If I were designing a live audio workflow from scratch, my intuition would be to sample at a somewhat high frequency (at least 48kHz but maybe 96kHz), do the math to figure out the actual latency / data rate tradeoff, but to also filter the data as needed to minimize high frequency content (again, being careful with latency and fidelity tradeoffs).

But I have never done this and don't have any plans to do so, so I'll let other people worry about it. But maybe some day I'll carry out my evil plot to write an alternative to brutefir that gets good asymptotic complexity without adding latency. :)