=== release 1.19.3 === 2021-11-03 15:43:36 +0000 Tim-Philipp Müller * NEWS: * RELEASE: * docs/gst_plugins_cache.json: * gst-rtsp-server.doap: * meson.build: Release 1.19.3 2021-11-03 15:43:32 +0000 Tim-Philipp Müller * ChangeLog: Update ChangeLogs for 1.19.3 2021-10-25 11:37:45 +0100 Tim-Philipp Müller * meson.build: meson: require matching GStreamer dep versions for unstable development releases Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929 Part-of: 2021-10-18 15:47:00 +0100 Tim-Philipp Müller * tests/check/meson.build: meson: update for meson.build_root() and .build_source() deprecation -> use meson.project_build_root() or .global_build_root() instead. Part-of: 2021-10-18 00:40:14 +0100 Tim-Philipp Müller * docs/meson.build: * tests/check/meson.build: meson: update for dep.get_pkgconfig_variable() deprecation ... in favour of dep.get_variable('foo', ..) which in some cases allows for further cleanups in future since we can extract variables from pkg-config dependencies as well as internal dependencies using this mechanism. Part-of: 2021-10-01 15:32:58 +0100 Tim-Philipp Müller * gst/rtsp-server/meson.build: * gst/rtsp-sink/meson.build: rtsp-server: define G_LOG_DOMAIN Fixes #634 Part-of: 2021-10-14 18:38:26 +0100 Tim-Philipp Müller * meson.build: meson: bump meson requirement to >= 0.59 For monorepo build and ugly/bad, for advanced feature option API like get_option('xyz').required(..) which we use in combination with the 'gpl' option. For rest of modules for consistency (people will likely use newer features based on the top-level requirement). Part-of: 2021-10-12 15:52:48 -0300 Thibault Saunier * docs/meson.build: meson: Streamline the way we detect when to build documentation Part-of: 2020-06-27 00:39:00 -0400 Thibault Saunier * docs/meson.build: * gst/rtsp-server/meson.build: * meson.build: meson: List libraries and their corresponding gir definition Introduces a `libraries` variable that contains all libraries in a list with the following format: ``` meson libraries = [ [pkg_name, { 'lib': library_object 'gir': [ {full gir definition in a dict } ] ], .... ] ``` It therefore refactors the way we build the gir so that we can reuse the same information to build them against 'gstreamer-full' in gst-build when linking statically Part-of: 2020-06-27 00:37:39 -0400 Thibault Saunier * gst/rtsp-server/meson.build: meson: Mark files as files() Making it more robust and future proof And fix issues that it creates Part-of: 2021-10-07 13:00:10 +0300 Sebastian Dröge * gst/rtsp-server/rtsp-media.c: rtsp-media: Unprepare suspended medias too Previously suspended medias immediately reached the UNPREPARED state without going through the media's unprepare() vfunc. This didn't allow the media subclass to do any additional cleanup, and for example the shutdown-eos property of GstRTSPMedia was ignored. Part-of: 2021-10-06 18:19:29 +0300 Sebastian Dröge * gst/rtsp-server/rtsp-media.c: rtsp-media: Only unprepare a media if it was not already unpreparing anyway Part-of: 2021-10-03 23:25:23 +0200 Ognyan Tonchev * gst/rtsp-server/rtsp-client.c: * gst/rtsp-server/rtsp-session.c: * gst/rtsp-server/rtsp-session.h: rtsp-client: make sure sessmedia will not get freed while used handle_*_request() functions were all retrieving the session media from the session by calling gst_rtsp_session_get_media () which is a transfer-none call. If a session timeout happens at that time, the session media may get freed making the pointer invalid.. Fixes #757 Part-of: 2021-10-05 19:37:40 +0300 Sebastian Dröge * gst/rtsp-server/rtsp-media.c: rtsp-media: Also mark receive-only (RECORD) medias as prepared when unsuspending Previously the status was only changed for other medias. Part-of: 2021-10-01 13:51:37 +0300 Sebastian Dröge * gst/rtsp-server/rtsp-session.c: rtsp-session: Don't unref medias twice if it is removed inside gst_rtsp_session_filter() while the mutex is shortly released Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757 Part-of: 2021-09-28 10:11:15 +1000 Brad Hards * RELEASE: doc: update IRC links to OFTC Part-of: 2021-09-26 01:07:02 +0100 Tim-Philipp Müller * docs/gst_plugins_cache.json: * meson.build: Back to development Part-of: === release 1.19.2 === 2021-09-23 01:35:27 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * docs/gst_plugins_cache.json: * gst-rtsp-server.doap: * meson.build: Release 1.19.2 2021-07-05 11:54:18 +0200 Göran Jönsson * 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: 2021-06-21 08:34:35 +0000 Corentin Damman * COPYING: * COPYING.LIB: Update COPYING.LIB, COPYING files Part-of: 2021-06-01 15:29:07 +0100 Tim-Philipp Müller * docs/gst_plugins_cache.json: * meson.build: Back to development === release 1.19.1 === 2021-06-01 00:15:08 +0100 Tim-Philipp Müller * 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 * gst/rtsp-server/rtsp-stream.c: rtsp-stream: use new gst_buffer_new_memdup() Part-of: 2021-05-04 20:47:18 -0400 Doug Nazar * 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: 2021-05-04 20:45:19 -0400 Doug Nazar * 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: 2021-04-21 10:43:41 +0200 François Laignel * gst/rtsp-server/rtsp-stream.c: Use gst_element_request_pad_simple... Instead of the deprecated gst_element_get_request_pad. Part-of: 2021-04-29 03:07:42 -0400 Doug Nazar * 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: 2021-04-27 09:22:21 +0200 Marc Leeman * docs/README: docs: minor spelling correction in README Part-of: 2021-04-27 09:05:39 +0200 Marc Leeman * examples/test-replay-server.c: test-replay-server: minor spelling corrections Bumped on these while investigating the example code. Part-of: 2021-04-22 23:26:02 -0400 Doug Nazar * 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: 2021-04-23 07:18:48 +0200 Edward Hervey * 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: 2021-04-16 14:35:02 -0400 Doug Nazar * 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: 2021-03-19 10:36:01 +0200 Sebastian Dröge * 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: 2021-02-15 12:07:15 +0000 Tim-Philipp Müller * tests/check/gst/rtspclientsink.c: tests: rtspclientsink: fix some leaks Part-of: 2021-02-15 12:26:30 +0000 Tim-Philipp Müller * gst/rtsp-sink/gstrtspclientsink.c: rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy Part-of: 2021-02-15 12:07:45 +0000 Tim-Philipp Müller * tests/check/gst/rtspclientsink.c: rtspclientsink: add unit test for potential shutdown deadlock Part-of: 2021-02-15 12:01:34 +0000 Tim-Philipp Müller * 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: 2021-02-01 12:16:46 +0100 Branko Subasic * 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: 2021-01-22 08:58:23 +0100 Branko Subasic * 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: 2020-12-15 11:07:01 +0200 Sebastian Dröge * 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: 2020-12-23 13:54:54 -0500 John Lindgren * tests/check/gst/client.c: Add test cases for mountpoint of '/' Part-of: 2020-11-05 16:02:49 -0500 John Lindgren * 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://:554 or rtsp://: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: 2020-12-15 10:18:16 +0200 Sebastian Dröge * 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: 2020-12-17 15:27:27 +0100 Tobias Ronge * 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: 2020-10-21 15:38:43 +0200 Jimmi Holst Christensen * gst/rtsp-sink/gstrtspclientsink.c: rtspclientsink add proper support for uri queries Part-of: 2020-12-14 14:12:38 +1300 Lawrence Troup * 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: 2020-11-18 20:36:50 +0100 Mathieu Duponchelle * 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: 2020-11-16 10:34:41 +0200 Sebastian Dröge * 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: 2020-11-03 16:56:28 +0100 David Phung * 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: 2020-10-28 21:48:06 +0100 Kristofer Björkström * 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: 2018-10-29 09:19:33 -0400 Xavier Claessens * 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: 2020-10-19 11:25:25 +0300 Sebastian Dröge * meson.build: meson: update glib minimum version to 2.56 Part-of: 2020-09-04 21:14:35 +0200 Mathieu Duponchelle * 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: 2020-10-08 23:45:24 +0200 Mathieu Duponchelle * hooks/pre-commit.hook: * meson.build: git: use our standard pre commit hook Part-of: 2020-10-08 22:17:16 +0200 Mathieu Duponchelle * 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: 2020-10-06 00:04:17 +0200 Mathieu Duponchelle * 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: 2020-09-27 20:09:22 +0900 Seungha Yang * 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: 2020-09-28 22:03:47 +0200 David Phung * 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: 2020-09-11 15:46:41 +0200 Guiqin Zou * 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: 2020-09-05 00:30:42 +0200 Mathieu Duponchelle * 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: 2020-08-21 03:02:40 +0200 Mathieu Duponchelle * 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: 2020-08-25 16:10:36 +0200 Mathieu Duponchelle * 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: 2020-09-08 17:30:49 +0100 Tim-Philipp Müller * .gitlab-ci.yml: ci: include template from gst-ci master branch again 2020-09-08 16:58:58 +0100 Tim-Philipp Müller * docs/gst_plugins_cache.json: * meson.build: Back to development === release 1.18.0 === 2020-09-08 00:08:29 +0100 Tim-Philipp Müller * .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 * 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 * 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: 2020-08-03 19:34:30 +0300 Jordan Petridis * 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: 2020-07-15 11:19:40 +0200 Srimanta Panda * 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: 2020-07-08 17:28:57 +0100 Tim-Philipp Müller * meson.build: * scripts/extract-release-date-from-doap-file.py: meson: set release date from .doap file for releases Part-of: 2020-07-02 23:52:47 +0200 Mathieu Duponchelle * 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: 2020-07-03 02:04:04 +0100 Tim-Philipp Müller * docs/gst_plugins_cache.json: * meson.build: Back to development === release 1.17.2 === 2020-07-03 00:33:54 +0100 Tim-Philipp Müller * 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 * docs/gst_plugins_cache.json: doc: Stop documenting properties from parents 2020-06-22 20:04:45 +0300 Sebastian Dröge * docs/gst_plugins_cache.json: docs: Fix version in the plugins cache Part-of: 2020-06-22 12:33:32 +0300 Sebastian Dröge * 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: 2020-06-20 00:28:28 +0100 Tim-Philipp Müller * meson.build: Back to development === release 1.17.1 === 2020-06-19 19:24:38 +0100 Tim-Philipp Müller * 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 * 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: 2020-06-15 19:45:21 +0300 Sebastian Dröge * gst/rtsp-server/rtsp-media.c: rtsp-media: Fix misleading comment Part-of: 2020-06-15 18:29:13 +0300 Sebastian Dröge * 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: 2020-06-15 18:17:40 +0300 Sebastian Dröge * 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: 2020-06-12 15:38:45 +0200 Lenny Jorissen * examples/test-onvif-server.c: test-onvif-server: cast ntp-offset property value to 64 bit Part-of: 2020-06-09 15:21:24 -0400 Thibault Saunier * docs/gst_plugins_cache.json: docs: Update plugins cache 2020-06-10 13:45:04 +0200 Mathieu Duponchelle * 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: 2020-06-08 10:59:34 -0400 Thibault Saunier * docs/gst_plugins_cache.json: docs: Update plugins cache 2020-06-08 09:45:15 +0200 Guillaume Desmottes * 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: 2020-06-06 00:41:51 +0200 Mathieu Duponchelle * gst/rtsp-sink/gstrtspclientsink.c: plugins: uddate gst_type_mark_as_plugin_api() calls 2020-06-03 18:36:25 -0400 Thibault Saunier * docs/meson.build: doc: Require hotdoc >= 0.11.0 2020-05-27 17:00:05 +0300 Sebastian Dröge * docs/gst_plugins_cache.json: docs: Update gst_plugins_cache.json 2020-05-30 23:23:51 +0300 Sebastian Dröge * 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 * 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: 2020-05-27 17:43:43 +0100 Tim-Philipp Müller * 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: 2020-05-27 17:33:24 +0100 Tim-Philipp Müller * 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: 2020-05-27 17:32:02 +0100 Tim-Philipp Müller * 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: 2020-05-27 17:29:18 +0100 Tim-Philipp Müller * tests/check/meson.build: tests: pick up rtsp-server plugins from build directory only Part-of: 2020-05-26 15:31:22 +0200 Ludvig Rappe * 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: 2020-05-04 13:43:00 +0200 Ludvig Rappe * 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: 2020-05-25 13:49:45 +0200 Kristofer Björkström * 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: 2015-03-03 14:42:07 +0100 Gregor Boirie * 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: 2020-05-14 10:08:32 +0300 Sebastian Dröge * 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: 2020-05-03 16:29:31 +0300 Sebastian Dröge * 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: 2020-05-03 10:17:41 +0000 Tim-Philipp Müller * examples/test-onvif-server.c: examples: test-onvif-server: fix compiler warnings on raspbian Fix printf format for 64-bit variables. Part-of: 2020-05-01 10:42:17 +0300 Sebastian Dröge * 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: 2020-04-28 23:33:49 +0300 Sebastian Dröge * 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: 2020-04-28 23:16:18 +0300 Sebastian Dröge * 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: 2020-04-27 23:25:22 +0300 Sebastian Dröge * 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: 2020-04-27 19:47:15 +0300 Sebastian Dröge * 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: 2020-04-27 13:49:55 +0300 Sebastian Dröge * 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: 2020-03-23 14:51:28 +0100 Michael Olbrich * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: rtsp-media: fix default latency 2020-01-15 17:06:41 +0100 Mathieu Duponchelle * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-auth.c: rtsp-auth: fix default token leak 2019-12-09 14:17:05 +0100 Adam x Nilsson * gst/rtsp-sink/gstrtspclientsink.c: gstrtspclientsink: unref transports when closing bin Fixes #91 2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-session.c: rtsp-session: clean up comment extra-timeout 2019-10-17 12:15:42 +0200 Muhammet Ilendemli * 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 * 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 * 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 * 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 * .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 * tests/check/gst/client.c: client: fix test mem leak in attach_rate_tweaking_probe 2019-10-07 10:14:52 +0200 Göran Jönsson * tests/check/gst/media.c: media: remove memleak in test test_media_seek 2019-10-07 10:07:54 +0200 Göran Jönsson * tests/check/gst/rtspserver.c: rtspserver: Remove memleak in test test_double_play 2019-09-17 13:45:57 +0200 Adam x Nilsson * 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 * 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 * 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 * 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 * 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 * examples/test-onvif-client.c: test-onvif-client: remove unused query 2019-08-30 14:00:52 +0200 Kristofer Björkström * 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 * 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 * 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 * 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) { \ ^ :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) { \ ^ :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) { \ ^ :12:1: note: expanded from here ONVIF_IS_FACTORY ^ 2019-08-23 16:21:36 +1000 Matthew Waters * 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 * examples/test-onvif-client.c: test-onvif-client: stdin is not defined in MSVC 2019-08-12 18:03:36 +0200 Mathieu Duponchelle * gst/rtsp-server/rtsp-media.c: rtsp-media: add missing Since tag 2019-08-08 15:52:53 +0200 Mathieu Duponchelle * 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 * examples/test-onvif-server.c: test-onvif-server: downgrade logging 2019-07-27 05:14:49 +0200 Mathieu Duponchelle * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * docs/README: * gst/rtsp-server/rtsp-media-factory-uri.h: Fix typos 2019-04-05 00:48:07 +0200 Mathieu Duponchelle * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: Scale implies TRICK_MODE 2018-11-07 00:32:29 +0100 Mathieu Duponchelle * gst/rtsp-server/rtsp-client.c: client: compare booleans, not pointers to them 2018-11-13 21:28:45 +0100 Nikita Bobkov * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-sink/gstrtspclientsink.c: doc: remove xml from comments 2019-05-16 09:23:53 -0400 Thibault Saunier * 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 * docs/gst_plugins_cache.json: docs: Update plugins documentation cache 2019-04-23 12:30:02 -0400 Thibault Saunier * 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 * .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 * 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 * 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 * 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 * 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 * 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 * 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 * RELEASE: * configure.ac: * meson.build: Back to development === release 1.16.0 === 2019-04-19 00:34:54 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: * meson.build: Release 1.16.0 2019-04-15 20:33:01 +0300 Sebastian Dröge * 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 * 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 * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: * meson.build: Release 1.15.90 2019-04-10 10:32:53 +0200 Ulf Olsson * 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 * 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 * 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 * 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 * meson.build: meson: pass -Wno-unused to compiler if gstreamer debug system is disabled 2019-03-14 07:37:26 +0100 Göran Jönsson * 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 * NEWS: * RELEASE: * configure.ac: * meson.build: Back to development === release 1.15.2 === 2019-02-26 11:58:53 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: * meson.build: Release 1.15.2 2019-02-19 09:45:08 +0100 Göran Jönsson * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: * meson.build: Release 1.15.1 2018-12-05 15:07:25 +0100 Patricia Muscalu * 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 * common: Automatic update of common submodule From ed78bee to 59cb678 2018-11-20 19:12:09 +0100 Ingo Randolf * examples/test-appsrc.c: examples: test-appsrc: fix coding style error 2018-11-20 11:07:48 +0100 Ingo Randolf * examples/test-appsrc.c: examples: test-appsrc: fix buffer leak 2018-11-17 19:19:54 +0100 Patricia Muscalu * 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 * 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 * tests/check/gst/rtspserver.c: tests: rtspserver: Add shared media test case for TCP 2018-11-06 18:21:54 +0100 Linus Svensson * 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 * 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 * .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 * .gitmodules: * gst-rtsp-server.doap: Update git locations to gitlab 2018-11-01 14:20:16 +0100 Mathieu Duponchelle * gst/rtsp-server/meson.build: meson: add new onvif types 2018-11-01 12:49:51 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/Makefile.am: autotools: fix distcheck 2018-09-12 11:55:15 +0200 Ognyan Tonchev * 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 * 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 * 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 * 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 * 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 * 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 * 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 * meson_options.txt: meson: re-arrange options 2018-09-01 11:21:15 +0530 Nirbheek Chauhan * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: * tests/check/gst/stream.c: Fix indentation again 2018-07-26 12:01:16 +0200 Patricia Muscalu * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * examples/test-auth-digest.c: examples: Fix indentation 2018-07-25 15:33:18 +0200 Patricia Muscalu * 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 * 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 * 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 * 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 * gst/rtsp-server/Makefile.am: rtsp-server: Add gstreamer-base gir dir in autotools 2018-07-25 19:54:55 +0200 Mathieu Duponchelle * 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 * 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 * gst/rtsp-server/rtsp-stream.c: rtsp-stream: Slightly simplify locking 2018-06-28 11:22:21 +0200 David Svensson Fors * 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 * 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 * 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 * 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 * 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 * 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 * 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 * .gitignore: .gitignore: add another example binary 2018-07-10 23:55:20 +0100 Tim-Philipp Müller * examples/meson.build: meson: add new test-appsrc2 example to meson build 2018-07-10 23:53:41 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * docs/libs/gst-rtsp-server-sections.txt: docs: add missing auth methods 2018-06-20 00:10:18 +0200 Mathieu Duponchelle * 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 * examples/meson.build: meson: build auth-digest example 2018-06-05 08:44:44 +0200 Patricia Muscalu * 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 * 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 * 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 * 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 * 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 * meson.build: meson: use -Wl,-Bsymbolic-functions where supported Just like the autotools build. 