GstWebRTC Enumerations
Enumerations
GstWebRTCBundlePolicy
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
Members
GST_WEBRTC_BUNDLE_POLICY_NONE
(0)
–
GST_WEBRTC_BUNDLE_POLICY_BALANCED
(1)
–
GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT
(2)
–
GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE
(3)
–
GstWebRTC.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
Members
GstWebRTC.WebRTCBundlePolicy.NONE
(0)
–
GstWebRTC.WebRTCBundlePolicy.BALANCED
(1)
–
GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT
(2)
–
GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE
(3)
–
GstWebRTC.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
Members
GstWebRTC.WebRTCBundlePolicy.NONE
(0)
–
GstWebRTC.WebRTCBundlePolicy.BALANCED
(1)
–
GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT
(2)
–
GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE
(3)
–
GstWebRTCDTLSSetup
Members
GST_WEBRTC_DTLS_SETUP_NONE
(0)
–
none
GST_WEBRTC_DTLS_SETUP_ACTPASS
(1)
–
actpass
GST_WEBRTC_DTLS_SETUP_ACTIVE
(2)
–
sendonly
GST_WEBRTC_DTLS_SETUP_PASSIVE
(3)
–
recvonly
GstWebRTC.WebRTCDTLSSetup
Members
GstWebRTC.WebRTCDTLSSetup.NONE
(0)
–
none
GstWebRTC.WebRTCDTLSSetup.ACTPASS
(1)
–
actpass
GstWebRTC.WebRTCDTLSSetup.ACTIVE
(2)
–
sendonly
GstWebRTC.WebRTCDTLSSetup.PASSIVE
(3)
–
recvonly
GstWebRTC.WebRTCDTLSSetup
Members
GstWebRTC.WebRTCDTLSSetup.NONE
(0)
–
none
GstWebRTC.WebRTCDTLSSetup.ACTPASS
(1)
–
actpass
GstWebRTC.WebRTCDTLSSetup.ACTIVE
(2)
–
sendonly
GstWebRTC.WebRTCDTLSSetup.PASSIVE
(3)
–
recvonly
GstWebRTCDTLSTransportState
Members
GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW
(0)
–
new
GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED
(1)
–
closed
GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED
(2)
–
failed
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING
(3)
–
connecting
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED
(4)
–
connected
GstWebRTC.WebRTCDTLSTransportState
Members
GstWebRTC.WebRTCDTLSTransportState.NEW
(0)
–
new
GstWebRTC.WebRTCDTLSTransportState.CLOSED
(1)
–
closed
GstWebRTC.WebRTCDTLSTransportState.FAILED
(2)
–
failed
GstWebRTC.WebRTCDTLSTransportState.CONNECTING
(3)
–
connecting
GstWebRTC.WebRTCDTLSTransportState.CONNECTED
(4)
–
connected
GstWebRTC.WebRTCDTLSTransportState
Members
GstWebRTC.WebRTCDTLSTransportState.NEW
(0)
–
new
GstWebRTC.WebRTCDTLSTransportState.CLOSED
(1)
–
closed
GstWebRTC.WebRTCDTLSTransportState.FAILED
(2)
–
failed
GstWebRTC.WebRTCDTLSTransportState.CONNECTING
(3)
–
connecting
GstWebRTC.WebRTCDTLSTransportState.CONNECTED
(4)
–
connected
GstWebRTCDataChannelState
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
Members
GST_WEBRTC_DATA_CHANNEL_STATE_NEW
(0)
–
GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING
(1)
–
GST_WEBRTC_DATA_CHANNEL_STATE_OPEN
(2)
–
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING
(3)
–
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED
(4)
–
GstWebRTC.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
Members
GstWebRTC.WebRTCDataChannelState.NEW
(0)
–
GstWebRTC.WebRTCDataChannelState.CONNECTING
(1)
–
GstWebRTC.WebRTCDataChannelState.OPEN
(2)
–
GstWebRTC.WebRTCDataChannelState.CLOSING
(3)
–
GstWebRTC.WebRTCDataChannelState.CLOSED
(4)
–
GstWebRTC.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
Members
GstWebRTC.WebRTCDataChannelState.NEW
