount_point);
+ tcase_add_test (tc, test_client_multicast_transport_404);
+ tcase_add_test (tc, test_client_multicast_transport);
+ tcase_add_test (tc, test_client_multicast_ignore_transport_specific);
+ tcase_add_test (tc, test_client_multicast_transport_specific);
+ tcase_add_test (tc, test_client_sdp_with_max_bitrate_tag);
+ tcase_add_test (tc, test_client_sdp_with_bitrate_tag);
+ tcase_add_test (tc, test_client_sdp_with_max_bitrate_and_bitrate_tags);
+ tcase_add_test (tc, test_client_sdp_with_no_bitrate_tags);
+ tcase_add_test (tc,
+ test_client_multicast_transport_specific_two_clients_shared_media);
+ tcase_add_test (tc, test_client_multicast_transport_specific_two_clients);
+#ifndef G_OS_WIN32
+ tcase_add_test (tc,
+ test_client_multicast_transport_specific_two_clients_same_ports);
+#else
+ /* skip the test on windows as the test restricts the multicast sockets to multicast traffic only,
+ * by specifying the multicast IP as the bind address and this currently doesn't work on Windows */
+ tcase_skip_broken_test (tc,
+ test_client_multicast_transport_specific_two_clients_same_ports);
+#endif
+ tcase_add_test (tc,
+ test_client_multicast_transport_specific_two_clients_same_destination);
+ tcase_add_test (tc,
+ test_client_multicast_transport_specific_two_clients_shared_media_same_transport);
+ tcase_add_test (tc, test_client_multicast_two_clients_shared_media);
+ tcase_add_test (tc,
+ test_client_multicast_two_clients_shared_media_teardown_play);
+ tcase_add_test (tc,
+ test_client_multicast_two_clients_not_shared_media_teardown_play);
+ tcase_add_test (tc,
+ test_client_multicast_two_clients_first_specific_transport_shared_media);
+ tcase_add_test (tc,
+ test_client_multicast_two_clients_second_specific_transport_shared_media);
+ tcase_add_test (tc,
+ test_client_multicast_transport_specific_no_address_in_pool);
+ tcase_add_test (tc, test_client_multicast_max_ttl_first_client);
+ tcase_add_test (tc, test_client_multicast_max_ttl_second_client);
+ tcase_add_test (tc, test_client_multicast_invalid_ttl);
+ tcase_add_test (tc, test_scale_and_speed);
+ tcase_add_test (tc, test_client_play);
+ tcase_add_test (tc, test_client_play_root_mount_point);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspclient);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-media-factory.h>
+
+/* Check if the media can return a SDP. We don't actually check whether
+ * the contents are valid or not */
+static gboolean
+media_has_sdp (GstRTSPMedia * media)
+{
+ GstSDPInfo info;
+ GstSDPMessage *sdp;
+ gchar *sdp_str;
+
+ info.is_ipv6 = FALSE;
+ info.server_ip = "0.0.0.0";
+
+ /* Check if media can generate a SDP */
+ gst_sdp_message_new (&sdp);
+ GST_DEBUG ("Getting SDP");
+ if (!gst_rtsp_sdp_from_media (sdp, &info, media)) {
+ GST_WARNING ("failed to get the SDP");
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+ sdp_str = gst_sdp_message_as_text (sdp);
+ GST_DEBUG ("Got SDP\n%s", sdp_str);
+ g_free (sdp_str);
+ gst_sdp_message_free (sdp);
+
+ return TRUE;
+}
+
+GST_START_TEST (test_media_seek)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream;
+ GstRTSPTimeRange *range;
+ gchar *str;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPTransport *transport;
+ gdouble rate = 0;
+ gdouble applied_rate = 0;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream != NULL);
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (media_has_sdp (media));
+
+ /* define transport */
+ fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
+ transport->lower_transport = GST_RTSP_LOWER_TRANS_TCP;
+
+ fail_unless (gst_rtsp_stream_complete_stream (stream, transport));
+
+ fail_unless (gst_rtsp_transport_free (transport) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_range_parse ("npt=5.0-", &range) == GST_RTSP_OK);
+
+ /* the media is seekable now */
+ fail_unless (gst_rtsp_media_seek (media, range));
+
+ str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ fail_unless (g_str_equal (str, "npt=5-"));
+ g_free (str);
+
+ /* seeking without rate should result in rate == 1.0 */
+ fail_unless (gst_rtsp_media_seek (media, range));
+ fail_unless (gst_rtsp_media_get_rates (media, &rate, &applied_rate));
+ fail_unless (rate == 1.0);
+ fail_unless (applied_rate == 1.0);
+
+ /* seeking with rate set to 1.5 should result in rate == 1.5 */
+ fail_unless (gst_rtsp_media_seek_trickmode (media, range,
+ GST_SEEK_FLAG_NONE, 1.5, 0));
+ fail_unless (gst_rtsp_media_get_rates (media, &rate, &applied_rate));
+ fail_unless (rate == 1.5);
+ fail_unless (applied_rate == 1.0);
+
+ gst_rtsp_range_free (range);
+
+ /* seeking with rate set to -2.0 should result in rate == -2.0 */
+ fail_unless (gst_rtsp_range_parse ("npt=10-5", &range) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_media_seek_trickmode (media, range,
+ GST_SEEK_FLAG_NONE, -2.0, 0));
+ fail_unless (gst_rtsp_media_get_rates (media, &rate, &applied_rate));
+ fail_unless (rate == -2.0);
+ fail_unless (applied_rate == 1.0);
+
+ gst_rtsp_range_free (range);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static void
+media_playback_seek_one_active_stream (const gchar * launch_line)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream1;
+ GstRTSPStream *stream2;
+ GstRTSPTimeRange *range;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPTransport *transport;
+ char *range_str;
+ GstRTSPTimeRange *play_range;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory, launch_line);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+ stream1 = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream1 != NULL);
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (media_has_sdp (media));
+
+ /* define transport */
+ fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
+ transport->lower_transport = GST_RTSP_LOWER_TRANS_TCP;
+
+ fail_unless_equals_int64 (gst_rtsp_media_seekable (media), G_MAXINT64);
+
+ /* video stream is complete and seekable */
+ fail_unless (gst_rtsp_stream_complete_stream (stream1, transport));
+ fail_unless (gst_rtsp_stream_seekable (stream1));
+
+ /* audio stream is blocked (it does not contain any transport based part),
+ * but it's seekable */
+ stream2 = gst_rtsp_media_get_stream (media, 1);
+ fail_unless (stream2 != NULL);
+ fail_unless (gst_rtsp_stream_seekable (stream2));
+
+ fail_unless (gst_rtsp_transport_free (transport) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_range_parse ("npt=3.0-5.0", &range) == GST_RTSP_OK);
+
+ /* the media is seekable now */
+ fail_unless (gst_rtsp_media_seek (media, range));
+
+ /* verify that we got the expected range, 'npt=3.0-5.0' */
+ range_str = gst_rtsp_media_get_range_string (media, TRUE, GST_RTSP_RANGE_NPT);
+ fail_unless (gst_rtsp_range_parse (range_str, &play_range) == GST_RTSP_OK);
+ fail_unless (play_range->min.seconds == range->min.seconds);
+ fail_unless (play_range->max.seconds == range->max.seconds);
+
+ gst_rtsp_range_free (range);
+ gst_rtsp_range_free (play_range);
+ g_free (range_str);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+/* case: media is complete and contains two streams but only one is active,
+ audio & video sources */
+GST_START_TEST (test_media_playback_seek_one_active_stream)
+{
+ media_playback_seek_one_active_stream
+ ("( videotestsrc ! rtpvrawpay pt=96 name=pay0 "
+ " audiotestsrc ! audioconvert ! rtpL16pay name=pay1 )");
+}
+
+GST_END_TEST;
+
+/* case: media is complete and contains two streams but only one is active,
+ demux */
+GST_START_TEST (test_media_playback_demux_seek_one_active_stream)
+{
+ /* FIXME: this test produces "Failed to push event" error messages in the
+ * GST_DEBUG logs because the incomplete stream has no sinks */
+ media_playback_seek_one_active_stream ("( filesrc location="
+ GST_TEST_FILES_PATH "/test.avi !"
+ " avidemux name=demux demux.audio_0 ! queue ! decodebin ! audioconvert !"
+ " audioresample ! rtpL16pay pt=97 name=pay1"
+ " demux.video_0 ! queue ! decodebin ! rtpvrawpay pt=96 name=pay0 )");
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_seek_no_sinks)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream;
+ GstRTSPTimeRange *range;
+ gchar *str;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream != NULL);
+
+ /* fails, need to be prepared */
+ str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ fail_unless (str == NULL);
+
+ fail_unless (gst_rtsp_range_parse ("npt=5.0-", &range) == GST_RTSP_OK);
+ /* fails, need to be prepared */
+ fail_if (gst_rtsp_media_seek (media, range));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (media_has_sdp (media));
+
+ str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ fail_unless (g_str_equal (str, "npt=0-"));
+ g_free (str);
+
+ str = gst_rtsp_media_get_range_string (media, TRUE, GST_RTSP_RANGE_NPT);
+ fail_unless (g_str_equal (str, "npt=0-"));
+ g_free (str);
+
+ /* fails, need to be prepared and contain sink elements */
+ fail_if (gst_rtsp_media_seek (media, range));
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ /* should fail again */
+ str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ fail_unless (str == NULL);
+ fail_if (gst_rtsp_media_seek (media, range));
+
+ gst_rtsp_range_free (range);
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media)
+{
+ GstRTSPMedia *media;
+ GstElement *bin, *e1, *e2;
+
+ bin = gst_bin_new ("bin");
+ fail_if (bin == NULL);
+
+ e1 = gst_element_factory_make ("videotestsrc", NULL);
+ fail_if (e1 == NULL);
+
+ e2 = gst_element_factory_make ("rtpvrawpay", "pay0");
+ fail_if (e2 == NULL);
+ g_object_set (e2, "pt", 96, NULL);
+
+ gst_bin_add_many (GST_BIN_CAST (bin), e1, e2, NULL);
+ gst_element_link_many (e1, e2, NULL);
+
+ media = gst_rtsp_media_new (bin);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ g_object_unref (media);
+}
+
+GST_END_TEST;
+
+static void
+test_prepare_reusable (const gchar * launch_line, gboolean is_live)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPThread *thread;
+ GstRTSPThreadPool *pool;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory, launch_line);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ g_object_set (G_OBJECT (media), "reusable", TRUE, NULL);
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (media_has_sdp (media));
+ if (is_live) { /* Live is not seekable */
+ fail_unless_equals_int64 (gst_rtsp_media_seekable (media), -1);
+ } else {
+ fail_unless_equals_int64 (gst_rtsp_media_seekable (media), G_MAXINT64);
+ }
+ fail_unless (gst_rtsp_media_unprepare (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (media_has_sdp (media));
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ g_object_unref (media);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+
+ g_object_unref (pool);
+}
+
+GST_START_TEST (test_media_reusable)
+{
+
+ /* test reusable media */
+ test_prepare_reusable ("( videotestsrc ! rtpvrawpay pt=96 name=pay0 )",
+ FALSE);
+ test_prepare_reusable
+ ("( videotestsrc is-live=true ! rtpvrawpay pt=96 name=pay0 )", TRUE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_prepare)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ pool = gst_rtsp_thread_pool_new ();
+
+ /* test non-reusable media first */
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (media_has_sdp (media));
+ fail_unless_equals_int64 (gst_rtsp_media_seekable (media), G_MAXINT64);
+ fail_unless (gst_rtsp_media_unprepare (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_if (gst_rtsp_media_prepare (media, thread));
+
+ g_object_unref (media);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+
+ g_object_unref (pool);
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+enum _SyncState
+{
+ SYNC_STATE_INIT,
+ SYNC_STATE_1,
+ SYNC_STATE_2,
+ SYNC_STATE_RACE
+};
+typedef enum _SyncState SyncState;
+
+struct _help_thread_data
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPMedia *media;
+ GstRTSPTransport *transport;
+ GstRTSPStream *stream;
+ SyncState *state;
+ GMutex *sync_mutex;
+ GCond *sync_cond;
+};
+typedef struct _help_thread_data help_thread_data;
+
+static gpointer
+help_thread_main (gpointer user_data)
+{
+ help_thread_data *data;
+ GstRTSPThread *thread;
+ GPtrArray *transports;
+ GstRTSPStreamTransport *stream_transport;
+
+ data = (help_thread_data *) user_data;
+ GST_INFO ("Another thread sharing media");
+
+ /* wait SYNC_STATE_1 */
+ g_mutex_lock (data->sync_mutex);
+ while (*data->state < SYNC_STATE_1)
+ g_cond_wait (data->sync_cond, data->sync_mutex);
+ g_mutex_unlock (data->sync_mutex);
+
+ /* prepare */
+ thread = gst_rtsp_thread_pool_get_thread (data->pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (data->media, thread));
+
+ /* set SYNC_STATE_2 */
+ g_mutex_lock (data->sync_mutex);
+ *data->state = SYNC_STATE_2;
+ g_cond_signal (data->sync_cond);
+ g_mutex_unlock (data->sync_mutex);
+
+ /* wait SYNC_STATE_RACE */
+ g_mutex_lock (data->sync_mutex);
+ while (*data->state < SYNC_STATE_RACE)
+ g_cond_wait (data->sync_cond, data->sync_mutex);
+ g_mutex_unlock (data->sync_mutex);
+
+ /* set state */
+ transports = g_ptr_array_new_with_free_func (g_object_unref);
+ fail_unless (transports != NULL);
+ stream_transport =
+ gst_rtsp_stream_transport_new (data->stream, data->transport);
+ fail_unless (stream_transport != NULL);
+ g_ptr_array_add (transports, stream_transport);
+ fail_unless (gst_rtsp_media_set_state (data->media, GST_STATE_NULL,
+ transports));
+
+ /* clean up */
+ GST_INFO ("Thread exit");
+ fail_unless (gst_rtsp_media_unprepare (data->media));
+ g_ptr_array_unref (transports);
+ return NULL;
+}
+
+GST_START_TEST (test_media_shared_race_test_unsuspend_vs_set_state_null)
+{
+ help_thread_data data;
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GThread *sharing_media_thread;
+ GstRTSPTransport *transport;
+ GstRTSPStream *stream;
+ SyncState state = SYNC_STATE_INIT;
+ GMutex sync_mutex;
+ GCond sync_cond;
+
+ g_mutex_init (&sync_mutex);
+ g_cond_init (&sync_cond);
+
+ pool = gst_rtsp_thread_pool_new ();
+
+ /* test non-reusable media first */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+ gst_rtsp_media_set_suspend_mode (media, GST_RTSP_SUSPEND_MODE_RESET);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream != NULL);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ /* help thread */
+ data.pool = pool;
+ data.media = media;
+ data.stream = stream;
+ data.state = &state;
+ data.sync_mutex = &sync_mutex;
+ data.sync_cond = &sync_cond;
+ sharing_media_thread = g_thread_new ("new thread", help_thread_main, &data);
+ fail_unless (sharing_media_thread != NULL);
+
+ /* set state SYNC_STATE_1 */
+ g_mutex_lock (&sync_mutex);
+ state = SYNC_STATE_1;
+ g_cond_signal (&sync_cond);
+ g_mutex_unlock (&sync_mutex);
+
+ /* wait SYNC_STATE_2 */
+ g_mutex_lock (&sync_mutex);
+ while (state < SYNC_STATE_2)
+ g_cond_wait (&sync_cond, &sync_mutex);
+ g_mutex_unlock (&sync_mutex);
+
+ gst_rtsp_media_suspend (media);
+
+ fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
+ transport->lower_transport = GST_RTSP_LOWER_TRANS_TCP;
+ fail_unless (gst_rtsp_stream_complete_stream (stream, transport));
+ data.transport = transport;
+
+ /* set state SYNC_STATE_RACE let the race begin unsuspend <-> set state GST_STATE_NULL */
+ g_mutex_lock (&sync_mutex);
+ state = SYNC_STATE_RACE;
+ g_cond_signal (&sync_cond);
+ g_mutex_unlock (&sync_mutex);
+
+ fail_unless (gst_rtsp_media_unsuspend (media));
+
+ /* sync end of other thread */
+ g_thread_join (sharing_media_thread);
+
+ /* clean up */
+ g_cond_clear (&sync_cond);
+ g_mutex_clear (&sync_mutex);
+ fail_unless (gst_rtsp_media_unprepare (media));
+ g_object_unref (media);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+ g_object_unref (pool);
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+
+#define FLAG_HAVE_CAPS GST_ELEMENT_FLAG_LAST
+static void
+on_notify_caps (GstPad * pad, GParamSpec * pspec, GstElement * pay)
+{
+ GstCaps *caps;
+
+ g_object_get (pad, "caps", &caps, NULL);
+
+ GST_DEBUG ("notify %" GST_PTR_FORMAT, caps);
+
+ if (caps) {
+ if (!GST_OBJECT_FLAG_IS_SET (pay, FLAG_HAVE_CAPS)) {
+ g_signal_emit_by_name (pay, "pad-added", pad);
+ g_signal_emit_by_name (pay, "no-more-pads", NULL);
+ GST_OBJECT_FLAG_SET (pay, FLAG_HAVE_CAPS);
+ }
+ gst_caps_unref (caps);
+ } else {
+ if (GST_OBJECT_FLAG_IS_SET (pay, FLAG_HAVE_CAPS)) {
+ g_signal_emit_by_name (pay, "pad-removed", pad);
+ GST_OBJECT_FLAG_UNSET (pay, FLAG_HAVE_CAPS);
+ }
+ }
+}
+
+GST_START_TEST (test_media_dyn_prepare)
+{
+ GstRTSPMedia *media;
+ GstElement *bin, *src, *pay;
+ GstElement *pipeline;
+ GstPad *srcpad;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ bin = gst_bin_new ("bin");
+ fail_if (bin == NULL);
+
+ src = gst_element_factory_make ("videotestsrc", NULL);
+ fail_if (src == NULL);
+
+ pay = gst_element_factory_make ("rtpvrawpay", "dynpay0");
+ fail_if (pay == NULL);
+ g_object_set (pay, "pt", 96, NULL);
+
+ gst_bin_add_many (GST_BIN_CAST (bin), src, pay, NULL);
+ gst_element_link_many (src, pay, NULL);
+
+ media = gst_rtsp_media_new (bin);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ g_object_set (G_OBJECT (media), "reusable", TRUE, NULL);
+
+ pipeline = gst_pipeline_new ("media-pipeline");
+ gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+ gst_rtsp_media_collect_streams (media);
+
+ srcpad = gst_element_get_static_pad (pay, "src");
+
+ g_signal_connect (srcpad, "notify::caps", (GCallback) on_notify_caps, pay);
+
+ pool = gst_rtsp_thread_pool_new ();
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+ fail_unless (media_has_sdp (media));
+ fail_unless_equals_int64 (gst_rtsp_media_seekable (media), G_MAXINT64);
+ fail_unless (gst_rtsp_media_unprepare (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+ fail_unless (media_has_sdp (media));
+ fail_unless_equals_int64 (gst_rtsp_media_seekable (media), G_MAXINT64);
+ fail_unless (gst_rtsp_media_unprepare (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+ gst_object_unref (srcpad);
+ g_object_unref (media);
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_take_pipeline)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstElement *pipeline;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+ gst_rtsp_media_factory_set_launch (factory,
+ "( fakesrc ! text/plain ! rtpgstpay name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ pipeline = gst_pipeline_new ("media-pipeline");
+ gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+ g_object_unref (media);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_reset)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ pool = gst_rtsp_thread_pool_new ();
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ gst_rtsp_url_parse ("rtsp://localhost:8554/test", &url);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (media_has_sdp (media));
+ fail_unless_equals_int64 (gst_rtsp_media_seekable (media), G_MAXINT64);
+ fail_unless (gst_rtsp_media_suspend (media));
+ fail_unless (gst_rtsp_media_unprepare (media));
+ g_object_unref (media);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ gst_rtsp_media_set_suspend_mode (media, GST_RTSP_SUSPEND_MODE_RESET);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (media_has_sdp (media));
+ fail_unless_equals_int64 (gst_rtsp_media_seekable (media), G_MAXINT64);
+ fail_unless (gst_rtsp_media_suspend (media));
+ fail_unless (gst_rtsp_media_unprepare (media));
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_multidyn_prepare)
+{
+ GstRTSPMedia *media;
+ GstElement *bin, *src0, *pay0, *src1, *pay1;
+ GstElement *pipeline;
+ GstPad *srcpad0, *srcpad1;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ bin = gst_bin_new ("bin");
+ fail_if (bin == NULL);
+
+ src0 = gst_element_factory_make ("videotestsrc", NULL);
+ fail_if (src0 == NULL);
+
+ pay0 = gst_element_factory_make ("rtpvrawpay", "dynpay0");
+ fail_if (pay0 == NULL);
+ g_object_set (pay0, "pt", 96, NULL);
+
+ src1 = gst_element_factory_make ("videotestsrc", NULL);
+ fail_if (src1 == NULL);
+
+ pay1 = gst_element_factory_make ("rtpvrawpay", "dynpay1");
+ fail_if (pay1 == NULL);
+ g_object_set (pay1, "pt", 97, NULL);
+
+ gst_bin_add_many (GST_BIN_CAST (bin), src0, pay0, src1, pay1, NULL);
+ gst_element_link_many (src0, pay0, NULL);
+ gst_element_link_many (src1, pay1, NULL);
+
+ media = gst_rtsp_media_new (bin);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ g_object_set (G_OBJECT (media), "reusable", TRUE, NULL);
+
+ pipeline = gst_pipeline_new ("media-pipeline");
+ gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+ gst_rtsp_media_collect_streams (media);
+
+ srcpad0 = gst_element_get_static_pad (pay0, "src");
+ srcpad1 = gst_element_get_static_pad (pay1, "src");
+
+ g_signal_connect (srcpad0, "notify::caps", (GCallback) on_notify_caps, pay0);
+ g_signal_connect (srcpad1, "notify::caps", (GCallback) on_notify_caps, pay1);
+
+ pool = gst_rtsp_thread_pool_new ();
+
+ fail_unless_equals_int (gst_rtsp_media_n_streams (media), 0);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless_equals_int (gst_rtsp_media_n_streams (media), 2);
+ fail_unless (media_has_sdp (media));
