--- /dev/null
+<?xml version="1.0"?>
+<!-- This file was automatically generated from C sources - DO NOT EDIT!
+To affect the contents of this file, edit the original C definitions,
+and/or use gtk-doc annotations. -->
+<repository version="1.2"
+ xmlns="http://www.gtk.org/introspection/core/1.0"
+ xmlns:c="http://www.gtk.org/introspection/c/1.0"
+ xmlns:glib="http://www.gtk.org/introspection/glib/1.0">
+ <include name="Gst" version="1.0"/>
+ <include name="GstSdp" version="1.0"/>
+ <package name="gstreamer-webrtc-1.0"/>
+ <c:include name="gst/webrtc/webrtc.h"/>
+ <namespace name="GstWebRTC"
+ version="1.0"
+ shared-library="libgstwebrtc-1.0.so.0"
+ c:identifier-prefixes="Gst"
+ c:symbol-prefixes="gst">
+ <enumeration name="WebRTCDTLSSetup"
+ glib:type-name="GstWebRTCDTLSSetup"
+ glib:get-type="gst_webrtc_dtls_setup_get_type"
+ c:type="GstWebRTCDTLSSetup">
+ <doc xml:space="preserve">GST_WEBRTC_DTLS_SETUP_NONE: none
+GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
+GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
+GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
+ <member name="none"
+ value="0"
+ c:identifier="GST_WEBRTC_DTLS_SETUP_NONE"
+ glib:nick="none">
+ </member>
+ <member name="actpass"
+ value="1"
+ c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS"
+ glib:nick="actpass">
+ </member>
+ <member name="active"
+ value="2"
+ c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE"
+ glib:nick="active">
+ </member>
+ <member name="passive"
+ value="3"
+ c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE"
+ glib:nick="passive">
+ </member>
+ </enumeration>
+ <class name="WebRTCDTLSTransport"
+ c:symbol-prefix="webrtc_dtls_transport"
+ c:type="GstWebRTCDTLSTransport"
+ parent="Gst.Object"
+ glib:type-name="GstWebRTCDTLSTransport"
+ glib:get-type="gst_webrtc_dtls_transport_get_type"
+ glib:type-struct="WebRTCDTLSTransportClass">
+ <constructor name="new" c:identifier="gst_webrtc_dtls_transport_new">
+ <return-value transfer-ownership="none">
+ <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+ </return-value>
+ <parameters>
+ <parameter name="session_id" transfer-ownership="none">
+ <type name="guint" c:type="guint"/>
+ </parameter>
+ <parameter name="rtcp" transfer-ownership="none">
+ <type name="gboolean" c:type="gboolean"/>
+ </parameter>
+ </parameters>
+ </constructor>
+ <method name="set_transport"
+ c:identifier="gst_webrtc_dtls_transport_set_transport">
+ <return-value transfer-ownership="none">
+ <type name="none" c:type="void"/>
+ </return-value>
+ <parameters>
+ <instance-parameter name="transport" transfer-ownership="none">
+ <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+ </instance-parameter>
+ <parameter name="ice" transfer-ownership="none">
+ <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+ </parameter>
+ </parameters>
+ </method>
+ <property name="certificate" writable="1" transfer-ownership="none">
+ <type name="utf8" c:type="gchar*"/>
+ </property>
+ <property name="client" writable="1" transfer-ownership="none">
+ <type name="gboolean" c:type="gboolean"/>
+ </property>
+ <property name="remote-certificate" transfer-ownership="none">
+ <type name="utf8" c:type="gchar*"/>
+ </property>
+ <property name="rtcp"
+ writable="1"
+ construct-only="1"
+ transfer-ownership="none">
+ <type name="gboolean" c:type="gboolean"/>
+ </property>
+ <property name="session-id"
+ writable="1"
+ construct-only="1"
+ transfer-ownership="none">
+ <type name="guint" c:type="guint"/>
+ </property>
+ <property name="state" transfer-ownership="none">
+ <type name="WebRTCDTLSTransportState"/>
+ </property>
+ <property name="transport" transfer-ownership="none">
+ <type name="WebRTCICETransport"/>
+ </property>
+ <field name="parent">
+ <type name="Gst.Object" c:type="GstObject"/>
+ </field>
+ <field name="transport">
+ <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+ </field>
+ <field name="state">
+ <type name="WebRTCDTLSTransportState"
+ c:type="GstWebRTCDTLSTransportState"/>
+ </field>
+ <field name="is_rtcp">
+ <type name="gboolean" c:type="gboolean"/>
+ </field>
+ <field name="client">
+ <type name="gboolean" c:type="gboolean"/>
+ </field>
+ <field name="session_id">
+ <type name="guint" c:type="guint"/>
+ </field>
+ <field name="dtlssrtpenc">
+ <type name="Gst.Element" c:type="GstElement*"/>
+ </field>
+ <field name="dtlssrtpdec">
+ <type name="Gst.Element" c:type="GstElement*"/>
+ </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="WebRTCDTLSTransportClass"
+ c:type="GstWebRTCDTLSTransportClass"
+ glib:is-gtype-struct-for="WebRTCDTLSTransport">
+ <field name="parent_class">
+ <type name="Gst.BinClass" c:type="GstBinClass"/>
+ </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="WebRTCDTLSTransportState"
+ glib:type-name="GstWebRTCDTLSTransportState"
+ glib:get-type="gst_webrtc_dtls_transport_state_get_type"
+ c:type="GstWebRTCDTLSTransportState">
+ <doc xml:space="preserve">GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
+GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
+GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
+GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
+GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
+ <member name="new"
+ value="0"
+ c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW"
+ glib:nick="new">
+ </member>
+ <member name="closed"
+ value="1"
+ c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED"
+ glib:nick="closed">
+ </member>
+ <member name="failed"
+ value="2"
+ c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED"
+ glib:nick="failed">
+ </member>
+ <member name="connecting"
+ value="3"
+ c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING"
+ glib:nick="connecting">
+ </member>
+ <member name="connected"
+ value="4"
+ c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED"
+ glib:nick="connected">
+ </member>
+ </enumeration>
+ <enumeration name="WebRTCFECType"
+ glib:type-name="GstWebRTCFECType"
+ glib:get-type="gst_webrtc_fec_type_get_type"
+ c:type="GstWebRTCFECType">
+ <doc xml:space="preserve">GST_WEBRTC_FEC_TYPE_NONE: none
+GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red</doc>
+ <member name="none"
+ value="0"
+ c:identifier="GST_WEBRTC_FEC_TYPE_NONE"
+ glib:nick="none">
+ </member>
+ <member name="ulp_red"
+ value="1"
+ c:identifier="GST_WEBRTC_FEC_TYPE_ULP_RED"
+ glib:nick="ulp-red">
+ </member>
+ </enumeration>
+ <enumeration name="WebRTCICEComponent"
+ glib:type-name="GstWebRTCICEComponent"
+ glib:get-type="gst_webrtc_ice_component_get_type"
+ c:type="GstWebRTCICEComponent">
+ <doc xml:space="preserve">GST_WEBRTC_ICE_COMPONENT_RTP,
+GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
+ <member name="rtp"
+ value="0"
+ c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP"
+ glib:nick="rtp">
+ </member>
+ <member name="rtcp"
+ value="1"
+ c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP"
+ glib:nick="rtcp">
+ </member>
+ </enumeration>
+ <enumeration name="WebRTCICEConnectionState"
+ glib:type-name="GstWebRTCICEConnectionState"
+ glib:get-type="gst_webrtc_ice_connection_state_get_type"
+ c:type="GstWebRTCICEConnectionState">
+ <doc xml:space="preserve">GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
+GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
+GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
+GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
+GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
+GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
+GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
+See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink></doc>
+ <member name="new"
+ value="0"
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW"
+ glib:nick="new">
+ </member>
+ <member name="checking"
+ value="1"
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING"
+ glib:nick="checking">
+ </member>
+ <member name="connected"
+ value="2"
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED"
+ glib:nick="connected">
+ </member>
+ <member name="completed"
+ value="3"
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED"
+ glib:nick="completed">
+ </member>
+ <member name="failed"
+ value="4"
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED"
+ glib:nick="failed">
+ </member>
+ <member name="disconnected"
+ value="5"
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED"
+ glib:nick="disconnected">
+ </member>
+ <member name="closed"
+ value="6"
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED"
+ glib:nick="closed">
+ </member>
+ </enumeration>
+ <enumeration name="WebRTCICEGatheringState"
+ glib:type-name="GstWebRTCICEGatheringState"
+ glib:get-type="gst_webrtc_ice_gathering_state_get_type"
+ c:type="GstWebRTCICEGatheringState">
+ <doc xml:space="preserve">GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
+GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
+GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
+See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink></doc>
+ <member name="new"
+ value="0"
+ c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW"
+ glib:nick="new">
+ </member>
+ <member name="gathering"
+ value="1"
+ c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING"
+ glib:nick="gathering">
+ </member>
+ <member name="complete"
+ value="2"
+ c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE"
+ glib:nick="complete">
+ </member>
+ </enumeration>
+ <enumeration name="WebRTCICERole"
+ glib:type-name="GstWebRTCICERole"
+ glib:get-type="gst_webrtc_ice_role_get_type"
+ c:type="GstWebRTCICERole">
+ <doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
+GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
+ <member name="controlled"
+ value="0"
+ c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED"
+ glib:nick="controlled">
+ </member>
+ <member name="controlling"
+ value="1"
+ c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING"
+ glib:nick="controlling">
+ </member>
+ </enumeration>
+ <class name="WebRTCICETransport"
+ c:symbol-prefix="webrtc_ice_transport"
+ c:type="GstWebRTCICETransport"
+ parent="Gst.Object"
+ abstract="1"
+ glib:type-name="GstWebRTCICETransport"
+ glib:get-type="gst_webrtc_ice_transport_get_type"
+ glib:type-struct="WebRTCICETransportClass">
+ <virtual-method name="gather_candidates">
+ <return-value transfer-ownership="none">
+ <type name="gboolean" c:type="gboolean"/>
+ </return-value>
+ <parameters>
+ <instance-parameter name="transport" transfer-ownership="none">
+ <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+ </instance-parameter>
+ </parameters>
+ </virtual-method>
+ <method name="connection_state_change"
+ c:identifier="gst_webrtc_ice_transport_connection_state_change">
+ <return-value transfer-ownership="none">
+ <type name="none" c:type="void"/>
+ </return-value>
+ <parameters>
+ <instance-parameter name="ice" transfer-ownership="none">
+ <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+ </instance-parameter>
+ <parameter name="new_state" transfer-ownership="none">
+ <type name="WebRTCICEConnectionState"
+ c:type="GstWebRTCICEConnectionState"/>
+ </parameter>
+ </parameters>
+ </method>
+ <method name="gathering_state_change"
+ c:identifier="gst_webrtc_ice_transport_gathering_state_change">
+ <return-value transfer-ownership="none">
+ <type name="none" c:type="void"/>
+ </return-value>
+ <parameters>
+ <instance-parameter name="ice" transfer-ownership="none">
+ <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+ </instance-parameter>
+ <parameter name="new_state" transfer-ownership="none">
+ <type name="WebRTCICEGatheringState"
+ c:type="GstWebRTCICEGatheringState"/>
+ </parameter>
+ </parameters>
+ </method>
+ <method name="new_candidate"
+ c:identifier="gst_webrtc_ice_transport_new_candidate">
+ <return-value transfer-ownership="none">
+ <type name="none" c:type="void"/>
+ </return-value>
+ <parameters>
+ <instance-parameter name="ice" transfer-ownership="none">
+ <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+ </instance-parameter>
+ <parameter name="stream_id" transfer-ownership="none">
+ <type name="guint" c:type="guint"/>
+ </parameter>
+ <parameter name="component" transfer-ownership="none">
+ <type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/>
+ </parameter>
+ <parameter name="attr" transfer-ownership="none">
+ <type name="utf8" c:type="gchar*"/>
+ </parameter>
+ </parameters>
+ </method>
+ <method name="selected_pair_change"
+ c:identifier="gst_webrtc_ice_transport_selected_pair_change">
+ <return-value transfer-ownership="none">
+ <type name="none" c:type="void"/>
+ </return-value>
+ <parameters>
+ <instance-parameter name="ice" transfer-ownership="none">
+ <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+ </instance-parameter>
+ </parameters>
+ </method>
+ <property name="component"
+ writable="1"
+ construct-only="1"
+ transfer-ownership="none">
+ <type name="WebRTCICEComponent"/>
+ </property>
+ <property name="gathering-state" transfer-ownership="none">
+ <type name="WebRTCICEGatheringState"/>
+ </property>
+ <property name="state" transfer-ownership="none">
+ <type name="WebRTCICEConnectionState"/>
+ </property>
+ <field name="parent">
+ <type name="Gst.Object" c:type="GstObject"/>
+ </field>
+ <field name="role">
+ <type name="WebRTCICERole" c:type="GstWebRTCICERole"/>
+ </field>
+ <field name="component">
+ <type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/>
+ </field>
+ <field name="state">
+ <type name="WebRTCICEConnectionState"
+ c:type="GstWebRTCICEConnectionState"/>
+ </field>
+ <field name="gathering_state">
+ <type name="WebRTCICEGatheringState"
+ c:type="GstWebRTCICEGatheringState"/>
+ </field>
+ <field name="src">
+ <type name="Gst.Element" c:type="GstElement*"/>
+ </field>
+ <field name="sink">
+ <type name="Gst.Element" c:type="GstElement*"/>
+ </field>
+ <field name="_padding">
+ <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+ <type name="gpointer" c:type="gpointer"/>
+ </array>
+ </field>
+ <glib:signal name="on-new-candidate" when="last">
+ <return-value transfer-ownership="none">
+ <type name="none" c:type="void"/>
+ </return-value>
+ <parameters>
+ <parameter name="object" transfer-ownership="none">
+ <type name="utf8" c:type="gchar*"/>
+ </parameter>
+ </parameters>
+ </glib:signal>
+ <glib:signal name="on-selected-candidate-pair-change" when="last">
+ <return-value transfer-ownership="none">
+ <type name="none" c:type="void"/>
+ </return-value>
+ </glib:signal>
+ </class>
+ <record name="WebRTCICETransportClass"
+ c:type="GstWebRTCICETransportClass"
+ glib:is-gtype-struct-for="WebRTCICETransport">
+ <field name="parent_class">
+ <type name="Gst.BinClass" c:type="GstBinClass"/>
+ </field>
+ <field name="gather_candidates">
