<?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="WebRTCBundlePolicy"
                 version="1.16"
                 glib:type-name="GstWebRTCBundlePolicy"
                 glib:get-type="gst_webrtc_bundle_policy_get_type"
                 c:type="GstWebRTCBundlePolicy">
      <doc xml:space="preserve"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="332">GST_WEBRTC_BUNDLE_POLICY_NONE: none
GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced
GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat
GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
for more information.</doc>
      <member name="none"
              value="0"
              c:identifier="GST_WEBRTC_BUNDLE_POLICY_NONE"
              glib:nick="none">
      </member>
      <member name="balanced"
              value="1"
              c:identifier="GST_WEBRTC_BUNDLE_POLICY_BALANCED"
              glib:nick="balanced">
      </member>
      <member name="max_compat"
              value="2"
              c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT"
              glib:nick="max-compat">
      </member>
      <member name="max_bundle"
              value="3"
              c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE"
              glib:nick="max-bundle">
      </member>
    </enumeration>
    <enumeration name="WebRTCDTLSSetup"
                 glib:type-name="GstWebRTCDTLSSetup"
                 glib:get-type="gst_webrtc_dtls_setup_get_type"
                 c:type="GstWebRTCDTLSSetup">
      <doc xml:space="preserve"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="210">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">
      <source-position filename="git/gst-libs/gst/webrtc/dtlstransport.h"
                       line="59"/>
      <constructor name="new" c:identifier="gst_webrtc_dtls_transport_new">
        <source-position filename="git/gst-libs/gst/webrtc/dtlstransport.h"
                         line="62"/>
        <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">
        <source-position filename="git/gst-libs/gst/webrtc/dtlstransport.h"
                         line="65"/>
        <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" fixed-size="4">
          <type name="gpointer" c:type="gpointer"/>
        </array>
      </field>
    </class>
    <record name="WebRTCDTLSTransportClass"
            c:type="GstWebRTCDTLSTransportClass"
            glib:is-gtype-struct-for="WebRTCDTLSTransport">
      <source-position filename="git/gst-libs/gst/webrtc/dtlstransport.h"
                       line="59"/>
      <field name="parent_class">
        <type name="Gst.ObjectClass" c:type="GstObjectClass"/>
      </field>
      <field name="_padding">
        <array zero-terminated="0" 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"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="57">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="WebRTCDataChannelState"
                 version="1.16"
                 glib:type-name="GstWebRTCDataChannelState"
                 glib:get-type="gst_webrtc_data_channel_state_get_type"
                 c:type="GstWebRTCDataChannelState">
      <doc xml:space="preserve"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="311">GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new
GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection
GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate&lt;/ulink&gt;</doc>
      <member name="new"
              value="0"
              c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_NEW"
              glib:nick="new">
      </member>
      <member name="connecting"
              value="1"
              c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING"
              glib:nick="connecting">
      </member>
      <member name="open"
              value="2"
              c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_OPEN"
              glib:nick="open">
      </member>
      <member name="closing"
              value="3"
              c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING"
              glib:nick="closing">
      </member>
      <member name="closed"
              value="4"
              c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED"
              glib:nick="closed">
      </member>
    </enumeration>
    <enumeration name="WebRTCFECType"
                 version="1.14.1"
                 glib:type-name="GstWebRTCFECType"
                 glib:get-type="gst_webrtc_fec_type_get_type"
                 c:type="GstWebRTCFECType">
      <member name="none"
              value="0"
              c:identifier="GST_WEBRTC_FEC_TYPE_NONE"
              glib:nick="none">
        <doc xml:space="preserve"
             filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
             line="262">none</doc>
      </member>
      <member name="ulp_red"
              value="1"
              c:identifier="GST_WEBRTC_FEC_TYPE_ULP_RED"
              glib:nick="ulp-red">
        <doc xml:space="preserve"
             filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
             line="263">ulpfec + red</doc>
      </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"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="165">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"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="89">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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate&lt;/ulink&gt;</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"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="74">GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate&lt;/ulink&gt;</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"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="154">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">
      <source-position filename="git/gst-libs/gst/webrtc/icetransport.h"
                       line="61"/>
      <virtual-method name="gather_candidates">
        <source-position filename="git/gst-libs/gst/webrtc/icetransport.h"
                         line="58"/>
        <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">
        <source-position filename="git/gst-libs/gst/webrtc/icetransport.h"
                         line="64"/>
        <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">
        <source-position filename="git/gst-libs/gst/webrtc/icetransport.h"
                         line="67"/>
        <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">
