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gir-files: Import from gstreamer-editing-services 1.18.2 at d55d1f5c

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Marijn Suijten 2021-01-05 02:50:54 +01:00
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33
gir-files/GES-1.0.gir
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@ -14992,7 +14992,7 @@ contains one frame)</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="VERSION_MICRO" value="0" c:type="GES_VERSION_MICRO">
<constant name="VERSION_MICRO" value="2" c:type="GES_VERSION_MICRO">
<type name="gint" c:type="gint"/>
</constant>
@ -15831,6 +15831,37 @@ formatter asset.</doc>
</parameter>
</parameters>
</function>
<docsection name="ges.h">
<doc xml:space="preserve">GES needs to be initialized after GStreamer itself. This section
contains the various functions to do so.</doc>
</docsection>
<docsection name="gestimeoverlayclip">
<doc xml:space="preserve">A #GESSourceClip that overlays timing information on top.
## Asset
The default asset ID is "time-overlay" (of type #GES_TYPE_SOURCE_CLIP),
but the framerate and video size can be overridden using an ID of the form:
```
time-overlay, framerate=60/1, width=1920, height=1080, max-duration=5.0
```
## Children properties
{{ libs/GESTimeOverlayClip-children-props.md }}
## Symbols</doc>
</docsection>
<docsection name="gestypes">
<doc xml:space="preserve">GStreamer Editing Services data types</doc>
</docsection>
<docsection name="gesuriasset">
<doc xml:space="preserve">The #GESUriClipAsset is a special #GESAsset that lets you handle
the media file to use inside the GStreamer Editing Services. It has APIs that
let you get information about the medias. Also, the tags found in the media file are
set as Metadata of the Asset.</doc>
</docsection>
<function name="init" c:identifier="ges_init">
<doc xml:space="preserve">Initialize the GStreamer Editing Service. Call this before any usage of
GES. You should take care of initilizing GStreamer before calling this

380
gir-files/GstRtspServer-1.0.gir
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@ -12098,6 +12098,386 @@ the media factory and retrieve the role with the same name.</doc>
<type name="utf8" c:type="const gchar*"/>
</field>
</record>
<docsection name="rtsp-address-pool">
<doc xml:space="preserve">The #GstRTSPAddressPool is an object that maintains a collection of network
addresses. It is used to allocate server ports and server multicast addresses
but also to reserve client provided destination addresses.
A range of addresses can be added with gst_rtsp_address_pool_add_range().
Both multicast and unicast addresses can be added.
With gst_rtsp_address_pool_acquire_address() an unused address and port range
can be acquired from the pool. With gst_rtsp_address_pool_reserve_address() a
specific address can be retrieved. Both methods return a boxed
#GstRTSPAddress that should be freed with gst_rtsp_address_free() after
usage, which brings the address back into the pool.
Last reviewed on 2013-07-16 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-auth">
<doc xml:space="preserve">The #GstRTSPAuth object is responsible for checking if the current user is
allowed to perform requested actions. The default implementation has some
reasonable checks but subclasses can implement custom security policies.
A new auth object is made with gst_rtsp_auth_new(). It is usually configured
on the #GstRTSPServer object.
The RTSP server will call gst_rtsp_auth_check() with a string describing the
check to perform. The possible checks are prefixed with
GST_RTSP_AUTH_CHECK_*. Depending on the check, the default implementation
will use the current #GstRTSPToken, #GstRTSPContext and
#GstRTSPPermissions on the object to check if an operation is allowed.
The default #GstRTSPAuth object has support for basic authentication. With
gst_rtsp_auth_add_basic() you can add a basic authentication string together
with the #GstRTSPToken that will become active when successfully
authenticated.
When a TLS certificate has been set with gst_rtsp_auth_set_tls_certificate(),
the default auth object will require the client to connect with a TLS
connection.
Last reviewed on 2013-07-16 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-client">
<doc xml:space="preserve">The client object handles the connection with a client for as long as a TCP
connection is open.
A #GstRTSPClient is created by #GstRTSPServer when a new connection is
accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
#GstRTSPAuth and #GstRTSPThreadPool from the server.
