mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer-rs.git
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877 lines
35 KiB
XML
877 lines
35 KiB
XML
<?xml version="1.0"?>
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<!-- This file was automatically generated from C sources - DO NOT EDIT!
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To affect the contents of this file, edit the original C definitions,
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and/or use gtk-doc annotations. -->
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<repository version="1.2"
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xmlns="http://www.gtk.org/introspection/core/1.0"
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xmlns:c="http://www.gtk.org/introspection/c/1.0"
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xmlns:glib="http://www.gtk.org/introspection/glib/1.0">
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<include name="Gst" version="1.0"/>
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<include name="GstSdp" version="1.0"/>
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<package name="gstreamer-webrtc-1.0"/>
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<c:include name="gst/webrtc/webrtc.h"/>
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<namespace name="GstWebRTC"
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version="1.0"
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shared-library="libgstwebrtc-1.0.so.0"
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c:identifier-prefixes="Gst"
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c:symbol-prefixes="gst">
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<enumeration name="WebRTCDTLSSetup" c:type="GstWebRTCDTLSSetup">
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<doc xml:space="preserve">GST_WEBRTC_DTLS_SETUP_NONE: none
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GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
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GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
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GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
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<member name="none" value="0" c:identifier="GST_WEBRTC_DTLS_SETUP_NONE">
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</member>
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<member name="actpass"
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value="1"
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c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS">
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</member>
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<member name="active"
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value="2"
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c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE">
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</member>
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<member name="passive"
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value="3"
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c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE">
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</member>
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</enumeration>
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<class name="WebRTCDTLSTransport"
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c:symbol-prefix="webrtc_dtls_transport"
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c:type="GstWebRTCDTLSTransport"
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parent="Gst.Object"
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glib:type-name="GstWebRTCDTLSTransport"
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glib:get-type="gst_webrtc_dtls_transport_get_type"
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glib:type-struct="WebRTCDTLSTransportClass">
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<constructor name="new" c:identifier="gst_webrtc_dtls_transport_new">
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<return-value transfer-ownership="none">
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<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
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</return-value>
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<parameters>
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<parameter name="session_id" transfer-ownership="none">
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<type name="guint" c:type="guint"/>
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</parameter>
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<parameter name="rtcp" transfer-ownership="none">
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<type name="gboolean" c:type="gboolean"/>
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</parameter>
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</parameters>
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</constructor>
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<method name="set_transport"
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c:identifier="gst_webrtc_dtls_transport_set_transport">
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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<parameters>
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<instance-parameter name="transport" transfer-ownership="none">
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<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
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</instance-parameter>
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<parameter name="ice" transfer-ownership="none">
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<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
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</parameter>
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</parameters>
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</method>
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<property name="certificate" writable="1" transfer-ownership="none">
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<type name="utf8" c:type="gchar*"/>
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</property>
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<property name="client" writable="1" transfer-ownership="none">
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<type name="gboolean" c:type="gboolean"/>
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</property>
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<property name="remote-certificate" transfer-ownership="none">
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<type name="utf8" c:type="gchar*"/>
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</property>
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<property name="rtcp"
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writable="1"
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construct-only="1"
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transfer-ownership="none">
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<type name="gboolean" c:type="gboolean"/>
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</property>
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<property name="session-id"
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writable="1"
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construct-only="1"
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transfer-ownership="none">
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<type name="guint" c:type="guint"/>
