[−][src]Crate gstreamer_webrtc
Modules
| prelude |
Structs
Enums
| WebRTCBundlePolicy | 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 |
| WebRTCDTLSSetup | |
| WebRTCDTLSTransportState | |
| WebRTCDataChannelState | 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 http://w3c.github.io/webrtc-pc/`dom`-rtcdatachannelstate |
| WebRTCFECType | |
| WebRTCICEComponent | |
| WebRTCICEConnectionState | See http://w3c.github.io/webrtc-pc/`dom`-rtciceconnectionstate |
| WebRTCICEGatheringState | See http://w3c.github.io/webrtc-pc/`dom`-rtcicegatheringstate |
| WebRTCICERole | |
| WebRTCICETransportPolicy | 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 |
| WebRTCPeerConnectionState | See http://w3c.github.io/webrtc-pc/`dom`-rtcpeerconnectionstate |
| WebRTCPriorityType | 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 http://w3c.github.io/webrtc-pc/`dom`-rtcprioritytype |
| WebRTCRTPTransceiverDirection | |
| WebRTCSCTPTransportState | 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 http://w3c.github.io/webrtc-pc/`dom`-rtcsctptransportstate |
| WebRTCSDPType | |
| WebRTCSignalingState | |
| WebRTCStatsType |