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authorSebastian Dröge <sebastian@centricular.com>2017-01-12 16:14:46 +0200
committerSebastian Dröge <sebastian@centricular.com>2017-01-12 16:14:46 +0200
commitc860590a8cdccda4a73b29d3d2a57e2a60ee6bd3 (patch)
tree5a05e323edfd21587729f053c11ef3ad5656d1b8
parentfb7833245de53bd6e409f5faf228be7899ce933f (diff)
Release 1.11.1
-rw-r--r--ChangeLog242
-rw-r--r--NEWS1115
-rw-r--r--RELEASE46
-rw-r--r--configure.ac12
-rw-r--r--gst-rtsp-server.doap10
-rw-r--r--win32/common/libgstrtspserver.def2
6 files changed, 292 insertions, 1135 deletions
diff --git a/ChangeLog b/ChangeLog
index 9e51434..610dd60 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,9 +1,247 @@
+=== release 1.11.1 ===
+
+2017-01-12 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.11.1
+
+2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: corrected if-statement in _get_server_port()
+ This bug was accidentally introduced while fixing a segfault
+ in _get_server_port() function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776345
+
+2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/stream.c:
+ rtsp-stream: fixed segmenation fault in _get_server_port()
+ Calling function gst_rtsp_stream_get_server_port() results in
+ segmenation fault in the RTP/RTSP/TCP case.
+ Port that the server will use to receive RTCP makes only
+ sense in the UDP case, however the function should handle
+ the TCP case in a nicer way.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776345
+
+2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ dosc: Fix a little typo
+ https://bugzilla.gnome.org/show_bug.cgi?id=777037
+
+2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/meson.build:
+ meson: generate pkg-config -uninstalled pc files
+ Generating those files is useful for users building the GStreamer stack
+ using meson and having to link it to another project which is still
+ using the autotools.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776810
+
+2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ pkgconfig: fix -uninstalled pc file
+ pcfiledir was never defined so the paths were wrong.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776867
+
+2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-stream: Fixed TCP transport case
+ Make sure that the appsink element is actually added to
+ the bin before trying to link it with the elements in it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776343
+
+2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ * gst-rtsp.spec.in:
+ Remove generated .spec file
+ Likely extremely bitrotten, and we should not ship this anyway.
+
+2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From f980fd9 to 39ac2f5
+
+2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Fix pt map caps
+ Since decryption is handled within rtpbin, all outcoming stream
+ caps will be application/x-rtp (i.e. regular rtp)
+ Fixes RECORD with SRTP streams
+
+2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: Create media objects with the proper transport mode
+ The function called immediately afterwards (collect_streams()) will
+ need it to work properly
+
+2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
+
+2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: Don't create a pipeline for the media pipeline string
+ We're going to put a pipeline into a pipeline otherwise, which is not
+ exactly ideal.
+
+2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Fix race condition around finish_unprepare() if called multiple time
+ https://bugzilla.gnome.org/show_bug.cgi?id=755329
+
+2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Don't leave stale pointer after unref
+ Fix a warning on shutdown - don't keep a pointer to an
+ alread-unreffed object.
+
+2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitmodules:
+ common: use https protocol for common submodule
+ https://bugzilla.gnome.org/show_bug.cgi?id=775110
+
+2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: block the output of rtpbin instead of the source pipeline
+ 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
+ detection of the srtp rollover counter to add to the SDP.
+ Unfortunately, it was incomplete for live pipelines where the logic
+ blocks the source bin before creating the SDP and thus would never have
+ the necessary informaiton to create a correct SDP with srtp encryption.
+ Move the pad blocks to rtpbin's output pads instead so that the
+ necessary information can be created before we need the information for
+ the SDP.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770239
+
+2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: add IDLE timeout, before session exists
+ The RTSP server will not timeout an idle RTSP connection
+ (note this is different from doing timeout on a RTSP
+ session).
+ At least for Apache this is a problem when running RTSP over
+ HTTPS since it uses one of the threads (there is a rather
+ limited number) that are available for handling requests.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771830
+
+2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ .gitignore more
+
+2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set close-socket FALSE on UDP src:es
+ With this RTSP server can use the sockets independent on the udpsrc
+ state.
+ When the udp src is finalized it will unref socket and when g_socket
+ is finalized the socket will be closed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765673
+
+2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Move to new helper function to parse authentication responses
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ * examples/test-auth-digest.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * win32/common/libgstrtspserver.def:
+ rtsp-auth: Add support for Digest authentication
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * meson.build:
+ * tests/check/meson.build:
+ * win32/MANIFEST:
+ * win32/common/libgstrtspserver.def:
+ Enable building with MSVC
+ https://bugzilla.gnome.org/show_bug.cgi?id=774640
+
+2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * meson.build:
+ meson: gstreamer gst_check_dep does not exist on windows
+
+2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: update do_send_message to match type GstRTSPClientSendFunc
+ This type mismatch fails building with MSVC
+ https://bugzilla.gnome.org/show_bug.cgi?id=774640
+
+2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Fix indentation
+
+2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only signal "new-state" if the state has actually changed
+ https://bugzilla.gnome.org/show_bug.cgi?id=774173
+
+2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: emit signal in the beginning of each rtsp request
+ These signals let the application validate the requests, configure the
+ media/stream in a certain way and also generate error status code in
+ case of error or bad request.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758062
+
+2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: update version
+
+=== release 1.11.0 ===
+
+2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
=== release 1.10.0 ===
-2016-11-01 Sebastian Dröge <slomo@coaxion.net>
+2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.10.0
+ * gst-rtsp-server.doap:
+ Release 1.10.0
2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
diff --git a/NEWS b/NEWS
index 547de7f..a940f7b 100644
--- a/NEWS
+++ b/NEWS
@@ -1,1114 +1 @@
-# GStreamer 1.10 Release Notes
-
-**GStreamer 1.10.0 was released on 1st November 2016.**
-
-The GStreamer team is proud to announce a new major feature release in the
-stable 1.x API series of your favourite cross-platform multimedia framework!
-
-As always, this release is again packed with new features, bug fixes and other
-improvements.
-
-See [https://gstreamer.freedesktop.org/releases/1.10/][latest] for the latest
-version of this document.
-
-*Last updated: Tuesday 1 Nov 2016, 15:00 UTC [(log)][gitlog]*
-
-[latest]: https://gstreamer.freedesktop.org/releases/1.10/
-[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.10/release-notes-1.10.md
-
-## Introduction
-
-The GStreamer team is proud to announce a new major feature release in the
-stable 1.x API series of your favourite cross-platform multimedia framework!
-
-As always, this release is again packed with new features, bug fixes and other
-improvements.
-
-## Highlights
-
-- Several convenience APIs have been added to make developers' lives easier
-- A new `GstStream` API provides applications a more meaningful view of the
- structure of streams, simplifying the process of dealing with media in
- complex container formats
-- Experimental `decodebin3` and `playbin3` elements which bring a number of
- improvements which were hard to implement within `decodebin` and `playbin`
-- A new `parsebin` element to automatically unpack and parse a stream, stopping
- just short of decoding
-- Experimental new `meson`-based build system, bringing faster build and much
- better Windows support (including for building with Visual Studio)
-- A new `gst-docs` module has been created, and we are in the process of moving
- our documentation to a markdown-based format for easier maintenance and
- updates
-- A new `gst-examples` module has been create, which contains example
- GStreamer applications and is expected to grow with many more examples in
- the future
-- Various OpenGL and OpenGL|ES-related fixes and improvements for greater
- efficiency on desktop and mobile platforms, and Vulkan support on Wayland was
- also added
-- Extensive improvements to the VAAPI plugins for improved robustness and
- efficiency
-- Lots of fixes and improvements across the board, spanning RTP/RTSP, V4L2,
- Bluetooth, audio conversion, echo cancellation, and more!
-
-## Major new features and changes
-
-### Noteworthy new API, features and other changes
-
-#### Core API additions
-
-##### Receive property change notifications via bus messages
-
-New API was added to receive element property change notifications via
-bus messages. So far, applications had to connect a callback to an element's
-`notify::property-name` signal via the GObject API, which was inconvenient for
-at least two reasons: one had to implement a signal callback function, and that
-callback function would usually be called from one of the streaming threads, so
-one had to marshal (send) any information gathered or pending requests to the
-main application thread which was tedious and error-prone.
