gstreamer/girs/GstWebRTC-1.0.gir

<?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">
    <function-macro name="IS_WEBRTC_DATA_CHANNEL"
                    c:identifier="GST_IS_WEBRTC_DATA_CHANNEL"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="34"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_DATA_CHANNEL_CLASS"
                    c:identifier="GST_IS_WEBRTC_DATA_CHANNEL_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="36"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_DTLS_TRANSPORT"
                    c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
                       line="33"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_DTLS_TRANSPORT_CLASS"
                    c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
                       line="35"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_ICE_TRANSPORT"
                    c:identifier="GST_IS_WEBRTC_ICE_TRANSPORT"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/icetransport.h"
                       line="32"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_ICE_TRANSPORT_CLASS"
                    c:identifier="GST_IS_WEBRTC_ICE_TRANSPORT_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/icetransport.h"
                       line="34"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_RTP_RECEIVER"
                    c:identifier="GST_IS_WEBRTC_RTP_RECEIVER"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="33"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_RTP_RECEIVER_CLASS"
                    c:identifier="GST_IS_WEBRTC_RTP_RECEIVER_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="35"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_RTP_SENDER"
                    c:identifier="GST_IS_WEBRTC_RTP_SENDER"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="33"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_RTP_SENDER_CLASS"
                    c:identifier="GST_IS_WEBRTC_RTP_SENDER_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="35"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_RTP_TRANSCEIVER"
                    c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
                       line="34"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="IS_WEBRTC_RTP_TRANSCEIVER_CLASS"
                    c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
                       line="36"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_DATA_CHANNEL"
                    c:identifier="GST_WEBRTC_DATA_CHANNEL"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="33"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_DATA_CHANNEL_CLASS"
                    c:identifier="GST_WEBRTC_DATA_CHANNEL_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="35"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_DATA_CHANNEL_GET_CLASS"
                    c:identifier="GST_WEBRTC_DATA_CHANNEL_GET_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="37"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_DATA_CHANNEL_LOCK"
                    c:identifier="GST_WEBRTC_DATA_CHANNEL_LOCK"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="39"/>
      <parameters>
        <parameter name="channel">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_DATA_CHANNEL_UNLOCK"
                    c:identifier="GST_WEBRTC_DATA_CHANNEL_UNLOCK"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="40"/>
      <parameters>
        <parameter name="channel">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_DTLS_TRANSPORT"
                    c:identifier="GST_WEBRTC_DTLS_TRANSPORT"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
                       line="32"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_DTLS_TRANSPORT_CLASS"
                    c:identifier="GST_WEBRTC_DTLS_TRANSPORT_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
                       line="34"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_DTLS_TRANSPORT_GET_CLASS"
                    c:identifier="GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
                       line="36"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_ICE_TRANSPORT"
                    c:identifier="GST_WEBRTC_ICE_TRANSPORT"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/icetransport.h"
                       line="31"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_ICE_TRANSPORT_CLASS"
                    c:identifier="GST_WEBRTC_ICE_TRANSPORT_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/icetransport.h"
                       line="33"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_ICE_TRANSPORT_GET_CLASS"
                    c:identifier="GST_WEBRTC_ICE_TRANSPORT_GET_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/icetransport.h"
                       line="35"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_RTP_RECEIVER"
                    c:identifier="GST_WEBRTC_RTP_RECEIVER"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="32"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_RTP_RECEIVER_CLASS"
                    c:identifier="GST_WEBRTC_RTP_RECEIVER_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="34"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_RTP_RECEIVER_GET_CLASS"
                    c:identifier="GST_WEBRTC_RTP_RECEIVER_GET_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="36"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_RTP_SENDER"
                    c:identifier="GST_WEBRTC_RTP_SENDER"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="32"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_RTP_SENDER_CLASS"
                    c:identifier="GST_WEBRTC_RTP_SENDER_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="34"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_RTP_SENDER_GET_CLASS"
                    c:identifier="GST_WEBRTC_RTP_SENDER_GET_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="36"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_RTP_TRANSCEIVER"
                    c:identifier="GST_WEBRTC_RTP_TRANSCEIVER"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
                       line="33"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_RTP_TRANSCEIVER_CLASS"
                    c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
                       line="35"/>
      <parameters>
        <parameter name="klass">
        </parameter>
      </parameters>
    </function-macro>
    <function-macro name="WEBRTC_RTP_TRANSCEIVER_GET_CLASS"
                    c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS"
                    introspectable="0">
      <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
                       line="37"/>
      <parameters>
        <parameter name="obj">
        </parameter>
      </parameters>
    </function-macro>
    <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="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="340">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">