2018-04-22 20:09:01 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 3fa2c9e to ed78bee 2018-04-04 10:06:06 +0300 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * NEWS: * RELEASE: * configure.ac: * meson.build: Back to development === release 1.14.0 === 2018-03-19 20:27:04 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: * meson.build: Release 1.13.90 2018-03-02 16:24:23 +0100 Mathieu Duponchelle * 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 * 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 * meson.build: meson: enable more warnings 2018-02-28 21:12:43 +0200 Sebastian Dröge * 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 * 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 * 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 * gst/rtsp-sink/gstrtspclientsink.c: rtspclientsink: fix extra unref of custom payloader 2018-02-26 22:57:39 +0100 Mathieu Duponchelle * gst/rtsp-sink/gstrtspclientsink.c: rspclientsink: fix recent code indentation 2018-02-26 20:27:57 +0100 Mathieu Duponchelle * gst/rtsp-sink/gstrtspclientsink.c: rtspclientsink: add missing get_type prototype 2018-02-24 03:52:15 +0100 Mathieu Duponchelle * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * configure.ac: * meson.build: Back to development === release 1.13.1 === 2018-02-15 17:15:40 +0000 Tim-Philipp Müller * NEWS: * configure.ac: * gst-rtsp-server.doap: * meson.build: Release 1.13.1 2018-02-14 17:11:19 +0100 Mathieu Duponchelle * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * tests/check/gst/rtspclientsink.c: tests: rtspclientsink: fix leak 2018-01-02 14:19:31 +0100 Branko Subasic * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From e8c7a71 to 3fa2c9e 2017-12-07 16:08:29 +0100 Edward Hervey * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 3f4aa96 to e8c7a71 2017-11-25 20:34:16 +0100 Patricia Muscalu * 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 * 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 * 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 * meson.build: meson: remove vs_module_defs_dir variable which is no longer needed 2017-11-26 14:46:05 +0000 Tim-Philipp Müller * gst/rtsp-server/rtsp-session.h: rtsp: fix distcheck 2017-11-26 12:53:42 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: rtsp-client: Don't leak addr CID #1422260 2017-11-21 09:53:08 +0100 Edward Hervey * 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 * 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 * gst/rtsp-server/rtsp-media.c: rtsp-media: Fix doc 2017-11-20 16:48:55 +0100 Edward Hervey * 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 * 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 * 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 * win32/common/libgstrtspserver.def: win32: Fix exported symbols list 2017-11-15 19:52:29 +0200 Sebastian Dröge * 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 * win32/common/libgstrtspserver.def: win32: Update exports 2017-10-23 09:49:09 +0200 Patricia Muscalu * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: rtsp-media: Initialize scalar variable CID 1418985 2017-10-06 10:27:34 +0200 Edward Hervey * win32/common/libgstrtspserver.def: win32: Update export file 2017-04-22 09:26:07 -0300 Thibault Saunier * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 48a5d85 to 3f4aa96 2017-08-14 21:04:23 +0300 Sebastian Dröge * gst/rtsp-server/rtsp-client.c: rtsp-client: Fix typo in debug message 2017-08-11 14:14:32 +0100 Tim-Philipp Müller * meson.build: meson: hide symbols by default unless explicitly exported 2017-08-10 14:20:12 +0100 Tim-Philipp Müller * 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 * 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 * 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 * meson.build: meson: Allow using glib as a subproject 2017-06-26 09:55:49 +0100 Tim-Philipp Müller * 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 * Makefile.am: Distribute meson_options.txt 2017-06-09 20:11:47 -0400 Nicolas Dufresne * Makefile.am: And config.h.meson is no longer dist either 2017-06-09 21:27:09 +0100 Tim-Philipp Müller * config.h.meson: * meson.build: meson: config.h.meson is no longer needed 2017-06-07 13:04:41 -0400 Thibault Saunier * tests/check/meson.build: * tests/meson.build: meson: Fix building tests and activate them again 2017-06-07 12:55:41 -0400 Thibault Saunier * 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 * 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 * 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 * 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 * configure.ac: * meson.build: Back to development === release 1.12.0 === 2017-05-04 15:40:46 +0300 Sebastian Dröge * 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 * 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 * common: Automatic update of common submodule From 60aeef6 to 48a5d85 2017-04-13 14:20:10 -0300 Thibault Saunier * 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 * gst/rtsp-server/meson.build: * meson.build: * meson_options.txt: meson: Build gir 2017-04-10 23:51:12 +0100 Tim-Philipp Müller * 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 * 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 * examples/test-launch.c: examples: make test-launch pipeline shared by default as well 2017-02-27 19:10:44 +0200 Sebastian Dröge * 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 * meson.build: meson: Update version 2017-02-24 15:37:49 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.11.2 === 2017-02-24 15:10:07 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.11.2 2017-02-14 20:40:26 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * meson.build: meson: bump version 2017-01-12 19:04:23 +0200 Sebastian Dröge * 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 * configure.ac: Back to development === release 1.11.1 === 2017-01-12 16:14:46 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * .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 * common: Automatic update of common submodule From f980fd9 to 39ac2f5 2016-12-02 15:40:09 +0100 Edward Hervey * 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 * 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 * 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 * 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 * 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 * 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 * .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 * 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 * 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 * .gitignore: .gitignore more 2016-11-21 13:05:50 +0100 Göran Jönsson * 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 * 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 * 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 * 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 * meson.build: meson: gstreamer gst_check_dep does not exist on windows 2016-11-17 09:43:37 -0800 Scott D Phillips * 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 * gst/rtsp-server/rtsp-sdp.c: rtsp-sdp: Fix indentation 2016-11-10 05:16:00 +0000 Neha Arora * 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 * 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 * meson.build: meson: update version === release 1.11.0 === 2016-11-01 18:53:15 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.10.0 === 2016-11-01 18:06:46 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.10.0 2016-10-28 18:38:01 +0100 Tim-Philipp Müller * 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 * 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 * 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 * meson.build: meson: add fallbacks for gst modules For gst-all. 2016-09-14 17:48:39 +0300 Nikita Bobkov * 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 * 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 * meson.build: meson: update version === release 1.9.90 === 2016-09-30 13:04:12 +0300 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.9.90 2016-09-17 13:17:19 +0100 Ian Jamison * 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 * 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 * configure.ac: configure: Depend on gstreamer 1.9.2.1 2016-09-10 20:52:31 +1000 Jan Schmidt * autogen.sh: * common: Automatic update of common submodule From b18d820 to f980fd9 2016-09-10 09:58:31 +1000 Jan Schmidt * autogen.sh: * common: Automatic update of common submodule From 6f2d209 to b18d820 2016-09-07 18:44:34 +0300 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-stream.c: rtsp-stream: Fix up various multicast related issues 2016-09-05 13:40:59 +0300 Sebastian Dröge * tests/check/gst/stream.c: tests: Fix compilation 2016-07-28 15:33:05 -0400 Xavier Claessens * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * configure.ac: Back to development === release 1.9.2 === 2016-09-01 12:32:51 +0300 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.9.2 2016-01-27 01:03:52 +0000 Tim-Philipp Müller * 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 * 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 * gst/rtsp-server/rtsp-client.c: rtsp-client: Fix leaking of media in error cases With additional fixes by Kseniya Vasilchuk 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 * 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 * common: Automatic update of common submodule From f363b32 to f49c55e 2016-07-06 13:51:15 +0300 Sebastian Dröge * configure.ac: Back to development === release 1.9.1 === 2016-07-06 13:28:12 +0300 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.9.1 2016-06-24 02:02:20 +0530 Nirbheek Chauhan * 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 * common: Automatic update of common submodule From ac2f647 to f363b32 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 6f2d209 to ac2f647 2016-04-06 10:09:46 +0200 Patricia Muscalu * 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 * 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 * 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 * gst/rtsp-server/Makefile.am: rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS) 2015-12-30 18:39:05 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * configure.ac: Back to development === release 1.8.0 === 2016-03-24 13:00:35 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.8.0 2016-03-16 23:35:09 +0200 Sebastian Dröge * 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 * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.7.91 2016-03-10 13:54:38 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.7.90 2016-02-26 12:42:51 +0200 Sebastian Dröge * common: Automatic update of common submodule From b64f03f to 6f2d209 2016-02-24 00:10:52 +1100 Jan Schmidt * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * configure.ac: Back to development === release 1.7.2 === 2016-02-19 12:03:18 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.7.2 2016-02-18 15:20:05 +0000 Julien Isorce * 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 * 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 * 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 * 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 * autogen.sh: * common: Automatic update of common submodule From 86e4663 to b64f03f 2016-02-02 09:01:51 +0100 Steven Hoving * 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 * 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 * 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 * .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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * examples/test-netclock.c: test-netclock: Use the new API to configure a clock directly 2015-12-30 16:31:13 +0200 Sebastian Dröge * 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 * 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 * gst/rtsp-server/rtsp-media-factory.c: rtsp-media-factory: Add FIXME for 2.0 2015-12-30 16:29:45 +0200 Sebastian Dröge * gst/rtsp-server/rtsp-stream.c: rtsp-stream: Fix indentation 2015-12-22 12:08:02 +0100 Sebastian Rasmussen * 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 * 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 * configure.ac: Back to development === release 1.7.1 === 2015-12-24 14:54:06 +0100 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.7.1 2015-12-21 00:43:49 +0100 Koop Mast * 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 * 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 * 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 * 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 * autogen.sh: * common: Automatic update of common submodule From b319909 to 86e4663 2015-11-18 11:14:39 +0100 Srimanta Panda * 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 * 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 * 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 * 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 * 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 * 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 * 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 * examples/.gitignore: * examples/Makefile.am: * examples/test-record-auth.c: Add test-record-auth example 2015-11-17 01:12:28 +1100 Jan Schmidt * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From b99800a to b319909 2015-10-20 