Statistical Model of Speech Signals Based on Composite Autoregressive System with Application to Blind Source Separation

Abstract. This paper presents a new statistical model for speech signals, which consists of a time-invariant dictionary incorporating a set of the power spectral densities of excitation signals and a set of all-pole filters where the gain of each pair of excitation and filter elements is allowed to vary over time. We use this model to develop a combined blind separation and dereverberation method for speech. Reasonably good separations were obtained under a highly reverberant condition