-
-Enter [`gst_element_add_property_notify_watch()`][notify-watch] and
-[`gst_element_add_property_deep_notify_watch()`][deep-notify-watch] which will
-watch for changes of a property on the specified element, either only for this
-element or recursively for a whole bin or pipeline. Whenever such a
-property change happens, a `GST_MESSAGE_PROPERTY_NOTIFY` message will be posted
-on the pipeline bus with details of the element, the property and the new
-property value, all of which can be retrieved later from the message in the
-application via [`gst_message_parse_property_notify()`][parse-notify]. Unlike
-the GstBus watch functions, this API does not rely on a running GLib main loop.
-
-The above can be used to be notified asynchronously of caps changes in the
-pipeline, or volume changes on an audio sink element, for example.
-
-[notify-watch]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-add-property-notify-watch
-[deep-notify-watch]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-add-property-deep-notify-watch
-[parse-notify]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-parse-property-notify
-
-##### GstBin "deep" element-added and element-removed signals
-
-GstBin has gained `"deep-element-added"` and `"deep-element-removed"` signals
-which makes it easier for applications and higher-level plugins to track when
-elements are added or removed from a complex pipeline with multiple sub-bins.
-
-`playbin` makes use of this to implement the new `"element-setup"` signal which
-can be used to configure elements as they are added to `playbin`, just like the
-existing `"source-setup"` signal which can be used to configure the source
-element created.
-
-##### Error messages can contain additional structured details
-
-It is often useful to provide additional, structured information in error,
-warning or info messages for applications (or higher-level elements) to make
-intelligent decisions based on them. To allow this, error, warning and info
-messages now have API for adding arbitrary additional information to them
-using a `GstStructure`:
-[`GST_ELEMENT_ERROR_WITH_DETAILS`][element-error-with-details] and
-corresponding API for the other message types.
-
-This is now used e.g. by the new [`GST_ELEMENT_FLOW_ERROR`][element-flow-error]
-API to include the actual flow error in the error message, and the
-[souphttpsrc element][souphttpsrc-detailed-errors] to provide the HTTP
-status code, and the URL (if any) to which a redirection has happened.
-
-[element-error-with-details]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GST-ELEMENT-ERROR-WITH-DETAILS:CAPS
-[element-flow-error]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GST-ELEMENT-FLOW-ERROR:CAPS
-[souphttpsrc-detailed-errors]: https://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/ext/soup/gstsouphttpsrc.c?id=60d30db912a1aedd743e66b9dcd2e21d71fbb24f#n1318
-
-##### Redirect messages have official API now
-
-Sometimes, elements need to redirect the current stream URL and tell the
-application to proceed with this new URL, possibly using a different
-protocol too (thus changing the pipeline configuration). Until now, this was
-informally implemented using `ELEMENT` messages on the bus.
-
-Now this has been formalized in the form of a new `GST_MESSAGE_REDIRECT` message.
-A new redirect message can be created using [`gst_message_new_redirect()`][new-redirect].
-If needed, multiple redirect locations can be specified by calling
-[`gst_message_add_redirect_entry()`][add-redirect] to add further redirect
-entries, all with metadata, so the application can decide which is
-most suitable (e.g. depending on the bitrate tags).
-
-[new-redirect]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-new-redirect
-[add-redirect]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-add-redirect-entry
-
-##### New pad linking convenience functions that automatically create ghost pads
-
-New pad linking convenience functions were added:
-[`gst_pad_link_maybe_ghosting()`][pad-maybe-ghost] and
-[`gst_pad_link_maybe_ghosting_full()`][pad-maybe-ghost-full] which were
-previously internal to GStreamer have now been exposed for general use.
-
-The existing pad link functions will refuse to link pads or elements at
-different levels in the pipeline hierarchy, requiring the developer to
-create ghost pads where necessary. These new utility functions will
-automatically create ghostpads as needed when linking pads at different
-levels of the hierarchy (e.g. from an element inside a bin to one that's at
-the same level in the hierarchy as the bin, or in another bin).
-
-[pad-maybe-ghost]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-link-maybe-ghosting
-[pad-maybe-ghost-full]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-link-maybe-ghosting-full
-
-##### Miscellaneous
-
-Pad probes: IDLE and BLOCK probes now work slightly differently in pull mode,
-so that push and pull mode have opposite scenarios for idle and blocking probes.
-In push mode, it will block with some data type and IDLE won't have any data.
-In pull mode, it will block _before_ getting a buffer and will be IDLE once some
-data has been obtained. ([commit][commit-pad-probes], [bug][bug-pad-probes])
-
-[commit-pad-probes]: https://cgit.freedesktop.org/gstreamer/gstreamer/commit/gst/gstpad.c?id=368ee8a336d0c868d81fdace54b24431a8b48cbf
-[bug-pad-probes]: https://bugzilla.gnome.org/show_bug.cgi?id=761211
-
-[`gst_parse_launch_full()`][parse-launch-full] can now be made to return a
-`GstBin` instead of a top-level pipeline by passing the new
-`GST_PARSE_FLAG_PLACE_IN_BIN` flag.
-
-[parse-launch-full]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstParse.html#gst-parse-launch-full
-
-The default GStreamer debug log handler can now be removed before
-calling `gst_init()`, so that it will never get installed and won't be active
-during initialization.
-
-A new [`STREAM_GROUP_DONE` event][stream-group-done-event] was added. In some
-ways it works similar to the `EOS` event in that it can be used to unblock
-downstream elements which may be waiting for further data, such as for example
-`input-selector`. Unlike `EOS`, further data flow may happen after the
-`STREAM_GROUP_DONE` event though (and without the need to flush the pipeline).
-This is used to unblock input-selector when switching between streams in
-adaptive streaming scenarios (e.g. HLS).
-
-[stream-group-done-event]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-new-stream-group-done
-
-The `gst-launch-1.0` command line tool will now print unescaped caps in verbose
-mode (enabled by the -v switch).
-
-[`gst_element_call_async()`][call-async] has been added as convenience API for
-plugin developers. It is useful for one-shot operations that need to be done
-from a thread other than the current streaming thread. It is backed by a
-thread-pool that is shared by all elements.
-
-[call-async]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-call-async
-
-Various race conditions have been fixed around the `GstPoll` API used by e.g.
-`GstBus` and `GstBufferPool`. Some of these manifested themselves primarily
-on Windows.
-
-`GstAdapter` can now keep track of discontinuities signalled via the `DISCONT`
-buffer flag, and has gained [new API][new-adapter-api] to track PTS, DTS and
-offset at the last discont. This is useful for plugins implementing advanced
-trick mode scenarios.
-
-[new-adapter-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstAdapter.html#gst-adapter-pts-at-discont
-
-`GstTestClock` gained a new [`"clock-type"` property][clock-type-prop].
-
-[clock-type-prop]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstTestClock.html#GstTestClock--clock-type
-
-#### GstStream API for stream announcement and stream selection
-
-New stream listing and stream selection API: new API has been added to
-provide high-level abstractions for streams ([`GstStream`][stream-api])
-and collections of streams ([`GstStreamCollections`][stream-collection-api]).
-
-##### Stream listing
-
-A [`GstStream`][stream-api] contains all the information pertinent to a stream,
-such as stream id, caps, tags, flags and stream type(s); it can represent a
-single elementary stream (e.g. audio, video, subtitles, etc.) or a container
-stream. This will depend on the context. In a decodebin3/playbin3 one
-it will typically be elementary streams that can be selected and unselected.
-
-A [`GstStreamCollection`][stream-collection-api] represents a group of streams
-and is used to announce or publish all available streams. A GstStreamCollection
-is immutable - once created it won't change. If the available streams change,
-e.g. because a new stream appeared or some streams disappeared, a new stream
-collection will be published. This new stream collection may contain streams
-from the previous collection if those streams persist, or completely new ones.
-Stream collections do not yet list all theoretically available streams,
-e.g. other available DVD angles or alternative resolutions/bitrate of the same
-stream in case of adaptive streaming.