+ <callback name="gather_candidates">
+ <return-value transfer-ownership="none">
+ <type name="gboolean" c:type="gboolean"/>
+ </return-value>
+ <parameters>
+ <parameter name="transport" transfer-ownership="none">
+ <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+ </parameter>
+ </parameters>
+ </callback>
+ </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="WebRTCPeerConnectionState"
+ glib:type-name="GstWebRTCPeerConnectionState"
+ glib:get-type="gst_webrtc_peer_connection_state_get_type"
+ c:type="GstWebRTCPeerConnectionState">
+ <doc xml:space="preserve">GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
+GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
+GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
+GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
+GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
+GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
+See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink></doc>
+ <member name="new"
+ value="0"
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW"
+ glib:nick="new">
+ </member>
+ <member name="connecting"
+ value="1"
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING"
+ glib:nick="connecting">
+ </member>
+ <member name="connected"
+ value="2"
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED"
+ glib:nick="connected">
+ </member>
+ <member name="disconnected"
+ value="3"
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED"
+ glib:nick="disconnected">
+ </member>
+ <member name="failed"
+ value="4"
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED"
+ glib:nick="failed">
+ </member>
+ <member name="closed"
+ value="5"
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED"
+ glib:nick="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"
+ glib:type-name="GstWebRTCRTPTransceiverDirection"
+ glib:get-type="gst_webrtc_rtp_transceiver_direction_get_type"
+ c:type="GstWebRTCRTPTransceiverDirection">
+ <member name="none"
+ value="0"
+ c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE"
+ glib:nick="none">
+ </member>
+ <member name="inactive"
+ value="1"
+ c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE"
+ glib:nick="inactive">
+ </member>
+ <member name="sendonly"
+ value="2"
+ c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY"
+ glib:nick="sendonly">
+ </member>
+ <member name="recvonly"
+ value="3"
+ c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY"
+ glib:nick="recvonly">
+ </member>
+ <member name="sendrecv"
+ value="4"
+ c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV"
+ glib:nick="sendrecv">
+ </member>
+ </enumeration>
+ <enumeration name="WebRTCSDPType"
+ glib:type-name="GstWebRTCSDPType"
+ glib:get-type="gst_webrtc_sdp_type_get_type"
+ 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"
+ glib:nick="offer">
+ </member>
+ <member name="pranswer"
+ value="2"
+ c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER"
+ glib:nick="pranswer">
+ </member>
+ <member name="answer"
+ value="3"
+ c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER"
+ glib:nick="answer">
+ </member>
+ <member name="rollback"
+ value="4"
+ c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK"
+ glib:nick="rollback">
+ </member>
+ <function name="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>
+ </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">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">
+ <doc xml:space="preserve">the #GstWebRTCSDPType of the description</doc>
+ <type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
+ </field>
+ <field name="sdp" writable="1">
+ <doc xml:space="preserve">the #GstSDPMessage of the description</doc>
+ <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"
+ glib:type-name="GstWebRTCSignalingState"
+ glib:get-type="gst_webrtc_signaling_state_get_type"
+ 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"
+ glib:nick="stable">
+ </member>
+ <member name="closed"
+ value="1"
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED"
+ glib:nick="closed">
+ </member>
+ <member name="have_local_offer"
+ value="2"
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER"
+ glib:nick="have-local-offer">
+ </member>
+ <member name="have_remote_offer"
+ value="3"
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER"
+ glib:nick="have-remote-offer">
+ </member>
+ <member name="have_local_pranswer"
+ value="4"
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER"
+ glib:nick="have-local-pranswer">
+ </member>
+ <member name="have_remote_pranswer"
+ value="5"
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER"
+ glib:nick="have-remote-pranswer">
+ </member>
+ </enumeration>
+ <enumeration name="WebRTCStatsType"
+ glib:type-name="GstWebRTCStatsType"
+ glib:get-type="gst_webrtc_stats_type_get_type"
+ 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"
+ glib:nick="codec">
+ </member>
+ <member name="inbound_rtp"
+ value="2"
+ c:identifier="GST_WEBRTC_STATS_INBOUND_RTP"
+ glib:nick="inbound-rtp">
+ </member>
+ <member name="outbound_rtp"
+ value="3"
+ c:identifier="GST_WEBRTC_STATS_OUTBOUND_RTP"
+ glib:nick="outbound-rtp">
+ </member>
+ <member name="remote_inbound_rtp"
+ value="4"
+ c:identifier="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP"
+ glib:nick="remote-inbound-rtp">
+ </member>
+ <member name="remote_outbound_rtp"
+ value="5"
+ c:identifier="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP"
+ glib:nick="remote-outbound-rtp">
+ </member>
+ <member name="csrc"
+ value="6"
+ c:identifier="GST_WEBRTC_STATS_CSRC"
+ glib:nick="csrc">
+ </member>
+ <member name="peer_connection"
+ value="7"
+ c:identifier="GST_WEBRTC_STATS_PEER_CONNECTION"
+ glib:nick="peer-connection">
+ </member>
+ <member name="data_channel"
+ value="8"
+ c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL"
+ glib:nick="data-channel">
+ </member>
+ <member name="stream"
+ value="9"
+ c:identifier="GST_WEBRTC_STATS_STREAM"
+ glib:nick="stream">
+ </member>
+ <member name="transport"
+ value="10"
+ c:identifier="GST_WEBRTC_STATS_TRANSPORT"
+ glib:nick="transport">
+ </member>
+ <member name="candidate_pair"
+ value="11"
+ c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR"
+ glib:nick="candidate-pair">
+ </member>
+ <member name="local_candidate"
+ value="12"
+ c:identifier="GST_WEBRTC_STATS_LOCAL_CANDIDATE"
+ glib:nick="local-candidate">
+ </member>
+ <member name="remote_candidate"
+ value="13"
+ c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE"
+ glib:nick="remote-candidate">
+ </member>
+ <member name="certificate"
+ value="14"
+ c:identifier="GST_WEBRTC_STATS_CERTIFICATE"
+ glib:nick="certificate">
+ </member>
+ </enumeration>
+ <function name="webrtc_sdp_type_to_string"
+ c:identifier="gst_webrtc_sdp_type_to_string"
+ moved-to="WebRTCSDPType.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>
['gstreamer-rtsp', ['gst-plugins-base', 'rtsp_dep'], 'gst_rtsp'],
['gstreamer-sdp', ['gst-plugins-base', 'sdp_dep'], 'gstsdp'],
['gstreamer-tag', ['gst-plugins-base', 'tag_dep'], 'gsttag'],
- ['gstreamer-video', ['gst-plugins-base', 'video_dep'], 'gstvideo'],]
+ ['gstreamer-video', ['gst-plugins-base', 'video_dep'], 'gstvideo'],
+ ['gstreamer-webrtc', ['gst-plugins-bad', 'gstwebrtc_dep'], 'gstwebrtc'],
+]
foreach dep: gst_deps_defs
gst_deps += [dependency(dep.get(0) + '-' + apiversion, version: gst_required_version,
run_target('bindinate_gstreamer',
command: [bindinate,
'--name=gstreamer', '--regenerate=true',
- '--merge-with=GstApp-1.0,GstAudio-1.0,GstBase-1.0,GstController-1.0,GstNet-1.0,GstPbutils-1.0,GstRtp-1.0,GstRtsp-1.0,GstSdp-1.0,GstTag-1.0,GstVideo-1.0',
+ '--merge-with=GstApp-1.0,GstAudio-1.0,GstBase-1.0,GstController-1.0,GstNet-1.0,GstPbutils-1.0,GstRtp-1.0,GstRtsp-1.0,GstSdp-1.0,GstTag-1.0,GstVideo-1.0,GstWebRTC-1.0',
'--gir=Gst-1.0',
'--copy-girs=@0@'.format(join_paths(meson.current_source_dir(), 'girs'))],
depends: []
run_target('update-all', command: [find_program('update_sources.py'), 'bindinate'])
else
warning('Bindinator not usable as some required dependencies are not avalaible.')
-endif
\ No newline at end of file
+endif
GLib.GType.Register (FractionRange.GType, typeof(FractionRange));
GLib.GType.Register (DateTime.GType, typeof(DateTime));
GLib.GType.Register (Gst.Array.GType, typeof(Gst.Array));
-
+ GLib.GType.Register(Promise.GType, typeof(Promise));
+ GLib.GType.Register(Gst.WebRTC.WebRTCSessionDescription.GType, typeof(Gst.WebRTC.WebRTCSessionDescription));
}
[DllImport("libgstreamer-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Collections;
+ using System.Collections.Generic;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ public partial class Constants {
+
+#endregion
+ }
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Collections;
+ using System.Collections.Generic;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ public partial class Global {
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_sdp_type_to_string(int type);
+
+ public static string WebrtcSdpTypeToString(Gst.WebRTC.WebRTCSDPType type) {
+ IntPtr raw_ret = gst_webrtc_sdp_type_to_string((int) type);
+ string ret = GLib.Marshaller.Utf8PtrToString (raw_ret);
+ return ret;
+ }
+
+#endregion
+ }
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+
+ public delegate void OnNewCandidateHandler(object o, OnNewCandidateArgs args);
+
+ public class OnNewCandidateArgs : GLib.SignalArgs {
+ public string Object{
+ get {
+ return (string) Args [0];
+ }
+ }
+
+ }
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCDTLSSetupGType))]
+ public enum WebRTCDTLSSetup {
+
+ None = 0,
+ Actpass = 1,
+ Active = 2,
+ Passive = 3,
+ }
+
+ internal class WebRTCDTLSSetupGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_dtls_setup_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_dtls_setup_get_type ());
+ }
+ }
+ }
+#endregion
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Collections;
+ using System.Collections.Generic;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ public partial class WebRTCDTLSTransport : Gst.Object {
+
+ public WebRTCDTLSTransport (IntPtr raw) : base(raw) {}
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_dtls_transport_new(uint session_id, bool rtcp);
+
+ public WebRTCDTLSTransport (uint session_id, bool rtcp) : base (IntPtr.Zero)
+ {
+ if (GetType () != typeof (WebRTCDTLSTransport)) {
+ var vals = new List<GLib.Value> ();
+ var names = new List<string> ();
+ names.Add ("session_id");
+ vals.Add (new GLib.Value (session_id));
+ names.Add ("rtcp");
+ vals.Add (new GLib.Value (rtcp));
+ CreateNativeObject (names.ToArray (), vals.ToArray ());
+ return;
+ }
+ Raw = gst_webrtc_dtls_transport_new(session_id, rtcp);
+ }
+
+ [GLib.Property ("certificate")]
+ public string Certificate {
+ get {
+ GLib.Value val = GetProperty ("certificate");
+ string ret = (string) val;
+ val.Dispose ();
+ return ret;
+ }
+ set {
+ GLib.Value val = new GLib.Value(value);
+ SetProperty("certificate", val);
+ val.Dispose ();
+ }
+ }
+
+ [GLib.Property ("client")]
+ public bool Client {
+ get {
+ GLib.Value val = GetProperty ("client");
+ bool ret = (bool) val;
+ val.Dispose ();
+ return ret;
+ }
+ set {
+ GLib.Value val = new GLib.Value(value);
+ SetProperty("client", val);
+ val.Dispose ();
+ }
+ }
+
+ [GLib.Property ("remote-certificate")]
+ public string RemoteCertificate {
+ get {
+ GLib.Value val = GetProperty ("remote-certificate");
+ string ret = (string) val;
+ val.Dispose ();
+ return ret;
+ }
+ }
+
+ [GLib.Property ("rtcp")]
+ public bool Rtcp {
+ get {
+ GLib.Value val = GetProperty ("rtcp");
+ bool ret = (bool) val;
+ val.Dispose ();
+ return ret;
+ }
+ }
+
+ [GLib.Property ("session-id")]
+ public uint SessionId {
+ get {
+ GLib.Value val = GetProperty ("session-id");
+ uint ret = (uint) val;
+ val.Dispose ();
+ return ret;
+ }
+ }
+
+ [GLib.Property ("state")]
+ public Gst.WebRTC.WebRTCDTLSTransportState State {
+ get {
+ GLib.Value val = GetProperty ("state");
+ Gst.WebRTC.WebRTCDTLSTransportState ret = (Gst.WebRTC.WebRTCDTLSTransportState) (Enum) val;
+ val.Dispose ();
+ return ret;
+ }
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern void gst_webrtc_dtls_transport_set_transport(IntPtr raw, IntPtr ice);
+
+ [GLib.Property ("transport")]
+ public Gst.WebRTC.WebRTCICETransport Transport {
+ get {
+ GLib.Value val = GetProperty ("transport");
+ Gst.WebRTC.WebRTCICETransport ret = (Gst.WebRTC.WebRTCICETransport) val;
+ val.Dispose ();
+ return ret;
+ }
+ set {
+ gst_webrtc_dtls_transport_set_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
+ }
+ }
+
+ public Gst.WebRTC.WebRTCICETransport TransportField {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("transport"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCICETransport;
+ }
+ }
+ }
+
+ public Gst.WebRTC.WebRTCDTLSTransportState StateField {
+ get {
+ unsafe {
+ int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("state"));
+ return (Gst.WebRTC.WebRTCDTLSTransportState) (*raw_ptr);
+ }
+ }
+ }
+
+ public bool IsRtcp {
+ get {
+ unsafe {
+ bool* raw_ptr = (bool*)(((byte*)Handle) + abi_info.GetFieldOffset("is_rtcp"));
+ return (*raw_ptr);
+ }
+ }
+ }
+
+ public bool ClientField {
+ get {
+ unsafe {
+ bool* raw_ptr = (bool*)(((byte*)Handle) + abi_info.GetFieldOffset("client"));
+ return (*raw_ptr);
+ }
+ }
+ }
+
+ public uint SessionIdField {
+ get {
+ unsafe {
+ uint* raw_ptr = (uint*)(((byte*)Handle) + abi_info.GetFieldOffset("session_id"));
+ return (*raw_ptr);
+ }
+ }
+ }
+
+ public Gst.Element Dtlssrtpenc {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("dtlssrtpenc"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
+ }
+ }
+ }
+
+ public Gst.Element Dtlssrtpdec {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("dtlssrtpdec"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
+ }
+ }
+ }
+
+
+ // Internal representation of the wrapped structure ABI.
+ static GLib.AbiStruct _class_abi = null;
+ static public new GLib.AbiStruct class_abi {
+ get {
+ if (_class_abi == null)
+ _class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{
+ new GLib.AbiField("_padding"
+ , Gst.Object.class_abi.Fields
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+ , null
+ , null
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ });
+
+ return _class_abi;
+ }
+ }
+
+
+ // End of the ABI representation.
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_dtls_transport_get_type();
+
+ public static new GLib.GType GType {
+ get {
+ IntPtr raw_ret = gst_webrtc_dtls_transport_get_type();
+ GLib.GType ret = new GLib.GType(raw_ret);
+ return ret;
+ }
+ }
+
+
+ static WebRTCDTLSTransport ()
+ {
+ GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
+ }
+
+ // Internal representation of the wrapped structure ABI.