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</property>
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<property name="state" introspectable="0" transfer-ownership="none">
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<type/>
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</property>
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<property name="transport" transfer-ownership="none">
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<type name="WebRTCICETransport"/>
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</property>
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<field name="parent">
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<type name="Gst.Object" c:type="GstObject"/>
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</field>
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<field name="transport">
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<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
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</field>
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<field name="state">
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<type name="WebRTCDTLSTransportState"
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c:type="GstWebRTCDTLSTransportState"/>
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</field>
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<field name="is_rtcp">
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<type name="gboolean" c:type="gboolean"/>
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</field>
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<field name="client">
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<type name="gboolean" c:type="gboolean"/>
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</field>
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<field name="session_id">
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<type name="guint" c:type="guint"/>
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</field>
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<field name="dtlssrtpenc">
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<type name="Gst.Element" c:type="GstElement*"/>
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</field>
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<field name="dtlssrtpdec">
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<type name="Gst.Element" c:type="GstElement*"/>
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</field>
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<field name="_padding">
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<array zero-terminated="0" c:type="gpointer" fixed-size="4">
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<type name="gpointer" c:type="gpointer"/>
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</array>
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</field>
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</class>
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<record name="WebRTCDTLSTransportClass"
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c:type="GstWebRTCDTLSTransportClass"
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glib:is-gtype-struct-for="WebRTCDTLSTransport">
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<field name="parent_class">
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<type name="Gst.BinClass" c:type="GstBinClass"/>
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</field>
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<field name="_padding">
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<array zero-terminated="0" c:type="gpointer" fixed-size="4">
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<type name="gpointer" c:type="gpointer"/>
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</array>
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</field>
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</record>
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<enumeration name="WebRTCDTLSTransportState"
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c:type="GstWebRTCDTLSTransportState">
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<doc xml:space="preserve">GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
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GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
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<member name="new"
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value="0"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW">
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</member>
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<member name="closed"
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value="1"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED">
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</member>
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<member name="failed"
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value="2"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED">
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</member>
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<member name="connecting"
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value="3"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING">
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</member>
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<member name="connected"
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value="4"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED">
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</member>
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</enumeration>
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<enumeration name="WebRTCICEComponent" c:type="GstWebRTCICEComponent">
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<doc xml:space="preserve">GST_WEBRTC_ICE_COMPONENT_RTP,
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GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
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<member name="rtp" value="0" c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP">
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</member>
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<member name="rtcp"
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value="1"
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c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP">
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</member>
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</enumeration>
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<enumeration name="WebRTCICEConnectionState"
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c:type="GstWebRTCICEConnectionState">
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<doc xml:space="preserve">GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
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GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
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GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
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GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
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GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
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GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
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GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
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See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink></doc>
