Speech Analysis-Synthesis System Based on Homomorphic Filtering

Abstract

A digital speechanalysis‐synthesis system based on a recently proposed approach to the deconvolution of speech is presented. The analyzer is based on a computation of the cepstrum considered as the inverse Fourier transform of the log magnitude of the Fourier transform. The transmitted parameters represent pitch and voiced unvoiced information and the low‐time portion of the cepstrum representing an approximation to the cepstrum of the vocal‐tract impulse response. In the synthesis, the low‐time cepstral information is transformed to an impulse response function, which is then convolved with a train of impulses during voiced portions or a noise waveform during unvoiced portions to reconstruct the speech. Since no phase information is retained in the analysis, phase must be regenerated during synthesis. Either a zero‐phase or minimum‐phase characteristic can be obtained by simple weighting of the cepstrum before transformation.