(0)
–
GstWebRTC.WebRTCDataChannelState.CONNECTING
(1)
–
GstWebRTC.WebRTCDataChannelState.OPEN
(2)
–
GstWebRTC.WebRTCDataChannelState.CLOSING
(3)
–
GstWebRTC.WebRTCDataChannelState.CLOSED
(4)
–
GstWebRTCFECType
Members
GST_WEBRTC_FEC_TYPE_NONE
(0)
–
none
GST_WEBRTC_FEC_TYPE_ULP_RED
(1)
–
ulpfec + red
GstWebRTC.WebRTCFECType
Members
GstWebRTC.WebRTCFECType.NONE
(0)
–
none
GstWebRTC.WebRTCFECType.ULP_RED
(1)
–
ulpfec + red
GstWebRTC.WebRTCFECType
Members
GstWebRTC.WebRTCFECType.NONE
(0)
–
none
GstWebRTC.WebRTCFECType.ULP_RED
(1)
–
ulpfec + red
GstWebRTCICEComponent
Members
GST_WEBRTC_ICE_COMPONENT_RTP
(0)
–
RTP component
GST_WEBRTC_ICE_COMPONENT_RTCP
(1)
–
RTCP component
GstWebRTC.WebRTCICEComponent
Members
GstWebRTC.WebRTCICEComponent.RTP
(0)
–
RTP component
GstWebRTC.WebRTCICEComponent.RTCP
(1)
–
RTCP component
GstWebRTC.WebRTCICEComponent
Members
GstWebRTC.WebRTCICEComponent.RTP
(0)
–
RTP component
GstWebRTC.WebRTCICEComponent.RTCP
(1)
–
RTCP component
GstWebRTCICEConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
Members
GST_WEBRTC_ICE_CONNECTION_STATE_NEW
(0)
–
new
GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING
(1)
–
checking
GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED
(2)
–
connected
GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED
(3)
–
completed
GST_WEBRTC_ICE_CONNECTION_STATE_FAILED
(4)
–
failed
GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED
(5)
–
disconnected
GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED
(6)
–
closed
GstWebRTC.WebRTCICEConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
Members
GstWebRTC.WebRTCICEConnectionState.NEW
(0)
–
new
GstWebRTC.WebRTCICEConnectionState.CHECKING
(1)
–
checking
GstWebRTC.WebRTCICEConnectionState.CONNECTED
(2)
–
connected
GstWebRTC.WebRTCICEConnectionState.COMPLETED
(3)
–
completed
GstWebRTC.WebRTCICEConnectionState.FAILED
(4)
–
failed
GstWebRTC.WebRTCICEConnectionState.DISCONNECTED
(5)
–
disconnected
GstWebRTC.WebRTCICEConnectionState.CLOSED
(6)
–
closed
GstWebRTC.WebRTCICEConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
Members
GstWebRTC.WebRTCICEConnectionState.NEW
(0)
–
new
GstWebRTC.WebRTCICEConnectionState.CHECKING
(1)
–
checking
GstWebRTC.WebRTCICEConnectionState.CONNECTED
(2)
–
connected
GstWebRTC.WebRTCICEConnectionState.COMPLETED
(3)
–
completed
GstWebRTC.WebRTCICEConnectionState.FAILED
(4)
–
failed
GstWebRTC.WebRTCICEConnectionState.DISCONNECTED
(5)
–
disconnected
GstWebRTC.WebRTCICEConnectionState.CLOSED
(6)
–
closed
GstWebRTCICEGatheringState
See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
Members
GST_WEBRTC_ICE_GATHERING_STATE_NEW
(0)
–
new
GST_WEBRTC_ICE_GATHERING_STATE_GATHERING
(1)
–
gathering
GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE
(2)
–
complete
GstWebRTC.WebRTCICEGatheringState
See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
Members
GstWebRTC.WebRTCICEGatheringState.NEW
(0)
–
new
GstWebRTC.WebRTCICEGatheringState.GATHERING
(1)
–
gathering
GstWebRTC.WebRTCICEGatheringState.COMPLETE
(2)
–
complete
GstWebRTC.WebRTCICEGatheringState
See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
Members
GstWebRTC.WebRTCICEGatheringState.NEW
(0)
–
new
GstWebRTC.WebRTCICEGatheringState.GATHERING
(1)
–
gathering
GstWebRTC.WebRTCICEGatheringState.COMPLETE
(2)
–
complete
GstWebRTCICERole
Members