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<member name="new"
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value="0"
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW">
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</member>
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<member name="checking"
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value="1"
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING">
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</member>
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<member name="connected"
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value="2"
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED">
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</member>
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<member name="completed"
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value="3"
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED">
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</member>
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<member name="failed"
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value="4"
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED">
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</member>
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<member name="disconnected"
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value="5"
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED">
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</member>
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<member name="closed"
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value="6"
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED">
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</member>
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</enumeration>
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<enumeration name="WebRTCICEGatheringState"
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c:type="GstWebRTCICEGatheringState">
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<doc xml:space="preserve">GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
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GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
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GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
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See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink></doc>
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<member name="new"
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value="0"
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c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW">
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</member>
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<member name="gathering"
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value="1"
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c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING">
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</member>
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<member name="complete"
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value="2"
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c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE">
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</member>
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</enumeration>
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<enumeration name="WebRTCICERole" c:type="GstWebRTCICERole">
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<doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
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GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
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<member name="controlled"
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value="0"
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c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED">
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</member>
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<member name="controlling"
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value="1"
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c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING">
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</member>
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</enumeration>
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<class name="WebRTCICETransport"
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c:symbol-prefix="webrtc_ice_transport"
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c:type="GstWebRTCICETransport"
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parent="Gst.Object"
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abstract="1"
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glib:type-name="GstWebRTCICETransport"
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glib:get-type="gst_webrtc_ice_transport_get_type"
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glib:type-struct="WebRTCICETransportClass">
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<virtual-method name="gather_candidates">
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<return-value transfer-ownership="none">
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<type name="gboolean" c:type="gboolean"/>
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</return-value>
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<parameters>
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<instance-parameter name="transport" transfer-ownership="none">
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<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
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</instance-parameter>
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</parameters>
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</virtual-method>
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<method name="connection_state_change"
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c:identifier="gst_webrtc_ice_transport_connection_state_change">
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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<parameters>
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<instance-parameter name="ice" transfer-ownership="none">
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<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
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</instance-parameter>
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<parameter name="new_state" transfer-ownership="none">
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<type name="WebRTCICEConnectionState"
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c:type="GstWebRTCICEConnectionState"/>
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</parameter>
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</parameters>
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</method>
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<method name="gathering_state_change"
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c:identifier="gst_webrtc_ice_transport_gathering_state_change">
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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<parameters>
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<instance-parameter name="ice" transfer-ownership="none">
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<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
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</instance-parameter>
