I don't think there's "one optimal" way for anything. For "naive" clipping it pretty much comes down to max(minValue,min(maxValue,value)) but then again such "naive" clipping aliases like hell, so personally I wouldn't ever want to clip audio in such a way, except maybe as a safe-guard in low-level driver interfacing code (for example to avoid integer wrap-around, which is even worse).
Typically you will need some utility stuff where you just want the "best" possible (meaning some trade-off between "fastest", "most precise" and "most flexible") approach, but I don't think you can build anything interesting out of such "utility only" code-base. You will probably need roughly one generic utility biquad (but with a plenty of ways to calculate parameters), and one or two FIR implementations, some sort of FFT, and so on. This is all stuff that you essentially implement once, and then you only touch that code, if you need more features or something. You want to learn it all though, so I wouldn't be surprised if schools made you "reinvent the wheels" here.
But you'll probably need just as much "custom code" for any non-trivial plugin: oscillators, musical filters, LFOs, envelope generators and followers, delay lines (both simple and modulated), waveshapers, etc. [edit: I'm not trying to imply that you wouldn't want to split such high-level components into various lower-level building blocks, but it gets very specific very quick]. Unless you're trying to do "yet another basic this-and-that" these will probably all require some custom approaches, specific to your requirements.
Remember that when it comes to music, the slightest differences can be meaningful (or completely meaningless; depends on both the differences themselves, and the context). I find that over the years, even my "standard utility code" has slowly drifted in a direction very specific to what I personally want to do.