+ static GLib.AbiStruct _abi_info = null;
+ static public new GLib.AbiStruct abi_info {
+ get {
+ if (_abi_info == null)
+ _abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{
+ new GLib.AbiField("transport"
+ , Gst.Object.abi_info.Fields
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // transport
+ , null
+ , "state"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("state"
+ , -1
+ , (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCDTLSTransportState))) // state
+ , "transport"
+ , "is_rtcp"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_stateAlign), "state")
+ , 0
+ ),
+ new GLib.AbiField("is_rtcp"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(bool)) // is_rtcp
+ , "state"
+ , "client"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_is_rtcpAlign), "is_rtcp")
+ , 0
+ ),
+ new GLib.AbiField("client"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(bool)) // client
+ , "is_rtcp"
+ , "session_id"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_clientAlign), "client")
+ , 0
+ ),
+ new GLib.AbiField("session_id"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(uint)) // session_id
+ , "client"
+ , "dtlssrtpenc"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_session_idAlign), "session_id")
+ , 0
+ ),
+ new GLib.AbiField("dtlssrtpenc"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // dtlssrtpenc
+ , "session_id"
+ , "dtlssrtpdec"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("dtlssrtpdec"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // dtlssrtpdec
+ , "dtlssrtpenc"
+ , "_padding"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("_padding"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+ , "dtlssrtpdec"
+ , null
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ });
+
+ return _abi_info;
+ }
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCDTLSTransport_stateAlign
+ {
+ sbyte f1;
+ private Gst.WebRTC.WebRTCDTLSTransportState state;
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCDTLSTransport_is_rtcpAlign
+ {
+ sbyte f1;
+ private bool is_rtcp;
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCDTLSTransport_clientAlign
+ {
+ sbyte f1;
+ private bool client;
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCDTLSTransport_session_idAlign
+ {
+ sbyte f1;
+ private uint session_id;
+ }
+
+
+ // End of the ABI representation.
+
+#endregion
+ }
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCDTLSTransportStateGType))]
+ public enum WebRTCDTLSTransportState {
+
+ New = 0,
+ Closed = 1,
+ Failed = 2,
+ Connecting = 3,
+ Connected = 4,
+ }
+
+ internal class WebRTCDTLSTransportStateGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_dtls_transport_state_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_dtls_transport_state_get_type ());
+ }
+ }
+ }
+#endregion
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCICEComponentGType))]
+ public enum WebRTCICEComponent {
+
+ Rtp = 0,
+ Rtcp = 1,
+ }
+
+ internal class WebRTCICEComponentGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_ice_component_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_ice_component_get_type ());
+ }
+ }
+ }
+#endregion
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCICEConnectionStateGType))]
+ public enum WebRTCICEConnectionState {
+
+ New = 0,
+ Checking = 1,
+ Connected = 2,
+ Completed = 3,
+ Failed = 4,
+ Disconnected = 5,
+ Closed = 6,
+ }
+
+ internal class WebRTCICEConnectionStateGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_ice_connection_state_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_ice_connection_state_get_type ());
+ }
+ }
+ }
+#endregion
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCICEGatheringStateGType))]
+ public enum WebRTCICEGatheringState {
+
+ New = 0,
+ Gathering = 1,
+ Complete = 2,
+ }
+
+ internal class WebRTCICEGatheringStateGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_ice_gathering_state_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_ice_gathering_state_get_type ());
+ }
+ }
+ }
+#endregion
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCICERoleGType))]
+ public enum WebRTCICERole {
+
+ Controlled = 0,
+ Controlling = 1,
+ }
+
+ internal class WebRTCICERoleGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_ice_role_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_ice_role_get_type ());
+ }
+ }
+ }
+#endregion
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Collections;
+ using System.Collections.Generic;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ public partial class WebRTCICETransport : Gst.Object {
+
+ protected WebRTCICETransport (IntPtr raw) : base(raw) {}
+
+ protected WebRTCICETransport() : base(IntPtr.Zero)
+ {
+ CreateNativeObject (new string [0], new GLib.Value [0]);
+ }
+
+ [GLib.Property ("component")]
+ public Gst.WebRTC.WebRTCICEComponent Component {
+ get {
+ GLib.Value val = GetProperty ("component");
+ Gst.WebRTC.WebRTCICEComponent ret = (Gst.WebRTC.WebRTCICEComponent) (Enum) val;
+ val.Dispose ();
+ return ret;
+ }
+ }
+
+ [GLib.Property ("gathering-state")]
+ public Gst.WebRTC.WebRTCICEGatheringState GatheringState {
+ get {
+ GLib.Value val = GetProperty ("gathering-state");
+ Gst.WebRTC.WebRTCICEGatheringState ret = (Gst.WebRTC.WebRTCICEGatheringState) (Enum) val;
+ val.Dispose ();
+ return ret;
+ }
+ }
+
+ [GLib.Property ("state")]
+ public Gst.WebRTC.WebRTCICEConnectionState State {
+ get {
+ GLib.Value val = GetProperty ("state");
+ Gst.WebRTC.WebRTCICEConnectionState ret = (Gst.WebRTC.WebRTCICEConnectionState) (Enum) val;
+ val.Dispose ();
+ return ret;
+ }
+ }
+
+ public Gst.WebRTC.WebRTCICERole Role {
+ get {
+ unsafe {
+ int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("role"));
+ return (Gst.WebRTC.WebRTCICERole) (*raw_ptr);
+ }
+ }
+ }
+
+ public Gst.WebRTC.WebRTCICEComponent ComponentField {
+ get {
+ unsafe {
+ int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("component"));
+ return (Gst.WebRTC.WebRTCICEComponent) (*raw_ptr);
+ }
+ }
+ }
+
+ public Gst.WebRTC.WebRTCICEConnectionState StateField {
+ get {
+ unsafe {
+ int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("state"));
+ return (Gst.WebRTC.WebRTCICEConnectionState) (*raw_ptr);
+ }
+ }
+ }
+
+ public Gst.WebRTC.WebRTCICEGatheringState GatheringStateField {
+ get {
+ unsafe {
+ int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("gathering_state"));
+ return (Gst.WebRTC.WebRTCICEGatheringState) (*raw_ptr);
+ }
+ }
+ }
+
+ public Gst.Element Src {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("src"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
+ }
+ }
+ }
+
+ public Gst.Element Sink {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("sink"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
+ }
+ }
+ }
+
+ [GLib.Signal("on-selected-candidate-pair-change")]
+ public event System.EventHandler OnSelectedCandidatePairChange {
+ add {
+ this.AddSignalHandler ("on-selected-candidate-pair-change", value);
+ }
+ remove {
+ this.RemoveSignalHandler ("on-selected-candidate-pair-change", value);
+ }
+ }
+
+ [GLib.Signal("on-new-candidate")]
+ public event Gst.WebRTC.OnNewCandidateHandler OnNewCandidate {
+ add {
+ this.AddSignalHandler ("on-new-candidate", value, typeof (Gst.WebRTC.OnNewCandidateArgs));
+ }
+ remove {
+ this.RemoveSignalHandler ("on-new-candidate", value);
+ }
+ }
+
+ static OnNewCandidateNativeDelegate OnNewCandidate_cb_delegate;
+ static OnNewCandidateNativeDelegate OnNewCandidateVMCallback {
+ get {
+ if (OnNewCandidate_cb_delegate == null)
+ OnNewCandidate_cb_delegate = new OnNewCandidateNativeDelegate (OnNewCandidate_cb);
+ return OnNewCandidate_cb_delegate;
+ }
+ }
+
+ static void OverrideOnNewCandidate (GLib.GType gtype)
+ {
+ OverrideOnNewCandidate (gtype, OnNewCandidateVMCallback);
+ }
+
+ static void OverrideOnNewCandidate (GLib.GType gtype, OnNewCandidateNativeDelegate callback)
+ {
+ OverrideVirtualMethod (gtype, "on-new-candidate", callback);
+ }
+ [UnmanagedFunctionPointer (CallingConvention.Cdecl)]
+ delegate void OnNewCandidateNativeDelegate (IntPtr inst, IntPtr _object);
+
+ static void OnNewCandidate_cb (IntPtr inst, IntPtr _object)
+ {
+ try {
+ WebRTCICETransport __obj = GLib.Object.GetObject (inst, false) as WebRTCICETransport;
+ __obj.OnOnNewCandidate (GLib.Marshaller.Utf8PtrToString (_object));
+ } catch (Exception e) {
+ GLib.ExceptionManager.RaiseUnhandledException (e, false);
+ }
+ }
+
+ [GLib.DefaultSignalHandler(Type=typeof(Gst.WebRTC.WebRTCICETransport), ConnectionMethod="OverrideOnNewCandidate")]
+ protected virtual void OnOnNewCandidate (string _object)
+ {
+ InternalOnNewCandidate (_object);
+ }
+
+ private void InternalOnNewCandidate (string _object)
+ {
+ GLib.Value ret = GLib.Value.Empty;
+ GLib.ValueArray inst_and_params = new GLib.ValueArray (2);
+ GLib.Value[] vals = new GLib.Value [2];
+ vals [0] = new GLib.Value (this);
+ inst_and_params.Append (vals [0]);
+ vals [1] = new GLib.Value (_object);
+ inst_and_params.Append (vals [1]);
+ g_signal_chain_from_overridden (inst_and_params.ArrayPtr, ref ret);
+ foreach (GLib.Value v in vals)
+ v.Dispose ();
+ }
+
+ static OnSelectedCandidatePairChangeNativeDelegate OnSelectedCandidatePairChange_cb_delegate;
+ static OnSelectedCandidatePairChangeNativeDelegate OnSelectedCandidatePairChangeVMCallback {
+ get {
+ if (OnSelectedCandidatePairChange_cb_delegate == null)
+ OnSelectedCandidatePairChange_cb_delegate = new OnSelectedCandidatePairChangeNativeDelegate (OnSelectedCandidatePairChange_cb);
+ return OnSelectedCandidatePairChange_cb_delegate;
+ }
+ }
+
+ static void OverrideOnSelectedCandidatePairChange (GLib.GType gtype)
+ {
+ OverrideOnSelectedCandidatePairChange (gtype, OnSelectedCandidatePairChangeVMCallback);
+ }
+
+ static void OverrideOnSelectedCandidatePairChange (GLib.GType gtype, OnSelectedCandidatePairChangeNativeDelegate callback)
+ {
+ OverrideVirtualMethod (gtype, "on-selected-candidate-pair-change", callback);
+ }
+ [UnmanagedFunctionPointer (CallingConvention.Cdecl)]
+ delegate void OnSelectedCandidatePairChangeNativeDelegate (IntPtr inst);
+
+ static void OnSelectedCandidatePairChange_cb (IntPtr inst)
+ {
+ try {
+ WebRTCICETransport __obj = GLib.Object.GetObject (inst, false) as WebRTCICETransport;
+ __obj.OnOnSelectedCandidatePairChange ();
+ } catch (Exception e) {
+ GLib.ExceptionManager.RaiseUnhandledException (e, false);
+ }
+ }
+
+ [GLib.DefaultSignalHandler(Type=typeof(Gst.WebRTC.WebRTCICETransport), ConnectionMethod="OverrideOnSelectedCandidatePairChange")]
+ protected virtual void OnOnSelectedCandidatePairChange ()
+ {
+ InternalOnSelectedCandidatePairChange ();
+ }
+
+ private void InternalOnSelectedCandidatePairChange ()
+ {
+ GLib.Value ret = GLib.Value.Empty;
+ GLib.ValueArray inst_and_params = new GLib.ValueArray (1);
+ GLib.Value[] vals = new GLib.Value [1];
+ vals [0] = new GLib.Value (this);
+ inst_and_params.Append (vals [0]);
+ g_signal_chain_from_overridden (inst_and_params.ArrayPtr, ref ret);
+ foreach (GLib.Value v in vals)
+ v.Dispose ();
+ }
+
+ static GatherCandidatesNativeDelegate GatherCandidates_cb_delegate;
+ static GatherCandidatesNativeDelegate GatherCandidatesVMCallback {
+ get {
+ if (GatherCandidates_cb_delegate == null)
+ GatherCandidates_cb_delegate = new GatherCandidatesNativeDelegate (GatherCandidates_cb);
+ return GatherCandidates_cb_delegate;
+ }
+ }
+
+ static void OverrideGatherCandidates (GLib.GType gtype)
+ {
+ OverrideGatherCandidates (gtype, GatherCandidatesVMCallback);
+ }
+
+ static void OverrideGatherCandidates (GLib.GType gtype, GatherCandidatesNativeDelegate callback)
+ {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((long) gtype.GetClassPtr()) + (long) class_abi.GetFieldOffset("gather_candidates"));
+ *raw_ptr = Marshal.GetFunctionPointerForDelegate((Delegate) callback);
+ }
+ }
+
+ [UnmanagedFunctionPointer (CallingConvention.Cdecl)]
+ delegate bool GatherCandidatesNativeDelegate (IntPtr inst);
+
+ static bool GatherCandidates_cb (IntPtr inst)
+ {
+ try {
+ WebRTCICETransport __obj = GLib.Object.GetObject (inst, false) as WebRTCICETransport;
+ bool __result;
+ __result = __obj.OnGatherCandidates ();
+ return __result;
+ } catch (Exception e) {
+ GLib.ExceptionManager.RaiseUnhandledException (e, true);
+ // NOTREACHED: above call does not return.
+ throw e;
+ }
+ }
+
+ [GLib.DefaultSignalHandler(Type=typeof(Gst.WebRTC.WebRTCICETransport), ConnectionMethod="OverrideGatherCandidates")]
+ protected virtual bool OnGatherCandidates ()
+ {
+ return InternalGatherCandidates ();
+ }
+
+ private bool InternalGatherCandidates ()
+ {
+ GatherCandidatesNativeDelegate unmanaged = null;
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((long) this.LookupGType().GetThresholdType().GetClassPtr()) + (long) class_abi.GetFieldOffset("gather_candidates"));
+ unmanaged = (GatherCandidatesNativeDelegate) Marshal.GetDelegateForFunctionPointer(*raw_ptr, typeof(GatherCandidatesNativeDelegate));
+ }
+ if (unmanaged == null) return false;
+
+ bool __result = unmanaged (this.Handle);
+ return __result;
+ }
+
+
+ // Internal representation of the wrapped structure ABI.
+ static GLib.AbiStruct _class_abi = null;
+ static public new GLib.AbiStruct class_abi {
+ get {
+ if (_class_abi == null)
+ _class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{
+ new GLib.AbiField("gather_candidates"
+ , Gst.Object.class_abi.Fields
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // gather_candidates
+ , null
+ , "_padding"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("_padding"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+ , "gather_candidates"
+ , null
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ });
+
+ return _class_abi;
+ }
+ }
+
+
+ // End of the ABI representation.
