gstreamer/subprojects/gstreamer-sharp/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="32"/>
      <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="34"/>
      <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_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="31"/>
      <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="33"/>
      <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="35"/>
      <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">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">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="342">none</doc>
      </member>
      <member name="balanced"
              value="1"
              c:identifier="GST_WEBRTC_BUNDLE_POLICY_BALANCED"
              glib:nick="balanced">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="343">balanced</doc>
      </member>
      <member name="max_compat"
              value="2"
              c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT"
              glib:nick="max-compat">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="344">max-compat</doc>
      </member>
      <member name="max_bundle"
              value="3"
              c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE"
              glib:nick="max-bundle">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="345">max-bundle</doc>
      </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/webrtc_fwd.h" line="46"/>
      <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="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>
    </class>
    <record name="WebRTCDTLSTransportClass"
            c:type="GstWebRTCDTLSTransportClass"
            disguised="1"
            glib:is-gtype-struct-for="WebRTCDTLSTransport">
      <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="46"/>
    </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"
           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/webrtc_fwd.h" line="63"/>
      <method name="close" c:identifier="gst_webrtc_data_channel_close">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="545">Close the @channel.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="46"/>
        <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="547">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="507">Send @data as a data message over @channel.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="40"/>
        <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="509">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="510">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="526">Send @str as a string message over @channel.</doc>
        <source-position filename="gst-libs/gst/webrtc/datachannel.h"
                         line="43"/>
        <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="528">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="529">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>
      <glib:signal name="close" when="last" action="1">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/datachannel.c"
             line="353">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="299">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="308">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="317">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="334">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="345">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"
            disguised="1"
            glib:is-gtype-struct-for="WebRTCDataChannel">
      <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="63"/>
    </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">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">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="321">new</doc>
      </member>
      <member name="connecting"
              value="1"
              c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING"
              glib:nick="connecting">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="322">connection</doc>
      </member>
      <member name="open"
              value="2"
              c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_OPEN"
              glib:nick="open">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="323">open</doc>
      </member>
      <member name="closing"
              value="3"
              c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING"
              glib:nick="closing">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="324">closing</doc>
      </member>
      <member name="closed"
              value="4"
              c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED"
              glib:nick="closed">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="325">closed</doc>
      </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/webrtc_fwd.h" line="49"/>
      <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>
      <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"
            disguised="1"
            glib:is-gtype-struct-for="WebRTCICETransport">
      <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="49"/>
    </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">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">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="362">all</doc>
      </member>
      <member name="relay"
              value="1"
              c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY"
              glib:nick="relay">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="363">relay</doc>
      </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">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">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="302">very-low</doc>
      </member>
      <member name="low"
              value="2"
              c:identifier="GST_WEBRTC_PRIORITY_TYPE_LOW"
              glib:nick="low">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="303">low</doc>
      </member>
      <member name="medium"
              value="3"
              c:identifier="GST_WEBRTC_PRIORITY_TYPE_MEDIUM"
              glib:nick="medium">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="304">medium</doc>
      </member>
      <member name="high"
              value="4"
              c:identifier="GST_WEBRTC_PRIORITY_TYPE_HIGH"
              glib:nick="high">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="305">high</doc>
      </member>
    </enumeration>
    <class name="WebRTCRTPReceiver"
           c:symbol-prefix="webrtc_rtp_receiver"
           c:type="GstWebRTCRTPReceiver"
           parent="Gst.Object"
           glib:type-name="GstWebRTCRTPReceiver"
           glib:get-type="gst_webrtc_rtp_receiver_get_type"
           glib:type-struct="WebRTCRTPReceiverClass">
      <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="52"/>
      <property name="transport" version="1.20" transfer-ownership="none">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtpreceiver.c"