        <source-position filename="git/gst-libs/gst/webrtc/icetransport.h"
                         line="72"/>
        <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">
        <source-position filename="git/gst-libs/gst/webrtc/icetransport.h"
                         line="70"/>
        <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" 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">
      <source-position filename="git/gst-libs/gst/webrtc/icetransport.h"
                       line="61"/>
      <field name="parent_class">
        <type name="Gst.ObjectClass" c:type="GstObjectClass"/>
      </field>
      <field name="gather_candidates">
        <callback name="gather_candidates">
          <source-position filename="git/gst-libs/gst/webrtc/icetransport.h"
                           line="58"/>
          <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" fixed-size="4">
          <type name="gpointer" c:type="gpointer"/>
        </array>
      </field>
    </record>
    <enumeration name="WebRTCICETransportPolicy"
                 version="1.16"
                 glib:type-name="GstWebRTCICETransportPolicy"
                 glib:get-type="gst_webrtc_ice_transport_policy_get_type"
                 c:type="GstWebRTCICETransportPolicy">
      <doc xml:space="preserve"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="352">GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
for more information.</doc>
      <member name="all"
              value="0"
              c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL"
              glib:nick="all">
      </member>
      <member name="relay"
              value="1"
              c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY"
              glib:nick="relay">
      </member>
    </enumeration>
    <enumeration name="WebRTCPeerConnectionState"
                 glib:type-name="GstWebRTCPeerConnectionState"
                 glib:get-type="gst_webrtc_peer_connection_state_get_type"
                 c:type="GstWebRTCPeerConnectionState">
      <doc xml:space="preserve"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="133">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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate&lt;/ulink&gt;</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>
    <enumeration name="WebRTCPriorityType"
                 version="1.16"
                 glib:type-name="GstWebRTCPriorityType"
                 glib:get-type="gst_webrtc_priority_type_get_type"
                 c:type="GstWebRTCPriorityType">
      <doc xml:space="preserve"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="292">GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low
GST_WEBRTC_PRIORITY_TYPE_LOW: low
GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium
GST_WEBRTC_PRIORITY_TYPE_HIGH: high
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype&lt;/ulink&gt;</doc>
      <member name="very_low"
              value="1"
              c:identifier="GST_WEBRTC_PRIORITY_TYPE_VERY_LOW"
              glib:nick="very-low">
      </member>
      <member name="low"
              value="2"
              c:identifier="GST_WEBRTC_PRIORITY_TYPE_LOW"
              glib:nick="low">
      </member>
      <member name="medium"
              value="3"
              c:identifier="GST_WEBRTC_PRIORITY_TYPE_MEDIUM"
              glib:nick="medium">
      </member>
      <member name="high"
              value="4"
              c:identifier="GST_WEBRTC_PRIORITY_TYPE_HIGH"
              glib:nick="high">
      </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">
      <source-position filename="git/gst-libs/gst/webrtc/rtpreceiver.h"
                       line="54"/>
      <constructor name="new" c:identifier="gst_webrtc_rtp_receiver_new">
        <source-position filename="git/gst-libs/gst/webrtc/rtpreceiver.h"
                         line="57"/>
        <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">
        <source-position filename="git/gst-libs/gst/webrtc/rtpreceiver.h"
                         line="62"/>
        <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">
        <source-position filename="git/gst-libs/gst/webrtc/rtpreceiver.h"
                         line="59"/>
        <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" fixed-size="4">
          <type name="gpointer" c:type="gpointer"/>
        </array>
      </field>
    </class>
    <record name="WebRTCRTPReceiverClass"
            c:type="GstWebRTCRTPReceiverClass"
            glib:is-gtype-struct-for="WebRTCRTPReceiver">
      <source-position filename="git/gst-libs/gst/webrtc/rtpreceiver.h"
                       line="54"/>
      <field name="parent_class">
        <type name="Gst.ObjectClass" c:type="GstObjectClass"/>
      </field>
      <field name="_padding">
        <array zero-terminated="0" 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">
      <source-position filename="git/gst-libs/gst/webrtc/rtpsender.h"
                       line="56"/>
      <constructor name="new" c:identifier="gst_webrtc_rtp_sender_new">
        <source-position filename="git/gst-libs/gst/webrtc/rtpsender.h"
                         line="59"/>
        <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">
        <source-position filename="git/gst-libs/gst/webrtc/rtpsender.h"
                         line="65"/>
        <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">
        <source-position filename="git/gst-libs/gst/webrtc/rtpsender.h"
                         line="62"/>
        <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" fixed-size="4">
          <type name="gpointer" c:type="gpointer"/>
        </array>
      </field>
    </class>
    <record name="WebRTCRTPSenderClass"
            c:type="GstWebRTCRTPSenderClass"
            glib:is-gtype-struct-for="WebRTCRTPSender">
      <source-position filename="git/gst-libs/gst/webrtc/rtpsender.h"
                       line="56"/>
      <field name="parent_class">
        <type name="Gst.ObjectClass" c:type="GstObjectClass"/>
      </field>
      <field name="_padding">
        <array zero-terminated="0" 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">
      <source-position filename="git/gst-libs/gst/webrtc/rtptransceiver.h"
                       line="62"/>
      <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" fixed-size="4">
          <type name="gpointer" c:type="gpointer"/>
        </array>
      </field>
    </class>
    <record name="WebRTCRTPTransceiverClass"
