Before you understand how foldback distortion works you may need to have a basic knowledge of audio distortion in general. Of course, there are several types of distortion but in this case, let's focus on hard-clipping distortion to keep things simple.

Okay, so in the digital domain, hard-clipping distortion is a result of pushing an audio signal beyond the maximum capacity of what a digital system can handle.

For instance, a DAW music program (e.g., FL Studio, Cubase, Ableton, Presonus Studio One, Logic, Pro Tools and etc) will typically have a limit of 0 dBFS (i.e. particularly for 16-bit or 24-bit audio) meaning in order to avoid hard-clipping your music shouldn't go past that point, otherwise, distortion will be audible on the master fader. However, this won't happen with 32-bit or 64-bit floating-point audio.

Foldback distortion definition​

Now If you increase the amplitude of a clean sine wave signal beyond 0 dBFS. This means on an oscilloscope you should notice the sine wave being flat-lined. That's because all the peaks above 0 dBFS are essentially chopped off (or squared off) in the digital audio domain.

But with foldback distortion, all the peaks that go past beyond the maximum capacity of a digital audio system aren't chopped off as they would in hard-clipping distortion.

Instead, foldback distortion means all those peaks are inverted upside down (or made to backflip in the opposite direction) hence the term foldback. The final result usually sounds like Frequency modulation synthesis (or FM synthesis in short).
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