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 2nd edition.

In the Second Edition of Digital Speech Transmission and Enhancement, 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.

使讀者能夠理解由於計算能力和設備微型化的進步而產生的語音增強/傳輸的最新發展。《數字語音傳輸和增強》第二版已經全面更新，提供了有關語音信號處理及其應用的理論和實踐的最新進展所需的所有細節，包括許多新的研究結果、標準、算法和最新的發展，這些發展即將應用於最先進的應用中。除了移動通信是第一版的主要應用領域外，過去十年中，聽力器材和人機界面的語音增強顯著提升，因此在這本更新和擴展的第二版中更加關注。在《數字語音傳輸和增強》第二版中，讀者可以期待找到以下信息和新方法：低延遲的頻譜分析合成、單通道和雙通道的降噪和去混響算法。多麥克風處理方法，現在在移動電話、助聽器和人機界面等應用中廣泛使用。近端聆聽增強算法，為使用者在噪聲接收端的移動電話上提供顯著提高的語音可懂度。語音信號處理、估計和機器學習的基礎，語音編碼、軟解碼的錯誤隱藏，以及語音信號的人工頻寬擴展。《數字語音傳輸和增強》是一本全面的單一來源指南，涵蓋了語音信號處理和語音通信技術的基本問題、算法、標準和趨勢，因此對通信、電氣工程和信息技術領域的工程師、研究人員、學者和研究生來說是一個寶貴的資源。

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

Professor Rainer Martin is Head of the Institute of Communication Acoustics at Ruhr-Universität Bochum, Germany. Prof. Martin is a Fellow of the IEEE.

Both authors have been actively involved in speech processing research and teaching over several decades.

彼得·瓦里教授是德國亞琛工業大學通信系統研究所的前所長。瓦里教授是IEEE、EURASIP和ITG的會士，並曾擔任IEEE信號處理學會的傑出講師。 雷納·馬丁教授是德國波鴻大學通信聲學研究所的所長。馬丁教授是IEEE的會士。這兩位作者在語音處理研究和教學方面積極參與了數十年。