Automatic Room Correction has won the Subwoofer Wars

I have ARC with my home theater system and also vastly more sophisticated and capable DEQX DSP with my two channel music system.  I have never been impressed with ARC, but the DEQX is a totally different matter. To begin with performing speaker correction before performing room correction is quite important to the final result.  I believe the total effect of the DEQX DSP is a big enough SQ improvement to outweigh any downside including an extra AD/DA for analog sources. Also it allows me to triamplify my my DIY fully horn loaded speakers for even more SQ improvement.

@dannad- I’ll let a copy/paste, from the information to which I earlier referred, be my last word on the subject as to whether an FFT program/algorithm can separate/discern arrival times:      "Once the impulse response has been obtained, it can be analysed to calculate information about how the room behaves. The simplest analysis is the FFT, to show the frequency response between the source and mic positions. However, we have some control over it. Altering which part of the impulse response is analysed by the FFT changes what aspect of the room’s response we see. The early part of the impulse response corresponds to the direct sound from the source to the mic, the shortest path between them. Sound that has bounced off the room’s surfaces has to travel further to reach the mic, which takes longer, so the later parts of the impulse response contain the contributions of the room. "Windowing" the impulse response to look at only the initial part shows us the frequency response of the direct sound with little or no contribution from the room. A window that includes later parts of the response lets us see how the room’s contribution alters the frequency response. The ability to separate the contributions of the direct and later (reflected) sound is a key difference between the frequency response derived from an impulse response and one we would get from an RTA, for example, which can only show the total combined response of source and room."         I’ll take their word for it, before yours!       Apologies, to Erik

The premise/purpose of the Fourier transform is simple: Any recorded sound can be broken up into discrete sinusoidal components. For instance, a square wave, no matter how perfectly square, can be decomposed into odd-order harmonic sine waves, despite the original square wave looking anything but sinusoidal.

It is not magical however, and the results will vary based on the portion of the recording analyzed.

FFT is also not a substitute for all digital signal analysis. You don’t need FFT to tell you what a visual inspection of an impulse response will, such as looking for reflections and time aligning speakers. Nor do you need FFT to create an algorithm to automatically set speaker delays.


https://en.wikipedia.org/wiki/Fourier_transform


Where the Fast Fourier Transform really transformed acoustics was in the nearly infinite resolution. We went from band limited (octave, 1/3 octave, etc) measurements to resolution bounded only by the sample length on the low end and the Nyquist frequency on the top. Outstanding.


https://en.wikipedia.org/wiki/Nyquist_frequency


And of course, let’s not forget the waterfall plots.

Best,

E