diff options
| -rw-r--r-- | ChangeLog | 1161 | ||||
| -rw-r--r-- | NEWS | 1302 | ||||
| -rw-r--r-- | RELEASE | 15 | ||||
| -rw-r--r-- | docs/gst_plugins_cache.json | 4 | ||||
| -rw-r--r-- | gst-rtsp-server.doap | 12 | ||||
| -rw-r--r-- | meson.build | 2 |
6 files changed, 1256 insertions, 1240 deletions
@@ -1,3 +1,1164 @@ +=== release 1.17.1 === + +2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com> + + * ChangeLog: + * NEWS: + * RELEASE: + * gst-rtsp-server.doap: + * meson.build: + Release 1.17.1 + +2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Add/configure transports when completing the pipeline + Otherwise the transports are not set up yet during the PLAY request + handling when unsuspending (and thus unblocking) the media. + In case of live pipelines this then causes the first few packets to go + to the sinks before they know what to do with them, and they simply + discard them which is rather suboptimal in case of keyframes. + For non-live pipelines this is not a problem because the sink will still + be PAUSED and as such not send out the data yet but wait until it goes + to PLAYING, which is late enough. + Adding the transports multiple times is not a problem: if the transport + is already added it won't be added another time and TRUE will be + returned. + This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0 + before 1.14.0. + Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135> + +2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Fix misleading comment + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135> + +2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering + The pad probes are not needed anymore at this point and later when + reaching buffering 100% only the state is changed, no unblocking + happens. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135> + +2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Remove duplicated media_unblock() function + It does literally the same as media_streams_set_blocked(FALSE). + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135> + +2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com> + + * examples/test-onvif-server.c: + test-onvif-server: cast ntp-offset property value to 64 bit + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134> + +2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com> + + * docs/gst_plugins_cache.json: + docs: Update plugins cache + +2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * examples/test-onvif-server.c: + * examples/test-onvif-server.h: + * gst/rtsp-server/rtsp-onvif-media-factory.h: + onvif-media-factory: define autoptr cleanup function + And have the factory in the onvif-server example inherit from + GstRTSPOnvifMediaFactory. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133> + +2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com> + + * docs/gst_plugins_cache.json: + docs: Update plugins cache + +2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com> + + * tests/check/gst/rtspserver.c: + tests: enforce I420 format + Test was not enforcing a video format on videotestsrc. I420 was picked as it + was the first format in GST_VIDEO_FORMATS_ALL which will no longer be + true (gst-plugins-base!689). + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129> + +2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + plugins: uddate gst_type_mark_as_plugin_api() calls + +2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com> + + * docs/meson.build: + doc: Require hotdoc >= 0.11.0 + +2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com> + + * docs/gst_plugins_cache.json: + docs: Update gst_plugins_cache.json + +2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types + +2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/meson.build: + meson: gir: remove bogus sources_top_dir kwarg + Doesn't actually exist. Was fixed differently in Meson + so that the user doesn't have to specify it. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127> + +2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/meson.build: + tests: put registry into tests/check not the gst/ subdir + Underscorify the test name before setting GST_REGISTRY, + so the registry actually ends up in the current build dir + and not some subdir. + For consistency with the other modules, but should also + avoid problems on windows. + Also fix indentation of environment block. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126> + +2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/meson.build: + tests: fix meson test env setup to make sure we use the right gst-plugin-scanner + If core is built as a subproject (e.g. as in gst-build), make sure to use + the gst-plugin-scanner from the built subproject. Without this, gstreamer + might accidentally use the gst-plugin-scanner from the install prefix if + that exists, which in turn might drag in gst library versions we didn't + mean to drag in. Those gst library versions might then be older than + what our current build needs, and might cause our newly-built plugins + to get blacklisted in the test registry because they rely on a symbol + that the wrongly-pulled in gst lib doesn't have. + This should fix running of unit tests in gst-build when invoking + meson test or ninja test from outside the devenv for the case where + there is an older or different-version gst-plugin-scanner installed + in the install prefix. + In case no gst-plugin-scanner is installed in the install prefix, this + will fix "GStreamer-WARNING: External plugin loader failed. This most + likely means that the plugin loader helper binary was not found or + could not be run. You might need to set the GST_PLUGIN_SCANNER + environment variable if your setup is unusual." warnings when running + the unit tests. + In the case where we find GStreamer core via pkg-config we use + a newly-added pkg-config var "pluginscannerdir" to get the right + directory. This has the benefit of working transparently for both + installed and uninstalled pkg-config files/setups. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126> + +2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/meson.build: + tests: gst-plugins-base and -bad plugins are required for the unit tests + Make hard requirement until we have more fine-grained control + in the unit tests. Of course the presence of the .pc file doesn't + imply that the plugins we need are actually there, but it's at + least a step in the right direction. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126> + +2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/meson.build: + tests: pick up rtsp-server plugins from build directory only + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126> + +2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: wait for all GstRTSPStreamBlocking messages + Make sure rtsp-media have received a GstRTSPStreamBlocking message from + each active stream when checking if all streams are blocked. + Without this change there will be a race condition when using two or + more streams and rtsp-media receives a GstRTSPStreamBlocking message + from one of the streams. This is because rtsp-media then checks if all + streams are blocked by calling gst_rtsp_stream_is_blocking() for each + stream. This function call returns TRUE if the stream has sent a + GstRTSPStreamBlocking message, however, rtsp-media may have yet to + receive this message. This would then result in that rtsp-media + erroneously thinks it is blocking all streams which could result in + rtsp-media changing state, from PREPARING to PREPARED. In the case of a + preroll, this could result in that rtsp-media thinks that the pipeline + is prerolled even though that might not be the case. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124> + +2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: update expected_async_done during suspend + Set expected_async_done to FALSE in default_suspend() if a state change + occurs and the return value from set_target_state() is something other + than GST_STATE_CHANGE_ASYNC. + Without this change there is a risk that expected_async_done will be + TRUE even though no asynchronous state change is taking place. This + could happen if the pipeline is set to PAUSED using + media_set_pipeline_state_locked(), an asynchronous state change starts + and then the media is suspended (which could result in a state change, + aborting the asynchronous state change). If the media is suspended + before the asynchronous state change ends then expected_async_done will + be TRUE but no asynchronous state change is taking place. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123> + +2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client + There was a race condition where client was being finalized and + concurrently in some other thread the rtsp ctrl timout was relying on + client data that was being freed. + When rtsp ctrl timeout is setup, a WeakRef on Client is set. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121> + +2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com> + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media-factory: complete DSCP QoS setting support + add dscp_qos setting support at factory and media level to setup IP DSCP + field of bounded UDP sinks. + Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120> + +2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Fix some race conditions around timeout source removal + We always need to take the lock while accessing it as otherwise another + thread might've removed it in the meantime. Also when destroying and + creating a new one, ensure that the mutex is not shortly unlocked in + between as during that time another one might potentially be created + already. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119> + +2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates() + And the same for gst_rtsp_stream_get_rates(). + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118> + +2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com> + + * examples/test-onvif-server.c: + examples: test-onvif-server: fix compiler warnings on raspbian + Fix printf format for 64-bit variables. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117> + +2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks + The old API is preserved now and new API was added that provides the + additional parameter to the callback. + Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116> + +2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Store the timeout source by pointer instead of id + That way we don't have to retrieve it again from the main context when + destroying it but can directly do so. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115> + +2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Clean up watch/watch context and related state consistently + And assert that it was cleaned up properly before the client is + finalized. If something is still around when the client is shut down + then something went very wrong before. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115> + +2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * tests/check/gst/rtspserver.c: + rtsp-client: Combine the pre-session and post-session timeout + They previously used the same state but different mechanisms and + functions, which was difficult to follow, error prone and simply + confusing. + Also adjust the test for the post-session timeout a bit to be less racy + now that the timing has slightly changed. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115> + +2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Don't ever close the client connection directly when a session is torn down + There might be other sessions that are running over the same RTSP + connection and we should not simply close the client directly if one of + them is torn down. + By default the connection will be closed once the client closes it or + the OS does. This behaviour can be adjusted with the + post-session-timeout property, which allows to close it automatically + from the server side after all sessions are gone and the given timeout + is reached. + This reverts the previous commit. