@fishjam-cloud/webrtc-client/dist/index.d.mts

Typescript client library for ExWebRTC/WebRTC endpoint in Membrane RTC Engine

import { BinaryWriter, BinaryReader } from '@bufbuild/protobuf/wire';
import TypedEmitter from 'typed-emitter';

declare enum Variant {
    VARIANT_UNSPECIFIED = 0,
    VARIANT_LOW = 1,
    VARIANT_MEDIUM = 2,
    VARIANT_HIGH = 3,
    UNRECOGNIZED = -1
}
/** Contains information about an ICE candidate which will be sent to the peer/server */
interface Candidate {
    candidate: string;
    sdpMLineIndex: number;
    sdpMid: string;
    usernameFragment: string;
}
declare const Candidate: MessageFns$2<Candidate>;
type Builtin$2 = Date | Function | Uint8Array | string | number | boolean | undefined;
type DeepPartial$2<T> = T extends Builtin$2 ? T : T extends globalThis.Array<infer U> ? globalThis.Array<DeepPartial$2<U>> : T extends ReadonlyArray<infer U> ? ReadonlyArray<DeepPartial$2<U>> : T extends {} ? {
    [K in keyof T]?: DeepPartial$2<T[K]>;
} : Partial<T>;
type KeysOfUnion$2<T> = T extends T ? keyof T : never;
type Exact$2<P, I extends P> = P extends Builtin$2 ? P : P & {
    [K in keyof P]: Exact$2<P[K], I[K]>;
} & {
    [K in Exclude<keyof I, KeysOfUnion$2<P>>]: never;
};
interface MessageFns$2<T> {
    encode(message: T, writer?: BinaryWriter): BinaryWriter;
    decode(input: BinaryReader | Uint8Array, length?: number): T;
    fromJSON(object: any): T;
    toJSON(message: T): unknown;
    create<I extends Exact$2<DeepPartial$2<T>, I>>(base?: I): T;
    fromPartial<I extends Exact$2<DeepPartial$2<T>, I>>(object: I): T;
}

/** Defines any type of message sent from Peer to Membrane RTC Engine */
interface MediaEvent$1 {
    connect?: MediaEvent_Connect | undefined;
    disconnect?: MediaEvent_Disconnect | undefined;
    updateEndpointMetadata?: MediaEvent_UpdateEndpointMetadata | undefined;
    updateTrackMetadata?: MediaEvent_UpdateTrackMetadata | undefined;
    renegotiateTracks?: MediaEvent_RenegotiateTracks | undefined;
    candidate?: Candidate | undefined;
    sdpOffer?: MediaEvent_SdpOffer | undefined;
    trackBitrates?: MediaEvent_TrackBitrates | undefined;
    enableTrackVariant?: MediaEvent_EnableTrackVariant | undefined;
    disableTrackVariant?: MediaEvent_DisableTrackVariant | undefined;
    setTargetTrackVariant?: MediaEvent_SetTargetTrackVariant | undefined;
    unmuteTrack?: MediaEvent_UnmuteTrack | undefined;
}
declare const MediaEvent$1: MessageFns$1<MediaEvent$1>;
interface MediaEvent_VariantBitrate {
    variant: Variant;
    bitrate: number;
}
declare const MediaEvent_VariantBitrate: MessageFns$1<MediaEvent_VariantBitrate>;
/** Sent when a peer wants to join WebRTC Endpoint. */
interface MediaEvent_Connect {
    metadataJson: string;
}
declare const MediaEvent_Connect: MessageFns$1<MediaEvent_Connect>;
/** Sent when a peer disconnects from WebRTC Endpoint. */
interface MediaEvent_Disconnect {
}
declare const MediaEvent_Disconnect: MessageFns$1<MediaEvent_Disconnect>;
/** Sent when a peer wants to update its metadata */
interface MediaEvent_UpdateEndpointMetadata {
    metadataJson: string;
}
declare const MediaEvent_UpdateEndpointMetadata: MessageFns$1<MediaEvent_UpdateEndpointMetadata>;
/** Sent when a peer wants to update its track's metadata */
interface MediaEvent_UpdateTrackMetadata {
    trackId: string;
    metadataJson: string;
}
declare const MediaEvent_UpdateTrackMetadata: MessageFns$1<MediaEvent_UpdateTrackMetadata>;
/** Sent when peer wants to renegatiate connection due to adding a track or removing a track */
interface MediaEvent_RenegotiateTracks {
}
declare const MediaEvent_RenegotiateTracks: MessageFns$1<MediaEvent_RenegotiateTracks>;
/**
 * Sent as a response to `offerData` media event during renegotiation
 * Maps contain only information about current peer's `sendonly` tracks.
