Fully Featured Audio DSP Firmware for the Raspberry Pi Pico

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. :)

Its an important consideration for everyone, since we all have our own tolerance levels for latency - think about it when you're on a Teams call and someone in your room starts speaking but you hear them a half second later on 'delay' ..

Latency is percievable by most people down to about 8-12ms .. lower than that and its harder to perceive, higher than that and you will get some people feeling like there are glitches in the audio ..

This is also important for musicians such as keyboard players, whose perception of their instrument is radically altered by that instruments latencies. Most modern synthesizers work very hard to get audio latency in their internal engines below 20ms ..

Why stop there? HDMI supports 1536kHz audio -- plenty to capture WLW's AM broadcast at 700KHz alongside the entire spectrum below it. :)

But even here in these comments, they're not arguing that 192kHz is insufficient. I haven't seen anyone ever make that argument, actually, so from my perspective this sampling rate represents a useful amount of overkill.

The extra data is a very small price to pay for silence.