+ fail_unless_equals_int64 (gst_rtsp_media_seekable (media), G_MAXINT64);
+ fail_unless (gst_rtsp_media_unprepare (media));
+ fail_unless_equals_int (gst_rtsp_media_n_streams (media), 0);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless_equals_int (gst_rtsp_media_n_streams (media), 2);
+ fail_unless (media_has_sdp (media));
+ fail_unless_equals_int64 (gst_rtsp_media_seekable (media), G_MAXINT64);
+ fail_unless (gst_rtsp_media_unprepare (media));
+ fail_unless_equals_int (gst_rtsp_media_n_streams (media), 0);
+
+ gst_object_unref (srcpad0);
+ gst_object_unref (srcpad1);
+ g_object_unref (media);
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+
+static Suite *
+rtspmedia_suite (void)
+{
+ Suite *s = suite_create ("rtspmedia");
+ TCase *tc = tcase_create ("general");
+ gboolean has_avidemux;
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+
+ has_avidemux = gst_registry_check_feature_version (gst_registry_get (),
+ "avidemux", GST_VERSION_MAJOR, GST_VERSION_MINOR, 0);
+
+ tcase_add_test (tc, test_media_seek);
+ tcase_add_test (tc, test_media_seek_no_sinks);
+ tcase_add_test (tc, test_media_playback_seek_one_active_stream);
+ if (has_avidemux) {
+ tcase_add_test (tc, test_media_playback_demux_seek_one_active_stream);
+ } else {
+ GST_INFO ("Skipping test, missing plugins: avidemux");
+ }
+ tcase_add_test (tc, test_media);
+ tcase_add_test (tc, test_media_prepare);
+ tcase_add_test (tc, test_media_shared_race_test_unsuspend_vs_set_state_null);
+ tcase_add_test (tc, test_media_reusable);
+ tcase_add_test (tc, test_media_dyn_prepare);
+ tcase_add_test (tc, test_media_take_pipeline);
+ tcase_add_test (tc, test_media_reset);
+ tcase_add_test (tc, test_media_multidyn_prepare);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspmedia);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-media-factory.h>
+
+GST_START_TEST (test_parse_error)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+
+ gst_rtsp_media_factory_set_launch (factory, "foo");
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+ ASSERT_CRITICAL (gst_rtsp_media_factory_create_element (factory, url));
+ ASSERT_CRITICAL (gst_rtsp_media_factory_construct (factory, url));
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_launch)
+{
+ GstRTSPMediaFactory *factory;
+ GstElement *element;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ fail_unless (GST_IS_BIN (element));
+ fail_if (GST_IS_PIPELINE (element));
+ gst_object_unref (element);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_launch_construct)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media, *media2;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ media2 = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media2));
+ fail_if (media == media2);
+
+ g_object_unref (media);
+ g_object_unref (media2);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_shared)
+{
+ GstRTSPMediaFactory *factory;
+ GstElement *element;
+ GstRTSPMedia *media, *media2;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ fail_unless (gst_rtsp_media_factory_is_shared (factory));
+
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ fail_unless (GST_IS_BIN (element));
+ fail_if (GST_IS_PIPELINE (element));
+ gst_object_unref (element);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ media2 = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media2));
+ fail_unless (media == media2);
+
+ g_object_unref (media);
+ g_object_unref (media2);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_addresspool)
+{
+ GstRTSPMediaFactory *factory;
+ GstElement *element;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPAddressPool *pool, *tmppool;
+ GstRTSPStream *stream;
+ GstRTSPAddress *addr;
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 "
+ " audiotestsrc ! audioconvert ! rtpL16pay name=pay1 )");
+
+ pool = gst_rtsp_address_pool_new ();
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "233.252.0.1", 5000, 5001, 3));
+
+ gst_rtsp_media_factory_set_address_pool (factory, pool);
+
+ tmppool = gst_rtsp_media_factory_get_address_pool (factory);
+ fail_unless (pool == tmppool);
+ g_object_unref (tmppool);
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ fail_unless (GST_IS_BIN (element));
+ fail_if (GST_IS_PIPELINE (element));
+ gst_object_unref (element);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ tmppool = gst_rtsp_media_get_address_pool (media);
+ fail_unless (pool == tmppool);
+ g_object_unref (tmppool);
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream != NULL);
+
+ tmppool = gst_rtsp_stream_get_address_pool (stream);
+ fail_unless (pool == tmppool);
+ g_object_unref (tmppool);
+
+ addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5000);
+ fail_unless (addr->n_ports == 2);
+ fail_unless (addr->ttl == 3);
+ gst_rtsp_address_free (addr);
+
+ stream = gst_rtsp_media_get_stream (media, 1);
+ fail_unless (stream != NULL);
+
+ tmppool = gst_rtsp_stream_get_address_pool (stream);
+ fail_unless (pool == tmppool);
+ g_object_unref (tmppool);
+
+ addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+ fail_unless (addr == NULL);
+
+
+ g_object_unref (media);
+
+ g_object_unref (pool);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_permissions)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPPermissions *perms;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ gst_rtsp_media_factory_add_role (factory, "admin",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ perms = gst_rtsp_media_factory_get_permissions (factory);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.access"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.construct"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
+ "media.factory.access"));
+ gst_rtsp_permissions_unref (perms);
+
+ perms = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (perms, "user",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, FALSE, NULL);
+ gst_rtsp_media_factory_set_permissions (factory, perms);
+ gst_rtsp_permissions_unref (perms);
+
+ perms = gst_rtsp_media_factory_get_permissions (factory);
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.access"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.construct"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user",
+ "media.factory.access"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user",
+ "media.factory.construct"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
+ "media.factory.access"));
+ gst_rtsp_permissions_unref (perms);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ perms = gst_rtsp_media_get_permissions (media);
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.access"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.construct"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user",
+ "media.factory.access"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user",
+ "media.factory.construct"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
+ "media.factory.access"));
+ gst_rtsp_permissions_unref (perms);
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_reset)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ gst_rtsp_url_parse ("rtsp://localhost:8554/test", &url);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ fail_if (gst_rtsp_media_get_suspend_mode (media) !=
+ GST_RTSP_SUSPEND_MODE_NONE);
+ g_object_unref (media);
+
+ gst_rtsp_media_factory_set_suspend_mode (factory,
+ GST_RTSP_SUSPEND_MODE_RESET);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ fail_if (gst_rtsp_media_get_suspend_mode (media) !=
+ GST_RTSP_SUSPEND_MODE_RESET);
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_mcast_ttl)
+{
+ GstRTSPMediaFactory *factory;
+ GstElement *element;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream;
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 "
+ " audiotestsrc ! audioconvert ! rtpL16pay name=pay1 )");
+
+ /* try to set an invalid ttl and make sure that the default ttl value (255) is
+ * set */
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory, 0);
+ fail_unless (gst_rtsp_media_factory_get_max_mcast_ttl (factory) == 255);
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory, 300);
+ fail_unless (gst_rtsp_media_factory_get_max_mcast_ttl (factory) == 255);
+
+ /* set a valid ttl value */
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory, 3);
+ fail_unless (gst_rtsp_media_factory_get_max_mcast_ttl (factory) == 3);
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ fail_unless (GST_IS_BIN (element));
+ fail_if (GST_IS_PIPELINE (element));
+ gst_object_unref (element);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 2);
+ fail_unless (gst_rtsp_media_get_max_mcast_ttl (media) == 3);
+
+ /* verify that the correct ttl value has been propageted to the media
+ * streams */
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream != NULL);
+ fail_unless (gst_rtsp_stream_get_max_mcast_ttl (stream) == 3);
+
+ stream = gst_rtsp_media_get_stream (media, 1);
+ fail_unless (stream != NULL);
+ fail_unless (gst_rtsp_stream_get_max_mcast_ttl (stream) == 3);
+
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_allow_bind_mcast)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream;
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 "
+ " audiotestsrc ! audioconvert ! rtpL16pay name=pay1 )");
+
+ /* verify that by default binding sockets to multicast addresses is not enabled */
+ fail_unless (gst_rtsp_media_factory_is_bind_mcast_address (factory) == FALSE);
+
+ /* allow multicast sockets to be bound to multicast addresses */
+ gst_rtsp_media_factory_set_bind_mcast_address (factory, TRUE);
+ /* verify that the socket binding to multicast address has been enabled */
+ fail_unless (gst_rtsp_media_factory_is_bind_mcast_address (factory) == TRUE);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ /* verify that the correct socket binding configuration has been propageted to the media */
+ fail_unless (gst_rtsp_media_is_bind_mcast_address (media) == TRUE);
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+ /* verify that the correct socket binding configuration has been propageted to the media streams */
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream != NULL);
+ fail_unless (gst_rtsp_stream_is_bind_mcast_address (stream) == TRUE);
+
+ stream = gst_rtsp_media_get_stream (media, 1);
+ fail_unless (stream != NULL);
+ fail_unless (gst_rtsp_stream_is_bind_mcast_address (stream) == TRUE);
+
+ g_object_unref (media);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspmediafactory_suite (void)
+{
+ Suite *s = suite_create ("rtspmediafactory");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_parse_error);
+ tcase_add_test (tc, test_launch);
+ tcase_add_test (tc, test_launch_construct);
+ tcase_add_test (tc, test_shared);
+ tcase_add_test (tc, test_addresspool);
+ tcase_add_test (tc, test_permissions);
+ tcase_add_test (tc, test_reset);
+ tcase_add_test (tc, test_mcast_ttl);
+ tcase_add_test (tc, test_allow_bind_mcast);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspmediafactory);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-mount-points.h>
+
+GST_START_TEST (test_create)
+{
+ GstRTSPMountPoints *mounts;
+ GstRTSPUrl *url, *url2;
+ GstRTSPMediaFactory *factory;
+
+ mounts = gst_rtsp_mount_points_new ();
+
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test2",
+ &url2) == GST_RTSP_OK);
+
+ fail_unless (gst_rtsp_mount_points_match (mounts, url->abspath,
+ NULL) == NULL);
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ fail_unless (gst_rtsp_mount_points_match (mounts, url->abspath,
+ NULL) == factory);
+ g_object_unref (factory);
+ fail_unless (gst_rtsp_mount_points_match (mounts, url2->abspath,
+ NULL) == NULL);
+
+ gst_rtsp_mount_points_remove_factory (mounts, "/test");
+
+ fail_unless (gst_rtsp_mount_points_match (mounts, url->abspath,
+ NULL) == NULL);
+ fail_unless (gst_rtsp_mount_points_match (mounts, url2->abspath,
+ NULL) == NULL);
+
+ gst_rtsp_url_free (url);
+ gst_rtsp_url_free (url2);
+
+ g_object_unref (mounts);
+}
+
+GST_END_TEST;
+
+static const gchar *paths[] = {
+ "/test",
+ "/booz/fooz",
+ "/booz/foo/zoop",
+ "/tark/bar",
+ "/tark/bar/baz",
+ "/tark/bar/baz/t",
+ "/boozop",
+ "/raw",
+ "/raw/video",
+ "/raw/snapshot",
+};
+
+GST_START_TEST (test_match)
+{
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *f[G_N_ELEMENTS (paths)], *tmp;
+ gint i, matched;
+
+ mounts = gst_rtsp_mount_points_new ();
+
+ for (i = 0; i < G_N_ELEMENTS (paths); i++) {
+ f[i] = gst_rtsp_media_factory_new ();
+ gst_rtsp_mount_points_add_factory (mounts, paths[i], f[i]);
+ }
+
+ tmp = gst_rtsp_mount_points_match (mounts, "/test", &matched);
+ fail_unless (tmp == f[0]);
+ fail_unless (matched == 5);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/test/stream=1", &matched);
+ fail_unless (tmp == f[0]);
+ fail_unless (matched == 5);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/booz", &matched);
+ fail_unless (tmp == NULL);
+ tmp = gst_rtsp_mount_points_match (mounts, "/booz/foo", &matched);
+ fail_unless (tmp == NULL);
+ tmp = gst_rtsp_mount_points_match (mounts, "/booz/fooz", &matched);
+ fail_unless (tmp == f[1]);
+ fail_unless (matched == 10);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/booz/fooz/zoo", &matched);
+ fail_unless (tmp == f[1]);
+ fail_unless (matched == 10);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/booz/foo/zoop", &matched);
+ fail_unless (tmp == f[2]);
+ fail_unless (matched == 14);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar", &matched);
+ fail_unless (tmp == f[3]);
+ fail_unless (matched == 9);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar/boo", &matched);
+ fail_unless (tmp == f[3]);
+ fail_unless (matched == 9);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar/ba", &matched);
+ fail_unless (tmp == f[3]);
+ fail_unless (matched == 9);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar/baz", &matched);
+ fail_unless (tmp == f[4]);
+ fail_unless (matched == 13);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/raw/video", &matched);
+ fail_unless (tmp == f[8]);
+ fail_unless (matched == 10);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/raw/snapshot", &matched);
+ fail_unless (tmp == f[9]);
+ fail_unless (matched == 13);
+ g_object_unref (tmp);
+
+ g_object_unref (mounts);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspmountpoints_suite (void)
+{
+ Suite *s = suite_create ("rtspmountpoints");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_create);
+ tcase_add_test (tc, test_match);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspmountpoints);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2018 Mathieu Duponchelle <mathieu@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/sdp/gstsdpmessage.h>
+#include <gst/rtsp/gstrtspmessage.h>
+#include <gst/base/gstpushsrc.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+#include <rtsp-onvif-client.h>
+#include <rtsp-onvif-media.h>
+#include <rtsp-onvif-media-factory.h>
+
+/* Test source implementation */
+
+#define FRAME_DURATION (GST_MSECOND)
+
+typedef struct
+{
+ GstPushSrc element;
+
+ GstSegment *segment;
+ /* In milliseconds */
+ guint trickmode_interval;
+ GstClockTime ntp_offset;
+} TestSrc;
+
+typedef struct
+{
+ GstPushSrcClass parent_class;
+} TestSrcClass;
+
+/**
+ * video/x-dumdum is a very simple encoded video format:
+ *
+ * - It has I-frames, P-frames and B-frames for the purpose
+ * of testing trick modes, and is infinitely scalable, mimicking server-side
+ * trick modes that would have the server reencode when a trick mode seek with
+ * an absolute rate different from 1.0 is requested.
+ *
+ * - The only source capable of outputting this format, `TestSrc`, happens
+ * to always output frames following this pattern:
+ *
+ * IBBBBPBBBBI
+ *
+ * Its framerate is 1000 / 1, each Group of Pictures is thus 10 milliseconds
+ * long. The first frame in the stream dates back to January the first,
+ * 1900, at exactly midnight. There are no gaps in the stream.
+ *
+ * A nice side effect of this for testing purposes is that as the resolution
+ * of UTC (clock=) seeks is a hundredth of a second, this coincides with the
+ * alignment of our Group of Pictures, which means we don't have to worry
+ * about synchronization points.
+ *
+ * - Size is used to distinguish the various frame types:
+ *
+ * * I frames: 20 bytes
+ * * P frames: 10 bytes
+ * * B frames: 5 bytes
+ *
+ */
+
+#define TEST_CAPS "video/x-dumdum"
+
+typedef enum
+{
+ FRAME_TYPE_I,
+ FRAME_TYPE_P,
+ FRAME_TYPE_B,
+} FrameType;
+
+static FrameType
+frame_type_for_index (guint64 index)
+{
+ FrameType ret;
+
+ if (index % 10 == 0)
+ ret = FRAME_TYPE_I;
+ else if (index % 5 == 0)
+ ret = FRAME_TYPE_P;
+ else
+ ret = FRAME_TYPE_B;
+
+ return ret;
+}
+
+static GstStaticPadTemplate test_src_template = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (TEST_CAPS)
+ );
+
+GType test_src_get_type (void);
+
+#define test_src_parent_class parent_class
+G_DEFINE_TYPE (TestSrc, test_src, GST_TYPE_PUSH_SRC);
+
+#define ROUND_UP_TO_10(x) (((x + 10 - 1) / 10) * 10)
+#define ROUND_DOWN_TO_10(x) (x - (x % 10))
+
+/*
+ * For now, the theoretical range of our test source is infinite.
+ *
+ * When creating a buffer, we use the current segment position to
+ * determine the PTS, and simply increment it afterwards.
+ *
+ * When the stop time of a buffer we have created reaches segment->stop,
+ * GstBaseSrc will take care of sending an EOS for us, which rtponviftimestamp
+ * will translate to setting the T flag in the RTP header extension.
+ */
+static GstFlowReturn
+test_src_create (GstPushSrc * psrc, GstBuffer ** buffer)
+{
+ GstFlowReturn ret = GST_FLOW_OK;
+ gsize buf_size;
+ TestSrc *src = (TestSrc *) psrc;
+ GstClockTime pts, duration;
+ FrameType ftype;
+ guint64 n_frames;
+
+ if (src->segment->rate < 1.0) {
+ if (src->segment->position < src->segment->start) {
+ ret = GST_FLOW_EOS;
+ goto done;
+ }
+ } else if ((src->segment->position >= src->segment->stop)) {
+ ret = GST_FLOW_EOS;
+ goto done;
+ }
+
+ pts = src->segment->position;
+ duration = FRAME_DURATION;
+
+ if ((src->segment->flags & GST_SEGMENT_FLAG_TRICKMODE_KEY_UNITS)) {
+ duration =
+ MAX (duration * 10,
+ duration * ROUND_UP_TO_10 (src->trickmode_interval));
+ } else if ((src->segment->
+ flags & GST_SEGMENT_FLAG_TRICKMODE_FORWARD_PREDICTED)) {
+ duration *= 5;
+ }
+
+ n_frames = gst_util_uint64_scale (src->segment->position, 1000, GST_SECOND);
+
+ ftype = frame_type_for_index (n_frames);
+
+ switch (ftype) {
+ case FRAME_TYPE_I:
+ buf_size = 20;
+ break;
+ case FRAME_TYPE_P:
+ buf_size = 10;
+ break;
+ case FRAME_TYPE_B:
+ buf_size = 5;
+ break;
+ }
+
+ *buffer = gst_buffer_new_allocate (NULL, buf_size, NULL);
+
+ if (ftype != FRAME_TYPE_I) {
+ GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DELTA_UNIT);
+ }
+
+ GST_BUFFER_PTS (*buffer) = pts;
+ GST_BUFFER_DURATION (*buffer) = duration;
+
+ src->segment->position = pts + duration;
+
+ if (!GST_CLOCK_TIME_IS_VALID (src->ntp_offset)) {
+ GstClock *clock = gst_system_clock_obtain ();
+ GstClockTime clock_time = gst_clock_get_time (clock);
+ guint64 real_time = g_get_real_time ();
+ GstStructure *s;
+ GstEvent *onvif_event;
+
+ real_time *= 1000;
+ real_time += (G_GUINT64_CONSTANT (2208988800) * GST_SECOND);
+ src->ntp_offset = real_time - clock_time;
+
+ s = gst_structure_new ("GstNtpOffset",
+ "ntp-offset", G_TYPE_UINT64, src->ntp_offset,
+ "discont", G_TYPE_BOOLEAN, FALSE, NULL);
+
+ onvif_event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM, s);
+
+ gst_element_send_event (GST_ELEMENT (src), onvif_event);
+ }
+
+ if (src->segment->rate < 1.0) {
+ guint64 next_n_frames =
+ gst_util_uint64_scale (src->segment->position, 1000, GST_SECOND);
+
+ if (src->segment->position > src->segment->stop
+ || next_n_frames / 10 > n_frames / 10) {
+ GstStructure *s;
+ GstEvent *onvif_event;
+ guint n_gops;
+
+ n_gops = MAX (1, ((int) src->trickmode_interval / 10));
+
+ next_n_frames = (n_frames / 10 - n_gops) * 10;
+
+ src->segment->position = next_n_frames * GST_MSECOND;
+ s = gst_structure_new ("GstNtpOffset",
+ "ntp-offset", G_TYPE_UINT64, src->ntp_offset,
+ "discont", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ onvif_event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM, s);
+
+ gst_element_send_event (GST_ELEMENT (src), onvif_event);
+ }
+ }
+
+done:
+ return ret;
+}
+
+static void
+test_src_init (TestSrc * src)
+{
+ gst_base_src_set_format (GST_BASE_SRC (src), GST_FORMAT_TIME);
+ gst_base_src_set_automatic_eos (GST_BASE_SRC (src), FALSE);
+ src->segment = NULL;
+ src->ntp_offset = GST_CLOCK_TIME_NONE;
+}
+
+/*
+ * We support seeking, both this method and GstBaseSrc.do_seek must
+ * be implemented for GstBaseSrc to report TRUE in the seeking query.
+ */
+static gboolean
+test_src_is_seekable (GstBaseSrc * bsrc)
+{
+ return TRUE;
+}
+
+/* Extremely simple seek handling for now, we simply update our
+ * segment, which will cause test_src_create to timestamp output
+ * buffers as expected.
