import { StatsReport, TrackStatsReport, CalculationStats, AudioExtraStats, VideoExtraStats, MediaStatisticStatsWithCalculationReport, NetworkGrade } from "../types";
declare class ExpFilter {
    static kValueUndefined: number;
    static AlphaForPacketLossFractionSmoother: number;
    /**
     * the smooth should be done when rtcp packet arrive/send, but we can't get the rtcp
     * packet arrive/send time. according to webrtc and media server, the default rtcp
     * interval is:
     *
     * kDefaultVideoReportInterval = 1000;
     * kDefaultAudioReportInterval = 5000;
     * so we define the value accordingly
     */
    static SMOOTH_VIDEO_INTERVAL: number;
    static SMOOTH_AUDIO_INTERVAL: number;
    mAlpha: number;
    mFiltered: number;
    mMax: number;
    mLastFilteredTimestamp: number;
    smooth_interval: number;
    applied: boolean;
    constructor(filterType: "audio" | "video", alpha?: number);
    Reset(alpha: number): void;
    Apply(sample: number): void;
    Filtered(): number;
    LastFilteredTime(): number;
    UpdateBase(alpha: number): void;
}
/**
 * TrackStatsReport and MediaStatisticStatsWithCalculationReport use different timer
 * So two filters for each here.
 */
export declare const getReportExpFilter: (id: string, filterType: "audio" | "video") => ExpFilter;
export declare const getStatisticExpFilter: (id: string, filterType: "audio" | "video") => ExpFilter;
export declare const defaultStatsReport: StatsReport;
export declare const defaultTrackStatsReport: () => TrackStatsReport;
export declare function getNetworkGrade(maxLostRate: number, maxRtt: number): NetworkGrade;
export declare function getStats(pc: RTCPeerConnection, track: MediaStreamTrack, pctype: "send" | "recv"): Promise<TrackStatsReport[]>;
export declare const defaultAudioExtraStats: AudioExtraStats;
export declare const defaultVideoExtraStats: VideoExtraStats;
export declare const defaultCalculationStats: CalculationStats;
export declare enum MediaType {
    Video = "video",
    Audio = "audio"
}
export declare enum StatsReportType {
    MediaSource = "media-source",
    Track = "track",
    OutBoundRtp = "outbound-rtp",
    InBoundRtp = "inbound-rtp",
    RemoteInBound = "remote-inbound-rtp"
}
export declare enum BoundType {
    In = "in",
    Out = "out"
}
export declare function getDefaultMediaStatisticWithCalculationReport(id: string, trackId: string, mediaType: MediaType): MediaStatisticStatsWithCalculationReport;
export declare function getMediaStatisticStats(pc: RTCPeerConnection): Promise<MediaStatisticStatsWithCalculationReport[]>;
/**
 * Note that:
 * - packetsLost means total number of RTP packets lost for this SSRC and can be negative if more packets are received than sent.
 * - packetsTrans stands for packetsSent in outbound-rtp or packetsReceived in inbound-rtp.
 *   - packetsSent means total number of RTP packets sent for this SSRC.
 *   - packetsReceived means total number of RTP packets received for this SSRC.
 *
 * So we calculate packets lost rate by following math:
 *    packetsLostRate = difference of packetsLost between two gatherings / difference of packetsTrans between two gatherings
 */
export declare function calculatePacketsLostRate(lastPacketsLost: number, currentPacketsLost: number, lastPacketsTrans: number, currentPacketsTrans: number): number;
export interface CalculationReport {
    framerate: number;
    kbps: number;
    packet_lost_rate: number;
}
export declare function calculationMediaStatisticReport(calculation: CalculationStats, lastCalculation: CalculationStats, expFilter: ExpFilter): CalculationReport;
export {};