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115> + +2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: If the TEARDOWN response can be sent directly, directly close the client + Instead of closing it never at all. Previously there was only code that + closed the client asynchronously if sending the response happened + asynchrously at a later time. + Thanks to Christian M for debugging this issue. + Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114> + +2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo + Otherwise no sink is found for multicast sreams and the less accurate + fallback is used to determine the current sequence number and timestamp. + +2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-auth.c: + rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header + When using the basic authentication scheme, we wouldn't validate that + the authorization field of the credentials is not NULL and pass it on + to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will + dereference the NULL pointer and crash. + A specially crafted (read: invalid) RTSP header can cause this to + happen. + As a solution, check for the authorization to be not NULL before + continuing processing it and if it is simply fail authentication. + This fixes CVE-2020-6095 and TALOS-2020-1018. + Discovered by Peter Wang of Cisco ASIG. + +2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Use watch_context before unref + Move the usage of priv->watch_context to beginning of function + gst_rtsp_client_finalize. Instead of use it after + g_main_context_unref (priv->watch_context). + +2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: fix deadlock on transport removal + We cannot take the RTSPStream lock while holding a transport backlog + lock, as remove_transport may be called externally, which will + take first the RTSPStream lock then the transport backlog lock. + +2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-server-internal.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: clear backlog when removing transport + This ensures we don't end up calling any of transports' callbacks + with a potentially unreffed user_data (in practice, a client that + may have been removed) + +2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: marshal calls to send_tcp_message to a single thread + In order to address the race condition pointed out at + https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579 + we get rid of the send thread pool, and instead spawn and manage + a single thread to pull samples from app sinks and add them to + the transport's backlogs. + Additionally, we now also always go through the backlogs in order + to simplify the logic. + +2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-server-internal.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: properly protect TCP backlog access + Fixes #97 + We cannot hold stream->lock while pushing data, but need + to consistently check the state of the backlog both from + the send_tcp_message function and the on_message_sent function, + which may or may not be called from the same thread. + This commit introduces internal API to allow for potentially + recursive locking of transport streams, addressing a race + condition where the RTSP stream could push items out of order + when popping them from the backlog. + +2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline() + It's taken ownership of by the media, and returned with `transfer none` + from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it + first then any bindings will wrongly take ownership of the pipeline once + it arrives in bindings code. + +2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com> + + * examples/test-onvif-client.c: + Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants + +2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: fix default latency + +2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: make closing more thread safe + + Take the watch lock prior to using priv->watch + + Flush both the watch and connection before closing / unreffing + gst_rtsp_connection_close() is not threadsafe on its own, this is + a workaround at the client level, where we control both the watch + and the connection + +2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com> + + * gst/rtsp-server/rtsp-latency-bin.c: + rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated + from glib + ``` + Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated + `your_type_get_instance_private()` function instead + ``` + +2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * tests/check/gst/rtspserver.c: + rtsp-client: add property post-session-timeout + This is a TCP connection timeout for client connections, in seconds. + If a positive value is set for this property, the client connection + will be kept alive for this amount of seconds after the last session + timeout. For negative values of this property the connection timeout + handling is delegated to the system (just as it was before). + Fixes #83 + +2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: check for NULL transports prior to ref'ing + +2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-server-internal.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: fix checking of TCP backpressure + The internal index of our appsinks, while it can be used to + determine whether a message is RTP or RTCP, is not necessarily + the same as the interleaved channel. Let the stream-transport + determine the channel to check backpressure for, the same way + it determines the channel according to whether it is sending + RTP or RTCP. + +2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-session.c: + rtsp-session: Butcher the file to please gst-indent in the CI + This should be reverted once the CI has an updated gst-indent. + +2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + * gst/rtsp-sink/gstrtspclientsink.c: + * gst/rtsp-sink/gstrtspclientsink.h: + rtsp-session & client: Remove deprecated GTimeVal + GTimeVal won't work past 2038 + +2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com> + + * gst/rtsp-server/rtsp-auth.c: + rtsp-auth: fix default token leak + +2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + gstrtspclientsink: unref transports when closing bin + Fixes #91 + +2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Force seek when flush flag is set + The commit "rtsp-client: define all seek accuracy flags from + setup_play_mode" changed the behaviour of when doing a seek. + Before that commit, having the flush flag set would result in a seek + (forced seek). + Even if no seek was needed. One reason to force seek is to flush old buffers + created in Describe requests. + Thus adding force seek also for flush flag will result in play request + with fresh buffers. + +2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Revitalize dead code + Leftover from 65d9aa327cd1844934836249cd4463edf09c725d + CID: 1455379 + +2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com> + + * gst/rtsp-server/rtsp-sdp.c: + rtsp-sdp: Don't try to use non-initialized values + Only attempt to use the various timing values iif gst_rtsp_stream_get_info() + returns TRUE. Also avoid the whole clock signalling block if we're not + dealing with senders. + CID: 1439524 + CID: 1439536 + CID: 1439520 + +2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com> + + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/stream.c: + rtsp-stream: Removing invalid transports returns false + When removing transports an assertion was that the transports passed in + for removal are present in the list, however that can't be assumed. + As an example if a transport was removed from a thread running + send_tcp_message, the main thread can try to remove the same transport + again if it gets a handle_pause_request. This will not effect the + transport list but it will effect n_tcp_transports as it will be + decrement and then have the wrong value. + +2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com> + + * tests/check/gst/client.c: + client test: add scale and speed negative tests + Negative tests for scale and speed should be done as well, verify that + the response code is "400 Bad request" when a bad request is done. + +2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-sink/gstrtspclientsink.c: + Don't pass default GLib marshallers for signals + By passing NULL to `g_signal_new` instead of a marshaller, GLib will + actually internally optimize the signal (if the marshaller is available + in GLib itself) by also setting the valist marshaller. This makes the + signal emission a bit more performant than the regular marshalling, + which still needs to box into `GValue` and call libffi in case of a + generic marshaller. + Note that for custom marshallers, one would use + `g_signal_set_va_marshaller()` with the valist marshaller instead. + +2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com> + + * gst/rtsp-server/rtsp-mount-points.c: + GstRTSPMountPoints: Remove any existing factory before adding a new one + The documentation of gst_rtsp_mount_points_add_factory() says "Any + previous mount point will be freed" which was true when it was + implemented using a GHashTable. But in 2012 it got rewrote using a + GSequence and since then it could have 2 factories for the same path. + Which one gets used is random, depending on the sorting order of 2 + identical items. + +2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-server-internal.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + * gst/rtsp-server/rtsp-stream.c: + stream: refactor TCP backpressure handling + The previous implementation stopped sending TCP messages to + all clients when a single one stopped consuming them, which + obviously created problems for shared media. + Instead, we now manage a backlog in stream-transport, and slow + clients are removed once this backlog exceeds a maximum duration, + currently hardcoded. + Fixes #80 + +2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com> + + * meson.build: + meson: build gir even when cross-compiling if introspection was enabled explicitly + This can be made to work in certain circumstances when + cross-compiling, so default to not building g-i stuff + when cross-compiling, but allow it if introspection was + enabled explicitly via -Dintrospection=enabled. + See gstreamer/gstreamer#454 and gstreamer/gstreamer#381. + +2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-session.c: + rtsp-session: clean up comment extra-timeout + +2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses + Instead of hardcoding the URI, take the actual URI (and especially the correct port) + from the RTSP context. + Fixes #84 + +2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + rtsp-client: Lock shared media + For shared media we got race conditions. Concurrently rtsp clients might + suspend or unsuspend the shared media and thus change the state without + the clients expecting that. + By introducing a lock that can be taken by callers such as rtsp_client + one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media, + to handle the media sequentially thus allowing one client to finish its + rtsp call before another client calls on the same media. + https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86 + Fixes #86 + +2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-session.c: + rtsp-session: add property extra-timeout + Extra time to add to the timeout, in seconds. This only + affects the time until a session is considered timed out + and is not signalled in the RTSP request responses. + Only the value of the timeout property is signalled in the + request responses. + +2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream : fix race condition in send_tcp_message + If one thread is inside the send_tcp_message function and are done + sending rtp or rtcp messages so the n_outstanding variable is zero + however have not exit the loop sending the messages. While sending its + messages, transports have been added or removed to the transport list, + so the cache should be updated. If now an additional thread comes to + the function send_tcp_message and trying to send rtp messages it will + first destroy the rtp cache that is still being iterated trough by the + first thread. + Fixes #81 + +2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com> + + * .gitignore: + * .gitmodules: + * Makefile.am: + * autogen.sh: + * common: + * configure.ac: + * docs/.gitignore: + * examples/.gitignore: + * examples/Makefile.am: + * gst/Makefile.am: + * gst/rtsp-server/.gitignore: + * gst/rtsp-server/Makefile.am: + * gst/rtsp-sink/Makefile.am: + * pkgconfig/.gitignore: + * pkgconfig/Makefile.am: + * tests/.gitignore: + * tests/Makefile.am: + * tests/check/Makefile.am: + Remove autotools build + Replaced by Meson. + Maybe we can now use the meson pkgconfig module + for .pc files? (Does it support uninstalled now?) + +2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com> + + * tests/check/gst/client.c: + client: fix test mem leak in attach_rate_tweaking_probe + +2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com> + + * tests/check/gst/media.c: + media: remove memleak in test test_media_seek + +2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com> + + * tests/check/gst/rtspserver.c: + rtspserver: Remove memleak in test test_double_play + +2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Use lock in gst_rtsp_media_is_receive_only + +2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-media.c: + * tests/check/gst/rtspserver.c: + rtsp-media: Unblock all streams + When unsuspending and going to PLAYING, unblock all streams instead of + only those that are linked (the linked streams are the ones for which + SETUP has been called). GST_FLOW_NOT_LINKED will be returned when + pushing buffers on unlinked streams. + This change is because playback using single-threaded demuxers like + matroska-demux could be blocked