 * The "mid" is an identifier used to associate an RTP packet with an MLine from the SDP offer/answer.
 */
interface MediaEvent_SdpOffer {
    /** The value of the `sessionDescription.sdp` */
    sdp: string;
    trackIdToMetadataJson: {
        [key: string]: string;
    };
    /** Maps track_id to its bitrate. The track_id in the TrackBitrates message is ignored (we use the map key), so it can be ommited. */
    trackIdToBitrates: {
        [key: string]: MediaEvent_TrackBitrates;
    };
    midToTrackId: {
        [key: string]: string;
    };
}
declare const MediaEvent_SdpOffer: MessageFns$1<MediaEvent_SdpOffer>;
/** Sent when Peer wants to update its track's bitrate */
interface MediaEvent_TrackBitrates {
    trackId: string;
    /** Bitrate of each variant. For non-simulcast tracks use VARIANT_UNSPECIFIED. */
    variantBitrates: MediaEvent_VariantBitrate[];
}
declare const MediaEvent_TrackBitrates: MessageFns$1<MediaEvent_TrackBitrates>;
/** Sent when client disables one of the track variants */
interface MediaEvent_DisableTrackVariant {
    trackId: string;
    variant: Variant;
}
declare const MediaEvent_DisableTrackVariant: MessageFns$1<MediaEvent_DisableTrackVariant>;
/** Sent when client enables one of the track variants */
interface MediaEvent_EnableTrackVariant {
    trackId: string;
    variant: Variant;
}
declare const MediaEvent_EnableTrackVariant: MessageFns$1<MediaEvent_EnableTrackVariant>;
interface MediaEvent_SetTargetTrackVariant {
    trackId: string;
    variant: Variant;
}
declare const MediaEvent_SetTargetTrackVariant: MessageFns$1<MediaEvent_SetTargetTrackVariant>;
interface MediaEvent_UnmuteTrack {
    trackId: string;
}
declare const MediaEvent_UnmuteTrack: MessageFns$1<MediaEvent_UnmuteTrack>;
type Builtin$1 = Date | Function | Uint8Array | string | number | boolean | undefined;
type DeepPartial$1<T> = T extends Builtin$1 ? T : T extends globalThis.Array<infer U> ? globalThis.Array<DeepPartial$1<U>> : T extends ReadonlyArray<infer U> ? ReadonlyArray<DeepPartial$1<U>> : T extends {} ? {
    [K in keyof T]?: DeepPartial$1<T[K]>;
} : Partial<T>;
type KeysOfUnion$1<T> = T extends T ? keyof T : never;
type Exact$1<P, I extends P> = P extends Builtin$1 ? P : P & {
    [K in keyof P]: Exact$1<P[K], I[K]>;
} & {
    [K in Exclude<keyof I, KeysOfUnion$1<P>>]: never;
};
interface MessageFns$1<T> {
    encode(message: T, writer?: BinaryWriter): BinaryWriter;
    decode(input: BinaryReader | Uint8Array, length?: number): T;
    fromJSON(object: any): T;
    toJSON(message: T): unknown;
    create<I extends Exact$1<DeepPartial$1<T>, I>>(base?: I): T;
    fromPartial<I extends Exact$1<DeepPartial$1<T>, I>>(object: I): T;
}

interface MediaEvent_Track_SimulcastConfig {
    enabled: boolean;
    enabledVariants: Variant[];
    disabledVariants: Variant[];
}
declare const MediaEvent_Track_SimulcastConfig: MessageFns<MediaEvent_Track_SimulcastConfig>;
interface MediaEvent_OfferData_TrackTypes {
    audio: number;
    video: number;
}
declare const MediaEvent_OfferData_TrackTypes: MessageFns<MediaEvent_OfferData_TrackTypes>;
type Builtin = Date | Function | Uint8Array | string | number | boolean | undefined;