-
-New events and messages have been added to notify or update other elements and
-the application about which streams are currently available and/or selected.
-This way, we can easily and seamlessly let the application know whenever the
-available streams change, as happens frequently with digital television streams
-for example. The new system is also more flexible. For example, it is now also
-possible for the application to select multiple streams of the same type
-(e.g. in a transcoding/transmuxing scenario).
-
-A [`STREAM_COLLECTION` message][stream-collection-msg] is posted on the bus
-to inform the parent bin (e.g. `playbin3`, `decodebin3`) and/or the application
-about what streams are available, so you no longer have to hunt for this
-information at different places. The available information includes number of
-streams of each type, caps, tags etc. Bins and/or the application can intercept
-the message synchronously to select and deselect streams before any data is
-produced - for the case where elements such as the demuxers support the new
-stream API, not necessarily in the parsebin compatibility fallback case.
-
-Similarly, there is also a [`STREAM_COLLECTION` event][stream-collection-event]
-to inform downstream elements of the available streams. This event can be used
-by elements to aggregate streams from multiple inputs into one single collection.
-
-The `STREAM_START` event was extended so that it can also contain a GstStream
-object with all information about the current stream, see
-[`gst_event_set_stream()`][event-set-stream] and
-[`gst_event_parse_stream()`][event-parse-stream].
-[`gst_pad_get_stream()`][pad-get-stream] is a new utility function that can be
-used to look up the GstStream from the `STREAM_START` sticky event on a pad.
-
-[stream-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstStream.html
-[stream-collection-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstStreamCollection.html
-[stream-collection-msg]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-new-stream-collection
-[stream-collection-event]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-new-stream-collection
-[event-set-stream]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-set-stream
-[event-parse-stream]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-parse-stream
-[pad-get-stream]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-get-stream
-
-##### Stream selection
-
-Once the available streams have been published, streams can be selected via
-their stream ID using the new `SELECT_STREAMS` event, which can be created
-with [`gst_event_new_select_streams()`][event-select-streams]. The new API
-supports selecting multiple streams per stream type. In the future, we may also
-implement explicit deselection of streams that will never be used, so
-elements can skip these and never expose them or output data for them in the
-first place.
-
-The application is then notified of the currently selected streams via the
-new `STREAMS_SELECTED` message on the pipeline bus, containing both the current
-stream collection as well as the selected streams. This might be posted in
-response to the application sending a `SELECT_STREAMS` event or when
-`decodebin3` or `playbin3` decide on the streams to be initially selected without
-application input.
-
-[event-select-streams]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-new-select-streams
-
-##### Further reading
-
-See further below for some notes on the new elements supporting this new
-stream API, namely: `decodebin3`, `playbin3` and `parsebin`.
-
-More information about the new API and the new elements can also be found here:
-
-- GStreamer [stream selection design docs][streams-design]
-- Edward Hervey's talk ["The new streams API: Design and usage"][streams-talk] ([slides][streams-slides])
-- Edward Hervey's talk ["Decodebin3: Dealing with modern playback use cases"][db3-talk] ([slides][db3-slides])
-
-[streams-design]: https://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-stream-selection.txt
-[streams-talk]: https://gstconf.ubicast.tv/videos/the-new-gststream-api-design-and-usage/
-[streams-slides]: https://gstreamer.freedesktop.org/data/events/gstreamer-conference/2016/Edward%20Hervey%20-%20The%20New%20Streams%20API%20Design%20and%20Usage.pdf
-[db3-talk]: https://gstconf.ubicast.tv/videos/decodebin3-or-dealing-with-modern-playback-use-cases/
-[db3-slides]: https://gstreamer.freedesktop.org/data/events/gstreamer-conference/2015/Edward%20Hervey%20-%20decodebin3.pdf
-
-#### Audio conversion and resampling API
-
-The audio conversion library received a completely new and rewritten audio
-resampler, complementing the audio conversion routines moved into the audio
-library in the [previous release][release-notes-1.8]. Integrating the resampler
-with the other audio conversion library allows us to implement generic
-conversion much more efficiently, as format conversion and resampling can now
-be done in the same processing loop instead of having to do it in separate
-steps (our element implementations do not make use of this yet though).
-
-The new audio resampler library is a combination of some of the best features
-of other samplers such as ffmpeg, speex and SRC. It natively supports S16, S32,
-F32 and F64 formats and uses optimized x86 and neon assembly for most of its
-processing. It also has support for dynamically changing sample rates by incrementally
-updating the filter tables using linear or cubic interpolation. According to
-some benchmarks, it's one of the fastest and most accurate resamplers around.
-
-The `audioresample` plugin has been ported to the new audio library functions
-to make use of the new resampler.
-
-[release-notes-1.8]: https://gstreamer.freedesktop.org/releases/1.8/
-
-#### Support for SMPTE timecodes
-
-Support for SMPTE timecodes was added to the GStreamer video library. This
-comes with an abstraction for timecodes, [`GstVideoTimeCode`][video-timecode]
-and a [`GstMeta`][video-timecode-meta] that can be placed on video buffers for
-carrying the timecode information for each frame. Additionally there is
-various API for making handling of timecodes easy and to do various
-calculations with them.
-
-A new plugin called [`timecode`][timecode-plugin] was added, that contains an
-element called `timecodestamper` for putting the timecode meta on video frames
-based on counting the frames and another element called `timecodewait` that
-drops all video (and audio) until a specific timecode is reached.
-
-Additionally support was added to the Decklink plugin for including the
-timecode information when sending video out or capturing it via SDI, the
-`qtmux` element is able to write timecode information into the MOV container,
-and the `timeoverlay` element can overlay timecodes on top of the video.
-
-More information can be found in the [talk about timecodes][timecode-talk] at
-the GStreamer Conference 2016.
-
-[video-timecode]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideo.html#GstVideoTimeCode
-[video-timecode-meta]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideometa.html#gst-buffer-add-video-time-code-meta
-[timecode-plugin]: https://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/gst/timecode
-[timecode-talk]: https://gstconf.ubicast.tv/videos/smpte-timecodes-in-gstreamer/
-
-#### GStreamer OpenMAX IL plugin
-
-The last gst-omx release, 1.2.0, was in July 2014. It was about time to get
-a new one out with all the improvements that have happened in the meantime.
-From now on, we will try to release gst-omx together with all other modules.
-
-This release features a lot of bugfixes, improved support for the Raspberry Pi
-and in general improved support for zerocopy rendering via EGL and a few minor
-new features.
-
-At this point, gst-omx is known to work best on the Raspberry Pi platform but
-it is also known to work on various other platforms. Unfortunately, we are
-not including configurations for any other platforms, so if you happen to use
-gst-omx: please send us patches with your configuration and code changes!
-
-### New Elements
-
-#### decodebin3, playbin3, parsebin (experimental)
-
-This release features new decoding and playback elements as experimental
-technology previews: `decodebin3` and `playbin3` will soon supersede the
-existing `decodebin` and `playbin` elements. We skipped the number 2 because
-it was already used back in the 0.10 days, which might cause confusion.
-Experimental technology preview means that everything should work fine already,
-but we can't guarantee there won't be minor behavioural changes in the
-next cycle. In any case, please test and report any problems back.
-
-Before we go into detail about what these new elements improve, let's look at
-the new [`parsebin`][parsebin] element. It works similarly to `decodebin` and
-`decodebin3`, only that it stops one step short and does not plug any actual
-decoder elements. It will only plug parsers, tag readers, demuxers and
-depayloaders. Also note that parsebin does not contain any queueing element.
-
-[`decodebin3`'s][decodebin3] internal architecture is slightly different from
-the existing `decodebin` element and fixes many long-standing issues with our
-decoding engine. For one, data is now fed into the internal `multiqueue` element
-*after* it has been parsed and timestamped, which means that the `multiqueue`
-element now has more knowledge and is able to calculate the interleaving of the
-various streams, thus minimizing memory requirements and doing away with magic
-values for buffering limits that were conceived when videos were 240p or 360p.
-Anyone who has tried to play back 4k video streams with decodebin2
-will have noticed the limitations of that approach. The improved timestamp
-tracking also enables `multiqueue` to keep streams of the same type (audio,
-video) aligned better, making sure switching between streams of the same type
-is very fast.