if SETUP was not called for all media. + Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux, + gstflvdemux, qtdemux, and matroska-demux) will handle + GST_FLOW_NOT_LINKED automatically. + Fixes #39 + +2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-media.c: + * tests/check/gst/rtspserver.c: + rtsp-media: Wait on async when needed. + Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode. + In the unit test the pause from adjust_play_mode will cause a preroll + and after that async-done will be produced. + Without this patch there are no one consuming this async-done and when + later when seek fluch is done in gst_rtsp_media_seek_trickmode then it + wait for async-done. But then it wrongly find the async-done prodused by + adjus_play_mode and continue executing without waiting for the preroll + to finish. + +2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: RTP Info when completed_sender + Change condition that should be fulfilled regarding RTPInfo. + Replace !gst_rtsp_media_is_receive_only with + gst_rtsp_media_has_completed_sender. It is more correct to actually look + for a sender pipeline that is complete. Only then a RTPInfo should + exist. + gst_rtsp_media_is_receive_only gives different answears depending on + state of server. + If Describe is called wth URL+options for backchannel SDP will give only + audio and only backchannel a=sendonly + If Describe is called on URL+options that gives both audio and video + direction from server to client, pipelines are created. Thus + receive_only will return false, even though Setup only would setup + backchannel. + RTP-Info is only for outgoing streams. Thus one should look if outgoing + streams are complete. + +2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * tests/check/gst/client.c: + rtsp-client: RTP Info exists conditionally in PLAY + If RTP Info is missing and it is not a receiver only, eg. audio + backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR. + In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional. + Since 1.14 there is audio backchannel support. Thus RTP-info is + conditional now. When audio backchannel only mode, there is no RTP-info. + Fixes #82 + +2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * examples/test-onvif-client.c: + test-onvif-client: remove unused query + +2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: RTP Info must exist in PLAY response + If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR + Fixes #76 + +2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * examples/test-onvif-client.c: + test-onvif-client: perform accurate seeks + See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336 + Also, modify how we compute the position: position queries in + PAUSED mode fail to account for the newly-prerolled frame, leading + to frame skips when performing seeks in that state. Instead, + compute the current position from the last sample. + +2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * tests/check/gst/rtspserver.c: + Use complete streams for scale and speed. + Without this patch it's always stream0 that is used to get segment event + that is used to set scale and speed. This even if client not doing SETUP + for stream0. At least in suspend mode reset this not working since then + it's just random if send_rtp_sink have got any segment event. There are + no check if send_rtp_sink for stream0 got any data before media is + prerolled after PLAY request. + +2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com> + + * examples/test-onvif-server.c: + * examples/test-onvif-server.h: + examples/onvif-server: fix werror build with clang + ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:346:65: warning: implicit conversion from enumeration type 'const GstSegmentFlags' to different enumeration type 'GstSeekFlags' [-Wenum-conversion] + self->incoming_segment->format, self->incoming_segment->flags, + ~~~~~~~~~~~~~~~~~~~~~~~~^~~~~ + ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function] + G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin); + ^ + /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE' + static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \ + ^ + <scratch space>:77:1: note: expanded from here + REPLAY_IS_BIN + ^ + ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function] + G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY, + ^ + /usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE' + static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \ + ^ + <scratch space>:9:1: note: expanded from here + ONVIF_FACTORY + ^ + ../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function] + /usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE' + static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \ + ^ + <scratch space>:12:1: note: expanded from here + ONVIF_IS_FACTORY + ^ + +2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com> + + * docs/meson.build: + meson: Don't generate doc cache when no plugins are enabled + Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled + +2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com> + + * examples/test-onvif-client.c: + test-onvif-client: stdin is not defined in MSVC + +2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: add missing Since tag + +2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * examples/test-onvif-client.c: + test-onvif-client: STDIN_FILENO is not portable + If not defined, define it to _fileno(stdin) on Windows, 0 + everywhere else + +2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * examples/test-onvif-server.c: + test-onvif-server: downgrade logging + +2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * examples/meson.build: + * examples/test-onvif-client.c: + * examples/test-onvif-server.c: + examples: add ONVIF client / server example + +2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + rtsp-client: define all seek accuracy flags from setup_play_mode + We then pass those to adjust_play_mode, which needs to operate + on the "final" seek flags, as previously the code in rtsp-media + was assuming that accuracy seek flags (accurate / key_unit) should + not be set if the flags passed to the seek method were already set. + +2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Try to get dynamic payloaders by name from their bin first + First try "pay", then "pay_%s" (where %s == pad name). And only then + fall back to the code that simply takes the first payloader that is + found. + The current code usually works (but is racy) because it will always take + the payloader that was last added (due to g_list_prepend() when adding + elements) in pad-added and that's usually the correct one. But if a new + payloader is added between pad-added and us trying to get it, we would + get the wrong payloader. + +2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * tests/check/gst/client.c: + client test: expect any port in transport + setup_multicast_client sets a 5000-5010 range for the client + ports, it is incorrect to expect the transport to always use + 5000-5001 + Fixes #73 + +2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * tests/check/gst/onvif.c: + onvif tests: use g_cond_wait() correctly + g_cond_wait() has to be called in a loop until required conditions + are met + Fixes #71 + +2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Not wait on receiver streams when pre-rolling + Without this patch there are problem pre-rolling when using audio back + channel. + Without this patch a probe will be created for all streams including + the stream for audio backchannel. To pre-roll all this pads have to + receive data. Since the stream for audio backchannel is a receiver this + will never happen. + The solution is to never create any probes for streams that are for + incomming data and instead set them as blocking already from beginning. + +2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/rtsp-onvif-media-factory.c: + * gst/rtsp-server/rtsp-onvif-media.c: + onvif-media: fix "void function returning a value" compiler warning + +2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: make sure streams are blocked when sending seek + The recent ONVIF work exposed a race condition when dealing with + multiple streams: one of the sinks may preroll before other streams + have started flushing. This led to the pipeline posting async-done + prematurely, when some streams were actually still in the middle + of performing a flushing seek. The newly-added code looks up a + sticky segment event on the first stream in order to respond to + the PLAY request with accurate Scale and Speed headers. In the + failure condition, the first stream was flushing, and thus had + no sticky segment event, leading to the PLAY request failing, + and in turn the test. + +2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de> + + * docs/README: + * gst/rtsp-server/rtsp-media-factory-uri.h: + Fix typos + +2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-onvif-client.c: + * gst/rtsp-server/rtsp-onvif-client.h: + * gst/rtsp-server/rtsp-onvif-media-factory.c: + * gst/rtsp-server/rtsp-onvif-media-factory.h: + * gst/rtsp-server/rtsp-onvif-media.c: + * gst/rtsp-server/rtsp-onvif-server.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/media.c: + * tests/check/gst/onvif.c: + * tests/check/meson.build: + onvif: Implement and test the Streaming Specification + https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf + +2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + rtsp-client: add gst_rtsp_client_get_stream_transport() + This will be used in the onvif tests in order to validate the + data transmitted over TCP: for streaming to continue after a + data message has been provided to client->send_func, the client + is responsible for marking the message as sent on the relevant + stream transport. + +2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + client: Scale implies TRICK_MODE + +2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + client: compare booleans, not pointers to them + +2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/media.c: + Reverse playback support + GStreamer plays segment from stop to start when doing reverse playback. + RTSP implies that media should be played from start of Range header to + its stop. Hence we swap start and stop times before passing them to + gst_element_seek. + Also make gst_rtsp_stream_query_stop always return value that can be + used as stop time of Range header. + +2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * tests/check/gst/client.c: + rtsp-client: add support for Scale and Speed header + Add support for the RTSP Scale and Speed headers by setting the rate in + the seek to (scale*speed). We then check the resulting segment for rate + and applied rate, and use them as values for the Speed and Scale headers + respectively. + https://bugzilla.gnome.org/show_bug.cgi?id=754575 + +2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + rtsp-client: allow sub classes to adjust the seek + Adds a new virtual function, adjust_play_mode(), that allows + sub classes to adjust the seek done on the media. The sub class can + modify the values of the the seek flags and the rate. + https://bugzilla.gnome.org/show_bug.cgi?id=754575 + +2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/media.c: + rtsp-media: allow specifying rate when seeking + Add new function gst_rtsp_media_seek_full_with_rate() which allows the + caller to specify the rate for the seek. Also added functions in + rtsp-stream and rtsp-media for retreiving current rate and applied rate. + https://bugzilla.gnome.org/show_bug.cgi?id=754575 + +2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com> + + * configure.ac: + * meson.build: + meson: Bump minimal GLib version to 2.44 + This means we can use some newer features and get rid of some + boilerplate code using the G_DECLARE_* macros. + As discussed on IRC, 2.44 is old enough by now to start depending on it. + +2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * docs/libs/.gitignore: + * docs/libs/Makefile.am: + * docs/libs/gst-rtsp-server-docs.sgml: + * docs/libs/gst-rtsp-server-sections.txt: + * docs/libs/gst-rtsp-server.types: + docs: remove obsolete gtk-doc related files + +2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + doc: remove xml from comments + +2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com> + + * docs/gst_plugins_cache.json: + * docs/meson.build: + docs: Stop building the doc cache by default + And update the cache + Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36 + +2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com> + + * docs/gst_plugins_cache.json: + docs: Update plugins documentation cache + +2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com> + + * docs/meson.build: + * gst/rtsp-server/rtsp-context.c: + * gst/rtsp-server/rtsp-session-pool.c: + doc: Fix some docstrings + +2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com> + + * .gitignore: + * Makefile.am: + * configure.ac: + * docs/Makefile.am: + * docs/gst_plugins_cache.json: + * docs/index.md: + * docs/meson.build: + * docs/plugin-index.md: + * docs/plugin-sitemap.txt: + * docs/sitemap.md: + * docs/sitemap.txt: + * docs/version.entities.in: + * gst/rtsp-server/meson.build: + * gst/rtsp-sink/meson.build: + * meson.build: + * meson_options.txt: + docs: Port to hotdoc + +2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-client.h: + rtsp-server: Fix various Since markers + +2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-stream.c: + rtsp-server: Add various Since: 1.14 markers + +2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + rtsp-server: Add various missing Since: 1.16 markers + +2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + rtspclientsink: Set async-handling=false for the internal bins + Without this we can easily run into a race condition with async state changes: + - the pipeline is doing an async state change + - we set the internal bins to PLAYING but that's ignored because an + async state change is currently pending + - the async state change finishes but does not change the state of the + internal bins because of locked_state==TRUE + - the internal bins stay in PAUSED forever + +2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + rtspclientsink: Use write_messages() API to send buffer lists in one go + And to write messages with multiple memories also via writev(). + +2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-server-object.h: + * gst/rtsp-server/rtsp-server.c: + rtsp-client: Handle Content-Length limitation + Add functionality to limit the Content-Length. + API addition, Enhancement. + Define an appropriate request size limit and reject requests + exceeding the limit with response status 413 Request Entity Too Large + Related to !182 + +2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com> + + * RELEASE: + * configure.ac: + * meson.build: + Back to development + === release 1.16.0 === 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com> @@ -1,14 +1,30 @@ -GSTREAMER 1.16 RELEASE NOTES +GSTREAMER 1.18 RELEASE NOTES -GStreamer 1.16.0 was originally released on 19 April 2019. +THESE RELEASE NOTES ARE A PLACEHOLDER, PLEASE BEAR WITH US WHILE WE +FINISH WRITING UP THE REAL THING. -See https://gstreamer.freedesktop.org/releases/1.16/ for the latest +GStreamer 1.18.0 has not yet been released. It is scheduled for release +in summer 2020 now. + +1.17.x is the unstable development series that is currently being +developed in the git master branch and which will eventually result in +1.18, and 1.17.1 is the current development release in that series. + +The schedule for the 1.18 development cycle is yet to be confirmed, but +it is expected that feature freeze will be in June/July 2020, followed +by several 1.17 pre-releases and then a new 1.18 stable release in +July/August 2020. + +1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10, +1.8, 1.6, 1.4, 1.2 and 1.0 release series. + +See https://gstreamer.freedesktop.org/releases/1.18/ for the latest version of this document. -_Last updated: Friday 19 April 2019, 00:00 UTC (log)_ +_Last updated: Thursday 18 June 2020, 16:00 UTC (log)_ Introduction @@ -23,1146 +39,133 @@ fixes and other improvements. Highlights -- GStreamer WebRTC stack gained support for data channels for - peer-to-peer communication based on SCTP, BUNDLE support, as well as - support for multiple TURN servers. - -- AV1 video codec support for Matroska and QuickTime/MP4 containers - and more configuration options and supported input formats for the - AOMedia AV1 encoder - -- Support for Closed Captions and other Ancillary Data in video - -- Support for planar (non-interleaved) raw audio - -- GstVideoAggregator, compositor and OpenGL mixer elements are now in - -base - -- New alternate fields interlace mode where each buffer carries a - single field - -- WebM and Matroska ContentEncryption support in the Matroska demuxer - -- new WebKit WPE-based web browser source element - -- Video4Linux: HEVC encoding and decoding, JPEG encoding, and improved - dmabuf import/export - -- Hardware-accelerated Nvidia video decoder gained support for VP8/VP9 - decoding, whilst the encoder gained support for H.265/HEVC encoding. - -- Many improvements to the Intel Media SDK based hardware-accelerated - video decoder and encoder plugin (msdk): dmabuf import/export for - zero-copy integration with other components; VP9 decoding; 10-bit - HEVC encoding; video post-processing (vpp) support including - deinterlacing; and the video decoder now handles dynamic resolution - changes. - -- The ASS/SSA subtitle overlay renderer can now handle multiple - subtitles that overlap in time and will show them on screen - simultaneously - -- The Meson build is now feature-complete (*) and it is now the - recommended build system on all platforms. The Autotools build is - scheduled to be removed in the next cycle. - -- The GStreamer Rust bindings and Rust plugins module are now - officially part of upstream GStreamer. - -- The GStreamer Editing Services gained a gesdemux element that allows - directly playing back serialized edit list with playbin or - (uri)decodebin - -- Many performance improvements +- FIXME Major new features and changes Noteworthy new API -- GstAggregator has a new "min-upstream-latency" property that forces - a minimum aggregate latency for the input branches of an aggregator. - This is useful for dynamic pipelines where branches with a higher - latency might be added later after the pipeline is already up and - running and where a change in the latency would be disruptive. This - only applies to the case where at least one of the input branches is - live though, it won’t force the aggregator into live mode in the - absence of any live inputs. - -- GstBaseSink gained a "processing-deadline" property and - setter/getter API to configure a processing deadline for live - pipelines. The processing deadline is the acceptable amount of time - to process the media in a live pipeline before it reaches the sink. - This is on top of the systemic latency that is normally reported by - the latency query. This defaults to 20ms and should make pipelines - such as v4l2src ! xvimagesink not claim that all frames are late in - the QoS events. Ideally, this should replace the "max-lateness" - property for most applications. - -- RTCP Extended Reports (XR) parsing according to RFC 3611: - Loss/Duplicate RLE, Packet Receipt Times, Receiver Reference Time, - Delay since the last Receiver (DLRR), Statistics Summary, and VoIP - Metrics reports. This only provides the ability to parse such - packets, generation of XR packets is not supported yet and XR - packets are not automatically parsed by rtpbin / rtpsession but must - be actively handled by the application. - -- a new mode for interlaced video was added where each buffer carries - a single field of interlaced video, with buffer flags indicating - whether the field is the top field or bottom field. Top and bottom - fields are expected to alternate in this mode. Caps for this - interlace mode must also carry a format:Interlaced caps feature to - ensure backwards compatibility. - -- The video library has gained support for three new raw pixel - formats: - - - Y410: packed 4:4:4 YUV, 10 bits per channel - - Y210: packed 4:2:2 YUV, 10 bits per channel - - NV12_10LE40: fully-packed 10-bit variant of NV12_10LE32, - i.e. without the padding bits - -- GstRTPSourceMeta is a new meta that can be used to transport - information about the origin of depayloaded or decoded RTP buffers, - e.g. when mixing audio from multiple sources into a single stream. A - new "source-info" property on the RTP depayloader base class - determines whether depayloaders should put this meta on outgoing - buffers. Similarly, the same property on RTP payloaders determines - whether they should use the information from this meta to construct - the CSRCs list on outgoing RTP buffers. - -- gst_sdp_message_from_text() is a convenience constructor to parse - SDPs from a string which is particularly useful for language - bindings. - -Support for Planar (Non-Interleaved) Raw Audio - -Raw audio samples are usually passed around in interleaved form in -GStreamer, which means that if there are multiple audio channels the -samples for each channel are interleaved in memory, e.g. -|LEFT|RIGHT|LEFT|RIGHT|LEFT|RIGHT| for stereo audio. A non-interleaved -or planar arrangement in memory would look like -|LEFT|LEFT|LEFT|RIGHT|RIGHT|RIGHT| instead, possibly with -|LEFT|LEFT|LEFT| and |RIGHT|RIGHT|RIGHT| residing in separate memory -chunks or separated by some padding. - -GStreamer has always had signalling for non-interleaved audio since -version 1.0, but it was never actually properly implemented in any -elements. audioconvert would advertise support for it, but wasn’t -actually able to handle it correctly. - -With this release we now have full support for non-interleaved audio as -well, which means more efficient integration with external APIs that -handle audio this way, but also more efficient processing of certain -operations like interleaving multiple 1-channel streams into a -multi-channel stream which can be done without memory copies now. - -New API to support this has been added to the GStreamer Audio support -library: There is now a new GstAudioMeta which describes how data is -laid out inside the buffer, and buffers with non-interleaved audio must -always carry this meta. To access the non-interleaved audio samples you -must map such buffers with gst_audio_buffer_map() which works much like -gst_buffer_map() or gst_video_frame_map() in that it will populate a -little GstAudioBuffer helper structure passed to it with the number of -samples, the number of planes and pointers to the start of each plane in -memory. This function can also be used to map interleaved audio buffers -in which case there will be only one plane of interleaved samples. - -Of course support for this has also been implemented in the various -audio helper and conversion APIs, base classes, and in elements such as -audioconvert, audioresample, audiotestsrc, audiorate. - -Support for Closed Captions and Other Ancillary Data in Video - -The video support library has gained support for detecting and -extracting Ancillary Data from videos as per the SMPTE S291M -specification, including: - -- a VBI (Vertical Blanking Interval) parser that can detect and - extract Ancillary Data from Vertical Blanking Interval lines of - component signals. This is currently supported for videos in v210 - and UYVY format. - -- a new GstMeta for closed captions: GstVideoCaptionMeta. This - supports the two types of closed captions, CEA-608 and CEA-708, - along with the four different ways they can be transported (other - systems are a superset of those). - -- a VBI (Vertical Blanking Interval) encoder for writing ancillary - data to the Vertical Blanking Interval lines of component signals. - -The new closedcaption plugin in gst-plugins-bad then makes use of all -this new infrastructure and provides the following elements: - -- cccombiner: a closed caption combiner that takes a closed captions - stream and another stream and adds the closed captions as - GstVideoCaptionMeta to the buffers of the other stream. - -- ccextractor: a closed caption extractor which will take - GstVideoCaptionMeta from input buffers and output them as a separate - closed captions stream. - -- ccconverter: a closed caption converter that can convert between - different formats - -- line21encoder, line21decoder: inject/extract line21 closed captions - to/from SD video streams - -- cc708overlay: decodes CEA 608/708 captions and overlays them on - video - -Additionally, the following elements have also gained Closed Caption -support: - -- qtdemux and qtmux support CEA 608/708 Closed Caption tracks - -- mpegvideoparse, h264parse extracts Closed Captions from MPEG-2/H.264 - video streams - -- avviddec, avvidenc, x264enc got support for extracting/injecting - Closed Captions - -- decklinkvideosink can output closed captions and decklinkvideosrc - can extract closed captions - -- playbin and playbin3 learned how to autoplug CEA 608/708 CC overlay - elements - -- the externally maintained ajavideosrc element for AJA capture cards - has support for extracting closed captions - -The rsclosedcaption plugin in the Rust plugins collection includes a -MacCaption (MCC) file parser and encoder. +- FIXME New Elements -- overlaycomposition: New element that allows applications to draw - GstVideoOverlayCompositions on a stream. The element will emit the - "draw" signal for each video buffer, and the application then - generates an overlay for that frame (or not). This is much more - performant than e.g. cairooverlay for many use cases, e.g. because - pixel format conversions can be avoided or the blitting of the - overlay can be delegated to downstream elements (such as - gloverlaycompositor). It’s particularly useful for cases where only - a small section of the video frame should be drawn on. - -- gloverlaycompositor: New OpenGL-based compositor element that - flattens any overlays