[][src]Crate gstreamer_webrtc

Modules

prelude

Structs

WebRTCDTLSTransport

Implements

WebRTCDTLSTransportClass
WebRTCICETransport

Implements

WebRTCICETransportClass
WebRTCRTPReceiver

Implements

WebRTCRTPReceiverClass
WebRTCRTPSender

Implements

WebRTCRTPSenderClass
WebRTCRTPTransceiver

Implements

WebRTCRTPTransceiverClass
WebRTCSessionDescription

See https://www.w3.org/TR/webrtc/rtcsessiondescription-class</ulink>

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-24section-4.1.1 for more information.

WebRTCDTLSSetup

GST_WEBRTC_DTLS_SETUP_NONE: none GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly

WebRTCDTLSTransportState

GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected

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</ulink>

WebRTCFECType
WebRTCICEComponent

GST_WEBRTC_ICE_COMPONENT_RTP, GST_WEBRTC_ICE_COMPONENT_RTCP,

WebRTCICEConnectionState

GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed See http://w3c.github.io/webrtc-pc/dom-rtciceconnectionstate</ulink>

WebRTCICEGatheringState

GST_WEBRTC_ICE_GATHERING_STATE_NEW: new GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete See http://w3c.github.io/webrtc-pc/dom-rtcicegatheringstate</ulink>

WebRTCICERole

GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling

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-24section-4.1.1 for more information.

WebRTCPeerConnectionState

GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed See http://w3c.github.io/webrtc-pc/dom-rtcpeerconnectionstate</ulink>

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</ulink>

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</ulink>

WebRTCSDPType

GST_WEBRTC_SDP_TYPE_OFFER: offer GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer GST_WEBRTC_SDP_TYPE_ANSWER: answer GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback See http://w3c.github.io/webrtc-pc/rtcsdptype``</ulink>

WebRTCSignalingState

GST_WEBRTC_SIGNALING_STATE_STABLE: stable GST_WEBRTC_SIGNALING_STATE_CLOSED: closed GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer See http://w3c.github.io/webrtc-pc/dom-rtcsignalingstate</ulink>

WebRTCStatsType

GST_WEBRTC_STATS_CODEC: codec GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp GST_WEBRTC_STATS_CSRC: csrc GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion GST_WEBRTC_STATS_DATA_CHANNEL: data-channel GST_WEBRTC_STATS_STREAM: stream GST_WEBRTC_STATS_TRANSPORT: transport GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate GST_WEBRTC_STATS_CERTIFICATE: certificate