+ */
+static gboolean
+test_src_do_seek (GstBaseSrc * bsrc, GstSegment * segment)
+{
+ TestSrc *src = (TestSrc *) bsrc;
+
+ if ((segment->flags & GST_SEGMENT_FLAG_TRICKMODE
+ && ABS (segment->rate) != 1.0)) {
+ segment->applied_rate = segment->rate;
+ segment->stop =
+ segment->start + ((segment->stop -
+ segment->start) / ABS (segment->rate));
+ segment->rate = segment->rate > 0 ? 1.0 : -1.0;
+ }
+
+ if (src->segment)
+ gst_segment_free (src->segment);
+
+ src->segment = gst_segment_copy (segment);
+
+ if (src->segment->rate < 0) {
+ guint64 n_frames =
+ ROUND_DOWN_TO_10 (gst_util_uint64_scale (src->segment->stop, 1000,
+ GST_SECOND));
+
+ src->segment->position = n_frames * GST_MSECOND;
+ }
+
+ return TRUE;
+}
+
+static gboolean
+test_src_event (GstBaseSrc * bsrc, GstEvent * event)
+{
+ TestSrc *src = (TestSrc *) bsrc;
+
+ if (GST_EVENT_TYPE (event) == GST_EVENT_SEEK) {
+ GstClockTime interval;
+
+ gst_event_parse_seek_trickmode_interval (event, &interval);
+
+ src->trickmode_interval = interval / 1000000;
+ }
+
+ return GST_BASE_SRC_CLASS (parent_class)->event (bsrc, event);
+}
+
+static void
+test_src_class_init (TestSrcClass * klass)
+{
+ gst_element_class_add_static_pad_template (GST_ELEMENT_CLASS (klass),
+ &test_src_template);
+ GST_PUSH_SRC_CLASS (klass)->create = test_src_create;
+ GST_BASE_SRC_CLASS (klass)->is_seekable = test_src_is_seekable;
+ GST_BASE_SRC_CLASS (klass)->do_seek = test_src_do_seek;
+ GST_BASE_SRC_CLASS (klass)->event = test_src_event;
+}
+
+static GstElement *
+test_src_new (void)
+{
+ return g_object_new (test_src_get_type (), NULL);
+}
+
+/* Test media factory */
+
+typedef struct
+{
+ GstRTSPMediaFactory factory;
+} TestMediaFactory;
+
+typedef struct
+{
+ GstRTSPMediaFactoryClass parent_class;
+} TestMediaFactoryClass;
+
+GType test_media_factory_get_type (void);
+
+G_DEFINE_TYPE (TestMediaFactory, test_media_factory,
+ GST_TYPE_RTSP_MEDIA_FACTORY);
+
+#define MAKE_AND_ADD(var, pipe, name, label, elem_name) \
+G_STMT_START { \
+ if (G_UNLIKELY (!(var = (gst_element_factory_make (name, elem_name))))) { \
+ GST_ERROR ("Could not create element %s", name); \
+ goto label; \
+ } \
+ if (G_UNLIKELY (!gst_bin_add (GST_BIN_CAST (pipe), var))) { \
+ GST_ERROR ("Could not add element %s", name); \
+ goto label; \
+ } \
+} G_STMT_END
+
+static GstElement *
+test_media_factory_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstElement *ret = gst_bin_new (NULL);
+ GstElement *pbin = gst_bin_new ("pay0");
+ GstElement *src, *pay, *onvifts, *queue;
+ GstPad *sinkpad, *srcpad;
+ GstPadLinkReturn link_ret;
+
+ src = test_src_new ();
+ gst_bin_add (GST_BIN (ret), src);
+ MAKE_AND_ADD (pay, pbin, "rtpgstpay", fail, NULL);
+ MAKE_AND_ADD (onvifts, pbin, "rtponviftimestamp", fail, NULL);
+ MAKE_AND_ADD (queue, pbin, "queue", fail, NULL);
+
+ gst_bin_add (GST_BIN (ret), pbin);
+ if (!gst_element_link_many (pay, onvifts, queue, NULL))
+ goto fail;
+
+ sinkpad = gst_element_get_static_pad (pay, "sink");
+ gst_element_add_pad (pbin, gst_ghost_pad_new ("sink", sinkpad));
+ gst_object_unref (sinkpad);
+
+ sinkpad = gst_element_get_static_pad (pbin, "sink");
+ srcpad = gst_element_get_static_pad (src, "src");
+ link_ret = gst_pad_link (srcpad, sinkpad);
+ gst_object_unref (srcpad);
+ gst_object_unref (sinkpad);
+
+ if (link_ret != GST_PAD_LINK_OK)
+ goto fail;
+
+ srcpad = gst_element_get_static_pad (queue, "src");
+ gst_element_add_pad (pbin, gst_ghost_pad_new ("src", srcpad));
+ gst_object_unref (srcpad);
+
+ g_object_set (pay, "timestamp-offset", 0, NULL);
+ g_object_set (onvifts, "set-t-bit", TRUE, NULL);
+
+done:
+ return ret;
+
+fail:
+ gst_object_unref (ret);
+ ret = NULL;
+ goto done;
+}
+
+static void
+test_media_factory_init (TestMediaFactory * factory)
+{
+}
+
+static void
+test_media_factory_class_init (TestMediaFactoryClass * klass)
+{
+ GST_RTSP_MEDIA_FACTORY_CLASS (klass)->create_element =
+ test_media_factory_create_element;
+}
+
+static GstRTSPMediaFactory *
+test_media_factory_new (void)
+{
+ GstRTSPMediaFactory *result;
+
+ result = g_object_new (test_media_factory_get_type (), NULL);
+
+ return result;
+}
+
+/* Actual tests implementation */
+
+static gchar *session_id;
+static gint cseq;
+static gboolean terminal_frame;
+static gboolean received_rtcp;
+
+static GstSDPMessage *
+sdp_from_message (GstRTSPMessage * msg)
+{
+ GstSDPMessage *sdp_message;
+ guint8 *body = NULL;
+ guint body_size;
+
+ fail_unless (gst_rtsp_message_get_body (msg, &body,
+ &body_size) == GST_RTSP_OK);
+ fail_unless (gst_sdp_message_new (&sdp_message) == GST_SDP_OK);
+ fail_unless (gst_sdp_message_parse_buffer (body, body_size,
+ sdp_message) == GST_SDP_OK);
+
+ return sdp_message;
+}
+
+static gboolean
+test_response_x_onvif_track (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstSDPMessage *sdp = sdp_from_message (response);
+ guint medias_len = gst_sdp_message_medias_len (sdp);
+ guint i;
+
+ fail_unless_equals_int (medias_len, 1);
+
+ for (i = 0; i < medias_len; i++) {
+ const GstSDPMedia *smedia = gst_sdp_message_get_media (sdp, i);
+ gchar *x_onvif_track = g_strdup_printf ("APPLICATION%03d", i);
+
+ fail_unless_equals_string (gst_sdp_media_get_attribute_val (smedia,
+ "x-onvif-track"), x_onvif_track);
+ }
+
+ gst_sdp_message_free (sdp);
+
+ return TRUE;
+}
+
+static gboolean
+test_setup_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *str;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPSession *session;
+ gchar **session_hdr_params;
+
+ fail_unless_equals_int (gst_rtsp_message_get_type (response),
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless_equals_int (code, GST_RTSP_STS_OK);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+ 0) == GST_RTSP_OK);
+ fail_unless (atoi (str) == cseq++);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION,
+ &str, 0) == GST_RTSP_OK);
+ session_hdr_params = g_strsplit (str, ";", -1);
+
+ /* session-id value */
+ fail_unless (session_hdr_params[0] != NULL);
+
+ session_pool = gst_rtsp_client_get_session_pool (client);
+ fail_unless (session_pool != NULL);
+
+ session = gst_rtsp_session_pool_find (session_pool, session_hdr_params[0]);
+ g_strfreev (session_hdr_params);
+
+ /* remember session id to be able to send teardown */
+ if (session_id)
+ g_free (session_id);
+ session_id = g_strdup (gst_rtsp_session_get_sessionid (session));
+ fail_unless (session_id != NULL);
+
+ fail_unless (session != NULL);
+ g_object_unref (session);
+
+ g_object_unref (session_pool);
+
+ return TRUE;
+}
+
+static gboolean
+test_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless_equals_int (gst_rtsp_message_get_type (response),
+ GST_RTSP_MESSAGE_RESPONSE);
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless_equals_int (code, GST_RTSP_STS_OK);
+
+ return TRUE;
+}
+
+typedef struct
+{
+ guint32 previous_ts;
+ gint32 expected_ts_interval;
+ gint32 expected_i_frame_ts_interval;
+ guint expected_n_buffers;
+ guint n_buffers;
+ guint expected_n_i_frames;
+ guint n_i_frames;
+ guint expected_n_p_frames;
+ guint n_p_frames;
+ guint expected_n_b_frames;
+ guint n_b_frames;
+ guint expected_n_clean_points;
+ guint n_clean_points;
+ gboolean timestamped_rtcp;
+} RTPCheckData;
+
+#define EXTENSION_ID 0xABAC
+#define EXTENSION_SIZE 3
+
+static gboolean
+test_play_response_200_and_check_data (GstRTSPClient * client,
+ GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ RTPCheckData *check = (RTPCheckData *) user_data;
+
+ /* We check data in the same send function because client->send_func cannot
+ * be changed from client->send_func
+ */
+ if (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_DATA) {
+ GstRTSPStreamTransport *trans;
+ guint8 channel = 42;
+
+ gst_rtsp_message_parse_data (response, &channel);
+ fail_unless (trans =
+ gst_rtsp_client_get_stream_transport (client, channel));
+
+ if (channel == 0) { /* RTP */
+ GstBuffer *buf;
+ GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+ guint8 *body = NULL;
+ guint body_size;
+ guint8 *data;
+ guint16 bits;
+ guint wordlen;
+ guint8 flags;
+ gint32 expected_interval;
+ gboolean is_custom_event = FALSE;
+
+ fail_unless (gst_rtsp_message_get_body (response, &body,
+ &body_size) == GST_RTSP_OK);
+
+ buf = gst_rtp_buffer_new_copy_data (body, body_size);
+
+ switch (body_size) {
+ case 115: /* Ignore our serialized custom events */
+ is_custom_event = TRUE;
+ break;
+ case 56:
+ expected_interval = check->expected_i_frame_ts_interval;
+ check->n_i_frames += 1;
+ break;
+ case 46:
+ expected_interval = check->expected_ts_interval;
+ check->n_p_frames += 1;
+ break;
+ case 41:
+ expected_interval = check->expected_ts_interval;
+ check->n_b_frames += 1;
+ break;
+ default:
+ fail ("Invalid body size %u", body_size);
+ }
+
+ if (!is_custom_event) {
+ fail_unless (gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp));
+
+ if (check->previous_ts) {
+ fail_unless_equals_int (gst_rtp_buffer_get_timestamp (&rtp) -
+ check->previous_ts, expected_interval);
+ }
+
+ check->previous_ts = gst_rtp_buffer_get_timestamp (&rtp);
+ check->n_buffers += 1;
+
+ fail_unless (gst_rtp_buffer_get_extension_data (&rtp, &bits,
+ (gpointer) & data, &wordlen));
+
+ fail_unless (bits == EXTENSION_ID && wordlen == EXTENSION_SIZE);
+
+ flags = GST_READ_UINT8 (data + 8);
+
+ gst_rtp_buffer_unmap (&rtp);
+
+ if (flags & (1 << 7)) {
+ check->n_clean_points += 1;
+ }
+
+ /* T flag is set, we are done */
+ if (flags & (1 << 4)) {
+ fail_unless_equals_int (check->expected_n_buffers, check->n_buffers);
+ fail_unless_equals_int (check->expected_n_i_frames,
+ check->n_i_frames);
+ fail_unless_equals_int (check->expected_n_p_frames,
+ check->n_p_frames);
+ fail_unless_equals_int (check->expected_n_b_frames,
+ check->n_b_frames);
+ fail_unless_equals_int (check->expected_n_clean_points,
+ check->n_clean_points);
+
+ terminal_frame = TRUE;
+
+ }
+ }
+
+ gst_buffer_unref (buf);
+ } else if (channel == 1) { /* RTCP */
+ GstBuffer *buf;
+ guint8 *body = NULL;
+ guint body_size;
+ GstRTCPPacket packet;
+ GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
+ guint32 ssrc, rtptime, packet_count, octet_count;
+ guint64 ntptime;
+
+ received_rtcp = TRUE;
+ fail_unless (gst_rtsp_message_get_body (response, &body,
+ &body_size) == GST_RTSP_OK);
+
+ buf = gst_rtp_buffer_new_copy_data (body, body_size);
+ gst_rtcp_buffer_map (buf, GST_MAP_READ, &rtcp);
+ gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
+
+ gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, &rtptime,
+ &packet_count, &octet_count);
+
+ if (check->timestamped_rtcp) {
+ fail_unless (rtptime != 0);
+ fail_unless (ntptime != 0);
+ } else {
+ fail_unless (rtptime == 0);
+ fail_unless (ntptime == 0);
+ }
+
+ gst_rtcp_buffer_unmap (&rtcp);
+ gst_buffer_unref (buf);
+ }
+
+ gst_rtsp_stream_transport_message_sent (trans);
+
+ if (terminal_frame && received_rtcp) {
+ g_mutex_lock (&check_mutex);
+ g_cond_broadcast (&check_cond);
+ g_mutex_unlock (&check_mutex);
+ }
+
+ return TRUE;
+ }
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OK);
+
+ return TRUE;
+}
+
+static gboolean
+test_teardown_response_200 (GstRTSPClient * client,
+ GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ /* We might still be seeing stray RTCP messages */
+ if (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_DATA)
+ return TRUE;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OK);
+ fail_unless (g_str_equal (reason, "OK"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static void
+send_teardown (GstRTSPClient * client)
+{
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ fail_unless (session_id != NULL);
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client, test_teardown_response_200,
+ NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ g_free (session_id);
+ session_id = NULL;
+}
+
+static GstRTSPClient *
+setup_client (const gchar * launch_line)
+{
+ GstRTSPClient *client;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPMountPoints *mount_points;
+ GstRTSPMediaFactory *factory;
+ GstRTSPThreadPool *thread_pool;
+
+ client = gst_rtsp_onvif_client_new ();
+
+ session_pool = gst_rtsp_session_pool_new ();
+ gst_rtsp_client_set_session_pool (client, session_pool);
+
+ mount_points = gst_rtsp_mount_points_new ();
+ factory = test_media_factory_new ();
+
+ gst_rtsp_media_factory_set_media_gtype (factory, GST_TYPE_RTSP_ONVIF_MEDIA);
+
+ gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
+ gst_rtsp_client_set_mount_points (client, mount_points);
+
+ thread_pool = gst_rtsp_thread_pool_new ();
+ gst_rtsp_client_set_thread_pool (client, thread_pool);
+
+ g_object_unref (mount_points);
+ g_object_unref (session_pool);
+ g_object_unref (thread_pool);
+
+ return client;
+}
+
+static void
+teardown_client (GstRTSPClient * client)
+{
+ gst_rtsp_client_set_thread_pool (client, NULL);
+ g_object_unref (client);
+}
+
+/**
+ * https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
+ * 6.2 RTSP describe
+ */
+GST_START_TEST (test_x_onvif_track)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_client (NULL);
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_x_onvif_track, NULL,
+ NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+static void
+create_connection (GstRTSPConnection ** conn)
+{
+ GSocket *sock;
+ GError *error = NULL;
+
+ sock = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_STREAM,
+ G_SOCKET_PROTOCOL_TCP, &error);
+ g_assert_no_error (error);
+ fail_unless (gst_rtsp_connection_create_from_socket (sock, "127.0.0.1", 444,
+ NULL, conn) == GST_RTSP_OK);
+ g_object_unref (sock);
+}
+
+static void
+test_seek (const gchar * range, const gchar * speed, const gchar * scale,
+ const gchar * frames, const gchar * rate_control, RTPCheckData * rtp_check)
+{
+ GstRTSPClient *client;
+ GstRTSPConnection *conn;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_client (NULL);
+ create_connection (&conn);
+ fail_unless (gst_rtsp_client_set_connection (client, conn));
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP/TCP;unicast");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, range);
+
+ if (scale) {
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, scale);
+ }
+
+ if (speed) {
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, speed);
+ }
+
+ if (frames) {
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_FRAMES, frames);
+ }
+
+ if (rate_control) {
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RATE_CONTROL,
+ rate_control);
+ }
+
+ gst_rtsp_client_set_send_func (client, test_play_response_200_and_check_data,
+ rtp_check, NULL);
+
+ terminal_frame = FALSE;
+ received_rtcp = FALSE;
+
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ g_mutex_lock (&check_mutex);
+ while (!terminal_frame || !received_rtcp)
+ g_cond_wait (&check_cond, &check_mutex);
+ g_mutex_unlock (&check_mutex);
+
+ send_teardown (client);
+
+ teardown_client (client);
+}
+
+GST_START_TEST (test_src_seek_simple)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 90;
+ rtp_check.expected_i_frame_ts_interval = 90;
+ rtp_check.expected_n_buffers = 100;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 10;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 80;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010000.00Z-19000101T010000.10Z", NULL, NULL, NULL,
+ NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+/**
+ * https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
+ * 6.4 RTSP Feature Tag
+ */
+GST_START_TEST (test_onvif_replay)
+{
+ GstRTSPClient *client;
+ GstRTSPConnection *conn;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_client (NULL);
+ create_connection (&conn);
+ fail_unless (gst_rtsp_client_set_connection (client, conn));
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP/TCP;unicast");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, "onvif-replay");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ send_teardown (client);
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_speed_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 45;
+ rtp_check.expected_i_frame_ts_interval = 45;
+ rtp_check.expected_n_buffers = 100;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 10;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 80;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010000.00Z-19000101T010000.10Z", "2.0", NULL, NULL,
+ NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_scale_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 90;
+ rtp_check.expected_i_frame_ts_interval = 90;
+ rtp_check.expected_n_buffers = 50;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 5;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 5;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 40;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 5;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010000.00Z-19000101T010000.10Z", NULL, "2.0", NULL,
+ NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_intra_frames_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 900;
+ rtp_check.expected_i_frame_ts_interval = 900;
+ rtp_check.expected_n_buffers = 10;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 0;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 0;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010000.00Z-19000101T010000.10Z", NULL, NULL,
+ "intra", NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_intra_frames_with_interval_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 1800;
+ rtp_check.expected_i_frame_ts_interval = 1800;
+ rtp_check.expected_n_buffers = 5;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 5;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 0;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 0;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 5;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010000.00Z-19000101T010000.10Z", NULL, NULL,
+ "intra/20", NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_predicted_frames_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 450;
+ rtp_check.expected_i_frame_ts_interval = 450;
+ rtp_check.expected_n_buffers = 20;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 10;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 0;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010000.00Z-19000101T010000.10Z", NULL, NULL,
+ "predicted", NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_reverse_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = -90;
+ rtp_check.expected_i_frame_ts_interval = 1710;
+ rtp_check.expected_n_buffers = 100;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 10;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 80;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010000.10Z-19000101T010000.00Z", NULL, "-1.0",
+ NULL, NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_speed_reverse_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = -45;
+ rtp_check.expected_i_frame_ts_interval = 855;
+ rtp_check.expected_n_buffers = 100;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 10;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 80;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010000.10Z-19000101T010000.00Z", "2.0", "-1.0",
+ NULL, NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_scale_reverse_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = -90;
+ rtp_check.expected_i_frame_ts_interval = 1710;
+ rtp_check.expected_n_buffers = 50;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 5;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 5;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 40;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 5;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010001.10Z-19000101T010001.00Z", NULL, "-2.0",
+ NULL, NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_intra_frames_reverse_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 0;
+ rtp_check.expected_i_frame_ts_interval = 900;
+ rtp_check.expected_n_buffers = 10;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 0;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 0;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010001.10Z-19000101T010001.00Z", NULL, "-1.0",
+ "intra", NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_predicted_frames_reverse_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = -450;
+ rtp_check.expected_i_frame_ts_interval = 1350;
+ rtp_check.expected_n_buffers = 20;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 10;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 0;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010001.10Z-19000101T010001.00Z", NULL, "-1.0",
+ "predicted", NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_intra_frames_with_interval_reverse_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 0;
+ rtp_check.expected_i_frame_ts_interval = 1800;
+ rtp_check.expected_n_buffers = 5;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 5;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 0;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 0;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 5;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = TRUE;
+
+ test_seek ("clock=19000101T010001.10Z-19000101T010001.00Z", NULL, "-1.0",
+ "intra/20", NULL, &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_rate_control_no_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 90;
+ rtp_check.expected_i_frame_ts_interval = 90;
+ rtp_check.expected_n_buffers = 100;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 10;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 80;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = FALSE;
+
+ test_seek ("clock=19000101T010000.00Z-19000101T010000.10Z", NULL, NULL, NULL,
+ "no", &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_rate_control_no_reverse_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 90;
+ rtp_check.expected_i_frame_ts_interval = -1710;
+ rtp_check.expected_n_buffers = 100;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 10;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 80;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = FALSE;
+
+ test_seek ("clock=19000101T010000.10Z-19000101T010000.00Z", NULL, "-1.0",
+ NULL, "no", &rtp_check);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_rate_control_no_frames_trick_mode)
+{
+ RTPCheckData rtp_check;
+
+ rtp_check.previous_ts = 0;
+ rtp_check.expected_ts_interval = 900;
+ rtp_check.expected_i_frame_ts_interval = 900;
+ rtp_check.expected_n_buffers = 10;
+ rtp_check.n_buffers = 0;
+ rtp_check.expected_n_i_frames = 10;
+ rtp_check.n_i_frames = 0;
+ rtp_check.expected_n_p_frames = 0;
+ rtp_check.n_p_frames = 0;
+ rtp_check.expected_n_b_frames = 0;
+ rtp_check.n_b_frames = 0;
+ rtp_check.expected_n_clean_points = 10;
+ rtp_check.n_clean_points = 0;
+ rtp_check.timestamped_rtcp = FALSE;
+
+ test_seek ("clock=19000101T010000.00Z-19000101T010000.10Z", NULL, NULL,
+ "intra", "no", &rtp_check);
+}
+
+GST_END_TEST;
+static Suite *
+onvif_suite (void)
+{
+ Suite *s = suite_create ("onvif");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+
+ tcase_add_test (tc, test_x_onvif_track);
+ tcase_add_test (tc, test_onvif_replay);
+ tcase_add_test (tc, test_src_seek_simple);
+ tcase_add_test (tc, test_speed_trick_mode);
+ tcase_add_test (tc, test_scale_trick_mode);
+ tcase_add_test (tc, test_intra_frames_trick_mode);
+ tcase_add_test (tc, test_predicted_frames_trick_mode);
+ tcase_add_test (tc, test_intra_frames_with_interval_trick_mode);
+ tcase_add_test (tc, test_reverse_trick_mode);
+ tcase_add_test (tc, test_speed_reverse_trick_mode);
+ tcase_add_test (tc, test_scale_reverse_trick_mode);
+ tcase_add_test (tc, test_intra_frames_reverse_trick_mode);
+ tcase_add_test (tc, test_predicted_frames_reverse_trick_mode);
+ tcase_add_test (tc, test_intra_frames_with_interval_reverse_trick_mode);
+ tcase_add_test (tc, test_rate_control_no_trick_mode);
+ tcase_add_test (tc, test_rate_control_no_reverse_trick_mode);
+ tcase_add_test (tc, test_rate_control_no_frames_trick_mode);
+
+ return s;
+}
+
+GST_CHECK_MAIN (onvif);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Sebastian Rasmussen <sebras@hotmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-permissions.h>
+
+GST_START_TEST (test_permissions)
+{
+ GstRTSPPermissions *perms;
+ GstRTSPPermissions *copy;
+ GstStructure *role_structure;
+
+ perms = gst_rtsp_permissions_new ();
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "missing", "permission1"));
+ gst_rtsp_permissions_unref (perms);
+
+ perms = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (perms, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "missing"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "missing", "permission1"));
+ copy = GST_RTSP_PERMISSIONS (gst_mini_object_copy (GST_MINI_OBJECT (perms)));
+ gst_rtsp_permissions_unref (perms);
+ fail_unless (gst_rtsp_permissions_is_allowed (copy, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (copy, "user", "permission2"));
+ gst_rtsp_permissions_unref (copy);
+
+ perms = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (perms, "admin",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_permissions_add_role (perms, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission2"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ gst_rtsp_permissions_unref (perms);
+
+ perms = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (perms, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ gst_rtsp_permissions_add_role (perms, "user",
+ "permission1", G_TYPE_BOOLEAN, FALSE,
+ "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ gst_rtsp_permissions_unref (perms);
+
+ perms = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (perms, "admin",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_permissions_add_role (perms, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission2"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ gst_rtsp_permissions_remove_role (perms, "user");
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission2"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+
+ /* _add_permission_for_role() should overwrite existing or create new role */
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+ gst_rtsp_permissions_add_permission_for_role (perms, "admin", "permission1",
+ FALSE);
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "tester", "permission1"));
+ gst_rtsp_permissions_add_permission_for_role (perms, "tester", "permission1",
+ TRUE);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "tester",
+ "permission1"));
+ gst_rtsp_permissions_add_permission_for_role (perms, "tester", "permission1",
+ FALSE);
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "tester", "permission1"));
+ gst_rtsp_permissions_add_permission_for_role (perms, "tester", "permission2",
+ TRUE);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "tester",
+ "permission2"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "tester", "permission3"));
+
+ gst_rtsp_permissions_add_role_empty (perms, "noone");
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "noone", "permission1"));
+
+ role_structure = gst_structure_new ("tester", "permission1", G_TYPE_BOOLEAN,
+ TRUE, NULL);
+ gst_rtsp_permissions_add_role_from_structure (perms, role_structure);
+ gst_structure_free (role_structure);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "tester",
+ "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "tester", "permission2"));
+
+ gst_rtsp_permissions_unref (perms);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtsppermissions_suite (void)
+{
+ Suite *s = suite_create ("rtsppermissions");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_permissions);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtsppermissions);
--- /dev/null
+/* GStreamer unit test for rtspclientsink
+ * Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
+ * @author David Svensson Fors <davidsf at axis dot com>
+ * Copyright (C) 2015 Centricular Ltd
+ * @author Tim-Philipp Müller <tim@centricular.com>
+ * @author Jan Schmidt <jan@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/sdp/gstsdpmessage.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+
+#include <stdio.h>
+#include <netinet/in.h>
+
+#include "rtsp-server.h"
+
