+=== 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>
-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
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
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_
projects / platform / upstream / gstreamer.git / commitdiff