type DeepPartial<T> = T extends Builtin ? T : T extends globalThis.Array<infer U> ? globalThis.Array<DeepPartial<U>> : T extends ReadonlyArray<infer U> ? ReadonlyArray<DeepPartial<U>> : T extends {} ? {
    [K in keyof T]?: DeepPartial<T[K]>;
} : Partial<T>;
type KeysOfUnion<T> = T extends T ? keyof T : never;
type Exact<P, I extends P> = P extends Builtin ? P : P & {
    [K in keyof P]: Exact<P[K], I[K]>;
} & {
    [K in Exclude<keyof I, KeysOfUnion<P>>]: never;
};
interface MessageFns<T> {
    encode(message: T, writer?: BinaryWriter): BinaryWriter;
    decode(input: BinaryReader | Uint8Array, length?: number): T;
    fromJSON(object: any): T;
    toJSON(message: T): unknown;
    create<I extends Exact<DeepPartial<T>, I>>(base?: I): T;
    fromPartial<I extends Exact<DeepPartial<T>, I>>(object: I): T;
}

type SerializedMediaEvent = Uint8Array;
interface MediaEvent {
    type: keyof MediaEvent$1;
    key?: string;
    data?: any;
}

type MediaStreamTrackId = string;
type TrackKind = 'audio' | 'video';
/**
 * Type describing Voice Activity Detection statuses.
 *
 * - `speech` - voice activity has been detected
 * - `silence` - lack of voice activity has been detected
 */
type VadStatus = 'speech' | 'silence';
/**
 * Type describing maximal bandwidth that can be used, in kbps. 0 is interpreted as unlimited bandwidth.
 */
type BandwidthLimit = number;
/**
 * Type describing bandwidth limit for simulcast track.
 * It is a mapping (encoding => BandwidthLimit).
 * If encoding isn't present in this mapping, it will be assumed that this particular encoding shouldn't have any bandwidth limit
 */
type SimulcastBandwidthLimit = Map<Variant, BandwidthLimit>;
/**
 * Type describing bandwidth limitation of a Track, including simulcast and non-simulcast tracks.
 * A sum type of `BandwidthLimit` and `SimulcastBandwidthLimit`
 */
type TrackBandwidthLimit = BandwidthLimit | SimulcastBandwidthLimit;
/**
 * Type describing possible reasons for currently selected encoding.
 * - `other` - the exact reason couldn't be determined
 * - `encodingInactive` - previously selected encoding became inactive
 * - `lowBandwidth` - there is no longer enough bandwidth to maintain previously selected encoding
 */
type EncodingReason = 'other' | 'encodingInactive' | 'lowBandwidth';
/**
 * Track's context i.e. all data that can be useful when operating on track.
 */
interface TrackContextFields {
    readonly track: MediaStreamTrack | null;
    /**
     * Stream this track belongs to.
     */
    readonly stream: MediaStream | null;
    /**
     * Endpoint this track comes from.
     */
    readonly endpoint: Endpoint;
    /**
     * Track id. It is generated by RTC engine and takes form `endpoint_id:<random_uuidv4>`.
     * It is WebRTC agnostic i.e. it does not contain `mid` or `stream id`.
     */
    readonly trackId: string;
    /**
     * Simulcast configuration.
     * Only present for local tracks.
     */
    readonly simulcastConfig?: MediaEvent_Track_SimulcastConfig;
    /**
     * Any info that was passed in {@link WebRTCEndpoint.addTrack}.
     */
    readonly metadata?: unknown;
    readonly maxBandwidth?: TrackBandwidthLimit;
    readonly vadStatus: VadStatus;
    /**
     * Encoding that is currently received.