-
-Another major improvement in `decodebin3` is that it will no longer decode
-streams that are not being used. With the old `decodebin` and `playbin`, when
-there were 8 audio streams we would always decode all 8 streams even
-if 7 were not actually used. This caused a lot of CPU overhead, which was
-particularly problematic on embedded devices. When switching between streams
-`decodebin3` will try hard to re-use existing decoders. This is useful when
-switching between multiple streams of the same type if they are encoded in the
-same format.
-
-Re-using decoders is also useful when the available streams change on the fly,
-as might happen with radio streams (chained Oggs), digital television
-broadcasts, when adaptive streaming streams change bitrate, or when switching
-gaplessly to the next title. In order to guarantee a seamless transition, the
-old `decodebin2` would plug a second decoder for the new stream while finishing
-up the old stream. With `decodebin3`, this is no longer needed - at least not
-when the new and old format are the same. This will be particularly useful
-on embedded systems where it is often not possible to run multiple decoders
-at the same time, or when tearing down and setting up decoders is fairly
-expensive.
-
-`decodebin3` also allows for multiple input streams, not just a single one.
-This will be useful, in the future, for gapless playback, or for feeding
-multiple external subtitle streams to decodebin/playbin.
-
-`playbin3` uses `decodebin3` internally, and will supercede `playbin`.
-It was decided that it would be too risky to make the old `playbin` use the
-new `decodebin3` in a backwards-compatible way. The new architecture
-makes it awkward, if not impossible, to maintain perfect backwards compatibility
-in some aspects, hence `playbin3` was born, and developers can migrate to the
-new element and new API at their own pace.
-
-All of these new elements make use of the new `GstStream` API for listing and
-selecting streams, as described above. `parsebin` provides backwards
-compatibility for demuxers and parsers which do not advertise their streams
-using the new API yet (which is most).
-
-The new elements are not entirely feature-complete yet: `playbin3` does not
-support so-called decodersinks yet where the data is not decoded inside
-GStreamer but passed directly for decoding to the sink. `decodebin3` is missing
-the various `autoplug-*` signals to influence which decoders get autoplugged
-in which order. We're looking to add back this functionality, but it will probably
-be in a different way, with a single unified signal and using GstStream perhaps.
-
-For more information on these new elements, check out Edward Hervey's talk
-[*decodebin3 - dealing with modern playback use cases*][db3-talk]
-
-[parsebin]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-parsebin.html
-[decodebin3]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-decodebin3.html
-[db3-talk]: https://gstconf.ubicast.tv/videos/decodebin3-or-dealing-with-modern-playback-use-cases/
-
-#### LV2 ported from 0.10 and switched from slv2 to lilv2
-
-The LV2 wrapper plugin has been ported to 1.0 and moved from using the
-deprecated slv2 library to its replacement liblv2. We support sources and
-filter elements. lv2 is short for *Linux Audio Developer's Simple Plugin API
-(LADSPA) version 2* and is an open standard for audio plugins which includes
-support for audio synthesis (generation), digital signal processing of digital
-audio, and MIDI. The new lv2 plugin supersedes the existing LADSPA plugin.
-
-#### WebRTC DSP Plugin for echo-cancellation, gain control and noise suppression
-
-A set of new elements ([webrtcdsp][webrtcdsp], [webrtcechoprobe][webrtcechoprobe])
-based on the WebRTC DSP software stack can now be used to improve your audio
-voice communication pipelines. They support echo cancellation, gain control,
-noise suppression and more. For more details you may read
-[Nicolas' blog post][webrtc-blog-post].
-
-[webrtcdsp]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-webrtcdsp.html
-[webrtcechoprobe]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-webrtcechoprobe.html
-[webrtc-blog-post]: https://ndufresne.ca/2016/06/gstreamer-echo-canceller/
-
-#### Fraunhofer FDK AAC encoder and decoder
-
-New encoder and decoder elements wrapping the Fraunhofer FDK AAC library have
-been added (`fdkaacdec`, `fdkaacdec`). The Fraunhofer FDK AAC encoder is
-generally considered to be a very high-quality AAC encoder, but unfortunately
-it comes under a non-free license with the option to obtain a paid, commercial
-license.
-
-### Noteworthy element features and additions
-
-#### Major RTP and RTSP improvements
-
-- The RTSP server and source element, as well as the RTP jitterbuffer now support
- remote clock synchronization according to [RFC7273][https://tools.ietf.org/html/rfc7273].
-- Support for application and profile specific RTCP packets was added.
-- The H265/HEVC payloader/depayloader is again in sync with the final RFC.
-- Seeking stability of the RTSP source and server was improved a lot and
- runs stably now, even when doing scrub-seeking.
-- The RTSP server received various major bugfixes, including for regressions that
- caused the IP/port address pool to not be considered, or NAT hole punching
- to not work anymore. [Bugzilla #766612][https://bugzilla.gnome.org/show_bug.cgi?id=766612]
-- Various other bugfixes that improve the stability of RTP and RTSP, including
- many new unit / integration tests.
-
-#### Improvements to splitmuxsrc and splitmuxsink
-
-- The splitmux element received reliability and error handling improvements,
- removing at least one deadlock case. `splitmuxsrc` now stops cleanly at the end
- of the segment when handling seeks with a stop time. We fixed a bug with large
- amounts of downstream buffering causing incorrect out-of-sequence playback.
-
-- `splitmuxsrc` now has a `"format-location"` signal to directly specify the list
- of files to play from.
-
-- `splitmuxsink` can now optionally send force-keyunit events to upstream
- elements to allow splitting files more accurately instead of having to wait
- for upstream to provide a new keyframe by itself.
-
-#### OpenGL/GLES improvements
-
-##### iOS and macOS (OS/X)
-
-- We now create OpenGL|ES 3.x contexts on iOS by default with a fallback to
- OpenGL|ES 2.x if that fails.
-- Various zerocopy decoding fixes and enhancements with the
- encoding/decoding/capturing elements.
-- libdispatch is now used on all Apple platforms instead of GMainLoop, removing
- the expensive poll()/pthread_*() overhead.
-
-##### New API
-
-- `GstGLFramebuffer` - for wrapping OpenGL frame buffer objects. It provides
- facilities for attaching `GstGLMemory` objects to the necessary attachment
- points, binding and unbinding and running a user-supplied function with the
- framebuffer bound.
-- `GstGLRenderbuffer` (a `GstGLBaseMemory` subclass) - for wrapping OpenGL
- render buffer objects that are typically used for depth/stencil buffers or
- for color buffers where we don't care about the output.
-- `GstGLMemoryEGL` (a `GstGLMemory` subclass) - for combining `EGLImage`s with a GL
- texture that replaces `GstEGLImageMemory` bringing the improvements made to the
- other `GstGLMemory` implementations. This fixes a performance regression in
- zerocopy decoding on the Raspberry Pi when used with an updated gst-omx.
-
-##### Miscellaneous improvements
-
-- `gltestsrc` is now usable on devices/platforms with OpenGL 3.x and OpenGL|ES
- and has completed or gained support for new patterns in line with the
- existing ones in `videotestsrc`.
-- `gldeinterlace` is now available on devices/platforms with OpenGL|ES
- implementations.
-- The dispmanx backend (used on the Raspberry Pi) now supports the
- `gst_video_overlay_set_window_handle()` and
- `gst_video_overlay_set_render_rectangle()` functions.
-- The `gltransformation` element now correctly transforms mouse coordinates (in
- window space) to stream coordinates for both perspective and orthographic
- projections.
-- The `gltransformation` element now detects if the
- `GstVideoAffineTransformationMeta` is supported downstream and will efficiently
- pass its transformation downstream. This is a performance improvement as it
- results in less processing being required.
-- The wayland implementation now uses the multi-threaded safe event-loop API
- allowing correct usage in applications that call wayland functions from
- multiple threads.
-- Support for native 90 degree rotations and horizontal/vertical flips
- in `glimagesink`.
-
-#### Vulkan
-
-- The Vulkan elements now work under Wayland and have received numerous
- bugfixes.
-
-#### QML elements
-
-- `qmlglsink` video sink now works on more platforms, notably, Windows, Wayland,
- and Qt's eglfs (for embedded devices with an OpenGL implementation) including
- the Raspberry Pi.
