/**
 * Sample-rate conversion.
 *
 * The default is a windowed-sinc (Kaiser) interpolator — the same design SoX
 * and libsamplerate use. Cheaper alternatives are offered explicitly, because
 * the naive ones are the reason so much web audio sounds bad:
 *
 * - **Dropping or duplicating samples** (what a bare index-scale loop does)
 *   aliases badly and adds audible grit.
 * - **Linear interpolation** is a poor low-pass: it rolls off the top octave and
 *   still passes significant aliasing. Fine for a preview, wrong for a file
 *   somebody keeps.
 * - **Windowed sinc** approximates the ideal reconstruction filter. It costs
 *   more arithmetic and is worth it for anything that gets saved or shipped.
 *
 * Downsampling additionally needs the anti-alias cutoff moved *below* the new
 * Nyquist frequency before decimation. Skipping that step folds high frequencies
 * back down as inharmonic noise, which is the single most common resampling bug.
 */
export type ResampleQuality = 'fast' | 'good' | 'best';
export interface ResampleOptions {
    /**
     * `'best'` (default) windowed sinc, `'good'` cubic, `'fast'` linear.
     *
     * Use `'fast'` for realtime previews and waveform thumbnails; keep `'best'`
     * for anything the user will download.
     */
    quality?: ResampleQuality;
}
/** Output frame count for a given conversion. */
export declare function resampledLength(inputFrames: number, fromRate: number, toRate: number): number;
/**
 * Resamples one channel.
 *
 * @param input Source samples.
 * @param fromRate Source sample rate in Hz.
 * @param toRate Target sample rate in Hz.
 */
export declare function resampleChannel(input: Float32Array, fromRate: number, toRate: number, options?: ResampleOptions): Float32Array;
/** Resamples every channel of a planar buffer. */
export declare function resamplePlanar(channels: readonly Float32Array[], fromRate: number, toRate: number, options?: ResampleOptions): Float32Array[];