GST_WEBRTC_ICE_ROLE_CONTROLLED
(0)
–
controlled
GST_WEBRTC_ICE_ROLE_CONTROLLING
(1)
–
controlling
GstWebRTC.WebRTCICERole
Members
GstWebRTC.WebRTCICERole.CONTROLLED
(0)
–
controlled
GstWebRTC.WebRTCICERole.CONTROLLING
(1)
–
controlling
GstWebRTC.WebRTCICERole
Members
GstWebRTC.WebRTCICERole.CONTROLLED
(0)
–
controlled
GstWebRTC.WebRTCICERole.CONTROLLING
(1)
–
controlling
GstWebRTCICETransportPolicy
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
Members
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL
(0)
–
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY
(1)
–
GstWebRTC.WebRTCICETransportPolicy
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
Members
GstWebRTC.WebRTCICETransportPolicy.ALL
(0)
–
GstWebRTC.WebRTCICETransportPolicy.RELAY
(1)
–
GstWebRTC.WebRTCICETransportPolicy
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
Members
GstWebRTC.WebRTCICETransportPolicy.ALL
(0)
–
GstWebRTC.WebRTCICETransportPolicy.RELAY
(1)
–
GstWebRTCPeerConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate
Members
GST_WEBRTC_PEER_CONNECTION_STATE_NEW
(0)
–
new
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING
(1)
–
connecting
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED
(2)
–
connected
GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED
(3)
–
disconnected
GST_WEBRTC_PEER_CONNECTION_STATE_FAILED
(4)
–
failed
GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED
(5)
–
closed
GstWebRTC.WebRTCPeerConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate
Members
GstWebRTC.WebRTCPeerConnectionState.NEW
(0)
–
new
GstWebRTC.WebRTCPeerConnectionState.CONNECTING
(1)
–
connecting
GstWebRTC.WebRTCPeerConnectionState.CONNECTED
(2)
–
connected
GstWebRTC.WebRTCPeerConnectionState.DISCONNECTED
(3)
–
disconnected
GstWebRTC.WebRTCPeerConnectionState.FAILED
(4)
–
failed
GstWebRTC.WebRTCPeerConnectionState.CLOSED
(5)
–
closed
GstWebRTC.WebRTCPeerConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate
Members
GstWebRTC.WebRTCPeerConnectionState.NEW
(0)
–
new
GstWebRTC.WebRTCPeerConnectionState.CONNECTING
(1)
–
connecting
GstWebRTC.WebRTCPeerConnectionState.CONNECTED
(2)
–
connected
GstWebRTC.WebRTCPeerConnectionState.DISCONNECTED
(3)
–
disconnected
GstWebRTC.WebRTCPeerConnectionState.FAILED
(4)
–
failed
GstWebRTC.WebRTCPeerConnectionState.CLOSED
(5)
–
closed
GstWebRTCPriorityType
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
Members
GST_WEBRTC_PRIORITY_TYPE_VERY_LOW
(1)
–
GST_WEBRTC_PRIORITY_TYPE_LOW
(2)
–
GST_WEBRTC_PRIORITY_TYPE_MEDIUM
(3)
–
GST_WEBRTC_PRIORITY_TYPE_HIGH
(4)
–
GstWebRTC.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
Members
GstWebRTC.WebRTCPriorityType.VERY_LOW
(1)
–
GstWebRTC.WebRTCPriorityType.LOW
(2)
–
GstWebRTC.WebRTCPriorityType.MEDIUM
(3)
–
GstWebRTC.WebRTCPriorityType.HIGH
(4)
–
GstWebRTC.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
Members
GstWebRTC.WebRTCPriorityType.VERY_LOW
(1)
–
GstWebRTC.WebRTCPriorityType.LOW
(2)
–
GstWebRTC.WebRTCPriorityType.MEDIUM
(3)
–
GstWebRTC.WebRTCPriorityType.HIGH
(4)
–
GstWebRTCRTPTransceiverDirection
Members
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE
(0)
–
none
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE
(1)
–
inactive
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY
(2)
–
sendonly
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY
(3)
–
recvonly
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV
(4)
–
sendrecv
GstWebRTC.WebRTCRTPTransceiverDirection
Members
GstWebRTC.WebRTCRTPTransceiverDirection.NONE
(0)
–
none
GstWebRTC.WebRTCRTPTransceiverDirection.INACTIVE
(1)
–
inactive
GstWebRTC.WebRTCRTPTransceiverDirection.SENDONLY
(2)
–
sendonly
GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY
(3)
–
recvonly
GstWebRTC.WebRTCRTPTransceiverDirection.SENDRECV
(4)
–
sendrecv
GstWebRTC.WebRTCRTPTransceiverDirection
Members
GstWebRTC.WebRTCRTPTransceiverDirection.NONE
(0)
–
none
GstWebRTC.WebRTCRTPTransceiverDirection.INACTIVE
(1)
–
inactive
GstWebRTC.WebRTCRTPTransceiverDirection.SENDONLY
(2)
–
sendonly
GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY
(3)
–
recvonly
GstWebRTC.WebRTCRTPTransceiverDirection.SENDRECV
(4)
–
sendrecv
GstWebRTCSCTPTransportState
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
Members
GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW
(0)
–
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING
(1)
–
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED
(2)
–
GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED
(3)
–
GstWebRTC.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
Members
GstWebRTC.WebRTCSCTPTransportState.NEW
(0)
–
GstWebRTC.WebRTCSCTPTransportState.CONNECTING
(1)
–
GstWebRTC.WebRTCSCTPTransportState.CONNECTED
(2)
–
GstWebRTC.WebRTCSCTPTransportState.CLOSED
(3)
–
GstWebRTC.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
Members
GstWebRTC.WebRTCSCTPTransportState.NEW
(0)
–
GstWebRTC.WebRTCSCTPTransportState.CONNECTING
(1)
–
GstWebRTC.WebRTCSCTPTransportState.CONNECTED
(2)
–
GstWebRTC.WebRTCSCTPTransportState.CLOSED
(3)
–
GstWebRTCSDPType
See http://w3c.github.io/webrtc-pc/#rtcsdptype
Members
GST_WEBRTC_SDP_TYPE_OFFER
(1)
–
offer
GST_WEBRTC_SDP_TYPE_PRANSWER
(2)
–
pranswer
GST_WEBRTC_SDP_TYPE_ANSWER
(3)
–
answer
GST_WEBRTC_SDP_TYPE_ROLLBACK
(4)
–
rollback
GstWebRTC.WebRTCSDPType
See http://w3c.github.io/webrtc-pc/#rtcsdptype
Members
GstWebRTC.WebRTCSDPType.OFFER
(1)
–
offer
GstWebRTC.WebRTCSDPType.PRANSWER
(2)
–
pranswer
GstWebRTC.WebRTCSDPType.ANSWER
(3)
–
answer
GstWebRTC.WebRTCSDPType.ROLLBACK
(4)
–
rollback
GstWebRTC.WebRTCSDPType
See http://w3c.github.io/webrtc-pc/#rtcsdptype
Members
GstWebRTC.WebRTCSDPType.OFFER
(1)
–
offer
GstWebRTC.WebRTCSDPType.PRANSWER
(2)
–
pranswer
GstWebRTC.WebRTCSDPType.ANSWER
(3)
–
answer
GstWebRTC.WebRTCSDPType.ROLLBACK
(4)
–
rollback
GstWebRTCSignalingState
See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
Members
GST_WEBRTC_SIGNALING_STATE_STABLE
(0)
–
stable
GST_WEBRTC_SIGNALING_STATE_CLOSED
(1)
–
closed
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER
(2)
–
have-local-offer
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER
(3)
–
have-remote-offer
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER
(4)
–
have-local-pranswer
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER
(5)
–
have-remote-pranswer
GstWebRTC.WebRTCSignalingState
See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
Members
GstWebRTC.WebRTCSignalingState.STABLE
(0)
–
stable
GstWebRTC.WebRTCSignalingState.CLOSED
(1)
–
closed
GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_OFFER
(2)
–
have-local-offer
GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_OFFER
(3)