The client connection should be configured with the #GstRTSPConnection using
gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
using gst_rtsp_client_attach(). From then on the client will handle requests
on the connection.
Use gst_rtsp_client_session_filter() to iterate or modify all the
#GstRTSPSession objects managed by the client object.
Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-context">
<doc xml:space="preserve">Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-media">
<doc xml:space="preserve">a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
streaming to the clients. The actual data transfer is done by the
#GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
client does a DESCRIBE or SETUP of a resource.
A media is created with gst_rtsp_media_new() that takes the element that will
provide the streaming elements. For each of the streams, a new #GstRTSPStream
object needs to be made with the gst_rtsp_media_create_stream() which takes
the payloader element and the source pad that produces the RTP stream.
The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
prepare method will add rtpbin and sinks and sources to send and receive RTP
and RTCP packets from the clients. Each stream srcpad is connected to an
input into the internal rtpbin.
It is also possible to dynamically create #GstRTSPStream objects during the
prepare phase. With gst_rtsp_media_get_status() you can check the status of
the prepare phase.
After the media is prepared, it is ready for streaming. It will usually be
managed in a session with gst_rtsp_session_manage_media(). See
#GstRTSPSession and #GstRTSPSessionMedia.
The state of the media can be controlled with gst_rtsp_media_set_state ().
Seeking can be done with gst_rtsp_media_seek(), or gst_rtsp_media_seek_full()
or gst_rtsp_media_seek_trickmode() for finer control of the seek.
With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
cleanly shut down.
With gst_rtsp_media_set_shared(), the media can be shared between multiple
clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
can be prepared again after an unprepare.
Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-media-factory">
<doc xml:space="preserve">The #GstRTSPMediaFactory is responsible for creating or recycling
#GstRTSPMedia objects based on the passed URL.
The default implementation of the object can create #GstRTSPMedia objects
containing a pipeline created from a launch description set with
gst_rtsp_media_factory_set_launch().
Media from a factory can be shared by setting the shared flag with
gst_rtsp_media_factory_set_shared(). When a factory is shared,
gst_rtsp_media_factory_construct() will return the same #GstRTSPMedia when
the url matches.
Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-media-factory-uri">
<doc xml:space="preserve">This specialized #GstRTSPMediaFactory constructs media pipelines from a URI,
given with gst_rtsp_media_factory_uri_set_uri().
It will automatically demux and payload the different streams found in the
media at URL.
Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-mount-points">
<doc xml:space="preserve">A #GstRTSPMountPoints object maintains a relation between paths
and #GstRTSPMediaFactory objects. This object is usually given to
#GstRTSPClient and used to find the media attached to a path.
With gst_rtsp_mount_points_add_factory () and
gst_rtsp_mount_points_remove_factory(), factories can be added and
removed.
With gst_rtsp_mount_points_match() you can find the #GstRTSPMediaFactory
object that completely matches the given path.
Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-onvif-media">
<doc xml:space="preserve">a #GstRTSPOnvifMedia contains the complete GStreamer pipeline to manage the
streaming to the clients. The actual data transfer is done by the
#GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
On top of #GstRTSPMedia this subclass adds special ONVIF features.
Special ONVIF features that are currently supported is a backchannel for
the client to send back media to the server in a normal PLAY media. To
handle the ONVIF backchannel, a #GstRTSPOnvifMediaFactory and
#GstRTSPOnvifServer has to be used.</doc>
</docsection>
<docsection name="rtsp-onvif-media-factory">
<doc xml:space="preserve">The #GstRTSPOnvifMediaFactory is responsible for creating or recycling
#GstRTSPMedia objects based on the passed URL. Different to
#GstRTSPMediaFactory, this supports special ONVIF features and can create
#GstRTSPOnvifMedia in addition to normal #GstRTSPMedia.
Special ONVIF features that are currently supported is a backchannel for
the client to send back media to the server in a normal PLAY media, see
gst_rtsp_onvif_media_factory_set_backchannel_launch() and
gst_rtsp_onvif_media_factory_set_backchannel_bandwidth().</doc>
</docsection>
<docsection name="rtsp-onvif-server">
<doc xml:space="preserve">The server object is the object listening for connections on a port and
creating #GstRTSPOnvifClient objects to handle those connections.