             line="107">The DTLS transport for this receiver</doc>
        <type name="WebRTCDTLSTransport"/>
      </property>
    </class>
    <record name="WebRTCRTPReceiverClass"
            c:type="GstWebRTCRTPReceiverClass"
            disguised="1"
            glib:is-gtype-struct-for="WebRTCRTPReceiver">
      <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="52"/>
    </record>
    <class name="WebRTCRTPSender"
           c:symbol-prefix="webrtc_rtp_sender"
           c:type="GstWebRTCRTPSender"
           parent="Gst.Object"
           glib:type-name="GstWebRTCRTPSender"
           glib:get-type="gst_webrtc_rtp_sender_get_type"
           glib:type-struct="WebRTCRTPSenderClass">
      <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="55"/>
      <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="61">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="39"/>
        <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="63">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="64">The priority of this sender</doc>
            <type name="WebRTCPriorityType" c:type="GstWebRTCPriorityType"/>
          </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="143">The priority from which to set the DSCP field on packets</doc>
        <type name="WebRTCPriorityType"/>
      </property>
      <property name="transport" version="1.20" transfer-ownership="none">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtpsender.c"
             line="158">The DTLS transport for this sender</doc>
        <type name="WebRTCDTLSTransport"/>
      </property>
    </class>
    <record name="WebRTCRTPSenderClass"
            c:type="GstWebRTCRTPSenderClass"
            disguised="1"
            glib:is-gtype-struct-for="WebRTCRTPSender">
      <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="55"/>
    </record>
    <class name="WebRTCRTPTransceiver"
           c:symbol-prefix="webrtc_rtp_transceiver"
           c:type="GstWebRTCRTPTransceiver"
           parent="Gst.Object"
           abstract="1"
           glib:type-name="GstWebRTCRTPTransceiver"
           glib:get-type="gst_webrtc_rtp_transceiver_get_type"
           glib:type-struct="WebRTCRTPTransceiverClass">
      <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="60"/>
      <property name="codec-preferences"
                version="1.20"
                writable="1"
                transfer-ownership="none">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.c"
             line="285">Caps representing the codec preferences.</doc>
        <type name="Gst.Caps"/>
      </property>
      <property name="current-direction"
                version="1.20"
                transfer-ownership="none">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.c"
             line="253">The transceiver's current directionality, or none if the
transceiver is stopped or has never participated in an exchange
of offers and answers. To change the transceiver's
directionality, set the value of the direction property.</doc>
        <type name="WebRTCRTPTransceiverDirection"/>
      </property>
      <property name="direction"
                version="1.18"
                writable="1"
                transfer-ownership="none">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.c"
             line="216">Direction of the transceiver.</doc>
        <type name="WebRTCRTPTransceiverDirection"/>
      </property>
      <property name="kind" version="1.20" transfer-ownership="none">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.c"
             line="271">The kind of media this transceiver transports</doc>
        <type name="WebRTCKind"/>
      </property>
      <property name="mid" version="1.20" transfer-ownership="none">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/rtptransceiver.c"
             line="231">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*"/>
      </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>
    </class>
    <record name="WebRTCRTPTransceiverClass"
            c:type="GstWebRTCRTPTransceiverClass"
            disguised="1"
            glib:is-gtype-struct-for="WebRTCRTPTransceiver">
      <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="60"/>
    </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">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">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="283">new</doc>
      </member>
      <member name="connecting"
              value="1"
              c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING"
              glib:nick="connecting">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="284">connecting</doc>
      </member>
      <member name="connected"
              value="2"
              c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED"
              glib:nick="connected">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="285">connected</doc>
      </member>
      <member name="closed"
              value="3"
              c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED"
              glib:nick="closed">
        <doc xml:space="preserve"
             filename="gst-libs/gst/webrtc/webrtc_fwd.h"
             line="286">closed</doc>
      </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>
    <docsection name="gstwebrtc-datachannel">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/datachannel.c"
           line="21">&lt;https://www.w3.org/TR/webrtc/#rtcdatachannel&gt;</doc>
    </docsection>
    <docsection name="gstwebrtc-dtlstransport">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/dtlstransport.c"
           line="20">&lt;https://www.w3.org/TR/webrtc/#rtcdtlstransport&gt;</doc>
    </docsection>
    <docsection name="gstwebrtc-icetransport">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/icetransport.c"
           line="20">&lt;https://www.w3.org/TR/webrtc/#rtcicetransport&gt;</doc>
    </docsection>
    <docsection name="gstwebrtc-receiver">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/rtpreceiver.c"
           line="20">&lt;https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface&gt;</doc>
    </docsection>
    <docsection name="gstwebrtc-sender">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/rtpsender.c"
           line="20">&lt;https://www.w3.org/TR/webrtc/#rtcrtpsender-interface&gt;</doc>
    </docsection>
    <docsection name="gstwebrtc-sessiondescription">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
           line="20">&lt;https://www.w3.org/TR/webrtc/#rtcsessiondescription-class&gt;</doc>
    </docsection>
    <docsection name="gstwebrtc-transceiver">
      <doc xml:space="preserve"
           filename="gst-libs/gst/webrtc/rtptransceiver.c"
           line="20">&lt;https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface&gt;</doc>
    </docsection>
    <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>