17:29:42 +0300 Sebastian Dröge * 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 * common: Automatic update of common submodule From 6babecd to b99800a 2015-10-02 22:25:47 +0300 Sebastian Dröge * configure.ac: Update GLib dependency to 2.40.0 2015-10-02 16:11:05 +0900 Hyunjun Ko * 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 * common: common: update for new suppression Makes check-valgrind pass with glib 2.46 2015-09-28 17:40:59 +0200 Sebastian Rasmussen * 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 * 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 * configure.ac: Back to development === release 1.6.0 === 2015-09-25 23:32:52 +0200 Sebastian Dröge * 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 * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.5.91 2015-09-17 20:07:34 +0100 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095 2015-08-31 22:47:45 +1000 Jan Schmidt * 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 * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.5.90 2015-08-12 14:33:44 +0900 Hyunjun Ko * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From f74b2df to 9aed1d7 2015-06-25 00:04:28 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.5.2 === 2015-06-24 23:44:37 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.5.2 2015-06-18 13:12:04 +0200 Ognyan Tonchev * 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 https://bugzilla.gnome.org/show_bug.cgi?id=749417 2015-06-16 17:50:26 -0400 Nicolas Dufresne * common: Automatic update of common submodule From 6015d26 to f74b2df 2015-06-11 17:39:00 +0200 Ognyan Tonchev * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From d9a3353 to 6015d26 2015-06-08 23:08:34 +0200 Stefan Sauer * common: Automatic update of common submodule From d37af32 to d9a3353 2015-06-07 23:07:31 +0200 Stefan Sauer * common: Automatic update of common submodule From 21ba2e5 to d37af32 2015-06-07 17:32:29 +0200 Stefan Sauer * common: Automatic update of common submodule From c408583 to 21ba2e5 2015-06-07 17:06:40 +0200 Stefan Sauer * 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 * common: Automatic update of common submodule From 44a3517 to c408583 2015-06-07 16:44:55 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.5.1 === 2015-06-07 11:20:01 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.5.1 2015-05-25 16:36:18 +0200 Göran Jönsson * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: rtsp-media: Mark some more functions static 2015-05-05 16:46:19 +0200 Sebastian Dröge * 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 * examples/test-video-rtx.c: examples: Use AVPF profile for the RTX example 2015-05-04 16:31:20 +0200 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From bc76a8b to c8fb372 2015-03-23 21:03:20 +0100 Sebastian Dröge * 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 * 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 * 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 * 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 * README: Fix typo in README 2015-03-10 09:39:22 +0000 Tim-Philipp Müller * gst/rtsp-server/rtsp-media-factory.c: * tests/check/gst/client.c: Fix double semicolons 2015-03-09 16:00:07 +0100 Sebastian Dröge * 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 * 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 * 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 should not be included. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335 Based on the patch for rtspsrc by Linus Svensson 2014-02-26 22:34:06 +0100 Linus Svensson * 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: -. The - part should be added to the caps in a payloader and the 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-stream.h: rtsp-stream: minor code formatting fix 2015-02-10 16:39:58 +0000 Luis de Bethencourt * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: rtsp-client: fix awkward if clause 2015-02-06 19:34:17 +0000 Tim-Philipp Müller * 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 * 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 * examples/.gitignore: examples: add new test-record to .gitignore 2015-01-28 18:54:01 +0100 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: rtsp-client: log interleaved data received 2015-01-19 20:18:20 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From f2c6b95 to bc76a8b 2014-12-31 13:04:57 +0000 Tim-Philipp Müller * Makefile.am: Fix 'make check' from top-level directory 2014-12-30 18:13:49 +0530 Nirbheek Chauhan * 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 * 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 * common: Automatic update of common submodule From ef1ffdc to f2c6b95 2014-12-17 20:02:05 +0100 Sebastian Rasmussen * 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 * 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 * gst/rtsp-server/rtsp-stream.c: rtsp-stream: Fix some minor memory leaks 2014-12-16 16:46:06 +0100 Sebastian Dröge * gst/rtsp-server/rtsp-media.c: rtsp-media: Some minor cleanup 2014-12-16 16:42:13 +0100 Sebastian Dröge * 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 * 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 * 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 * common: Automatic update of common submodule From 7bb2bce to ef1ffdc 2014-11-07 12:48:53 +0100 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: refactor cleanup of cached media 2014-10-23 13:39:10 +0200 Linus Svensson * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 84d06cd to 7bb2bce 2014-10-24 17:48:04 +0100 Tim-Philipp Müller * Makefile.am: Parallelise 'make check-valgrind' 2014-10-21 13:04:14 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From a8c8939 to 84d06cd 2014-10-21 13:00:49 +0200 Stefan Sauer * common: Automatic update of common submodule From 36388a1 to a8c8939 2014-10-01 07:12:30 -0400 Vincent Penquerc'h * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: rtsp-media: Set state to UNPREPARING in all cases 2014-09-30 19:17:04 +0200 Ognyan Tonchev * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * examples/test-mp4.c: test: add example of dumping RTCP reports 2014-09-08 09:26:23 +0200 Srimanta Panda * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: signals: Fix copy-pasto in target-state signal offset 2014-08-01 10:46:44 +0200 Edward Hervey * 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 * 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 * configure.ac: Back to development === release 1.4.0 === 2014-07-19 17:56:31 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.4.0 2014-07-16 20:39:42 +0900 Hyunjun Ko * 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 * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.3.91 2014-07-10 17:37:45 +0200 Wim Taymans * docs/libs/gst-rtsp-server-sections.txt: docs: update docs 2014-07-10 17:10:06 +0200 Wim Taymans * gst/rtsp-server/rtsp-server.c: server: implement client REMOVE filter 2014-07-10 17:05:13 +0200 Wim Taymans * 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 * 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 * 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 * 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 * tests/check/gst/client.c: client tests: send teardown to cleanup session 2014-07-09 14:17:46 +0200 Ognyan Tonchev * tests/check/gst/rtspserver.c: server tests: send teardown to cleanup session 2014-07-09 15:01:31 +0200 Ognyan Tonchev * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: fix build 2014-07-01 14:41:14 +0200 Wim Taymans * 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 * 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 See https://bugzilla.gnome.org/show_bug.cgi?id=732226 2014-06-30 15:14:34 +0200 Wim Taymans * 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 * 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 * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.3.90 2014-06-27 16:54:22 +0200 Wim Taymans * gst/rtsp-server/rtsp-stream.c: stream: crypto can be NULL 2014-06-11 16:42:08 -0700 Evan Nemerson * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-context.h: context: add type macro 2014-06-24 09:34:50 +0200 Wim Taymans * 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 * configure.ac: Back to development === release 1.3.3 === 2014-06-22 19:36:14 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.3.3 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: Destroy the rtsp watch after connection close 2014-06-13 16:46:06 +0200 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: fix confusing comment 2014-05-27 12:36:52 +0200 Göran Jönsson * 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 * configure.ac: Back to development === release 1.3.2 === 2014-05-21 13:06:36 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * common: * configure.ac: * gst-rtsp-server.doap: Release 1.3.2 2014-05-21 10:54:05 +0200 Sebastian Dröge * common: Automatic update of common submodule From 211fa5f to 1f5d3c3 2014-05-20 15:57:30 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * configure.ac: Back to development === release 1.3.1 === 2014-05-03 18:40:24 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-rtsp-server.doap: Release 1.3.1 2014-05-03 10:18:00 +0200 Sebastian Dröge * common: Automatic update of common submodule From bcb1518 to 211fa5f 2014-05-02 19:58:15 +0100 Tim-Philipp Müller * .gitignore: Update .gitignore 2014-05-02 19:57:23 +0100 Tim-Philipp Müller * tests/check/gst/sessionmedia.c: tests: fix memory leak in sessionmedia unit test 2014-05-01 06:17:06 +0200 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: fix typo in comment 2014-04-14 15:17:14 +0200 Ognyan Tonchev * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * tests/check/gst/media.c: media test: cleanup the thread pool in tests 2014-04-01 13:16:26 +0200 Linus Svensson * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: cleanup error paths 2014-04-02 12:27:24 +0200 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: fix docs 2014-03-25 12:42:39 +0100 Wim Taymans * 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 * examples/test-video.c: test: print an error on failure 2014-03-13 17:35:21 +0100 Wim Taymans * 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 * 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 * 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 * 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 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * .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 * gst/rtsp-server/rtsp-stream.c: stream: release some locks in error cases 2014-03-02 05:12:10 +0100 Sebastian Rasmussen 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 * common: Automatic update of common submodule From fe1672e to bcb1518 2014-02-26 22:15:51 +0100 Stefan Sauer * common: Automatic update of common submodule From 1a07da9 to fe1672e 2014-02-25 15:13:40 +0000 Tim-Philipp Müller * examples/Makefile.am: examples: use LDADD for libs instead of LDFLAGS 2014-02-25 14:42:09 +0000 Tim-Philipp Müller * configure.ac: configure: make sure releases are in .doap file 2014-02-25 14:11:00 +0000 Tim-Philipp Müller * 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 * examples/.gitignore: examples: add cgroup test binary to .gitignore 2014-02-25 14:06:47 +0000 Tim-Philipp Müller * 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 * .gitignore: .gitignore: ignore temp files created in the course of 'make check' 2014-02-18 09:44:34 +0100 Branko Subasic * 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 * 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 * 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 * 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 * 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 * Makefile.am: build: Ship gst-rtsp-server.doap file 2014-02-09 10:47:09 +0100 Sebastian Dröge * tests/check/gst/rtspserver.c: tests: Fix another compiler warning with gcc 2014-02-09 10:45:28 +0100 Sebastian Dröge * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-stream.c: stream: handle NULL seqnum and rtptime arguments 2014-01-31 15:02:22 +0100 Ognyan Tonchev * 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 * 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 * common: Automatic update of common submodule From f7bc1c3 to 1a07da9 2014-01-28 14:51:26 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-stream.c: stream: fix compilation 2014-01-07 12:21:09 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 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 * 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 * 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 * autogen.sh: * common: * configure.ac: Fix broken gettext setup which is not used anyway 2013-12-22 22:36:06 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From dbedaa0 to d48bed3 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué * 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 * 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 * 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 * 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 * .gitignore: Make git ignore more unit test binaries 2013-12-12 00:36:07 +0000 Tim-Philipp Müller * 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 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 * 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 * gst/rtsp-server/rtsp-stream-transport.c: stream-transport: free url in finalize 2013-11-29 15:50:23 +0100 Ognyan Tonchev * gst/rtsp-server/rtsp-media.c: media: also do state change in suspended state 2013-11-29 10:53:08 +0100 Wim Taymans * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: suspend after SDP and unsuspend before PLAYING Based on patches by Ognyan Tonchev Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257 2013-11-28 14:10:19 +0100 Wim Taymans * 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 See https://bugzilla.gnome.org/show_bug.cgi?id=711257 2013-11-28 14:06:53 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-media.c: media: refactor starting and waiting for preroll Based on patches from Ognyan Tonchev See https://bugzilla.gnome.org/show_bug.cgi?id=711257 2013-11-28 13:42:21 +0100 Wim Taymans * 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 See https://bugzilla.gnome.org/show_bug.cgi?id=711257 2013-11-27 15:42:45 +0100 Edward Hervey * 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 * gst/rtsp-server/rtsp-media.c: media: move default implementations to where they are used 2013-11-26 16:25:37 +0100 Wim Taymans * 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 * 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 * 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 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 * 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 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 * 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 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-auth.c: auth: small typos 2013-10-19 19:25:27 +0200 Sebastian Rasmussen * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 865aa20 to dbedaa0 2013-10-14 12:03:07 +0200 Ognyan Tonchev * 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 * 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 * 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 * .gitignore: gitignore: Add new test binary 2013-10-09 15:19:12 +0200 Ognyan Tonchev * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: rtsp-media: remove old line 2013-10-01 13:15:19 +0200 Jonas Holmberg * 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 * 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 * gst/rtsp-server/rtsp-media.c: media: Return FALSE if seeking is not supported 2013-10-01 17:16:11 +0200 Wim Taymans * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 6b03ba7 to 865aa20 2013-09-23 14:28:04 +0200 Jonas Holmberg * 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 * 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 * 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 * common: Automatic update of common submodule From b613661 to 6b03ba7 2013-09-19 18:46:14 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From 74a6857 to b613661 2013-09-19 17:39:24 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From 01a7a46 to 74a6857 2013-09-19 15:44:26 +0200 Jonas Holmberg * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: map URL to path in requests 2013-09-16 16:47:40 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * examples/Makefile.am: Makefile: add rule for appsrc example 2013-08-23 15:14:29 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: media: set protocols on streams 2013-08-16 16:16:31 +0200 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: use protocols supported by stream 2013-08-16 16:16:00 +0200 Wim Taymans * 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 * gst/rtsp-server/rtsp-media.c: media: configure protocols in new streams 2013-08-16 16:08:43 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.h: small documentation fix 2013-07-11 16:11:55 -0400 Youness Alaoui * gst/rtsp-server/rtsp-client.c: Do not take range header if range is invalid 2013-08-02 16:57:26 +0200 Wim Taymans * 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 * 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 * 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 * examples/test-cgroups.c: tests: add some more docs 2013-07-22 14:25:04 +0200 Wim Taymans * 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 * 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 * examples/test-auth.c: * gst/rtsp-server/rtsp-auth.c: auth: use defines when possible 2013-07-18 11:44:21 +0200 Wim Taymans * gst/rtsp-server/rtsp-address-pool.c: address-pool: improve docs 2013-07-18 12:26:45 +0200 Wim Taymans * gst/rtsp-server/rtsp-permissions.c: permissions: add the role to the copy 2013-07-17 19:35:33 -0400 Olivier Crête * gst/rtsp-server/rtsp-permissions.c: permissions: Also copy the roles 2013-07-17 19:32:09 -0400 Olivier Crête * gst/rtsp-server/rtsp-permissions.c: permissions: Make it build 2013-07-16 12:36:56 +0200 Wim Taymans * gst/rtsp-server/rtsp-address-pool.h: docs: small fixes 2013-07-16 12:32:51 +0200 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-permissions.c: permissions: implement _remove_role 2013-07-15 17:12:43 +0200 Wim Taymans * gst/rtsp-server/rtsp-permissions.c: permissions: update docs 2013-07-15 16:48:37 +0200 Wim Taymans * 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 * 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 * docs/README: README: update readme 2013-07-15 15:25:00 +0200 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-server.c: server: add small debug 2013-07-15 11:58:58 +0200 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-mount-points.c: mount-points: add some debug 2013-07-12 17:26:55 +0200 Wim Taymans * tests/check/gst/client.c: tests: almost fix test 2013-07-12 17:07:53 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-auth.c: auth: fix typo 2013-07-12 15:19:29 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-mount-points.c: mount-points: fix debug 2013-07-11 17:28:17 +0200 Wim Taymans * tests/check/gst/media.c: tests: fix media test 2013-07-11 17:28:04 +0200 Wim Taymans * gst/rtsp-server/rtsp-thread-pool.c: thread-pool: we don't require a state 2013-07-11 17:18:58 +0200 Wim Taymans * 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 * tests/check/gst/client.c: tests: fix client test 2013-07-11 16:57:14 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * configure.ac: * examples/Makefile.am: * examples/test-cgroups.c: examples: add cgroups example 2013-07-10 20:55:03 +0200 Wim Taymans * tests/check/gst/rtspserver.c: tests: fix compilation 2013-07-10 20:48:47 +0200 Wim Taymans * gst/rtsp-server/rtsp-thread-pool.c: thread-pool: fix vmethod invocation 2013-07-10 20:48:18 +0200 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-server.c: server: configure client thread pool 2013-07-10 17:06:36 +0200 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-auth.c: auth: debug authorization check 2013-07-09 20:44:51 +0200 Wim Taymans * 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 * gst/rtsp-server/rtsp-media-factory.c: factory: pass permissions to media by default 2013-07-09 16:09:07 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/Makefile.am: rtsp-server: Allow building of static library 2013-07-02 15:59:16 +0200 Wim Taymans * tests/check/gst/mediafactory.c: tests: fix compilation 2013-07-02 15:54:43 +0200 Wim Taymans * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: fix comment 2013-07-01 14:45:49 +0200 Wim Taymans * 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 * gst/rtsp-server/rtsp-stream.c: stream: improve docs 2013-07-01 12:04:45 +0200 Wim Taymans * gst/rtsp-server/rtsp-stream-transport.c: stream-transport: remove old if 0 block 2013-06-27 11:21:42 +0200 Patricia Muscalu * 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 * 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 * 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 * gst/rtsp-server/rtsp-media-factory.h: media-factory: fix typo 2013-06-26 14:42:15 +0200 Ognyan Tonchev * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: media: fix docs 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 098c0d7 to 01a7a46 2013-05-29 13:45:00 +0200 David Svensson Fors * 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 * 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 * 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 * 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 * 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 * 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 * examples/Makefile.am: * examples/test-video.c: example: add TLS example 2013-05-31 11:42:36 +0200 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-server.c: server: don't crash on NULL GError 2013-05-30 10:46:33 +0200 Patricia Muscalu * 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 * gst/rtsp-server/rtsp-client.c: client: improve debug 2013-05-30 07:18:22 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 5edcd85 to 098c0d7 2013-04-23 11:28:39 +0200 Ognyan Tonchev * tests/check/gst/media.c: tests: add dynamic payloader prepare/unprepare check 2013-04-23 10:27:35 +0200 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: release lock when removing fakesink 2013-04-23 10:16:17 +0200 Wim Taymans * 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 * common: Automatic update of common submodule From 3cb3d3c to 5edcd85 2013-04-22 17:34:37 +0200 Wim Taymans * 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 * examples/test-sdp.c: tests: add example of reusable pipelines 2013-04-22 17:32:31 +0200 Ognyan Tonchev * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From aed87ae to 3cb3d3c 2013-04-12 11:34:38 +0100 Wim Taymans * 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 * 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 * common: Automatic update of common submodule From 04c7a1e to aed87ae 2013-04-09 20:39:58 +0200 Wim Taymans * 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 * gst/rtsp-server/rtsp-media.c: media: small cleanup 2013-03-20 12:33:54 +0100 David Svensson Fors * 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 * 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 * 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 * 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 * 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 * 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 * configure.ac: configure: remove out-of-date comment 2013-03-12 00:05:49 +0000 Tim-Philipp Müller * .gitignore: .gitignore: ignore more build files 2013-03-12 00:03:36 +0000 Tim-Philipp Müller * tests/check/Makefile.am: tests: use right _LIBS variable for gst-plugins-base libs 2013-03-11 11:35:14 +0100 Wim Taymans * tests/check/Makefile.am: check: add librtp to libs 2013-02-20 19:37:51 -0500 Olivier Crête * 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 * tests/check/gst/rtspserver.c: tests: Make sure packets are actually received 2013-02-19 18:27:20 -0500 Olivier Crête * gst/rtsp-server/rtsp-stream.c: stream: Select unicast address from pool if appropriate 2013-02-19 16:43:08 -0500 Olivier Crête * gst/rtsp-server/rtsp-stream.c: stream: Properties are always there in Gst 1.0 2013-02-19 16:36:20 -0500 Olivier Crête * tests/check/gst/addresspool.c: tests: Add tests for unicast addresses in pool 2013-02-20 14:26:03 -0500 Olivier Crête * 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 * 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 * 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 * 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 * tests/check/gst/rtspserver.c: tests: Add test for client disconnection 2013-02-18 20:15:41 -0500 Olivier Crête * tests/check/gst/rtspserver.c: tests: Test client and session timeouts with multiple threads 2013-02-18 14:59:58 -0500 Olivier Crête * 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 * tests/check/gst/rtspserver.c: tests: Test that slow DESCRIBE don't block other clients 2013-02-14 19:52:09 -0500 Olivier Crête * tests/check/gst/client.c: tests: Add tests for client-requested multicast address 2013-02-14 13:44:54 -0500 Olivier Crête * 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 * 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 * 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 * 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 * 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 * tests/check/gst/addresspool.c: tests: Add more tests for the address pool 2013-02-12 16:29:25 -0500 Olivier Crête * 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 * common: Automatic update of common submodule From 2de221c to 04c7a1e 2013-02-07 16:18:08 -0600 George McCollister * 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 * common: Automatic update of common submodule From a942293 to 2de221c 2013-01-28 10:31:50 +0100 Wim Taymans * 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 * 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 * common: Automatic update of common submodule From acb04d9 to a942293 2012-12-14 11:58:29 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: media: improve debug 2012-11-30 12:53:02 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-media.c: media: use g_object_ref/unref for GObjects 2012-11-30 07:05:25 +0100 Alessandro Decina * 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 * gst/rtsp-server/rtsp-media.c: Fix compiler warning 2012-11-30 06:14:49 +0100 Alessandro Decina * 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 * gst/rtsp-server/rtsp-session-media.h: small cleanup 2012-11-29 17:20:56 +0100 Wim Taymans * gst/rtsp-server/rtsp-media.c: * tests/check/gst/media.c: media: avoid element leak 2012-11-29 17:20:26 +0100 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: require an element in media constructor 2012-11-29 17:07:30 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-client.c: client: TEARDOWN brings that state to Init again 2012-11-29 11:11:05 +0100 Wim Taymans * 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 * tests/check/Makefile.am: * tests/check/gst/media.c: tests: add media tests 2012-11-28 14:45:30 +0100 Wim Taymans * 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 * tests/check/Makefile.am: * tests/check/gst/mediafactory.c: tests: add mediafactory test 2012-11-28 12:40:18 +0100 Wim Taymans * gst/rtsp-server/rtsp-stream.c: stream: improve debug 2012-11-28 12:39:37 +0100 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-media-factory.c: media-factory: require an url 2012-11-28 11:40:33 +0100 Wim Taymans * examples/test-uri.c: examples: fix include 2012-11-28 11:17:27 +0100 Wim Taymans * gst/rtsp-server/rtsp-server.h: server: remove unused include 2012-11-28 11:07:57 +0100 Wim Taymans * tests/check/Makefile.am: * tests/check/gst/mountpoints.c: tests: add test for mountpoints 2012-11-28 11:05:08 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-mount-points.c: mounts: add g_return_if guards 2012-11-27 12:51:55 +0100 Wim Taymans * tests/check/gst/client.c: tests: add more tests 2012-11-27 12:33:02 +0100 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: improve debug 2012-11-27 12:24:21 +0100 Wim Taymans * 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 * common: update common 2012-11-27 12:13:59 +0100 Wim Taymans * tests/check/gst/client.c: test: add test for session in options request 2012-11-27 12:11:41 +0100 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: small cleanup 2012-11-26 17:34:35 +0100 Wim Taymans * gst/rtsp-server/rtsp-client.h: client: remove unused include 2012-11-26 17:34:24 +0100 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: fix compilation 2012-11-26 17:28:29 +0100 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: call destroy without the lock 2012-11-26 17:20:39 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-client.c: * gst/rtsp-server/rtsp-client.h: client: small cleanup 2012-11-26 16:39:26 +0100 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: log more errors 2012-11-26 13:35:48 +0100 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: fix compilation 2012-11-26 13:16:59 +0100 Wim Taymans * 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 * 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 * configure.ac: configure: bump required version of -base 2012-11-21 17:21:28 +0100 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: fix seeking 2012-11-21 16:41:56 +0100 Wim Taymans * 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 * examples/test-launch.c: examples: fix whitespace 2012-11-20 13:34:46 +0100 Wim Taymans * 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 * examples/test-uri.c: test-uri: remove mapping example 2012-11-20 12:47:20 +0100 Wim Taymans * examples/test-uri.c: test-uri: fix callback signature 2012-11-20 12:29:55 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-media-factory-uri.c: factory-uri: add some debug 2012-11-20 12:24:13 +0100 Wim Taymans * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-server.c: server: fix small leak 2012-11-20 09:42:51 +0100 Wim Taymans * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 6bb6951 to a72faea 2012-11-17 00:11:27 +0000 Tim-Philipp Müller * 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 * 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 * 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 * 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 * examples/test-multicast.c: test: set shared 2012-11-15 16:18:29 +0100 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: media: configure address pool in new streams 2012-11-15 15:36:21 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-address-pool.c: address-pool: small cleanups 2012-11-14 15:50:42 +0100 Wim Taymans * tests/check/Makefile.am: * tests/check/gst/addresspool.c: tests: add addresspool unit test 2012-11-14 15:49:06 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-server.c: server: set default max-threads property 2012-11-13 11:54:17 +0100 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-stream.c: * gst/rtsp-server/rtsp-stream.h: stream: add locking 2012-11-12 17:11:18 +0100 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-session.c: * gst/rtsp-server/rtsp-session.h: session: add locking 2012-11-12 16:30:16 +0100 Wim Taymans * gst/rtsp-server/rtsp-server.c: server: free old socket 2012-11-12 16:18:57 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-media-factory.c: media-factory: add locking 2012-11-12 16:03:21 +0100 Wim Taymans * gst/rtsp-server/rtsp-auth.c: * gst/rtsp-server/rtsp-auth.h: auth: add locking 2012-11-12 15:53:28 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: small cleanup 2012-10-27 23:49:24 +0200 Wim Taymans * 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 * gst/rtsp-server/rtsp-client.c: client: refactor transport parsing 2012-10-27 21:05:03 +0200 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-stream.c: stream: transports must already have been removed 2012-10-26 17:28:10 +0200 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-media.c: media: signal unprepared when we actually finish 2012-10-26 15:19:23 +0200 Wim Taymans * 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 * 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 * 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 * docs/README: docs: update docs 2012-10-26 11:24:55 +0100 Tim-Philipp Müller * configure.ac: configure: bump version number after refactoring 2012-10-25 21:29:58 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/Makefile.am: Explicitly link against gio. Fix link error on mac. 2012-10-10 11:13:10 +0200 Ognyan Tonchev * 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 * 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 * common: Automatic update of common submodule From 6c0b52c to 6bb6951 2012-10-01 16:13:50 +0200 Patricia Muscalu * 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 * common: Automatic update of common submodule From 4f962f7 to 6c0b52c 2012-09-10 16:25:57 +0200 Ognyan Tonchev * gst/rtsp-server/rtsp-media.c: media: fix check for seekability 2012-09-07 17:14:30 +0200 Wim Taymans * 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 * gst/rtsp-server/rtsp-server.c: server: remove obsolete includes 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque 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 * 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 * 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 * common: Automatic update of common submodule From 668acee to 4f962f7 2012-08-15 15:54:32 +0200 Patricia Muscalu * 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 * common: Automatic update of common submodule From 94ccf4c to 668acee 2012-07-18 15:54:49 +0200 Patricia Muscalu * 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 * common: Automatic update of common submodule From 98e386f to 94ccf4c 2012-07-10 11:39:58 +0200 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: fix docs 2012-07-03 18:06:00 +0200 Ognyan Tonchev * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: don't leak transports 2012-06-12 14:45:39 +0200 David Svensson Fors * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * common: Automatic update of common submodule From 03a0e57 to 98e386f 2012-06-06 18:20:49 +0200 Edward Hervey * common: Automatic update of common submodule From 1fab359 to 03a0e57 2012-06-06 14:49:02 +0200 David Svensson Fors * 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 * common: Automatic update of common submodule From f1b5a96 to 1fab359 2012-05-31 13:11:43 +0200 Sebastian Dröge * common: Automatic update of common submodule From 92b7266 to f1b5a96 2012-05-30 12:48:51 +0200 Sebastian Dröge * common: Automatic update of common submodule From ec1c4a8 to 92b7266 2012-05-30 11:27:31 +0200 Sebastian Dröge * common: Automatic update of common submodule From 3429ba6 to ec1c4a8 2012-05-22 15:37:25 +0200 David Svensson Fors * 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 * common: Automatic update of common submodule From dc70203 to 3429ba6 2012-05-11 09:42:47 +0200 Wim Taymans * 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 * common: Automatic update of common submodule From 6db25be to dc70203 2012-04-13 15:27:22 +0200 Sebastian Dröge * 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 * 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 * common: Automatic update of common submodule From 464fe15 to 6db25be 2012-04-05 18:45:43 +0200 Sebastian Dröge * common: Automatic update of common submodule From 7fda524 to 464fe15 2012-04-04 14:45:55 +0200 Sebastian Dröge * 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 Merge remote-tracking branch 'origin/0.10' Conflicts: gst/rtsp-server/rtsp-session-pool.c 2012-03-27 10:13:20 +0200 Sebastian Dröge * gst/rtsp-server/rtsp-session-pool.c: rtsp-server: Don't use deprecated GLib API 2012-03-26 12:23:36 +0200 Wim Taymans Replace master with 0.11 2012-03-26 12:22:05 +0200 Wim Taymans Merge branch 'master' into 0.11 2012-03-26 12:20:51 +0200 Wim Taymans Merge branch 'master' into 0.11 2012-03-19 10:48:09 +0000 Vincent Penquerc'h * docs/README: A couple minor typo fixes 2012-03-13 18:10:53 +0100 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: fix state of the appqueue 2012-03-13 16:06:50 +0100 Wim Taymans * gst/rtsp-server/rtsp-media-factory-uri.c: factory: use videoconvert 2012-03-13 16:02:47 +0100 Wim Taymans * gst/rtsp-server/rtsp-media-factory-uri.c: factory: change to new style caps 2012-03-07 15:03:55 +0100 Wim Taymans * 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 * configure.ac: configure: fix build 2012-02-29 15:56:06 +0000 Tim-Philipp Müller * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: rtsp-client: update for new map API 2012-02-13 10:37:37 +0000 Tim-Philipp Müller * .