mediasoup
Version:
Cutting Edge WebRTC Video Conferencing
34 lines • 1.13 kB
TypeScript
export declare enum Event {
TRANSPORT_SEND_RTCP = 0,
PRODUCER_SEND = 1,
DATAPRODUCER_SEND = 2,
WORKER_RUNNING = 3,
TRANSPORT_SCTP_STATE_CHANGE = 4,
TRANSPORT_TRACE = 5,
WEBRTCTRANSPORT_ICE_SELECTED_TUPLE_CHANGE = 6,
WEBRTCTRANSPORT_ICE_STATE_CHANGE = 7,
WEBRTCTRANSPORT_DTLS_STATE_CHANGE = 8,
PLAINTRANSPORT_TUPLE = 9,
PLAINTRANSPORT_RTCP_TUPLE = 10,
DIRECTTRANSPORT_RTCP = 11,
PRODUCER_SCORE = 12,
PRODUCER_TRACE = 13,
PRODUCER_VIDEO_ORIENTATION_CHANGE = 14,
CONSUMER_PRODUCER_PAUSE = 15,
CONSUMER_PRODUCER_RESUME = 16,
CONSUMER_PRODUCER_CLOSE = 17,
CONSUMER_LAYERS_CHANGE = 18,
CONSUMER_RTP = 19,
CONSUMER_SCORE = 20,
CONSUMER_TRACE = 21,
DATACONSUMER_BUFFERED_AMOUNT_LOW = 22,
DATACONSUMER_SCTP_SENDBUFFER_FULL = 23,
DATACONSUMER_DATAPRODUCER_PAUSE = 24,
DATACONSUMER_DATAPRODUCER_RESUME = 25,
DATACONSUMER_DATAPRODUCER_CLOSE = 26,
DATACONSUMER_MESSAGE = 27,
ACTIVESPEAKEROBSERVER_DOMINANT_SPEAKER = 28,
AUDIOLEVELOBSERVER_SILENCE = 29,
AUDIOLEVELOBSERVER_VOLUMES = 30
}
//# sourceMappingURL=event.d.ts.map