-- New element `qmlglsrc` to record a QML scene into a GStreamer pipeline.
-
-#### KMS video sink
-
-- New element `kmssink` to render video using Direct Rendering Manager
- (DRM) and Kernel Mode Setting (KMS) subsystems in the Linux
- kernel. It is oriented to be used mostly in embedded systems.
-
-#### Wayland video sink
-
-- `waylandsink` now supports the wl_viewporter extension allowing
- video scaling and cropping to be delegated to the Wayland
- compositor. This extension is also been made optional, so that it can
- also work on current compositors that don't support it. It also now has
- support for the video meta, allowing zero-copy operations in more
- cases.
-
-#### DVB improvements
-
-- `dvbsrc` now has better delivery-system autodetection and several
- new parameter sanity-checks to improve its resilience to configuration
- omissions and errors. Superfluous polling continues to be trimmed down,
- and the debugging output has been made more consistent and precise.
- Additionally, the channel-configuration parser now supports the new dvbv5
- format, enabling `dvbbasebin` to automatically playback content transmitted
- on delivery systems that previously required manual description, like ISDB-T.
-
-#### DASH, HLS and adaptivedemux
-
-- HLS now has support for Alternate Rendition audio and video tracks. Full
- support for Alternate Rendition subtitle tracks will be in an upcoming release.
-- DASH received support for keyframe-only trick modes if the
- `GST_SEEK_FLAG_TRICKMODE_KEY_UNITS` flag is given when seeking. It will
- only download keyframes then, which should help with high-speed playback.
- Changes to skip over multiple frames based on bandwidth and other metrics
- will be added in the near future.
-- Lots of reliability fixes around seek handling and bitrate switching.
-
-#### Bluetooth improvements
-
-- The `avdtpsrc` element now supports metadata such as track title, artist
- name, and more, which devices can send via AVRCP. These are published as
- tags on the pipeline.
-- The `a2dpsink` element received some love and was cleaned up so that it
- actually works after the initial GStreamer 1.0 port.
-
-#### GStreamer VAAPI
-
-- All the decoders have been split, one plugin feature per codec. So
- far, the available ones, depending on the driver, are:
- `vaapimpeg2dec`, `vaapih264dec`, `vaapih265dec`, `vaapivc1dec`, `vaapivp8dec`,
- `vaapivp9dec` and `vaapijpegdec` (which already was split).
-- Improvements when mapping VA surfaces into memory. It now differentiates
- between negotiation caps and allocations caps, since the allocation
- memory for surfaces may be bigger than one that is going to be
- mapped.
-- `vaapih265enc` now supports constant bitrate mode (CBR).
-- Since several VA drivers are unmaintained, we decide to keep a whitelist
- with the va drivers we actually test, which is mostly the i915 and to a lesser
- degree gallium from the mesa project. Exporting the environment variable
- `GST_VAAPI_ALL_DRIVERS` disables the whitelist.
-- Plugin features are registered at run-time, according to their support by
- the loaded VA driver. So only the decoders and encoder supported by the
- system are registered. Since the driver can change, some dependencies are
- tracked to invalidate the GStreamer registry and reload the plugin.
-- `dmabuf` importation from upstream has been improved, gaining performance.
-- `vaapipostproc` now can negotiate buffer transformations via caps.
-- Decoders now can do I-frame only reverse playback. This decodes I-frames
- only because the surface pool is smaller than the required by the GOP to show all the
- frames.
-- The upload of frames onto native GL textures has been optimized too, keeping
- a cache of the internal structures for the offered textures by the sink.
-
-#### V4L2 changes
-
-- More pixels formats are now supported
-- Decoder is now using `G_SELECTION` instead of the deprecated `G_CROP`
-- Decoder now uses the `STOP` command to handle EOS
-- Transform element can now scale the pixel aspect ratio
-- Colorimetry support has been improved even more
-- We now support the `OUTPUT_OVERLAY` type of video node in v4l2sink
-
-#### Miscellaneous
-
-- `multiqueue`'s input pads gained a new `"group-id"` property which
- can be used to group input streams. Typically one will assign
- different id numbers to audio, video and subtitle streams for
- example. This way `multiqueue` can make sure streams of the same
- type advance in lockstep if some of the streams are unlinked and the
- `"sync-by-running-time"` property is set. This is used in
- decodebin3/playbin3 to implement almost-instantaneous stream
- switching. The grouping is required because different downstream
- paths (audio, video, etc.) may have different buffering/latency
- etc. so might be consuming data from multiqueue with a slightly
- different phase, and if we track different stream groups separately
- we minimize stream switching delays and buffering inside the
- `multiqueue`.
-- `alsasrc` now supports ALSA drivers without a position for each
- channel, this is common in some professional or industrial hardware.
-- `libvpx` based decoders (`vp8dec` and `vp9dec`) now create multiple threads on
- computers with multiple CPUs automatically.
-- `rfbsrc` - used for capturing from a VNC server - has seen a lot of
- debugging. It now supports the latest version of the RFB
- protocol and uses GIO everywhere.
-- `tsdemux` can now read ATSC E-AC-3 streams.
-- New `GstVideoDirection` video orientation interface for rotating, flipping
- and mirroring video in 90° steps. It is implemented by the `videoflip` and
- `glvideoflip` elements currently.
-- It is now possible to give `appsrc` a duration in time, and there is now a
- non-blocking try-pull API for `appsink` that returns NULL if nothing is
- available right now.
-- `x264enc` has support now for chroma-site and colorimetry settings
-- A new JPEG2000 parser element was added, and the JPEG2000 caps were cleaned
- up and gained more information needed in combination with RTP and various
- container formats.
-- Reverse playback support for `videorate` and `deinterlace` was implemented
-- Various improvements everywhere for reverse playback and `KEY_UNITS` trick mode
-- New cleaned up `rawaudioparse` and `rawvideoparse` elements that replace the
- old `audioparse` and `videoparse` elements. There are compatibility element
- factories registered with the old names to allow existing code to continue
- to work.
-- The Decklink plugin gained support for 10 bit video SMPTE timecodes, and
- generally got many bugfixes for various issues.
-- New API in `GstPlayer` for setting the multiview mode for stereoscopic
- video, setting an HTTP/RTSP user agent and a time offset between audio and
- video. In addition to that, there were various bugfixes and the new
- gst-examples module contains Android, iOS, GTK+ and Qt example applications.
-- `GstBin` has new API for suppressing various `GstElement` or `GstObject`
- flags that would otherwise be affected by added/removed child elements. This
- new API allows `GstBin` subclasses to handle for themselves if they
- should be considered a sink or source element, for example.
-- The `subparse` element can handle WebVTT streams now.
-- A new `sdpsrc` element was added that can read an SDP from a file, or get it
- as a string as property and then sets up an RTP pipeline accordingly.
-
-### Plugin moves
-
-No plugins were moved this cycle. We'll make up for it next cycle, promise!
-
-### Rewritten memory leak tracer
-
-GStreamer has had basic functionality to trace allocation and freeing of
-both mini-objects (buffers, events, caps, etc.) and objects in the form of the
-internal `GstAllocTrace` tracing system. This API was never exposed in the
-1.x API series though. When requested, this would dump a list of objects and
-mini-objects at exit time which had still not been freed at that point,
-enabled with an environment variable. This subsystem has now been removed
-in favour of a new implementation based on the recently-added tracing framework.
-
-Tracing hooks have been added to trace the creation and destruction of
-GstObjects and mini-objects, and a new tracer plugin has been written using
-those new hooks to track which objects are still live and which are not. If
-GStreamer has been compiled against the libunwind library, the new leaks tracer
-will remember where objects were allocated from as well. By default the leaks
-tracer will simply output a warning if leaks have been detected on `gst_deinit()`.
-
-If the `GST_LEAKS_TRACER_SIG` environment variable is set, the leaks tracer
-will also handle the following UNIX signals:
-
- - `SIGUSR1`: log alive objects
- - `SIGUSR2`: create a checkpoint and print a list of objects created and
- destroyed since the previous checkpoint.
-
-Unfortunately this will not work on Windows due to no signals, however.
