import TypedEmitter from "typed-emitter";
export type WillowClientEvents = {
    /** Emitted whenever the client wants to log something. This is verbose. */
    onLog: (msg: string) => void;
    /** Emitted whenever the client encounters an error. */
    onError: (msg: string) => void;
    /** Emitted whenever the client receives a data channel message from the server */
    onMessage: (msg: DataChannelMessage) => void;
    /** Emitted whenever the client receives a speech to text inference result from the server */
    onInfer: (results: {
        text: string;
        time: number;
    }) => void;
    /** Emitted when the WebRTC connection is closed for any reason. */
    onClose: () => void;
    /** Emitted when the WebRTC connection is first opened. */
    onOpen: () => void;
};
/** Config class for willow client.  */
export interface WillowClientConfig {
    /** See https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia */
    constraints?: MediaStreamConstraints;
    /** See https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection */
    rtcConfig?: RTCConfiguration;
    /** The host to connect to for infrance. This should include path. */
    host?: string;
}
/** Type of any data channel message returned by the server. */
export interface DataChannelMessage<T = any> {
    type: string;
    message?: string;
    obj?: T;
}
declare const WillowClient_base: new () => TypedEmitter<WillowClientEvents>;
/**
 * The main class for interacting with library. An event emitter that establishes and maintains connection.
 */
export declare class WillowClient extends WillowClient_base {
    config: WillowClientConfig;
    pc: RTCPeerConnection;
    dc: RTCDataChannel;
    stream: MediaStream | undefined;
    private lastStop;
    recording: boolean;
    constructor(config: WillowClientConfig);
    get track(): MediaStreamTrack | undefined;
    get sender(): RTCRtpSender | undefined;
    /** Indicates if the client is currently connected. */
    get connected(): boolean;
    /** Mutes the current client. Generally you want to call `stop()` instead. */
    mute(mute: boolean): Promise<void>;
    /** Starts sending recorded voice to the server. Call `stop()` to trigger inference. */
    start(): Promise<void>;
    /** Used internally. Can also be called if you want to send custom message to server on data channel. */
    sendMessage(message: DataChannelMessage): Promise<void>;
    /** Stops recording and triggers inference which will be emmited via "onInfer" event. */
    stop(): Promise<void>;
    /** Forces the WebRTC connection closed */
    disconnect(): Promise<void>;
    /** After the client is constructed call this to initalize and connect to ASR server.*/
    init(): Promise<void>;
    private negotiate;
}
export {};