标准化语音编码算法基础与演进：标准编解码器详解

下载需积分: 0 | PDF格式 | 8.44MB | 更新于2025-01-09 | 118 浏览量 | 举报

《语音编码算法基础与标准化编码器的发展》一书由WAIC.CHU在MobileMediaLaboratory和DoCoMoUSALabs位于加利福尼亚州圣何塞的实验室编撰，是John Wiley & Sons, Inc.出版的作品。本书深入探讨了语音编码算法的基石以及其标准化历程。随着通信技术的快速发展，语音编码在音频压缩、无线传输和多媒体通信中扮演着关键角色。该书涵盖了编码原理、编码方法、以及标准制定如G.711、G.729、G.723等在国际电信联盟（ITU）框架下的演变。语音编码算法的基础部分主要介绍编码的数学原理，如线性预测编码（Linear Predictive Coding, LPC）、脉冲编码调制（Pulse Code Modulation, PCM）、以及声码器（Vocoder）的工作原理。这些编码技术利用人类听觉系统的特性，将连续的模拟声音信号转换成数字信号，以减小数据量并实现高效的通信效率。随着技术的进步，标准化编码器的发展历程着重讲述了一系列国际电信标准的制定过程。G.711是第一个广泛使用的全速率编码标准，主要用于电话网络，其后G.729和G.723等低比特率编码器被设计用于移动通信，以适应带宽有限的环境。这些标准不仅考虑了编码效率，还兼顾了语音质量的保真度，如信噪比和语音可懂度。书中可能还涉及编码器的性能指标，如编码速率、解码延时、误码率控制和复杂度分析，以及不同编码技术之间的比较和优化策略。此外，标准化编码器的未来趋势，如向更先进的编码方式如MPEG-4 AAC和VoIP优化的编码算法的转变，也将在书中有所提及。《语音编码算法基础与标准化编码器的发展》是一本全面而深入的指南，为读者揭示了语音编码技术从基础理论到实际应用的发展轨迹，对于从事通信工程、信号处理或音频技术的专业人士，以及对语音通信技术感兴趣的读者来说，都具有很高的参考价值。版权信息表明，所有内容未经许可不得复制或传播，除非符合美国1976年版权法的相关规定。

PREFACE

My ﬁrst contact with speech coding was in 1993 when I was a Field Application

Engineer at Texas Instruments, Inc. Soon after joining the company I was assigned

to design a demo prototype for the digital telephone answering device project.

Initially I was in charge of hardware including circuit design and printed circuit

board layout. The core of the board consisted of a microcontroller sending

commands to a mixed signal processor, where all the signal processing tasks—

including speech coding—were performed. In those days a major concern was the

excessive cost associated with random-access memory (RAM), and compressing

the digital speech before storing was almost a mandatory requirement, as this

greatly improved cost-effectiveness.

Soon after the hardware was ﬁnished, the focus switched to software (or ﬁrmware)

design, mainly dealing with the control of various on-board peripheral devices. My

true interest, however, was the program code inside the mixed signal processor,

which was developed by a separate team of ‘‘advanced’’ engineers. I was told that

voice signals were compressed using a code-excited linear prediction (CELP)

algorithm. Also, it was possible to play back ﬁxed announcement messages—such

as numbers and days of the week—with the messages stored in the linear prediction

coding (LPC) format. I had no idea what these algorithms were, nor how they

worked to compress speech. However, I was eager to learn the details, and decided

to g o back to school and pursue a PhD with concentration in speech coding.

This book is the result of my personal experience as a researcher and practitioner

in the ﬁeld of speech coding. Four years ago I decided to put in extra hours, usually

late nights and early mornings as well as weekends, to organize the literature in

speech coding and develop it into a logical presentation in terms of content and

terminology. Speech coding has evolved into a highly matured branch of signal

xiii

processing, with deployment in a plethora of products, such as, cellular phones,

answering machines, communication devices, and more recently, voice over

internet protocol (VoIP). It is obvious that a thorough textbook is necessary

for students, professors, and engineering professionals to handle the subj ect

appropriately. My sincere hope is that the availability of a book that collects

many of the techniques used in speech coding and presents them in an accessible

fashion will create excitement and enthusiasm, ensuring continuous rapid advances

in the ﬁeld.