-
-If the `GST_LEAKS_TRACER_STACK_TRACE` environment variable is set, the leaks
-tracer will also log the creation stack trace of leaked objects. This may
-significantly increase memory consumption however.
-
-New `MAY_BE_LEAKED` flags have been added to GstObject and GstMiniObject, so
-that objects and mini-objects that are likely to stay around forever can be
-flagged and blacklisted from the leak output.
-
-To give the new leak tracer a spin, simply call any GStreamer application such
-as `gst-launch-1.0` or `gst-play-1.0` like this:
-
- GST_TRACERS=leaks gst-launch-1.0 videotestsrc num-buffers=10 ! fakesink
-
-If there are any leaks, a warning will be raised at the end.
-
-It is also possible to trace only certain types of objects or mini-objects:
-
- GST_TRACERS="leaks(GstEvent,GstMessage)" gst-launch-1.0 videotestsrc num-buffers=10 ! fakesink
-
-This dedicated leaks tracer is much much faster than valgrind since all code is
-executed natively instead of being instrumented. This makes it very suitable
-for use on slow machines or embedded devices. It is however limited to certain
-types of leaks and won't catch memory leaks when the allocation has been made
-via plain old `malloc()` or `g_malloc()` or other means. It will also not trace
-non-GstObject GObjects.
-
-The goal is to enable leak tracing on GStreamer's Continuous-Integration and
-testing system, both for the regular unit tests (make check) and media tests
-(gst-validate), so that accidental leaks in common code paths can be detected
-and fixed quickly.
-
-For more information about the new tracer, check out Guillaume Desmottes's
-["Tracking Memory Leaks"][leaks-talk] talk or his [blog post][leaks-blog] about
-the topic.
-
-[leaks-talk]: https://gstconf.ubicast.tv/videos/tracking-memory-leaks/
-[leaks-blog]: https://blog.desmottes.be/?post/2016/06/20/GStreamer-leaks-tracer
-
-### GES and NLE changes
-
-- Clip priorities are now handled by the layers, and the GESTimelineElement
- priority property is now deprecated and unused
-- Enhanced (de)interlacing support to always use the `deinterlace` element
- and expose needed properties to users
-- Allow reusing clips children after removing the clip from a layer
-- We are now testing many more rendering formats in the gst-validate
- test suite, and failures have been fixed.
-- Also many bugs have been fixed in this cycle!
-
-### GStreamer validate changes
-
-This cycle has been focused on making GstValidate more than just a validating
-tool, but also a tool to help developers debug their GStreamer issues. When
-reporting issues, we try to gather as much information as possible and expose
-it to end users in a useful way. For an example of such enhancements, check out
-Thibault Saunier's [blog post](improving-debugging-gstreamer-validate) about
-the new Not Negotiated Error reporting mechanism.
-
-Playbin3 support has been added so we can run validate tests with `playbin3`
-instead of playbin.
-
-We are now able to properly communicate between `gst-validate-launcher` and
-launched subprocesses with actual IPC between them. That has enabled the test
-launcher to handle failing tests specifying the exact expected issue(s).
-
-[improving-debugging-gstreamer-validate]: https://blogs.s-osg.org/improving-debugging-gstreamer-validate/
-
-### gst-libav changes
-
-gst-libav uses the recently released ffmpeg 3.2 now, which brings a lot of
-improvements and bugfixes from the ffmpeg team in addition to various new
-codec mappings on the GStreamer side and quite a few bugfixes to the GStreamer
-integration to make it more robust.
-
-## Build and Dependencies
-
-### Experimental support for Meson as build system
-
-#### Overview
-
-We have have added support for building GStreamer using the
-[Meson build system][meson]. This is currently experimental, but should work
-fine at least on Linux using the gcc or clang toolchains and on Windows using
-the MingW or MSVC toolchains.
-
-Autotools remains the primary build system for the time being, but we hope to
-someday replace it and will steadily work towards that goal.
-
-More information about the background and implications of all this and where
-we're hoping to go in future with this can be found in [Tim's mail][meson-mail]
-to the gstreamer-devel mailing list.
-
-For more information on Meson check out [these videos][meson-videos] and also
-the [Meson talk][meson-gstconf] at the GStreamer Conference.
-
-Immediate benefits for Linux users are faster builds and rebuilds. At the time
-of writing the Meson build of GStreamer is used by default in GNOME's jhbuild
-system.
-
-The Meson build currently still lacks many of the fine-grained configuration
-options to enable/disable specific plugins. These will be added back in due
-course.
-
-Note: The meson build files are not distributed in the source tarballs, you will
-need to get GStreamer from git if you want try it out.
-
-[meson]: http://mesonbuild.com/
-[meson-mail]: https://lists.freedesktop.org/archives/gstreamer-devel/2016-September/060231.html
-[meson-videos]: http://mesonbuild.com/videos.html
-[meson-gstconf]: https://gstconf.ubicast.tv/videos/gstreamer-development-on-windows-ans-faster-builds-everywhere-with-meson/
-
-#### Windows Visual Studio toolchain support
-
-Windows users might appreciate being able to build GStreamer using the MSVC
-toolchain, which is not possible using autotools. This means that it will be
-possible to debug GStreamer and applications in Visual Studio, for example.
-We require VS2015 or newer for this at the moment.
-
-There are two ways to build GStreamer using the MSVC toolchain:
-
-1. Using the MSVC command-line tools (`cl.exe` etc.) via Meson's "ninja" backend.
-2. Letting Meson's "vs2015" backend generate Visual Studio project files that
- can be opened in Visual Studio and compiled from there.
-
-This is currently only for adventurous souls though. All the bits are in place,
-but support for all of this has not been merged into GStreamer's cerbero build
-tool yet at the time of writing. This will hopefully happen in the next cycle,
-but for now this means that those wishing to compile GStreamer with MSVC will
-have to get their hands dirty.
-
-There are also no binary SDK builds using the MSVC toolchain yet.
-
-For more information on GStreamer builds using Meson and the Windows toolchain
-check out Nirbheek Chauhan's blog post ["Building and developing GStreamer using Visual Studio"][msvc-blog].
-
-[msvc-blog]: http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
-
-### Dependencies
-
-#### gstreamer
-
-libunwind was added as an optional dependency. It is used only for debugging
-and tracing purposes.
-
-The `opencv` plugin in gst-plugins-bad can now be built against OpenCV
-version 3.1, previously only 2.3-2.5 were supported.
-
-#### gst-plugins-ugly
-
-- `mpeg2dec` now requires at least libmpeg2 0.5.1 (from 2008).
-
-#### gst-plugins-bad
-
-- `gltransformation` now requires at least graphene 1.4.0.
-
-- `lv2` now plugin requires at least lilv 0.16 instead of slv2.
-
-### Packaging notes
-
-Packagers please note that the `gst/gstconfig.h` public header file in the
-GStreamer core library moved back from being an architecture dependent include
-to being architecture independent, and thus it is no longer installed into
-`$(libdir)/gstreamer-1.0/include/gst` but into the normal include directory
-where it lives happily ever after with all the other public header files. The
-reason for this is that we now check whether the target supports unaligned
-memory access based on predefined compiler macros at compile time instead of
-checking it at configure time.
-
-## Platform-specific improvements
-
-### Android
-
-#### New universal binaries for all supported ABIs
-
-We now provide a "universal" tarball to allow building apps against all the
-architectures currently supported (x86, x86-64, armeabi, armeabi-v7a,
-armeabi-v8a). This is needed for building with recent versions of the Android
-NDK which defaults to building against all supported ABIs. Use [the Android
-player example][android-player-example-build] as a reference for the required
-changes.
-
-[android-player-example-build]: https://cgit.freedesktop.org/gstreamer/gst-examples/commit/playback/player/android?id=a5cdde9119f038a1eb365aca20faa9741a38e788
-
-#### Miscellaneous
-
-- New `ahssrc` element that allows reading the hardware sensors, e.g. compass
- or accelerometer.
-
-### macOS (OS/X) and iOS
-
-- Support for querying available devices on OS/X via the GstDeviceProvider
- API was added.
-- It is now possible to create OpenGL|ES 3.x contexts on iOS and use them in
- combination with the VideoToolbox based decoder element.