The only different to #GstRTSPServer is that #GstRTSPOnvifServer creates
#GstRTSPOnvifClient that have special handling for ONVIF specific features,
like a backchannel that allows clients to send back media to the server.</doc>
</docsection>
<docsection name="rtsp-params">
<doc xml:space="preserve">Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-permissions">
<doc xml:space="preserve">The #GstRTSPPermissions object contains an array of roles and associated
permissions. The roles are represented with a string and the permissions with
a generic #GstStructure.
The permissions are deliberately kept generic. The possible values of the
roles and #GstStructure keys and values are only determined by the #GstRTSPAuth
object that performs the checks on the permissions and the current
#GstRTSPToken.
As a convenience function, gst_rtsp_permissions_is_allowed() can be used to
check if the permissions contains a role that contains the boolean value
%TRUE for the the given key.
Last reviewed on 2013-07-15 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-sdp">
<doc xml:space="preserve">Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-server">
<doc xml:space="preserve">The server object is the object listening for connections on a port and
creating #GstRTSPClient objects to handle those connections.
The server will listen on the address set with gst_rtsp_server_set_address()
and the port or service configured with gst_rtsp_server_set_service().
Use gst_rtsp_server_set_backlog() to configure the amount of pending requests
that the server will keep. By default the server listens on the current
network (0.0.0.0) and port 8554.
The server will require an SSL connection when a TLS certificate has been
set in the auth object with gst_rtsp_auth_set_tls_certificate().
To start the server, use gst_rtsp_server_attach() to attach it to a
#GMainContext. For more control, gst_rtsp_server_create_source() and
gst_rtsp_server_create_socket() can be used to get a #GSource and #GSocket
respectively.
gst_rtsp_server_transfer_connection() can be used to transfer an existing
socket to the RTSP server, for example from an HTTP server.
Once the server socket is attached to a mainloop, it will start accepting
connections. When a new connection is received, a new #GstRTSPClient object
is created to handle the connection. The new client will be configured with
the server #GstRTSPAuth, #GstRTSPMountPoints, #GstRTSPSessionPool and
#GstRTSPThreadPool.
The server uses the configured #GstRTSPThreadPool object to handle the
remainder of the communication with this client.
Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-session">
<doc xml:space="preserve">The #GstRTSPSession is identified by an id, unique in the
#GstRTSPSessionPool that created the session and manages media and its
configuration.
A #GstRTSPSession has a timeout that can be retrieved with
gst_rtsp_session_get_timeout(). You can check if the sessions is expired with
gst_rtsp_session_is_expired(). gst_rtsp_session_touch() will reset the
expiration counter of the session.
When a client configures a media with SETUP, a session will be created to
keep track of the configuration of that media. With
gst_rtsp_session_manage_media(), the media is added to the managed media
in the session. With gst_rtsp_session_release_media() the media can be
released again from the session. Managed media is identified in the sessions
with a url. Use gst_rtsp_session_get_media() to get the media that matches
(part of) the given url.
The media in a session can be iterated with gst_rtsp_session_filter().
Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-session-media">
<doc xml:space="preserve">The #GstRTSPSessionMedia object manages a #GstRTSPMedia with a given path.
With gst_rtsp_session_media_get_transport() and
gst_rtsp_session_media_set_transport() the transports of a #GstRTSPStream of
the managed #GstRTSPMedia can be retrieved and configured.
Use gst_rtsp_session_media_set_state() to control the media state and
transports.
Last reviewed on 2013-07-16 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-session-pool">
<doc xml:space="preserve">The #GstRTSPSessionPool object manages a list of #GstRTSPSession objects.
The maximum number of sessions can be configured with
gst_rtsp_session_pool_set_max_sessions(). The current number of sessions can
be retrieved with gst_rtsp_session_pool_get_n_sessions().
Use gst_rtsp_session_pool_create() to create a new #GstRTSPSession object.
The session object can be found again with its id and
gst_rtsp_session_pool_find().
All sessions can be iterated with gst_rtsp_session_pool_filter().
Run gst_rtsp_session_pool_cleanup() periodically to remove timed out sessions
or use gst_rtsp_session_pool_create_watch() to be notified when session
cleanup should be performed.
Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-stream">
<doc xml:space="preserve">The #GstRTSPStream object manages the data transport for one stream. It
is created from a payloader element and a source pad that produce the RTP
packets for the stream.
With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
The #GstRTSPStream will use the configured addresspool, as set with
gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
stream. With gst_rtsp_stream_get_multicast_address() you can get the
configured address.
With gst_rtsp_stream_get_server_port () you can get the port that the server
will use to receive RTCP. This is the part that the clients will use to send
RTCP to.
With gst_rtsp_stream_add_transport() destinations can be added where the
stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
the destination again.
Each #GstRTSPStreamTransport spawns one queue that will serve as a backlog of a
controllable maximum size when the reflux from the TCP connection's backpressure
starts spilling all over.
Unlike the backlog in rtspconnection, which we have decided should only contain
at most one RTP and one RTCP data message in order to allow control messages to
go through unobstructed, this backlog only consists of data messages, allowing
us to fill it up without concern.
When multiple TCP transports exist, for example in the context of a shared media,
we only pop samples from our appsinks when at least one of the transports doesn't
experience back pressure: this allows us to pace our sample popping to the speed
of the fastest client.
When a sample is popped, it is either sent directly on transports that don't
experience backpressure, or queued on the transport's backlog otherwise. Samples
are then popped from that backlog when the transport reports it has sent the message.
Once the backlog reaches an overly large duration, the transport is dropped as
the client was deemed too slow.</doc>
</docsection>
<docsection name="rtsp-stream-transport">
<doc xml:space="preserve">The #GstRTSPStreamTransport configures the transport used by a
#GstRTSPStream. It is usually manages by a #GstRTSPSessionMedia object.
With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured
to handle the RTP and RTCP packets from the stream, for example when they
need to be sent over TCP.
With gst_rtsp_stream_transport_set_active() the transports are added and
removed from the stream.
A #GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP
is received from the client. It will also call
gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out.
A #GstRTSPClient will call gst_rtsp_stream_transport_message_sent() when it
has sent a data message for the transport.
Last reviewed on 2013-07-16 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-thread-pool">
<doc xml:space="preserve">A #GstRTSPThreadPool manages reusable threads for various server tasks.
Currently the defined thread types can be found in #GstRTSPThreadType.
Threads of type #GST_RTSP_THREAD_TYPE_CLIENT are used to handle requests from
a connected client. With gst_rtsp_thread_pool_get_max_threads() a maximum
number of threads can be set after which the pool will start to reuse the
same thread for multiple clients.
Threads of type #GST_RTSP_THREAD_TYPE_MEDIA will be used to perform the state
changes of the media pipelines and handle its bus messages.
gst_rtsp_thread_pool_get_thread() can be used to create a #GstRTSPThread
object of the right type. The thread object contains a mainloop and context
that run in a seperate thread and can be used to attached sources to.
gst_rtsp_thread_reuse() can be used to reuse a thread for multiple purposes.
If all gst_rtsp_thread_reuse() calls are matched with a
gst_rtsp_thread_stop() call, the mainloop will be quit and the thread will
stop.
To configure the threads, a subclass of this object should be made and the
virtual methods should be overriden to implement the desired functionality.
Last reviewed on 2013-07-11 (1.0.0)</doc>
</docsection>
<docsection name="rtsp-token">
<doc xml:space="preserve">A #GstRTSPToken contains the permissions and roles of the user
performing the current request. A token is usually created when a user is
authenticated by the #GstRTSPAuth object and is then placed as the current
token for the current request.
#GstRTSPAuth can use the token and its contents to check authorization for
various operations by comparing the token to the #GstRTSPPermissions of the
object.
The accepted values of the token are entirely defined by the #GstRTSPAuth
object that implements the security policy.
Last reviewed on 2013-07-15 (1.0.0)</doc>
</docsection>
<function name="rtsp_context_get_current" c:identifier="gst_rtsp_context_get_current" moved-to="RTSPContext.get_current" introspectable="0">
<doc xml:space="preserve">Get the current #GstRTSPContext. This object is retrieved from the
current thread that is handling the request for a client.</doc>