+#define TEST_MOUNT_POINT "/test"
+
+/* tested rtsp server */
+static GstRTSPServer *server = NULL;
+
+/* tcp port that the test server listens for rtsp requests on */
+static gint test_port = 0;
+static gint server_send_rtcp_port;
+
+/* id of the server's source within the GMainContext */
+static guint source_id;
+
+/* iterate the default main context until there are no events to dispatch */
+static void
+iterate (void)
+{
+ while (g_main_context_iteration (NULL, FALSE)) {
+ GST_DEBUG ("iteration");
+ }
+}
+
+/* start the testing rtsp server for RECORD mode */
+static GstRTSPMediaFactory *
+start_record_server (const gchar * launch_line)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMountPoints *mounts;
+ gchar *service;
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = gst_rtsp_media_factory_new ();
+
+ gst_rtsp_media_factory_set_transport_mode (factory,
+ GST_RTSP_TRANSPORT_MODE_RECORD);
+ gst_rtsp_media_factory_set_launch (factory, launch_line);
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+ g_object_unref (mounts);
+
+ /* set port to any */
+ gst_rtsp_server_set_service (server, "0");
+
+ /* attach to default main context */
+ source_id = gst_rtsp_server_attach (server, NULL);
+ fail_if (source_id == 0);
+
+ /* get port */
+ service = gst_rtsp_server_get_service (server);
+ test_port = atoi (service);
+ fail_unless (test_port != 0);
+ g_free (service);
+
+ GST_DEBUG ("rtsp server listening on port %d", test_port);
+ return factory;
+}
+
+/* stop the tested rtsp server */
+static void
+stop_server (void)
+{
+ g_source_remove (source_id);
+ source_id = 0;
+
+ GST_DEBUG ("rtsp server stopped");
+}
+
+/* fixture setup function */
+static void
+setup (void)
+{
+ server = gst_rtsp_server_new ();
+}
+
+/* fixture clean-up function */
+static void
+teardown (void)
+{
+ if (server) {
+ g_object_unref (server);
+ server = NULL;
+ }
+ test_port = 0;
+}
+
+/* create an rtsp connection to the server on test_port */
+static gchar *
+get_server_uri (gint port, const gchar * mount_point)
+{
+ gchar *address;
+ gchar *uri_string;
+ GstRTSPUrl *url = NULL;
+
+ address = gst_rtsp_server_get_address (server);
+ uri_string = g_strdup_printf ("rtsp://%s:%d%s", address, port, mount_point);
+ g_free (address);
+
+ fail_unless (gst_rtsp_url_parse (uri_string, &url) == GST_RTSP_OK);
+ gst_rtsp_url_free (url);
+
+ return uri_string;
+}
+
+static GstRTSPFilterResult
+check_transport (GstRTSPStream * stream, GstRTSPStreamTransport * strans,
+ gpointer user_data)
+{
+ const GstRTSPTransport *trans =
+ gst_rtsp_stream_transport_get_transport (strans);
+
+ server_send_rtcp_port = trans->client_port.max;
+
+ return GST_RTSP_FILTER_KEEP;
+}
+
+static void
+new_state_cb (GstRTSPMedia * media, gint state, gpointer user_data)
+{
+ if (state == GST_STATE_PLAYING) {
+ GstRTSPStream *stream = gst_rtsp_media_get_stream (media, 0);
+
+ gst_rtsp_stream_transport_filter (stream,
+ (GstRTSPStreamTransportFilterFunc) check_transport, user_data);
+ }
+}
+
+static void
+media_constructed_cb (GstRTSPMediaFactory * mfactory, GstRTSPMedia * media,
+ gpointer user_data)
+{
+ GstElement **p_sink = user_data;
+ GstElement *bin;
+
+ g_signal_connect (media, "new-state", G_CALLBACK (new_state_cb), user_data);
+
+ bin = gst_rtsp_media_get_element (media);
+ *p_sink = gst_bin_get_by_name (GST_BIN (bin), "sink");
+ GST_INFO ("media constructed!: %" GST_PTR_FORMAT, *p_sink);
+ gst_object_unref (bin);
+}
+
+#define AUDIO_PIPELINE "audiotestsrc num-buffers=%d ! " \
+ "audio/x-raw,rate=8000 ! alawenc ! rtspclientsink name=sink location=%s"
+#define RECORD_N_BUFS 10
+
+GST_START_TEST (test_record)
+{
+ GstRTSPMediaFactory *mfactory;
+ GstElement *server_sink = NULL;
+ gint i;
+
+ mfactory =
+ start_record_server ("( rtppcmadepay name=depay0 ! appsink name=sink )");
+
+ g_signal_connect (mfactory, "media-constructed",
+ G_CALLBACK (media_constructed_cb), &server_sink);
+
+ /* Create an rtspclientsink and send some data */
+ {
+ gchar *uri = get_server_uri (test_port, TEST_MOUNT_POINT);
+ gchar *pipe_str;
+ GstMessage *msg;
+ GstElement *pipeline;
+ GstBus *bus;
+
+ pipe_str = g_strdup_printf (AUDIO_PIPELINE, RECORD_N_BUFS, uri);
+ g_free (uri);
+
+ pipeline = gst_parse_launch (pipe_str, NULL);
+ g_free (pipe_str);
+
+ fail_unless (pipeline != NULL);
+
+ bus = gst_element_get_bus (pipeline);
+ fail_if (bus == NULL);
+
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+ msg = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1);
+ fail_if (GST_MESSAGE_TYPE (msg) != GST_MESSAGE_EOS);
+ gst_message_unref (msg);
+
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+ gst_object_unref (bus);
+ }
+
+ iterate ();
+
+ fail_unless (server_send_rtcp_port != 0);
+
+ /* check received data (we assume every buffer created by audiotestsrc and
+ * subsequently encoded by mulawenc results in exactly one RTP packet) */
+ for (i = 0; i < RECORD_N_BUFS; ++i) {
+ GstSample *sample = NULL;
+
+ g_signal_emit_by_name (G_OBJECT (server_sink), "pull-sample", &sample);
+ GST_INFO ("%2d recv sample: %p", i, sample);
+ if (sample) {
+ gst_sample_unref (sample);
+ sample = NULL;
+ }
+ }
+ gst_object_unref (server_sink);
+
+ /* clean up and iterate so the clean-up can finish */
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+/* Make sure we can shut down rtspclientsink while it's still waiting for
+ * the initial preroll data */
+GST_START_TEST (test_record_no_data)
+{
+
+ start_record_server ("( rtppcmadepay name=depay0 ! fakesink )");
+
+ /* Create an rtspclientsink and send some data */
+ {
+ gchar *uri = get_server_uri (test_port, TEST_MOUNT_POINT);
+ gchar *pipe_str;
+ GstMessage *msg;
+ GstElement *pipeline;
+ GstBus *bus;
+
+ pipe_str = g_strdup_printf ("appsrc caps=audio/x-alaw,rate=8000,channels=1"
+ " ! rtspclientsink name=sink location=%s", uri);
+ g_free (uri);
+
+ pipeline = gst_parse_launch (pipe_str, NULL);
+ g_free (pipe_str);
+
+ fail_unless (pipeline != NULL);
+
+ bus = gst_element_get_bus (pipeline);
+ fail_if (bus == NULL);
+
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+ /* wait for a bit */
+ msg = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_ERROR,
+ 500 * GST_MSECOND);
+ fail_unless (msg == NULL);
+
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+ gst_object_unref (bus);
+ }
+
+ /* clean up and iterate so the clean-up can finish */
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspclientsink_suite (void)
+{
+ Suite *s = suite_create ("rtspclientsink");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_add_checked_fixture (tc, setup, teardown);
+ tcase_set_timeout (tc, 120);
+ tcase_add_test (tc, test_record);
+ tcase_add_test (tc, test_record_no_data);
+ return s;
+}
+
+GST_CHECK_MAIN (rtspclientsink);
--- /dev/null
+/* GStreamer unit test for GstRTSPServer
+ * Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
+ * @author David Svensson Fors <davidsf at axis dot com>
+ * Copyright (C) 2015 Centricular Ltd
+ * @author Tim-Philipp Müller <tim@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/sdp/gstsdpmessage.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+
+#include <stdio.h>
+#include <netinet/in.h>
+
+#include "rtsp-server.h"
+
+#define ERRORIGNORE "errorignore ignore-error=false ignore-notlinked=true " \
+ "ignore-notnegotiated=false convert-to=ok"
+#define VIDEO_PIPELINE "videotestsrc ! " \
+ ERRORIGNORE " ! " \
+ "video/x-raw,format=I420,width=352,height=288 ! " \
+ "rtpgstpay name=pay0 pt=96"
+#define AUDIO_PIPELINE "audiotestsrc ! " \
+ ERRORIGNORE " ! " \
+ "audio/x-raw,rate=8000 ! " \
+ "rtpgstpay name=pay1 pt=97"
+
+#define TEST_MOUNT_POINT "/test"
+#define TEST_PROTO "RTP/AVP"
+#define TEST_ENCODING "X-GST"
+#define TEST_CLOCK_RATE "90000"
+
+/* tested rtsp server */
+static GstRTSPServer *server = NULL;
+
+/* tcp port that the test server listens for rtsp requests on */
+static gint test_port = 0;
+
+/* id of the server's source within the GMainContext */
+static guint source_id;
+
+/* iterate the default main loop until there are no events to dispatch */
+static void
+iterate (void)
+{
+ while (g_main_context_iteration (NULL, FALSE)) {
+ GST_DEBUG ("iteration");
+ }
+}
+
+static void
+get_client_ports_full (GstRTSPRange * range, GSocket ** rtp_socket,
+ GSocket ** rtcp_socket)
+{
+ GSocket *rtp = NULL;
+ GSocket *rtcp = NULL;
+ gint rtp_port = 0;
+ gint rtcp_port;
+ GInetAddress *anyaddr = g_inet_address_new_any (G_SOCKET_FAMILY_IPV4);
+ GSocketAddress *sockaddr;
+ gboolean bound;
+
+ for (;;) {
+ if (rtp_port != 0)
+ rtp_port += 2;
+
+ rtp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ fail_unless (rtp != NULL);
+
+ sockaddr = g_inet_socket_address_new (anyaddr, rtp_port);
+ fail_unless (sockaddr != NULL);
+ bound = g_socket_bind (rtp, sockaddr, FALSE, NULL);
+ g_object_unref (sockaddr);
+ if (!bound) {
+ g_object_unref (rtp);
+ continue;
+ }
+
+ sockaddr = g_socket_get_local_address (rtp, NULL);
+ fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
+ rtp_port =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
+ g_object_unref (sockaddr);
+
+ if (rtp_port % 2 != 0) {
+ rtp_port += 1;
+ g_object_unref (rtp);
+ continue;
+ }
+
+ rtcp_port = rtp_port + 1;
+
+ rtcp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ fail_unless (rtcp != NULL);
+
+ sockaddr = g_inet_socket_address_new (anyaddr, rtcp_port);
+ fail_unless (sockaddr != NULL);
+ bound = g_socket_bind (rtcp, sockaddr, FALSE, NULL);
+ g_object_unref (sockaddr);
+ if (!bound) {
+ g_object_unref (rtp);
+ g_object_unref (rtcp);
+ continue;
+ }
+
+ sockaddr = g_socket_get_local_address (rtcp, NULL);
+ fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
+ fail_unless (rtcp_port ==
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr)));
+ g_object_unref (sockaddr);
+
+ break;
+ }
+
+ range->min = rtp_port;
+ range->max = rtcp_port;
+ if (rtp_socket)
+ *rtp_socket = rtp;
+ else
+ g_object_unref (rtp);
+ if (rtcp_socket)
+ *rtcp_socket = rtcp;
+ else
+ g_object_unref (rtcp);
+ GST_DEBUG ("client_port=%d-%d", range->min, range->max);
+ g_object_unref (anyaddr);
+}
+
+/* get a free rtp/rtcp client port pair */
+static void
+get_client_ports (GstRTSPRange * range)
+{
+ get_client_ports_full (range, NULL, NULL);
+}
+
+/* start the tested rtsp server */
+static void
+start_server (gboolean set_shared_factory)
+{
+ GstRTSPMountPoints *mounts;
+ gchar *service;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAddressPool *pool;
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = gst_rtsp_media_factory_new ();
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+ g_object_unref (mounts);
+
+ /* use an address pool for multicast */
+ pool = gst_rtsp_address_pool_new ();
+ gst_rtsp_address_pool_add_range (pool,
+ "224.3.0.0", "224.3.0.10", 5500, 5510, 16);
+ gst_rtsp_address_pool_add_range (pool, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
+ GST_RTSP_ADDRESS_POOL_ANY_IPV4, 6000, 6010, 0);
+ gst_rtsp_media_factory_set_address_pool (factory, pool);
+ gst_rtsp_media_factory_set_shared (factory, set_shared_factory);
+ gst_object_unref (pool);
+
+ /* set port to any */
+ gst_rtsp_server_set_service (server, "0");
+
+ /* attach to default main context */
+ source_id = gst_rtsp_server_attach (server, NULL);
+ fail_if (source_id == 0);
+
+ /* get port */
+ service = gst_rtsp_server_get_service (server);
+ test_port = atoi (service);
+ fail_unless (test_port != 0);
+ g_free (service);
+
+ GST_DEBUG ("rtsp server listening on port %d", test_port);
+}
+
+static void
+start_tcp_server (gboolean set_shared_factory)
+{
+ GstRTSPMountPoints *mounts;
+ gchar *service;
+ GstRTSPMediaFactory *factory;
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = gst_rtsp_media_factory_new ();
+
+ gst_rtsp_media_factory_set_protocols (factory, GST_RTSP_LOWER_TRANS_TCP);
+ gst_rtsp_media_factory_set_launch (factory,
+ "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+ gst_rtsp_media_factory_set_shared (factory, set_shared_factory);
+ g_object_unref (mounts);
+
+ /* set port to any */
+ gst_rtsp_server_set_service (server, "0");
+
+ /* attach to default main context */
+ source_id = gst_rtsp_server_attach (server, NULL);
+ fail_if (source_id == 0);
+
+ /* get port */
+ service = gst_rtsp_server_get_service (server);
+ test_port = atoi (service);
+ fail_unless (test_port != 0);
+ g_free (service);
+
+ GST_DEBUG ("rtsp server listening on port %d", test_port);
+
+}
+
+/* start the testing rtsp server for RECORD mode */
+static GstRTSPMediaFactory *
+start_record_server (const gchar * launch_line)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMountPoints *mounts;
+ gchar *service;
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = gst_rtsp_media_factory_new ();
+
+ gst_rtsp_media_factory_set_transport_mode (factory,
+ GST_RTSP_TRANSPORT_MODE_RECORD);
+ gst_rtsp_media_factory_set_launch (factory, launch_line);
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+ g_object_unref (mounts);
+
+ /* set port to any */
+ gst_rtsp_server_set_service (server, "0");
+
+ /* attach to default main context */
+ source_id = gst_rtsp_server_attach (server, NULL);
+ fail_if (source_id == 0);
+
+ /* get port */
+ service = gst_rtsp_server_get_service (server);
+ test_port = atoi (service);
+ fail_unless (test_port != 0);
+ g_free (service);
+
+ GST_DEBUG ("rtsp server listening on port %d", test_port);
+ return factory;
+}
+
+/* stop the tested rtsp server */
+static void
+stop_server (void)
+{
+ g_source_remove (source_id);
+ source_id = 0;
+
+ GST_DEBUG ("rtsp server stopped");
+}
+
+/* create an rtsp connection to the server on test_port */
+static GstRTSPConnection *
+connect_to_server (gint port, const gchar * mount_point)
+{
+ GstRTSPConnection *conn = NULL;
+ gchar *address;
+ gchar *uri_string;
+ GstRTSPUrl *url = NULL;
+
+ address = gst_rtsp_server_get_address (server);
+ uri_string = g_strdup_printf ("rtsp://%s:%d%s", address, port, mount_point);
+ g_free (address);
+ fail_unless (gst_rtsp_url_parse (uri_string, &url) == GST_RTSP_OK);
+ g_free (uri_string);
+
+ fail_unless (gst_rtsp_connection_create (url, &conn) == GST_RTSP_OK);
+ gst_rtsp_url_free (url);
+
+ fail_unless (gst_rtsp_connection_connect (conn, NULL) == GST_RTSP_OK);
+
+ return conn;
+}
+
+/* create an rtsp request */
+static GstRTSPMessage *
+create_request (GstRTSPConnection * conn, GstRTSPMethod method,
+ const gchar * control)
+{
+ GstRTSPMessage *request = NULL;
+ gchar *base_uri;
+ gchar *full_uri;
+
+ base_uri = gst_rtsp_url_get_request_uri (gst_rtsp_connection_get_url (conn));
+ full_uri = g_strdup_printf ("%s/%s", base_uri, control ? control : "");
+ g_free (base_uri);
+ if (gst_rtsp_message_new_request (&request, method, full_uri) != GST_RTSP_OK) {
+ GST_DEBUG ("failed to create request object");
+ g_free (full_uri);
+ return NULL;
+ }
+ g_free (full_uri);
+ return request;
+}
+
+/* send an rtsp request */
+static gboolean
+send_request (GstRTSPConnection * conn, GstRTSPMessage * request)
+{
+ if (gst_rtsp_connection_send (conn, request, NULL) != GST_RTSP_OK) {
+ GST_DEBUG ("failed to send request");
+ return FALSE;
+ }
+ return TRUE;
+}
+
+/* read rtsp response. response must be freed by the caller */
+static GstRTSPMessage *
+read_response (GstRTSPConnection * conn)
+{
+ GstRTSPMessage *response = NULL;
+ GstRTSPMsgType type;
+
+ if (gst_rtsp_message_new (&response) != GST_RTSP_OK) {
+ GST_DEBUG ("failed to create response object");
+ return NULL;
+ }
+ if (gst_rtsp_connection_receive (conn, response, NULL) != GST_RTSP_OK) {
+ GST_DEBUG ("failed to read response");
+ gst_rtsp_message_free (response);
+ return NULL;
+ }
+ type = gst_rtsp_message_get_type (response);
+ fail_unless (type == GST_RTSP_MESSAGE_RESPONSE
+ || type == GST_RTSP_MESSAGE_DATA);
+ return response;
+}
+
+/* send an rtsp request and receive response. gchar** parameters are out
+ * parameters that have to be freed by the caller */
+static GstRTSPStatusCode
+do_request_full (GstRTSPConnection * conn, GstRTSPMethod method,
+ const gchar * control, const gchar * session_in, const gchar * transport_in,
+ const gchar * range_in, const gchar * require_in,
+ gchar ** content_type, gchar ** content_base, gchar ** body,
+ gchar ** session_out, gchar ** transport_out, gchar ** range_out,
+ gchar ** unsupported_out)
+{
+ GstRTSPMessage *request;
+ GstRTSPMessage *response;
+ GstRTSPStatusCode code;
+ gchar *value;
+ GstRTSPMsgType msg_type;
+
+ /* create request */
+ request = create_request (conn, method, control);
+
+ /* add headers */
+ if (session_in) {
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session_in);
+ }
+ if (transport_in) {
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_TRANSPORT, transport_in);
+ }
+ if (range_in) {
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_RANGE, range_in);
+ }
+ if (require_in) {
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_REQUIRE, require_in);
+ }
+
+ /* send request */
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+
+ iterate ();
+
+ /* read response */
+ response = read_response (conn);
+ fail_unless (response != NULL);
+
+ msg_type = gst_rtsp_message_get_type (response);
+
+ if (msg_type == GST_RTSP_MESSAGE_DATA) {
+ do {
+ gst_rtsp_message_free (response);
+ response = read_response (conn);
+ msg_type = gst_rtsp_message_get_type (response);
+ } while (msg_type == GST_RTSP_MESSAGE_DATA);
+ }
+
+ fail_unless (msg_type == GST_RTSP_MESSAGE_RESPONSE);
+
+ /* check status line */
+ gst_rtsp_message_parse_response (response, &code, NULL, NULL);
+ if (code != GST_RTSP_STS_OK) {
+ if (unsupported_out != NULL && code == GST_RTSP_STS_OPTION_NOT_SUPPORTED) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_UNSUPPORTED,
+ &value, 0);
+ *unsupported_out = g_strdup (value);
+ }
+ gst_rtsp_message_free (response);
+ return code;
+ }
+
+ /* get information from response */
+ if (content_type) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_TYPE,
+ &value, 0);
+ *content_type = g_strdup (value);
+ }
+ if (content_base) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
+ &value, 0);
+ *content_base = g_strdup (value);
+ }
+ if (body) {
+ *body = g_malloc (response->body_size + 1);
+ strncpy (*body, (gchar *) response->body, response->body_size);
+ }
+ if (session_out) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &value, 0);
+
+ value = g_strdup (value);
+
+ /* Remove the timeout */
+ if (value) {
+ char *pos = strchr (value, ';');
+ if (pos)
+ *pos = 0;
+ }
+ if (session_in) {
+ /* check that we got the same session back */
+ fail_unless (!g_strcmp0 (value, session_in));
+ }
+ *session_out = value;
+ }
+ if (transport_out) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT, &value, 0);
+ *transport_out = g_strdup (value);
+ }
+ if (range_out) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_RANGE, &value, 0);
+ *range_out = g_strdup (value);
+ }
+
+ gst_rtsp_message_free (response);
+ return code;
+}
+
+/* send an rtsp request and receive response. gchar** parameters are out
+ * parameters that have to be freed by the caller */
+static GstRTSPStatusCode
+do_request (GstRTSPConnection * conn, GstRTSPMethod method,
+ const gchar * control, const gchar * session_in,
+ const gchar * transport_in, const gchar * range_in,
+ gchar ** content_type, gchar ** content_base, gchar ** body,
+ gchar ** session_out, gchar ** transport_out, gchar ** range_out)
+{
+ return do_request_full (conn, method, control, session_in, transport_in,
+ range_in, NULL, content_type, content_base, body, session_out,
+ transport_out, range_out, NULL);
+}
+
+/* send an rtsp request with a method and a session, and receive response */
+static GstRTSPStatusCode
+do_simple_request (GstRTSPConnection * conn, GstRTSPMethod method,
+ const gchar * session)
+{
+ return do_request (conn, method, NULL, session, NULL, NULL, NULL,
+ NULL, NULL, NULL, NULL, NULL);
+}
+
+/* send an rtsp request with a method,session and range in,
+ * and receive response. range_in is the Range in req header */
+static GstRTSPStatusCode
+do_simple_request_rangein (GstRTSPConnection * conn, GstRTSPMethod method,
+ const gchar * session, const gchar * rangein)
+{
+ return do_request (conn, method, NULL, session, NULL, rangein, NULL,
+ NULL, NULL, NULL, NULL, NULL);
+}
+
+/* send a DESCRIBE request and receive response. returns a received
+ * GstSDPMessage that must be freed by the caller */
+static GstSDPMessage *
+do_describe (GstRTSPConnection * conn, const gchar * mount_point)
+{
+ GstSDPMessage *sdp_message;
+ gchar *content_type = NULL;
+ gchar *content_base = NULL;
+ gchar *body = NULL;
+ gchar *address;
+ gchar *expected_content_base;
+
+ /* send DESCRIBE request */
+ fail_unless (do_request (conn, GST_RTSP_DESCRIBE, NULL, NULL, NULL, NULL,
+ &content_type, &content_base, &body, NULL, NULL, NULL) ==
+ GST_RTSP_STS_OK);
+
+ /* check response values */
+ fail_unless (!g_strcmp0 (content_type, "application/sdp"));
+ address = gst_rtsp_server_get_address (server);
+ expected_content_base =
+ g_strdup_printf ("rtsp://%s:%d%s/", address, test_port, mount_point);
+ fail_unless (!g_strcmp0 (content_base, expected_content_base));
+
+ /* create sdp message */
+ fail_unless (gst_sdp_message_new (&sdp_message) == GST_SDP_OK);
+ fail_unless (gst_sdp_message_parse_buffer ((guint8 *) body,
+ strlen (body), sdp_message) == GST_SDP_OK);
+
+ /* clean up */
+ g_free (content_type);
+ g_free (content_base);
+ g_free (body);
+ g_free (address);
+ g_free (expected_content_base);
+
+ return sdp_message;
+}
+
+/* send a SETUP request and receive response. if *session is not NULL,
+ * it is used in the request. otherwise, *session is set to a returned
+ * session string that must be freed by the caller. the returned
+ * transport must be freed by the caller. */
+static GstRTSPStatusCode
+do_setup_full (GstRTSPConnection * conn, const gchar * control,
+ GstRTSPLowerTrans lower_transport, const GstRTSPRange * client_ports,
+ const gchar * require, gchar ** session, GstRTSPTransport ** transport,
+ gchar ** unsupported)
+{
+ GstRTSPStatusCode code;
+ gchar *session_in = NULL;
+ GString *transport_string_in = NULL;
+ gchar **session_out = NULL;
+ gchar *transport_string_out = NULL;
+
+ /* prepare and send SETUP request */
+ if (session) {
+ if (*session) {
+ session_in = *session;
+ } else {
+ session_out = session;
+ }
+ }
+
+ transport_string_in = g_string_new (TEST_PROTO);
+ switch (lower_transport) {
+ case GST_RTSP_LOWER_TRANS_UDP:
+ transport_string_in =
+ g_string_append (transport_string_in, "/UDP;unicast");
+ break;
+ case GST_RTSP_LOWER_TRANS_UDP_MCAST:
+ transport_string_in =
+ g_string_append (transport_string_in, "/UDP;multicast");
+ break;
+ case GST_RTSP_LOWER_TRANS_TCP:
+ transport_string_in =
+ g_string_append (transport_string_in, "/TCP;unicast");
+ break;
+ default:
+ g_assert_not_reached ();
+ break;
+ }
+
+ if (client_ports) {
+ g_string_append_printf (transport_string_in, ";client_port=%d-%d",
+ client_ports->min, client_ports->max);
+ }
+
+ code =
+ do_request_full (conn, GST_RTSP_SETUP, control, session_in,
+ transport_string_in->str, NULL, require, NULL, NULL, NULL, session_out,
+ &transport_string_out, NULL, unsupported);
+ g_string_free (transport_string_in, TRUE);
+
+ if (transport_string_out) {
+ /* create transport */
+ fail_unless (gst_rtsp_transport_new (transport) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_transport_parse (transport_string_out,
+ *transport) == GST_RTSP_OK);
+ g_free (transport_string_out);
+ }
+ GST_INFO ("code=%d", code);
+ return code;
+}
+
+/* send a SETUP request and receive response. if *session is not NULL,
+ * it is used in the request. otherwise, *session is set to a returned
+ * session string that must be freed by the caller. the returned
+ * transport must be freed by the caller. */
+static GstRTSPStatusCode
+do_setup (GstRTSPConnection * conn, const gchar * control,
+ const GstRTSPRange * client_ports, gchar ** session,
+ GstRTSPTransport ** transport)
+{
+ return do_setup_full (conn, control, GST_RTSP_LOWER_TRANS_UDP, client_ports,
+ NULL, session, transport, NULL);
+}
+
+/* fixture setup function */
+static void
+setup (void)
+{
+ server = gst_rtsp_server_new ();
+}
+
+/* fixture clean-up function */
+static void
+teardown (void)
+{
+ if (server) {
+ g_object_unref (server);
+ server = NULL;
+ }
+ test_port = 0;
+}
+
+GST_START_TEST (test_connect)
+{
+ GstRTSPConnection *conn;
+
+ start_server (FALSE);
+
+ /* connect to server */
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* clean up */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+
+ /* iterate so the clean-up can finish */
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_describe)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ gint32 format;
+ gchar *expected_rtpmap;
+ const gchar *rtpmap;
+ const gchar *control_video;
+ const gchar *control_audio;
+
+ start_server (FALSE);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* send DESCRIBE request */
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+
+ /* check video sdp */
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ fail_unless (!g_strcmp0 (gst_sdp_media_get_proto (sdp_media), TEST_PROTO));
+ fail_unless (gst_sdp_media_formats_len (sdp_media) == 1);
+ sscanf (gst_sdp_media_get_format (sdp_media, 0), "%" G_GINT32_FORMAT,
+ &format);
+ expected_rtpmap =
+ g_strdup_printf ("%d " TEST_ENCODING "/" TEST_CLOCK_RATE, format);
+ rtpmap = gst_sdp_media_get_attribute_val (sdp_media, "rtpmap");
+ fail_unless (!g_strcmp0 (rtpmap, expected_rtpmap));
+ g_free (expected_rtpmap);
+ control_video = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ fail_unless (!g_strcmp0 (control_video, "stream=0"));
+
+ /* check audio sdp */
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ fail_unless (!g_strcmp0 (gst_sdp_media_get_proto (sdp_media), TEST_PROTO));
+ fail_unless (gst_sdp_media_formats_len (sdp_media) == 1);
+ sscanf (gst_sdp_media_get_format (sdp_media, 0), "%" G_GINT32_FORMAT,
+ &format);
+ expected_rtpmap =
+ g_strdup_printf ("%d " TEST_ENCODING "/" TEST_CLOCK_RATE, format);
+ rtpmap = gst_sdp_media_get_attribute_val (sdp_media, "rtpmap");
+ fail_unless (!g_strcmp0 (rtpmap, expected_rtpmap));
+ g_free (expected_rtpmap);
+ control_audio = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ fail_unless (!g_strcmp0 (control_audio, "stream=1"));
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_describe_record_media)
+{
+ GstRTSPConnection *conn;
+
+ start_record_server ("( fakesink name=depay0 )");
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* send DESCRIBE request */
+ fail_unless_equals_int (do_request (conn, GST_RTSP_DESCRIBE, NULL, NULL, NULL,
+ NULL, NULL, NULL, NULL, NULL, NULL, NULL),
+ GST_RTSP_STS_METHOD_NOT_ALLOWED);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_describe_non_existing_mount_point)
+{
+ GstRTSPConnection *conn;
+
+ start_server (FALSE);
+
+ /* send DESCRIBE request for a non-existing mount point
+ * and check that we get a 404 Not Found */
+ conn = connect_to_server (test_port, "/non-existing");
+ fail_unless (do_simple_request (conn, GST_RTSP_DESCRIBE, NULL)
+ == GST_RTSP_STS_NOT_FOUND);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+static void
+do_test_setup (GstRTSPLowerTrans lower_transport)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_ports = { 0 };
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+
+ start_server (FALSE);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_ports);
+
+ /* send SETUP request for video */
+ fail_unless (do_setup_full (conn, video_control, lower_transport,
+ &client_ports, NULL, &session, &video_transport,
+ NULL) == GST_RTSP_STS_OK);
+ GST_DEBUG ("set up video %s, got session '%s'", video_control, session);
+
+ /* check response from SETUP */
+ fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
+ fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
+ fail_unless (video_transport->lower_transport == lower_transport);
+ fail_unless (video_transport->mode_play);
+ gst_rtsp_transport_free (video_transport);
+
+ /* send SETUP request for audio */
+ fail_unless (do_setup_full (conn, audio_control, lower_transport,
+ &client_ports, NULL, &session, &audio_transport,
+ NULL) == GST_RTSP_STS_OK);
