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This is a bane of digital audio systems. The sample rate of a system defines how fast the individual measurements (samples) that reconstruct a sound are recorded or played back. The bandwidth of that audio file (which corresponds to the highest frequency that can be reproduced) is in practice a bit less than half of the sample rate.

If you try to digitally record a signal that has harmonics above half of the sample rate, you either need to filter those out, or they will be mirrored at that half-sample-rate pivot point and folded back into the recorded signal. This mirror image can be thought of as an “alias” of the original sound’s content.

If you play back a digital audio file where half of the sample rate is an audible pitch, you will also hear a mirror image of the sound’s harmonic content reproduced started at that half-sample-rate pivot (unless, again, some excellent filtering has taken place). Aliasing can create some interesting, occasionally useful artifacts; on other occasions it’s better to use a high enough sample rate that aliasing never becomes audible.

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