A question might come to mind to readers who are not familiar with this subject, as to why the subset of ‘Morphologies’ that is known as ‘Convolutions’ – i.e. ‘Linear Filters’ – is advantageous in filtering signals.
This is because even though such a static system of coefficients, applied constantly to input samples, will often produce spectral changes in the signal, they will not produce frequency components that were not present before. If new frequency components are produced, this is referred to as ‘distortion’, while otherwise all we get is spectral errors – i.e. ‘coloration of the sound’. The latter type of error is gentler on the ear.
For this reason, the mere realization that certain polynomial approximations can be converted into systems, that entirely produce linear products of the input samples, makes those more interesting.
OTOH, If each sampling of a continuous polynomial curve was at a random, irregular point in time – thus truly revealing it to be a polynomial – then additional errors get introduced, which might resemble ‘noise’, because those may not have deterministic frequencies with respect to the input.
And, the fact that the output samples are being generated at a frequency which is a multiple of the original sample-rate, also means that new frequency components will be generated, that go up to the same multiple.
In the case of digital signal processing, the most common type of distortion is ‘Aliasing’, while with analog methods it used to be ‘Total Harmonic Distortion’, followed by ‘Intermodulation Distortion’.
If we up-sample a digital stream and apply a filter, which consistently underestimates the sub-sampled, then the resulting distortion will consist of unwanted modulations of the higher Nyquist Frequency.