Philosophy and Appro ach

Speech Coding Algorithms reﬂects the core subject of the book, since most coding

techniques are implemented as algorithms, or computational procedures performed

by a processor. However, this is by no means an exhaustive documentation of all

methods developed in this ﬁeld; it is rather the study of the most successful

techniques, deﬁned as those incorporated in a standard. By doing so we concentrate

our effort on understanding the most inﬂuential ideas, which is a rather efﬁcient

manner to navigate this vast territory of knowledge.

In my own personal learning curve, I found that there is a different and

refreshing lesson to be found in every standard. To understand a new standard it

is often necessary to look back into the developed techniques adopted by past

standards or studies. Attempting to lea rn by reading the ofﬁcial documentat ion

describing the standard is very often a frustrating experience, since the assumption

made in preparing those materials is that the audience consists of experts in the

subject, and hence the logical order and justiﬁcation of a given approach is

routinely omitted. Therefore the origin and the reason behind a certain practice

cannot be fully understood. This might not be a problem if one’s objective is to

implement the algorithm without comprehending it. However, for those researchers

eager to delve deeply into its roots, alternative reference sources must be explored,

which can be a strenuous and prolonged process. In this book I have summarized

the knowledge acquired over an extended period of time, with the intention of

ﬁlling the void between principles and imple mentations.

In writing this book, a balance is sought between theory and practice, and

between intuition and rigor. Theoretical ideas are include d only if they are used to

solve practical problems, and thorough proof s are provided. Speech coding is

related to human perception, and therefore a degree of fuzziness exists, in the sense

that no absolute right or wrong can be established for certain situations; in other

words, no mathematical proofs are obtainable. In these cases, solutions are often

found and justiﬁed on an intuitive basis. For the most part, the book is meant to be

pragmatic, since the discussed techniques are widely used in industry.

Prerequisites

The minimum background required to understand the book is explained, with

reference to popular textbooks where the relevant subjects can be found.

xiv PREFACE

 Advanced calculus, including complex variables [Churchill and Brown, 1990].

 Discrete-time signals and systems, Fourier transforms, z-transforms, ﬁltering,

and convolution [Oppenheim and Schafer, 1989; Stearns and Hush, 1990].

 Random variables and stochastic processes, expectation, probability, and

wide-sense stationarity [Papoulis, 1991; Peebles 1993].

 Linear algebra, including linear equations, matrices, and vectors [Strang,

1988].

 Experience with high-level programming using a language such as C.

The above list is covered in most undergraduate Electrical Engineering curricula;

with this background, the book is self-contained.

Organization

The text is divided into 19 chapters. Cha pter 1 provides an overview of the subjects

covered, with references to various aspects of speech coding, standards, algorithms,

and comments on notation and terminology. Cha pter 2 is a review of some signal

processing techniques, some are very general, but others are less known outside

speech coding literature. Chapter 3 contains some foundation for stochastic

processes and mode ls, which are important for an understanding of the theoretical

aspects. Chapter 4 is about linear prediction, the integral part of almost all modern

speech coders. Chapter 5 reviews the various aspects of scalar quantization, which

are utilized routinely by many speech coding algorithms. One of the earliest digital

coding techniques is pulse code modulation (PCM); it and its variants are the topic

of Chapter 6. Chapter 7 deals with vector quantization, which has become more and

more important for the achievement of high efﬁciency in coding systems. Linear

prediction coefﬁcients (LPC) are normally quantized for transmission as part of the

compressed bit-stream; Chapter 8 covers the various methods for scalar quantiza-

tion of these coefﬁcients. One of the landmarks in low bit-rate speech coding is the

linear prediction coding (LPC) algorithm, discussed in Chapter 9. Chapter 10

is devoted to regular pulse excitation coders, with a thorough description of the