-- many OpenGL/GLES improvements, see OpenGL section above
-
-### Windows
-
-- gstconfig.h: Always use dllexport/import on Windows with MSVC
-- Miscellaneous fixes to make libs and plugins compile with the MVSC toolchain
-- MSVC toolchain support (see Meson section above for more details)
-
-## New Modules for Documentation, Examples, Meson Build
-
-Three new git modules have been added recently:
-
-### gst-docs
-
-This is a new module where we will maintain documentation in the markdown
-format.
-
-It contains the former gstreamer.com SDK tutorials which have kindly been made
-available by Fluendo under a Creative Commons license. The tutorials have been
-reviewed and updated for GStreamer 1.x and will be available as part of the
-[official GStreamer documentation][doc] going forward. The old gstreamer.com
-site will then be shut down with redirects pointing to the updated tutorials.
-
-Some of the existing docbook XML-formatted documentation from the GStreamer
-core module such as the *Application Development Manual* and the *Plugin
-Writer's Guide* have been converted to markdown as well and will be maintained
-in the gst-docs module in future. They will be removed from the GStreamer core
-module in the next cycle.
-
-This is just the beginning. Our goal is to provide a more cohesive documentation
-experience for our users going forward, and easier to create and maintain
-documentation for developers. There is a lot more work to do, get in touch if
-you want to help out.
-
-If you encounter any problems or spot any omissions or outdated content in the
-new documentation, please [file a bug in bugzilla][doc-bug] to let us know.
-
-We will probably release gst-docs as a separate tarball for distributions to
-package in the next cycle.
-
-[doc]: http://gstreamer.freedesktop.org/documentation/
-[doc-bug]: https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=documentation
-
-### gst-examples
-
-A new [module][examples-git] has been added for examples. It does not contain
-much yet, currently it only contains a small [http-launch][http-launch] utility
-that serves a pipeline over http as well as various [GstPlayer playback frontends][puis]
-for Android, iOS, Gtk+ and Qt.
-
-More examples will be added over time. The examples in this repository should
-be more useful and more substantial than most of the examples we ship as part
-of our other modules, and also written in a way that makes them good example
-code. If you have ideas for examples, let us know.
-
-No decision has been made yet if this module will be released and/or packaged.
-It probably makes sense to do so though.
-
-[examples-git]: https://cgit.freedesktop.org/gstreamer/gst-examples/tree/
-[http-launch]: https://cgit.freedesktop.org/gstreamer/gst-examples/tree/network/http-launch/
-[puis]: https://cgit.freedesktop.org/gstreamer/gst-examples/tree/playback/player
-
-### gst-build
-
-[gst-build][gst-build-git] is a new meta module to build GStreamer using the
-new Meson build system. This module is not required to build GStreamer with
-Meson, it is merely for convenience and aims to provide a development setup
-similar to the existing `gst-uninstalled` setup.
-
-gst-build makes use of Meson's [subproject feature][meson-subprojects] and sets
-up the various GStreamer modules as subprojects, so they can all be updated and
-built in parallel.
-
-This module is still very new and highly experimental. It should work at least
-on Linux and Windows (OS/X needs some build fixes). Let us know of any issues
-you encounter by popping into the `#gstreamer` IRC channel or by
-[filing a bug][gst-build-bug].
-
-This module will probably not be released or packaged (does not really make sense).
-
-[gst-build-git]: https://cgit.freedesktop.org/gstreamer/gst-build/tree/
-[gst-build-bug]: https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-build
-[meson-subprojects]: https://github.com/mesonbuild/meson/wiki/Subprojects
-
-## Contributors
-
-Aaron Boxer, Aleix Conchillo Flaqué, Alessandro Decina, Alexandru Băluț, Alex
-Ashley, Alex-P. Natsios, Alistair Buxton, Allen Zhang, Andreas Naumann, Andrew
-Eikum, Andy Devar, Anthony G. Basile, Arjen Veenhuizen, Arnaud Vrac, Artem
-Martynovich, Arun Raghavan, Aurélien Zanelli, Barun Kumar Singh, Bernhard
-Miller, Brad Lackey, Branko Subasic, Carlos Garcia Campos, Carlos Rafael
-Giani, Christoffer Stengren, Daiki Ueno, Damian Ziobro, Danilo Cesar Lemes de
-Paula, David Buchmann, Dimitrios Katsaros, Duncan Palmer, Edward Hervey,
-Emmanuel Poitier, Enrico Jorns, Enrique Ocaña González, Fabrice Bellet,
-Florian Zwoch, Florin Apostol, Francisco Velazquez, Frédéric Bertolus, Fredrik
-Fornwall, Gaurav Gupta, George Kiagiadakis, Georg Lippitsch, Göran Jönsson,
-Graham Leggett, Gregoire Gentil, Guillaume Desmottes, Gwang Yoon Hwang, Haakon
-Sporsheim, Haihua Hu, Havard Graff, Heinrich Fink, Hoonhee Lee, Hyunjun Ko,
-Iain Lane, Ian, Ian Jamison, Jagyum Koo, Jake Foytik, Jakub Adam, Jan
-Alexander Steffens (heftig), Jan Schmidt, Javier Martinez Canillas, Jerome
-Laheurte, Jesper Larsen, Jie Jiang, Jihae Yi, Jimmy Ohn, Jinwoo Ahn, Joakim
-Johansson, Joan Pau Beltran, Jonas Holmberg, Jonathan Matthew, Jonathan Roy,
-Josep Torra, Julien Isorce, Jun Ji, Jürgen Slowack, Justin Kim, Kazunori
-Kobayashi, Kieran Bingham, Kipp Cannon, Koop Mast, Kouhei Sutou, Kseniia, Kyle
-Schwarz, Kyungyong Kim, Linus Svensson, Luis de Bethencourt, Marcin Kolny,
-Marcin Lewandowski, Marianna Smidth Buschle, Mario Sanchez Prada, Mark
-Combellack, Mark Nauwelaerts, Martin Kelly, Matej Knopp, Mathieu Duponchelle,
-Mats Lindestam, Matthew Gruenke, Matthew Waters, Michael Olbrich, Michal Lazo,
-Miguel París Díaz, Mikhail Fludkov, Minjae Kim, Mohan R, Munez, Nicola Murino,
-Nicolas Dufresne, Nicolas Huet, Nikita Bobkov, Nirbheek Chauhan, Olivier
-Crête, Paolo Pettinato, Patricia Muscalu, Paulo Neves, Peng Liu, Peter
-Seiderer, Philippe Normand, Philippe Renon, Philipp Zabel, Pierre Lamot, Piotr
-Drąg, Prashant Gotarne, Raffaele Rossi, Ray Strode, Reynaldo H. Verdejo
-Pinochet, Santiago Carot-Nemesio, Scott D Phillips, Sebastian Dröge, Sebastian
-Rasmussen, Sergei Saveliev, Sergey Borovkov, Sergey Mamonov, Sergio Torres
-Soldado, Seungha Yang, sezero, Song Bing, Sreerenj Balachandran, Stefan Sauer,
-Stephen, Steven Hoving, Stian Selnes, Thiago Santos, Thibault Saunier, Thijs
-Vermeir, Thomas Bluemel, Thomas Jones, Thomas Klausner, Thomas Scheuermann,
-Tim-Philipp Müller, Ting-Wei Lan, Tom Schoonjans, Ursula Maplehurst, Vanessa
-Chipirras Navalon, Víctor Manuel Jáquez Leal, Vincent Penquerc'h, Vineeth TM,
-Vivia Nikolaidou, Vootele Vesterblom, Wang Xin-yu (王昕宇), William Manley,
-Wim Taymans, Wonchul Lee, Xabier Rodriguez Calvar, Xavier Claessens, xlazom00,
-Yann Jouanin, Zaheer Abbas Merali
-
-... and many others who have contributed bug reports, translations, sent
-suggestions or helped testing.
-
-## Bugs fixed in 1.10
-
-More than [750 bugs][bugs-fixed-in-1.10] have been fixed during
-the development of 1.10.
-
-This list does not include issues that have been cherry-picked into the
-stable 1.8 branch and fixed there as well, all fixes that ended up in the
-1.8 branch are also included in 1.10.
