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.

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.