     * Only present for remote tracks.
     */
    readonly encoding?: Variant;
    /**
     * The reason of currently selected encoding.
     * Only present for remote tracks.
     */
    readonly encodingReason?: EncodingReason;
}
interface TrackContextEvents {
    /**
     * Emitted each time track encoding has changed.
     *
     * Track encoding can change in the following cases:
     * - when user requested a change
     * - when sender stopped sending some encoding (because of bandwidth change)
     * - when receiver doesn't have enough bandwidth
     *
     * Some of those reasons are indicated in {@link TrackContext.encodingReason}.
     */
    encodingChanged: (context: TrackContext) => void;
    /**
     * Emitted every time an update about voice activity is received from the server.
     */
    voiceActivityChanged: (context: TrackContext) => void;
}
interface TrackContext extends TrackContextFields, TypedEmitter<Required<TrackContextEvents>> {
}
type TrackNegotiationStatus = 'awaiting' | 'offered' | 'done';
/**
 * Events emitted by the {@link WebRTCEndpoint} instance.
 */
interface WebRTCEndpointEvents {
    /**
     * Emitted each time WebRTCEndpoint need to send some data to the server.
     */
    sendMediaEvent: (mediaEvent: SerializedMediaEvent) => void;
    /**
     * Emitted when endpoint of this {@link WebRTCEndpoint} instance is ready. Triggered by {@link WebRTCEndpoint.connect}
     */
    connected: (endpointId: string, otherEndpoints: Endpoint[]) => void;
    /**
     * Emitted when endpoint of this {@link WebRTCEndpoint} instance was removed.
     */
    disconnected: () => void;
    /**
     * Emitted when data in a new track arrives.
     *
     * This event is always emitted after {@link trackAdded}.
     * It informs the user that data related to the given track arrives and can be played or displayed.
     */
    trackReady: (ctx: TrackContext) => void;
    /**
     * Emitted each time the endpoint which was already in the room, adds new track. Fields track and stream will be set to null.
     * These fields will be set to non-null value in {@link trackReady}
     */
    trackAdded: (ctx: TrackContext) => void;
    /**
     * Emitted when some track will no longer be sent.
     *
     * It will also be emitted before {@link endpointRemoved} for each track of this endpoint.
     */
    trackRemoved: (ctx: TrackContext) => void;
    /**
     * Emitted each time endpoint has its track metadata updated.
     */
    trackUpdated: (ctx: TrackContext) => void;
    /**
     * Emitted each time new endpoint is added to the room.
     */
    endpointAdded: (endpoint: Endpoint) => void;
    /**
     * Emitted each time endpoint is removed, emitted only for other endpoints.
     */
    endpointRemoved: (endpoint: Endpoint) => void;
    /**
     * Emitted each time endpoint has its metadata updated.
     */
    endpointUpdated: (endpoint: Endpoint) => void;
    /**
     * Emitted in case of errors related to multimedia session e.g. ICE connection.
     */
    connectionError: (error: {
        message: string;
        event: Event;
    }) => void;
    /**
     * Emitted in case of errors related to multimedia session e.g. ICE connection.
     */
    signalingError: (error: {
        message: string;
    }) => void;
    /**
     * Currently, this event is only emitted when DisplayManager in RTC Engine is
     * enabled and simulcast is disabled.
     *
     * Emitted when priority of video tracks have changed.
     * @param enabledTracks - list of tracks which will be sent to client from SFU
     * @param disabledTracks - list of tracks which will not be sent to client from SFU
     */
    tracksPriorityChanged: (enabledTracks: TrackContext[], disabledTracks: TrackContext[]) => void;
    /**
     * Emitted every time the server estimates client's bandwidth.
     *
     * @param {bigint} estimation - client's available incoming bitrate estimated
     * by the server. It's measured in bits per second.
     */
    bandwidthEstimationChanged: (estimation: bigint) => void;
    /**
     * Emitted each time track encoding has been disabled.