      <member name="none"
              value="0"
              c:identifier="GST_WEBRTC_DTLS_SETUP_NONE"
              glib:nick="none">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="220">none</doc>
      </member>
      <member name="actpass"
              value="1"
              c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS"
              glib:nick="actpass">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="221">actpass</doc>
      </member>
      <member name="active"
              value="2"
              c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE"
              glib:nick="active">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="222">sendonly</doc>
      </member>
      <member name="passive"
              value="3"
              c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE"
              glib:nick="passive">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="223">recvonly</doc>
      </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="gst-libs/gst/webrtc/dtlstransport.h"
                       line="62"/>
      <constructor name="new" c:identifier="gst_webrtc_dtls_transport_new">
        <source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
                         line="65"/>
        <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="gst-libs/gst/webrtc/dtlstransport.h"
                         line="68"/>
        <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="gst-libs/gst/webrtc/dtlstransport.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="WebRTCDTLSTransportState"
                 glib:type-name="GstWebRTCDTLSTransportState"
                 glib:get-type="gst_webrtc_dtls_transport_state_get_type"
                 c:type="GstWebRTCDTLSTransportState">
      <member name="new"
              value="0"
              c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW"
              glib:nick="new">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="67">new</doc>
      </member>
      <member name="closed"
              value="1"
              c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED"
              glib:nick="closed">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="68">closed</doc>
      </member>
      <member name="failed"
              value="2"
              c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED"
              glib:nick="failed">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="69">failed</doc>
      </member>
      <member name="connecting"
              value="3"
              c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING"
              glib:nick="connecting">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="70">connecting</doc>
      </member>
      <member name="connected"
              value="4"
              c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED"
              glib:nick="connected">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="71">connected</doc>
      </member>
    </enumeration>
    <class name="WebRTCDataChannel"
           c:symbol-prefix="webrtc_data_channel"
           c:type="GstWebRTCDataChannel"
           version="1.18"
           parent="GObject.Object"
           abstract="1"
           glib:type-name="GstWebRTCDataChannel"
           glib:get-type="gst_webrtc_data_channel_get_type"
           glib:type-struct="WebRTCDataChannelClass">
      <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="82"/>
      <virtual-method name="close" invoker="close">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="544">Close the @channel.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="79"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="546">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
        </parameters>
      </virtual-method>
      <virtual-method name="send_data" invoker="send_data">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="506">Send @data as a data message over @channel.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="77"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="508">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
          <parameter name="data"
                     transfer-ownership="none"
                     nullable="1"
                     allow-none="1">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="509">a #GBytes or %NULL</doc>
            <type name="GLib.Bytes" c:type="GBytes*"/>
          </parameter>
        </parameters>
      </virtual-method>
      <virtual-method name="send_string" invoker="send_string">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="525">Send @str as a string message over @channel.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="78"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="527">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
          <parameter name="str"
                     transfer-ownership="none"
                     nullable="1"
                     allow-none="1">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="528">a string or %NULL</doc>
            <type name="utf8" c:type="const gchar*"/>
          </parameter>
        </parameters>
      </virtual-method>
      <method name="close" c:identifier="gst_webrtc_data_channel_close">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="544">Close the @channel.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="109"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="546">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
        </parameters>
      </method>
      <method name="on_buffered_amount_low"
              c:identifier="gst_webrtc_data_channel_on_buffered_amount_low">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="490">Signal that the data channel reached a low buffered amount. Should only be used by subclasses.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="100"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="492">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
        </parameters>
      </method>
      <method name="on_close" c:identifier="gst_webrtc_data_channel_on_close">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="405">Signal that the data channel was closed. Should only be used by subclasses.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="88"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="407">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
        </parameters>
      </method>
      <method name="on_error" c:identifier="gst_webrtc_data_channel_on_error">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="434">Signal that the data channel had an error. Should only be used by subclasses.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="91"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="436">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
          <parameter name="error" transfer-ownership="full">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="437">a #GError</doc>