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_ice_transport_get_type();
+
+ public static new GLib.GType GType {
+ get {
+ IntPtr raw_ret = gst_webrtc_ice_transport_get_type();
+ GLib.GType ret = new GLib.GType(raw_ret);
+ return ret;
+ }
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern void gst_webrtc_ice_transport_connection_state_change(IntPtr raw, int new_state);
+
+ public void ConnectionStateChange(Gst.WebRTC.WebRTCICEConnectionState new_state) {
+ gst_webrtc_ice_transport_connection_state_change(Handle, (int) new_state);
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern void gst_webrtc_ice_transport_gathering_state_change(IntPtr raw, int new_state);
+
+ public void GatheringStateChange(Gst.WebRTC.WebRTCICEGatheringState new_state) {
+ gst_webrtc_ice_transport_gathering_state_change(Handle, (int) new_state);
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern void gst_webrtc_ice_transport_new_candidate(IntPtr raw, uint stream_id, int component, IntPtr attr);
+
+ public void NewCandidate(uint stream_id, Gst.WebRTC.WebRTCICEComponent component, string attr) {
+ IntPtr native_attr = GLib.Marshaller.StringToPtrGStrdup (attr);
+ gst_webrtc_ice_transport_new_candidate(Handle, stream_id, (int) component, native_attr);
+ GLib.Marshaller.Free (native_attr);
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern void gst_webrtc_ice_transport_selected_pair_change(IntPtr raw);
+
+ public void SelectedPairChange() {
+ gst_webrtc_ice_transport_selected_pair_change(Handle);
+ }
+
+
+ static WebRTCICETransport ()
+ {
+ GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
+ }
+
+ // Internal representation of the wrapped structure ABI.
+ static GLib.AbiStruct _abi_info = null;
+ static public new GLib.AbiStruct abi_info {
+ get {
+ if (_abi_info == null)
+ _abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{
+ new GLib.AbiField("role"
+ , Gst.Object.abi_info.Fields
+ , (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICERole))) // role
+ , null
+ , "component"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_roleAlign), "role")
+ , 0
+ ),
+ new GLib.AbiField("component"
+ , -1
+ , (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICEComponent))) // component
+ , "role"
+ , "state"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_componentAlign), "component")
+ , 0
+ ),
+ new GLib.AbiField("state"
+ , -1
+ , (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICEConnectionState))) // state
+ , "component"
+ , "gathering_state"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_stateAlign), "state")
+ , 0
+ ),
+ new GLib.AbiField("gathering_state"
+ , -1
+ , (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICEGatheringState))) // gathering_state
+ , "state"
+ , "src"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_gathering_stateAlign), "gathering_state")
+ , 0
+ ),
+ new GLib.AbiField("src"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // src
+ , "gathering_state"
+ , "sink"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("sink"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // sink
+ , "src"
+ , "_padding"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("_padding"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+ , "sink"
+ , null
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ });
+
+ return _abi_info;
+ }
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCICETransport_roleAlign
+ {
+ sbyte f1;
+ private Gst.WebRTC.WebRTCICERole role;
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCICETransport_componentAlign
+ {
+ sbyte f1;
+ private Gst.WebRTC.WebRTCICEComponent component;
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCICETransport_stateAlign
+ {
+ sbyte f1;
+ private Gst.WebRTC.WebRTCICEConnectionState state;
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCICETransport_gathering_stateAlign
+ {
+ sbyte f1;
+ private Gst.WebRTC.WebRTCICEGatheringState gathering_state;
+ }
+
+
+ // End of the ABI representation.
+
+#endregion
+ }
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCPeerConnectionStateGType))]
+ public enum WebRTCPeerConnectionState {
+
+ New = 0,
+ Connecting = 1,
+ Connected = 2,
+ Disconnected = 3,
+ Failed = 4,
+ Closed = 5,
+ }
+
+ internal class WebRTCPeerConnectionStateGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_peer_connection_state_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_peer_connection_state_get_type ());
+ }
+ }
+ }
+#endregion
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Collections;
+ using System.Collections.Generic;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ public partial class WebRTCRTPReceiver : Gst.Object {
+
+ public WebRTCRTPReceiver (IntPtr raw) : base(raw) {}
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_rtp_receiver_new();
+
+ public WebRTCRTPReceiver () : base (IntPtr.Zero)
+ {
+ if (GetType () != typeof (WebRTCRTPReceiver)) {
+ CreateNativeObject (new string [0], new GLib.Value[0]);
+ return;
+ }
+ Raw = gst_webrtc_rtp_receiver_new();
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern void gst_webrtc_rtp_receiver_set_transport(IntPtr raw, IntPtr transport);
+
+ public Gst.WebRTC.WebRTCDTLSTransport Transport {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("transport"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
+ }
+ }
+ set {
+ gst_webrtc_rtp_receiver_set_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
+ }
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern void gst_webrtc_rtp_receiver_set_rtcp_transport(IntPtr raw, IntPtr transport);
+
+ public Gst.WebRTC.WebRTCDTLSTransport RtcpTransport {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("rtcp_transport"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
+ }
+ }
+ set {
+ gst_webrtc_rtp_receiver_set_rtcp_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
+ }
+ }
+
+
+ // Internal representation of the wrapped structure ABI.
+ static GLib.AbiStruct _class_abi = null;
+ static public new GLib.AbiStruct class_abi {
+ get {
+ if (_class_abi == null)
+ _class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{
+ new GLib.AbiField("_padding"
+ , Gst.Object.class_abi.Fields
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+ , null
+ , null
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ });
+
+ return _class_abi;
+ }
+ }
+
+
+ // End of the ABI representation.
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_rtp_receiver_get_type();
+
+ public static new GLib.GType GType {
+ get {
+ IntPtr raw_ret = gst_webrtc_rtp_receiver_get_type();
+ GLib.GType ret = new GLib.GType(raw_ret);
+ return ret;
+ }
+ }
+
+
+ static WebRTCRTPReceiver ()
+ {
+ GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
+ }
+
+ // Internal representation of the wrapped structure ABI.
+ static GLib.AbiStruct _abi_info = null;
+ static public new GLib.AbiStruct abi_info {
+ get {
+ if (_abi_info == null)
+ _abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{
+ new GLib.AbiField("transport"
+ , Gst.Object.abi_info.Fields
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // transport
+ , null
+ , "rtcp_transport"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("rtcp_transport"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // rtcp_transport
+ , "transport"
+ , "_padding"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("_padding"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+ , "rtcp_transport"
+ , null
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ });
+
+ return _abi_info;
+ }
+ }
+
+
+ // End of the ABI representation.
+
+#endregion
+ }
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Collections;
+ using System.Collections.Generic;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ public partial class WebRTCRTPSender : Gst.Object {
+
+ public WebRTCRTPSender (IntPtr raw) : base(raw) {}
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_rtp_sender_new();
+
+ public WebRTCRTPSender () : base (IntPtr.Zero)
+ {
+ if (GetType () != typeof (WebRTCRTPSender)) {
+ CreateNativeObject (new string [0], new GLib.Value[0]);
+ return;
+ }
+ Raw = gst_webrtc_rtp_sender_new();
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern void gst_webrtc_rtp_sender_set_transport(IntPtr raw, IntPtr transport);
+
+ public Gst.WebRTC.WebRTCDTLSTransport Transport {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("transport"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
+ }
+ }
+ set {
+ gst_webrtc_rtp_sender_set_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
+ }
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern void gst_webrtc_rtp_sender_set_rtcp_transport(IntPtr raw, IntPtr transport);
+
+ public Gst.WebRTC.WebRTCDTLSTransport RtcpTransport {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("rtcp_transport"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
+ }
+ }
+ set {
+ gst_webrtc_rtp_sender_set_rtcp_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
+ }
+ }
+
+
+ // Internal representation of the wrapped structure ABI.
+ static GLib.AbiStruct _class_abi = null;
+ static public new GLib.AbiStruct class_abi {
+ get {
+ if (_class_abi == null)
+ _class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{
+ new GLib.AbiField("_padding"
+ , Gst.Object.class_abi.Fields
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+ , null
+ , null
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ });
+
+ return _class_abi;
+ }
+ }
+
+
+ // End of the ABI representation.
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_rtp_sender_get_type();
+
+ public static new GLib.GType GType {
+ get {
+ IntPtr raw_ret = gst_webrtc_rtp_sender_get_type();
+ GLib.GType ret = new GLib.GType(raw_ret);
+ return ret;
+ }
+ }
+
+
+ static WebRTCRTPSender ()
+ {
+ GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
+ }
+
+ // Internal representation of the wrapped structure ABI.
+ static GLib.AbiStruct _abi_info = null;
+ static public new GLib.AbiStruct abi_info {
+ get {
+ if (_abi_info == null)
+ _abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{
+ new GLib.AbiField("transport"
+ , Gst.Object.abi_info.Fields
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // transport
+ , null
+ , "rtcp_transport"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("rtcp_transport"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // rtcp_transport
+ , "transport"
+ , "send_encodings"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("send_encodings"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // send_encodings
+ , "rtcp_transport"
+ , "_padding"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("_padding"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+ , "send_encodings"
+ , null
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ });
+
+ return _abi_info;
+ }
+ }
+
+
+ // End of the ABI representation.
+
+#endregion
+ }
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Collections;
+ using System.Collections.Generic;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ public partial class WebRTCRTPTransceiver : Gst.Object {
+
+ protected WebRTCRTPTransceiver (IntPtr raw) : base(raw) {}
+
+ protected WebRTCRTPTransceiver() : base(IntPtr.Zero)
+ {
+ CreateNativeObject (new string [0], new GLib.Value [0]);
+ }
+
+ [GLib.Property ("mlineindex")]
+ public uint Mlineindex {
+ get {
+ GLib.Value val = GetProperty ("mlineindex");
+ uint ret = (uint) val;
+ val.Dispose ();
+ return ret;
+ }
+ }
+
+ [GLib.Property ("receiver")]
+ public Gst.WebRTC.WebRTCRTPReceiver Receiver {
+ get {
+ GLib.Value val = GetProperty ("receiver");
+ Gst.WebRTC.WebRTCRTPReceiver ret = (Gst.WebRTC.WebRTCRTPReceiver) val;
+ val.Dispose ();
+ return ret;
+ }
+ }
+
+ [GLib.Property ("sender")]
+ public Gst.WebRTC.WebRTCRTPSender Sender {
+ get {
+ GLib.Value val = GetProperty ("sender");
+ Gst.WebRTC.WebRTCRTPSender ret = (Gst.WebRTC.WebRTCRTPSender) val;
+ val.Dispose ();
+ return ret;
+ }
+ }
+
+ public uint Mline {
+ get {
+ unsafe {
+ uint* raw_ptr = (uint*)(((byte*)Handle) + abi_info.GetFieldOffset("mline"));
+ return (*raw_ptr);
+ }
+ }
+ }
+
+ public string Mid {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("mid"));
+ return GLib.Marshaller.Utf8PtrToString ((*raw_ptr));
+ }
+ }
+ }
+
+ public bool Stopped {
+ get {
+ unsafe {
+ bool* raw_ptr = (bool*)(((byte*)Handle) + abi_info.GetFieldOffset("stopped"));
+ return (*raw_ptr);
+ }
+ }
+ }
+
+ public Gst.WebRTC.WebRTCRTPSender SenderField {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("sender"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCRTPSender;
+ }
+ }
+ }
+
+ public Gst.WebRTC.WebRTCRTPReceiver ReceiverField {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("receiver"));
+ return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCRTPReceiver;
+ }
+ }
+ }
+
+ public Gst.WebRTC.WebRTCRTPTransceiverDirection Direction {
+ get {
+ unsafe {
+ int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("direction"));
+ return (Gst.WebRTC.WebRTCRTPTransceiverDirection) (*raw_ptr);
+ }
+ }
+ }
+
+ public Gst.WebRTC.WebRTCRTPTransceiverDirection CurrentDirection {
+ get {
+ unsafe {
+ int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("current_direction"));
+ return (Gst.WebRTC.WebRTCRTPTransceiverDirection) (*raw_ptr);
+ }
+ }
+ }
+
+ public Gst.Caps CodecPreferences {
+ get {
+ unsafe {
+ IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("codec_preferences"));
+ return (*raw_ptr) == IntPtr.Zero ? null : (Gst.Caps) GLib.Opaque.GetOpaque ((*raw_ptr), typeof (Gst.Caps), false);
+ }
+ }
+ }
+
+
+ // Internal representation of the wrapped structure ABI.
+ static GLib.AbiStruct _class_abi = null;
+ static public new GLib.AbiStruct class_abi {
+ get {
+ if (_class_abi == null)
+ _class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{
+ new GLib.AbiField("_padding"
+ , Gst.Object.class_abi.Fields
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+ , null
+ , null
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ });
+
+ return _class_abi;
+ }
+ }
+
+
+ // End of the ABI representation.
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_rtp_transceiver_get_type();
+
+ public static new GLib.GType GType {
+ get {
+ IntPtr raw_ret = gst_webrtc_rtp_transceiver_get_type();
+ GLib.GType ret = new GLib.GType(raw_ret);
+ return ret;
+ }
+ }
+
+
+ static WebRTCRTPTransceiver ()
+ {
+ GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
+ }
+
+ // Internal representation of the wrapped structure ABI.
+ static GLib.AbiStruct _abi_info = null;
+ static public new GLib.AbiStruct abi_info {
+ get {
+ if (_abi_info == null)
+ _abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{
+ new GLib.AbiField("mline"
+ , Gst.Object.abi_info.Fields
+ , (uint) Marshal.SizeOf(typeof(uint)) // mline
+ , null
+ , "mid"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_mlineAlign), "mline")
+ , 0
+ ),
+ new GLib.AbiField("mid"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // mid
+ , "mline"
+ , "stopped"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("stopped"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(bool)) // stopped
+ , "mid"
+ , "sender"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_stoppedAlign), "stopped")
+ , 0
+ ),
+ new GLib.AbiField("sender"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // sender
+ , "stopped"
+ , "receiver"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("receiver"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // receiver
+ , "sender"
+ , "direction"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("direction"
+ , -1
+ , (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCRTPTransceiverDirection))) // direction
+ , "receiver"
+ , "current_direction"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_directionAlign), "direction")
+ , 0
+ ),
+ new GLib.AbiField("current_direction"
+ , -1
+ , (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCRTPTransceiverDirection))) // current_direction
+ , "direction"
+ , "codec_preferences"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_current_directionAlign), "current_direction")
+ , 0
+ ),
+ new GLib.AbiField("codec_preferences"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) // codec_preferences
+ , "current_direction"
+ , "_padding"
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ new GLib.AbiField("_padding"
+ , -1
+ , (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+ , "codec_preferences"
+ , null
+ , (uint) Marshal.SizeOf(typeof(IntPtr))
+ , 0
+ ),
+ });
+
+ return _abi_info;
+ }
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCRTPTransceiver_mlineAlign
+ {
+ sbyte f1;
+ private uint mline;
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCRTPTransceiver_stoppedAlign
+ {
+ sbyte f1;
+ private bool stopped;
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCRTPTransceiver_directionAlign
+ {
+ sbyte f1;
+ private Gst.WebRTC.WebRTCRTPTransceiverDirection direction;
+ }
+
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCRTPTransceiver_current_directionAlign
+ {
+ sbyte f1;
+ private Gst.WebRTC.WebRTCRTPTransceiverDirection current_direction;
+ }
+
+
+ // End of the ABI representation.
