Audio Sample Rate Converter

Resample audio between 44.1kHz, 48kHz and more

Drag & drop or click to upload

WAV · MP3 · OGG · M4A · FLAC · OPUS

What is Audio Sample Rate Converter?

This tool resamples audio files between standard sample rates (8, 16, 22.05, 44.1, 48, and 96 kHz) entirely on your device. Fix mismatched rates before video editing, prepare voice recordings for speech APIs that require 16 kHz, or standardize a podcast's source files. Output is a WAV at the rate you choose.

Nine presets run from 8 kHz up to 96 kHz, with output in 16-bit, 24-bit or 32-bit float and a mono, stereo or keep-original channel choice. Source files up to 200 MB are accepted in WAV, MP3, FLAC, OGG, M4A and several more. The estimated output size is shown before you start the conversion.

How to use

  1. Drop in an audio file — WAV, MP3, OGG, or M4A are all accepted.
  2. Pick the target sample rate and choose mono or stereo output.
  3. Convert, preview the result, and download the resampled WAV file.

When to use

  • Matching a voice-over to a video timeline that runs at 48 kHz.
  • Downsampling recordings to 16 kHz mono for a speech-to-text service.
  • Getting a batch of podcast source files onto one shared rate.

Result

A developer preparing voice samples for a speech-recognition API drops in a 44.1 kHz stereo recording, selects 16 kHz mono, and downloads a WAV that matches the API's exact input requirements.

FAQ

Does converting 44.1 kHz up to 96 kHz make the audio sound better?
No. Resampling upward interpolates between samples that already exist, and nothing above the original Nyquist limit comes back. Raise the rate only when a mixer, plugin chain or hardware device insists on that exact number.
Which should I choose for my project, 44.1 kHz or 48 kHz, and why?
Music heading for CD and most streaming catalogues stays at 44.1 kHz. Anything travelling with picture, from a YouTube upload to broadcast, runs at 48 kHz. Matching the destination saves a second resample later in the chain.
Why is the downloaded WAV so much bigger than the MP3 I uploaded?
WAV stores every sample uncompressed while MP3 throws most of that data away. A stereo minute at 44.1 kHz and 16-bit runs about 10 MB, against roughly 1 MB for a 128 kbps MP3. Switching to mono or 16-bit trims it back.
Which bit depth should I pick, and is 32-bit float worth using?
16-bit suits any finished file: it matches CD and plays on everything. Choose 24-bit if the file goes back into a mix with more processing ahead. 32-bit float mainly helps when a later stage might push levels past 0 dBFS.

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