Computer modeling of voice signals with adjustable pitch and formant frequencies

Abstract

Digital encoding of speech to allow more efficient transmission at low data rates involves the decomposition of the speech waveform into various parameters which are related to the physical structure of the speech production process. In this thesis, linear predictive coding is used to produce a set of coefficients for the characteristic polynomial of successive 25 msec. segments of the voice track, in the z-domain. The location of the poles in the z-plane and the excitation pitch period are then shifted and the signal reformulated to cause changes of the overall frequency characteristics of the speech waveform, while maintaining the perceived sounds and information content. The resulting audio tapes confirm the theory and conjectures of the thesis.