+
+#endregion
+ }
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCRTPTransceiverDirectionGType))]
+ public enum WebRTCRTPTransceiverDirection {
+
+ None = 0,
+ Inactive = 1,
+ Sendonly = 2,
+ Recvonly = 3,
+ Sendrecv = 4,
+ }
+
+ internal class WebRTCRTPTransceiverDirectionGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_rtp_transceiver_direction_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_rtp_transceiver_direction_get_type ());
+ }
+ }
+ }
+#endregion
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCSDPTypeGType))]
+ public enum WebRTCSDPType {
+
+ Offer = 1,
+ Pranswer = 2,
+ Answer = 3,
+ Rollback = 4,
+ }
+
+ internal class WebRTCSDPTypeGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_sdp_type_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_sdp_type_get_type ());
+ }
+ }
+ }
+#endregion
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Collections;
+ using System.Collections.Generic;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [StructLayout(LayoutKind.Sequential)]
+ public partial struct WebRTCSessionDescription : IEquatable<WebRTCSessionDescription> {
+
+ public Gst.WebRTC.WebRTCSDPType Type;
+ private IntPtr _sdp;
+ public Gst.Sdp.SDPMessage Sdp {
+ get {
+ return _sdp == IntPtr.Zero ? null : (Gst.Sdp.SDPMessage) GLib.Opaque.GetOpaque (_sdp, typeof (Gst.Sdp.SDPMessage), false);
+ }
+ set {
+ _sdp = value == null ? IntPtr.Zero : value.Handle;
+ }
+ }
+
+ public static Gst.WebRTC.WebRTCSessionDescription Zero = new Gst.WebRTC.WebRTCSessionDescription ();
+
+ public static Gst.WebRTC.WebRTCSessionDescription New(IntPtr raw) {
+ if (raw == IntPtr.Zero)
+ return Gst.WebRTC.WebRTCSessionDescription.Zero;
+ return (Gst.WebRTC.WebRTCSessionDescription) Marshal.PtrToStructure (raw, typeof (Gst.WebRTC.WebRTCSessionDescription));
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_session_description_new(int type, IntPtr sdp);
+
+ public static WebRTCSessionDescription New(Gst.WebRTC.WebRTCSDPType type, Gst.Sdp.SDPMessage sdp)
+ {
+ WebRTCSessionDescription result = WebRTCSessionDescription.New (gst_webrtc_session_description_new((int) type, sdp == null ? IntPtr.Zero : sdp.Handle));
+ return result;
+ }
+
+ [DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_session_description_get_type();
+
+ public static GLib.GType GType {
+ get {
+ IntPtr raw_ret = gst_webrtc_session_description_get_type();
+ GLib.GType ret = new GLib.GType(raw_ret);
+ return ret;
+ }
+ }
+
+ public bool Equals (WebRTCSessionDescription other)
+ {
+ return true && Type.Equals (other.Type) && Sdp.Equals (other.Sdp);
+ }
+
+ public override bool Equals (object other)
+ {
+ return other is WebRTCSessionDescription && Equals ((WebRTCSessionDescription) other);
+ }
+
+ public override int GetHashCode ()
+ {
+ return this.GetType ().FullName.GetHashCode () ^ Type.GetHashCode () ^ Sdp.GetHashCode ();
+ }
+
+ public static explicit operator GLib.Value (Gst.WebRTC.WebRTCSessionDescription boxed)
+ {
+ GLib.Value val = GLib.Value.Empty;
+ val.Init (Gst.WebRTC.WebRTCSessionDescription.GType);
+ val.Val = boxed;
+ return val;
+ }
+
+ public static explicit operator Gst.WebRTC.WebRTCSessionDescription (GLib.Value val)
+ {
+ return (Gst.WebRTC.WebRTCSessionDescription) val.Val;
+ }
+#endregion
+ }
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCSignalingStateGType))]
+ public enum WebRTCSignalingState {
+
+ Stable = 0,
+ Closed = 1,
+ HaveLocalOffer = 2,
+ HaveRemoteOffer = 3,
+ HaveLocalPranswer = 4,
+ HaveRemotePranswer = 5,
+ }
+
+ internal class WebRTCSignalingStateGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_signaling_state_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_signaling_state_get_type ());
+ }
+ }
+ }
+#endregion
+}
--- /dev/null
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+ using System;
+ using System.Runtime.InteropServices;
+
+#region Autogenerated code
+ [GLib.GType (typeof (Gst.WebRTC.WebRTCStatsTypeGType))]
+ public enum WebRTCStatsType {
+
+ Codec = 1,
+ InboundRtp = 2,
+ OutboundRtp = 3,
+ RemoteInboundRtp = 4,
+ RemoteOutboundRtp = 5,
+ Csrc = 6,
+ PeerConnection = 7,
+ DataChannel = 8,
+ Stream = 9,
+ Transport = 10,
+ CandidatePair = 11,
+ LocalCandidate = 12,
+ RemoteCandidate = 13,
+ Certificate = 14,
+ }
+
+ internal class WebRTCStatsTypeGType {
+ [DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_stats_type_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_stats_type_get_type ());
+ }
+ }
+ }
+#endregion
+}
GLib.GType.Register (Gst.Video.VideoEncoder.GType, typeof (Gst.Video.VideoEncoder));
GLib.GType.Register (Gst.Video.VideoFilter.GType, typeof (Gst.Video.VideoFilter));
GLib.GType.Register (Gst.Video.VideoSink.GType, typeof (Gst.Video.VideoSink));
+ GLib.GType.Register (Gst.WebRTC.WebRTCDTLSTransport.GType, typeof (Gst.WebRTC.WebRTCDTLSTransport));
+ GLib.GType.Register (Gst.WebRTC.WebRTCICETransport.GType, typeof (Gst.WebRTC.WebRTCICETransport));
+ GLib.GType.Register (Gst.WebRTC.WebRTCRTPReceiver.GType, typeof (Gst.WebRTC.WebRTCRTPReceiver));
+ GLib.GType.Register (Gst.WebRTC.WebRTCRTPSender.GType, typeof (Gst.WebRTC.WebRTCRTPSender));
+ GLib.GType.Register (Gst.WebRTC.WebRTCRTPTransceiver.GType, typeof (Gst.WebRTC.WebRTCRTPTransceiver));
}
}
}
#include <gst/video/video.h>
#include <gst/video/gstvideoaffinetransformationmeta.h>
#include <gst/net/gstnetcontrolmessagemeta.h>
+#include <gst/webrtc/webrtc.h>
int main (int argc, char *argv[]) {
g_print("\"sizeof(GstAllocatorClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstAllocatorClass));
g_print("\"GstVideoInfo.fps_d\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstVideoInfo, fps_d));
g_print("\"GstVideoInfo.offset\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstVideoInfo, offset));
g_print("\"GstVideoInfo.stride\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstVideoInfo, stride));
+ g_print("\"sizeof(GstWebRTCDTLSTransportClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCDTLSTransportClass));
+ g_print("\"GstWebRTCDTLSTransportClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransportClass, _padding));
+ g_print("\"sizeof(GstWebRTCDTLSTransport)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCDTLSTransport));
+ g_print("\"GstWebRTCDTLSTransport.transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, transport));
+ g_print("\"GstWebRTCDTLSTransport.state\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, state));
+ g_print("\"GstWebRTCDTLSTransport.is_rtcp\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, is_rtcp));
+ g_print("\"GstWebRTCDTLSTransport.client\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, client));
+ g_print("\"GstWebRTCDTLSTransport.session_id\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, session_id));
+ g_print("\"GstWebRTCDTLSTransport.dtlssrtpenc\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, dtlssrtpenc));
+ g_print("\"GstWebRTCDTLSTransport.dtlssrtpdec\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, dtlssrtpdec));
+ g_print("\"GstWebRTCDTLSTransport._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, _padding));
+ g_print("\"sizeof(GstWebRTCICETransportClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCICETransportClass));
+ g_print("\"GstWebRTCICETransportClass.gather_candidates\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransportClass, gather_candidates));
+ g_print("\"GstWebRTCICETransportClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransportClass, _padding));
+ g_print("\"sizeof(GstWebRTCICETransport)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCICETransport));
+ g_print("\"GstWebRTCICETransport.role\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, role));
+ g_print("\"GstWebRTCICETransport.component\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, component));
+ g_print("\"GstWebRTCICETransport.state\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, state));
+ g_print("\"GstWebRTCICETransport.gathering_state\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, gathering_state));
+ g_print("\"GstWebRTCICETransport.src\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, src));
+ g_print("\"GstWebRTCICETransport.sink\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, sink));
+ g_print("\"GstWebRTCICETransport._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, _padding));
+ g_print("\"sizeof(GstWebRTCRTPReceiverClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPReceiverClass));
+ g_print("\"GstWebRTCRTPReceiverClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiverClass, _padding));
+ g_print("\"sizeof(GstWebRTCRTPReceiver)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPReceiver));
+ g_print("\"GstWebRTCRTPReceiver.transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiver, transport));
+ g_print("\"GstWebRTCRTPReceiver.rtcp_transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiver, rtcp_transport));
+ g_print("\"GstWebRTCRTPReceiver._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiver, _padding));
+ g_print("\"sizeof(GstWebRTCRTPSenderClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPSenderClass));
+ g_print("\"GstWebRTCRTPSenderClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSenderClass, _padding));
+ g_print("\"sizeof(GstWebRTCRTPSender)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPSender));
+ g_print("\"GstWebRTCRTPSender.transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, transport));
+ g_print("\"GstWebRTCRTPSender.rtcp_transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, rtcp_transport));
+ g_print("\"GstWebRTCRTPSender.send_encodings\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, send_encodings));
+ g_print("\"GstWebRTCRTPSender._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, _padding));
+ g_print("\"sizeof(GstWebRTCRTPTransceiverClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPTransceiverClass));
+ g_print("\"GstWebRTCRTPTransceiverClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiverClass, _padding));
+ g_print("\"sizeof(GstWebRTCRTPTransceiver)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPTransceiver));
+ g_print("\"GstWebRTCRTPTransceiver.mline\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, mline));
+ g_print("\"GstWebRTCRTPTransceiver.mid\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, mid));
+ g_print("\"GstWebRTCRTPTransceiver.stopped\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, stopped));
+ g_print("\"GstWebRTCRTPTransceiver.sender\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, sender));
+ g_print("\"GstWebRTCRTPTransceiver.receiver\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, receiver));
+ g_print("\"GstWebRTCRTPTransceiver.direction\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, direction));
+ g_print("\"GstWebRTCRTPTransceiver.current_direction\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, current_direction));
+ g_print("\"GstWebRTCRTPTransceiver.codec_preferences\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, codec_preferences));
+ g_print("\"GstWebRTCRTPTransceiver._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, _padding));
return 0;
}
Console.WriteLine("\"GstVideoInfo.fps_d\": \"" + Gst.Video.VideoInfo.abi_info.GetFieldOffset("fps_d") + "\"");
Console.WriteLine("\"GstVideoInfo.offset\": \"" + Gst.Video.VideoInfo.abi_info.GetFieldOffset("offset") + "\"");
Console.WriteLine("\"GstVideoInfo.stride\": \"" + Gst.Video.VideoInfo.abi_info.GetFieldOffset("stride") + "\"");
+ Console.WriteLine("\"sizeof(GstWebRTCDTLSTransportClass)\": \"" + Gst.WebRTC.WebRTCDTLSTransport.class_abi.Size + "\"");
+ Console.WriteLine("\"GstWebRTCDTLSTransportClass._padding\": \"" + Gst.WebRTC.WebRTCDTLSTransport.class_abi.GetFieldOffset("_padding") + "\"");
+ Console.WriteLine("\"sizeof(GstWebRTCDTLSTransport)\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.Size + "\"");
+ Console.WriteLine("\"GstWebRTCDTLSTransport.transport\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("transport") + "\"");
+ Console.WriteLine("\"GstWebRTCDTLSTransport.state\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("state") + "\"");
+ Console.WriteLine("\"GstWebRTCDTLSTransport.is_rtcp\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("is_rtcp") + "\"");
+ Console.WriteLine("\"GstWebRTCDTLSTransport.client\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("client") + "\"");
+ Console.WriteLine("\"GstWebRTCDTLSTransport.session_id\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("session_id") + "\"");
+ Console.WriteLine("\"GstWebRTCDTLSTransport.dtlssrtpenc\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("dtlssrtpenc") + "\"");
+ Console.WriteLine("\"GstWebRTCDTLSTransport.dtlssrtpdec\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("dtlssrtpdec") + "\"");
+ Console.WriteLine("\"GstWebRTCDTLSTransport._padding\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("_padding") + "\"");
+ Console.WriteLine("\"sizeof(GstWebRTCICETransportClass)\": \"" + Gst.WebRTC.WebRTCICETransport.class_abi.Size + "\"");
+ Console.WriteLine("\"GstWebRTCICETransportClass.gather_candidates\": \"" + Gst.WebRTC.WebRTCICETransport.class_abi.GetFieldOffset("gather_candidates") + "\"");
+ Console.WriteLine("\"GstWebRTCICETransportClass._padding\": \"" + Gst.WebRTC.WebRTCICETransport.class_abi.GetFieldOffset("_padding") + "\"");
+ Console.WriteLine("\"sizeof(GstWebRTCICETransport)\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.Size + "\"");
+ Console.WriteLine("\"GstWebRTCICETransport.role\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("role") + "\"");
+ Console.WriteLine("\"GstWebRTCICETransport.component\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("component") + "\"");
+ Console.WriteLine("\"GstWebRTCICETransport.state\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("state") + "\"");
+ Console.WriteLine("\"GstWebRTCICETransport.gathering_state\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("gathering_state") + "\"");
+ Console.WriteLine("\"GstWebRTCICETransport.src\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("src") + "\"");
+ Console.WriteLine("\"GstWebRTCICETransport.sink\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("sink") + "\"");
+ Console.WriteLine("\"GstWebRTCICETransport._padding\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("_padding") + "\"");
+ Console.WriteLine("\"sizeof(GstWebRTCRTPReceiverClass)\": \"" + Gst.WebRTC.WebRTCRTPReceiver.class_abi.Size + "\"");
+ Console.WriteLine("\"GstWebRTCRTPReceiverClass._padding\": \"" + Gst.WebRTC.WebRTCRTPReceiver.class_abi.GetFieldOffset("_padding") + "\"");
+ Console.WriteLine("\"sizeof(GstWebRTCRTPReceiver)\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.Size + "\"");