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<parameter name="new_state" transfer-ownership="none">
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<type name="WebRTCICEGatheringState"
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c:type="GstWebRTCICEGatheringState"/>
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</parameter>
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</parameters>
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</method>
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<method name="new_candidate"
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c:identifier="gst_webrtc_ice_transport_new_candidate">
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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<parameters>
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<instance-parameter name="ice" transfer-ownership="none">
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<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
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</instance-parameter>
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<parameter name="stream_id" transfer-ownership="none">
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<type name="guint" c:type="guint"/>
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</parameter>
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<parameter name="component" transfer-ownership="none">
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<type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/>
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</parameter>
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<parameter name="attr" transfer-ownership="none">
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<type name="utf8" c:type="gchar*"/>
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</parameter>
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</parameters>
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</method>
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<method name="selected_pair_change"
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c:identifier="gst_webrtc_ice_transport_selected_pair_change">
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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<parameters>
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<instance-parameter name="ice" transfer-ownership="none">
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<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
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</instance-parameter>
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</parameters>
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</method>
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<property name="component"
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introspectable="0"
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writable="1"
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construct-only="1"
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transfer-ownership="none">
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<type/>
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</property>
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<property name="gathering-state"
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introspectable="0"
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transfer-ownership="none">
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<type/>
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</property>
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<property name="state" introspectable="0" transfer-ownership="none">
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<type/>
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</property>
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<field name="parent">
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<type name="Gst.Object" c:type="GstObject"/>
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</field>
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<field name="role">
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<type name="WebRTCICERole" c:type="GstWebRTCICERole"/>
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</field>
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<field name="component">
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<type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/>
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</field>
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<field name="state">
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<type name="WebRTCICEConnectionState"
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c:type="GstWebRTCICEConnectionState"/>
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</field>
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<field name="gathering_state">
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<type name="WebRTCICEGatheringState"
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c:type="GstWebRTCICEGatheringState"/>
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</field>
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<field name="src">
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<type name="Gst.Element" c:type="GstElement*"/>
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</field>
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<field name="sink">
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<type name="Gst.Element" c:type="GstElement*"/>
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</field>
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<field name="_padding">
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<array zero-terminated="0" c:type="gpointer" fixed-size="4">
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<type name="gpointer" c:type="gpointer"/>
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</array>
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</field>
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<glib:signal name="on-new-candidate" when="last">
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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<parameters>
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<parameter name="object" transfer-ownership="none">
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<type name="utf8" c:type="gchar*"/>
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</parameter>
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</parameters>
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</glib:signal>
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<glib:signal name="on-selected-candidate-pair-change" when="last">
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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</glib:signal>
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</class>
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<record name="WebRTCICETransportClass"
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c:type="GstWebRTCICETransportClass"
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glib:is-gtype-struct-for="WebRTCICETransport">
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<field name="parent_class">
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<type name="Gst.BinClass" c:type="GstBinClass"/>
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</field>
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<field name="gather_candidates">
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<callback name="gather_candidates">
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<return-value transfer-ownership="none">