            <type name="GLib.Error" c:type="GError*"/>
          </parameter>
        </parameters>
      </method>
      <method name="on_message_data"
              c:identifier="gst_webrtc_data_channel_on_message_data">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="454">Signal that the data channel received a data message. Should only be used by subclasses.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="94"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="456">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
          <parameter name="data"
                     transfer-ownership="none"
                     nullable="1"
                     allow-none="1">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="457">a #GBytes or %NULL</doc>
            <type name="GLib.Bytes" c:type="GBytes*"/>
          </parameter>
        </parameters>
      </method>
      <method name="on_message_string"
              c:identifier="gst_webrtc_data_channel_on_message_string">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="472">Signal that the data channel received a string message. Should only be used by subclasses.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="97"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="474">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
          <parameter name="str"
                     transfer-ownership="none"
                     nullable="1"
                     allow-none="1">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="475">a string or %NULL</doc>
            <type name="utf8" c:type="const gchar*"/>
          </parameter>
        </parameters>
      </method>
      <method name="on_open" c:identifier="gst_webrtc_data_channel_on_open">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="371">Signal that the data channel was opened. Should only be used by subclasses.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="85"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="373">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
        </parameters>
      </method>
      <method name="send_data"
              c:identifier="gst_webrtc_data_channel_send_data">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="506">Send @data as a data message over @channel.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="103"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="508">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
          <parameter name="data"
                     transfer-ownership="none"
                     nullable="1"
                     allow-none="1">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="509">a #GBytes or %NULL</doc>
            <type name="GLib.Bytes" c:type="GBytes*"/>
          </parameter>
        </parameters>
      </method>
      <method name="send_string"
              c:identifier="gst_webrtc_data_channel_send_string">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="525">Send @str as a string message over @channel.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="106"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="channel" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="527">a #GstWebRTCDataChannel</doc>
            <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
          </instance-parameter>
          <parameter name="str"
                     transfer-ownership="none"
                     nullable="1"
                     allow-none="1">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="528">a string or %NULL</doc>
            <type name="utf8" c:type="const gchar*"/>
          </parameter>
        </parameters>
      </method>
      <property name="buffered-amount" transfer-ownership="none">
        <type name="guint64" c:type="guint64"/>
      </property>
      <property name="buffered-amount-low-threshold"
                writable="1"
                transfer-ownership="none">
        <type name="guint64" c:type="guint64"/>
      </property>
      <property name="id"
                writable="1"
                construct-only="1"
                transfer-ownership="none">
        <type name="gint" c:type="gint"/>
      </property>
      <property name="label"
                writable="1"
                construct-only="1"
                transfer-ownership="none">
        <type name="utf8" c:type="gchar*"/>
      </property>
      <property name="max-packet-lifetime"
                writable="1"
                construct-only="1"
                transfer-ownership="none">
        <type name="gint" c:type="gint"/>
      </property>
      <property name="max-retransmits"
                writable="1"
                construct-only="1"
                transfer-ownership="none">
        <type name="gint" c:type="gint"/>
      </property>
      <property name="negotiated"
                writable="1"
                construct-only="1"
                transfer-ownership="none">
        <type name="gboolean" c:type="gboolean"/>
      </property>
      <property name="ordered"
                writable="1"
                construct-only="1"
                transfer-ownership="none">
        <type name="gboolean" c:type="gboolean"/>
      </property>
      <property name="priority"
                writable="1"
                construct-only="1"
                transfer-ownership="none">
        <type name="WebRTCPriorityType"/>
      </property>
      <property name="protocol"
                writable="1"
                construct-only="1"
                transfer-ownership="none">
        <type name="utf8" c:type="gchar*"/>
      </property>
      <property name="ready-state" transfer-ownership="none">
        <type name="WebRTCDataChannelState"/>
      </property>
      <field name="parent">
        <type name="GObject.Object" c:type="GObject"/>
      </field>
      <field name="lock">
        <type name="GLib.Mutex" c:type="GMutex"/>
      </field>
      <field name="label">
        <type name="utf8" c:type="gchar*"/>
      </field>
      <field name="ordered">
        <type name="gboolean" c:type="gboolean"/>
      </field>
      <field name="max_packet_lifetime">
        <type name="guint" c:type="guint"/>
      </field>
      <field name="max_retransmits">
        <type name="guint" c:type="guint"/>
      </field>
      <field name="protocol">
        <type name="utf8" c:type="gchar*"/>
      </field>
      <field name="negotiated">
        <type name="gboolean" c:type="gboolean"/>
      </field>
      <field name="id">
        <type name="gint" c:type="gint"/>
      </field>
      <field name="priority">
        <type name="WebRTCPriorityType" c:type="GstWebRTCPriorityType"/>
      </field>
      <field name="ready_state">
        <type name="WebRTCDataChannelState"
              c:type="GstWebRTCDataChannelState"/>
      </field>
      <field name="buffered_amount">
        <type name="guint64" c:type="guint64"/>
      </field>
      <field name="buffered_amount_low_threshold">
        <type name="guint64" c:type="guint64"/>
      </field>