+ Console.WriteLine("\"GstWebRTCRTPReceiver.transport\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.GetFieldOffset("transport") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPReceiver.rtcp_transport\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.GetFieldOffset("rtcp_transport") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPReceiver._padding\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.GetFieldOffset("_padding") + "\"");
+ Console.WriteLine("\"sizeof(GstWebRTCRTPSenderClass)\": \"" + Gst.WebRTC.WebRTCRTPSender.class_abi.Size + "\"");
+ Console.WriteLine("\"GstWebRTCRTPSenderClass._padding\": \"" + Gst.WebRTC.WebRTCRTPSender.class_abi.GetFieldOffset("_padding") + "\"");
+ Console.WriteLine("\"sizeof(GstWebRTCRTPSender)\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.Size + "\"");
+ Console.WriteLine("\"GstWebRTCRTPSender.transport\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("transport") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPSender.rtcp_transport\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("rtcp_transport") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPSender.send_encodings\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("send_encodings") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPSender._padding\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("_padding") + "\"");
+ Console.WriteLine("\"sizeof(GstWebRTCRTPTransceiverClass)\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.class_abi.Size + "\"");
+ Console.WriteLine("\"GstWebRTCRTPTransceiverClass._padding\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.class_abi.GetFieldOffset("_padding") + "\"");
+ Console.WriteLine("\"sizeof(GstWebRTCRTPTransceiver)\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.Size + "\"");
+ Console.WriteLine("\"GstWebRTCRTPTransceiver.mline\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("mline") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPTransceiver.mid\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("mid") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPTransceiver.stopped\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("stopped") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPTransceiver.sender\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("sender") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPTransceiver.receiver\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("receiver") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPTransceiver.direction\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("direction") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPTransceiver.current_direction\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("current_direction") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPTransceiver.codec_preferences\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("codec_preferences") + "\"");
+ Console.WriteLine("\"GstWebRTCRTPTransceiver._padding\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("_padding") + "\"");
}
}
}
<constant value="16" ctype="gint" gtype="gint" name="VIDEO_TILE_Y_TILES_SHIFT" />
</object>
</namespace>
+ <namespace name="Gst.WebRTC" library="libgstwebrtc-1.0-0.dll">
+ <enum name="WebRTCDTLSSetup" cname="GstWebRTCDTLSSetup" type="enum" gtype="gst_webrtc_dtls_setup_get_type">
+ <member cname="GST_WEBRTC_DTLS_SETUP_NONE" name="None" value="0" />
+ <member cname="GST_WEBRTC_DTLS_SETUP_ACTPASS" name="Actpass" value="1" />
+ <member cname="GST_WEBRTC_DTLS_SETUP_ACTIVE" name="Active" value="2" />
+ <member cname="GST_WEBRTC_DTLS_SETUP_PASSIVE" name="Passive" value="3" />
+ </enum>
+ <enum name="WebRTCDTLSTransportState" cname="GstWebRTCDTLSTransportState" type="enum" gtype="gst_webrtc_dtls_transport_state_get_type">
+ <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW" name="New" value="0" />
+ <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED" name="Closed" value="1" />
+ <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED" name="Failed" value="2" />
+ <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING" name="Connecting" value="3" />
+ <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED" name="Connected" value="4" />
+ </enum>
+ <enum name="WebRTCICEComponent" cname="GstWebRTCICEComponent" type="enum" gtype="gst_webrtc_ice_component_get_type">
+ <member cname="GST_WEBRTC_ICE_COMPONENT_RTP" name="Rtp" value="0" />
+ <member cname="GST_WEBRTC_ICE_COMPONENT_RTCP" name="Rtcp" value="1" />
+ </enum>
+ <enum name="WebRTCICEConnectionState" cname="GstWebRTCICEConnectionState" type="enum" gtype="gst_webrtc_ice_connection_state_get_type">
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_NEW" name="New" value="0" />
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING" name="Checking" value="1" />
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED" name="Connected" value="2" />
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED" name="Completed" value="3" />
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED" name="Failed" value="4" />
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="5" />
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED" name="Closed" value="6" />
+ </enum>
+ <enum name="WebRTCICEGatheringState" cname="GstWebRTCICEGatheringState" type="enum" gtype="gst_webrtc_ice_gathering_state_get_type">
+ <member cname="GST_WEBRTC_ICE_GATHERING_STATE_NEW" name="New" value="0" />
+ <member cname="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING" name="Gathering" value="1" />
+ <member cname="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE" name="Complete" value="2" />
+ </enum>
+ <enum name="WebRTCICERole" cname="GstWebRTCICERole" type="enum" gtype="gst_webrtc_ice_role_get_type">
+ <member cname="GST_WEBRTC_ICE_ROLE_CONTROLLED" name="Controlled" value="0" />
+ <member cname="GST_WEBRTC_ICE_ROLE_CONTROLLING" name="Controlling" value="1" />
+ </enum>
+ <enum name="WebRTCPeerConnectionState" cname="GstWebRTCPeerConnectionState" type="enum" gtype="gst_webrtc_peer_connection_state_get_type">
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_NEW" name="New" value="0" />
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING" name="Connecting" value="1" />
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED" name="Connected" value="2" />
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="3" />
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED" name="Failed" value="4" />
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED" name="Closed" value="5" />
+ </enum>
+ <enum name="WebRTCRTPTransceiverDirection" cname="GstWebRTCRTPTransceiverDirection" type="enum" gtype="gst_webrtc_rtp_transceiver_direction_get_type">
+ <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE" name="None" value="0" />
+ <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE" name="Inactive" value="1" />
+ <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY" name="Sendonly" value="2" />
+ <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY" name="Recvonly" value="3" />
+ <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV" name="Sendrecv" value="4" />
+ </enum>
+ <enum name="WebRTCSDPType" cname="GstWebRTCSDPType" type="enum" gtype="gst_webrtc_sdp_type_get_type">
+ <member cname="GST_WEBRTC_SDP_TYPE_OFFER" name="Offer" value="1" />
+ <member cname="GST_WEBRTC_SDP_TYPE_PRANSWER" name="Pranswer" value="2" />
+ <member cname="GST_WEBRTC_SDP_TYPE_ANSWER" name="Answer" value="3" />
+ <member cname="GST_WEBRTC_SDP_TYPE_ROLLBACK" name="Rollback" value="4" />
+ </enum>
+ <enum name="WebRTCSignalingState" cname="GstWebRTCSignalingState" type="enum" gtype="gst_webrtc_signaling_state_get_type">
+ <member cname="GST_WEBRTC_SIGNALING_STATE_STABLE" name="Stable" value="0" />
+ <member cname="GST_WEBRTC_SIGNALING_STATE_CLOSED" name="Closed" value="1" />
+ <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER" name="HaveLocalOffer" value="2" />
+ <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER" name="HaveRemoteOffer" value="3" />
+ <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER" name="HaveLocalPranswer" value="4" />
+ <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER" name="HaveRemotePranswer" value="5" />
+ </enum>
+ <enum name="WebRTCStatsType" cname="GstWebRTCStatsType" type="enum" gtype="gst_webrtc_stats_type_get_type">
+ <member cname="GST_WEBRTC_STATS_CODEC" name="Codec" value="1" />
+ <member cname="GST_WEBRTC_STATS_INBOUND_RTP" name="InboundRtp" value="2" />
+ <member cname="GST_WEBRTC_STATS_OUTBOUND_RTP" name="OutboundRtp" value="3" />
+ <member cname="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP" name="RemoteInboundRtp" value="4" />
+ <member cname="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP" name="RemoteOutboundRtp" value="5" />
+ <member cname="GST_WEBRTC_STATS_CSRC" name="Csrc" value="6" />
+ <member cname="GST_WEBRTC_STATS_PEER_CONNECTION" name="PeerConnection" value="7" />
+ <member cname="GST_WEBRTC_STATS_DATA_CHANNEL" name="DataChannel" value="8" />
+ <member cname="GST_WEBRTC_STATS_STREAM" name="Stream" value="9" />
+ <member cname="GST_WEBRTC_STATS_TRANSPORT" name="Transport" value="10" />
+ <member cname="GST_WEBRTC_STATS_CANDIDATE_PAIR" name="CandidatePair" value="11" />
+ <member cname="GST_WEBRTC_STATS_LOCAL_CANDIDATE" name="LocalCandidate" value="12" />
+ <member cname="GST_WEBRTC_STATS_REMOTE_CANDIDATE" name="RemoteCandidate" value="13" />
+ <member cname="GST_WEBRTC_STATS_CERTIFICATE" name="Certificate" value="14" />
+ </enum>
+ <object name="WebRTCDTLSTransport" cname="GstWebRTCDTLSTransport" opaque="false" hidden="false" parent="GstObject">
+ <class_struct cname="GstWebRTCDTLSTransportClass">
+ <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+ </class_struct>
+ <method name="GetType" cname="gst_webrtc_dtls_transport_get_type" shared="true">
+ <return-type type="GType" />
+ </method>
+ <constructor cname="gst_webrtc_dtls_transport_new">
+ <parameters>
+ <parameter name="session_id" type="guint" />
+ <parameter name="rtcp" type="gboolean" />
+ </parameters>
+ </constructor>
+ <method name="SetTransport" cname="gst_webrtc_dtls_transport_set_transport">
+ <return-type type="void" />
+ <parameters>
+ <parameter name="ice" type="GstWebRTCICETransport*" />
+ </parameters>
+ </method>
+ <property name="Certificate" cname="certificate" type="gchar*" readable="true" writeable="true" construct="false" construct-only="false" />
+ <property name="Client" cname="client" type="gboolean" readable="true" writeable="true" construct="false" construct-only="false" />
+ <property name="RemoteCertificate" cname="remote-certificate" type="gchar*" readable="true" writeable="false" construct="false" construct-only="false" />
+ <property name="Rtcp" cname="rtcp" type="gboolean" readable="true" writeable="true" construct="false" construct-only="true" />
+ <property name="SessionId" cname="session-id" type="guint" readable="true" writeable="true" construct="false" construct-only="true" />
+ <property name="State" cname="state" type="GstWebRTCDTLSTransportState" readable="true" writeable="false" construct="false" construct-only="false" />
+ <property name="Transport" cname="transport" type="GstWebRTCICETransport*" readable="true" writeable="false" construct="false" construct-only="false" />
+ <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
+ <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="TransportField" type="GstWebRTCICETransport*" />
+ <field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCDTLSTransportState" />
+ <field cname="is_rtcp" access="public" writeable="false" readable="true" is_callback="false" name="IsRtcp" type="gboolean" />
+ <field cname="client" access="public" writeable="false" readable="true" is_callback="false" name="ClientField" type="gboolean" />
+ <field cname="session_id" access="public" writeable="false" readable="true" is_callback="false" name="SessionIdField" type="guint" />
+ <field cname="dtlssrtpenc" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpenc" type="GstElement*" />
+ <field cname="dtlssrtpdec" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpdec" type="GstElement*" />
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+ </object>
+ <object name="WebRTCICETransport" cname="GstWebRTCICETransport" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
+ <class_struct cname="GstWebRTCICETransportClass">
+ <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <method vm="gather_candidates" />
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+ </class_struct>
+ <method name="GetType" cname="gst_webrtc_ice_transport_get_type" shared="true">
+ <return-type type="GType" />
+ </method>
+ <virtual_method name="GatherCandidates" cname="gather_candidates">
+ <return-type type="gboolean" />
+ <parameters />
+ </virtual_method>
+ <method name="ConnectionStateChange" cname="gst_webrtc_ice_transport_connection_state_change">
+ <return-type type="void" />
+ <parameters>
+ <parameter name="new_state" type="GstWebRTCICEConnectionState" />
+ </parameters>
+ </method>
+ <method name="GatheringStateChange" cname="gst_webrtc_ice_transport_gathering_state_change">
+ <return-type type="void" />
+ <parameters>
+ <parameter name="new_state" type="GstWebRTCICEGatheringState" />
+ </parameters>
+ </method>
+ <method name="NewCandidate" cname="gst_webrtc_ice_transport_new_candidate">
+ <return-type type="void" />
+ <parameters>
+ <parameter name="stream_id" type="guint" />
+ <parameter name="component" type="GstWebRTCICEComponent" />
+ <parameter name="attr" type="const-gchar*" />
+ </parameters>
+ </method>
+ <method name="SelectedPairChange" cname="gst_webrtc_ice_transport_selected_pair_change">
+ <return-type type="void" />
+ <parameters />
+ </method>
+ <property name="Component" cname="component" type="GstWebRTCICEComponent" readable="true" writeable="true" construct="false" construct-only="true" />
+ <property name="GatheringState" cname="gathering-state" type="GstWebRTCICEGatheringState" readable="true" writeable="false" construct="false" construct-only="false" />
+ <property name="State" cname="state" type="GstWebRTCICEConnectionState" readable="true" writeable="false" construct="false" construct-only="false" />
+ <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
+ <field cname="role" access="public" writeable="false" readable="true" is_callback="false" name="Role" type="GstWebRTCICERole" />
+ <field cname="component" access="public" writeable="false" readable="true" is_callback="false" name="ComponentField" type="GstWebRTCICEComponent" />
+ <field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCICEConnectionState" />
+ <field cname="gathering_state" access="public" writeable="false" readable="true" is_callback="false" name="GatheringStateField" type="GstWebRTCICEGatheringState" />
