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Sample Rate Converter Calculator

Calculate audio file sizes, bandwidth requirements, and Nyquist frequencies. Understand aliasing effects when converting between 44.1kHz, 48kHz, and 96kHz.

sec
Original File Size
-- MB
Nyquist: 22.05 kHz
Target File Size
-- MB
Nyquist: 24 kHz
Storage Difference
0%
No change

Interactive Aliasing & Sampling Simulator

Simulate how a sound frequency is captured and reconstructed at the target sample rate.

Original Sine Wave Reconstructed Wave
1,000 Hz
Bass (50 Hz)Midrange (1 kHz)Audible limit (20 kHz)Ultrasonic (50 kHz)

What is the Sample Rate Converter Calculator?

The Sample Rate Converter Calculator is an educational tool designed for audio engineers, podcasters, and music producers. It helps you calculate file sizes, signal bandwidths, and determine the Nyquist frequency limit when converting digital audio files between standard rates like 44.1kHz, 48kHz, and 96kHz.

Digital audio records sound waves as a series of snapshots, or samples, taken at a regular frequency. When converting an audio file to a lower sample rate, high-frequency details exceeding the new rate's Nyquist limit must be low-pass filtered out. Otherwise, they fold back into the audible spectrum, creating harsh harmonic distortion known as aliasing.

This calculator lets you simulate varying input frequencies against custom sample rates in real-time, clearly showing the point at which a reconstructed waveform becomes aliased. All calculations are performed locally in your browser with zero data sent to external servers.

How to Use the Calculator

  1. Select the Original Sample Rate and target bit depth for your source audio file.
  2. Select the Target Sample Rate you wish to convert the file into.
  3. Set the Channels configuration (mono, stereo, or surround) and enter the file duration.
  4. Review the calculated file sizes, storage difference, and respective Nyquist limits.
  5. Use the frequency slider below to observe how different frequencies are captured and see where aliasing begins.

The Nyquist Limit and Aliasing Explained

The Nyquist-Shannon sampling theorem states that to accurately reconstruct a signal, the sampling frequency must be greater than twice the highest frequency component of the signal. The threshold where the Nyquist limit equals half the sample rate is called the Nyquist frequency.

For example:

  • 44.1 kHz (CD) has a Nyquist limit of 22.05 kHz.
  • 48 kHz (Video) has a Nyquist limit of 24 kHz.
  • 96 kHz (Studio) has a Nyquist limit of 48 kHz.

If a signal is sampled at 44.1kHz but contains a high-frequency tone at 25kHz, it exceeds the 22.05kHz limit. The sampling points fail to track the speed of the wave, and it is reconstructed as a lower-frequency foldback tone at 19.1 kHz (calculated as 44.1 kHz minus 25 kHz). To prevent this, converters apply an anti-aliasing brickwall filter at the Nyquist frequency before downsampling.

Frequently Asked Questions

What is a sample rate in digital audio?

The sample rate is the number of samples of audio carried per second, measured in Hertz (Hz) or kilohertz (kHz). For example, a sample rate of 44.1kHz means the analog audio signal is sampled 44,100 times per second to create the digital representation.

Why are 44.1kHz and 48kHz the standard audio sample rates?

The human hearing range is approximately 20Hz to 20kHz. According to the Nyquist-Shannon sampling theorem, the sample rate must be at least twice the highest frequency we wish to record (which is 40kHz). 44.1kHz was chosen for CDs to fit digital audio onto video tape recorders, while 48kHz is the standard for professional video and television production.

What is aliasing in sample rate conversion?

Aliasing is a form of digital distortion that occurs when a signal contains frequencies higher than half of the sampling rate (the Nyquist limit). These high frequencies cannot be represented correctly and fold back or "alias" into the audible spectrum, creating artificial, harsh-sounding tones. An anti-aliasing low-pass filter is required to remove these frequencies before converting to a lower sample rate.

Does converting a 44.1kHz file to 96kHz improve its quality?

No. Up-sampling an audio file does not recreate lost high-frequency details. It simply increases the file size and spreads the existing frequency information across a larger number of samples. The audio quality will remain identical to the original 44.1kHz file, though it will consume more storage space.

How is audio file size calculated?

Audio file size is calculated by multiplying the sample rate (samples per second) by the bit depth (bits per sample), the number of channels (1 for mono, 2 for stereo), and the duration of the audio in seconds. The formula is: Size in bytes = (Sample Rate * Bit Depth * Channels * Duration) / 8.

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