# Audio Resampler

> Change sample rate (8 / 16 / 22.05 / 32 / 44.1 / 48 / 96 / 192 kHz) for compatibility or storage. High-quality resampling, no aliasing artefacts.

Canonical: https://helpers.aibrush.co/en/audio-resampler

## What it does

Pick a target sample rate. Downsampling shrinks the file roughly proportionally to the rate ratio — useful for voice content. Upsampling rarely improves anything but is occasionally required by picky downstream tools.

Resampling runs through the WebCodecs audio resampler, which applies a polyphase filter to avoid aliasing in the result.

## Sample-rate quick guide

48 kHz — video/film standard. 44.1 kHz — CD/streaming standard. 22.05 / 32 — voice content. 16 / 8 — telephony.

96 / 192 — pro-audio mastering. Don't upsample for production; downsample for the target medium.

## How to use it

1. **Drop the audio** — Any common audio format.
2. **Pick rate and format** — Output container can stay WAV or change to AAC/M4A/Ogg Opus. Sample-rate change forces a re-encode regardless.
3. **Process and download** — Filename includes the new rate (e.g. song-44100Hz.wav) so you can keep both versions.

## When to use it

### Speech-to-text input

Most STT services prefer 16 kHz mono. Resample before feeding in.

### Matching a video soundtrack

Video wants 48 kHz. If your audio source is 44.1, resample first to avoid the encoder doing it for you with worse quality.

### Pro audio mastering

Downsample a 96 kHz master to 44.1 kHz for CD distribution.

## FAQ

### Will resampling reduce quality?

Downsampling loses high frequencies above the new Nyquist limit. The polyphase filter avoids aliasing. Upsampling adds samples by interpolation — no new information, no quality gain.