In this paper we describe a cepstral model of the vocal tract which models both formants and antiformants.
The investigated model is more precise compared to the linear prediction model, which models
only the formants of the vocal tract. The expone
ntial function is used for the inverse transformation.
However, it is difficult to implement this function on a digital signal processor. To solve this issue we use a
continued fraction expansion to approximate the exponential function. The transfer function that
approximates the exponential function is realized by using the Infinite Impulse Response (IIR) digital
filter, in which branches type Finite Impulse Response (FIR) digital filters are included. The coefficients
of the FIR digital filters are just the coefficients of the real speech cepstrum. The state-space difference
equations are proposed and implemented on a DSP56300 fixed-point digital signal processor (Motorola).
Finally, the results of the digital signal processor implementation for chosen vowels and consonants are
evaluated.
