Audio Sample Rate and Nyquist Calculator | Plan Your Session Format and Storage
Choose a sample rate, bit depth, and channel count to get the Nyquist frequency (the theoretical reproducible ceiling), the dynamic range that bit depth buys you, and the bitrate and uncompressed PCM file size. Nine presets from 8 kHz telephone audio up to 192 kHz mastering sit side by side.
Why use this tool?
Every time you spin up a DAW session you face the same questions: 44.1 or 48 kHz? How many gigabytes will a 24-bit, half-hour multitrack eat? Instead of digging through forum threads, this calculator turns the sampling theorem into a number you can act on:
- Know your ceiling: Nyquist frequency is just sample rate / 2, so a 44.1 kHz file tops out at 22.05 kHz — comfortably above the 20 kHz limit of human hearing.
- See the headroom: Dynamic range follows 6.02 × bits + 1.76 dB, the gap between your loudest sample and the quantization noise floor.
- Budget the disk: Bitrate is rate × bits × channels, and file size is that bitrate times your duration — handy before you hit record on a long take.
FAQ
Why is the CD standard 44.1 kHz? Human hearing tops out near 20 kHz. A 44.1 kHz rate puts the Nyquist frequency at 22.05 kHz, just above that limit, while leaving room for the anti-aliasing filter to roll off cleanly.
Does 24-bit actually sound better than 16-bit? The audible payoff is dynamic range, not frequency. 16-bit gives roughly 98 dB, 24-bit roughly 146 dB. That extra headroom is why tracking and mixing engineers capture at 24-bit even when the final delivery is 16-bit.
Is 96 kHz worth the extra storage? It pushes Nyquist to 48 kHz, well past anything you can hear. The real benefits are gentler filter design and cleaner pitch shifting — and the file is more than double the size of 44.1 kHz, which you can confirm here before committing.
Will the file size match my real WAV? It is the theoretical figure for uncompressed PCM (rate × bits × channels × seconds / 8). A real WAV or AIFF is slightly larger because of the header, and compressed formats like MP3 or AAC come out smaller.
📚 Fun Facts: Aliasing, the digital-only gremlin
The sampling theorem says you must sample at least twice as fast as the highest frequency you want to keep. Feed in something above the Nyquist frequency and it does not just disappear — it folds back down as aliasing, a phantom tone at a lower, wrong frequency. Analog tape never did this; it is a uniquely digital artifact.
That is why every analog-to-digital converter sits behind a low-pass anti-aliasing filter that strips everything above Nyquist before sampling. A big reason hi-res formats use such high rates is not extra audible bandwidth but the breathing room to design that filter with a gentle, less phase-distorting slope.