Spinoff
Company of
The
University of Parma
Computation of filter coefficients
•No theory is assumed: the set of hi,j filters are derived
directly from a set of impulse response measurements, designed according to a
least-squares principle.
•STEP1: a matrix C of impulse responses
is measured,
•STEP2: the target polar
pattern P of the virtual microphone is defined
•STEP3: the processing
filters H are found by imposing that
and
inverting the matrix.
•This way, the outputs of the microphone array are maximally
close to the ideal responses
prescribed
•This method also inherently corrects for transducer
deviations and acoustical artifacts
(shielding, diffractions, reflections, etc.)
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