Re: Seeking Help Developing a Crude Room Correction Algorithm

If I interpreted what you said correctly, you have a stimulus (audio
being played through a speaker) and a microphone in a room, and you
would like to correct for the distortion caused by the room at the
microphone.

Look up "Adaptive Filter." The gist of it is that you run your source
through a filter prior to it leaving the speaker. The coefficients of
the filter are changed based on the error between the original source
and the sound from the microphone. It's pretty easy to grasp once you
try a concrete example. Other things to google: "LMS adaptive filter".
Most books on adaptive filters introduce LMS in a fairly
straightforward way. There are better alternatives, but it's easiest
to grasp at first.

Beware: try this in Matlab first with a short impulse response. Room
impulse responses are fairly long and require a bit more effort to
work with.

Chris

On Apr 27, 4:38 pm, "gener" <gyp...@xxxxxxx> wrote:
Hi all, first time poster here, and was hoping to get some advice on class
project. What I'm trying to do is to correct the impulse response of my
room by filtering the stimulus signal so that the new IR would be "flat".
Then, hopefully sending out the new stimulus signal would result in a flat
frequency IR — this is crude (i'm a beginner), and I'll probably avoid
dealing with phase at this point.

If I want to scale the frequency bins of the stinmulus signal by the
inverse of the magnitudes of the IR, any suggestions on how to make sure
the bins in the stimulus and the IR are the same as I take the FFT of
each?

Basically I want to convolve (.*) the ffts of the stim and the IR and then
take the ifft of that result.

Thanks!

.

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