GSM 6.10 standard. Principles of code-excited linear predictio n (CELP) are given

in Chapter 11, covering the various aspects of analysi s-by-synthesis, signal

calculation, postﬁlter design, and efﬁciency. Chapters 12 and 13 present the

structure of two standardized CELP coders: FS1016 and IS54, respectively; these

are both milestones in speech coding development. Chapter 14 is dedicated to

the G.72 8 low-delay CELP standar d, with thorough explanations of strategies for

delay reduction and detailed structures of the coder. Vector quantization of LPC

is included in Chapter 15, representing a huge advance with respect to scalar

quantization techniques covered in Chapter 8, and methods used by various

standardized coders are analyzed. The highly inﬂuential algebraic CELP (ACELP)

algorithm is covered in Chapter 16, where several ACELP-based standards are

described, with focus on the G.729 standard. The mixed excitation linear prediction

(MELP) algorithm is discussed in Chapter 17, and is shown to be an improvement

PREFACE xv

upon the LPC coder, covered in Chapter 9. Chapter 18 is devoted to the IS96

variable bit-rate CELP algorithm, which is a source-controlled multimode coder

with the operating mode selected by the input characteristics of the speech signal.

Finally, Chapter 19 is concerned with various methods to assess the quality of

speech signals, especially those processed by a speech coding algorithm.

The following table summarizes the chapter s and their prerequisites.

Chapter Title Prerequisites

1 Introduction

2 Signal Processing Techniques 1

3 Stochastic Processes and Models

4 Linear Prediction 1, 2, 3

5 Scalar Quantization

6 Pulse Code Modulation and its Variants 4, 5

7 Vector Quantization 5

8 Scalar Quantization of Linear Prediction Coefﬁcients 4, 5

9 Linear Prediction Coding 4, 8

10 Regular-Pulse Excitation Coders 4, 8

11 Code-Excited Linear Prediction 2, 4

12 The Federal Standard Version of CELP 2, 8, 11

13 Vector Sum Excited Linear Prediction 8, 12

14 Low-Delay CELP 4, 11

15 Vector Quantization of Linear Prediction Coefﬁcients 7 , 8

16 Algebraic CELP 7, 12, 15

17 Mixed Excitation Linear Prediction 9, 15

18 Source-Controlled Variable Bit-Rate CELP 11

19 Speech Quality Assessment 1

Acknowledgments

Throughout my professional career, I have had the opportunity to work with and

learn from a number of people whom I should like to publicly acknowledge. My

former advisor Dr. Nirmal K. Bose at the Pennsylvania State University had

provided me with invaluable instruction, trust, and friendship during my graduate

studies; his methodical style, hard-working spirit, and commitment toward educa-

tion have served as a role model to follow. I am grateful to my former supervisor

Dr. Tandhoni S. Rao at Texas Instruments Inc., who had guided me thr ough projects

involving adaptive ﬁlters, speech coding, and programming of digital signal

processors.

I would like to dedicate this book to my parents who have always encouraged

my academic interests and provided the moral support throughout my life and

career. I am deeply indebted to my cousin Chi-Ming Chu and wife Kam-Chi Chu

for their help and support during my graduate studies at Stevens Tech; their

industriousness and candid spirit have given me a great deal of positive inﬂuence.

xvi PREFACE

剩余584页未读，继续阅读

点点吃得太多了

粉丝: 180
资源: 680

标准化语音编码算法基础与演进：标准编解码器详解

语音编码算法基础与标准化编码器的发展

MATLAB实现LPC音频编码技术概述

"无失真信源编码方法及应用研究：理论与实践

Wiley.Speech.Coding.Algorithms.Foundation.and.Evolution.of.Standardized.Coders.eBook-KB.pdf

《SPEECH CODING ALGORITHMS》

Channel Coding Algorithms Theory and Applications

High Efficiency Video Coding Algorithms and Architectures

Speech Coding Using Sub-Bands:Speech Coding Using Sub-Bands-matlab开发

Coding Theory: Algorithms, Architectures and Applications

Turbo Coding For Generic RSC Coders.:Transmitter and Receiver Turbo Coding scheme with generic RSC Coders。-matlab开发

最新资源