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Add missing versions to new WebRTC enum types
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@ -16,6 +16,7 @@ and/or use gtk-doc annotations. -->
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c:identifier-prefixes="Gst"
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c:symbol-prefixes="gst">
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<enumeration name="WebRTCBundlePolicy"
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version="1.16"
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glib:type-name="GstWebRTCBundlePolicy"
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glib:get-type="gst_webrtc_bundle_policy_get_type"
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c:type="GstWebRTCBundlePolicy">
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@ -215,6 +216,7 @@ GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
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</member>
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</enumeration>
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<enumeration name="WebRTCDataChannelState"
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version="1.16"
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glib:type-name="GstWebRTCDataChannelState"
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glib:get-type="gst_webrtc_data_channel_state_get_type"
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c:type="GstWebRTCDataChannelState">
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@ -251,6 +253,7 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate">h
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</member>
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</enumeration>
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<enumeration name="WebRTCFECType"
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version="1.14.1"
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glib:type-name="GstWebRTCFECType"
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glib:get-type="gst_webrtc_fec_type_get_type"
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c:type="GstWebRTCFECType">
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@ -533,6 +536,7 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
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</field>
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</record>
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<enumeration name="WebRTCICETransportPolicy"
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version="1.16"
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glib:type-name="GstWebRTCICETransportPolicy"
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glib:get-type="gst_webrtc_ice_transport_policy_get_type"
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c:type="GstWebRTCICETransportPolicy">
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@ -594,6 +598,7 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g
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</member>
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</enumeration>
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<enumeration name="WebRTCPriorityType"
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version="1.16"
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glib:type-name="GstWebRTCPriorityType"
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glib:get-type="gst_webrtc_priority_type_get_type"
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c:type="GstWebRTCPriorityType">
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@ -866,6 +871,7 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype">http:
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</member>
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</enumeration>
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<enumeration name="WebRTCSCTPTransportState"
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version="1.16"
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glib:type-name="GstWebRTCSCTPTransportState"
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glib:get-type="gst_webrtc_sctp_transport_state_get_type"
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c:type="GstWebRTCSCTPTransportState">
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