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 * common: Automatic update of common submodule From c463bc0 to 7fda524 2012-01-25 11:40:59 +0100 Sebastian Dröge * common: Automatic update of common submodule From 2a59016 to c463bc0 2012-01-18 16:48:41 +0100 Sebastian Dröge * common: Automatic update of common submodule From 0807187 to 2a59016 2012-01-04 19:56:02 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 11f0cd5 to 0807187 2011-12-09 11:00:46 +0100 Wim Taymans * examples/test-auth.c: example: update for new caps 2011-12-09 10:53:30 +0100 Wim Taymans * 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 Merge branch 'master' into 0.11 2011-11-03 16:06:23 +0100 Fabian Deutsch * 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 2011-11-03 16:06:23 +0100 Fabian Deutsch * 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 2011-11-03 12:58:42 +0100 Wim Taymans Merge branch 'master' into 0.11 2011-11-03 12:55:24 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-media.c: Disallow seek in live media 2011-11-03 11:58:42 +0100 Wim Taymans Merge branch 'master' into 0.11 2011-11-03 10:48:40 +0100 mat * gst/rtsp-server/rtsp-server.c: #ifdef statements for windows socket creation were missing 2011-09-06 21:53:46 +0200 Stefan Sauer * common: Automatic update of common submodule From a39eb83 to 11f0cd5 2011-09-06 16:07:18 +0200 Stefan Sauer * common: Automatic update of common submodule From 605cd9a to a39eb83 2011-08-16 16:39:26 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-08-16 16:07:04 +0200 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: use method to access property 2011-08-16 15:15:19 +0200 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: use method to access property 2011-08-16 15:15:19 +0200 Wim Taymans * 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 * 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 Merge branch 'master' into 0.11 2011-08-16 13:43:44 +0200 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: use media multicast group 2011-08-16 13:37:50 +0200 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-media-factory.c: media-factory: configure multicast in media 2011-08-16 13:25:16 +0200 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: use media multicast group 2011-08-16 13:37:50 +0200 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-media-factory.c: media-factory: configure multicast in media 2011-08-16 13:25:16 +0200 Wim Taymans * 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 * 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 * 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 Merge branch 'master' into 0.11 2011-08-16 12:09:48 +0200 Robert Krakora * 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 Merge branch 'master' into 0.11 2011-06-30 10:13:59 +0200 Emmanuel Pacaud * 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 Merge branch 'master' into 0.11 2011-08-11 18:07:08 -0700 David Schleef * gst/rtsp-server/rtsp-server.c: rtsp-server: hold on to reference while using object 2011-08-04 08:59:17 +0200 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: use new api 2011-08-04 08:58:58 +0200 Wim Taymans * configure.ac: configure: use unstable api 2011-06-27 11:26:26 -0700 David Schleef * gst/rtsp-server/rtsp-client.c: client: fix reference counting 2011-07-20 17:16:42 +0200 Thijs Vermeir * 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 * 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 * common: Automatic update of common submodule From 69b981f to 605cd9a 2011-06-13 19:05:57 +0200 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: update for buffer API change 2011-06-07 10:54:26 +0200 Edward Hervey * gst/rtsp-server/Makefile.am: Makefile.am: 0.10 => @GST_MAJORMINOR@ 2011-06-07 10:59:16 +0200 Edward Hervey * 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 * gst/rtsp-server/.gitignore: .gitignore: 0.10 => 0.11 2011-06-07 10:54:26 +0200 Edward Hervey * gst/rtsp-server/Makefile.am: Makefile.am: 0.10 => @GST_MAJORMINOR@ 2011-05-24 18:26:06 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-05-19 23:00:52 +0300 Stefan Kost * common: Automatic update of common submodule From 9e5bbd5 to 69b981f 2011-05-18 16:14:10 +0300 Stefan Kost * common: Automatic update of common submodule From fd35073 to 9e5bbd5 2011-05-18 12:27:35 +0300 Stefan Kost * common: Automatic update of common submodule From 46dfcea to fd35073 2011-05-17 09:48:13 +0200 Wim Taymans * 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 Merge branch 'master' into 0.11 2011-05-03 21:13:15 +0200 Fabian Deutsch * bindings/vala/gst-rtsp-server-0.10.vapi: Updated Vala bindings. Signed-off-by: Fabian Deutsch 2011-05-03 16:24:28 +0200 Fabian Deutsch * gst/rtsp-server/rtsp-server.c: * gst/rtsp-server/rtsp-server.h: Add a signal for newly connected clients. Signed-off-by: Fabian Deutsch 2011-05-08 13:15:19 +0200 Alessandro Decina * bindings/python/rtspserver.override: python: override gst_rtsp_media_mapping_add_factory to fix refcounting 2011-04-26 19:22:50 +0200 Wim Taymans * 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 * common: add common 2011-04-26 19:07:13 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: common configure.ac 2011-04-24 14:07:11 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From c3cafe1 to 46dfcea 2011-04-20 11:19:38 +0200 Alessandro Decina * bindings/python/Makefile.am: * bindings/python/rtspserver.defs: python bindings: wrap GstRTSPMediaFactoryClass vfuncs 2011-04-20 11:13:56 +0200 Alessandro Decina * bindings/python/arg-types.py: python bindings: add GstRTSPUrlParam Needed to implement MediaFactory virtual proxies 2011-04-20 10:19:46 +0200 Alessandro Decina * bindings/python/arg-types.py: python bindings: fix returning GstRTSPUrl types 2011-04-20 10:17:07 +0200 Alessandro Decina * bindings/python/arg-types.py: python bindings: add arg type for GstRTSPUrl 2011-04-20 10:16:08 +0200 Alessandro Decina * bindings/python/rtspserver.defs: python bindings: fix the definition of MediaFactory.collect_stream 2011-04-04 15:59:50 +0300 Stefan Kost * common: Automatic update of common submodule From 1ccbe09 to c3cafe1 2011-03-25 22:38:06 +0100 Sebastian Dröge * common: Automatic update of common submodule From 193b717 to 1ccbe09 2011-03-25 14:58:34 +0200 Stefan Kost * common: Automatic update of common submodule From b77e2bf to 193b717 2011-03-25 10:04:57 +0100 Sebastian Dröge * Makefile.am: build: Include lcov.mak to allow test coverage report generation 2011-03-25 09:35:15 +0100 Sebastian Dröge * common: Automatic update of common submodule From d8814b6 to b77e2bf 2011-03-25 09:11:40 +0100 Sebastian Dröge * common: Automatic update of common submodule From 6aaa286 to d8814b6 2011-03-24 18:51:37 +0200 Stefan Kost * common: Automatic update of common submodule From 6aec6b9 to 6aaa286 2011-03-18 19:34:57 +0100 Luis de Bethencourt * autogen.sh: autogen: wingo signed comment 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya * 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 * gst/rtsp-server/Makefile.am: rtsp-server: Don't install the funnel header 2011-02-28 18:35:03 +0100 Mark Nauwelaerts * common: Automatic update of common submodule From 1de7f6a to 6aec6b9 2011-02-26 19:58:02 +0000 Tim-Philipp Müller * 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 * common: Automatic update of common submodule From f94d739 to 1de7f6a 2011-02-02 15:37:03 +0100 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: remove more unused code 2011-02-02 15:30:45 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * docs/Makefile.am: docs: recursive into sub-directories on 'make upload' 2011-01-24 11:53:17 +0000 Tim-Philipp Müller * 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 * configure.ac: back to development === release 0.10.8 === 2011-01-24 11:57:12 +0100 Wim Taymans * configure.ac: release 0.10.8 2011-01-19 15:29:55 +0100 Wim Taymans * gst/rtsp-server/rtsp-server.c: rtsp-server: clarify docs a little 2011-01-13 18:57:15 +0100 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: init debug category before starting thread 2011-01-13 18:40:48 +0100 Wim Taymans * gst/rtsp-server/rtsp-auth.c: auth: add realm to make it more spec compliant 2011-01-12 18:57:41 +0100 Wim Taymans * gst/rtsp-server/rtsp-server.c: * gst/rtsp-server/rtsp-server.h: server: add locking 2011-01-12 18:33:49 +0100 Wim Taymans * examples/test-video.c: example: improve example docs a little 2011-01-12 18:26:57 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-server.c: server: simpify channel function 2011-01-12 18:18:13 +0100 Wim Taymans * 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 * Makefile.am: * configure.ac: build tests 2011-01-12 18:16:46 +0100 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-server.c: server: chain up to the parent finalize 2010-09-21 17:04:02 -0300 André Dieb Martins * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: unref auth in finalize 2011-01-12 12:07:04 +0100 Wim Taymans * gst/rtsp-server/rtsp-server.c: server: unref auth in finalize 2011-01-12 11:07:26 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.h: rtsp-media.h: Minor corrections in comments. Fixes #638944 2011-01-11 15:52:44 +0200 Stefan Kost * common: Automatic update of common submodule From e572c87 to f94d739 2011-01-11 13:01:44 +0100 Edward Hervey * .gitignore: * docs/.gitignore: * docs/libs/.gitignore: * examples/.gitignore: * gst/rtsp-server/.gitignore: gitignore: updates 2011-01-11 12:58:39 +0100 Edward Hervey * 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 * common: Automatic update of common submodule From ccbaa85 to e572c87 2011-01-10 14:56:39 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 46445ad to ccbaa85 2011-01-10 15:10:53 +0100 Wim Taymans * 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 * 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 * 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 * .