/**
 * One live dictation session: microphone → WebRTC → realtime transcription.
 * The browser talks to the transcription service directly with a short-lived
 * ephemeral token minted by the backend — audio and transcripts never touch
 * OpenCX servers, and the real API key never reaches the browser.
 *
 * WebRTC (not WebSocket) because the browser stack handles Opus encoding,
 * jitter buffering and echo cancellation natively — no PCM plumbing, lowest
 * practical latency.
 *
 * Errors are CODES (the widget renders them through its translation table).
 */
export type DictationSessionState = 'idle' | 'connecting' | 'listening' | 'stopped' | 'error';
export type DictationErrorCode = 
/** No getUserMedia, permission denied, or no device. */
'microphone'
/** Mint, handshake, or transport failure. */
 | 'unavailable';
export interface DictationMint {
    token: string;
    expiresAt: string;
    model: string;
}
export declare class DictationSession {
    private readonly mintToken;
    private readonly onDelta;
    private readonly onState;
    private readonly onError;
    private readonly onLevel;
    private state;
    private pc;
    private dc;
    private stream;
    private audioContext;
    private levelRaf;
    private timers;
    private loggedParseFailure;
    constructor({ mintToken, onDelta, onState, onError, onLevel, }: {
        mintToken: () => Promise<DictationMint>;
        onDelta: (delta: string) => void;
        onState: (state: DictationSessionState) => void;
        onError: (code: DictationErrorCode) => void;
        onLevel?: (level: number) => void;
    });
    start(): Promise<void>;
    stop(): void;
    private requestMicrophone;
    private handleServerEvent;
    private startLevelMeter;
    private isTornDown;
    private fail;
    private teardown;
    private setState;
}
//# sourceMappingURL=dictation-session.d.ts.map