from GstVideoOverlayCompositionMetas into the - video stream. This element is also always part of glimagesink. - -- glalpha: New element that adds an alpha channel to a video stream. - The values of the alpha channel can either be set to a constant or - can be dynamically calculated via chroma keying. It is similar to - the existing alpha element but based on OpenGL. Calculations are - done in floating point so results may not be identical to the output - of the existing alpha element. - -- rtpfunnel funnels together RTP streams into a single session. Use - cases include multiplexing and bundle. webrtcbin uses it to - implement BUNDLE support. - -- testsrcbin is a source element that provides an audio and/or video - stream and also announces them using the recently-introduced - GstStream API. This is useful for testing elements such as playbin3 - or uridecodebin3 etc. - -- New closed caption elements: cccombiner, ccextractor, ccconverter, - line21encoder, line21decoder and cc708overlay (see above) - -- wpesrc: new source element acting as a Web Browser based on WebKit - WPE - -- Two new OpenCV-based elements: cameracalibrate and cameraundistort - that can communicate to figure out distortion correction parameters - for a camera and correct for the distortion. - -- New sctp plugin based on usrsctp with sctpenc and sctpdec elements. - These elements are used inside webrtcbin for implementing data - channels. +- FIXME New element features and additions -- playbin3, playbin and playsink have gained a new "text-offset" - property to adjust the positioning of the selected subtitle stream - vis-a-vis the audio and video streams. This uses subtitleoverlay’s - new "subtitle-ts-offset" property. GstPlayer has gained matching API - for this, namely gst_player_get_text_video_offset(). - -- playbin3 buffering improvements: in network playback scenarios there - may be multiple inputs to decodebin3, and buffering will be done - before decodebin3 using queue2 or downloadbuffer elements inside - urisourcebin. Since this is before any parsers or demuxers there may - not be any bitrate information available for the various streams, so - it was difficult to configure the buffering there smartly within - global constraints. This was improved now: The queue2 elements - inside urisourcebin will now use the new bitrate query to figure out - a bitrate estimate for the stream if no bitrate was provided by - upstream, and urisourcebin will use the bitrates of the individual - queues to distribute the globally-set "buffer-size" budget in bytes - to the various queues. urisourcebin also gained "low-watermark" and - "high-watermark" properties which will be proxied to the internal - queues, as well as a read-only "statistics" property which allows - querying of the minimum/maximum/average byte and time levels of the - queues inside the urisourcebin in question. - -- splitmuxsink has gained a couple of new features: - - - new "async-finalize" mode: This mode is useful for muxers or - outputs that can take a long time to finalize a file. Instead of - blocking the whole upstream pipeline while the muxer is doing - its stuff, we can unlink it and spawn a new muxer + sink - combination to continue running normally. This requires us to - receive the muxer and sink (if needed) as factories via the new - "muxer-factory" and "sink-factory" properties, optionally - accompanied by their respective properties structures (set via - the new "muxer-properties" and "sink-properties" properties). - There are also new "muxer-added" and "sink-added" signals in - case custom code has to be called for them to configure them. - - - "split-at-running-time" action signal: When called by the user, - this action signal ends the current file (and starts a new one) - as soon as the given running time is reached. If called multiple - times, running times are queued up and processed in the order - they were given. - - - "split-after" action signal to finish outputting the current GOP - to the current file and then start a new file as soon as the GOP - is finished and a new GOP is opened (unlike the existing - "split-now" which immediately finishes the current file and - writes the current GOP into the next newly-started file). - - - "reset-muxer" property: when unset, the muxer is reset using - flush events instead of setting its state to NULL and back. This - means the muxer can keep state across resets, e.g. mpegtsmux - will keep the continuity counter continuous across segments as - required by hlssink2. - -- qtdemux gained PIFF track encryption box support in addition to the - already-existing PIFF sample encryption support, and also allows - applications to select which encryption system to use via a - "drm-preferred-decryption-system-id" context in case there are - multiple options. - -- qtmux: the "start-gap-threshold" property determines now whether an - edit list will be created to account for small gaps or offsets at - the beginning of a stream in case the start timestamps of tracks - don’t line up perfectly. Previously the threshold was hard-coded to - 1% of the (video) frame duration, now it is 0 by default (so edit - list will be created even for small differences), but fully - configurable. - -- rtpjitterbuffer has improved end-of-stream handling - -- rtpmp4vpay will be prefered over rtpmp4gpay for MPEG-4 video in - autoplugging scenarios now - -- rtspsrc now allows applications to send RTSP SET_PARAMETER and - GET_PARAMETER requests using action signals. - -- rtspsrc has a small (100ms) configurable teardown delay by default - to try and make sure an RTSP TEARDOWN request gets sent out when the - source element shuts down. This will block the downward PAUSED to - READY state change for a short time, but can be disabled where it’s - a problem. Some servers only allow a limited number of concurrent - clients, so if no proper TEARDOWN is sent new clients may have - problems connecting to the server for a while. - -- souphttpsrc behaves better with low bitrate streams now. Before it - would increase the read block size too quickly which could lead to - it not reading any data from the socket for a very long time with - low bitrate streams that are output live downstream. This could lead - to servers kicking off the client. - -- filesink: do internal buffering to avoid performance regression with - small writes since we bypass libc buffering by using writev() - instead of fwrite() - -- identity: add "eos-after" property and fix "error-after" property - when the element is reused - -- input-selector: lets context queries pass through, so that - e.g. upstream OpenGL elements can use contexts and displays - advertised by downstream elements - -- queue2: avoid ping-pong between 0% and 100% buffering messages if - upstream is pushing buffers larger than one of its limits, plus - performance optimisations - -- opusdec: new "phase-inversion" property to control phase inversion. - When enabled, this will slightly increase stereo quality, but - produces a stream that when downmixed to mono will suffer audio - distortions. - -- The x265enc HEVC encoder also exposes a "key-int-max" property to - configure the maximum allowed GOP size now. - -- decklinkvideosink has seen stability improvements for long-running - pipelines (potential crash due to overflow of leaked clock refcount) - and clock-slaving improvements when performing flushing seeks - (causing stalls in the output timeline), pausing and/or buffering. - -- srtpdec, srtpenc: add support for MKIs which allow multiple keys to - be used with a single SRTP stream - -- srtpdec, srtpenc: add support for AES-GCM and also add support for - it in gst-rtsp-server and rtspsrc. - -- The srt Secure Reliable Transport plugin has integrated server and - client elements srt{client,server}{src,sink} into one (srtsrc and - srtsink), since SRT connection mode can be changed by uri - parameters. - -- h264parse and h265parse will handle SEI recovery point messages and - mark recovery points as keyframes as well (in addition to IDR - frames) - -- webrtcbin: "add-turn-server" action signal to pass multiple ICE - relays (TURN servers). - -- The removesilence element has received various new features and - properties, such as a "threshold" property, detecting silence only - after minimum silence time/buffers, a "silent" property to control - bus message notifications as well as a "squash" property. - -- AOMedia AV1 decoder gained support for 10/12bit decoding whilst the - AV1 encoder supports more image formats and subsamplings now and - acquired support for rate control and profile related configuration. - -- The Fraunhofer fdkaac plugin can now be built against the 2.0.0 - version API and has improved multichannel support - -- kmssink now supports unpadded 24-bit RGB and can configure mode - setting from video info, which enables display of multi-planar - formats such as I420 or NV12 with modesetting. It has also gained a - number of new properties: The "restore-crtc" property does what it - says on the tin and is enabled by default. "plane-properties" and - "connector-properties" can be used to pass custom properties to the - DRM. - -- waylandsink has a "fullscreen" property now and supports the - XDG-Shell protocol. - -- decklinkvideosink, decklinkvideosrc support selecting between - half/full duplex - -- The vulkan plugin gained support for macOS and iOS via MoltenVK in - addition to the existing support for X11 and Wayland - -- imagefreeze has a new num-buffers property to limit the number of - buffers that are produced and to send an EOS event afterwards - -- webrtcbin has a new, introspectable get-transceiver signal in - addition to the old get-transceivers signal that couldn’t be used - from bindings - -- Support for per-element latency information was added to the latency - tracer +- FIXME Plugin and library moves -- The stereo element was moved from -bad into the existing audiofx - plugin in -good. If you get duplicate type registration warnings - when upgrading, check that you don’t have a stale stereoplugin lying - about somewhere. - -GstVideoAggregator, compositor, and OpenGL mixer elements moved from -bad to -base - -GstVideoAggregator is a new base class for raw video mixers and muxers -and is based on GstAggregator. It provides defined-latency mixing of raw -video inputs and ensures that the pipeline won’t stall even if one of -the input streams stops producing data. - -As part of the move to stabilise the API there were some last-minute API -changes and clean-ups, but those should mostly affect internal elements. -Most notably, the "ignore-eos" pad property was renamed to -"repeat-after-eos" and the conversion code was moved to a -GstVideoAggregatorConvertPad subclass to avoid code duplication, make -things less awkward for subclasses like the OpenGL-based video mixer, -and make the API more consistent with the audio aggregator API. - -It is used by the compositor element, which is a replacement for -‘videomixer’ which did not handle live inputs very well. compositor -should behave much better in that respect and generally behave as one -would expected in most scenarios. - -The compositor element has gained support for per-pad blending mode -operators (SOURCE, OVER, ADD) which determines what operator to use for -blending this pad over the previous ones. This can be used to implement -crossfading and the available operators can be extended in the future as -needed. - -A number of OpenGL-based video mixer elements (glvideomixer, glmixerbin, -glvideomixerelement, glstereomix, glmosaic) which are built on top of -GstVideoAggregator have also been moved from -bad to -base now. These -elements have been merged into the existing OpenGL plugin, so if you get -duplicate type registration warnings when upgrading, check that you -don’t have a stale openglmixers plugin lying about somewhere. +- FIXME Plugin removals The following plugins have been removed from gst-plugins-bad: -- The experimental daala plugin has been removed, since it’s not so - useful now that all effort is focused on AV1 instead, and it had to - be enabled explicitly with --enable-experimental anyway. - -- The spc plugin has been removed. It has been replaced by the gme - plugin. - -- The acmmp3dec and acmenc plugins for Windows have been removed. ACM - is an ancient legacy API and there was no point in keeping the - plugins around for a licensed MP3 decoder now that the MP3 patents - have expired and we have a decoder in -good. We also didn’t ship - these in our cerbero-built Windows packages, so it’s unlikely that - they’ll be missed. +- FIXME Miscellaneous API additions -- GstBitwriter: new generic bit writer API to complement the existing - bit reader - -- gst_buffer_new_wrapped_bytes() creates a wrap buffer from a GBytes - -- gst_caps_set_features_simple() sets a caps feature on all the - structures of a GstCaps - -- New GST_QUERY_BITRATE query: This allows determining from downstream - what the expected bitrate of a stream may be which is useful in - queue2 for setting time based limits when upstream