Adaptive multi‐band filter structure‐based far‐end speech enhancement

Abstract

Speech signals degraded by noise tend to lose their quality and intelligibility. Therefore, the goal of speech enhancement algorithms is to restore these attributes of speech. The current work proposes a dynamic filter structure, dynamic over every utterance, that would vary simultaneously based on (i) the class of sound units in a given noisy/degraded signal, to improve intelligibility, and (ii) the noise components present, to improve quality. This filter structure is employed in the temporal-domain filtering-based multi-band speech enhancement algorithm, proposed by Jeeva et al.. The performance of the algorithm is evaluated subjectively and objectively, in terms of quality and intelligibility, and the algorithm is observed to successfully improve both attributes of degraded speech. Since the improvement in intelligibility depends on the effective restoration of sound units in the utterance, this process is language-specific. In this regard, an analysis is performed to study the influence of phone class distribution on the intelligibility improvement achieved for four Indian languages, namely Tamil, Hindi, Telugu, and Malayalam, and Indian English.