-
-This list also does not include issues that have been fixed without a bug
-report in bugzilla, so the actual number of fixes is much higher.
-
-[bugs-fixed-in-1.10]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=164074&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.8.1&target_milestone=1.8.2&target_milestone=1.8.3&target_milestone=1.8.4&target_milestone=1.9.1&target_milestone=1.9.2&target_milestone=1.9.90&target_milestone=1.10.0
-
-## Stable 1.10 branch
-
-After the 1.10.0 release there will be several 1.10.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 1.10.x bug-fix releases will be made from the git 1.10 branch,
-which is a stable branch.
-
-### 1.10.0
-
-1.10.0 was released on 1st November 2016.
-
-## Known Issues
-
-- iOS builds with iOS 6 SDK and old C++ STL. You need to select iOS 6 instead
- of 7 or 8 in your projects settings to be able to link applications.
- [Bug #766366](https://bugzilla.gnome.org/show_bug.cgi?id=766366)
-- Code signing for Apple platforms has some problems currently, requiring
- manual work to get your application signed. [Bug #771860](https://bugzilla.gnome.org/show_bug.cgi?id=771860)
-- Building applications with Android NDK r13 on Windows does not work. Other
- platforms and earlier/later versions of the NDK are not affected.
- [Bug #772842](https://bugzilla.gnome.org/show_bug.cgi?id=772842)
-- The new leaks tracer may deadlock the application (or exhibit other undefined
- behaviour) when `SIGUSR` handling is enabled via the `GST_LEAKS_TRACER_SIG`
- environment variable. [Bug #770373](https://bugzilla.gnome.org/show_bug.cgi?id=770373)
-- vp8enc crashes on 32 bit Windows, but was working fine in 1.6. 64 bit Windows is unaffected.
- [Bug #763663](https://bugzilla.gnome.org/show_bug.cgi?id=763663)
-
-## Schedule for 1.12
-
-Our next major feature release will be 1.12, and 1.11 will be the unstable
-development version leading up to the stable 1.12 release. The development
-of 1.11/1.12 will happen in the git master branch.
-
-The plan for the 1.12 development cycle is yet to be confirmed, but it is
-expected that feature freeze will be around early/mid-January,
-followed by several 1.11 pre-releases and the new 1.12 stable release
-in March.
-
-1.12 will be backwards-compatible to the stable 1.10, 1.8, 1.6, 1.4, 1.2 and
-1.0 release series.
-
-- - -
-
-*These release notes have been prepared by Olivier Crête, Sebastian Dröge,
-Nicolas Dufresne, Edward Hervey, Víctor Manuel Jáquez Leal, Tim-Philipp
-Müller, Reynaldo H. Verdejo Pinochet, Arun Raghavan, Thibault Saunier,
-Jan Schmidt, Wim Taymans, Matthew Waters*
-
-*License: [CC BY-SA 4.0](http://creativecommons.org/licenses/by-sa/4.0/)*
-
+This is GStreamer 1.11.1.
diff --git a/RELEASE b/RELEASE
index 8b76e38..d536918 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,23 +1,32 @@
-Release notes for GStreamer RTSP Server Library 1.10.0
+Release notes for GStreamer RTSP Server Library 1.11.1
-The GStreamer team is pleased to announce the first release of the new stable
-1.10 release series. The 1.10 release series is adding new features on top of
-the 1.0, 1.2, 1.4, 1.6 and 1.8 series and is part of the API and ABI-stable 1.x
-release series of the GStreamer multimedia framework.
+The GStreamer team is pleased to announce the first release of the unstable
+1.11 release series. The 1.11 release series is adding new features on top of
+the 1.0, 1.2, 1.4, 1.6, 1.8 and 1.10 series and is part of the API and ABI-stable 1.x release
+series of the GStreamer multimedia framework. The unstable 1.11 release series
+will lead to the stable 1.12 release series in the next weeks. Any newly added
+API can still change until that point.
-Binaries for Android, iOS, Mac OS X and Windows will be provided shortly after
-the source release by the GStreamer project during the stable 1.10 release
-series.
+Full release notes will be provided at some point during the 1.11 release
+cycle, highlighting all the new features, bugfixes, performance optimizations
+and other important changes.
+
+
+Binaries for Android, iOS, Mac OS X and Windows will be provided in the next days.
Bugs fixed in this release
- * 771983 : Deadlock when closing session and backlog is full.
- * 772478 : Missing video stream from SDP
- * 773640 : rtspclient unit test failures
+ * 758062 : rtsp-client: emit new rtsp request signals in the beginning of each request
+ * 771830 : There is no time out in idle connection RTSP server
+ * 774173 : media: emit signal SIGNAL_NEW_STATE only when state change happens
+ * 774640 : gst-rtsp-server: Enable building with MSVC
+ * 776867 : pkgconfig: fix -uninstalled pc file
+ * 777037 : rtsp-factory: just fixing a little typo in comments
+ * 774416 : RTSP digest Authentification for gst-rtsp-server
==== Download ====
@@ -54,8 +63,19 @@ subscribe to the gstreamer-devel list.
Contributors to this release
+ * Aleksandr Slobodeniuk
+ * Branko Subasic
+ * Dag Gullberg
+ * Edward Hervey
+ * Guillaume Desmottes
* Göran Jönsson
- * Nikita Bobkov
+ * Jan Schmidt
+ * Kseniia Vasilchuk
+ * Matthew Waters
+ * Neha Arora
+ * Patricia Muscalu
+ * Scott D Phillips
+ * Sebastian Dröge
+ * Thibault Saunier
* Tim-Philipp Müller
- * Xavier Claessens
  \ No newline at end of file
diff --git a/configure.ac b/configure.ac
index 8167ff9..012531b 100644
--- a/configure.ac
+++ b/configure.ac
@@ -2,7 +2,7 @@ AC_PREREQ(2.69)
dnl initialize autoconf
dnl when going to/from release please set the nano (fourth number) right !
dnl releases only do Wall, cvs and prerelease does Werror too
-AC_INIT([GStreamer RTSP Server Library], [1.11.0.1],
+AC_INIT([GStreamer RTSP Server Library], [1.11.1],
[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
[gst-rtsp-server])
AG_GST_INIT
@@ -53,13 +53,13 @@ dnl 1.2.5 => 205
dnl 1.10.9 (who knows) => 1009
dnl
dnl sets GST_LT_LDFLAGS
-AS_LIBTOOL(GST, 1100, 0, 1100)
+AS_LIBTOOL(GST, 1101, 0, 1101)
dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.11.0.1
-GSTPB_REQ=1.11.0.1
-GSTPG_REQ=1.11.0.1
-GSTPD_REQ=1.11.0.1
+GST_REQ=1.11.1
+GSTPB_REQ=1.11.1
+GSTPG_REQ=1.11.1
+GSTPD_REQ=1.11.1
dnl *** autotools stuff ****
diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap
index 27bb2ba..33fe0f8 100644
--- a/gst-rtsp-server.doap
+++ b/gst-rtsp-server.doap
@@ -32,6 +32,16 @@ RTSP server library based on GStreamer
<release>
<Version>
+ <revision>1.11.1</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2017-01-12</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.11.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.10.0</revision>
<branch>master</branch>
<name></name>
diff --git a/win32/common/libgstrtspserver.def b/win32/common/libgstrtspserver.def
index 18318e3..93f12fa 100644
--- a/win32/common/libgstrtspserver.def
+++ b/win32/common/libgstrtspserver.def
@@ -197,6 +197,7 @@ EXPORTS
gst_rtsp_session_get_sessionid
gst_rtsp_session_get_timeout
gst_rtsp_session_get_type
+ gst_rtsp_session_is_expired
gst_rtsp_session_is_expired_usec
gst_rtsp_session_manage_media
gst_rtsp_session_media_alloc_channels
@@ -212,6 +213,7 @@ EXPORTS
gst_rtsp_session_media_set_state
gst_rtsp_session_media_set_transport
gst_rtsp_session_new
+ gst_rtsp_session_next_timeout
gst_rtsp_session_next_timeout_usec
gst_rtsp_session_pool_cleanup
gst_rtsp_session_pool_create