+ <field cname="src" access="public" writeable="false" readable="true" is_callback="false" name="Src" type="GstElement*" />
+ <field cname="sink" access="public" writeable="false" readable="true" is_callback="false" name="Sink" type="GstElement*" />
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+ <signal name="OnNewCandidate" cname="on-new-candidate" when="last">
+ <return-type type="void" />
+ <parameters>
+ <parameter name="_object" type="const-gchar*" />
+ </parameters>
+ </signal>
+ <signal name="OnSelectedCandidatePairChange" cname="on-selected-candidate-pair-change" when="last">
+ <return-type type="void" />
+ <parameters />
+ </signal>
+ </object>
+ <object name="WebRTCRTPReceiver" cname="GstWebRTCRTPReceiver" opaque="false" hidden="false" parent="GstObject">
+ <class_struct cname="GstWebRTCRTPReceiverClass">
+ <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+ </class_struct>
+ <method name="GetType" cname="gst_webrtc_rtp_receiver_get_type" shared="true">
+ <return-type type="GType" />
+ </method>
+ <constructor cname="gst_webrtc_rtp_receiver_new" disable_void_ctor="" />
+ <method name="SetRtcpTransport" cname="gst_webrtc_rtp_receiver_set_rtcp_transport">
+ <return-type type="void" />
+ <parameters>
+ <parameter name="transport" type="GstWebRTCDTLSTransport*" />
+ </parameters>
+ </method>
+ <method name="SetTransport" cname="gst_webrtc_rtp_receiver_set_transport">
+ <return-type type="void" />
+ <parameters>
+ <parameter name="transport" type="GstWebRTCDTLSTransport*" />
+ </parameters>
+ </method>
+ <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
+ <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*" />
+ <field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*" />
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+ </object>
+ <object name="WebRTCRTPSender" cname="GstWebRTCRTPSender" opaque="false" hidden="false" parent="GstObject">
+ <class_struct cname="GstWebRTCRTPSenderClass">
+ <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+ </class_struct>
+ <method name="GetType" cname="gst_webrtc_rtp_sender_get_type" shared="true">
+ <return-type type="GType" />
+ </method>
+ <constructor cname="gst_webrtc_rtp_sender_new" disable_void_ctor="" />
+ <method name="SetRtcpTransport" cname="gst_webrtc_rtp_sender_set_rtcp_transport">
+ <return-type type="void" />
+ <parameters>
+ <parameter name="transport" type="GstWebRTCDTLSTransport*" />
+ </parameters>
+ </method>
+ <method name="SetTransport" cname="gst_webrtc_rtp_sender_set_transport">
+ <return-type type="void" />
+ <parameters>
+ <parameter name="transport" type="GstWebRTCDTLSTransport*" />
+ </parameters>
+ </method>
+ <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
+ <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*" />
+ <field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*" />
+ <field cname="send_encodings" access="public" writeable="false" readable="true" is_callback="false" name="SendEncodings" type="GArray*" array="true" null_term_array="true" />
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+ </object>
+ <object name="WebRTCRTPTransceiver" cname="GstWebRTCRTPTransceiver" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
+ <class_struct cname="GstWebRTCRTPTransceiverClass">
+ <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+ </class_struct>
+ <method name="GetType" cname="gst_webrtc_rtp_transceiver_get_type" shared="true">
+ <return-type type="GType" />
+ </method>
+ <property name="Mlineindex" cname="mlineindex" type="guint" readable="true" writeable="true" construct="false" construct-only="true" />
+ <property name="Receiver" cname="receiver" type="GstWebRTCRTPReceiver*" readable="true" writeable="true" construct="false" construct-only="true" />
+ <property name="Sender" cname="sender" type="GstWebRTCRTPSender*" readable="true" writeable="true" construct="false" construct-only="true" />
+ <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
+ <field cname="mline" access="public" writeable="false" readable="true" is_callback="false" name="Mline" type="guint" />
+ <field cname="mid" access="public" writeable="false" readable="true" is_callback="false" name="Mid" type="gchar*" />
+ <field cname="stopped" access="public" writeable="false" readable="true" is_callback="false" name="Stopped" type="gboolean" />
+ <field cname="sender" access="public" writeable="false" readable="true" is_callback="false" name="SenderField" type="GstWebRTCRTPSender*" />
+ <field cname="receiver" access="public" writeable="false" readable="true" is_callback="false" name="ReceiverField" type="GstWebRTCRTPReceiver*" />
+ <field cname="direction" access="public" writeable="false" readable="true" is_callback="false" name="Direction" type="GstWebRTCRTPTransceiverDirection" />
+ <field cname="current_direction" access="public" writeable="false" readable="true" is_callback="false" name="CurrentDirection" type="GstWebRTCRTPTransceiverDirection" />
+ <field cname="codec_preferences" access="public" writeable="false" readable="true" is_callback="false" name="CodecPreferences" type="GstCaps*">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+ </object>
+ <boxed name="WebRTCSessionDescription" cname="GstWebRTCSessionDescription" opaque="false" hidden="false">
+ <method name="GetType" cname="gst_webrtc_session_description_get_type" shared="true">
+ <return-type type="GType" />
+ </method>
+ <field cname="type" access="public" writeable="true" readable="true" is_callback="false" name="Type" type="GstWebRTCSDPType" />
+ <field cname="sdp" access="public" writeable="true" readable="true" is_callback="false" name="Sdp" type="GstSDPMessage*">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <constructor cname="gst_webrtc_session_description_new">
+ <parameters>
+ <parameter name="type" type="GstWebRTCSDPType" />
+ <parameter name="sdp" type="GstSDPMessage*">
+ <warning>missing glib:type-name</warning>
+ </parameter>
+ </parameters>
+ </constructor>
+ <method name="Copy" cname="gst_webrtc_session_description_copy">
+ <return-type type="GstWebRTCSessionDescription*" owned="true">
+ <warning>missing glib:type-name</warning>
+ </return-type>
+ <parameters />
+ </method>
+ <method name="Free" cname="gst_webrtc_session_description_free">
+ <return-type type="void" />
+ <parameters />
+ </method>
+ </boxed>
+ <object name="Global" cname="GstWebRTCGlobal" opaque="true">
+ <method name="WebrtcSdpTypeToString" cname="gst_webrtc_sdp_type_to_string" shared="true">
+ <return-type type="const-gchar*" />
+ <parameters>
+ <parameter name="type" type="GstWebRTCSDPType" />
+ </parameters>
+ </method>
+ </object>
+ <object name="Constants" cname="GstWebRTCConstants" opaque="true" />
+ </namespace>
</api>
\ No newline at end of file
'Gst.Rtsp/Gst.RtspSharp.RTSPConnectionAcceptCertificateFuncNative.cs',
'Gst.Audio/AudioStreamAlign.cs',
'Gst.Video/VideoOverlayProperties.cs',
+ 'Gst.WebRTC/WebRTCPeerConnectionState.cs',
+ 'Gst.WebRTC/WebRTCSessionDescription.cs',
+ 'Gst.WebRTC/WebRTCICEGatheringState.cs',
+ 'Gst.WebRTC/WebRTCRTPTransceiverDirection.cs',
+ 'Gst.WebRTC/WebRTCRTPTransceiver.cs',
+ 'Gst.WebRTC/OnNewCandidateHandler.cs',
+ 'Gst.WebRTC/WebRTCICERole.cs',
+ 'Gst.WebRTC/Global.cs',
+ 'Gst.WebRTC/WebRTCICEComponent.cs',
+ 'Gst.WebRTC/WebRTCICEConnectionState.cs',
+ 'Gst.WebRTC/WebRTCDTLSTransport.cs',
+ 'Gst.WebRTC/WebRTCICETransport.cs',
+ 'Gst.WebRTC/WebRTCSDPType.cs',
+ 'Gst.WebRTC/WebRTCRTPSender.cs',
+ 'Gst.WebRTC/WebRTCSignalingState.cs',
+ 'Gst.WebRTC/WebRTCDTLSTransportState.cs',
+ 'Gst.WebRTC/WebRTCDTLSSetup.cs',
+ 'Gst.WebRTC/WebRTCRTPReceiver.cs',
+ 'Gst.WebRTC/WebRTCStatsType.cs',
+ 'Gst.WebRTC/Constants.cs',
]
run_target('update_gstreamer_code',
<constant value="16" ctype="gint" gtype="gint" name="VIDEO_TILE_Y_TILES_SHIFT"/>
</object>
</namespace>
+ <namespace name="GstWebRTC" library="gstwebrtc-1.0">
+ <enum name="WebRTCDTLSSetup" cname="GstWebRTCDTLSSetup" type="enum" gtype="gst_webrtc_dtls_setup_get_type">
+ <member cname="GST_WEBRTC_DTLS_SETUP_NONE" name="None" value="0"/>
+ <member cname="GST_WEBRTC_DTLS_SETUP_ACTPASS" name="Actpass" value="1"/>
+ <member cname="GST_WEBRTC_DTLS_SETUP_ACTIVE" name="Active" value="2"/>
+ <member cname="GST_WEBRTC_DTLS_SETUP_PASSIVE" name="Passive" value="3"/>
+ </enum>
+ <enum name="WebRTCDTLSTransportState" cname="GstWebRTCDTLSTransportState" type="enum" gtype="gst_webrtc_dtls_transport_state_get_type">
+ <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW" name="New" value="0"/>
+ <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED" name="Closed" value="1"/>
+ <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED" name="Failed" value="2"/>
+ <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING" name="Connecting" value="3"/>
+ <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED" name="Connected" value="4"/>
+ </enum>
+ <enum name="WebRTCICEComponent" cname="GstWebRTCICEComponent" type="enum" gtype="gst_webrtc_ice_component_get_type">
+ <member cname="GST_WEBRTC_ICE_COMPONENT_RTP" name="Rtp" value="0"/>
+ <member cname="GST_WEBRTC_ICE_COMPONENT_RTCP" name="Rtcp" value="1"/>
+ </enum>
+ <enum name="WebRTCICEConnectionState" cname="GstWebRTCICEConnectionState" type="enum" gtype="gst_webrtc_ice_connection_state_get_type">
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_NEW" name="New" value="0"/>
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING" name="Checking" value="1"/>
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED" name="Connected" value="2"/>
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED" name="Completed" value="3"/>
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED" name="Failed" value="4"/>
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="5"/>
+ <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED" name="Closed" value="6"/>
+ </enum>
+ <enum name="WebRTCICEGatheringState" cname="GstWebRTCICEGatheringState" type="enum" gtype="gst_webrtc_ice_gathering_state_get_type">
+ <member cname="GST_WEBRTC_ICE_GATHERING_STATE_NEW" name="New" value="0"/>
+ <member cname="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING" name="Gathering" value="1"/>
+ <member cname="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE" name="Complete" value="2"/>
+ </enum>
+ <enum name="WebRTCICERole" cname="GstWebRTCICERole" type="enum" gtype="gst_webrtc_ice_role_get_type">
+ <member cname="GST_WEBRTC_ICE_ROLE_CONTROLLED" name="Controlled" value="0"/>
+ <member cname="GST_WEBRTC_ICE_ROLE_CONTROLLING" name="Controlling" value="1"/>
+ </enum>
+ <enum name="WebRTCPeerConnectionState" cname="GstWebRTCPeerConnectionState" type="enum" gtype="gst_webrtc_peer_connection_state_get_type">
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_NEW" name="New" value="0"/>
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING" name="Connecting" value="1"/>
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED" name="Connected" value="2"/>
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="3"/>
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED" name="Failed" value="4"/>
+ <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED" name="Closed" value="5"/>
+ </enum>
+ <enum name="WebRTCRTPTransceiverDirection" cname="GstWebRTCRTPTransceiverDirection" type="enum" gtype="gst_webrtc_rtp_transceiver_direction_get_type">
+ <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE" name="None" value="0"/>
+ <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE" name="Inactive" value="1"/>
+ <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY" name="Sendonly" value="2"/>
+ <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY" name="Recvonly" value="3"/>
+ <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV" name="Sendrecv" value="4"/>
+ </enum>
+ <enum name="WebRTCSDPType" cname="GstWebRTCSDPType" type="enum" gtype="gst_webrtc_sdp_type_get_type">
+ <member cname="GST_WEBRTC_SDP_TYPE_OFFER" name="Offer" value="1"/>
+ <member cname="GST_WEBRTC_SDP_TYPE_PRANSWER" name="Pranswer" value="2"/>
+ <member cname="GST_WEBRTC_SDP_TYPE_ANSWER" name="Answer" value="3"/>
+ <member cname="GST_WEBRTC_SDP_TYPE_ROLLBACK" name="Rollback" value="4"/>
+ </enum>
+ <enum name="WebRTCSignalingState" cname="GstWebRTCSignalingState" type="enum" gtype="gst_webrtc_signaling_state_get_type">
+ <member cname="GST_WEBRTC_SIGNALING_STATE_STABLE" name="Stable" value="0"/>
+ <member cname="GST_WEBRTC_SIGNALING_STATE_CLOSED" name="Closed" value="1"/>
+ <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER" name="HaveLocalOffer" value="2"/>
+ <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER" name="HaveRemoteOffer" value="3"/>
+ <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER" name="HaveLocalPranswer" value="4"/>
+ <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER" name="HaveRemotePranswer" value="5"/>
+ </enum>
+ <enum name="WebRTCStatsType" cname="GstWebRTCStatsType" type="enum" gtype="gst_webrtc_stats_type_get_type">
+ <member cname="GST_WEBRTC_STATS_CODEC" name="Codec" value="1"/>
+ <member cname="GST_WEBRTC_STATS_INBOUND_RTP" name="InboundRtp" value="2"/>
+ <member cname="GST_WEBRTC_STATS_OUTBOUND_RTP" name="OutboundRtp" value="3"/>
+ <member cname="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP" name="RemoteInboundRtp" value="4"/>
+ <member cname="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP" name="RemoteOutboundRtp" value="5"/>
+ <member cname="GST_WEBRTC_STATS_CSRC" name="Csrc" value="6"/>
+ <member cname="GST_WEBRTC_STATS_PEER_CONNECTION" name="PeerConnection" value="7"/>
+ <member cname="GST_WEBRTC_STATS_DATA_CHANNEL" name="DataChannel" value="8"/>
+ <member cname="GST_WEBRTC_STATS_STREAM" name="Stream" value="9"/>
+ <member cname="GST_WEBRTC_STATS_TRANSPORT" name="Transport" value="10"/>
+ <member cname="GST_WEBRTC_STATS_CANDIDATE_PAIR" name="CandidatePair" value="11"/>
+ <member cname="GST_WEBRTC_STATS_LOCAL_CANDIDATE" name="LocalCandidate" value="12"/>
+ <member cname="GST_WEBRTC_STATS_REMOTE_CANDIDATE" name="RemoteCandidate" value="13"/>
+ <member cname="GST_WEBRTC_STATS_CERTIFICATE" name="Certificate" value="14"/>
+ </enum>
+ <object name="WebRTCDTLSTransport" cname="GstWebRTCDTLSTransport" opaque="false" hidden="false" parent="GstObject">
+ <class_struct cname="GstWebRTCDTLSTransportClass">
+ <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+ </class_struct>
+ <method name="GetType" cname="gst_webrtc_dtls_transport_get_type" shared="true">
+ <return-type type="GType"/>
+ </method>
+ <constructor cname="gst_webrtc_dtls_transport_new">
+ <parameters>
+ <parameter name="session_id" type="guint"/>
+ <parameter name="rtcp" type="gboolean"/>
+ </parameters>
+ </constructor>
+ <method name="SetTransport" cname="gst_webrtc_dtls_transport_set_transport">
