Change the sample rate of WAV, MP3, FLAC and other audio files to any rate from 8 kHz to 96 kHz, and set the bit depth at the same time. A high-quality windowed-sinc filter prevents aliasing, and whole batches are processed on your own device.
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Output format
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How to use the Audio Sample Rate Converter
Add your audio files: drop them on the box or click to choose. The current sample rate of each file is shown.
Choose the target: pick a sample rate (or a preset such as CD or Video) and the bit depth.
Pick the quality: High is excellent for almost everything; Best uses a longer filter for critical work.
Press Convert. Each file shows the change, for example 48 kHz → 44.1 kHz.
Download each file, or press Download all (ZIP).
Common sample rates and where they are used
Sample rate
Highest frequency (Nyquist)
Typical use
16-bit stereo WAV per minute
8 kHz
4 kHz
Telephone, intercoms
1.9 MB
16 kHz
8 kHz
Speech recognition, VoIP, wideband calls
3.8 MB
22.05 kHz
11.025 kHz
Old games, low-bandwidth audio
5.3 MB
32 kHz
16 kHz
FM radio, DV camcorders
7.7 MB
44.1 kHz
22.05 kHz
CD, music streaming
10.6 MB
48 kHz
24 kHz
Video, broadcast, most DAWs
11.5 MB
96 kHz
48 kHz
Hi-res recording and mastering
23 MB
WAV size per second = sample rate × channels × bit depth ÷ 8. For example 48,000 × 2 × 24 ÷ 8 = 288,000 bytes per second, about 17.3 MB per minute.
How the resampler works
Each output sample is computed as a weighted sum of nearby input samples, where the weights come from a sinc function shaped by a Kaiser window. When downsampling, the filter's cut-off is placed just below the new Nyquist frequency, so content that cannot be represented is removed instead of aliasing. High quality uses 16 input samples on each side of every output sample (32 taps), Best uses 64 taps for a steeper filter. Fast uses the browser's built-in resampler, which is quicker but may be less precise depending on your browser.
Tips
Match the rate of your project: mixing a 44.1 kHz clip into a 48 kHz video project without converting it makes it play at the wrong speed and pitch in some editors.
Convert sample rate once, from the original file. Repeated conversions add small errors each time.
For speech-to-text, 16 kHz mono 16-bit WAV is the usual requirement.
If a file is louder than full scale after resampling (possible with heavily limited masters), the tool lowers the level just enough to avoid clipping and tells you by how much.
Frequently asked questions
What is the sample rate of audio?
It is how many times per second the waveform is measured. 44,100 Hz (44.1 kHz) is the CD and music standard, 48 kHz is the standard for video and most audio interfaces, 16 kHz is common for speech recognition and VoIP, and 8 kHz is telephone quality. A recording can only contain frequencies up to half its sample rate, the Nyquist frequency.
How do I convert 48 kHz to 44.1 kHz (or the reverse)?
Add the file, choose 44.1 kHz (or 48 kHz) and press Convert. The ratio between these rates is 147:160, so every output sample is calculated by interpolation with a band-limited sinc filter rather than by dropping or repeating samples.
Does upsampling improve quality?
No. Converting 44.1 kHz to 96 kHz makes the file larger but adds no detail above the original 22.05 kHz limit. Upsample only when software or hardware requires a specific rate.
What happens to high frequencies when I downsample?
Everything above the new Nyquist frequency is removed by the anti-aliasing filter: at 16 kHz the audio keeps content up to about 8 kHz. Without this filter those frequencies would fold back as audible distortion called aliasing.
What bit depth should I choose?
16-bit is CD quality with about 96 dB of dynamic range and is right for finished audio. 24-bit (about 144 dB) is standard for recording and editing. 32-bit float never clips during processing. 8-bit is only for retro or very low-fidelity uses.
What is dither and should I use it?
When reducing bit depth, rounding creates a small distortion that follows the music. Dither adds a tiny amount of random noise (TPDF, about ±1 least significant bit) that turns this distortion into a constant, much less noticeable hiss. Keep it on for 16-bit and 8-bit output.
Are my files uploaded?
No. Decoding, resampling and encoding run in your browser using the Web Audio API and JavaScript, so the audio never leaves your device.
Are all the tools on FreeToolCabin tested for quality and accuracy?
Yes! Every tool on our site undergoes automated Q/A testing in real browser environments to ensure high precision, correct output, and zero errors. If you happen to spot a bug or have a suggestion, please contact us at [email protected] and our team will review it right away.