import { LocalCameraStream, LocalMicrophoneStream, LocalDisplayStream, LocalSystemAudioStream, RemoteStream } from '@webex/media-helpers';
import { ClientEvent } from '@webex/internal-plugin-metrics';
import { throttle } from 'lodash';
export type MediaDirection = {
    sendAudio: boolean;
    sendVideo: boolean;
    sendShare: boolean;
    receiveAudio: boolean;
    receiveVideo: boolean;
    receiveShare: boolean;
};
export type IPVersion = ClientEvent['payload']['ipVersion'];
/**
 * @class MediaProperties
 */
export default class MediaProperties {
    audioStream?: LocalMicrophoneStream;
    mediaDirection: MediaDirection;
    mediaSettings: any;
    webrtcMediaConnection: any;
    remoteAudioStream: RemoteStream;
    remoteQualityLevel: any;
    remoteShareStream: RemoteStream;
    remoteVideoStream: RemoteStream;
    shareVideoStream?: LocalDisplayStream;
    shareAudioStream?: LocalSystemAudioStream;
    videoDeviceId: any;
    videoStream?: LocalCameraStream;
    namespace: string;
    mediaIssueCounters: {
        [key: string]: number;
    };
    throttledSendMediaIssueMetric: ReturnType<typeof throttle>;
    /**
     * @param {Object} [options] -- to auto construct
     * @returns {MediaProperties}
     */
    constructor();
    /**
     * Retrieves the preferred video input device
     * @returns {Object|null}
     */
    getVideoDeviceId(): any;
    setMediaDirection(mediaDirection: any): void;
    setMediaSettings(type: any, values: any): void;
    setMediaPeerConnection(mediaPeerConnection: any): void;
    setLocalVideoStream(videoStream?: LocalCameraStream): void;
    setLocalAudioStream(audioStream?: LocalMicrophoneStream): void;
    setLocalShareVideoStream(shareVideoStream?: LocalDisplayStream): void;
    setLocalShareAudioStream(shareAudioStream?: LocalSystemAudioStream): void;
    setRemoteQualityLevel(remoteQualityLevel: any): void;
    setRemoteShareStream(remoteShareStream: RemoteStream): void;
    /**
     * Sets the remote audio stream
     * @param {RemoteStream} remoteAudioStream RemoteStream to save
     * @returns {void}
     */
    setRemoteAudioStream(remoteAudioStream: RemoteStream): void;
    /**
     * Sets the remote video stream
     * @param {RemoteStream} remoteVideoStream RemoteStream to save
     * @returns {void}
     */
    setRemoteVideoStream(remoteVideoStream: RemoteStream): void;
    /**
     * Stores the preferred video input device
     * @param {string} deviceId Preferred video input device
     * @returns {void}
     */
    setVideoDeviceId(deviceId: string): void;
    /**
     * Clears the webrtcMediaConnection. This method should be called after
     * peer connection is closed and no longer needed.
     * @returns {void}
     */
    unsetPeerConnection(): void;
    /**
     * Removes both remote audio and video from class instance
     * @returns {void}
     */
    unsetRemoteMedia(): void;
    unsetRemoteShareStream(): void;
    /**
     * Unsets all remote streams
     * @returns {void}
     */
    unsetRemoteStreams(): void;
    /**
     * Returns if we have at least one local share stream or not.
     * @returns {Boolean}
     */
    hasLocalShareStream(): boolean;
    /**
     * Waits for the webrtc media connection to be connected.
     *
     * @param {string} correlationId
     * @returns {Promise<void>}
     */
    waitForMediaConnectionConnected(correlationId: string): Promise<void>;
    /**
     * Returns ICE transport information:
     * - selectedCandidatePairChanges - number of times the selected candidate pair was changed, it should be at least 1 for successful connections
     *  it will be -1 if browser doesn't supply this information
     * - numTransports - number of transports (should be 1 if we're using bundle)
     *
     * @param {Array<any>} allStatsReports array of RTC stats reports
     * @returns {Object}
     */
    private getTransportInfo;
    /**
     * Checks if the given IP address is IPv6
     * @param {string} ip address to check
     * @returns {boolean} true if the address is IPv6, false otherwise
     */
    private isIPv6;
    /** Finds out if we connected using IPv4 or IPv6
     * @param {RTCPeerConnection} webrtcMediaConnection
     * @param {Array<any>} allStatsReports array of RTC stats reports
     * @returns {string} IPVersion
     */
    private getConnectionIpVersion;
    /**
     * Returns the type of a connection that has been established
     * It should be 'UDP' | 'TCP' | 'TURN-TLS' | 'TURN-TCP' | 'TURN-UDP' | 'unknown'
     *
     * If connection was not established, it returns 'unknown'
     *
     * @param {Array<any>} allStatsReports array of RTC stats reports
     * @returns {string}
     */
    private getConnectionType;
    /**
     * Returns information about current webrtc media connection
     *
     * @returns {Promise<Object>}
     */
    getCurrentConnectionInfo(): Promise<{
        connectionType: string;
        ipVersion?: IPVersion;
        selectedCandidatePairChanges: number;
        numTransports: number;
    }>;
    /**
     * Sends a metric about a media issue. Metrics are throttled so that we don't
     * send too many of them, but include a count so that we know how many issues
     * were detected.
     *
     * @param {string} issueType
     * @param {string} issueSubType
     * @param {string} correlationId
     * @returns {void}
     */
    sendMediaIssueMetric(issueType: string, issueSubType: string, correlationId: any): void;
}