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<type name="gboolean" c:type="gboolean"/>
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</return-value>
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<parameters>
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<parameter name="transport" transfer-ownership="none">
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<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
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</parameter>
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</parameters>
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</callback>
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</field>
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<field name="_padding">
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<array zero-terminated="0" c:type="gpointer" fixed-size="4">
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<type name="gpointer" c:type="gpointer"/>
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</array>
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</field>
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</record>
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<enumeration name="WebRTCPeerConnectionState"
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c:type="GstWebRTCPeerConnectionState">
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<doc xml:space="preserve">GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
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GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
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GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
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GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
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GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
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GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
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See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink></doc>
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<member name="new"
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value="0"
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c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW">
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</member>
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<member name="connecting"
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value="1"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING">
|
|
</member>
|
|
<member name="connected"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED">
|
|
</member>
|
|
<member name="disconnected"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED">
|
|
</member>
|
|
<member name="failed"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED">
|
|
</member>
|
|
<member name="closed"
|
|
value="5"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED">
|
|
</member>
|
|
</enumeration>
|
|
<class name="WebRTCRTPReceiver"
|
|
c:symbol-prefix="webrtc_rtp_receiver"
|
|
c:type="GstWebRTCRTPReceiver"
|
|
parent="Gst.Object"
|
|
glib:type-name="GstWebRTCRTPReceiver"
|
|
glib:get-type="gst_webrtc_rtp_receiver_get_type"
|
|
glib:type-struct="WebRTCRTPReceiverClass">
|
|
<constructor name="new" c:identifier="gst_webrtc_rtp_receiver_new">
|
|
<return-value transfer-ownership="none">
|
|
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
|
|
</return-value>
|
|
</constructor>
|
|
<method name="set_rtcp_transport"
|
|
c:identifier="gst_webrtc_rtp_receiver_set_rtcp_transport">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="receiver" transfer-ownership="none">
|
|
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
|
|
</instance-parameter>
|
|
<parameter name="transport" transfer-ownership="none">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_transport"
|
|
c:identifier="gst_webrtc_rtp_receiver_set_transport">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="receiver" transfer-ownership="none">
|
|
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
|
|
</instance-parameter>
|
|
<parameter name="transport" transfer-ownership="none">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<field name="parent">
|
|
<type name="Gst.Object" c:type="GstObject"/>
|
|
</field>
|
|
<field name="transport">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</field>
|
|
<field name="rtcp_transport">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="WebRTCRTPReceiverClass"
|
|
c:type="GstWebRTCRTPReceiverClass"
|
|
glib:is-gtype-struct-for="WebRTCRTPReceiver">
|
|
<field name="parent_class">
|
|
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<class name="WebRTCRTPSender"
|
|
c:symbol-prefix="webrtc_rtp_sender"
|
|
c:type="GstWebRTCRTPSender"
|
|
parent="Gst.Object"
|
|
glib:type-name="GstWebRTCRTPSender"
|
|
glib:get-type="gst_webrtc_rtp_sender_get_type"
|
|
glib:type-struct="WebRTCRTPSenderClass">
|
|
<constructor name="new" c:identifier="gst_webrtc_rtp_sender_new">
|
|
<return-value transfer-ownership="none">
|
|
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
|
|
</return-value>
|
|
</constructor>
|
|
<method name="set_rtcp_transport"
|
|
c:identifier="gst_webrtc_rtp_sender_set_rtcp_transport">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sender" transfer-ownership="none">
|
|
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
|
|
</instance-parameter>
|
|
<parameter name="transport" transfer-ownership="none">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_transport"
|
|
c:identifier="gst_webrtc_rtp_sender_set_transport">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sender" transfer-ownership="none">
|
|
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
|
|
</instance-parameter>
|
|
<parameter name="transport" transfer-ownership="none">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<field name="parent">
|
|
<type name="Gst.Object" c:type="GstObject"/>
|
|
</field>
|
|
<field name="transport">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</field>
|
|
<field name="rtcp_transport">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</field>
|
|
<field name="send_encodings">
|
|
<array name="GLib.Array" c:type="GArray*">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="WebRTCRTPSenderClass"
|
|
c:type="GstWebRTCRTPSenderClass"
|
|
glib:is-gtype-struct-for="WebRTCRTPSender">
|
|
<field name="parent_class">
|
|
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<class name="WebRTCRTPTransceiver"
|
|
c:symbol-prefix="webrtc_rtp_transceiver"
|
|
c:type="GstWebRTCRTPTransceiver"
|
|
parent="Gst.Object"
|
|
abstract="1"
|
|
glib:type-name="GstWebRTCRTPTransceiver"
|
|
glib:get-type="gst_webrtc_rtp_transceiver_get_type"
|
|
glib:type-struct="WebRTCRTPTransceiverClass">
|
|
<property name="mlineindex"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</property>
|
|
<property name="receiver"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="WebRTCRTPReceiver"/>
|
|
</property>
|
|
<property name="sender"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="WebRTCRTPSender"/>
|
|
</property>
|
|
<field name="parent">
|
|
<type name="Gst.Object" c:type="GstObject"/>
|
|
</field>
|
|
<field name="mline">
|
|
<type name="guint" c:type="guint"/>
|
|
</field>
|
|
<field name="mid">
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</field>
|
|
<field name="stopped">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="sender">
|
|
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
|
|
</field>
|
|
<field name="receiver">
|
|
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
|
|
</field>
|
|
<field name="direction">
|
|
<type name="WebRTCRTPTransceiverDirection"
|
|
c:type="GstWebRTCRTPTransceiverDirection"/>
|