     */
    trackEncodingDisabled: (context: TrackContext, encoding: Variant) => void;
    /**
     * Emitted each time track encoding has been enabled.
     */
    trackEncodingEnabled: (context: TrackContext, encoding: Variant) => void;
    targetTrackEncodingRequested: (event: {
        trackId: string;
        variant: Variant;
    }) => void;
    disconnectRequested: (event: any) => void;
    localTrackAdded: (event: {
        trackId: string;
        track: MediaStreamTrack;
        stream: MediaStream;
        trackMetadata?: unknown;
        simulcastConfig: MediaEvent_Track_SimulcastConfig;
        maxBandwidth: TrackBandwidthLimit;
    }) => void;
    localTrackRemoved: (event: {
        trackId: string;
    }) => void;
    localTrackReplaced: (event: {
        trackId: string;
        track: MediaStreamTrack | null;
    }) => void;
    localTrackMuted: (event: {
        trackId: string;
    }) => void;
    localTrackUnmuted: (event: {
        trackId: string;
    }) => void;
    localTrackBandwidthSet: (event: {
        trackId: string;
        bandwidth: BandwidthLimit;
    }) => void;
    localTrackEncodingBandwidthSet: (event: {
        trackId: string;
        rid: Variant;
        bandwidth: BandwidthLimit;
    }) => void;
    localTrackEncodingEnabled: (event: {
        trackId: string;
        encoding: Variant;
    }) => void;
    localTrackEncodingDisabled: (event: {
        trackId: string;
        encoding: Variant;
    }) => void;
    localEndpointMetadataChanged: (event: {
        metadata: unknown;
    }) => void;
    localTrackMetadataChanged: (event: {
        trackId: string;
        metadata: unknown;
    }) => void;
}
/**
 * Interface describing Endpoint.
 */
interface Endpoint {
    /**
     * Endpoint's id. It is assigned by user in custom logic that use backend API.
     */
    id: string;
    /**
     * Type of the endpoint, e.g. "webrtc", "hls" or "rtsp".
     */
    type: string;
    /**
     * Any information that was provided in {@link WebRTCEndpoint.connect}.
     */
    metadata?: unknown;
    /**
     * List of tracks that are sent by the endpoint.
     */
    tracks: Map<string, TrackContext>;
}

declare class ConnectionManager {
    private readonly connection;
    constructor(iceServers: RTCIceServer[]);
    isConnectionUnstable: () => boolean;
    getConnection: () => RTCPeerConnection;
    addTransceiversIfNeeded: (serverTracks: MediaEvent_OfferData_TrackTypes) => void;
    addTransceiver: (track: MediaStreamTrack, transceiverConfig: RTCRtpTransceiverInit) => void;
    setOnTrackReady: (onTrackReady: (event: RTCTrackEvent) => void) => void;
    setRemoteDescription: (data: RTCSessionDescriptionInit) => Promise<void>;
    isTrackInUse: (track: MediaStreamTrack) => boolean;
    removeTrack: (sender: RTCRtpSender) => void;
    findSender: (mediaStreamTrackId: MediaStreamTrackId) => RTCRtpSender;
    addIceCandidate: (iceCandidate: RTCIceCandidate) => Promise<void>;
}

declare const TrackContextImpl_base: new () => TypedEmitter<Required<TrackContextEvents>>;
declare class TrackContextImpl extends TrackContextImpl_base implements TrackContext {
    endpoint: Endpoint;
    trackId: string;
    track: MediaStreamTrack | null;
    trackKind: TrackKind | null;
    stream: MediaStream | null;
    metadata?: unknown;
    metadataParsingError?: any;
    simulcastConfig?: MediaEvent_Track_SimulcastConfig;
    maxBandwidth: TrackBandwidthLimit;
    encoding?: Variant;
    encodingReason?: EncodingReason;
    vadStatus: VadStatus;
    negotiationStatus: TrackNegotiationStatus;
    pendingMetadataUpdate: boolean;
    constructor(endpoint: Endpoint, trackId: string, metadata: any, simulcastConfig?: MediaEvent_Track_SimulcastConfig);
}
type EndpointWithTrackContext = Omit<Endpoint, 'tracks'> & {
    tracks: Map<string, TrackContextImpl>;
};

declare const WebRTCEndpoint_base: new () => TypedEmitter<Required<WebRTCEndpointEvents>>;
/**
 * Main class that is responsible for connecting to the RTC Engine, sending and receiving media.
