There is a field of study called Auditory Scene Analysis (ASA) that postulates that humans interpret “scenes” using sound just as they do using vision. I am not sure that it necessarily has any particular bearing on the way that audio hardware should be designed: basically the scene is all the clearer if the reproduction of the audio is clean in terms of noise, channel separation, distortion, frequency response and (seemingly controversial to hi-fi folk) the time domain.
However, the seminal work in this field includes the following analogy for hearing:
Your friend digs two narrow channels up from the side of a lake. Each is a few feet long and a few inches wide and they are spaced a few feet apart. Halfway up each one, your friend stretches a handkerchief and fastens it to the sides of the channel. As the waves reach the side of the lake they travel up the channels and cause the two handkerchiefs to go into motion. You are allowed to look only at the handkerchiefs and from their motions to answer a series of questions: How many boats are there on the lake and where are they? Which is the most powerful one? Which one is closer? Is the wind blowing? Has any large object been dropped suddenly into the lake?
Of course, when we listen to reproduced music with an audio system we are, in effect, duplicating the motion of the handkerchiefs using two paddles in another lake (our listening room) and watching the motion of a new pair of handkerchiefs. Amazingly, it works! But the key to this is that the two lakes are well-defined linear systems. Our brains can ‘work back’ to the original sounds using a process akin to ‘blind deconvolution’.
If we want to, we can eliminate the ‘second lake’ by using headphones, or we can almost eliminate it by using an anechoic chamber. We could theoretically eliminate it at a single point in space by deconvolving the reproduced signal with the measured impulse response of the room at that point. Listening with headphones works OK, but listening to speakers in a dead acoustic sounds terrible – probably to do with ‘head related transfer function’ (HRTF) telling us that we are listening to a ‘real’ acoustic but with an absence of the expected acoustic cues when we move our heads. By adding the ‘second lake’ we create enough ‘real acoustic’ to overcome that.
But here is why ‘room correction’ is flawed. The logical conclusion of room correction is to simulate headphones, but this cannot be achieved – and is not what most listeners want anyway, even if they don’t know it. Instead, an incomplete ‘correction’ is implemented based on the idea of trying to make the motion of the two sets of ‘handkerchiefs’ closer to each other than they (in naive measurements) appear to be. If the idea of the brain ‘working back’ to the original sound is correct, it will ‘work back’ to a seemingly arbitrarily modified recording. Modifying the physical acoustics of the room is valid whereas modifying the signal is not.
I think the problem stems ultimately from an engineering tool (frequency domain measurement) proliferating due to cheap computing power. There is a huge difference in levels of understanding between the author of the ASA book and the audiophiles and manufacturers who think that the sound is improved by tweaking graphic equalisers in an attempt to compensate for delays that the brain has compensated for already.