|
</field>
|
|
<field name="current_direction">
|
|
<type name="WebRTCRTPTransceiverDirection"
|
|
c:type="GstWebRTCRTPTransceiverDirection"/>
|
|
</field>
|
|
<field name="codec_preferences">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="WebRTCRTPTransceiverClass"
|
|
c:type="GstWebRTCRTPTransceiverClass"
|
|
glib:is-gtype-struct-for="WebRTCRTPTransceiver">
|
|
<field name="parent_class">
|
|
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<enumeration name="WebRTCRTPTransceiverDirection"
|
|
c:type="GstWebRTCRTPTransceiverDirection">
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE">
|
|
</member>
|
|
<member name="inactive"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE">
|
|
</member>
|
|
<member name="sendonly"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY">
|
|
</member>
|
|
<member name="recvonly"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY">
|
|
</member>
|
|
<member name="sendrecv"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCSDPType" c:type="GstWebRTCSDPType">
|
|
<doc xml:space="preserve">GST_WEBRTC_SDP_TYPE_OFFER: offer
|
|
GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
|
|
GST_WEBRTC_SDP_TYPE_ANSWER: answer
|
|
GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
|
|
See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink></doc>
|
|
<member name="offer" value="1" c:identifier="GST_WEBRTC_SDP_TYPE_OFFER">
|
|
</member>
|
|
<member name="pranswer"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER">
|
|
</member>
|
|
<member name="answer"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER">
|
|
</member>
|
|
<member name="rollback"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK">
|
|
</member>
|
|
</enumeration>
|
|
<record name="WebRTCSessionDescription"
|
|
c:type="GstWebRTCSessionDescription"
|
|
glib:type-name="GstWebRTCSessionDescription"
|
|
glib:get-type="gst_webrtc_session_description_get_type"
|
|
c:symbol-prefix="webrtc_session_description">
|
|
<doc xml:space="preserve">sdp: the #GstSDPMessage of the description
|
|
See <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink></doc>
|
|
<field name="type" writable="1">
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</field>
|
|
<field name="sdp" writable="1">
|
|
<type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
|
|
</field>
|
|
<constructor name="new"
|
|
c:identifier="gst_webrtc_session_description_new">
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">a new #GstWebRTCSessionDescription from @type
|
|
and @sdp</doc>
|
|
<type name="WebRTCSessionDescription"
|
|
c:type="GstWebRTCSessionDescription*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="type" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstWebRTCSDPType</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</parameter>
|
|
<parameter name="sdp" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstSDPMessage</doc>
|
|
<type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</constructor>
|
|
<method name="copy" c:identifier="gst_webrtc_session_description_copy">
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">a new copy of @src</doc>
|
|
<type name="WebRTCSessionDescription"
|
|
c:type="GstWebRTCSessionDescription*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstWebRTCSessionDescription</doc>
|
|
<type name="WebRTCSessionDescription"
|
|
c:type="const GstWebRTCSessionDescription*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="free" c:identifier="gst_webrtc_session_description_free">
|
|
<doc xml:space="preserve">Free @desc and all associated resources</doc>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="desc" transfer-ownership="full">
|
|
<doc xml:space="preserve">a #GstWebRTCSessionDescription</doc>
|
|
<type name="WebRTCSessionDescription"
|
|
c:type="GstWebRTCSessionDescription*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
</record>
|
|
<enumeration name="WebRTCSignalingState" c:type="GstWebRTCSignalingState">
|
|
<doc xml:space="preserve">GST_WEBRTC_SIGNALING_STATE_STABLE: stable
|
|
GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
|
|
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
|
|
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
|
|
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
|
|
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
|
|
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate</ulink></doc>
|
|
<member name="stable"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE">
|
|
</member>
|
|
<member name="closed"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED">
|
|
</member>
|
|
<member name="have_local_offer"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER">
|
|
</member>
|
|
<member name="have_remote_offer"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER">
|
|
</member>
|
|
<member name="have_local_pranswer"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER">
|
|
</member>
|
|
<member name="have_remote_pranswer"
|
|
value="5"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCStatsType" c:type="GstWebRTCStatsType">
|
|
<doc xml:space="preserve">GST_WEBRTC_STATS_CODEC: codec
|
|
GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
|
|
GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
|
|
GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
|
|
GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
|
|
GST_WEBRTC_STATS_CSRC: csrc
|
|
GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
|
|
GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
|
|
GST_WEBRTC_STATS_STREAM: stream
|
|
GST_WEBRTC_STATS_TRANSPORT: transport
|
|
GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
|
|
GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
|
|
GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
|
|
GST_WEBRTC_STATS_CERTIFICATE: certificate</doc>
|
|
<member name="codec" value="1" c:identifier="GST_WEBRTC_STATS_CODEC">
|
|
</member>
|
|
<member name="inbound_rtp"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_STATS_INBOUND_RTP">
|
|
</member>
|
|
<member name="outbound_rtp"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_STATS_OUTBOUND_RTP">
|
|
</member>
|
|
<member name="remote_inbound_rtp"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP">
|
|
</member>
|
|
<member name="remote_outbound_rtp"
|
|
value="5"
|
|
c:identifier="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP">
|
|
</member>
|
|
<member name="csrc" value="6" c:identifier="GST_WEBRTC_STATS_CSRC">
|
|
</member>
|
|
<member name="peer_connection"
|
|
value="7"
|
|
c:identifier="GST_WEBRTC_STATS_PEER_CONNECTION">
|
|
</member>
|
|
<member name="data_channel"
|
|
value="8"
|
|
c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL">
|
|
</member>
|
|
<member name="stream" value="9" c:identifier="GST_WEBRTC_STATS_STREAM">
|
|
</member>
|
|
<member name="transport"
|
|
value="10"
|
|
c:identifier="GST_WEBRTC_STATS_TRANSPORT">
|
|
</member>
|
|
<member name="candidate_pair"
|
|
value="11"
|
|
c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR">
|
|
</member>
|
|
<member name="local_candidate"
|
|
value="12"
|
|
c:identifier="GST_WEBRTC_STATS_LOCAL_CANDIDATE">
|
|
</member>
|
|
<member name="remote_candidate"
|
|
value="13"
|
|
c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE">
|
|
</member>
|
|
<member name="certificate"
|
|
value="14"
|
|
c:identifier="GST_WEBRTC_STATS_CERTIFICATE">
|
|
</member>
|
|
</enumeration>
|
|
<function name="webrtc_sdp_type_to_string"
|
|
c:identifier="gst_webrtc_sdp_type_to_string">
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">the string representation of @type or "unknown" when @type is not
|
|
recognized.</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="type" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstWebRTCSDPType</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</namespace>
|
|
</repository>
|