 */
declare class WebRTCEndpoint extends WebRTCEndpoint_base {
    private readonly localTrackManager;
    private readonly remote;
    private readonly local;
    private readonly commandsQueue;
    private proposedIceServers;
    bandwidthEstimation: bigint;
    connectionManager?: ConnectionManager;
    private clearConnectionCallbacks;
    constructor();
    /**
     * Tries to connect to the RTC Engine. If user is successfully connected then {@link WebRTCEndpointEvents.connected}
     * will be emitted.
     *
     * @param metadata - Any information that other endpoints will receive in {@link WebRTCEndpointEvents.endpointAdded}
     * after accepting this endpoint
     *
     * @example
     * ```ts
     * let webrtc = new WebRTCEndpoint();
     * webrtc.connect({displayName: "Bob"});
     * ```
     */
    connect: (metadata: unknown) => void;
    /**
     * Feeds media event received from RTC Engine to {@link WebRTCEndpoint}.
     * This function should be called whenever some media event from RTC Engine
     * was received and can result in {@link WebRTCEndpoint} generating some other
     * media events.
     *
     * @param mediaEvent - String data received over custom signalling layer.
     *
     * @example
     * This example assumes phoenix channels as signalling layer.
     * As phoenix channels require objects, RTC Engine encapsulates binary data into
     * map with one field that is converted to object with one field on the TS side.
     * ```ts
     * webrtcChannel.on("mediaEvent", (event) => webrtc.receiveMediaEvent(event.data));
     * ```
     */
    receiveMediaEvent: (mediaEvent: SerializedMediaEvent) => Promise<void>;
    private getEndpointId;
    private onTrackReady;
    /**
     * Retrieves statistics related to the RTCPeerConnection.
     * These statistics provide insights into the performance and status of the connection.
     *
     * @return {Promise<RTCStatsReport>}
     *
     * @see {@link https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getStats | MDN Web Docs: RTCPeerConnection.getStats()}
     */
    getStatistics(selector?: MediaStreamTrack | null): Promise<RTCStatsReport>;
    /**
     * Returns a snapshot of currently received remote tracks.
     *
     * @example
     * if (webRTCEndpoint.getRemoteTracks()[trackId]?.simulcastConfig?.enabled) {
     *   webRTCEndpoint.setTargetTrackEncoding(trackId, encoding);
     * }
     */
    getRemoteTracks(): Record<string, TrackContext>;
    /**
     * Returns a snapshot of currently received remote endpoints.
     */
    getRemoteEndpoints(): Record<string, EndpointWithTrackContext>;
    getLocalEndpoint(): EndpointWithTrackContext;
    getBandwidthEstimation(): bigint;
    private handleMediaEvent;
    private onSdpAnswer;
    /**
     * Adds track that will be sent to the RTC Engine.
     * @param track - Audio or video track e.g. from your microphone or camera.
     * @param trackMetadata - Any information about this track that other endpoints will
     * receive in {@link WebRTCEndpointEvents.endpointAdded}. E.g. this can source of the track - whether it's
     * screensharing, webcam or some other media device.
     * @param simulcastConfig - Simulcast configuration. By default simulcast is disabled.
     * For more information refer to {@link SimulcastConfig}.
     * @param maxBandwidth - maximal bandwidth this track can use.
     * Defaults to 0 which is unlimited.
     * This option has no effect for simulcast and audio tracks.
     * For simulcast tracks use `{@link WebRTCEndpoint.setTrackBandwidth}.
