import { Subscription } from "rxjs";
import { CallRB } from "../../models/call.model";
import { Service } from "../../services/service";
export declare const WEBRTC_CONFIG_UTILS_SVC = "WebrtcConfigAndUtils";
export interface IWebrtcConfigAndUtils {
    RTC: any;
}
/**
 * This module contains all the webrtc configuration tools and utils that can be used to manages webrtc audio/video calls
 * This module can be used to set the video quality and frame rate for all P2P calls
 * The lifecycle of this module is managed by the WebRTC P2P Service
 */
export declare class WebrtcConfigAndUtils extends Service implements IWebrtcConfigAndUtils {
    /**
     * The current webrtc configuration used (TURN servers and DC)
     */
    webrtcConfig: any;
    /**
     * The current webrtc company configuration (if set for the company)
     */
    companyConfig: any;
    /**
     * The current webrtc video and sharing quality configuration when in call with mobile phone
     */
    mobileVideoQualityConfiguration: {
        video: {
            fixedFrameRate: number;
            fixedVideoWidth: number;
            fixedVideoHeight: number;
        };
        sharing: {
            fixedFrameRate: number;
            fixedVideoWidth: number;
            fixedVideoHeight: number;
        };
    };
    /**
     * The current webrtc video and sharing quality configuration
     */
    videoQualityConfiguration: {
        video: {
            fixedFrameRate: number;
            fixedVideoWidth: number;
            fixedVideoHeight: number;
        };
        sharing: {
            fixedFrameRate: number;
            fixedVideoWidth: number;
            fixedVideoHeight: number;
        };
    };
    /**
     * The RTC object that contains the attach and detach functions to add stream to DOM element based on the platform
     * The attachSessionMediaToElement is used for session that have only 1 media (for conferences case); We need to send the HTML element ID + the session that we want to "attach" to that element
     */
    RTC: any;
    /**
     * Subscribe to updates on the current calls sessions (all events are of RBEvent type `{ name: string, data:any }`);
     * All public events that are fired by this service are described in the documentation.
     * NB There might be some events that are not documented here, they should be ignored
     * @param handler - The call-back function that will be subscribed to the RxJS subject
     */
    subscribe(handler: any): Subscription;
    /**
     * Send an event to the RX Subject (sends always a RBEvent type)
     * @param string - name The name of the event
     * @param data - The data of the event
     */
    private sendEvent;
    /**
     * The Calls Stats object that contains all the statistics for the current call (that will be uploaded to the server at the end of the call)
     */
    callsStats: {};
    askToAddSharing(__conversationId: string, __shouldAskForAudio?: boolean): boolean;
    private logger;
    private settingsService;
    private authService;
    private xmppService;
    private mainService;
    private contactService;
    private eventService;
    private deviceManagementService;
    private profileService;
    private rxSubject;
    private webrtcSessionStatsHandler;
    private jingleSubscription;
    private localPeerConnection;
    private ttl;
    private ttlTimeout;
    private started;
    private statsWaitingQueue;
    static getInstance(): WebrtcConfigAndUtils;
    static build(): WebrtcConfigAndUtils;
    private constructor();
    start(): Promise<void>;
    stop(): void;
    private onConnectionStateChangeEvent;
    private onJingleMessage;
    private makeLocalEchoTest;
    /**
     * This method builds the local peer connection
     */
    private buildLocalPeerConnection;
    private destroyLocalPeerConnection;
    getBrowserMethodHandler(): any;
    private jingleTurnCredentialExpire;
    setBrowserMethodHandler(): any;
    getDesktopSharingSource(): any;
    private getConstraintsForMedia;
    getBrowserMedia(userMedia: any[], fixedVideoWidth?: number, fixedVideoHeight?: number, fixedFrameRate?: number, referer?: any, getAudioForSharing?: boolean, isMediaPillarCall?: boolean): Promise<MediaStream | undefined>;
    /**
     * Disables and removes all local streams for this session
     * @param session - The session (or any object that contains localStreams that will be removed)
     */
    disableAllStreamsForSession(session: any): void;
    /**
     * Disables MediaStream
     * @param streams - An array of MediaStream to be stopped and removed
     */
    disableMediaStreams(streams: MediaStream[]): void;
    /**
     * Disables MediaStream
     * @param stream - The MediaStream to be stopped and removed
     */
    disableMediaStream(stream: MediaStream): void;
    getMobileMediaConstraints(type: string): any;
    createOrUpdateStatisticsForCall(call: CallRB): Promise<void>;
    updateStatisticsForCall(id: string, callId: string, callState: string, metrics: any): Promise<void>;
    createStatisticsForSession(callId: string, peerId: string, callState: string): Promise<string>;
    setWebrtcConfiguration(): Promise<void>;
    /**
     * Get the Media stream from the browser w/o any constraints (except for the selected current device);
     * @param userMedia - An array of medias to get (accepted values inside the array are : "audio" and "video");
     */
    getBrowserMediaWithNoConstraints(userMedia: string[]): Promise<MediaStream>;
    private getDefaultConstraintsForMedia;
    private setIceConfig;
    private manageIceServers;
    private getIceConfig;
    private setCompanyWebrtcConfig;
    private setVideoQualityConfigurationForCompany;
    private getCompanyWebrtcConfig;
    getStatsForSession(session: any, interval?: number, webinarId?: string): any;
    /**
     * For QA Tests Auto
     * Set/reset an interval for generating WebRTC statistics that will be used instead of the default 5s interval.
     * Set by default to 1000ms.
     * Usage: rainbow.services.WebrtcConfigAndUtils.setWebrtcStatsInterval(intervalMs);
     */
    setWebrtcStatsInterval(interval?: number): void;
    resetWebrtcStatsInterval(): void;
}
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