does not provide - timing information. tsdemux, qtdemux and matroskademux have basic - support for this query on their sink pads. - -- elements: there is a new “Hardware” class specifier. Elements - interacting with hardware devices should specify this classifier in - their element factory class metadata. This is useful to advertise as - one might need to put such elements into READY state to test if the - hardware is present in the system for example. - -- protection: Add a new definition for unspecified system protection, - GST_PROTECTION_UNSPECIFIED_SYSTEM_ID - -- take functions for various mini objects that didn’t have them yet: - gst_query_take(), gst_message_take(), gst_tag_list_take(), - gst_buffer_list_take(). Unlike the various _replace() functions - _take() does not increase the reference count but takes ownership of - the mini object passed. - -- clear functions for various mini object types and GstObject which - unrefs the object or mini object (if non-NULL) and sets the variable - pointed to to NULL: gst_clear_structure(), gst_clear_tag_list(), - gst_clear_query(), gst_clear_message(), gst_clear_event(), - gst_clear_caps(), gst_clear_buffer_list(), gst_clear_buffer(), - gst_clear_mini_object(), gst_clear_object() - -- miniobject: new API gst_mini_object_add_parent() and - gst_mini_object_remove_parent() to set parent pointers on mini - objects to ensure correct writability: Every container of - miniobjects now needs to store itself as parent in the child object, - and remove itself again later. A mini object is then only writable - if there is at most one parent, that parent is writable itself, and - the reference count of the mini object is 1. GstBuffer (for - memories), GstBufferList (for buffers), GstSample (for caps, buffer, - bufferlist), and GstVideoOverlayComposition were updated - accordingly. Without this it was possible to have e.g. a buffer list - with a refcount of 2 used in two places at once that both modify the - same buffer with refcount 1 at the same time wrongly thinking it is - writable even though it’s really not. - -- poll: add API to watch for POLLPRI and stop treating POLLPRI as a - read. This is useful to wait for video4linux events which are - signalled via POLLPRI. - -- sample: new API to update the contents of a GstSample and make it - writable: gst_sample_set_buffer(), gst_sample_set_caps(), - gst_sample_set_segment(), gst_sample_set_info(), plus - gst_sample_is_writable() and gst_sample_make_writable(). This makes - it possible to reuse a sample object and avoid unnecessary memory - allocations, for example in appsink. - -- ClockIDs now keep a weak reference to underlying clock to avoid - crashes in basesink in corner cases where a clock goes away while - the ClockID is still in use, plus some new API - (gst_clock_id_get_clock(), gst_clock_id_uses_clock()) to check the - clock a ClockID is linked to. - -- The GstCheck unit test library gained a - fail_unless_equals_clocktime() convenience macro as well as some new - GstHarness API for for proposing meta APIs from the allocation - query: gst_harness_add_propose_allocation_meta(). ASSERT_CRITICAL() - checks in unit tests are now skipped if GStreamer was compiled with - GST_DISABLE_GLIB_CHECKS. - -- gst_audio_buffer_truncate() convenience function to truncate a raw - audio buffer - -- GstDiscoverer has support for caching the results of discovery in - the default cache directory. This can be enabled with the use-cache - property and is disabled by default. - -- GstMeta that are attached to GstBuffers are now always stored in the - order in which they were added. - -- Additional support for signalling ONVIF specific features were - added: the SEEK event can store a trickmode-interval now and support - for the Rate-Control and Frames RTSP headers was added to the RTSP - library. +- FIXME Miscellaneous performance and memory optimisations As always there have been many performance and memory usage improvements -across all components and modules. Some of them (such as dmabuf -import/export) have already been mentioned elsewhere so won’t be -repeated here. +across all components and modules. Some of them have already been +mentioned elsewhere so won’t be repeated here. The following list is only a small snapshot of some of the more interesting optimisations that haven’t been mentioned in other contexts yet: -- The GstVideoEncoder and GstVideoDecoder base classes now release the - STREAM_LOCK when pushing out buffers, which means (multi-threaded) - encoders and decoders can now receive and continue to process input - buffers whilst waiting for downstream elements in the pipeline to - process the buffer that was pushed out. This increases throughput - and reduces processing latency, also and especially for - hardware-accelerated encoder/decoder elements. - -- GstQueueArray has seen a few API additions - (gst_queue_array_peek_nth(), gst_queue_array_set_clear_func(), - gst_queue_array_clear()) so that it can be used in other places like - GstAdapter instead of a GList, which reduces allocations and - improves performance. - -- appsink now reuses the sample object in pull_sample() if possible - -- rtpsession only starts the RTCP thread when it’s actually needed now - -- udpsrc uses a buffer pool now and the GstUdpSrc object structure was - optimised for better cache performance +- FIXME GstPlayer -- API was added to fine-tune the synchronisation offset between - subtitles and video +- FIXME Miscellaneous changes -- As a result of moving to newer FFmpeg APIs, encoder and decoder - elements exposed by the GStreamer FFmpeg wrapper plugin (gst-libav) - may have seen possibly incompatible changes to property names and/or - types, and not all properties exposed might be functional. We are - still reviewing the new properties and aim to minimise breaking - changes at least for the most commonly-used properties, so please - report any issues you run into! +- FIXME OpenGL integration -- The OpenGL mixer elements have been moved from -bad to - gst-plugins-base (see above) - -- The Mesa GBM backend now supports headless mode - -- gloverlaycompositor: New OpenGL-based compositor element that - flattens any overlays from GstVideoOverlayCompositionMetas into the - video stream. - -- glalpha: New element that adds an alpha channel to a video stream. - The values of the alpha channel can either be set to a constant or - can be dynamically calculated via chroma keying. It is similar to - the existing alpha element but based on OpenGL. Calculations are - done in floating point so results may not be identical to the output - of the existing alpha element. - -- glupload: Implement direct dmabuf uploader, the idea being that some - GPUs (like the Vivante series) can actually perform the YUV->RGB - conversion internally, so no custom conversion shaders are needed. - To make use of this feature, we need an additional uploader that can - import DMABUF FDs and also directly pass the pixel format, relying - on the GPU to do the conversion. - -- The OpenGL library no longer restores the OpenGL viewport. This is a - performance optimization to not require performing multiple - expensive glGet*() function calls per frame. This affects any - application or plugin use of the following functions and objects: - - glcolorconvert library object (not the element) - - glviewconvert library object (not the element) - - gst_gl_framebuffer_draw_to_texture() - - custom GstGLWindow implementations +- FIXME Tracing framework and debugging improvements -- There is now a GDB PRETTY PRINTER FOR VARIOUS GSTREAMER TYPES: For - GstObject pointers the type and name is added, e.g. - 0x5555557e4110 [GstDecodeBin|decodebin0]. For GstMiniObject pointers - the object type is added, e.g. 0x7fffe001fc50 [GstBuffer]. For - GstClockTime and GstClockTimeDiff the time is also printed in human - readable form, e.g. 150116219955 [+0:02:30.116219955]. - -- GDB EXTENSION WITH TWO CUSTOM GDB COMMANDS gst-dot AND gst-print: - - - gst-dot creates dot files that a very close to what - GST_DEBUG_BIN_TO_DOT_FILE() produces, but object properties and - buffer contents such as codec-data in caps are not available. - - - gst-print produces high-level information about a GStreamer - object. This is currently limited to pads for GstElements and - events for the pads. The output may look like this: - -- gst_structure_to_string() now serialises the actual value of - pointers when serialising GstStructures instead of claiming they’re - NULL. This makes debug logging in various places less confusing, - because it’s clear now that structure fields actually hold valid - objects. Such object pointer values will never be deserialised - however. +- FIXME Tools -- gst-inspect-1.0 has coloured output now and will automatically use a - pager if the output does not fit on a page. This only works in a - UNIX environment and if the output is not piped, and on Windows 10 - build 16257 or newer. If you don’t like the colours you can disable - them by setting the GST_INSPECT_NO_COLORS=1 environment variable or - passing the --no-color command line option. +- FIXME GStreamer RTSP server -- Improved backlog handling when using TCP interleaved for data - transport. Before there was a fixed maximum size for backlog - messages, which was prone to deadlocks and made it difficult to - control memory usage with the watch backlog. The RTSP server now - limits queued TCP data messages to one per stream, moving queuing of - the data into the pipeline and leaving the RTSP connection - responsive to RTSP messages in both directions, preventing all those - problems. - -- Initial ULP Forward Error Correction support in rtspclientsink and - for RECORD mode in the server. - -- API to explicitly enable retransmission requests (RTX) - -- Lots of multicast-related fixes - -- rtsp-auth: Add support for parsing .htdigest files +- FIXME GStreamer VAAPI -- Support Wayland’s display for context sharing, so the application - can pass its own wl_display in order to be used for the VAAPI - display creation. - -- A lot of work to support new Intel hardware using media-driver as VA - backend. - -- For non-x86 devices, VAAPI display can instantiate, through DRM, - with no PCI bus. This enables the usage of libva-v4l2-request - driver. - -- Added support for XDG-shell protocol as wl_shell replacement which - is currently deprecated. This change add as dependency - wayland-protocol. - -- GstVaapiFilter, GstVaapiWindow, and GstVaapiDecoder classes now - inherit from GstObject, gaining all the GStreamer’s instrumentation - support. - -- The metadata now specifies the plugin as Hardware class. - -- H264 decoder is more stable with problematic streams. - -- In H265 decoder added support for profiles main-422-10 (P010_10LE), - main-444 (AYUV) and main-444-10 (Y410) - -- JPEG decoder handles dynamic resolution changes. - -- More specification adherence in H264 and H265 encoders. +- FIXME GStreamer OMX -- Add support of NV16 format to video encoders input. - -- Video decoders now handle the ALLOCATION query to tell upstream - about the number of buffers they require. Video encoders will also - use this query to adjust their number of allocated buffers - preventing starvation when using dynamic buffer mode. - -- The OMX_PERFORMANCE debug category has been renamed to OMX_API_TRACE - and can now be used to track a widder variety of interactions - between OMX and GStreamer. - -- Video encoders will now detect frame rate only changes and will - inform OMX about it rather than doing a full format reset. - -- Various Zynq UltraScale+ specific improvements: - - Video encoders are now able to import dmabuf from upstream. - - Support for HEVC range extension profiles and more AVC profiles. - - We can now request video encoders to generate an IDR using the - force key unit event. +- FIXME GStreamer Editing Services and NLE -- Added a gesdemux element, it is an auto pluggable element that - allows decoding edit list like files supported by GES - -- Added gessrc which wraps a GESTimeline as a standard source element - (implementing the ges protocol handler) - -- Added basic support for videorate::rate property potentially - allowing changing playback speed - -- Layer priority is now fully automatic and they should be moved with - the new ges_timeline_move_layer method, ges_layer_set_priority is - now deprecated. - -- Added a ges_timeline_element_get_layer_priority so we can simply get - all information about GESTimelineElement position in the timeline - -- GESVideoSource now auto orientates the images if it is defined in a - meta (overridable). - -- Added some PyGObject overrides to make the API more pythonic - -- The threading model has been made more explicit with safe guard to - make sure not thread safe APIs are not used from the wrong threads. - It is also now possible to properly handle in what thread the API - should be used. - -- Optimized GESClip and GESTrackElement creation - -- Added a way to compile out the old, unused and deprecated - GESPitiviFormatter - -- Re implemented the timeline editing API making it faster