     * @returns {string} Returns id of added track
     * @example
     * ```ts
     * let localStream: MediaStream = new MediaStream();
     * try {
     *   localAudioStream = await navigator.mediaDevices.getUserMedia(
     *     AUDIO_CONSTRAINTS
     *   );
     *   localAudioStream
     *     .getTracks()
     *     .forEach((track) => localStream.addTrack(track));
     * } catch (error) {
     *   console.error("Couldn't get microphone permission:", error);
     * }
     *
     * try {
     *   localVideoStream = await navigator.mediaDevices.getUserMedia(
     *     VIDEO_CONSTRAINTS
     *   );
     *   localVideoStream
     *     .getTracks()
     *     .forEach((track) => localStream.addTrack(track));
     * } catch (error) {
     *  console.error("Couldn't get camera permission:", error);
     * }
     *
     * localStream
     *  .getTracks()
     *  .forEach((track) => webrtc.addTrack(track, localStream));
     * ```
     */
    addTrack(track: MediaStreamTrack, trackMetadata?: unknown, simulcastConfig?: MediaEvent_Track_SimulcastConfig, maxBandwidth?: TrackBandwidthLimit): Promise<string>;
    /**
     * Replaces a track that is being sent to the RTC Engine.
     * @param trackId - Audio or video track.
     * @param {string} trackId - Id of audio or video track to replace.
     * @param {MediaStreamTrack} newTrack
     * @returns {Promise<boolean>} success
     * @example
     * ```ts
     * // setup camera
     * let localStream: MediaStream = new MediaStream();
     * try {
     *   localVideoStream = await navigator.mediaDevices.getUserMedia(
     *     VIDEO_CONSTRAINTS
     *   );
     *   localVideoStream
     *     .getTracks()
     *     .forEach((track) => localStream.addTrack(track));
     * } catch (error) {
     *   console.error("Couldn't get camera permission:", error);
     * }
     * let oldTrackId;
     * localStream
     *  .getTracks()
     *  .forEach((track) => trackId = webrtc.addTrack(track, localStream));
     *
     * // change camera
     * const oldTrack = localStream.getVideoTracks()[0];
     * let videoDeviceId = "abcd-1234";
     * navigator.mediaDevices.getUserMedia({
     *      video: {
     *        ...(VIDEO_CONSTRAINTS as {}),
     *        deviceId: {
     *          exact: videoDeviceId,
     *        },
     *      }
     *   })
     *   .then((stream) => {
     *     let videoTrack = stream.getVideoTracks()[0];
     *     webrtc.replaceTrack(oldTrackId, videoTrack);
     *   })
     *   .catch((error) => {
     *     console.error('Error switching camera', error);
     *   })
     * ```
     */
    replaceTrack(trackId: string, newTrack: MediaStreamTrack | null): Promise<void>;
    /**
     * Updates maximum bandwidth for the track identified by trackId.
     * This value directly translates to quality of the stream and, in case of video, to the amount of RTP packets being sent.
     * In case trackId points at the simulcast track bandwidth is split between all of the variant streams proportionally to their resolution.
     *
     * @param {string} trackId
     * @param {BandwidthLimit} bandwidth in kbps
     * @returns {Promise<boolean>} success
     */
    setTrackBandwidth(trackId: string, bandwidth: BandwidthLimit): Promise<void>;
    /**
     * Updates maximum bandwidth for the given simulcast encoding of the given track.
     *
     * @param {string} trackId - id of the track
     * @param {string} rid - rid of the encoding
     * @param {BandwidthLimit} bandwidth - desired max bandwidth used by the encoding (in kbps)
     * @returns
     */
    setEncodingBandwidth(trackId: string, rid: Variant, bandwidth: BandwidthLimit): Promise<void>;
    /**
     * Removes a track from connection that was sent to the RTC Engine.
     * @param {string} trackId - Id of audio or video track to remove.
