Two identical sounds coming from two separate sources arriving at different times (due to differences in path lengths from speaker to listener) create comb filtering. Notches in the frequency response that move as you move around the space. In home theatre, the listening position rarely moves and is usually fairly small in variation from sources. In live audio, people move around an audience and the listening area is much greater in scale. Even without movement, the comb filtering will result in changes in frequency response from one seating position to another. Therefore, the mix engineer cannot “fix” things with EQ, because any EQ done to correct one position could be making it worse in another.
In nature, all sources of sound rarely come from the same point, the way they do with a sound reinforcement system. But once you collect all those sources with microphones, mix them to a nice balance and project them from one point in space, the interactions from other speakers as well as reflections from acoustic boundaries will degrade audio quality.
Google “comb filtering” and you’ll have days of reading ahead.
For software, it’s nearly impossible without a degree and a LOT of experience and data from the space (materials, angles, surface areas) to truly predict an indoor acoustic environment. People spend their lives honing those skills, but on site there needs to be verification and final adjustments done to the processing of the system to correct for assumptions. Therefore most software is for modelling the direct sound, since that’s the part that can be mathematically calculated with angles/distances and measured data from the speaker manufacturer.
Danley Sound Labs have a software program called Direct that is very good, it only has Danley loudspeakers in it. As the name implies, it does not model indoor spaces (the direct sound only, not reflected). Things in a space like this will change based on how many people are in the room (reflections will always be worse in an empty room). If you play around in Direct, make sure to look at the “splay” at numerous frequencies, it’s revealing to see just how little control even a large horn has on performance. Try high Q and low Q speakers and experiment. It should help visualize the invisible to connect the dots in our head of some of the concepts and expected commonalities.
DSP used in sound reinforcement systems is limited in that it cannot be so complex that it adds unnecessary signal delay which would then introduce new problems since the band is live in the room. This is especially critical for monitors. Some of the main things used to implement a good system design are polarity, signal delay, and crossovers. Very little EQ other than for box correction would be used. The term “EQ to the room” is fundamentally flawed in live audio. I say this so you don’t go chasing your tail trying to fix things with EQ where speaker positioning and selection would have been the correct tools.