and making - the code much more maintainable - -- Simplified usage of nlecomposition outside GES by removing quirks in - it API usage and removing the need to treat it specially from an - application perspective. - -- ges-launch-1.0: - - - Added support to add titles to the timeline - - Enhance the help auto generating it from the code - -- Deprecate ges_timeline_load_from_uri as loading the timeline should - be done through a project now - -- MANY leaks have been plugged and the unit testsuite is now “leak - free” +- FIXME GStreamer validate -- Added an action type to verify the checksum of the sink last-sample - -- Added an include keyword to validate scenarios - -- Added the notion of variable in scenarios, with the set-vars keyword - -- Started adding support for “performance” like tests by allowing to - define the number of dropped buffers or the minimum buffer frequency - on a specific pad - -- Added a validateflow plugin which allows defining the data flow to - be seen on a particular pad and verifying that following runs match - the expectations - -- Added support for appsrc based test definition so we can instrument - the data pushed into the pipeline from scenarios - -- Added a mockdecryptor allowing adding tests with on encrypted files, - the element will potentially be instrumented with a validate - scenario - -- gst-validate-launcher: - - - Cleaned up output - - - Changed the default for “muting” tests as user doesn’t expect - hundreds of windows to show up when running the testsuite - - - Fixed the outputted xunit files to be compatible with GitLab - - - Added support to run tests on media files in push mode (using - pushfile://) - - - Added support for running inside gst-build - - - Added support for running ssim tests on rendered files - - - Added a way to simply define tests on pipelines through a simple - .json file - - - Added a python app to easily run python testsuite reusing all - the launcher features - - - Added flatpak knowledge so we can print backtrace even when - running from within flatpak - - - Added a way to automatically generated “known issues” - suppressions lines - - - Added a way to rerun tests to check if they are flaky and added - a way to tolerate tests known to be flaky - - - Add a way to output html log files +- FIXME GStreamer Python Bindings -- add binding for gst_pad_set_caps() - -- pygobject dependency requirement was bumped to >= 3.8 - -- new audiotestsrc, audioplot, and mixer plugin examples, and a - dynamic pipeline example +- FIXME GStreamer C# Bindings -- bindings for the GstWebRTC library +- FIXME GStreamer Rust Bindings -The GStreamer Rust bindings are now officially part of the GStreamer -project and are also maintained in the GStreamer GitLab. - -The releases will generally not be synchronized with the releases of -other GStreamer parts due to dependencies on other projects. - -Also unlike the other GStreamer libraries, the bindings will not commit -to full API stability but instead will follow the approach that is -generally taken by Rust projects, e.g.: - -1) 0.12.X will be completely API compatible with all other 0.12.Y - versions. -2) 0.12.X+1 will contain bugfixes and compatible new feature additions. -3) 0.13.0 will _not_ be backwards compatible with 0.12.X but projects - will be able to stay at 0.12.X without any problems as long as they - don’t need newer features. - -The current stable release is 0.12.2 and the next release series will be -0.13, probably around March 2019. - -At this point the bindings cover most of GStreamer core (except for most -notably GstAllocator and GstMemory), and most parts of the app, audio, -base, check, editing-services, gl, net. pbutils, player, rtsp, -rtsp-server, sdp, video and webrtc libraries. - -Also included is support for creating subclasses of the following types -and writing GStreamer plugins: - -- gst::Element -- gst::Bin and gst::Pipeline -- gst::URIHandler and gst::ChildProxy -- gst::Pad, gst::GhostPad -- gst_base::Aggregator and gst_base::AggregatorPad -- gst_base::BaseSrc and gst_base::BaseSink -- gst_base::BaseTransform - -Changes to 0.12.X since 0.12.0 - -Fixed - -- PTP clock constructor actually creates a PTP instead of NTP clock - -Added - -- Bindings for GStreamer Editing Services -- Bindings for GStreamer Check testing library -- Bindings for the encoding profile API (encodebin) - -- VideoFrame, VideoInfo, AudioInfo, StructureRef implements Send and - Sync now -- VideoFrame has a function to get the raw FFI pointer -- From impls from the Error/Success enums to the combined enums like - FlowReturn -- Bin-to-dot file functions were added to the Bin trait -- gst_base::Adapter implements SendUnique now -- More complete bindings for the gst_video::VideoOverlay interface, - especially - gst_video::is_video_overlay_prepare_window_handle_message() - -Changed - -- All references were updated from GitHub to freedesktop.org GitLab -- Fix various links in the README.md -- Link to the correct location for the documentation -- Remove GitLab badge as that only works with gitlab.com currently - -Changes in git master for 0.13 - -Fixed - -- gst::tag::Album is the album tag now instead of artist sortname - -Added - -- Subclassing infrastructure was moved directly into the bindings, - making the gst-plugin crate deprecated. This involves many API - changes but generally cleans up code and makes it more flexible. - Take a look at the gst-plugins-rs crate for various examples. - -- Bindings for CapsFeatures and Meta -- Bindings for - ParentBufferMeta,VideoMetaandVideoOverlayCompositionMeta` -- Bindings for VideoOverlayComposition and VideoOverlayRectangle -- Bindings for VideoTimeCode - -- UniqueFlowCombiner and UniqueAdapter wrappers that make use of the - Rust compile-time mutability checks and expose more API in a safe - way, and as a side-effect implement Sync and Send now - -- More complete bindings for Allocation Query -- pbutils functions for codec descriptions -- TagList::iter() for iterating over all tags while getting a single - value per tag. The old ::iter_tag_list() function was renamed to - ::iter_generic() and still provides access to each value for a tag -- Bus::iter() and Bus::iter_timed() iterators around the corresponding - ::pop\*() functions - -- serde serialization of Value can also handle Buffer now - -- Extensive comments to all examples with explanations -- Transmuxing example showing how to use typefind, multiqueue and - dynamic pads -- basic-tutorial-12 was ported and added - -Changed - -- Rust 1.31 is the minimum supported Rust version now -- Update to latest gir code generator and glib bindings - -- Functions returning e.g. gst::FlowReturn or other “combined” enums - were changed to return split enums like - Result<gst::FlowSuccess, gst::FlowError> to allow usage of the - standard Rust error handling. - -- MiniObject subclasses are now newtype wrappers around the underlying - GstRc<FooRef> wrapper. This does not change the API in any breaking - way for the current usages, but allows MiniObjects to also be - implemented in other crates and makes sure rustdoc places the - documentation in the right places. - -- BinExt extension trait was renamed to GstBinExt to prevent conflicts - with gtk::Bin if both are imported - -- Buffer::from_slice() can’t possible return None - -- Various clippy warnings +- FIXME GStreamer Rust Plugins -Like the GStreamer Rust bindings, the Rust plugins are now officially -part of the GStreamer project and are also maintained in the GStreamer -GitLab. - -In the 0.3.x versions this contained infrastructure for writing -GStreamer plugins in Rust, and a set of plugins. - -In git master that infrastructure was moved to the GLib and GStreamer -bindings directly, together with many other improvements that were made -possible by this, so the gst-plugins-rs repository only contains -GStreamer elements now. - -Elements included are: - -- Tutorials plugin: identity, rgb2gray and sinesrc with extensive - comments - -- rsaudioecho, a port of the audiofx element - -- rsfilesrc, rsfilesink - -- rsflvdemux, a FLV demuxer. Not feature-equivalent with flvdemux yet - -- threadshare plugin: ts-appsrc, ts-proxysrc/sink, ts-queue, ts-udpsrc - and ts-tcpclientsrc elements that use a fixed number of threads and - share them between instances. For more background about these - elements see Sebastian’s talk “When adding more threads adds more - problems - Thread-sharing between elements in GStreamer” at the - GStreamer Conference 2017. +- FIXME -- rshttpsrc, a HTTP source around the hyper/reqwest Rust libraries. - Not feature-equivalent with souphttpsrc yet. -- togglerecord, an element that allows to start/stop recording at any - time and keeps all audio/video streams in sync. - -- mccparse and mccenc, parsers and encoders for the MCC closed caption - file format. - -Changes to 0.3.X since 0.3.0 - -- All references were updated from GitHub to freedesktop.org GitLab -- Fix various links in the README.md -- Link to the correct location for the documentation - -Changes in git master for 0.4 - -- togglerecord: Switch to parking_lot crate for mutexes/condition - variables for lower overhead -- Merge threadshare plugin here -- New closedcaption plugin with mccparse and mccenc elements -- New identity element for the tutorials plugin - -- Register plugins statically in tests instead of relying on the - plugin loader to find the shared library in a specific place - -- Update to the latest API changes in the GLib and GStreamer bindings -- Update to the latest versions of all crates +Build and Dependencies +- The Autotools build system has finally been removed in favour of the + Meson build system. Developers who currently use gst-uninstalled + should move to gst-build. -Build and Dependencies +- API and plugin documentation are no longer built with gtk_doc. The + gtk_doc documentation has been removed in favour of a new unified + documentation module built with hotdoc. The intention is to + distribute the generated documentation in form of tarballs alongside + releases. -- The MESON BUILD SYSTEM BUILD IS NOW FEATURE-COMPLETE (*) and it is - now the recommended build system on all platforms and also used by - Cerbero to build GStreamer on all platforms. The Autotools build is - scheduled to be removed in the next cycle. Developers who currently - use gst-uninstalled should move to gst-build. The build option - naming has been cleaned up and made consistent and there are now - feature options to enable/disable plugins and various other features - on a case-by-case basis. (*) with the exception of plugin docs which - will be handled differently in future - -- Symbol export in libraries is now controlled via explicit exports - using symbol visibility or export defines where supported, to ensure - consistency across all platforms. This also allows libraries to have - exports that vary based on detected platform features and configure - options as is the case with the GStreamer OpenGL integration library - for example. A few symbols that had been exported by accident in - earlier versions may no longer be exported. These symbols will not - have had declarations in any public header files then though and - would not have been usable. - -- The GStreamer FFmpeg wrapper plugin (gst-libav) now depends on - FFmpeg 4.x and uses the new FFmpeg 4.x API and stopped relying on - ancient API that was removed with the FFmpeg 4.x release. This means - that it is no longer possible to build this module against an older - system-provided FFmpeg 3.x version. Use the internal FFmpeg 4.x copy - instead if you build using autotools, or use gst-libav 1.14.x - instead which targets the FFmpeg 3.x API and _should_ work fine in - combination with a newer GStreamer. It’s difficult for us to support - both old and new FFmpeg APIs at the same time, apologies for any - inconvenience caused. - -- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and - nvenc can be built against CUDA Toolkit versions 9 and 10.0 now. The - dynlink interface has been dropped since it’s deprecated in 10.0. - -- The (optional) OpenCV requirement has been bumped to >= 3.0.0 and - the plugin can also be built against OpenCV 4.x now. - -- New sctp plugin based on usrsctp (for WebRTC data channels) +- FIXME Cerbero @@ -1172,221 +175,66 @@ Windows, Android, iOS and macOS. Cerbero has seen a number of improvements: -- Cerbero has been ported to Python 3 and requires Python 3.5 or newer - now - -- Source tarballs are now protected by checksums in the recipes to - guard against download errors and malicious takeover of projects or - websites. In addition, downloads are only allowed via secure - transports now and plain HTTP, FTP and git:// transports are not - allowed anymore. - -- There is now a new fetch-bootstrap command which downloads sources - required for bootstrapping, with an optional --build-tools-only - argument to match the bootstrap --build-tools-only command. - -- The bootstrap, build, package and bundle-source commands gained a - new --offline switch that ensures that only sources from the cache - are used and never downloaded via the network. This is useful in - combination with the fetch and fetch-bootstrap commands that acquire - sources ahead of time before any build steps are executed. This - allows more control over the sources used and when sources are - updated, and is particularly useful for build environments that - don’t have network access. - -- bootstrap --assume-yes will automatically say ‘yes’ to any - interactive prompts during the bootstrap stage, such as those from - apt-get or yum. - -- bootstrap --system-only will only bootstrap the system without build - tools. - -- Manifest support: The build manifest can be used in continuous - integration (CI) systems to fixate the Git revision of certain - projects so that all builds of a pipeline are on the same reference. - This is used in GStreamer’s gitlab CI for example. It can also be - used in order to re-produce a specific build. To set a manifest, you - can set manifest = 'my_manifest.xml' in your configuration file, or - use the --manifest command line option. The command line option will - take precendence over anything specific in the configuration file. - -- The new build-deps command can be used to build only the - dependencies of a recipe, without the recipe itself. - -- new --list-variants command to list available variants - -- variants can now be set on the command line via the -v option as a - comma-separated list. This overrides any variants set in any - configuration files. - -- new qt5, intelmsdk and nvidia variants for enabling Qt5 and hardware - codec support. See the Enabling Optional Features with Variants - section in the Cerbero documentation for more details how to enable - and use these variants. - -- A new -t / --timestamp command line switch makes commands print - timestamps +- FIXME Platform-specific changes and improvements Android -- toolchain: update compiler to clang and NDKr18. NDK r18 removed the - armv5 target and only has Android platforms that target at least - armv7 so the armv5 target is not useful anymore. - -- The way that GIO modules are named has changed due to upstream GLib - natively adding support for loading static GIO modules. This means - that any GStreamer application using gnutls for SSL/TLS on the - Android or iOS platforms (or any other setup using static libraries) - will fail to link looking for the g_io_module_gnutls_load_static() - function. The new function name is now - g_io_gnutls_load(gpointer data). data can be NULL for a static - library. Look at this commit for the necessary change in the - examples. - -- various build issues on Android have been fixed. +- FIXME macOS and iOS -- various build issues on iOS have been fixed. - -- the minimum required iOS version is now 9.0. The difference in - adoption between 8.0 and 9.0 is 0.1% and the bump to 9.0 fixes some - build issues. - -- The way that GIO modules are named has changed due to upstream GLib - natively adding support for loading static GIO modules. This means - that any GStreamer application using gnutls for SSL/TLS on the - Android or iOS platforms (or any other setup using static libraries) - will fail to link looking for the g_io_module_gnutls_load_static() - function. The new function name is now - g_io_gnutls_load(gpointer data). data can be NULL for a static - library. Look at this commit for the necessary change in the - examples. +- FIXME Windows -- The webrtcdsp element is shipped again as part of the Windows binary - packages, the build system issue has been resolved. - -- ‘Inconsistent DLL linkage’ warnings when building with MSVC have - been fixed - -- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and - nvenc build on Windows now, also with MSVC and using Meson. - -- The ksvideosrc camera capture plugin supports 16-bit grayscale video - now +- toolchain upgrade -- The wasapisrc audio capture element implements loopback recording - from another output device or sink - -- wasapisink recover from low buffer levels in shared mode and some - exclusive mode fixes - -- dshowsrc now implements the GstDeviceMonitor interface +- FIXME Contributors -Aaron Boxer, Aleix Conchillo Flaqué, Alessandro Decina, Alexandru Băluț, -Alex Ashley, Alexey Chernov, Alicia Boya García, Amit Pandya, Andoni -Morales Alastruey, Andreas Frisch, Andre McCurdy, Andy Green, Anthony -Violo, Antoine Jacoutot, Antonio Ospite, Arun Raghavan, Aurelien Jarno, -Aurélien Zanelli, ayaka, Bananahemic, Bastian Köcher, Branko Subasic, -Brendan Shanks, Carlos Rafael Giani, Charlie Turner, Christoph Reiter, -Corentin Noël, Daeseok Youn, Damian Vicino, Dan Kegel, Daniel Drake, -Daniel Klamt, Danilo Spinella, Dardo D Kleiner, David Ing, David -Svensson Fors, Devarsh Thakkar, Dimitrios Katsaros, Edward Hervey, -Emilio Pozuelo Monfort, Enrique Ocaña González, Erlend Eriksen, Ezequiel -Garcia, Fabien Dessenne, Fabrizio Gennari, Florent Thiéry, Francisco -Velazquez, Freyr666, Garima Gaur, Gary Bisson, George Kiagiadakis, Georg -Lippitsch, Georg Ottinger, Geunsik Lim, Göran Jönsson, Guillaume -Desmottes, H1Gdev, Haihao Xiang, Haihua Hu, Harshad Khedkar, Havard -Graff, He Junyan, Hoonhee Lee, Hosang Lee, Hyunjun Ko, Ilya Smelykh, -Ingo Randolf, Iñigo Huguet, Jakub Adam, James Stevenson, Jan Alexander -Steffens, Jan Schmidt, Jerome Laheurte, Jimmy Ohn, Joakim Johansson, -Jochen Henneberg, Johan Bjäreholt, John-Mark Bell, John Bassett, John -Nikolaides, Jonathan Karlsson, Jonny Lamb, Jordan Petridis, Josep Torra, -Joshua M. Doe, Jos van Egmond, Juan Navarro, Julian Bouzas, Jun Xie, -Junyan He, Justin Kim, Kai Kang, Kim Tae Soo, Kirill Marinushkin, Kyrylo -Polezhaiev, Lars Petter Endresen, Linus Svensson, Louis-Francis -Ratté-Boulianne, Lucas Stach, Luis de Bethencourt, Luz Paz, Lyon Wang, -Maciej Wolny, Marc-André Lureau, Marc Leeman, Marco Trevisan (Treviño), -Marcos Kintschner, Marian Mihailescu, Marinus Schraal, Mark Nauwelaerts, -Marouen Ghodhbane, Martin Kelly, Matej Knopp, Mathieu Duponchelle, -Matteo Valdina, Matthew Waters, Matthias Fend, memeka, Michael Drake, -Michael Gruner, Michael Olbrich, Michael Tretter, Miguel Paris, Mike -Wey, Mikhail Fludkov, Naveen Cherukuri, Nicola Murino, Nicolas Dufresne, -Niels De Graef, Nirbheek Chauhan, Norbert Wesp, Ognyan Tonchev, Olivier -Crête, Omar Akkila, Pat DeSantis, Patricia Muscalu, Patrick Radizi, -Patrik Nilsson, Paul Kocialkowski, Per Forlin, Peter Körner, Peter -Seiderer, Petr Kulhavy, Philippe Normand, Philippe Renon, Philipp Zabel, -Pierre Labastie, Piotr Drąg, Roland Jon, Roman Sivriver, Roman Shpuntov, -Rosen Penev, Russel Winder, Sam Gigliotti, Santiago Carot-Nemesio, -Sean-Der, Sebastian Dröge, Seungha Yang, Shi Yan, Sjoerd Simons, Snir -Sheriber, Song Bing, Soon, Thean Siew, Sreerenj Balachandran, Stefan -Ringel, Stephane Cerveau, Stian Selnes, Suhas Nayak, Takeshi Sato, -Thiago Santos, Thibault Saunier, Thomas Bluemel, Tianhao Liu, -Tim-Philipp Müller, Tobias Ronge, Tomasz Andrzejak, Tomislav Tustonić, -U. Artie Eoff, Ulf Olsson, Varunkumar Allagadapa, Víctor Guzmán, Víctor -Manuel Jáquez Leal, Vincenzo Bono, Vineeth T M, Vivia Nikolaidou, Wang -Fei, wangzq, Whoopie, Wim Taymans, Wind Yuan, Wonchul Lee, Xabier -Rodriguez Calvar, Xavier Claessens, Haihao Xiang, Yacine Bandou, -Yeongjin Jeong, Yuji Kuwabara, Zeeshan Ali, +- FIXME … and many others who have contributed bug reports, translations, sent suggestions or helped testing. -Stable 1.16 branch +Stable 1.18 branch -After the 1.16.0 release there will be several 1.16.x bug-fix releases +After the 1.18.0 release there will be several 1.18.x bug-fix releases which will contain bug fixes which have been deemed suitable for a stable branch, but no new features or intrusive changes will be added to -a bug-fix release usually. The 1.16.x bug-fix releases will be made from -the git 1.16 branch, which is a stable branch. +a bug-fix release usually. The 1.18.x bug-fix releases will be made from +the git 1.18 branch, which will be a stable branch. -1.16.0 +1.18.0 -1.16.0 was released on 19 April 2019. +1.18.0 has not been released yet. Known Issues -- possibly breaking/incompatible changes to properties of wrapped - FFmpeg decoders and encoders (see above). - -- The way that GIO modules are named has changed due to upstream GLib - natively adding support for loading static GIO modules. This means - that any GStreamer application using gnutls for SSL/TLS on the - Android or iOS platforms (or any other setup using static libraries) - will fail to link looking for the g_io_module_gnutls_load_static() - function. The new function name is now - g_io_gnutls_load(gpointer data). See Android/iOS sections above for - further details. +- FIXME -Schedule for 1.18 +Schedule for 1.20 -Our next major feature release will be 1.18, and 1.17 will be the -unstable development version leading up to the stable 1.18 release. The -development of 1.17/1.18 will happen in the git master branch. +Our next major feature release will be 1.20, and 1.19 will be the +unstable development version leading up to the stable 1.20 release. The +development of 1.19/1.20 will happen in the git master branch. -The plan for the 1.18 development cycle is yet to be confirmed, but it -is possible that the next cycle will be a short one in which case -feature freeze would be perhaps around August 2019 with a new 1.18 -stable release in September. +The plan for the 1.20 development cycle is yet to be confirmed. -1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10, -1.8, 1.6, 1.4, 1.2 and 1.0 release series. +1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12, +1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series. ------------------------------------------------------------------------ _These release notes have been prepared by Tim-Philipp Müller with_ -_contributions from Sebastian Dröge, Guillaume Desmottes, Matthew -Waters, _ _Thibault Saunier, and Víctor Manuel Jáquez Leal._ +_contributions from … (FIXME)_ _License: CC BY-SA 4.0_ @@ -1,18 +1,15 @@ -This is GStreamer gst-rtsp-server 1.17.0.1. +This is GStreamer gst-rtsp-server 1.17.1. -The GStreamer team is thrilled to announce a new major feature release in the -stable 1.0 API series of your favourite cross-platform multimedia framework! +GStreamer 1.17 is the development branch leading up to the next major +stable version which will be 1.18. -As always, this release is again packed with new features, bug fixes and -other improvements. - -The 1.16 release series adds new features on top of the 1.14 series and is +The 1.17 development series adds new features on top of the 1.16 series and is part of the API and ABI-stable 1.x release series of the GStreamer multimedia framework. Full release notes will one day be found at: - https://gstreamer.freedesktop.org/releases/1.16/ + https://gstreamer.freedesktop.org/releases/1.18/ Binaries for Android, iOS, Mac OS X and Windows will usually be provided shortly after the release. @@ -60,7 +57,7 @@ You can find source releases of gstreamer in the download directory: https://gstreamer.freedesktop.org/src/gstreamer/ The git repository and details how to clone it can be found at -https://cgit.freedesktop.org/gstreamer/gstreamer/ +https://gitlab.freedesktop.org/gstreamer/ ==== Homepage ==== diff --git a/docs/gst_plugins_cache.json b/docs/gst_plugins_cache.json index b838208..e05b899 100644 --- a/docs/gst_plugins_cache.json +++ b/docs/gst_plugins_cache.json @@ -345,7 +345,7 @@ "construct": false, "construct-only": false, "controllable": false, - "default": "GStreamer/1.17.0.1", + "default": "GStreamer/1.17.1", "mutable": "null", "readable": true, "type": "gchararray", @@ -510,7 +510,7 @@ } } }, - "package": "GStreamer RTSP Server Library git", + "package": "GStreamer RTSP Server Library", "source": "gst-rtsp-server", "tracers": {}, "url": "Unknown package origin" diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap index 430c0b2..d09320b 100644 --- a/gst-rtsp-server.doap +++ b/gst-rtsp-server.doap @@ -16,7 +16,7 @@ RTSP server library based on GStreamer RTSP server library based on GStreamer </description> <category></category> - <bug-database rdf:resource="http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-rtsp-server" /> + <bug-database rdf:resource="https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/" /> <screenshots></screenshots> <mailing-list rdf:resource="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" /> <programming-language>C</programming-language> @@ -32,6 +32,16 @@ RTSP server library based on GStreamer <release> <Version> + <revision>1.17.1</revision> + <branch>master</branch> + <name></name> + <created>2020-06-19</created> + <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.17.1.tar.xz" /> + </Version> + </release> + + <release> + <Version> <revision>1.16.0</revision> <branch>master</branch> <name></name> diff --git a/meson.build b/meson.build index 3f45f0e..26a2342 100644 --- a/meson.build +++ b/meson.build @@ -1,5 +1,5 @@ project('gst-rtsp-server', 'c', - version : '1.17.0.1', + version : '1.17.1', meson_version : '>= 0.48', default_options : ['warning_level=1', 'buildtype=debugoptimized']) |