     * @example
     * ```ts
     * // setup camera
     * let localStream: MediaStream = new MediaStream();
     * try {
     *   localVideoStream = await navigator.mediaDevices.getUserMedia(
     *     VIDEO_CONSTRAINTS
     *   );
     *   localVideoStream
     *     .getTracks()
     *     .forEach((track) => localStream.addTrack(track));
     * } catch (error) {
     *   console.error("Couldn't get camera permission:", error);
     * }
     *
     * let trackId
     * localStream
     *  .getTracks()
     *  .forEach((track) => trackId = webrtc.addTrack(track, localStream));
     *
     * // remove track
     * webrtc.removeTrack(trackId)
     * ```
     */
    removeTrack(trackId: string): Promise<void>;
    /**
     * Sets track variant that server should send to the client library.
     *
     * The variant will be sent whenever it is available.
     * If chosen variant is temporarily unavailable, some other variant
     * will be sent until the chosen variant becomes active again.
     *
     * @param {string} trackId - id of track
     * @param {Encoding} variant - variant to receive
     * @example
     * ```ts
     * webrtc.setTargetTrackEncoding(incomingTrackCtx.trackId, "l")
     * ```
     */
    setTargetTrackEncoding(trackId: string, variant: Variant): void;
    /**
     * Enables track encoding so that it will be sent to the server.
     * @param {string} trackId - id of track
     * @param {Encoding} encoding - encoding that will be enabled
     * @example
     * ```ts
     * const trackId = webrtc.addTrack(track, stream, {}, {enabled: true, activeEncodings: ["l", "m", "h"]});
     * webrtc.disableTrackEncoding(trackId, "l");
     * // wait some time
     * webrtc.enableTrackEncoding(trackId, "l");
     * ```
     */
    enableTrackEncoding: (trackId: string, encoding: Variant) => Promise<void>;
    /**
     * Disables track encoding so that it will be no longer sent to the server.
     * @param {string} trackId - id of track
     * @param {Encoding} encoding - encoding that will be disabled
     * @example
     * ```ts
     * const trackId = webrtc.addTrack(track, stream, {}, {enabled: true, activeEncodings: ["l", "m", "h"]});
     * webrtc.disableTrackEncoding(trackId, "l");
     * ```
     */
    disableTrackEncoding: (trackId: string, encoding: Variant) => Promise<void>;
    /**
     * Updates the metadata for the current endpoint.
     * @param metadata - Data about this endpoint that other endpoints will receive upon being added.
     *
     * If the metadata is different from what is already tracked in the room, the optional
     * event `endpointUpdated` will be emitted for other endpoint in the room.
     */
    updateEndpointMetadata: (metadata: any) => void;
    /**
     * Updates the metadata for a specific track.
     * @param trackId - trackId (generated in addTrack) of audio or video track.
     * @param trackMetadata - Data about this track that other endpoint will receive upon being added.
     *
     * If the metadata is different from what is already tracked in the room, the optional
     * event `trackUpdated` will be emitted for other endpoints in the room.
     */
    updateTrackMetadata: (trackId: string, trackMetadata: any) => void;
    /**
     * Disconnects from the room. This function should be called when user disconnects from the room
     * in a clean way e.g. by clicking a dedicated, custom button `disconnect`.
     * As a result there will be generated one more media event that should be
     * sent to the RTC Engine. Thanks to it each other endpoint will be notified
     * that endpoint was removed in {@link WebRTCEndpointEvents.endpointRemoved},
     */
    disconnect: () => void;
    /**
     * Cleans up {@link WebRTCEndpoint} instance.
     */
    cleanUp: () => void;
    private getTrackId;
    private sendMediaEvent;
    private createAndSendOffer;
    private onOfferData;
    private setupConnectionListeners;
    private createNewConnection;
    private onRemoteCandidate;
    private onLocalCandidate;
    private onIceCandidateError;
    private onConnectionStateChange;
    private onIceConnectionStateChange;
}

export { type BandwidthLimit, type EncodingReason, type Endpoint, type MediaEvent, type SerializedMediaEvent, type SimulcastBandwidthLimit, MediaEvent_Track_SimulcastConfig as SimulcastConfig, type TrackBandwidthLimit, type TrackContext, type TrackContextEvents, type TrackKind, type VadStatus, Variant, WebRTCEndpoint, type WebRTCEndpointEvents };