+ GST_DEBUG ("set up audio %s with session '%s'", audio_control, session);
+
+ /* check response from SETUP */
+ fail_unless (audio_transport->trans == GST_RTSP_TRANS_RTP);
+ fail_unless (audio_transport->profile == GST_RTSP_PROFILE_AVP);
+ fail_unless (audio_transport->lower_transport == lower_transport);
+ fail_unless (audio_transport->mode_play);
+ gst_rtsp_transport_free (audio_transport);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_free (session);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_START_TEST (test_setup_udp)
+{
+ do_test_setup (GST_RTSP_LOWER_TRANS_UDP);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_tcp)
+{
+ do_test_setup (GST_RTSP_LOWER_TRANS_TCP);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_udp_mcast)
+{
+ do_test_setup (GST_RTSP_LOWER_TRANS_UDP_MCAST);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_twice)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ GstRTSPRange client_ports;
+ GstRTSPTransport *video_transport = NULL;
+ gchar *session1 = NULL;
+ gchar *session2 = NULL;
+
+ start_server (FALSE);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* we wan't more than one session for this connection */
+ gst_rtsp_connection_set_remember_session_id (conn, FALSE);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get the control url */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_ports);
+
+ /* send SETUP request for one session */
+ fail_unless (do_setup (conn, video_control, &client_ports, &session1,
+ &video_transport) == GST_RTSP_STS_OK);
+ GST_DEBUG ("set up video %s, got session '%s'", video_control, session1);
+
+ /* check response from SETUP */
+ fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
+ fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
+ fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
+ fail_unless (video_transport->mode_play);
+ gst_rtsp_transport_free (video_transport);
+
+ /* send SETUP request for another session */
+ fail_unless (do_setup (conn, video_control, &client_ports, &session2,
+ &video_transport) == GST_RTSP_STS_OK);
+ GST_DEBUG ("set up video %s, got session '%s'", video_control, session2);
+
+ /* check response from SETUP */
+ fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
+ fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
+ fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
+ fail_unless (video_transport->mode_play);
+ gst_rtsp_transport_free (video_transport);
+
+ /* session can not be the same */
+ fail_unless (strcmp (session1, session2));
+
+ /* send TEARDOWN request for the first session */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session1) == GST_RTSP_STS_OK);
+
+ /* send TEARDOWN request for the second session */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session2) == GST_RTSP_STS_OK);
+
+ g_free (session1);
+ g_free (session2);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_with_require_header)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ GstRTSPRange client_ports;
+ gchar *session = NULL;
+ gchar *unsupported = NULL;
+ GstRTSPTransport *video_transport = NULL;
+
+ start_server (FALSE);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_ports);
+
+ /* send SETUP request for video, with single Require header */
+ fail_unless_equals_int (do_setup_full (conn, video_control,
+ GST_RTSP_LOWER_TRANS_UDP, &client_ports, "funky-feature", &session,
+ &video_transport, &unsupported), GST_RTSP_STS_OPTION_NOT_SUPPORTED);
+ fail_unless_equals_string (unsupported, "funky-feature");
+ g_free (unsupported);
+ unsupported = NULL;
+
+ /* send SETUP request for video, with multiple Require headers */
+ fail_unless_equals_int (do_setup_full (conn, video_control,
+ GST_RTSP_LOWER_TRANS_UDP, &client_ports,
+ "funky-feature, foo-bar, superburst", &session, &video_transport,
+ &unsupported), GST_RTSP_STS_OPTION_NOT_SUPPORTED);
+ fail_unless_equals_string (unsupported, "funky-feature, foo-bar, superburst");
+ g_free (unsupported);
+ unsupported = NULL;
+
+ /* ok, just do a normal setup then (make sure that still works) */
+ fail_unless_equals_int (do_setup (conn, video_control, &client_ports,
+ &session, &video_transport), GST_RTSP_STS_OK);
+
+ GST_DEBUG ("set up video %s, got session '%s'", video_control, session);
+
+ /* check response from SETUP */
+ fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
+ fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
+ fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
+ fail_unless (video_transport->mode_play);
+ gst_rtsp_transport_free (video_transport);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_free (session);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_non_existing_stream)
+{
+ GstRTSPConnection *conn;
+ GstRTSPRange client_ports;
+
+ start_server (FALSE);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ get_client_ports (&client_ports);
+
+ /* send SETUP request with a non-existing stream and check that we get a
+ * 404 Not Found */
+ fail_unless (do_setup (conn, "stream=7", &client_ports, NULL,
+ NULL) == GST_RTSP_STS_NOT_FOUND);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+static void
+receive_rtp (GSocket * socket, GSocketAddress ** addr)
+{
+ GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
+
+ for (;;) {
+ gssize bytes;
+ GstMapInfo map = GST_MAP_INFO_INIT;
+ GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
+
+ gst_buffer_map (buffer, &map, GST_MAP_WRITE);
+ bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
+ map.maxsize, NULL, NULL);
+ fail_unless (bytes > 0);
+ gst_buffer_unmap (buffer, &map);
+ gst_buffer_set_size (buffer, bytes);
+
+ if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer)) {
+ gst_rtp_buffer_unmap (&rtpbuffer);
+ break;
+ }
+
+ if (addr)
+ g_clear_object (addr);
+ }
+
+ gst_buffer_unref (buffer);
+}
+
+static void
+receive_rtcp (GSocket * socket, GSocketAddress ** addr, GstRTCPType type)
+{
+ GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
+
+ for (;;) {
+ gssize bytes;
+ GstMapInfo map = GST_MAP_INFO_INIT;
+
+ gst_buffer_map (buffer, &map, GST_MAP_WRITE);
+ bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
+ map.maxsize, NULL, NULL);
+ fail_unless (bytes > 0);
+ gst_buffer_unmap (buffer, &map);
+ gst_buffer_set_size (buffer, bytes);
+
+ if (gst_rtcp_buffer_validate (buffer)) {
+ GstRTCPBuffer rtcpbuffer = GST_RTCP_BUFFER_INIT;
+ GstRTCPPacket packet;
+
+ if (type) {
+ fail_unless (gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcpbuffer));
+ fail_unless (gst_rtcp_buffer_get_first_packet (&rtcpbuffer, &packet));
+ do {
+ if (gst_rtcp_packet_get_type (&packet) == type) {
+ gst_rtcp_buffer_unmap (&rtcpbuffer);
+ goto done;
+ }
+ } while (gst_rtcp_packet_move_to_next (&packet));
+ gst_rtcp_buffer_unmap (&rtcpbuffer);
+ } else {
+ break;
+ }
+ }
+
+ if (addr)
+ g_clear_object (addr);
+ }
+
+done:
+
+ gst_buffer_unref (buffer);
+}
+
+static void
+do_test_play_tcp_full (const gchar * range)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ gchar *range_out = NULL;
+ GstRTSPLowerTrans lower_transport = GST_RTSP_LOWER_TRANS_TCP;
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+ get_client_ports (&client_port);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup_full (conn, video_control, lower_transport,
+ &client_port, NULL, &session, &video_transport,
+ NULL) == GST_RTSP_STS_OK);
+ fail_unless (do_setup_full (conn, audio_control, lower_transport,
+ &client_port, NULL, &session, &audio_transport,
+ NULL) == GST_RTSP_STS_OK);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_request (conn, GST_RTSP_PLAY, NULL, session, NULL, range,
+ NULL, NULL, NULL, NULL, NULL, &range_out) == GST_RTSP_STS_OK);
+
+ if (range)
+ fail_unless_equals_string (range, range_out);
+ g_free (range_out);
+
+ {
+ GstRTSPMessage *message;
+ fail_unless (gst_rtsp_message_new (&message) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_connection_receive (conn, message,
+ NULL) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_message_get_type (message) == GST_RTSP_MESSAGE_DATA);
+ gst_rtsp_message_free (message);
+ }
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* FIXME: The rtsp-server always disconnects the transport before
+ * sending the RTCP BYE
+ * receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
+ */
+
+ /* clean up and iterate so the clean-up can finish */
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+}
+
+static void
+do_test_play_full (const gchar * range, GstRTSPLowerTrans lower_transport,
+ GMutex * lock)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GSocket *rtp_socket, *rtcp_socket;
+ gchar *range_out = NULL;
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup_full (conn, video_control, lower_transport,
+ &client_port, NULL, &session, &video_transport,
+ NULL) == GST_RTSP_STS_OK);
+ fail_unless (do_setup_full (conn, audio_control, lower_transport,
+ &client_port, NULL, &session, &audio_transport,
+ NULL) == GST_RTSP_STS_OK);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_request (conn, GST_RTSP_PLAY, NULL, session, NULL, range,
+ NULL, NULL, NULL, NULL, NULL, &range_out) == GST_RTSP_STS_OK);
+ if (range)
+ fail_unless_equals_string (range, range_out);
+ g_free (range_out);
+
+ for (;;) {
+ receive_rtp (rtp_socket, NULL);
+ receive_rtcp (rtcp_socket, NULL, 0);
+
+ if (lock != NULL) {
+ if (g_mutex_trylock (lock) == TRUE) {
+ g_mutex_unlock (lock);
+ break;
+ }
+ } else {
+ break;
+ }
+
+ }
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* FIXME: The rtsp-server always disconnects the transport before
+ * sending the RTCP BYE
+ * receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
+ */
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (rtp_socket);
+ g_object_unref (rtcp_socket);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+}
+
+static void
+do_test_play (const gchar * range)
+{
+ do_test_play_full (range, GST_RTSP_LOWER_TRANS_UDP, NULL);
+}
+
+GST_START_TEST (test_play)
+{
+ start_server (FALSE);
+
+ do_test_play (NULL);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_play_tcp)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_ports = { 0 };
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+
+ start_tcp_server (FALSE);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* send DESCRIBE request */
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_ports);
+
+ /* send SETUP request for the first media */
+ fail_unless (do_setup_full (conn, video_control, GST_RTSP_LOWER_TRANS_TCP,
+ &client_ports, NULL, &session, &video_transport,
+ NULL) == GST_RTSP_STS_OK);
+
+ /* check response from SETUP */
+ fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
+ fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
+ fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP);
+ fail_unless (video_transport->mode_play);
+ gst_rtsp_transport_free (video_transport);
+
+ /* send SETUP request for the second media */
+ fail_unless (do_setup_full (conn, audio_control, GST_RTSP_LOWER_TRANS_TCP,
+ &client_ports, NULL, &session, &audio_transport,
+ NULL) == GST_RTSP_STS_OK);
+
+ /* check response from SETUP */
+ fail_unless (audio_transport->trans == GST_RTSP_TRANS_RTP);
+ fail_unless (audio_transport->profile == GST_RTSP_PROFILE_AVP);
+ fail_unless (audio_transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP);
+ fail_unless (audio_transport->mode_play);
+ gst_rtsp_transport_free (audio_transport);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session) == GST_RTSP_STS_OK);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_free (session);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_play_without_session)
+{
+ GstRTSPConnection *conn;
+
+ start_server (FALSE);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* send PLAY request without a session and check that we get a
+ * 454 Session Not Found */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ NULL) == GST_RTSP_STS_SESSION_NOT_FOUND);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_bind_already_in_use)
+{
+ GstRTSPServer *serv;
+ GSocketService *service;
+ GError *error = NULL;
+ guint16 port;
+ gchar *port_str;
+
+ serv = gst_rtsp_server_new ();
+ service = g_socket_service_new ();
+
+ /* bind service to port */
+ port =
+ g_socket_listener_add_any_inet_port (G_SOCKET_LISTENER (service), NULL,
+ &error);
+ g_assert_no_error (error);
+
+ port_str = g_strdup_printf ("%d\n", port);
+
+ /* try to bind server to the same port */
+ g_object_set (serv, "service", port_str, NULL);
+ g_free (port_str);
+
+ /* attach to default main context */
+ fail_unless (gst_rtsp_server_attach (serv, NULL) == 0);
+
+ /* cleanup */
+ g_object_unref (serv);
+ g_socket_service_stop (service);
+ g_object_unref (service);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_multithreaded)
+{
+ GstRTSPThreadPool *pool;
+
+ pool = gst_rtsp_server_get_thread_pool (server);
+ gst_rtsp_thread_pool_set_max_threads (pool, 2);
+ g_object_unref (pool);
+
+ start_server (FALSE);
+
+ do_test_play (NULL);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+enum
+{
+ BLOCK_ME,
+ BLOCKED,
+ UNBLOCK
+};
+
+
+static void
+media_constructed_block (GstRTSPMediaFactory * factory,
+ GstRTSPMedia * media, gpointer user_data)
+{
+ gint *block_state = user_data;
+
+ g_mutex_lock (&check_mutex);
+
+ *block_state = BLOCKED;
+ g_cond_broadcast (&check_cond);
+
+ while (*block_state != UNBLOCK)
+ g_cond_wait (&check_cond, &check_mutex);
+ g_mutex_unlock (&check_mutex);
+}
+
+
+GST_START_TEST (test_play_multithreaded_block_in_describe)
+{
+ GstRTSPConnection *conn;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ gint block_state = BLOCK_ME;
+ GstRTSPMessage *request;
+ GstRTSPMessage *response;
+ GstRTSPStatusCode code;
+ GstRTSPThreadPool *pool;
+
+ pool = gst_rtsp_server_get_thread_pool (server);
+ gst_rtsp_thread_pool_set_max_threads (pool, 2);
+ g_object_unref (pool);
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+ fail_unless (mounts != NULL);
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
+ g_signal_connect (factory, "media-constructed",
+ G_CALLBACK (media_constructed_block), &block_state);
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT "2", factory);
+ g_object_unref (mounts);
+
+ start_server (FALSE);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT "2");
+ iterate ();
+
+ /* do describe, it will not return now as we've blocked it */
+ request = create_request (conn, GST_RTSP_DESCRIBE, NULL);
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+
+ g_mutex_lock (&check_mutex);
+ while (block_state != BLOCKED)
+ g_cond_wait (&check_cond, &check_mutex);
+ g_mutex_unlock (&check_mutex);
+
+ /* Do a second connection while the first one is blocked */
+ do_test_play (NULL);
+
+ /* Now unblock the describe */
+ g_mutex_lock (&check_mutex);
+ block_state = UNBLOCK;
+ g_cond_broadcast (&check_cond);
+ g_mutex_unlock (&check_mutex);
+
+ response = read_response (conn);
+ gst_rtsp_message_parse_response (response, &code, NULL, NULL);
+ fail_unless (code == GST_RTSP_STS_OK);
+ gst_rtsp_message_free (response);
+
+
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+
+}
+
+GST_END_TEST;
+
+
+static void
+new_session_timeout_one (GstRTSPClient * client,
+ GstRTSPSession * session, gpointer user_data)
+{
+ gst_rtsp_session_set_timeout (session, 1);
+
+ g_signal_handlers_disconnect_by_func (client, new_session_timeout_one,
+ user_data);
+}
+
+static void
+session_connected_new_session_cb (GstRTSPServer * server,
+ GstRTSPClient * client, gpointer user_data)
+{
+
+ g_signal_connect (client, "new-session", user_data, NULL);
+}
+
+GST_START_TEST (test_play_multithreaded_timeout_client)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GstRTSPSessionPool *pool;
+ GstRTSPThreadPool *thread_pool;
+
+ thread_pool = gst_rtsp_server_get_thread_pool (server);
+ gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
+ g_object_unref (thread_pool);
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ g_signal_connect (server, "client-connected",
+ G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
+
+ start_server (FALSE);
+
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_port);
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup_full (conn, video_control, GST_RTSP_LOWER_TRANS_UDP,
+ &client_port, NULL, &session, &video_transport,
+ NULL) == GST_RTSP_STS_OK);
+ fail_unless (do_setup_full (conn, audio_control, GST_RTSP_LOWER_TRANS_UDP,
+ &client_port, NULL, &session, &audio_transport,
+ NULL) == GST_RTSP_STS_OK);
+
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session) == GST_RTSP_STS_OK);
+
+ sleep (7);
+
+ fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 0);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (pool);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_multithreaded_timeout_session)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session1 = NULL;
+ gchar *session2 = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GstRTSPSessionPool *pool;
+ GstRTSPThreadPool *thread_pool;
+
+ thread_pool = gst_rtsp_server_get_thread_pool (server);
+ gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
+ g_object_unref (thread_pool);
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ g_signal_connect (server, "client-connected",
+ G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
+
+ start_server (FALSE);
+
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ gst_rtsp_connection_set_remember_session_id (conn, FALSE);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_port);
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup (conn, video_control, &client_port, &session1,
+ &video_transport) == GST_RTSP_STS_OK);
+ fail_unless (do_setup (conn, audio_control, &client_port, &session2,
+ &audio_transport) == GST_RTSP_STS_OK);
+
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 2);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session1) == GST_RTSP_STS_OK);
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session2) == GST_RTSP_STS_OK);
+
+ sleep (7);
+
+ fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
+
+ /* send TEARDOWN request and check that we get 454 Session Not found */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session1) == GST_RTSP_STS_SESSION_NOT_FOUND);
+
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session2) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (pool);
+ g_free (session1);
+ g_free (session2);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+static void
+new_connection_and_session_timeout_one (GstRTSPClient * client,
+ GstRTSPSession * session, gpointer user_data)
+{
+ gint ps_timeout = 0;
+
+ g_object_set (G_OBJECT (client), "post-session-timeout", 1, NULL);
+ g_object_get (G_OBJECT (client), "post-session-timeout", &ps_timeout, NULL);
+ fail_unless_equals_int (ps_timeout, 1);
+
+ g_object_set (G_OBJECT (session), "extra-timeout", 0, NULL);
+ gst_rtsp_session_set_timeout (session, 1);
+
+ g_signal_handlers_disconnect_by_func (client,
+ new_connection_and_session_timeout_one, user_data);
+}
+
+GST_START_TEST (test_play_timeout_connection)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPSessionPool *pool;
+ GstRTSPThreadPool *thread_pool;
+ GstRTSPMessage *request;
+ GstRTSPMessage *response;
+
+ thread_pool = gst_rtsp_server_get_thread_pool (server);
+ g_object_unref (thread_pool);
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ g_signal_connect (server, "client-connected",
+ G_CALLBACK (session_connected_new_session_cb),
+ new_connection_and_session_timeout_one);
+
+ start_server (FALSE);
+
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ gst_rtsp_connection_set_remember_session_id (conn, FALSE);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_port);
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup (conn, video_control, &client_port, &session,
+ &video_transport) == GST_RTSP_STS_OK);
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session) == GST_RTSP_STS_OK);
+ sleep (2);
+ fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
+ sleep (3);
+
+ request = create_request (conn, GST_RTSP_TEARDOWN, NULL);
+
+ /* add headers */
+ if (session) {
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session);
+ }
+
+ /* send request */
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+
+ iterate ();
+
+ /* read response */
+ response = read_response (conn);
+ fail_unless (response == NULL);
+
+ if (response) {
+ gst_rtsp_message_free (response);
+ }
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (pool);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_no_session_timeout)
+{
+ GstRTSPSession *session;
+ gint64 now;
+ gboolean is_expired;
+
+ session = gst_rtsp_session_new ("test-session");
+ gst_rtsp_session_set_timeout (session, 0);
+
+ now = g_get_monotonic_time ();
+ /* add more than the extra 5 seconds that are usually added in
+ * gst_rtsp_session_next_timeout_usec */
+ now += 7000000;
+
+ is_expired = gst_rtsp_session_is_expired_usec (session, now);
+ fail_unless (is_expired == FALSE);
+
+ g_object_unref (session);
+}
+
+GST_END_TEST;
+
+/* media contains two streams: video and audio but only one
+ * stream is requested */
+GST_START_TEST (test_play_one_active_stream)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPSessionPool *pool;
+ GstRTSPThreadPool *thread_pool;
+
+ thread_pool = gst_rtsp_server_get_thread_pool (server);
+ gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
+ g_object_unref (thread_pool);
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ g_signal_connect (server, "client-connected",
+ G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
+
+ start_server (FALSE);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ gst_rtsp_connection_set_remember_session_id (conn, FALSE);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_port);
+
+ /* do SETUP for video only */
+ fail_unless (do_setup (conn, video_control, &client_port, &session,
+ &video_transport) == GST_RTSP_STS_OK);
+
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session) == GST_RTSP_STS_OK);
+
+
+ /* send TEARDOWN request */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (pool);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_play_disconnect)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ g_signal_connect (server, "client-connected",
+ G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
+
+ start_server (FALSE);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_port);
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup (conn, video_control, &client_port, &session,
+ &video_transport) == GST_RTSP_STS_OK);
+ fail_unless (do_setup (conn, audio_control, &client_port, &session,
+ &audio_transport) == GST_RTSP_STS_OK);
+
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session) == GST_RTSP_STS_OK);
+
+ gst_rtsp_connection_free (conn);
+
+ sleep (7);
+
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+ fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
+
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (pool);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+/* Only different with test_play is the specific ports selected */
+
+GST_START_TEST (test_play_specific_server_port)
+{
+ GstRTSPMountPoints *mounts;
+ gchar *service;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAddressPool *pool;
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GSocket *rtp_socket, *rtcp_socket;
+ GSocketAddress *rtp_address, *rtcp_address;
+ guint16 rtp_port, rtcp_port;
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = gst_rtsp_media_factory_new ();
+ /* we have to suspend media after SDP in order to make sure that
+ * we can reconfigure UDP sink with new UDP ports */
+ gst_rtsp_media_factory_set_suspend_mode (factory,
+ GST_RTSP_SUSPEND_MODE_RESET);
+ pool = gst_rtsp_address_pool_new ();
+ gst_rtsp_address_pool_add_range (pool, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
+ GST_RTSP_ADDRESS_POOL_ANY_IPV4, 7770, 7780, 0);
+ gst_rtsp_media_factory_set_address_pool (factory, pool);
+ g_object_unref (pool);
+ gst_rtsp_media_factory_set_launch (factory, "( " VIDEO_PIPELINE " )");
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+ g_object_unref (mounts);
+
+ /* set port to any */
+ gst_rtsp_server_set_service (server, "0");
+
+ /* attach to default main context */
+ source_id = gst_rtsp_server_attach (server, NULL);
+ fail_if (source_id == 0);
+
+ /* get port */
+ service = gst_rtsp_server_get_service (server);
+ test_port = atoi (service);
+ fail_unless (test_port != 0);
+ g_free (service);
+
+ GST_DEBUG ("rtsp server listening on port %d", test_port);
+
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 1);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
+
+ /* do SETUP for video */
+ fail_unless (do_setup (conn, video_control, &client_port, &session,
+ &video_transport) == GST_RTSP_STS_OK);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session) == GST_RTSP_STS_OK);
+
+ receive_rtp (rtp_socket, &rtp_address);
+ receive_rtcp (rtcp_socket, &rtcp_address, 0);
+
+ fail_unless (G_IS_INET_SOCKET_ADDRESS (rtp_address));
+ fail_unless (G_IS_INET_SOCKET_ADDRESS (rtcp_address));
+ rtp_port =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_address));
+ rtcp_port =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtcp_address));
+ fail_unless (rtp_port >= 7770 && rtp_port <= 7780 && rtp_port % 2 == 0);
+ fail_unless (rtcp_port >= 7770 && rtcp_port <= 7780 && rtcp_port % 2 == 1);
+ fail_unless (rtp_port + 1 == rtcp_port);
+
+ g_object_unref (rtp_address);
+ g_object_unref (rtcp_address);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* FIXME: The rtsp-server always disconnects the transport before
+ * sending the RTCP BYE
+ * receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
+ */
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (rtp_socket);
+ g_object_unref (rtcp_socket);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_smpte_range)
+{
+ start_server (FALSE);
+
+ do_test_play ("npt=5-");
+ do_test_play ("smpte=0:00:00-");
+ do_test_play ("smpte=1:00:00-");
+ do_test_play ("smpte=1:00:03-");
+ do_test_play ("clock=20120321T152256Z-");
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_play_smpte_range_tcp)
+{
+ start_tcp_server (FALSE);
+
+ do_test_play_tcp_full ("npt=5-");
+ do_test_play_tcp_full ("smpte=0:00:00-");
+ do_test_play_tcp_full ("smpte=1:00:00-");
+ do_test_play_tcp_full ("smpte=1:00:03-");
+ do_test_play_tcp_full ("clock=20120321T152256Z-");
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+static gpointer
+thread_func_udp (gpointer data)
+{
+ do_test_play_full (NULL, GST_RTSP_LOWER_TRANS_UDP, (GMutex *) data);
+ return NULL;
+}
+
+static gpointer
+thread_func_tcp (gpointer data)
+{
+ do_test_play_tcp_full (NULL);
+ return NULL;
+}
+
+static void
+test_shared (gpointer (thread_func) (gpointer data))
+{
+ GMutex lock1, lock2, lock3, lock4;
+ GThread *thread1, *thread2, *thread3, *thread4;
+
+ /* Locks for each thread. Each thread will keep reading data as long as the
+ * thread is locked. */
+ g_mutex_init (&lock1);
+ g_mutex_init (&lock2);
+ g_mutex_init (&lock3);
+ g_mutex_init (&lock4);
+
+ if (thread_func == thread_func_tcp)
+ start_tcp_server (TRUE);
+ else
+ start_server (TRUE);
+
+ /* Start the first receiver thread. */
+ g_mutex_lock (&lock1);
+ thread1 = g_thread_new ("thread1", thread_func, &lock1);
+
+ /* Connect and disconnect another client. */
+ g_mutex_lock (&lock2);
+ thread2 = g_thread_new ("thread2", thread_func, &lock2);
+ g_mutex_unlock (&lock2);
+ g_mutex_clear (&lock2);
+ g_thread_join (thread2);
+
+ /* Do it again. */
+ g_mutex_lock (&lock3);
+ thread3 = g_thread_new ("thread3", thread_func, &lock3);
+ g_mutex_unlock (&lock3);
+ g_mutex_clear (&lock3);
+ g_thread_join (thread3);
+
+ /* Disconnect the last client. This will clean up the media. */
+ g_mutex_unlock (&lock1);
+ g_mutex_clear (&lock1);
+ g_thread_join (thread1);
+
+ /* Connect and disconnect another client. This will create and clean up the
+ * media. */
+ g_mutex_lock (&lock4);
+ thread4 = g_thread_new ("thread4", thread_func, &lock4);
+ g_mutex_unlock (&lock4);
+ g_mutex_clear (&lock4);
+ g_thread_join (thread4);
+
+ stop_server ();
+ iterate ();
+}
+
+/* Test adding and removing clients to a 'Shared' media.
