When a signal is sampled, it is inherently band-limited in frequency.
In other words, when a signal is sampled by a finite number of points, it cannot represent an infinite range of frequencies.
A conventional D-to-A converter for audio will only create signals within a specific frequency range that is determined by the sampling rate.
If there are any recorded frequencies outside of this range, they are interpreted by the converter and mapped to frequencies within this range. This is aliasing — when one frequency is coded as a different frequency.
Nyquist frequency
The sampling rate determines this frequency range because it sets the Nyquist frequency.
The Nyquist frequency is the maximum frequency that can be recorded by a specific sampling rate. The Nyquist frequency is half of the sampling rate.
When it comes to audio recording, if the sampling rate is 48,000 samples per second, the Nyquist frequency is 24,000 Hz. If the sampling rate is 44,100 samples per second, the Nyquist frequency is 22,050 Hz.
If a signal contains any frequencies greater than the Nyquist frequency, they are interpreted by the converter and mapped to frequencies less than the Nyquist frequency.
Nyquist limit = 1/2 of the Sample Rate (S.R.)
(you don’t want to process limit)
Anti-Aliasing filter
Filtering to prevent sounds above Nyquist from being recorded.
Aliasing would be a big problem for digital audio, because it is usually not desired for frequencies to change in a signal. The good thing is that there is a dedicated component to prevent aliasing as part of the analog-to-digital and digital-to-analog conversion process. This component is called an anti-aliasing filter.
Conceptually, the anti-aliasing filter blocks frequencies above the Nyquist frequency from being converted. This is going to prevent any signals from changing frequency during the conversion process.
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