We're not talking about the waveform here. A digital reverb does not see a waveform nor a sound. It does not see the samples as connected in any way, shape, or form. It applies the process to each individual sample as its own entity. All laws related to waveform capture and reproduction aren't really applicable in digital processing.
Digital processors that aren't very, very bad like the one demonstrated above use interpolation, but it is cheap and full of aliasing. The only reverb I have ever seen that doesn't alias without resampling is 2C-Audio's Aether. It's clean for the whole band and has zero aliasing, but processing time can be up to two hours for a 5 minute audio file.
I'm not sure I understand your question. You say that digital reverb only sees the samples, rather than the waveforms but then you show diagrams of how the reverb would see the waveform from the samples?
I agree of course that the reverb would only see the samples, now the actual digital signal processing mathematics itself I'm afraid is beyond my knowledge. Mention Fourier or Laplace transforms and my eyes glaze over! All I can tell you is that oversampling would be useful for a few digital processing tasks where the output of the process would result in frequencies above (or near to) the Nyquist Point, but algorithmic reverb is not (to my knowledge) one of them.