Change the sample rate. Match your project, fix mismatch issues.
Need to convert a 48 kHz file to 44.1 kHz to bounce a CD master? A 44.1 kHz file to 48 kHz to drop into a video project? Drop the audio, pick the target rate, get back the same audio resampled cleanly with a proper anti-aliasing filter. No DAW round-trip needed.
drop your audio file
WAV, MP3, FLAC, M4A, anything — we'll handle the decode. Batch supported.
Common sample rates and what they're for
- 8 kHz — Telephone-quality audio. Tin can sound. Used for very small voicemail files.
- 16 kHz — Speech recognition. OpenAI Whisper and most ASR engines downsample to this internally.
- 22.05 kHz — Voice memo, old phone audio. Half-CD-quality.
- 44.1 kHz — CD audio, music streaming (Spotify, Apple Music, Bandcamp). The default for finished music.
- 48 kHz — Video, broadcast, YouTube, film, DAW projects for video work. The default for finished audio-for-picture.
- 88.2 kHz — 2× CD rate, sometimes used in mastering for clean integer-ratio downsampling.
- 96 kHz — 2× video rate, high-res audio downloads, demanding DAW projects.
- 192 kHz — Very high-res. Rarely audibly different from 96 kHz. Sometimes required by audiophile labels.
Why resampling matters
Your DAW project is locked to one sample rate (set when you created the project). Audio imported at a different rate will either be auto-resampled on the fly (usually fine, sometimes wrong) or played at the wrong speed and pitch (bad). Matching the sample rate up front avoids the question.
Some specific cases:
- Music for video: original at 44.1 → convert to 48 before delivering to a video editor.
- CD master from a 48 kHz session: convert your final bounce from 48 to 44.1 before sending to a CD authoring tool.
- Voice for Whisper: any rate → 16 kHz. We have a dedicated preset for that.
- Sampler upload: SP-404 wants 48 kHz, MPC wants 44.1, Splice expects 44.1. Sampler presets page picks the right one for you.
What we do under the hood
We decode the file using the browser's audio engine, then resample through an OfflineAudioContext at the target rate. The Web Audio API uses a high-quality polyphase filter for resampling — better than naive linear interpolation. The output is then re-encoded as either WAV (lossless) or MP3 (lossy but smaller).
FAQ
Will resampling change how my audio sounds?
For typical music and voice, no audible difference. For very content-heavy material at very low target rates (e.g., music at 16 kHz), you'll hear loss of high frequencies — that's the point of going to 16 kHz. For 44.1 ↔ 48 ↔ 96, you'll hear nothing.
Should I go up to 96 kHz from a 44.1 source?
It doesn't add quality (you can't invent samples you didn't record) but doesn't hurt either if a destination requires 96. Just makes the file bigger.
Why does the file get smaller when I downsample?
Fewer samples per second = fewer bytes per second. A WAV at 22 kHz is half the size of a WAV at 44 kHz. Useful for archiving voice content where high frequencies don't matter.
What's the right rate for an audiobook?
22.05 kHz mono is plenty for narration. Cuts file size by 75% versus 44.1 stereo with no audible loss for voice.
What's the right rate for a podcast?
44.1 kHz stereo or mono. Most podcast hosts accept 44.1 / 48 either way; 44.1 is slightly more space-efficient.
Does anything upload?
No. Resampling and re-encoding both run in your browser.