Bitcrusher

A Bitcrusher is a lo-fi (low fidelity) digital audio effect, which produces a distortion by the reduction of the resolution or bandwidth of digital audio data. The resulting quantization noise may produce a “warmer” sound impression, or a harsh one, depending on the amount of reduction.

A typical bitcrusher uses two methods to reduce audio fidelity: sample rate reduction and resolution reduction.

Digital audio is composed of a rapid series of numeric samples that encode the changing amplitude of an audio waveform. To accurately represent a wideband waveform of substantial duration, digital audio requires a large number of samples at a high sample rate. The higher the rate, the more accurate the waveform; a lower rate requires the source analog signal to be low-pass filtered to limit the maximum frequency component in the signal, or else high-frequency components of the signal will be aliased. Specifically, the frequency of sampling (a.k.a. the sample rate) must be at least twice the maximum frequency component in the signal; this maximum signal frequency of one half the sampling frequency is called the Nyquist limit.

Though it is a common misconception that the sample rate affects the "smoothness" of the digitally represented waveform, this is not true; sampling theory guarantees that up to the maximum signal frequency supported by the sample rate (i.e. the Nyquist limit), the digital (discrete) signal will exactly represent the analog (continuous-wave) source, except for the distortion of quantization noise resulting from the finite precision of the individual samples. The original signal can be exactly reconstructed simply by passing the low-pass discrete signal through an ideal low-pass filter (with a perfect vertical cutoff profile). However, as an ideal filter is impossible to build, a real filter, with a gradual transition between the passband and the stopband, must be used, with the consequence that it is impossible to accurately record all frequencies right up to the Nyquist limit for a given sample rate. The solution is to increase the sample rate by an amount that accommodates the transition bands of the filters used both for sampling and for continuous-wave reconstruction; this is why, for example, Compact Discs use a sampling rate of 44.1 kHz to record audio that seldom exceeds 20 kHz, even though the Nyquist limit for this sample rate is 22.05 kHz. Another consideration is that for perfect reconstruction, the samples should be rendered as ideal impulses of infinitesimal duration, but all real hardware generates rectangular pulses for the samples; some lower-quality digital-to-analog conversion devices use step-wave conversion, which essentially outputs the samples as rectangular pulses that have a duration equal to the sampling period. In this case, too, an increase in the sample rate can reduce and compensate for the resultant distortion. Even so, it cannot be overemphasized that, regardless of its motivation, an extra margin added to the sampling frequency does /not/ make the reconstructed waveform smoother, it merely prevents aliasing of the frequencies in the transition band to lower frequencies, which would distort the signal nonlinearly.

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