      <field name="_padding">
        <array zero-terminated="0" fixed-size="4">
          <type name="gpointer" c:type="gpointer"/>
        </array>
      </field>
      <glib:signal name="close" when="last" action="1">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="352">Close the data channel</doc>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
      </glib:signal>
      <glib:signal name="on-buffered-amount-low" when="last">
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
      </glib:signal>
      <glib:signal name="on-close" when="last">
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
      </glib:signal>
      <glib:signal name="on-error" when="last">
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <parameter name="error" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="298">the #GError thrown</doc>
            <type name="GLib.Error"/>
          </parameter>
        </parameters>
      </glib:signal>
      <glib:signal name="on-message-data" when="last">
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <parameter name="data"
                     transfer-ownership="none"
                     nullable="1"
                     allow-none="1">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="307">a #GBytes of the data received</doc>
            <type name="GLib.Bytes"/>
          </parameter>
        </parameters>
      </glib:signal>
      <glib:signal name="on-message-string" when="last">
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <parameter name="data"
                     transfer-ownership="none"
                     nullable="1"
                     allow-none="1">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="316">the data received as a string</doc>
            <type name="utf8" c:type="gchar*"/>
          </parameter>
        </parameters>
      </glib:signal>
      <glib:signal name="on-open" when="last">
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
      </glib:signal>
      <glib:signal name="send-data" when="last" action="1">
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <parameter name="data"
                     transfer-ownership="none"
                     nullable="1"
                     allow-none="1">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="333">a #GBytes with the data</doc>
            <type name="GLib.Bytes"/>
          </parameter>
        </parameters>
      </glib:signal>
      <glib:signal name="send-string" when="last" action="1">
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <parameter name="data"
                     transfer-ownership="none"
                     nullable="1"
                     allow-none="1">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/datachannel.c"
                 line="344">the data to send as a string</doc>
            <type name="utf8" c:type="gchar*"/>
          </parameter>
        </parameters>
      </glib:signal>
    </class>
    <record name="WebRTCDataChannelClass"
            c:type="GstWebRTCDataChannelClass"
            glib:is-gtype-struct-for="WebRTCDataChannel"
            version="1.18">
      <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="82"/>
      <field name="parent_class">
        <type name="GObject.ObjectClass" c:type="GObjectClass"/>
      </field>
      <field name="send_data">
        <callback name="send_data">
          <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                           line="77"/>
          <return-value transfer-ownership="none">
            <type name="none" c:type="void"/>
          </return-value>
          <parameters>
            <parameter name="channel" transfer-ownership="none">
              <doc xml:space="preserve"
                   filename="gst-libs/gst/webrtc/datachannel.c"
                   line="508">a #GstWebRTCDataChannel</doc>
              <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
            </parameter>
            <parameter name="data"
                       transfer-ownership="none"
                       nullable="1"
                       allow-none="1">
              <doc xml:space="preserve"
                   filename="gst-libs/gst/webrtc/datachannel.c"
                   line="509">a #GBytes or %NULL</doc>
              <type name="GLib.Bytes" c:type="GBytes*"/>
            </parameter>
          </parameters>
        </callback>
      </field>
      <field name="send_string">
        <callback name="send_string">
          <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                           line="78"/>
          <return-value transfer-ownership="none">
            <type name="none" c:type="void"/>
          </return-value>
          <parameters>
            <parameter name="channel" transfer-ownership="none">
              <doc xml:space="preserve"
                   filename="gst-libs/gst/webrtc/datachannel.c"
                   line="527">a #GstWebRTCDataChannel</doc>
              <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
            </parameter>
            <parameter name="str"
                       transfer-ownership="none"
                       nullable="1"
                       allow-none="1">
              <doc xml:space="preserve"
                   filename="gst-libs/gst/webrtc/datachannel.c"
                   line="528">a string or %NULL</doc>
              <type name="utf8" c:type="const gchar*"/>
            </parameter>
          </parameters>
        </callback>
      </field>
      <field name="close">
        <callback name="close">
          <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                           line="79"/>
          <return-value transfer-ownership="none">
            <type name="none" c:type="void"/>
          </return-value>
          <parameters>
            <parameter name="channel" transfer-ownership="none">
              <doc xml:space="preserve"
                   filename="gst-libs/gst/webrtc/datachannel.c"
                   line="546">a #GstWebRTCDataChannel</doc>
              <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
            </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="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="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="319">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;http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate&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="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="270">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="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="271">ulpfec + red</doc>
      </member>
    </enumeration>
    <enumeration name="WebRTCICEComponent"
                 glib:type-name="GstWebRTCICEComponent"
                 glib:get-type="gst_webrtc_ice_component_get_type"
                 c:type="GstWebRTCICEComponent">
      <member name="rtp"
              value="0"
              c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP"
