We've been using Nyquist's work and anti-aliasing filters for nearly as long as we've been using digital audio at all. Your DAW (or whatever) may be able to sh

We've been using Nyquist's work and anti-aliasing filters for nearly as long as we've been using digital audio at all. Your DAW (or whatever) may be able to show you the stairsteps of individual samples on a screen, but with a functional playback system it is never that way at all by the time things become analog again. Instead, it's always smoothed out by an anti-aliasing filter. It works this way regardless of sampling rate. The stairsteps don't make it outside of number-land. You can run your DAW at 48KHz, 96KHz, or 192KHz, and signals below the least-common-denominator cutoff frequency will be identical on an oscilloscope -- and free of stairsteps. (Try it sometime. It's fun.) Aliasing is a solved problem that has been solved for a long time. Your analogy about scaling and diagonal lines is actually a decent visual representation of how this stuff works, except it has already been working that way without being deliberately clever with overkill sampling rates. Meanwhile: This Pi Pico DSP stack is structured very heavily towards being the last digital stage of a listening system. As-constructed, it's quite clearly evident that it is really not meant to be anything else. A person can certainly bend it to be other things (yay open source!), but you've probably already got a set of filters well-integrated into your existing toolchain that work superbly. But if that's what you want, then by all means: Use it. Integer sampling rate conversions are trivial operations to get correct. To get the 96KHz that this project works with from your your 192KHz workflow, it's just a matter of throwing away half of the samples and playing back whatever remains. Any aliasing is out-of-band, and is removed by the anti-aliasing filter that is part of the digital-to-analog stage. You can't hear it if it isn't there. :)