How does the resampling work?

The file is decoded to PCM, then re-rendered through an OfflineAudioContext created at the target sample rate. The browser's built-in resampler handles the interpolation, and the result is encoded to a standard WAV.

Why is the output always a WAV?

WAV is uncompressed PCM that every browser can write reliably, so there is no quality loss from re-encoding to a lossy format. You can convert the WAV to MP3 or AAC afterward in your editor if you need a smaller file.

Will downsampling lose quality?

Lowering the sample rate reduces the highest frequency the file can represent (the Nyquist limit is half the rate). The resampler low-pass filters to prevent aliasing, but very low rates like 8000 Hz will sound noticeably duller.

No. Decoding, resampling, and WAV encoding all happen in your browser. Nothing is sent to a server, so the converter works offline and keeps your audio private.

Does it change the duration or pitch?

No. Resampling changes only how many samples per second represent the audio; the playback duration and pitch stay the same.

Audio Sample Rate Converter

Convert audio between sample rates locally

A sample-rate converter changes how many samples per second an audio file uses — for example from a 48000 Hz video master to 44100 Hz for CD, or down to 22050 Hz to shrink a voice recording. This tool does it entirely in your browser and gives you a ready-to-use WAV.

How it works

The input is decoded to raw PCM, then played through an OfflineAudioContext that is constructed at the target sample rate. Because the offline context renders at that rate, the browser’s high-quality resampler produces the converted samples for you. The new length is computed as ceil(duration × targetRate) so the full audio is captured.

The rendered buffer is then written to a 16-bit PCM WAV: a standard 44-byte RIFF/WAVE header followed by interleaved, clamped little-endian samples. The result is offered as a download via an object URL.

Tips and notes

Use 44100 Hz for music distribution (CD, most streaming) and 48000 Hz for anything paired with video.
Downsampling is lossy in the sense that high frequencies above the new Nyquist limit are removed; upsampling cannot add detail that was not there.
The output WAV is uncompressed, so files are larger than the source MP3 — re-encode to MP3/AAC afterward if size matters.
Everything is local, so very long files use significant memory; convert in sections if the browser slows down.