              glib:nick="rtp">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="175">RTP component</doc>
      </member>
      <member name="rtcp"
              value="1"
              c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP"
              glib:nick="rtcp">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="176">RTCP component</doc>
      </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="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="97">See &lt;http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate&gt;</doc>
      <member name="new"
              value="0"
              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW"
              glib:nick="new">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="99">new</doc>
      </member>
      <member name="checking"
              value="1"
              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING"
              glib:nick="checking">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="100">checking</doc>
      </member>
      <member name="connected"
              value="2"
              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED"
              glib:nick="connected">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="101">connected</doc>
      </member>
      <member name="completed"
              value="3"
              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED"
              glib:nick="completed">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="102">completed</doc>
      </member>
      <member name="failed"
              value="4"
              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED"
              glib:nick="failed">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="103">failed</doc>
      </member>
      <member name="disconnected"
              value="5"
              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED"
              glib:nick="disconnected">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="104">disconnected</doc>
      </member>
      <member name="closed"
              value="6"
              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED"
              glib:nick="closed">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="105">closed</doc>
      </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="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="82">See &lt;http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate&gt;</doc>
      <member name="new"
              value="0"
              c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW"
              glib:nick="new">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="84">new</doc>
      </member>
      <member name="gathering"
              value="1"
              c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING"
              glib:nick="gathering">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="85">gathering</doc>
      </member>
      <member name="complete"
              value="2"
              c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE"
              glib:nick="complete">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="86">complete</doc>
      </member>
    </enumeration>
    <enumeration name="WebRTCICERole"
                 glib:type-name="GstWebRTCICERole"
                 glib:get-type="gst_webrtc_ice_role_get_type"
                 c:type="GstWebRTCICERole">
      <member name="controlled"
              value="0"
              c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED"
              glib:nick="controlled">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="164">controlled</doc>
      </member>
      <member name="controlling"
              value="1"
              c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING"
              glib:nick="controlling">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="165">controlling</doc>
      </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="gst-libs/gst/webrtc/icetransport.h"
                       line="64"/>
      <virtual-method name="gather_candidates">
        <source-position filename="gst-libs/gst/webrtc/icetransport.h"
                         line="61"/>
        <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="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="WebRTCICEConnectionState"
                  c:type="GstWebRTCICEConnectionState"/>
          </parameter>
        </parameters>
      </method>
      <method name="gathering_state_change"
              c:identifier="gst_webrtc_ice_transport_gathering_state_change">
        <source-position filename="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>
          <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="gst-libs/gst/webrtc/icetransport.h"
                         line="75"/>
        <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="gst-libs/gst/webrtc/icetransport.h"
                         line="73"/>
        <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="gst-libs/gst/webrtc/icetransport.h"
                       line="64"/>
      <field name="parent_class">
        <type name="Gst.ObjectClass" c:type="GstObjectClass"/>
      </field>
      <field name="gather_candidates">
        <callback name="gather_candidates">
          <source-position filename="gst-libs/gst/webrtc/icetransport.h"
                           line="61"/>
          <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="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="360">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="WebRTCKind"
                 version="1.20"
                 glib:type-name="GstWebRTCKind"
                 glib:get-type="gst_webrtc_kind_get_type"
                 c:type="GstWebRTCKind">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="376">https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind</doc>
      <member name="unknown"
              value="0"
              c:identifier="GST_WEBRTC_KIND_UNKNOWN"
              glib:nick="unknown">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="378">Kind has not yet been set</doc>
      </member>
      <member name="audio"
              value="1"
              c:identifier="GST_WEBRTC_KIND_AUDIO"
              glib:nick="audio">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="379">Kind is audio</doc>
      </member>
      <member name="video"
              value="2"
              c:identifier="GST_WEBRTC_KIND_VIDEO"
              glib:nick="video">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="380">Kind is audio</doc>
      </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="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="141">See &lt;http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate&gt;</doc>
      <member name="new"
              value="0"
              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW"
              glib:nick="new">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="143">new</doc>