–
have-remote-offer
GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_PRANSWER
(4)
–
have-local-pranswer
GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_PRANSWER
(5)
–
have-remote-pranswer
GstWebRTC.WebRTCSignalingState
See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
Members
GstWebRTC.WebRTCSignalingState.STABLE
(0)
–
stable
GstWebRTC.WebRTCSignalingState.CLOSED
(1)
–
closed
GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_OFFER
(2)
–
have-local-offer
GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_OFFER
(3)
–
have-remote-offer
GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_PRANSWER
(4)
–
have-local-pranswer
GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_PRANSWER
(5)
–
have-remote-pranswer
GstWebRTCStatsType
Members
GST_WEBRTC_STATS_CODEC
(1)
–
codec
GST_WEBRTC_STATS_INBOUND_RTP
(2)
–
inbound-rtp
GST_WEBRTC_STATS_OUTBOUND_RTP
(3)
–
outbound-rtp
GST_WEBRTC_STATS_REMOTE_INBOUND_RTP
(4)
–
remote-inbound-rtp
GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP
(5)
–
remote-outbound-rtp
GST_WEBRTC_STATS_CSRC
(6)
–
csrc
GST_WEBRTC_STATS_PEER_CONNECTION
(7)
–
peer-connectiion
GST_WEBRTC_STATS_DATA_CHANNEL
(8)
–
data-channel
GST_WEBRTC_STATS_STREAM
(9)
–
stream
GST_WEBRTC_STATS_TRANSPORT
(10)
–
transport
GST_WEBRTC_STATS_CANDIDATE_PAIR
(11)
–
candidate-pair
GST_WEBRTC_STATS_LOCAL_CANDIDATE
(12)
–
local-candidate
GST_WEBRTC_STATS_REMOTE_CANDIDATE
(13)
–
remote-candidate
GST_WEBRTC_STATS_CERTIFICATE
(14)
–
certificate
GstWebRTC.WebRTCStatsType
Members
GstWebRTC.WebRTCStatsType.CODEC
(1)
–
codec
GstWebRTC.WebRTCStatsType.INBOUND_RTP
(2)
–
inbound-rtp
GstWebRTC.WebRTCStatsType.OUTBOUND_RTP
(3)
–
outbound-rtp
GstWebRTC.WebRTCStatsType.REMOTE_INBOUND_RTP
(4)
–
remote-inbound-rtp
GstWebRTC.WebRTCStatsType.REMOTE_OUTBOUND_RTP
(5)
–
remote-outbound-rtp
GstWebRTC.WebRTCStatsType.CSRC
(6)
–
csrc
GstWebRTC.WebRTCStatsType.PEER_CONNECTION
(7)
–
peer-connectiion
GstWebRTC.WebRTCStatsType.DATA_CHANNEL
(8)
–
data-channel
GstWebRTC.WebRTCStatsType.STREAM
(9)
–
stream
GstWebRTC.WebRTCStatsType.TRANSPORT
(10)
–
transport
GstWebRTC.WebRTCStatsType.CANDIDATE_PAIR
(11)
–
candidate-pair
GstWebRTC.WebRTCStatsType.LOCAL_CANDIDATE
(12)
–
local-candidate
GstWebRTC.WebRTCStatsType.REMOTE_CANDIDATE
(13)
–
remote-candidate
GstWebRTC.WebRTCStatsType.CERTIFICATE
(14)
–
certificate
GstWebRTC.WebRTCStatsType
Members
GstWebRTC.WebRTCStatsType.CODEC
(1)
–
codec
GstWebRTC.WebRTCStatsType.INBOUND_RTP
(2)
–
inbound-rtp
GstWebRTC.WebRTCStatsType.OUTBOUND_RTP
(3)
–
outbound-rtp
GstWebRTC.WebRTCStatsType.REMOTE_INBOUND_RTP
(4)
–
remote-inbound-rtp
GstWebRTC.WebRTCStatsType.REMOTE_OUTBOUND_RTP
(5)
–
remote-outbound-rtp
GstWebRTC.WebRTCStatsType.CSRC
(6)
–
csrc
GstWebRTC.WebRTCStatsType.PEER_CONNECTION
(7)
–
peer-connectiion
GstWebRTC.WebRTCStatsType.DATA_CHANNEL
(8)
–
data-channel
GstWebRTC.WebRTCStatsType.STREAM
(9)
–
stream
GstWebRTC.WebRTCStatsType.TRANSPORT
(10)
–
transport
GstWebRTC.WebRTCStatsType.CANDIDATE_PAIR
(11)
–
candidate-pair
GstWebRTC.WebRTCStatsType.LOCAL_CANDIDATE
(12)
–
local-candidate
GstWebRTC.WebRTCStatsType.REMOTE_CANDIDATE
(13)
–
remote-candidate
GstWebRTC.WebRTCStatsType.CERTIFICATE
(14)
–
certificate
Constants
GST_WEBRTC_API
#define GST_WEBRTC_API GST_API_EXPORT /* from config.h */
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