forked from mirrors/gstreamer-rs
Update for GstWebRTCICERole enum rename
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2 changed files with 19 additions and 17 deletions
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@ -168,6 +168,8 @@ GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
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</member>
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</enumeration>
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<enumeration name="WebRTCICEComponent" c:type="GstWebRTCICEComponent">
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<doc xml:space="preserve">GST_WEBRTC_ICE_COMPONENT_RTP,
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GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
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<member name="rtp" value="0" c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP">
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</member>
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<member name="rtcp"
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@ -233,6 +235,18 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">
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c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE">
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</member>
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</enumeration>
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<enumeration name="WebRTCICERole" c:type="GstWebRTCICERole">
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<doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
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GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
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<member name="controlled"
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value="0"
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c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED">
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</member>
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<member name="controlling"
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value="1"
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c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING">
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</member>
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</enumeration>
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<class name="WebRTCICETransport"
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c:symbol-prefix="webrtc_ice_transport"
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c:type="GstWebRTCICETransport"
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@ -331,7 +345,7 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">
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<type name="Gst.Object" c:type="GstObject"/>
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</field>
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<field name="role">
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<type name="WebRTCIceRole" c:type="GstWebRTCIceRole"/>
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<type name="WebRTCICERole" c:type="GstWebRTCICERole"/>
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</field>
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<field name="component">
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<type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/>
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@ -395,18 +409,6 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">
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</array>
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</field>
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</record>
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<enumeration name="WebRTCIceRole" c:type="GstWebRTCIceRole">
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<doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
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GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
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<member name="controlled"
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value="0"
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c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED">
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</member>
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<member name="controlling"
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value="1"
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c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING">
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</member>
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</enumeration>
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<enumeration name="WebRTCPeerConnectionState"
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c:type="GstWebRTCPeerConnectionState">
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<doc xml:space="preserve">GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
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@ -50,9 +50,9 @@ pub const GST_WEBRTC_ICE_GATHERING_STATE_NEW: GstWebRTCICEGatheringState = 0;
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pub const GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: GstWebRTCICEGatheringState = 1;
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pub const GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: GstWebRTCICEGatheringState = 2;
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pub type GstWebRTCIceRole = c_int;
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pub const GST_WEBRTC_ICE_ROLE_CONTROLLED: GstWebRTCIceRole = 0;
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pub const GST_WEBRTC_ICE_ROLE_CONTROLLING: GstWebRTCIceRole = 1;
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pub type GstWebRTCICERole = c_int;
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pub const GST_WEBRTC_ICE_ROLE_CONTROLLED: GstWebRTCICERole = 0;
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pub const GST_WEBRTC_ICE_ROLE_CONTROLLING: GstWebRTCICERole = 1;
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pub type GstWebRTCPeerConnectionState = c_int;
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pub const GST_WEBRTC_PEER_CONNECTION_STATE_NEW: GstWebRTCPeerConnectionState = 0;
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@ -233,7 +233,7 @@ impl ::std::fmt::Debug for GstWebRTCDTLSTransport {
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#[derive(Copy, Clone)]
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pub struct GstWebRTCICETransport {
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pub parent: gst::GstObject,
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pub role: GstWebRTCIceRole,
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pub role: GstWebRTCICERole,
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pub component: GstWebRTCICEComponent,
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pub state: GstWebRTCICEConnectionState,
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pub gathering_state: GstWebRTCICEGatheringState,
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