Digital Speech Transmission and Enhancement

DIGITAL SPEECH TRANSMISSION AND ENHANCEMENT

Enables readers to understand the latest developments in speech enhancement/transmission due to advances in computational
power and device miniaturization

The Second Edition of Digital Speech Transmission and Enhancement has been updated throughout to provide all the necessary details on the latest advances in the theory and practice in speech signal processing and its applications, including many new research results, standards, algorithms, and developments which have recently appeared and are on their way into state-of-the-art applications.

Besides mobile communications, which constituted the main application domain of the first edition, speech enhancement for hearing instruments and man-machine interfaces has gained significantly more prominence in the past decade, and as such receives greater focus in this updated and expanded second edition.

Readers can expect to find information and novel methods on:

• Low-latency spectral analysis-synthesis, single-channel and dual-channel algorithms for noise reduction and dereverberation

• Multi-microphone processing methods, which are now widely used in applications such as mobile phones, hearing aids, and man-computer interfaces

• Algorithms for near-end listening enhancement, which provide a significantly increased speech intelligibility for users at the noisy receiving side of their mobile phone

• Fundamentals of speech signal processing, estimation and machine learning, speech coding, error concealment by soft decoding, and artificial bandwidth extension of speech signals

Digital Speech Transmission and Enhancement is a single-source, comprehensive guide to the fundamental issues, algorithms, standards, and trends in speech signal processing and speech communication technology, and as such is an invaluable resource for engineers, researchers, academics, and graduate students in the areas of communications, electrical engineering, and information technology.

Auteur

Peter Vary is former Head of the Institute of Communication Systems at RWTH Aachen University, Germany. Professor Vary is a Fellow of IEEE, EURASIP, and ITG, and has been a Distinguished Lecturer of the IEEE Signal Processing Society.

Rainer Martin is Head of the Institute of Communication Acoustics at Ruhr-Universität Bochum, Germany. Professor Martin is a Fellow of the IEEE.
Both authors have been actively involved in speech processing research and teaching over several decades.