I'm curious if people incorporate deconvolution into their room correction algorithms? From a linear systems theory standpoint, it seems that it would be fairly straightforward to measure the impulse response of the speaker in the room, and then use this function as a kernel for a deconvolution filter. But I've read other information that the system is not linear (or at least not minimum phase), which doesn't make sense to me.
Can you shed light on if this is possible, and if so, do any of your plugins allow you to enter your impulse response function to use it as a real-time filter?
You can use Voxengo Deconvolver software to produce "inverse" filter that can be loaded in convolver.
The main problem of such approach is the presense of discontinuities in the spectrum (the points where signal turns into zero). Such discontinuities will translate into infinite boosts in the inverse filter.
Another problem is that frequency response changes in a spot-to-spot manner. So, you can create a good filter for one listening spot which may not be optimal in the other.
The best apporach is to build approximate room correction filters by means of spectrum matching. You record a white-noise signal in the room and then match the recording to the ideal white-noise spectrum. You can use CurveEQ for that purpose.
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