Jacob Benesty

This book brings together many advanced topics in network and acoustic echo cancellation which are aimed towards enhancing the echo cancellation performance of next-generation telecommunication systems. The general subject nature relates to algorithms with increased convergence speed, improved detection of double-talk from near-end speech, robust immunity to undetected double-talk, increased computational efficiency, and multi-channel capability. The resulting compendium provides a coherent treatment of such topics not found otherwise in journals or other books. The chapters are related with a common terminology, but still can be read independently.

Microphone arrays have the potential to solve many important problems in both human-machine and human-human interfaces for different kinds of communications. The main objective of this book is to derive and explain the most fundamental algorithms from a strictly broadband (signals and/or processing) viewpoint.

This book is devoted to the study of the problem of speech enhancement whose objective is the recovery of a signal of interest (i.e., speech) from noisy observations. Typically, the recovery process is accomplished by passing the noisy observations through a linear filter (or a linear transformation). Since both the desired speech and undesired noise are filtered at the same time, the most critical issue of speech enhancement resides in how to design a proper optimal filter that can fully take advantage of the difference between the speech and noise statistics to mitigate the noise effect as much as possible while maintaining the speech perception identical to its original form. The optimal filters can be designed either in the time domain or in a transform space. As the title indicates, this book will focus on developing and analyzing optimal filters in the Karhunen-Loève expansion (KLE) domain. We begin by describing the basic problem of speech enhancement and the fundamental principles to solve it in the time domain. We then explain how the problem can be equivalently formulated in the KLE domain. Next, we divide the general problem in the KLE domain into four groups, depending on whether interframe and interband information is accounted for, leading to four linear models for speech enhancement in the KLE domain. For each model, we introduce signal processing measures to quantify the performance of speech enhancement, discuss the formation of different cost functions, and address the optimization of these cost functions for the derivation of different optimal filters. Both theoretical analysis and experiments will be provided to study the performance of these filters and the links between the KLE-domain and time-domain optimal filters will be examined.