Speech analysis system

Claim 1

In a speech analysis system, the method of determining the formant parameters for a recursive digital all-pole filter whereby a function derived from the filter approaches, as closely as possible, a function derived from the speech, the method comprising the steps: sampling, at a predetermined rate, segments, of a specified duration,, of the speech signal; determining the auto-correlation coefficients r k from the signal samples s j , wherein: ##EQU6## determining the filter coefficients a j from the autocorrelation coefficients r k , wherein: ##EQU7## determining the coefficient combinations p i and q i of the n second-order sections of the digital all-pole filter, wherein the transfer function thereof is split into n second-order transfer functions: ##EQU8## where z -1 =exp (-sT), s being the complex frequency s=+jw and T the sampling period; transforming the coefficient combinations p i and q i into the coefficients c i and r i in accordance with the equations: ##EQU9## limiting the values of the coefficients c i and r i to values located in an area limited by the values c=-2, c=2, r=1 and r=0; arranging the coefficient combinations c i and r i in order of increasing values of c i ; and determining the formant parameters F i and B i using the equations: r.sub.i =e.sup.-πB.sbsp.i.sup.T, C.sub.i =-2 cos 2πF.sub.i T controlling said fiter utilizing said formant parameters to generate said filter-derived speech function.