+ * CASE: unicast UDP */
+GST_START_TEST (test_shared_udp)
+{
+ test_shared (thread_func_udp);
+}
+
+GST_END_TEST;
+
+/* Test adding and removing clients to a 'Shared' media.
+ * CASE: unicast TCP */
+GST_START_TEST (test_shared_tcp)
+{
+ test_shared (thread_func_tcp);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_announce_without_sdp)
+{
+ GstRTSPConnection *conn;
+ GstRTSPStatusCode status;
+ GstRTSPMessage *request;
+ GstRTSPMessage *response;
+
+ start_record_server ("( fakesink name=depay0 )");
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* create and send ANNOUNCE request */
+ request = create_request (conn, GST_RTSP_ANNOUNCE, NULL);
+
+ fail_unless (send_request (conn, request));
+
+ iterate ();
+
+ response = read_response (conn);
+
+ /* check response */
+ gst_rtsp_message_parse_response (response, &status, NULL, NULL);
+ fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
+ gst_rtsp_message_free (response);
+
+ /* try again, this type with content-type, but still no SDP */
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/sdp");
+
+ fail_unless (send_request (conn, request));
+
+ iterate ();
+
+ response = read_response (conn);
+
+ /* check response */
+ gst_rtsp_message_parse_response (response, &status, NULL, NULL);
+ fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
+ gst_rtsp_message_free (response);
+
+ /* try again, this type with an unknown content-type */
+ gst_rtsp_message_remove_header (request, GST_RTSP_HDR_CONTENT_TYPE, -1);
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/x-something");
+
+ fail_unless (send_request (conn, request));
+
+ iterate ();
+
+ response = read_response (conn);
+
+ /* check response */
+ gst_rtsp_message_parse_response (response, &status, NULL, NULL);
+ fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
+ gst_rtsp_message_free (response);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_message_free (request);
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+static GstRTSPStatusCode
+do_announce (GstRTSPConnection * conn, GstSDPMessage * sdp)
+{
+ GstRTSPMessage *request;
+ GstRTSPMessage *response;
+ GstRTSPStatusCode code;
+ gchar *str;
+
+ /* create request */
+ request = create_request (conn, GST_RTSP_ANNOUNCE, NULL);
+
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/sdp");
+
+ /* add SDP to the response body */
+ str = gst_sdp_message_as_text (sdp);
+ gst_rtsp_message_take_body (request, (guint8 *) str, strlen (str));
+ gst_sdp_message_free (sdp);
+
+ /* send request */
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+
+ iterate ();
+
+ /* read response */
+ response = read_response (conn);
+
+ /* check status line */
+ gst_rtsp_message_parse_response (response, &code, NULL, NULL);
+
+ gst_rtsp_message_free (response);
+ return code;
+}
+
+static void
+media_constructed_cb (GstRTSPMediaFactory * mfactory, GstRTSPMedia * media,
+ gpointer user_data)
+{
+ GstElement **p_sink = user_data;
+ GstElement *bin;
+
+ bin = gst_rtsp_media_get_element (media);
+ *p_sink = gst_bin_get_by_name (GST_BIN (bin), "sink");
+ GST_INFO ("media constructed!: %" GST_PTR_FORMAT, *p_sink);
+ gst_object_unref (bin);
+}
+
+#define RECORD_N_BUFS 10
+
+GST_START_TEST (test_record_tcp)
+{
+ GstRTSPMediaFactory *mfactory;
+ GstRTSPConnection *conn;
+ GstRTSPStatusCode status;
+ GstRTSPMessage *response;
+ GstRTSPMessage *request;
+ GstSDPMessage *sdp;
+ GstRTSPResult rres;
+ GSocketAddress *sa;
+ GInetAddress *ia;
+ GstElement *server_sink = NULL;
+ GSocket *conn_socket;
+ const gchar *proto;
+ gchar *client_ip, *sess_id, *session = NULL;
+ gint i;
+
+ mfactory =
+ start_record_server
+ ("( rtppcmadepay name=depay0 ! appsink name=sink async=false )");
+
+ g_signal_connect (mfactory, "media-constructed",
+ G_CALLBACK (media_constructed_cb), &server_sink);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ conn_socket = gst_rtsp_connection_get_read_socket (conn);
+
+ sa = g_socket_get_local_address (conn_socket, NULL);
+ ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
+ client_ip = g_inet_address_to_string (ia);
+ if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV6)
+ proto = "IP6";
+ else if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV4)
+ proto = "IP4";
+ else
+ g_assert_not_reached ();
+ g_object_unref (sa);
+
+ gst_sdp_message_new (&sdp);
+
+ /* some standard things first */
+ gst_sdp_message_set_version (sdp, "0");
+
+ /* session ID doesn't have to be super-unique in this case */
+ sess_id = g_strdup_printf ("%u", g_random_int ());
+ gst_sdp_message_set_origin (sdp, "-", sess_id, "1", "IN", proto, client_ip);
+ g_free (sess_id);
+ g_free (client_ip);
+
+ gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
+ gst_sdp_message_set_information (sdp, "rtsp-server-test");
+ gst_sdp_message_add_time (sdp, "0", "0", NULL);
+ gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
+
+ /* add stream 0 */
+ {
+ GstSDPMedia *smedia;
+
+ gst_sdp_media_new (&smedia);
+ gst_sdp_media_set_media (smedia, "audio");
+ gst_sdp_media_add_format (smedia, "8"); /* pcma/alaw */
+ gst_sdp_media_set_port_info (smedia, 0, 1);
+ gst_sdp_media_set_proto (smedia, "RTP/AVP");
+ gst_sdp_media_add_attribute (smedia, "rtpmap", "8 PCMA/8000");
+ gst_sdp_message_add_media (sdp, smedia);
+ gst_sdp_media_free (smedia);
+ }
+
+ /* send ANNOUNCE request */
+ status = do_announce (conn, sdp);
+ fail_unless_equals_int (status, GST_RTSP_STS_OK);
+
+ /* create and send SETUP request */
+ request = create_request (conn, GST_RTSP_SETUP, NULL);
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP/TCP;interleaved=0;mode=record");
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+ iterate ();
+ response = read_response (conn);
+ gst_rtsp_message_parse_response (response, &status, NULL, NULL);
+ fail_unless_equals_int (status, GST_RTSP_STS_OK);
+
+ rres =
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &session, 0);
+ session = g_strdup (session);
+ fail_unless_equals_int (rres, GST_RTSP_OK);
+ gst_rtsp_message_free (response);
+
+ /* send RECORD */
+ request = create_request (conn, GST_RTSP_RECORD, NULL);
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session);
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+ iterate ();
+ response = read_response (conn);
+ gst_rtsp_message_parse_response (response, &status, NULL, NULL);
+ fail_unless_equals_int (status, GST_RTSP_STS_OK);
+ gst_rtsp_message_free (response);
+
+ /* send some data */
+ {
+ GstElement *pipeline, *src, *enc, *pay, *sink;
+
+ pipeline = gst_pipeline_new ("send-pipeline");
+ src = gst_element_factory_make ("audiotestsrc", NULL);
+ g_object_set (src, "num-buffers", RECORD_N_BUFS,
+ "samplesperbuffer", 1000, NULL);
+ enc = gst_element_factory_make ("alawenc", NULL);
+ pay = gst_element_factory_make ("rtppcmapay", NULL);
+ sink = gst_element_factory_make ("appsink", NULL);
+ fail_unless (pipeline && src && enc && pay && sink);
+ gst_bin_add_many (GST_BIN (pipeline), src, enc, pay, sink, NULL);
+ gst_element_link_many (src, enc, pay, sink, NULL);
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+ do {
+ GstRTSPMessage *data_msg;
+ GstMapInfo map = GST_MAP_INFO_INIT;
+ GstRTSPResult rres;
+ GstSample *sample = NULL;
+ GstBuffer *buf;
+
+ g_signal_emit_by_name (G_OBJECT (sink), "pull-sample", &sample);
+ if (sample == NULL)
+ break;
+ buf = gst_sample_get_buffer (sample);
+ rres = gst_rtsp_message_new_data (&data_msg, 0);
+ fail_unless_equals_int (rres, GST_RTSP_OK);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ GST_INFO ("sending %u bytes of data on channel 0", (guint) map.size);
+ GST_MEMDUMP ("data on channel 0", map.data, map.size);
+ rres = gst_rtsp_message_set_body (data_msg, map.data, map.size);
+ fail_unless_equals_int (rres, GST_RTSP_OK);
+ gst_buffer_unmap (buf, &map);
+ rres = gst_rtsp_connection_send (conn, data_msg, NULL);
+ fail_unless_equals_int (rres, GST_RTSP_OK);
+ gst_rtsp_message_free (data_msg);
+ gst_sample_unref (sample);
+ } while (TRUE);
+
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+ }
+
+ /* check received data (we assume every buffer created by audiotestsrc and
+ * subsequently encoded by mulawenc results in exactly one RTP packet) */
+ for (i = 0; i < RECORD_N_BUFS; ++i) {
+ GstSample *sample = NULL;
+
+ g_signal_emit_by_name (G_OBJECT (server_sink), "pull-sample", &sample);
+ GST_INFO ("%2d recv sample: %p", i, sample);
+ gst_sample_unref (sample);
+ }
+
+ fail_unless_equals_int (GST_STATE (server_sink), GST_STATE_PLAYING);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+ g_free (session);
+}
+
+GST_END_TEST;
+
+static void
+do_test_multiple_transports (GstRTSPLowerTrans trans1, GstRTSPLowerTrans trans2)
+{
+ GstRTSPConnection *conn1;
+ GstRTSPConnection *conn2;
+ GstSDPMessage *sdp_message1 = NULL;
+ GstSDPMessage *sdp_message2 = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port1, client_port2;
+ gchar *session1 = NULL;
+ gchar *session2 = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GSocket *rtp_socket, *rtcp_socket;
+
+ conn1 = connect_to_server (test_port, TEST_MOUNT_POINT);
+ conn2 = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message1 = do_describe (conn1, TEST_MOUNT_POINT);
+
+ get_client_ports_full (&client_port1, &rtp_socket, &rtcp_socket);
+ /* get control strings from DESCRIBE response */
+ sdp_media = gst_sdp_message_get_media (sdp_message1, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message1, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup_full (conn1, video_control, trans1,
+ &client_port1, NULL, &session1, &video_transport,
+ NULL) == GST_RTSP_STS_OK);
+ fail_unless (do_setup_full (conn1, audio_control, trans1,
+ &client_port1, NULL, &session1, &audio_transport,
+ NULL) == GST_RTSP_STS_OK);
+
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+
+ sdp_message2 = do_describe (conn2, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ sdp_media = gst_sdp_message_get_media (sdp_message2, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message2, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports_full (&client_port2, NULL, NULL);
+ /* do SETUP for video and audio */
+ fail_unless (do_setup_full (conn2, video_control, trans2,
+ &client_port2, NULL, &session2, &video_transport,
+ NULL) == GST_RTSP_STS_OK);
+ fail_unless (do_setup_full (conn2, audio_control, trans2,
+ &client_port2, NULL, &session2, &audio_transport,
+ NULL) == GST_RTSP_STS_OK);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_request (conn1, GST_RTSP_PLAY, NULL, session1, NULL, NULL,
+ NULL, NULL, NULL, NULL, NULL, NULL) == GST_RTSP_STS_OK);
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_request (conn2, GST_RTSP_PLAY, NULL, session2, NULL, NULL,
+ NULL, NULL, NULL, NULL, NULL, NULL) == GST_RTSP_STS_OK);
+
+
+ /* receive UDP data */
+ receive_rtp (rtp_socket, NULL);
+ receive_rtcp (rtcp_socket, NULL, 0);
+
+ /* receive TCP data */
+ {
+ GstRTSPMessage *message;
+ fail_unless (gst_rtsp_message_new (&message) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_connection_receive (conn2, message,
+ NULL) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_message_get_type (message) == GST_RTSP_MESSAGE_DATA);
+ gst_rtsp_message_free (message);
+ }
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn1, GST_RTSP_TEARDOWN,
+ session1) == GST_RTSP_STS_OK);
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn2, GST_RTSP_TEARDOWN,
+ session2) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (rtp_socket);
+ g_object_unref (rtcp_socket);
+ g_free (session1);
+ g_free (session2);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message1);
+ gst_sdp_message_free (sdp_message2);
+ gst_rtsp_connection_free (conn1);
+ gst_rtsp_connection_free (conn2);
+}
+
+GST_START_TEST (test_multiple_transports)
+{
+ start_server (TRUE);
+ do_test_multiple_transports (GST_RTSP_LOWER_TRANS_UDP,
+ GST_RTSP_LOWER_TRANS_TCP);
+ stop_server ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_suspend_mode_reset_only_audio)
+{
+ GstRTSPMountPoints *mounts;
+ gchar *service;
+ GstRTSPMediaFactory *factory;
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GSocket *rtp_socket, *rtcp_socket;
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_suspend_mode (factory,
+ GST_RTSP_SUSPEND_MODE_RESET);
+ gst_rtsp_media_factory_set_launch (factory,
+ "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+ g_object_unref (mounts);
+
+ /* set port to any */
+ gst_rtsp_server_set_service (server, "0");
+
+ /* attach to default main context */
+ source_id = gst_rtsp_server_attach (server, NULL);
+ fail_if (source_id == 0);
+
+ /* get port */
+ service = gst_rtsp_server_get_service (server);
+ test_port = atoi (service);
+ fail_unless (test_port != 0);
+ g_free (service);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
+
+ /* do SETUP for audio */
+ fail_unless (do_setup (conn, audio_control, &client_port, &session,
+ &audio_transport) == GST_RTSP_STS_OK);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session) == GST_RTSP_STS_OK);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_free (session);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+
+static GstRTSPStatusCode
+adjust_play_mode (GstRTSPClient * client, GstRTSPContext * ctx,
+ GstRTSPTimeRange ** range, GstSeekFlags * flags, gdouble * rate,
+ GstClockTime * trickmode_interval, gboolean * enable_rate_control)
+{
+ GstRTSPState rtspstate;
+
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (ctx->sessmedia);
+ if (rtspstate == GST_RTSP_STATE_PLAYING) {
+ if (!gst_rtsp_session_media_set_state (ctx->sessmedia, GST_STATE_PAUSED))
+ return GST_RTSP_STS_INTERNAL_SERVER_ERROR;
+
+ if (!gst_rtsp_media_unsuspend (ctx->media))
+ return GST_RTSP_STS_INTERNAL_SERVER_ERROR;
+ }
+
+ return GST_RTSP_STS_OK;
+}
+
+GST_START_TEST (test_double_play)
+{
+ GstRTSPMountPoints *mounts;
+ gchar *service;
+ GstRTSPMediaFactory *factory;
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GSocket *rtp_socket, *rtcp_socket;
+ GstRTSPClient *client;
+ GstRTSPClientClass *klass;
+
+ client = gst_rtsp_client_new ();
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ klass->adjust_play_mode = adjust_play_mode;
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+ g_object_unref (mounts);
+
+
+ /* set port to any */
+ gst_rtsp_server_set_service (server, "0");
+
+ /* attach to default main context */
+ source_id = gst_rtsp_server_attach (server, NULL);
+ fail_if (source_id == 0);
+
+ /* get port */
+ service = gst_rtsp_server_get_service (server);
+ test_port = atoi (service);
+ fail_unless (test_port != 0);
+ g_free (service);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
+
+ /* do SETUP for video */
+ fail_unless (do_setup (conn, video_control, &client_port, &session,
+ &video_transport) == GST_RTSP_STS_OK);
+
+ /* do SETUP for audio */
+ fail_unless (do_setup (conn, audio_control, &client_port, &session,
+ &audio_transport) == GST_RTSP_STS_OK);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request_rangein (conn, GST_RTSP_PLAY,
+ session, "npt=0-") == GST_RTSP_STS_OK);
+
+ /* let it play for a while, so it needs to seek
+ * for next play (npt=0-) */
+ g_usleep (30000);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request_rangein (conn, GST_RTSP_PLAY,
+ session, "npt=0-") == GST_RTSP_STS_OK);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (rtp_socket);
+ g_object_unref (rtcp_socket);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+
+static Suite *
+rtspserver_suite (void)
+{
+ Suite *s = suite_create ("rtspserver");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_add_checked_fixture (tc, setup, teardown);
+ tcase_set_timeout (tc, 120);
+ tcase_add_test (tc, test_connect);
+ tcase_add_test (tc, test_describe);
+ tcase_add_test (tc, test_describe_non_existing_mount_point);
+ tcase_add_test (tc, test_describe_record_media);
+ tcase_add_test (tc, test_setup_udp);
+ tcase_add_test (tc, test_setup_tcp);
+ tcase_add_test (tc, test_setup_udp_mcast);
+ tcase_add_test (tc, test_setup_twice);
+ tcase_add_test (tc, test_setup_with_require_header);
+ tcase_add_test (tc, test_setup_non_existing_stream);
+ tcase_add_test (tc, test_play);
+ tcase_add_test (tc, test_play_tcp);
+ tcase_add_test (tc, test_play_without_session);
+ tcase_add_test (tc, test_bind_already_in_use);
+ tcase_add_test (tc, test_play_multithreaded);
+ tcase_add_test (tc, test_play_multithreaded_block_in_describe);
+ tcase_add_test (tc, test_play_multithreaded_timeout_client);
+ tcase_add_test (tc, test_play_multithreaded_timeout_session);
+ tcase_add_test (tc, test_play_timeout_connection);
+ tcase_add_test (tc, test_no_session_timeout);
+ tcase_add_test (tc, test_play_one_active_stream);
+ tcase_add_test (tc, test_play_disconnect);
+ tcase_add_test (tc, test_play_specific_server_port);
+ tcase_add_test (tc, test_play_smpte_range);
+ tcase_add_test (tc, test_play_smpte_range_tcp);
+ tcase_add_test (tc, test_shared_udp);
+ tcase_add_test (tc, test_shared_tcp);
+ tcase_add_test (tc, test_announce_without_sdp);
+ tcase_add_test (tc, test_record_tcp);
+ tcase_add_test (tc, test_multiple_transports);
+ tcase_add_test (tc, test_suspend_mode_reset_only_audio);
+ tcase_add_test (tc, test_double_play);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspserver);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Branko Subasic <branko.subasic@axis.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-media-factory.h>
+#include <rtsp-session-media.h>
+
+#define TEST_PATH "rtsp://localhost:8554/test"
+#define SETUP_URL1 TEST_PATH "/stream=0"
+#define SETUP_URL2 TEST_PATH "/stream=1"
+
+GST_START_TEST (test_setup_url)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url, *setup_url;
+ GstRTSPStream *stream;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPSessionMedia *sm;
+ GstRTSPStreamTransport *trans;
+ GstRTSPTransport *ct;
+ gint match_len;
+ gchar *url_str, *url_str2;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (GST_IS_RTSP_STREAM (stream));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ /* create session media and make sure it matches test path
+ * note that gst_rtsp_session_media_new takes ownership of the media
+ * thus no need to unref it at the bottom of function */
+ sm = gst_rtsp_session_media_new (TEST_PATH, media);
+ fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+ fail_unless (gst_rtsp_session_media_matches (sm, TEST_PATH, &match_len));
+ fail_unless (match_len == strlen (TEST_PATH));
+ fail_unless (gst_rtsp_session_media_get_media (sm) == media);
+
+ /* make a transport for the stream */
+ gst_rtsp_transport_new (&ct);
+ trans = gst_rtsp_session_media_set_transport (sm, stream, ct);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+
+ /* make sure there's no setup url stored initially */
+ fail_unless (gst_rtsp_stream_transport_get_url (trans) == NULL);
+
+ /* now store a setup url and make sure it can be retrieved and that it's correct */
+ fail_unless (gst_rtsp_url_parse (SETUP_URL1, &setup_url) == GST_RTSP_OK);
+ gst_rtsp_stream_transport_set_url (trans, setup_url);
+
+ url_str = gst_rtsp_url_get_request_uri (setup_url);
+ url_str2 =
+ gst_rtsp_url_get_request_uri (gst_rtsp_stream_transport_get_url (trans));
+ fail_if (g_strcmp0 (url_str, url_str2) != 0);
+ g_free (url_str);
+ g_free (url_str2);
+
+ /* check that it's ok to try to store the same url again */
+ gst_rtsp_stream_transport_set_url (trans, setup_url);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ gst_rtsp_url_free (setup_url);
+ gst_rtsp_url_free (url);
+
+ g_object_unref (sm);
+
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_rtsp_state)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPSessionMedia *sm;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (GST_IS_RTSP_STREAM (stream));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ sm = gst_rtsp_session_media_new (TEST_PATH, media);
+ fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+ fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+ GST_RTSP_STATE_INIT);
+
+ gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_READY);
+ fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+ GST_RTSP_STATE_READY);
+
+ gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_SEEKING);
+ fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+ GST_RTSP_STATE_SEEKING);
+
+ gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_PLAYING);
+ fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+ GST_RTSP_STATE_PLAYING);
+
+ gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_RECORDING);
+ fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+ GST_RTSP_STATE_RECORDING);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ gst_rtsp_url_free (url);
+
+ g_object_unref (sm);
+
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_transports)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream1, *stream2;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPSessionMedia *sm;
+ GstRTSPStreamTransport *trans;
+ GstRTSPTransport *ct1, *ct2, *ct3, *ct4;
+ gint match_len;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 audiotestsrc ! rtpgstpay pt=97 name=pay1 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+ stream1 = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (GST_IS_RTSP_STREAM (stream1));
+
+ stream2 = gst_rtsp_media_get_stream (media, 1);
+ fail_unless (GST_IS_RTSP_STREAM (stream2));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ sm = gst_rtsp_session_media_new (TEST_PATH, media);
+ fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+ fail_unless (gst_rtsp_session_media_matches (sm, TEST_PATH, &match_len));
+ fail_unless (match_len == strlen (TEST_PATH));
+
+ gst_rtsp_transport_new (&ct1);
+ trans = gst_rtsp_session_media_set_transport (sm, stream1, ct1);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+
+ gst_rtsp_transport_new (&ct2);
+ trans = gst_rtsp_session_media_set_transport (sm, stream1, ct2);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+
+ gst_rtsp_transport_new (&ct3);
+ trans = gst_rtsp_session_media_set_transport (sm, stream2, ct3);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 1) == trans);
+
+ gst_rtsp_transport_new (&ct4);
+ trans = gst_rtsp_session_media_set_transport (sm, stream2, ct4);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 1) == trans);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ gst_rtsp_url_free (url);
+
+ g_object_unref (sm);
+
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_time_and_rtpinfo)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream1, *stream2;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPSessionMedia *sm;
+ GstClockTime base_time;
+ gchar *rtpinfo;
+ GstRTSPTransport *ct1;
+ GstRTSPStreamTransport *trans;
+ GstRTSPUrl *setup_url;
+ gchar **streaminfo;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc do-timestamp=true timestamp-offset=0 ! rtpvrawpay pt=96 name=pay0 "
+ "audiotestsrc do-timestamp=true timestamp-offset=1000000000 ! rtpgstpay pt=97 name=pay1 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+ stream1 = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (GST_IS_RTSP_STREAM (stream1));
+
+ stream2 = gst_rtsp_media_get_stream (media, 1);
+ fail_unless (GST_IS_RTSP_STREAM (stream2));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ sm = gst_rtsp_session_media_new (TEST_PATH, media);
+ fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+
+ base_time = gst_rtsp_session_media_get_base_time (sm);
+ fail_unless_equals_int64 (base_time, 0);
+
+ rtpinfo = gst_rtsp_session_media_get_rtpinfo (sm);
+ fail_unless (rtpinfo == NULL);
+
+ gst_rtsp_transport_new (&ct1);
