Crate gstreamer_webrtc[−][src]

Re-exports

pub use ffi;
pub use glib;
pub use gst;
pub use gst_sdp;

Modules

prelude

Structs

WebRTCDTLSTransport

Implements

WebRTCDataChannel

This is an Abstract Base Class, you cannot instantiate it.

WebRTCICETransport

This is an Abstract Base Class, you cannot instantiate it.

WebRTCRTPReceiver

Implements

WebRTCRTPSender

Implements

WebRTCRTPTransceiver

This is an Abstract Base Class, you cannot instantiate it.

WebRTCSessionDescription

See https://www.w3.org/TR/webrtc/`rtcsessiondescription`-class

Enums

WebRTCBundlePolicyv1_16

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-24section-4.1.1 for more information.

WebRTCDTLSSetup
WebRTCDTLSTransportState
WebRTCDataChannelStatev1_16

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

WebRTCFECTypev1_14_1
WebRTCICEComponent
WebRTCICEConnectionState

See http://w3c.github.io/webrtc-pc/`dom`-rtciceconnectionstate

WebRTCICEGatheringState

See http://w3c.github.io/webrtc-pc/`dom`-rtcicegatheringstate

WebRTCICERole
WebRTCICETransportPolicyv1_16

GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24section-4.1.1 for more information.

WebRTCPeerConnectionState

See http://w3c.github.io/webrtc-pc/`dom`-rtcpeerconnectionstate

WebRTCPriorityTypev1_16

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
WebRTCSCTPTransportStatev1_16

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

See http://w3c.github.io/webrtc-pc/`rtcsdptype`

WebRTCSignalingState

See http://w3c.github.io/webrtc-pc/`dom`-rtcsignalingstate

WebRTCStatsType