      </member>
      <member name="connecting"
              value="1"
              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING"
              glib:nick="connecting">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="144">connecting</doc>
      </member>
      <member name="connected"
              value="2"
              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED"
              glib:nick="connected">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="145">connected</doc>
      </member>
      <member name="disconnected"
              value="3"
              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED"
              glib:nick="disconnected">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="146">disconnected</doc>
      </member>
      <member name="failed"
              value="4"
              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED"
              glib:nick="failed">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="147">failed</doc>
      </member>
      <member name="closed"
              value="5"
              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED"
              glib:nick="closed">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="148">closed</doc>
      </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="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="300">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;http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype&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"
           version="1.16"
           parent="Gst.Object"
           glib:type-name="GstWebRTCRTPReceiver"
           glib:get-type="gst_webrtc_rtp_receiver_get_type"
           glib:type-struct="WebRTCRTPReceiverClass">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/rtpreceiver.h"
           line="38">An object to track the receiving aspect of the stream

Mostly matches the WebRTC RTCRtpReceiver interface.</doc>
      <source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="65"/>
      <constructor name="new" c:identifier="gst_webrtc_rtp_receiver_new">
        <source-position filename="gst-libs/gst/webrtc/rtpreceiver.h"
                         line="68"/>
        <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="gst-libs/gst/webrtc/rtpreceiver.h"
                         line="73"/>
        <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="gst-libs/gst/webrtc/rtpreceiver.h"
                         line="70"/>
        <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">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtpreceiver.h"
             line="40">The transport for RTP packets</doc>
        <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
      </field>
      <field name="rtcp_transport">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtpreceiver.h"
             line="41">The transport for RTCP packets without rtcp-mux</doc>
        <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="gst-libs/gst/webrtc/rtpreceiver.h" line="65"/>
      <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"
           version="1.16"
           parent="Gst.Object"
           glib:type-name="GstWebRTCRTPSender"
           glib:get-type="gst_webrtc_rtp_sender_get_type"
           glib:type-struct="WebRTCRTPSenderClass">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/rtpsender.h"
           line="38">An object to track the sending aspect of the stream

Mostly matches the WebRTC RTCRtpSender interface.</doc>
      <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="70"/>
      <constructor name="new" c:identifier="gst_webrtc_rtp_sender_new">
        <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="73"/>
        <return-value transfer-ownership="none">
          <type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
        </return-value>
      </constructor>
      <method name="set_priority"
              c:identifier="gst_webrtc_rtp_sender_set_priority"
              version="1.20">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtpsender.c"
             line="85">Sets the content of the IPv4 Type of Service (ToS), also known as DSCP
(Differentiated Services Code Point).
This also sets the Traffic Class field of IPv6.</doc>
        <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="82"/>
        <return-value transfer-ownership="none">
          <type name="none" c:type="void"/>
        </return-value>
        <parameters>
          <instance-parameter name="sender" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/rtpsender.c"
                 line="87">a #GstWebRTCRTPSender</doc>
            <type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
          </instance-parameter>
          <parameter name="priority" transfer-ownership="none">
            <doc xml:space="preserve"
                 filename="gst-libs/gst/webrtc/rtpsender.c"
                 line="88">The priority of this sender</doc>
            <type name="WebRTCPriorityType" c:type="GstWebRTCPriorityType"/>
          </parameter>
        </parameters>
      </method>
      <method name="set_rtcp_transport"
              c:identifier="gst_webrtc_rtp_sender_set_rtcp_transport">
        <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="79"/>
        <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="gst-libs/gst/webrtc/rtpsender.h" line="76"/>
        <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>
      <property name="priority"
                version="1.20"
                writable="1"
                transfer-ownership="none">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtpsender.c"
             line="166">The priority from which to set the DSCP field on packets</doc>
        <type name="WebRTCPriorityType"/>
      </property>
      <field name="parent">
        <type name="Gst.Object" c:type="GstObject"/>
      </field>
      <field name="transport">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtpsender.h"
             line="40">The transport for RTP packets</doc>
        <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
      </field>
      <field name="rtcp_transport">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtpsender.h"
             line="41">The transport for RTCP packets without rtcp-mux</doc>
        <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
      </field>
      <field name="send_encodings">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtpsender.h"
             line="42">Unused</doc>
        <array name="GLib.Array" c:type="GArray*">
          <type name="gpointer" c:type="gpointer"/>
        </array>
      </field>
      <field name="priority">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtpsender.h"
             line="43">The priority of the stream (Since: 1.20)</doc>
        <type name="WebRTCPriorityType" c:type="GstWebRTCPriorityType"/>
      </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="gst-libs/gst/webrtc/rtpsender.h" line="70"/>