+ trans = gst_rtsp_session_media_set_transport (sm, stream1, ct1);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+ fail_unless (gst_rtsp_url_parse (SETUP_URL1, &setup_url) == GST_RTSP_OK);
+ gst_rtsp_stream_transport_set_url (trans, setup_url);
+
+ base_time = gst_rtsp_session_media_get_base_time (sm);
+ fail_unless_equals_int64 (base_time, 0);
+
+ rtpinfo = gst_rtsp_session_media_get_rtpinfo (sm);
+ streaminfo = g_strsplit (rtpinfo, ",", 1);
+ g_free (rtpinfo);
+
+ fail_unless (g_strstr_len (streaminfo[0], -1, "url=") != NULL);
+ fail_unless (g_strstr_len (streaminfo[0], -1, "seq=") != NULL);
+ fail_unless (g_strstr_len (streaminfo[0], -1, "rtptime=") != NULL);
+ fail_unless (g_strstr_len (streaminfo[0], -1, SETUP_URL1) != NULL);
+
+ g_strfreev (streaminfo);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ rtpinfo = gst_rtsp_session_media_get_rtpinfo (sm);
+ fail_unless (rtpinfo == NULL);
+
+ gst_rtsp_url_free (setup_url);
+ gst_rtsp_url_free (url);
+
+ g_object_unref (sm);
+
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_allocate_channels)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPSessionMedia *sm;
+ GstRTSPRange range;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (GST_IS_RTSP_STREAM (stream));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ sm = gst_rtsp_session_media_new (TEST_PATH, media);
+ fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+
+ fail_unless (gst_rtsp_session_media_alloc_channels (sm, &range));
+ fail_unless_equals_int (range.min, 0);
+ fail_unless_equals_int (range.max, 1);
+
+ fail_unless (gst_rtsp_session_media_alloc_channels (sm, &range));
+ fail_unless_equals_int (range.min, 2);
+ fail_unless_equals_int (range.max, 3);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ gst_rtsp_url_free (url);
+
+ g_object_unref (sm);
+
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+static Suite *
+rtspsessionmedia_suite (void)
+{
+ Suite *s = suite_create ("rtspsessionmedia");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_setup_url);
+ tcase_add_test (tc, test_rtsp_state);
+ tcase_add_test (tc, test_transports);
+ tcase_add_test (tc, test_time_and_rtpinfo);
+ tcase_add_test (tc, test_allocate_channels);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspsessionmedia);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2014 Sebastian Rasmussen <sebras@hotmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <rtsp-session-pool.h>
+
+typedef struct
+{
+ GstRTSPSession *sessions[3];
+ GstRTSPFilterResult response[3];
+} Responses;
+
+static GstRTSPFilterResult
+filter_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
+ gpointer user_data)
+{
+ Responses *responses = (Responses *) user_data;
+ gint i;
+
+ for (i = 0; i < 3; i++)
+ if (session == responses->sessions[i])
+ return responses->response[i];
+
+ return GST_RTSP_FILTER_KEEP;
+}
+
+GST_START_TEST (test_pool)
+{
+ GstRTSPSessionPool *pool;
+ GstRTSPSession *session1, *session2, *session3;
+ GstRTSPSession *compare;
+ gchar *session1id, *session2id, *session3id;
+ GList *list;
+ guint maxsessions;
+ GSource *source;
+ guint sourceid;
+
+ pool = gst_rtsp_session_pool_new ();
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 0);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 0);
+
+ gst_rtsp_session_pool_set_max_sessions (pool, 3);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+
+ session1 = gst_rtsp_session_pool_create (pool);
+ fail_unless (GST_IS_RTSP_SESSION (session1));
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 1);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+ session1id = g_strdup (gst_rtsp_session_get_sessionid (session1));
+
+ session2 = gst_rtsp_session_pool_create (pool);
+ fail_unless (GST_IS_RTSP_SESSION (session2));
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 2);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+ session2id = g_strdup (gst_rtsp_session_get_sessionid (session2));
+
+ session3 = gst_rtsp_session_pool_create (pool);
+ fail_unless (GST_IS_RTSP_SESSION (session3));
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 3);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+ session3id = g_strdup (gst_rtsp_session_get_sessionid (session3));
+
+ fail_if (GST_IS_RTSP_SESSION (gst_rtsp_session_pool_create (pool)));
+
+ compare = gst_rtsp_session_pool_find (pool, session1id);
+ fail_unless (compare == session1);
+ g_object_unref (compare);
+ compare = gst_rtsp_session_pool_find (pool, session2id);
+ fail_unless (compare == session2);
+ g_object_unref (compare);
+ compare = gst_rtsp_session_pool_find (pool, session3id);
+ fail_unless (compare == session3);
+ g_object_unref (compare);
+ fail_unless (gst_rtsp_session_pool_find (pool, "") == NULL);
+
+ fail_unless (gst_rtsp_session_pool_remove (pool, session2));
+ g_object_unref (session2);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 2);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+
+ gst_rtsp_session_pool_set_max_sessions (pool, 2);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 2);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 2);
+
+ session2 = gst_rtsp_session_pool_create (pool);
+ fail_if (GST_IS_RTSP_SESSION (session2));
+
+ {
+ list = gst_rtsp_session_pool_filter (pool, NULL, NULL);
+ fail_unless_equals_int (g_list_length (list), 2);
+ fail_unless (g_list_find (list, session1) != NULL);
+ fail_unless (g_list_find (list, session3) != NULL);
+ g_list_free_full (list, (GDestroyNotify) g_object_unref);
+ }
+
+ {
+ Responses responses = {
+ {session1, session2, session3}
+ ,
+ {GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP}
+ ,
+ };
+
+ list = gst_rtsp_session_pool_filter (pool, filter_func, &responses);
+ fail_unless (list == NULL);
+ }
+
+ {
+ Responses responses = {
+ {session1, session2, session3}
+ ,
+ {GST_RTSP_FILTER_REF, GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP}
+ ,
+ };
+
+ list = gst_rtsp_session_pool_filter (pool, filter_func, &responses);
+ fail_unless_equals_int (g_list_length (list), 1);
+ fail_unless (g_list_nth_data (list, 0) == session1);
+ g_list_free_full (list, (GDestroyNotify) g_object_unref);
+ }
+
+ {
+ Responses responses = {
+ {session1, session2, session3}
+ ,
+ {GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_REMOVE}
+ ,
+ };
+
+ list = gst_rtsp_session_pool_filter (pool, filter_func, &responses);
+ fail_unless_equals_int (g_list_length (list), 0);
+ g_list_free (list);
+ }
+
+ compare = gst_rtsp_session_pool_find (pool, session1id);
+ fail_unless (compare == session1);
+ g_object_unref (compare);
+ fail_unless (gst_rtsp_session_pool_find (pool, session2id) == NULL);
+ fail_unless (gst_rtsp_session_pool_find (pool, session3id) == NULL);
+
+ g_object_get (pool, "max-sessions", &maxsessions, NULL);
+ fail_unless_equals_int (maxsessions, 2);
+
+ g_object_set (pool, "max-sessions", 3, NULL);
+ g_object_get (pool, "max-sessions", &maxsessions, NULL);
+ fail_unless_equals_int (maxsessions, 3);
+
+ fail_unless_equals_int (gst_rtsp_session_pool_cleanup (pool), 0);
+
+ gst_rtsp_session_set_timeout (session1, 1);
+
+ source = gst_rtsp_session_pool_create_watch (pool);
+ fail_unless (source != NULL);
+
+ sourceid = g_source_attach (source, NULL);
+ fail_unless (sourceid != 0);
+
+ while (!g_main_context_iteration (NULL, TRUE));
+
+ g_source_unref (source);
+
+ g_object_unref (session1);
+ g_object_unref (session3);
+
+ g_free (session1id);
+ g_free (session2id);
+ g_free (session3id);
+
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspsessionpool_suite (void)
+{
+ Suite *s = suite_create ("rtspsessionpool");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 15);
+ tcase_add_test (tc, test_pool);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspsessionpool);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Axis Communications AB <dev-gstreamer at axis dot com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-stream.h>
+#include <rtsp-address-pool.h>
+
+static void
+get_sockets (GstRTSPLowerTrans lower_transport, GSocketFamily socket_family)
+{
+ GstPad *srcpad;
+ GstElement *pay;
+ GstRTSPStream *stream;
+ GstBin *bin;
+ GstElement *rtpbin;
+ GstRTSPAddressPool *pool;
+ GSocket *socket;
+ gboolean have_ipv4;
+ gboolean have_ipv6;
+ GstRTSPTransport *transport;
+
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ gst_pad_set_active (srcpad, TRUE);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+ rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+ fail_unless (rtpbin != NULL);
+ bin = GST_BIN (gst_bin_new ("testbin"));
+ fail_unless (bin != NULL);
+ fail_unless (gst_bin_add (bin, rtpbin));
+
+ /* configure address pool for IPv4 and IPv6 unicast addresses */
+ pool = gst_rtsp_address_pool_new ();
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ GST_RTSP_ADDRESS_POOL_ANY_IPV4, GST_RTSP_ADDRESS_POOL_ANY_IPV4, 50000,
+ 60000, 0));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ GST_RTSP_ADDRESS_POOL_ANY_IPV6, GST_RTSP_ADDRESS_POOL_ANY_IPV6, 50000,
+ 60000, 0));
+ fail_unless (gst_rtsp_address_pool_add_range (pool, "233.252.0.0",
+ "233.252.0.0", 50000, 60000, 1));
+ fail_unless (gst_rtsp_address_pool_add_range (pool, "FF11:DB8::1",
+ "FF11:DB8::1", 50000, 60000, 1));
+ gst_rtsp_stream_set_address_pool (stream, pool);
+
+ fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+ /* allocate udp ports first */
+ fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
+ transport->lower_transport = lower_transport;
+
+ /* no ports allocated, complete stream should fail */
+ fail_if (gst_rtsp_stream_complete_stream (stream, transport));
+
+ /* allocate ports */
+ fail_unless (gst_rtsp_stream_allocate_udp_sockets (stream,
+ socket_family, transport, FALSE));
+
+ fail_unless (gst_rtsp_stream_complete_stream (stream, transport));
+ fail_unless (gst_rtsp_transport_free (transport) == GST_RTSP_OK);
+
+ if (lower_transport == GST_RTSP_LOWER_TRANS_UDP)
+ socket = gst_rtsp_stream_get_rtp_socket (stream, G_SOCKET_FAMILY_IPV4);
+ else
+ socket = gst_rtsp_stream_get_rtp_multicast_socket (stream,
+ G_SOCKET_FAMILY_IPV4);
+ have_ipv4 = (socket != NULL);
+ if (have_ipv4) {
+ fail_unless (g_socket_get_fd (socket) >= 0);
+ g_object_unref (socket);
+ }
+
+ if (lower_transport == GST_RTSP_LOWER_TRANS_UDP)
+ socket = gst_rtsp_stream_get_rtcp_socket (stream, G_SOCKET_FAMILY_IPV4);
+ else
+ socket = gst_rtsp_stream_get_rtcp_multicast_socket (stream,
+ G_SOCKET_FAMILY_IPV4);
+ if (have_ipv4) {
+ fail_unless (socket != NULL);
+ fail_unless (g_socket_get_fd (socket) >= 0);
+ g_object_unref (socket);
+ } else {
+ fail_unless (socket == NULL);
+ }
+
+ if (lower_transport == GST_RTSP_LOWER_TRANS_UDP)
+ socket = gst_rtsp_stream_get_rtp_socket (stream, G_SOCKET_FAMILY_IPV6);
+ else
+ socket = gst_rtsp_stream_get_rtp_multicast_socket (stream,
+ G_SOCKET_FAMILY_IPV6);
+ have_ipv6 = (socket != NULL);
+ if (have_ipv6) {
+ fail_unless (g_socket_get_fd (socket) >= 0);
+ g_object_unref (socket);
+ }
+
+ if (lower_transport == GST_RTSP_LOWER_TRANS_UDP)
+ socket = gst_rtsp_stream_get_rtcp_socket (stream, G_SOCKET_FAMILY_IPV6);
+ else
+ socket = gst_rtsp_stream_get_rtcp_multicast_socket (stream,
+ G_SOCKET_FAMILY_IPV6);
+ if (have_ipv6) {
+ fail_unless (socket != NULL);
+ fail_unless (g_socket_get_fd (socket) >= 0);
+ g_object_unref (socket);
+ } else {
+ fail_unless (socket == NULL);
+ }
+
+ /* check that at least one family is available */
+ fail_unless (have_ipv4 || have_ipv6);
+
+ g_object_unref (pool);
+
+ fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+
+ gst_object_unref (bin);
+ gst_object_unref (stream);
+}
+
+GST_START_TEST (test_get_sockets_udp_ipv4)
+{
+ get_sockets (GST_RTSP_LOWER_TRANS_UDP, G_SOCKET_FAMILY_IPV4);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_get_sockets_udp_ipv6)
+{
+ get_sockets (GST_RTSP_LOWER_TRANS_UDP, G_SOCKET_FAMILY_IPV6);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_get_sockets_mcast_ipv4)
+{
+ get_sockets (GST_RTSP_LOWER_TRANS_UDP_MCAST, G_SOCKET_FAMILY_IPV4);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_get_sockets_mcast_ipv6)
+{
+ get_sockets (GST_RTSP_LOWER_TRANS_UDP_MCAST, G_SOCKET_FAMILY_IPV6);
+}
+
+GST_END_TEST;
+
+/* The purpose of this test is to make sure that it's not possible to allocate
+ * multicast UDP ports if the address pool does not contain multicast UDP
+ * addresses. */
+GST_START_TEST (test_allocate_udp_ports_fail)
+{
+ GstPad *srcpad;
+ GstElement *pay;
+ GstRTSPStream *stream;
+ GstBin *bin;
+ GstElement *rtpbin;
+ GstRTSPAddressPool *pool;
+ GstRTSPTransport *transport;
+
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ gst_pad_set_active (srcpad, TRUE);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+ rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+ fail_unless (rtpbin != NULL);
+ bin = GST_BIN (gst_bin_new ("testbin"));
+ fail_unless (bin != NULL);
+ fail_unless (gst_bin_add (bin, rtpbin));
+
+ pool = gst_rtsp_address_pool_new ();
+ fail_unless (gst_rtsp_address_pool_add_range (pool, "192.168.1.1",
+ "192.168.1.1", 6000, 6001, 0));
+ gst_rtsp_stream_set_address_pool (stream, pool);
+
+ fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+ fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
+ transport->lower_transport = GST_RTSP_LOWER_TRANS_UDP_MCAST;
+ fail_if (gst_rtsp_stream_allocate_udp_sockets (stream, G_SOCKET_FAMILY_IPV4,
+ transport, FALSE));
+ fail_unless (gst_rtsp_transport_free (transport) == GST_RTSP_OK);
+
+ g_object_unref (pool);
+ fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+ gst_object_unref (bin);
+ gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_get_multicast_address)
+{
+ GstPad *srcpad;
+ GstElement *pay;
+ GstRTSPStream *stream;
+ GstRTSPAddressPool *pool;
+ GstRTSPAddress *addr1;
+ GstRTSPAddress *addr2;
+
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ gst_pad_set_active (srcpad, TRUE);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+
+ pool = gst_rtsp_address_pool_new ();
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.0", "233.252.0.0", 5100, 5101, 1));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "FF11:DB8::1", "FF11:DB8::1", 5102, 5103, 1));
+ gst_rtsp_stream_set_address_pool (stream, pool);
+
+ addr1 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+ fail_unless (addr1 != NULL);
+ fail_unless_equals_string (addr1->address, "233.252.0.0");
+ fail_unless_equals_int (addr1->port, 5100);
+ fail_unless_equals_int (addr1->n_ports, 2);
+
+ addr2 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+ fail_unless (addr2 != NULL);
+ fail_unless_equals_string (addr2->address, "233.252.0.0");
+ fail_unless_equals_int (addr2->port, 5100);
+ fail_unless_equals_int (addr2->n_ports, 2);
+
+ gst_rtsp_address_free (addr1);
+ gst_rtsp_address_free (addr2);
+
+ addr1 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
+ fail_unless (addr1 != NULL);
+ fail_unless (!g_ascii_strcasecmp (addr1->address, "FF11:DB8::1"));
+ fail_unless_equals_int (addr1->port, 5102);
+ fail_unless_equals_int (addr1->n_ports, 2);
+
+ addr2 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
+ fail_unless (addr2 != NULL);
+ fail_unless (!g_ascii_strcasecmp (addr2->address, "FF11:DB8::1"));
+ fail_unless_equals_int (addr2->port, 5102);
+ fail_unless_equals_int (addr2->n_ports, 2);
+
+ gst_rtsp_address_free (addr1);
+ gst_rtsp_address_free (addr2);
+
+ g_object_unref (pool);
+
+ gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+/* test case: address pool only contains multicast addresses,
+ * but the client is requesting unicast udp */
+GST_START_TEST (test_multicast_address_and_unicast_udp)
+{
+ GstPad *srcpad;
+ GstElement *pay;
+ GstRTSPStream *stream;
+ GstBin *bin;
+ GstElement *rtpbin;
+ GstRTSPAddressPool *pool;
+
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ gst_pad_set_active (srcpad, TRUE);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+ rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+ fail_unless (rtpbin != NULL);
+ bin = GST_BIN (gst_bin_new ("testbin"));
+ fail_unless (bin != NULL);
+ fail_unless (gst_bin_add (bin, rtpbin));
+
+ pool = gst_rtsp_address_pool_new ();
+ /* add a multicast addres to the address pool */
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.0", "233.252.0.0", 5200, 5201, 1));
+ gst_rtsp_stream_set_address_pool (stream, pool);
+
+ fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+ g_object_unref (pool);
+ fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+ gst_object_unref (bin);
+ gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_allocate_udp_ports_multicast)
+{
+ GstPad *srcpad;
+ GstElement *pay;
+ GstRTSPStream *stream;
+ GstBin *bin;
+ GstElement *rtpbin;
+ GstRTSPAddressPool *pool;
+ GstRTSPAddress *addr;
+
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ gst_pad_set_active (srcpad, TRUE);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+ rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+ fail_unless (rtpbin != NULL);
+ bin = GST_BIN (gst_bin_new ("testbin"));
+ fail_unless (bin != NULL);
+ fail_unless (gst_bin_add (bin, rtpbin));
+
+ pool = gst_rtsp_address_pool_new ();
+ /* add multicast addresses to the address pool */
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "233.252.0.1", 6000, 6001, 1));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "FF11:DB8::1", "FF11:DB8::1", 6002, 6003, 1));
+ gst_rtsp_stream_set_address_pool (stream, pool);
+
+ fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+ /* check the multicast address and ports for IPv4 */
+ addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+ fail_unless (addr != NULL);
+ fail_unless_equals_string (addr->address, "233.252.0.1");
+ fail_unless_equals_int (addr->port, 6000);
+ fail_unless_equals_int (addr->n_ports, 2);
+ gst_rtsp_address_free (addr);
+
+ /* check the multicast address and ports for IPv6 */
+ addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
+ fail_unless (addr != NULL);
+ fail_unless (!g_ascii_strcasecmp (addr->address, "FF11:DB8::1"));
+ fail_unless_equals_int (addr->port, 6002);
+ fail_unless_equals_int (addr->n_ports, 2);
+ gst_rtsp_address_free (addr);
+
+ g_object_unref (pool);
+ fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+ gst_object_unref (bin);
+ gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_allocate_udp_ports_client_settings)
+{
+ GstPad *srcpad;
+ GstElement *pay;
+ GstRTSPStream *stream;
+ GstBin *bin;
+ GstElement *rtpbin;
+ GstRTSPAddressPool *pool;
+ GstRTSPAddress *addr;
+
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ gst_pad_set_active (srcpad, TRUE);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+ rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+ fail_unless (rtpbin != NULL);
+ bin = GST_BIN (gst_bin_new ("testbin"));
+ fail_unless (bin != NULL);
+ fail_unless (gst_bin_add (bin, rtpbin));
+
+ pool = gst_rtsp_address_pool_new ();
+ /* add multicast addresses to the address pool */
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "233.252.0.1", 6000, 6001, 1));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "FF11:DB7::1", "FF11:DB7::1", 6004, 6005, 1));
+ /* multicast address specified by the client */
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.2", "233.252.0.2", 6002, 6003, 1));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "FF11:DB8::1", "FF11:DB8::1", 6006, 6007, 1));
+ gst_rtsp_stream_set_address_pool (stream, pool);
+
+ fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+ /* Reserve IPV4 mcast address */
+ addr = gst_rtsp_stream_reserve_address (stream, "233.252.0.2", 6002, 2, 1);
+ fail_unless (addr != NULL);
+ gst_rtsp_address_free (addr);
+
+ /* verify that the multicast address and ports correspond to the requested client
+ * transport information for IPv4 */
+ addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+ fail_unless (addr != NULL);
+ fail_unless_equals_string (addr->address, "233.252.0.2");
+ fail_unless_equals_int (addr->port, 6002);
+ fail_unless_equals_int (addr->n_ports, 2);
+ gst_rtsp_address_free (addr);
+
+ /* Reserve IPV6 mcast address */
+ addr = gst_rtsp_stream_reserve_address (stream, "FF11:DB8::1", 6006, 2, 1);
+ fail_unless (addr != NULL);
+ gst_rtsp_address_free (addr);
+
+ /* verify that the multicast address and ports correspond to the requested client
+ * transport information for IPv6 */
+ addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
+ fail_unless (addr != NULL);
+ fail_unless (!g_ascii_strcasecmp (addr->address, "FF11:DB8::1"));
+ fail_unless_equals_int (addr->port, 6006);
+ fail_unless_equals_int (addr->n_ports, 2);
+ gst_rtsp_address_free (addr);
+
+ g_object_unref (pool);
+ fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+ gst_object_unref (bin);
+ gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_tcp_transport)
+{
+ GstPad *srcpad;
+ GstElement *pay;
+ GstRTSPStream *stream;
+ GstBin *bin;
+ GstElement *rtpbin;
+ GstRTSPRange server_port;
+
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ gst_pad_set_active (srcpad, TRUE);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+ rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+ fail_unless (rtpbin != NULL);
+ bin = GST_BIN (gst_bin_new ("testbin"));
+ fail_unless (bin != NULL);
+ fail_unless (gst_bin_add (bin, rtpbin));
+
+ /* TCP transport */
+ gst_rtsp_stream_set_protocols (stream, GST_RTSP_LOWER_TRANS_TCP);
+ fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+ /* port that the server will use to receive RTCP makes only sense in the UDP
+ * case so verify that the received server port is 0 in the TCP case */
+ gst_rtsp_stream_get_server_port (stream, &server_port, G_SOCKET_FAMILY_IPV4);
+ fail_unless_equals_int (server_port.min, 0);
+ fail_unless_equals_int (server_port.max, 0);
+
+ fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+ gst_object_unref (bin);
+ gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+static void
+check_multicast_client_address (const gchar * destination, guint port,
+ const gchar * expected_addr_str, gboolean expected_res)
+{
+ GstPad *srcpad;
+ GstElement *pay;
+ GstRTSPStream *stream;
+ GstBin *bin;
+ GstElement *rtpbin;
+ GstRTSPTransport *transport;
+ GstRTSPRange ports = { 0 };
+ gchar *addr_str = NULL;
+
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ gst_pad_set_active (srcpad, TRUE);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+ rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+ fail_unless (rtpbin != NULL);
+ bin = GST_BIN (gst_bin_new ("testbin"));
+ fail_unless (bin != NULL);
+ fail_unless (gst_bin_add (bin, rtpbin));
+
+ fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+ fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
+ transport->lower_transport = GST_RTSP_LOWER_TRANS_UDP_MCAST;
+ transport->destination = g_strdup (destination);
+ transport->ttl = 1;
+ ports.min = port;
+ ports.max = port + 1;
+ transport->port = ports;
+
+ /* allocate ports */
+ fail_unless (gst_rtsp_stream_allocate_udp_sockets (stream,
+ G_SOCKET_FAMILY_IPV4, transport, TRUE) == expected_res);
+
+ fail_unless (gst_rtsp_stream_add_multicast_client_address (stream,
+ destination, ports.min, ports.max,
+ G_SOCKET_FAMILY_IPV4) == expected_res);
+
+ fail_unless (gst_rtsp_stream_complete_stream (stream,
+ transport) == expected_res);
+
+ fail_unless (gst_rtsp_transport_free (transport) == GST_RTSP_OK);
+ addr_str = gst_rtsp_stream_get_multicast_client_addresses (stream);
+
+ fail_unless (g_str_equal (addr_str, expected_addr_str));
+ g_free (addr_str);
+
+ fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+
+ gst_object_unref (bin);
+ gst_object_unref (stream);
+}
+
+/* test if the provided transport destination is correct.