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 * 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 * examples/test-uri.c: examples: add some more options and comments 2011-01-07 11:24:39 +0100 Wim Taymans * gst/rtsp-server/rtsp-media-factory-uri.c: factory-uri: use right property type 2011-01-05 12:07:42 +0100 Wim Taymans * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: * gst/rtsp-server/rtsp-media.h: docs: improve docs 2010-12-29 16:26:41 +0100 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.h: client: cleanup headers 2010-12-28 12:18:41 +0100 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: fix typo 2010-12-23 18:53:01 +0100 Wim Taymans * 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 * 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 * 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 * common: Automatic update of common submodule From 169462a to 46445ad 2010-12-18 11:24:48 +0100 Wim Taymans * 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 * 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 * common: Automatic update of common submodule From 011bcc8 to 169462a 2010-12-13 16:38:09 +0100 Andy Wingo 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 Merge branch 'master' into 0.11 Conflicts: common configure.ac 2010-12-12 15:48:47 +0100 Wim Taymans * 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 * 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 * examples/Makefile.am: * examples/test-uri.c: example: add example of the uri factory 2010-12-11 18:01:53 +0100 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-client.c: rtsp-server: Remove unused variable and dead assignment 2010-12-11 10:49:30 +0100 Edward Hervey * 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 * 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 * 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 * Makefile.am: Makefile: Add cruft-cleaning support 2010-12-07 18:52:15 +0100 Edward Hervey * 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 * gst/rtsp-server/Makefile.am: Makefile.am: Use standard GIR make behaviour 2010-12-07 18:14:22 +0100 Edward Hervey * autogen.sh: * configure.ac: autogen/configure: Bring more in sync to standard gst module behaviour 2010-12-06 19:29:53 +0100 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: warn and fail when gstrtpbin is not found 2010-12-06 12:40:30 +0100 Wim Taymans * configure.ac: configure: open 0.11 branch 2010-12-01 20:00:22 +0100 Edward Hervey * .gitmodules: * common: Add common submodule 2010-12-01 19:58:49 +0100 Edward Hervey * 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 * common/m4/introspection.m4: Update introspection.m4 to match usage 2010-10-30 13:26:12 +0200 Wim Taymans * README: README: update Remove old stuff from the README 2010-10-11 11:12:11 +0200 Wim Taymans * configure.ac: back to development === release 0.10.7 === 2010-10-11 11:05:40 +0200 Wim Taymans * configure.ac: release 0.10.7 2010-10-04 17:16:40 +0200 Wim Taymans * 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 * 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 * 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 * 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 * bindings/vala/Makefile.am: build: fix distcheck 2010-09-22 18:24:12 +0200 Sebastian Dröge * 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 * 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 * configure.ac: back to development === release 0.10.6 === 2010-09-22 16:22:49 +0200 Wim Taymans * configure.ac: configure: release 0.10.6 2010-09-22 16:15:56 +0200 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: help the compiler a little 2010-08-24 16:47:30 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * docs/README: README: add blurb about shared media factories 2010-08-09 12:56:23 -0700 David Schleef * gst/rtsp-server/rtsp-media.c: Add stdlib.h for atoi() 2010-05-20 14:33:24 +0100 Tim-Philipp Müller * 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 * 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 * configure.ac: configure: fail if GStreamer core/base requirements are not met 2010-04-06 17:08:40 +0200 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-session.c: session: small cleanups 2010-04-06 11:13:51 +0200 Wim Taymans * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-session.c: session: indent 2010-03-16 18:33:23 +0100 Wim Taymans * gst/rtsp-server/rtsp-client.c: client: use right size for malloc 2010-03-10 11:45:30 +0100 Wim Taymans * gst/rtsp-server/rtsp-server.c: server: comment ipv6 server listening address 2010-03-10 11:45:06 +0100 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: allow for ipv6 sockets 2010-03-09 13:49:00 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-media.h: media: update comments a little 2010-03-09 13:43:29 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-client.c: client: guard against invalid paths 2010-03-09 13:41:33 +0100 Wim Taymans * examples/test-video.c: test: catch server bind errors 2010-03-09 10:27:38 +0100 Alessandro Decina * 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 * gst/rtsp-server/rtsp-media.c: media: collect media position when seek completes 2010-03-05 18:37:17 +0100 Luca Ognibene * gst/rtsp-server/rtsp-client.c: client: call unlink_streams in client finalize Fixes #599027 2010-03-05 18:23:18 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-sdp.c: sdp: reindent and check for prepared status 2010-03-05 17:51:26 +0100 Wim Taymans * 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 * gst/rtsp-server/rtsp-media.c: media: reindent 2010-03-05 13:34:15 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-sdp.c: sdp: make server work better when behind a proxy 2009-11-21 01:17:25 +0100 Sebastian Pölsterl * 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 * 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 * 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 * configure.ac: back to development === release 0.10.5 === 2009-11-05 11:20:45 +0100 Wim Taymans * configure.ac: release 0.10.5 2009-10-14 12:11:31 +0200 Wim Taymans * configure.ac: configure: bump required versions 2009-10-11 13:57:54 +0200 Luca Ognibene * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: media: add some docs 2009-08-24 13:27:00 +0200 Peter Kjellerstedt * 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 * configure.ac: back to development === release 0.10.4 === 2009-08-05 11:44:49 +0200 Wim Taymans * configure.ac: Release 0.10.4 2009-07-27 19:42:44 +0200 Wim Taymans * 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 * 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 * 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 * 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 * configure.ac: configure: bump core/base requirements to release 2009-06-18 16:05:18 +0200 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: fix indentation 2009-06-14 23:12:13 +0200 Sebastian Pölsterl * 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 * gst/rtsp-server/rtsp-media.c: set state and remove elements of media in for loop 2009-06-13 14:38:39 +0200 Sebastian * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: media: don't leak session pads 2009-06-04 18:32:15 +0200 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: clean up the messages a bit 2009-06-03 12:13:21 +0200 Wim Taymans * gst/rtsp-server/rtsp-sdp.c: sdp: warn and skip streams without media 2009-05-30 14:38:34 +0200 Sebastian Pölsterl * 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 * 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 * gst/rtsp-server/rtsp-media.c: media: be less verbose and leak less 2009-05-26 19:05:07 +0200 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: don't leak the destination address 2009-05-26 19:01:10 +0200 Wim Taymans * 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 * 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 * 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 * 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 * examples/.gitignore: * examples/Makefile.am: * examples/test-sdp.c: example: add SDP relay example 2009-05-24 19:56:45 +0200 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: also count active TCP connections 2009-05-24 19:34:52 +0200 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-client.c: client: warn when we can't do RTP-Info 2009-05-23 16:18:04 +0200 Wim Taymans * gst/rtsp-server/rtsp-media-factory.c: factory: factor out the stream construction 2009-05-23 16:17:02 +0200 Wim Taymans * 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 * configure.ac: back to development === release 0.10.3 === 2009-05-17 13:59:10 +0200 Wim Taymans * 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 * 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 * 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 * configure.ac: back to development === release 0.10.2 === 2009-05-11 10:50:31 +0200 Wim Taymans * configure.ac: release 0.10.2 2009-05-11 10:38:44 +0200 Wim Taymans * .gitignore: * common/m4/.gitignore: * examples/.gitignore: * pkgconfig/.gitignore: add some .gitignore files 2009-04-29 17:24:46 +0200 Wim Taymans * gst/rtsp-server/rtsp-media.c: media: seek to key frames 2009-04-21 22:44:05 +0200 Wim Taymans * 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 * 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 * 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 * 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 * 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 * examples/test-ogg.c: tests: set the payload type correctly 2009-04-03 22:46:22 +0200 Wim Taymans * 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 * 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 * 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 * examples/test-readme.c: examples: mark the example as shared for testing 2009-04-03 19:44:37 +0200 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-session.c: docs: fix typo in API docs 2009-03-13 15:57:42 +0100 Wim Taymans * 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 * 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 * gst/rtsp-server/rtsp-client.c: allow pause requests for now. -- 2009-03-11 20:03:06 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: Free the pipeline before other things --- 2009-03-04 16:33:21 +0100 Wim Taymans * gst/rtsp-server/rtsp-client.c: Only free the pending tunnel if there is one -- 2009-03-04 12:44:01 +0100 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-media.c: Add better debug info Add some better debug info. 2009-02-13 20:00:34 +0100 Wim Taymans * examples/test-video.c: Time out sessions Add support for session timeouts in the example. 2009-02-13 19:58:17 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server 2009-02-04 20:10:39 +0100 Wim Taymans * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * configure.ac: Back to development Back to 0.10.1.1 === release 0.10.1 === 2009-01-30 11:20:18 +0100 Wim Taymans * configure.ac: Make 0.10.1 release Release 0.10.1 2009-01-29 15:19:01 +0100 Wim Taymans * bindings/vala/Makefile.am: Fix make dist Add more directories and files to the dist. 2009-01-24 14:34:35 +0100 Sebastian Pölsterl * bindings/python/Makefile.am: * bindings/python/rtspserver.override: Fixed compile error of python bindings 2009-01-23 21:03:53 +0100 Sebastian Pölsterl * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * gst/rtsp-server/Makefile.am: Add Makefile entry for the media factory 2009-01-20 19:44:45 +0100 Wim Taymans * 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 * 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 * 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 * gst/rtsp-server/rtsp-session.h: Add some comments. 2009-01-20 13:57:47 +0100 Wim Taymans * 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 * 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 * 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 * 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 * gst/rtsp-server/rtsp-session-pool.c: Do some more cleanup of the session pool. 2009-01-08 16:28:24 +0100 Wim Taymans * 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 * gst/rtsp-server/Makefile.am: Install rtsp-session and rtsp-session-pool headers 2009-01-08 14:57:55 +0100 Wim Taymans * .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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * examples/Makefile.am: * gst/rtsp-server/Makefile.am: Put GStreamer version in library name 2009-01-08 13:51:26 +0100 Wim Taymans * examples/Makefile.am: * gst/rtsp-server/Makefile.am: Fix some issues to pass distcheck 2009-01-08 13:41:33 +0100 Wim Taymans * gst/rtsp-server/rtsp-server.c: Added port property to GstRTSPServer class. 2009-01-08 13:18:55 +0100 Wim Taymans * 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 * src/rtsp-client.h: Removed obsolete variable 2008-11-10 21:03:15 +0100 Sebastian Pölsterl * 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 * 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 * src/rtsp-session.c: Initialize some more vars. 2008-10-23 12:14:55 +0200 Wim Taymans * src/rtsp-session.c: Initialize variable to avoid compiler warning. 2008-10-09 13:30:47 +0100 Simon McVittie * .gitignore: Add a reasonable generic .gitignore