      <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"
           version="1.16"
           parent="Gst.Object"
           abstract="1"
           glib:type-name="GstWebRTCRTPTransceiver"
           glib:get-type="gst_webrtc_rtp_transceiver_get_type"
           glib:type-struct="WebRTCRTPTransceiverClass">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/rtptransceiver.h"
           line="39">Mostly matches the WebRTC RTCRtpTransceiver interface.</doc>
      <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
                       line="96"/>
      <property name="direction"
                version="1.18"
                writable="1"
                transfer-ownership="none">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.c"
             line="188">Direction of the transceiver.</doc>
        <type name="WebRTCRTPTransceiverDirection"/>
      </property>
      <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">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.h"
             line="41">the mline number this transceiver corresponds to</doc>
        <type name="guint" c:type="guint"/>
      </field>
      <field name="mid">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.h"
             line="42">The media ID of the m-line associated with this
transceiver. This association is established, when possible,
whenever either a local or remote description is applied. This
field is NULL if neither a local or remote description has been
applied, or if its associated m-line is rejected by either a remote
offer or any answer.</doc>
        <type name="utf8" c:type="gchar*"/>
      </field>
      <field name="stopped">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.h"
             line="48">Indicates whether or not sending and receiving using the paired
#GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled,
either due to SDP offer/answer</doc>
        <type name="gboolean" c:type="gboolean"/>
      </field>
      <field name="sender">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.h"
             line="51">The #GstWebRTCRTPSender object responsible sending  data to the
remote peer</doc>
        <type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
      </field>
      <field name="receiver">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.h"
             line="53">The #GstWebRTCRTPReceiver object responsible for receiver data from
the remote peer.</doc>
        <type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
      </field>
      <field name="direction">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.h"
             line="55">The transceiver's desired direction.</doc>
        <type name="WebRTCRTPTransceiverDirection"
              c:type="GstWebRTCRTPTransceiverDirection"/>
      </field>
      <field name="current_direction">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.h"
             line="56">The transceiver's current direction (read-only)</doc>
        <type name="WebRTCRTPTransceiverDirection"
              c:type="GstWebRTCRTPTransceiverDirection"/>
      </field>
      <field name="codec_preferences">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.h"
             line="57">A caps representing the codec preferences (read-only)</doc>
        <type name="Gst.Caps" c:type="GstCaps*"/>
      </field>
      <field name="kind">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.h"
             line="58">Type of media (Since: 1.20)</doc>
        <type name="WebRTCKind" c:type="GstWebRTCKind"/>
      </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="gst-libs/gst/webrtc/rtptransceiver.h"
                       line="96"/>
      <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">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="203">none</doc>
      </member>
      <member name="inactive"
              value="1"
              c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE"
              glib:nick="inactive">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="204">inactive</doc>
      </member>
      <member name="sendonly"
              value="2"
              c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY"
              glib:nick="sendonly">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="205">sendonly</doc>
      </member>
      <member name="recvonly"
              value="3"
              c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY"
              glib:nick="recvonly">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="206">recvonly</doc>
      </member>
      <member name="sendrecv"
              value="4"
              c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV"
              glib:nick="sendrecv">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="207">sendrecv</doc>
      </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="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="281">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;http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate&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="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="184">See &lt;http://w3c.github.io/webrtc-pc/#rtcsdptype&gt;</doc>
      <member name="offer"
              value="1"
              c:identifier="GST_WEBRTC_SDP_TYPE_OFFER"
              glib:nick="offer">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="186">offer</doc>
      </member>
      <member name="pranswer"
              value="2"
              c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER"
              glib:nick="pranswer">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="187">pranswer</doc>
      </member>
      <member name="answer"
              value="3"
              c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER"
              glib:nick="answer">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="188">answer</doc>
      </member>
      <member name="rollback"
              value="4"
              c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK"
              glib:nick="rollback">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="189">rollback</doc>
      </member>
      <function name="to_string" c:identifier="gst_webrtc_sdp_type_to_string">
        <source-position filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
                         line="30"/>
        <return-value transfer-ownership="none">
          <doc xml:space="preserve"
               filename="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="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="gst-libs/gst/webrtc/rtcsessiondescription.h"
           line="36">See &lt;https://www.w3.org/TR/webrtc/#rtcsessiondescription-class&gt;</doc>
      <source-position filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
                       line="47"/>
      <field name="type" writable="1">
        <doc xml:space="preserve"
             filename="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="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="gst-libs/gst/webrtc/rtcsessiondescription.h"