+ * CASE: valid multicast address */
+GST_START_TEST (test_multicast_client_address)
+{
+ const gchar *addr = "233.252.0.1";
+ guint port = 50000;
+ const gchar *expected_addr_str = "233.252.0.1:50000";
+ gboolean expected_res = TRUE;
+
+ check_multicast_client_address (addr, port, expected_addr_str, expected_res);
+}
+
+GST_END_TEST;
+
+/* test if the provided transport destination is correct.
+ * CASE: invalid multicast address */
+GST_START_TEST (test_multicast_client_address_invalid)
+{
+ const gchar *addr = "1.2.3.4";
+ guint port = 50000;
+ const gchar *expected_addr_str = "";
+ gboolean expected_res = FALSE;
+
+ check_multicast_client_address (addr, port, expected_addr_str, expected_res);
+}
+
+GST_END_TEST;
+
+static void
+add_transports (gboolean add_twice)
+{
+ GstRTSPTransport *transport;
+ GstRTSPStream *stream;
+ GstRTSPStreamTransport *tr;
+ GstPad *srcpad;
+ GstElement *pay;
+ GstBin *bin;
+ GstElement *rtpbin;
+
+ fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
+ transport->lower_transport = GST_RTSP_LOWER_TRANS_TCP;
+ transport->destination = g_strdup ("127.0.0.1");
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+ rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+ fail_unless (rtpbin != NULL);
+ bin = GST_BIN (gst_bin_new ("testbin"));
+ fail_unless (bin != NULL);
+ fail_unless (gst_bin_add (bin, rtpbin));
+
+ /* TCP transport */
+ gst_rtsp_stream_set_protocols (stream, GST_RTSP_LOWER_TRANS_TCP);
+ fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+ tr = gst_rtsp_stream_transport_new (stream, transport);
+ fail_unless (tr);
+
+ if (add_twice) {
+ fail_unless (gst_rtsp_stream_add_transport (stream, tr));
+ fail_unless (gst_rtsp_stream_add_transport (stream, tr));
+ fail_unless (gst_rtsp_stream_remove_transport (stream, tr));
+ } else {
+ fail_unless (gst_rtsp_stream_add_transport (stream, tr));
+ fail_unless (gst_rtsp_stream_remove_transport (stream, tr));
+ fail_if (gst_rtsp_stream_remove_transport (stream, tr));
+ }
+
+ fail_unless (gst_rtsp_transport_free (transport) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+ gst_object_unref (bin);
+ gst_object_unref (stream);
+}
+
+
+GST_START_TEST (test_add_transport_twice)
+{
+ add_transports (TRUE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_remove_transport_twice)
+{
+ add_transports (FALSE);
+}
+
+GST_END_TEST;
+
+static gboolean
+is_ipv6_supported (void)
+{
+ GError *err = NULL;
+ GSocket *sock;
+
+ sock =
+ g_socket_new (G_SOCKET_FAMILY_IPV6, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_DEFAULT, &err);
+ if (sock) {
+ g_object_unref (sock);
+ return TRUE;
+ }
+
+ if (!g_error_matches (err, G_IO_ERROR, G_IO_ERROR_NOT_SUPPORTED)) {
+ GST_WARNING ("Unabled to create IPv6 socket: %s", err->message);
+ }
+ g_clear_error (&err);
+
+ return FALSE;
+}
+
+static Suite *
+rtspstream_suite (void)
+{
+ Suite *s = suite_create ("rtspstream");
+ TCase *tc = tcase_create ("general");
+ gboolean have_ipv6 = is_ipv6_supported ();
+
+ suite_add_tcase (s, tc);
+ tcase_add_test (tc, test_get_sockets_udp_ipv4);
+ tcase_add_test (tc, test_get_sockets_mcast_ipv4);
+ if (have_ipv6) {
+ tcase_add_test (tc, test_get_sockets_udp_ipv6);
+ tcase_add_test (tc, test_get_sockets_mcast_ipv6);
+ }
+ tcase_add_test (tc, test_allocate_udp_ports_fail);
+ tcase_add_test (tc, test_get_multicast_address);
+ tcase_add_test (tc, test_multicast_address_and_unicast_udp);
+ tcase_add_test (tc, test_allocate_udp_ports_multicast);
+ tcase_add_test (tc, test_allocate_udp_ports_client_settings);
+ tcase_add_test (tc, test_tcp_transport);
+ tcase_add_test (tc, test_multicast_client_address);
+ tcase_add_test (tc, test_multicast_client_address_invalid);
+ tcase_add_test (tc, test_add_transport_twice);
+ tcase_add_test (tc, test_remove_transport_twice);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspstream);
--- /dev/null
+/* GStreamer
+ * unit tests for GstRTSPThreadPool
+ * Copyright (C) 2013 Axis Communications <dev-gstreamer at axis dot com>
+ * @author Ognyan Tonchev <ognyan at axis dot com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-thread-pool.h>
+
+GST_START_TEST (test_pool_get_thread)
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ pool = gst_rtsp_thread_pool_new ();
+ fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+ thread = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread));
+ /* one ref is hold by the pool */
+ fail_unless_equals_int (GST_MINI_OBJECT_REFCOUNT (thread), 2);
+
+ gst_rtsp_thread_stop (thread);
+ g_object_unref (pool);
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_get_media_thread)
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ pool = gst_rtsp_thread_pool_new ();
+ fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+ thread = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_MEDIA,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread));
+ /* one ref is hold by the pool */
+ fail_unless_equals_int (GST_MINI_OBJECT_REFCOUNT (thread), 2);
+
+ gst_rtsp_thread_stop (thread);
+ g_object_unref (pool);
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_get_thread_reuse)
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread1;
+ GstRTSPThread *thread2;
+
+ pool = gst_rtsp_thread_pool_new ();
+ fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+ gst_rtsp_thread_pool_set_max_threads (pool, 1);
+
+ thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread1));
+
+ thread2 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread2));
+
+ fail_unless (thread2 == thread1);
+ /* one ref is hold by the pool */
+ fail_unless_equals_int (GST_MINI_OBJECT_REFCOUNT (thread1), 3);
+
+ gst_rtsp_thread_stop (thread1);
+ gst_rtsp_thread_stop (thread2);
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static void
+do_test_pool_max_thread (gboolean use_property)
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread1;
+ GstRTSPThread *thread2;
+ GstRTSPThread *thread3;
+ gint max_threads;
+
+ pool = gst_rtsp_thread_pool_new ();
+ fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+ if (use_property) {
+ g_object_get (pool, "max-threads", &max_threads, NULL);
+ fail_unless_equals_int (max_threads, 1);
+ } else {
+ fail_unless_equals_int (gst_rtsp_thread_pool_get_max_threads (pool), 1);
+ }
+
+ thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread1));
+
+ thread2 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread2));
+
+ fail_unless (thread1 == thread2);
+
+ gst_rtsp_thread_stop (thread1);
+ gst_rtsp_thread_stop (thread2);
+
+ if (use_property) {
+ g_object_set (pool, "max-threads", 2, NULL);
+ g_object_get (pool, "max-threads", &max_threads, NULL);
+ fail_unless_equals_int (max_threads, 2);
+ } else {
+ gst_rtsp_thread_pool_set_max_threads (pool, 2);
+ fail_unless_equals_int (gst_rtsp_thread_pool_get_max_threads (pool), 2);
+ }
+
+ thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread1));
+
+ thread2 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread2));
+
+ thread3 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread3));
+
+ fail_unless (thread2 != thread1);
+ fail_unless (thread3 == thread2 || thread3 == thread1);
+
+ gst_rtsp_thread_stop (thread1);
+ gst_rtsp_thread_stop (thread2);
+ gst_rtsp_thread_stop (thread3);
+
+ if (use_property) {
+ g_object_set (pool, "max-threads", 0, NULL);
+ g_object_get (pool, "max-threads", &max_threads, NULL);
+ fail_unless_equals_int (max_threads, 0);
+ } else {
+ gst_rtsp_thread_pool_set_max_threads (pool, 0);
+ fail_unless_equals_int (gst_rtsp_thread_pool_get_max_threads (pool), 0);
+ }
+
+ thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_if (GST_IS_RTSP_THREAD (thread1));
+
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_START_TEST (test_pool_max_threads)
+{
+ do_test_pool_max_thread (FALSE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_max_threads_property)
+{
+ do_test_pool_max_thread (TRUE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_thread_copy)
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread1;
+ GstRTSPThread *thread2;
+
+ pool = gst_rtsp_thread_pool_new ();
+ fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+ thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread1));
+ fail_unless (GST_IS_MINI_OBJECT_TYPE (thread1, GST_TYPE_RTSP_THREAD));
+
+ thread2 = GST_RTSP_THREAD (gst_mini_object_copy (GST_MINI_OBJECT (thread1)));
+ fail_unless (GST_IS_RTSP_THREAD (thread2));
+ fail_unless (GST_IS_MINI_OBJECT_TYPE (thread2, GST_TYPE_RTSP_THREAD));
+
+ gst_rtsp_thread_stop (thread1);
+ gst_rtsp_thread_stop (thread2);
+ g_object_unref (pool);
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspthreadpool_suite (void)
+{
+ Suite *s = suite_create ("rtspthreadpool");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_pool_get_thread);
+ tcase_add_test (tc, test_pool_get_media_thread);
+ tcase_add_test (tc, test_pool_get_thread_reuse);
+ tcase_add_test (tc, test_pool_max_threads);
+ tcase_add_test (tc, test_pool_max_threads_property);
+ tcase_add_test (tc, test_pool_thread_copy);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspthreadpool);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Sebastian Rasmussen <sebras@hotmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-token.h>
+
+GST_START_TEST (test_token)
+{
+ GstRTSPToken *token;
+ GstRTSPToken *token2;
+ GstRTSPToken *copy;
+ GstStructure *str;
+
+ token = gst_rtsp_token_new_empty ();
+ fail_if (gst_rtsp_token_is_allowed (token, "missing"));
+ gst_rtsp_token_unref (token);
+
+ token = gst_rtsp_token_new ("role", G_TYPE_STRING, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+ fail_unless (gst_rtsp_token_is_allowed (token, "permission1"));
+ fail_if (gst_rtsp_token_is_allowed (token, "permission2"));
+ fail_if (gst_rtsp_token_is_allowed (token, "missing"));
+ copy = GST_RTSP_TOKEN (gst_mini_object_copy (GST_MINI_OBJECT (token)));
+ gst_rtsp_token_unref (token);
+ fail_unless_equals_string (gst_rtsp_token_get_string (copy, "role"), "user");
+ fail_unless (gst_rtsp_token_is_allowed (copy, "permission1"));
+ fail_if (gst_rtsp_token_is_allowed (copy, "permission2"));
+ fail_if (gst_rtsp_token_is_allowed (copy, "missing"));
+ gst_rtsp_token_unref (copy);
+
+ token = gst_rtsp_token_new ("role", G_TYPE_STRING, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+ fail_unless (gst_rtsp_token_is_allowed (token, "permission1"));
+ fail_if (gst_rtsp_token_is_allowed (token, "permission2"));
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+
+ fail_unless (gst_mini_object_is_writable (GST_MINI_OBJECT (token)));
+ fail_unless (gst_rtsp_token_writable_structure (token) != NULL);
+ fail_unless (gst_rtsp_token_get_structure (token) != NULL);
+
+ token2 = gst_rtsp_token_ref (token);
+
+ fail_if (gst_mini_object_is_writable (GST_MINI_OBJECT (token)));
+ ASSERT_CRITICAL (fail_unless (gst_rtsp_token_writable_structure (token) ==
+ NULL));
+ fail_unless (gst_rtsp_token_get_structure (token) != NULL);
+
+ gst_rtsp_token_unref (token2);
+
+ fail_unless (gst_mini_object_is_writable (GST_MINI_OBJECT (token)));
+ fail_unless (gst_rtsp_token_writable_structure (token) != NULL);
+ fail_unless (gst_rtsp_token_get_structure (token) != NULL);
+
+ str = gst_rtsp_token_writable_structure (token);
+ gst_structure_set (str, "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+ fail_unless (gst_rtsp_token_is_allowed (token, "permission1"));
+ fail_unless (gst_rtsp_token_is_allowed (token, "permission2"));
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+
+ gst_rtsp_token_set_bool (token, "permission3", FALSE);
+ fail_unless (!gst_rtsp_token_is_allowed (token, "permission3"));
+ gst_rtsp_token_set_bool (token, "permission4", TRUE);
+ fail_unless (gst_rtsp_token_is_allowed (token, "permission4"));
+
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+ gst_rtsp_token_set_string (token, "role", "admin");
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"),
+ "admin");
+
+ gst_rtsp_token_unref (token);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtsptoken_suite (void)
+{
+ Suite *s = suite_create ("rtsptoken");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_token);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtsptoken);
--- /dev/null
+pluginsdirs = []
+if gst_dep.type_name() == 'pkgconfig'
+ pbase = dependency('gstreamer-plugins-base-' + api_version, required: true)
+ pbad = dependency('gstreamer-plugins-bad-' + api_version, required: true)
+
+ pluginsdirs = [gst_dep.get_pkgconfig_variable('pluginsdir'),
+ pbase.get_pkgconfig_variable('pluginsdir'),
+ pbad.get_pkgconfig_variable('pluginsdir')]
+
+ gst_plugin_scanner_dir = gst_dep.get_pkgconfig_variable('pluginscannerdir')
+else
+ gst_plugin_scanner_dir = subproject('gstreamer').get_variable('gst_scanner_dir')
+endif
+gst_plugin_scanner_path = join_paths(gst_plugin_scanner_dir, 'gst-plugin-scanner')
+
+test_c_args = [
+ '-UG_DISABLE_ASSERT',
+ '-UG_DISABLE_CAST_CHECKS',
+ '-DGST_CHECK_TEST_ENVIRONMENT_BEACON="GST_PLUGIN_LOADING_WHITELIST"',
+ '-DGST_TEST_FILES_PATH="' + meson.current_source_dir() + '/../files"',
+]
+
+rtsp_server_tests = [
+ 'gst/addresspool',
+ 'gst/client',
+ 'gst/mountpoints',
+ 'gst/mediafactory',
+ 'gst/media',
+ 'gst/permissions',
+ 'gst/rtspserver',
+ 'gst/sessionmedia',
+ 'gst/sessionpool',
+ 'gst/stream',
+ 'gst/threadpool',
+ 'gst/token',
+ 'gst/onvif',
+]
+
+if not get_option('rtspclientsink').disabled()
+ rtsp_server_tests += ['gst/rtspclientsink']
+endif
+
+foreach test_name : rtsp_server_tests
+ fname = '@0@.c'.format(test_name)
+ test_name = test_name.underscorify()
+
+ env = environment()
+ env.set('GST_PLUGIN_SYSTEM_PATH_1_0', '')
+ env.set('GST_STATE_IGNORE_ELEMENTS', '')
+ env.set('GST_PLUGIN_LOADING_WHITELIST', 'gstreamer:gst-plugins-base:gst-plugins-good:gst-plugins-bad:gst-rtsp-server@' + meson.build_root())
+ env.set('CK_DEFAULT_TIMEOUT', '120')
+ env.set('GST_REGISTRY', join_paths(meson.current_build_dir(), '@0@.registry'.format(test_name)))
+ env.set('GST_PLUGIN_PATH_1_0', [meson.build_root()] + pluginsdirs)
+ env.set('GST_PLUGIN_SCANNER_1_0', gst_plugin_scanner_path)
+
+ exe = executable(test_name, fname,
+ include_directories : rtspserver_incs,
+ c_args : rtspserver_args + test_c_args,
+ dependencies : [gstcheck_dep, gstrtsp_dep, gstrtp_dep, gst_rtsp_server_dep]
+ )
+ test(test_name, exe,
+ env : env,
+ timeout : 120,
+ is_parallel: false
+ )
+endforeach
--- /dev/null
+# FIXME: make check work on windows
+if host_machine.system() != 'windows' and gstcheck_dep.found()
+ subdir('check')
+endif
+
+test_cleanup_exe = executable('test-cleanup', 'test-cleanup.c',
+ dependencies: gst_rtsp_server_dep)
+
+test_reuse_exe = executable('test-reuse', 'test-reuse.c',
+ dependencies: gst_rtsp_server_dep)
+
+test('test-cleanup', test_cleanup_exe)
+test('test-reuse', test_reuse_exe)
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+static gboolean
+timeout (GMainLoop * loop)
+{
+ g_main_loop_quit (loop);
+ return FALSE;
+}
+
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ guint id;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* We just want to bind any port here, so that tests can run in parallel */
+ gst_rtsp_server_set_service (server, "0");
+
+ /* attach the server to the default maincontext */
+ if ((id = gst_rtsp_server_attach (server, NULL)) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, loop);
+
+ /* start serving */
+ g_main_loop_run (loop);
+
+ /* cleanup */
+ g_source_remove (id);
+ g_object_unref (server);
+ g_main_loop_unref (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+#define TIMEOUT 2
+
+static gboolean timeout_1 (GMainLoop * loop);
+
+static guint id;
+static gint rounds = 3;
+static GstRTSPServer *server;
+
+static gboolean
+timeout_2 (GMainLoop * loop)
+{
+ rounds--;
+ if (rounds > 0) {
+ id = gst_rtsp_server_attach (server, NULL);
+ g_print ("have attached\n");
+ g_timeout_add_seconds (TIMEOUT, (GSourceFunc) timeout_1, loop);
+ } else {
+ g_main_loop_quit (loop);
+ }
+ return FALSE;
+}
+
+static gboolean
+timeout_1 (GMainLoop * loop)
+{
+ g_source_remove (id);
+ g_print ("have removed\n");
+ g_timeout_add_seconds (TIMEOUT, (GSourceFunc) timeout_2, loop);
+ return FALSE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* attach the server to the default maincontext */
+ if ((id = gst_rtsp_server_attach (server, NULL)) == 0)
+ goto failed;
+ g_print ("have attached\n");
+
+ g_timeout_add_seconds (TIMEOUT, (GSourceFunc) timeout_1, loop);
+
+ /* start serving */
+ g_main_loop_run (loop);
+
+ /* cleanup */
+ g_object_unref (server);
+ g_main_loop_unref (loop);
+
+ g_print ("quit\n");
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
projects / platform / upstream / gstreamer.git / commitdiff