            c:type="GstWebRTCRTPTransceiverClass"
            glib:is-gtype-struct-for="WebRTCRTPTransceiver">
      <source-position filename="git/gst-libs/gst/webrtc/rtptransceiver.h"
                       line="62"/>
      <field name="parent_class">
        <type name="Gst.ObjectClass" c:type="GstObjectClass"/>
      </field>
      <field name="_padding">
        <array zero-terminated="0" 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="WebRTCSCTPTransportState"
                 version="1.16"
                 glib:type-name="GstWebRTCSCTPTransportState"
                 glib:get-type="gst_webrtc_sctp_transport_state_get_type"
                 c:type="GstWebRTCSCTPTransportState">
      <doc xml:space="preserve"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="273">GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected
GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate&lt;/ulink&gt;</doc>
      <member name="new"
              value="0"
              c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW"
              glib:nick="new">
      </member>
      <member name="connecting"
              value="1"
              c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING"
              glib:nick="connecting">
      </member>
      <member name="connected"
              value="2"
              c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED"
              glib:nick="connected">
      </member>
      <member name="closed"
              value="3"
              c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED"
              glib:nick="closed">
      </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"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="176">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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype"&gt;http://w3c.github.io/webrtc-pc/#rtcsdptype&lt;/ulink&gt;</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">
        <source-position filename="git/gst-libs/gst/webrtc/rtcsessiondescription.h"
                         line="30"/>
        <return-value transfer-ownership="none">
          <doc xml:space="preserve"
               filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
               line="41">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"
                 filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
                 line="39">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"
           filename="git/gst-libs/gst/webrtc/rtcsessiondescription.h"
           line="36">See &lt;ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class"&gt;https://www.w3.org/TR/webrtc/#rtcsessiondescription-class&lt;/ulink&gt;</doc>
      <source-position filename="git/gst-libs/gst/webrtc/rtcsessiondescription.h"
                       line="47"/>
      <field name="type" writable="1">
        <doc xml:space="preserve"
             filename="git/gst-libs/gst/webrtc/rtcsessiondescription.h"
             line="38">the #GstWebRTCSDPType of the description</doc>
        <type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
      </field>
      <field name="sdp" writable="1">
        <doc xml:space="preserve"
             filename="git/gst-libs/gst/webrtc/rtcsessiondescription.h"
             line="39">the #GstSDPMessage of the description</doc>
        <type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
      </field>
      <constructor name="new"
                   c:identifier="gst_webrtc_session_description_new">
        <source-position filename="git/gst-libs/gst/webrtc/rtcsessiondescription.h"
                         line="50"/>
        <return-value transfer-ownership="full">
          <doc xml:space="preserve"
               filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
               line="103">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"
                 filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
                 line="100">a #GstWebRTCSDPType</doc>
            <type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
          </parameter>
          <parameter name="sdp" transfer-ownership="full">
            <doc xml:space="preserve"
                 filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
                 line="101">a #GstSDPMessage</doc>
            <type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
          </parameter>
        </parameters>
      </constructor>
      <method name="copy" c:identifier="gst_webrtc_session_description_copy">
        <source-position filename="git/gst-libs/gst/webrtc/rtcsessiondescription.h"
                         line="52"/>
        <return-value transfer-ownership="full">
          <doc xml:space="preserve"
               filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
               line="65">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"
                 filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
                 line="63">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"
             filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
             line="83">Free @desc and all associated resources</doc>
        <source-position filename="git/gst-libs/gst/webrtc/rtcsessiondescription.h"
                         line="54"/>
        <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"
                 filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
                 line="85">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"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="112">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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate&lt;/ulink&gt;</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"
           filename="git/gst-libs/gst/webrtc/webrtc_fwd.h"
           line="225">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">
      <source-position filename="git/gst-libs/gst/webrtc/rtcsessiondescription.h"
                       line="30"/>
      <return-value transfer-ownership="none">
        <doc xml:space="preserve"
             filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
             line="41">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"
               filename="git/gst-libs/gst/webrtc/rtcsessiondescription.c"
               line="39">a #GstWebRTCSDPType</doc>
          <type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
        </parameter>
      </parameters>
    </function>
  </namespace>
</repository>