                         line="50"/>
        <return-value transfer-ownership="full">
          <doc xml:space="preserve"
               filename="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="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="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="gst-libs/gst/webrtc/rtcsessiondescription.h"
                         line="52"/>
        <return-value transfer-ownership="full">
          <doc xml:space="preserve"
               filename="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="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="gst-libs/gst/webrtc/rtcsessiondescription.c"
             line="83">Free @desc and all associated resources</doc>
        <source-position filename="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="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="gst-libs/gst/webrtc/webrtc_fwd.h"
           line="120">See &lt;http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate&gt;</doc>
      <member name="stable"
              value="0"
              c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE"
              glib:nick="stable">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="122">stable</doc>
      </member>
      <member name="closed"
              value="1"
              c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED"
              glib:nick="closed">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="123">closed</doc>
      </member>
      <member name="have_local_offer"
              value="2"
              c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER"
              glib:nick="have-local-offer">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="124">have-local-offer</doc>
      </member>
      <member name="have_remote_offer"
              value="3"
              c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER"
              glib:nick="have-remote-offer">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="125">have-remote-offer</doc>
      </member>
      <member name="have_local_pranswer"
              value="4"
              c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER"
              glib:nick="have-local-pranswer">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="126">have-local-pranswer</doc>
      </member>
      <member name="have_remote_pranswer"
              value="5"
              c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER"
              glib:nick="have-remote-pranswer">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="127">have-remote-pranswer</doc>
      </member>
    </enumeration>
    <enumeration name="WebRTCStatsType"
                 glib:type-name="GstWebRTCStatsType"
                 glib:get-type="gst_webrtc_stats_type_get_type"
                 c:type="GstWebRTCStatsType">
      <member name="codec"
              value="1"
              c:identifier="GST_WEBRTC_STATS_CODEC"
              glib:nick="codec">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="235">codec</doc>
      </member>
      <member name="inbound_rtp"
              value="2"
              c:identifier="GST_WEBRTC_STATS_INBOUND_RTP"
              glib:nick="inbound-rtp">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="236">inbound-rtp</doc>
      </member>
      <member name="outbound_rtp"
              value="3"
              c:identifier="GST_WEBRTC_STATS_OUTBOUND_RTP"
              glib:nick="outbound-rtp">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="237">outbound-rtp</doc>
      </member>
      <member name="remote_inbound_rtp"
              value="4"
              c:identifier="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP"
              glib:nick="remote-inbound-rtp">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="238">remote-inbound-rtp</doc>
      </member>
      <member name="remote_outbound_rtp"
              value="5"
              c:identifier="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP"
              glib:nick="remote-outbound-rtp">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="239">remote-outbound-rtp</doc>
      </member>
      <member name="csrc"
              value="6"
              c:identifier="GST_WEBRTC_STATS_CSRC"
              glib:nick="csrc">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="240">csrc</doc>
      </member>
      <member name="peer_connection"
              value="7"
              c:identifier="GST_WEBRTC_STATS_PEER_CONNECTION"
              glib:nick="peer-connection">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="241">peer-connectiion</doc>
      </member>
      <member name="data_channel"
              value="8"
              c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL"
              glib:nick="data-channel">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="242">data-channel</doc>
      </member>
      <member name="stream"
              value="9"
              c:identifier="GST_WEBRTC_STATS_STREAM"
              glib:nick="stream">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="243">stream</doc>
      </member>
      <member name="transport"
              value="10"
              c:identifier="GST_WEBRTC_STATS_TRANSPORT"
              glib:nick="transport">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="244">transport</doc>
      </member>
      <member name="candidate_pair"
              value="11"
              c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR"
              glib:nick="candidate-pair">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="245">candidate-pair</doc>
      </member>
      <member name="local_candidate"
              value="12"
              c:identifier="GST_WEBRTC_STATS_LOCAL_CANDIDATE"
              glib:nick="local-candidate">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="246">local-candidate</doc>
      </member>
      <member name="remote_candidate"
              value="13"
              c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE"
              glib:nick="remote-candidate">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="247">remote-candidate</doc>
      </member>
      <member name="certificate"
              value="14"
              c:identifier="GST_WEBRTC_STATS_CERTIFICATE"
              glib:nick="certificate">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="248">certificate</doc>
      </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="gst-libs/gst/webrtc/rtcsessiondescription.h"
                       line="30"/>
      <return-value transfer-ownership="none">
        <doc xml:space="preserve"
             filename="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="gst-libs/gst/webrtc/rtcsessiondescription.c"
               line="39">a #GstWebRTCSDPType</doc>
          <type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
        </parameter>
      </parameters>
    </function>
  </namespace>
</repository>