+ <return-type type="void"/>
+ <parameters>
+ <parameter name="ice" type="GstWebRTCICETransport*"/>
+ </parameters>
+ </method>
+ <property name="Certificate" cname="certificate" type="gchar*" readable="true" writeable="true" construct="false" construct-only="false"/>
+ <property name="Client" cname="client" type="gboolean" readable="true" writeable="true" construct="false" construct-only="false"/>
+ <property name="RemoteCertificate" cname="remote-certificate" type="gchar*" readable="true" writeable="false" construct="false" construct-only="false"/>
+ <property name="Rtcp" cname="rtcp" type="gboolean" readable="true" writeable="true" construct="false" construct-only="true"/>
+ <property name="SessionId" cname="session-id" type="guint" readable="true" writeable="true" construct="false" construct-only="true"/>
+ <property name="State" cname="state" type="GstWebRTCDTLSTransportState" readable="true" writeable="false" construct="false" construct-only="false"/>
+ <property name="Transport" cname="transport" type="GstWebRTCICETransport*" readable="true" writeable="false" construct="false" construct-only="false"/>
+ <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
+ <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="TransportField" type="GstWebRTCICETransport*"/>
+ <field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCDTLSTransportState"/>
+ <field cname="is_rtcp" access="public" writeable="false" readable="true" is_callback="false" name="IsRtcp" type="gboolean"/>
+ <field cname="client" access="public" writeable="false" readable="true" is_callback="false" name="ClientField" type="gboolean"/>
+ <field cname="session_id" access="public" writeable="false" readable="true" is_callback="false" name="SessionIdField" type="guint"/>
+ <field cname="dtlssrtpenc" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpenc" type="GstElement*"/>
+ <field cname="dtlssrtpdec" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpdec" type="GstElement*"/>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+ </object>
+ <object name="WebRTCICETransport" cname="GstWebRTCICETransport" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
+ <class_struct cname="GstWebRTCICETransportClass">
+ <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <method vm="gather_candidates"/>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+ </class_struct>
+ <method name="GetType" cname="gst_webrtc_ice_transport_get_type" shared="true">
+ <return-type type="GType"/>
+ </method>
+ <virtual_method name="GatherCandidates" cname="gather_candidates">
+ <return-type type="gboolean"/>
+ <parameters/>
+ </virtual_method>
+ <method name="ConnectionStateChange" cname="gst_webrtc_ice_transport_connection_state_change">
+ <return-type type="void"/>
+ <parameters>
+ <parameter name="new_state" type="GstWebRTCICEConnectionState"/>
+ </parameters>
+ </method>
+ <method name="GatheringStateChange" cname="gst_webrtc_ice_transport_gathering_state_change">
+ <return-type type="void"/>
+ <parameters>
+ <parameter name="new_state" type="GstWebRTCICEGatheringState"/>
+ </parameters>
+ </method>
+ <method name="NewCandidate" cname="gst_webrtc_ice_transport_new_candidate">
+ <return-type type="void"/>
+ <parameters>
+ <parameter name="stream_id" type="guint"/>
+ <parameter name="component" type="GstWebRTCICEComponent"/>
+ <parameter name="attr" type="const-gchar*"/>
+ </parameters>
+ </method>
+ <method name="SelectedPairChange" cname="gst_webrtc_ice_transport_selected_pair_change">
+ <return-type type="void"/>
+ <parameters/>
+ </method>
+ <property name="Component" cname="component" type="GstWebRTCICEComponent" readable="true" writeable="true" construct="false" construct-only="true"/>
+ <property name="GatheringState" cname="gathering-state" type="GstWebRTCICEGatheringState" readable="true" writeable="false" construct="false" construct-only="false"/>
+ <property name="State" cname="state" type="GstWebRTCICEConnectionState" readable="true" writeable="false" construct="false" construct-only="false"/>
+ <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
+ <field cname="role" access="public" writeable="false" readable="true" is_callback="false" name="Role" type="GstWebRTCICERole"/>
+ <field cname="component" access="public" writeable="false" readable="true" is_callback="false" name="ComponentField" type="GstWebRTCICEComponent"/>
+ <field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCICEConnectionState"/>
+ <field cname="gathering_state" access="public" writeable="false" readable="true" is_callback="false" name="GatheringStateField" type="GstWebRTCICEGatheringState"/>
+ <field cname="src" access="public" writeable="false" readable="true" is_callback="false" name="Src" type="GstElement*"/>
+ <field cname="sink" access="public" writeable="false" readable="true" is_callback="false" name="Sink" type="GstElement*"/>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+ <signal name="OnNewCandidate" cname="on-new-candidate" when="last">
+ <return-type type="void"/>
+ <parameters>
+ <parameter name="_object" type="const-gchar*"/>
+ </parameters>
+ </signal>
+ <signal name="OnSelectedCandidatePairChange" cname="on-selected-candidate-pair-change" when="last">
+ <return-type type="void"/>
+ <parameters/>
+ </signal>
+ </object>
+ <object name="WebRTCRTPReceiver" cname="GstWebRTCRTPReceiver" opaque="false" hidden="false" parent="GstObject">
+ <class_struct cname="GstWebRTCRTPReceiverClass">
+ <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+ </class_struct>
+ <method name="GetType" cname="gst_webrtc_rtp_receiver_get_type" shared="true">
+ <return-type type="GType"/>
+ </method>
+ <constructor cname="gst_webrtc_rtp_receiver_new" disable_void_ctor=""/>
+ <method name="SetRtcpTransport" cname="gst_webrtc_rtp_receiver_set_rtcp_transport">
+ <return-type type="void"/>
+ <parameters>
+ <parameter name="transport" type="GstWebRTCDTLSTransport*"/>
+ </parameters>
+ </method>
+ <method name="SetTransport" cname="gst_webrtc_rtp_receiver_set_transport">
+ <return-type type="void"/>
+ <parameters>
+ <parameter name="transport" type="GstWebRTCDTLSTransport*"/>
+ </parameters>
+ </method>
+ <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
+ <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*"/>
+ <field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*"/>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+ </object>
+ <object name="WebRTCRTPSender" cname="GstWebRTCRTPSender" opaque="false" hidden="false" parent="GstObject">
+ <class_struct cname="GstWebRTCRTPSenderClass">
+ <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+ </class_struct>
+ <method name="GetType" cname="gst_webrtc_rtp_sender_get_type" shared="true">
+ <return-type type="GType"/>
+ </method>
+ <constructor cname="gst_webrtc_rtp_sender_new" disable_void_ctor=""/>
+ <method name="SetRtcpTransport" cname="gst_webrtc_rtp_sender_set_rtcp_transport">
+ <return-type type="void"/>
+ <parameters>
+ <parameter name="transport" type="GstWebRTCDTLSTransport*"/>
+ </parameters>
+ </method>
+ <method name="SetTransport" cname="gst_webrtc_rtp_sender_set_transport">
+ <return-type type="void"/>
+ <parameters>
+ <parameter name="transport" type="GstWebRTCDTLSTransport*"/>
+ </parameters>
+ </method>
+ <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
+ <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*"/>
+ <field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*"/>
+ <field cname="send_encodings" access="public" writeable="false" readable="true" is_callback="false" name="SendEncodings" type="GArray*" array="true" null_term_array="true"/>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+ </object>
+ <object name="WebRTCRTPTransceiver" cname="GstWebRTCRTPTransceiver" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
+ <class_struct cname="GstWebRTCRTPTransceiverClass">
+ <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+ </class_struct>
+ <method name="GetType" cname="gst_webrtc_rtp_transceiver_get_type" shared="true">
+ <return-type type="GType"/>
+ </method>
+ <property name="Mlineindex" cname="mlineindex" type="guint" readable="true" writeable="true" construct="false" construct-only="true"/>
+ <property name="Receiver" cname="receiver" type="GstWebRTCRTPReceiver*" readable="true" writeable="true" construct="false" construct-only="true"/>
+ <property name="Sender" cname="sender" type="GstWebRTCRTPSender*" readable="true" writeable="true" construct="false" construct-only="true"/>
+ <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
+ <field cname="mline" access="public" writeable="false" readable="true" is_callback="false" name="Mline" type="guint"/>
+ <field cname="mid" access="public" writeable="false" readable="true" is_callback="false" name="Mid" type="gchar*"/>
+ <field cname="stopped" access="public" writeable="false" readable="true" is_callback="false" name="Stopped" type="gboolean"/>
+ <field cname="sender" access="public" writeable="false" readable="true" is_callback="false" name="SenderField" type="GstWebRTCRTPSender*"/>
+ <field cname="receiver" access="public" writeable="false" readable="true" is_callback="false" name="ReceiverField" type="GstWebRTCRTPReceiver*"/>
+ <field cname="direction" access="public" writeable="false" readable="true" is_callback="false" name="Direction" type="GstWebRTCRTPTransceiverDirection"/>
+ <field cname="current_direction" access="public" writeable="false" readable="true" is_callback="false" name="CurrentDirection" type="GstWebRTCRTPTransceiverDirection"/>
+ <field cname="codec_preferences" access="public" writeable="false" readable="true" is_callback="false" name="CodecPreferences" type="GstCaps*">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+ </object>
+ <boxed name="WebRTCSessionDescription" cname="GstWebRTCSessionDescription" opaque="false" hidden="false">
+ <method name="GetType" cname="gst_webrtc_session_description_get_type" shared="true">
+ <return-type type="GType"/>
+ </method>
+ <field cname="type" access="public" writeable="true" readable="true" is_callback="false" name="Type" type="GstWebRTCSDPType"/>
+ <field cname="sdp" access="public" writeable="true" readable="true" is_callback="false" name="Sdp" type="GstSDPMessage*">
+ <warning>missing glib:type-name</warning>
+ </field>
+ <constructor cname="gst_webrtc_session_description_new">
+ <parameters>
+ <parameter name="type" type="GstWebRTCSDPType"/>
+ <parameter name="sdp" type="GstSDPMessage*">
+ <warning>missing glib:type-name</warning>
+ </parameter>
+ </parameters>
+ </constructor>
+ <method name="Copy" cname="gst_webrtc_session_description_copy">
+ <return-type type="GstWebRTCSessionDescription*" owned="true">
+ <warning>missing glib:type-name</warning>
+ </return-type>
+ <parameters/>
+ </method>
+ <method name="Free" cname="gst_webrtc_session_description_free">
+ <return-type type="void"/>
+ <parameters/>
+ </method>
+ </boxed>
+ <object name="Global" cname="GstWebRTCGlobal" opaque="true">
+ <method name="WebrtcSdpTypeToString" cname="gst_webrtc_sdp_type_to_string" shared="true">
+ <return-type type="const-gchar*"/>
+ <parameters>
+ <parameter name="type" type="GstWebRTCSDPType"/>
+ </parameters>
+ </method>
+ </object>
+ <object name="Constants" cname="GstWebRTCConstants" opaque="true"/>
+ </namespace>
</api>
<dllmap dll="libgstrtp-1.0-0.dll" target="libgstrtp-1.0.so.0" os="linux"/>
<dllmap dll="libgstrtsp-1.0-0.dll" target="libgstrtsp-1.0.so.0" os="linux"/>
<dllmap dll="libgstsdp-1.0-0.dll" target="libgstsdp-1.0.so.0" os="linux"/>
+ <dllmap dll="libgstwebrtc-1.0-0.dll" target="libgstwebrtc-1.0.so.0" os="linux"/>
<dllmap dll="libgstcontroller-1.0-0.dll" target="libgstcontroller-1.0.so.0" os="linux"/>
<dllmap dll="libglib-2.0-0.dll" target="libglib-2.0.so.0" os="linux"/>
<dllmap dll="libgobject-2.0-0.dll" target="libgobject-2.0.so.0" os="linux"/>
<dllmap dll="libgstrtp-1.0-0.dll" target="libgstrtp-1.0.dylib" os="osx"/>
<dllmap dll="libgstrtsp-1.0-0.dll" target="libgstrtsp-1.0.dylib" os="osx"/>
<dllmap dll="libgstsdp-1.0-0.dll" target="libgstsdp-1.0.dylib" os="osx"/>
+ <dllmap dll="libgstwebrtc-1.0-0.dll" target="libgstwebrtc-1.0.dylib" os="osx"/>
<dllmap dll="libgstcontroller-1.0-0.dll" target="libgstcontroller-1.0.dylib" os="osx"/>
<dllmap dll="libglib-2.0-0.dll" target="libglib-2.0.dylib" os="osx"/>
<dllmap dll="libgobject-2.0-0.dll" target="libgobject-2.0.dylib" os="osx"/>
<attr path="/api/namespace[@name='GstRtp']" name="name">Gst.Rtp</attr>
<attr path="/api/namespace[@name='GstRtsp']" name="name">Gst.Rtsp</attr>
<attr path="/api/namespace[@name='GstSdp']" name="name">Gst.Sdp</attr>
+ <attr path="/api/namespace[@name='GstWebRTC']" name="name">Gst.WebRTC</attr>
<attr path="/api/namespace" name="library">libgstreamer-1.0-0.dll</attr>
<attr path="/api/namespace[@name='Gst.Base']" name="library">libgstbase-1.0-0.dll</attr>
<attr path="/api/namespace[@name='Gst.Rtp']" name="library">libgstrtp-1.0-0.dll</attr>
<attr path="/api/namespace[@name='Gst.Rtsp']" name="library">libgstrtsp-1.0-0.dll</attr>
<attr path="/api/namespace[@name='Gst.Sdp']" name="library">libgstsdp-1.0-0.dll</attr>
+ <attr path="/api/namespace[@name='Gst.WebRTC']" name="library">libgstwebrtc-1.0-0.dll</attr>
<!-- DoubleRange and Fraction are in Value.cs -->
<attr path="//struct[@name='DoubleRange' or @name='Fraction' or @name='IntRange' or @name='FractionRange']" name="hidden">true</attr>
raw_api_fname = join_paths(meson.current_source_dir(), meson.project_name() + '-api.raw')
metadata = files(meson.project_name() + '.metadata')
-abi_includes = 'glib.h,gst/gst.h,gst/video/video.h,gst/audio/audio.h,gst/rtsp/rtsp.h,gst/app/app.h,gst/audio/audio.h,gst/base/base.h,gst/controller/controller.h,gst/fft/fft.h,gst/net/net.h,gst/pbutils/gstaudiovisualizer.h,gst/pbutils/pbutils.h,gst/rtp/rtp.h,gst/rtsp/rtsp.h,gst/sdp/sdp.h,gst/tag/tag.h,gst/video/video.h,gst/video/gstvideoaffinetransformationmeta.h,gst/net/gstnetcontrolmessagemeta.h'
+abi_includes = 'glib.h,gst/gst.h,gst/video/video.h,gst/audio/audio.h,gst/rtsp/rtsp.h,gst/app/app.h,gst/audio/audio.h,gst/base/base.h,gst/controller/controller.h,gst/fft/fft.h,gst/net/net.h,gst/pbutils/gstaudiovisualizer.h,gst/pbutils/pbutils.h,gst/rtp/rtp.h,gst/rtsp/rtsp.h,gst/sdp/sdp.h,gst/tag/tag.h,gst/video/video.h,gst/video/gstvideoaffinetransformationmeta.h,gst/net/gstnetcontrolmessagemeta.h,gst/webrtc/webrtc.h'
sources = [
'custom/Adapter.cs',
if add_languages('c', required: false) and csc.get_id() == 'mono'
c_abi_exe = executable('gst_sharp_c_abi', c_abi,
- cs_args: ['-nowarn:169', '-nowarn:108', '-nowarn:114', '-unsafe'],
+ c_args: ['-DGST_USE_UNSTABLE_API'],
dependencies: [gst_deps])
cs_abi_exe = executable('gst_sharp_cs_abi', cs_abi,
projects / platform / upstream / gstreamer.git / commitdiff