标题：数字语音处理理论与应用
图书信息
出版社: 电子工业出版社; 第1版 (2011年1月1日)
外文书名: Theory and Applications of Digital Speech Processing
丛书名: 国外电子与通信教材系列
平装: 1042页
正文语种: 英语
开本: 16
ISBN: 7121124092, 9787121124099
条形码: 9787121124099
尺寸: 23.4 x 18.4 x 4.4 cm
重量: 1.4 Kg
作者简介
作者：(美国)劳伦斯 R.拉比纳(Lawrence R. Rabiner) (美国)罗纳德 W.谢弗(Ronald W. Schafer)
劳伦斯 R.拉比纳(Lawrence R.Rabiner)，美国工程院和美国科学院院士，美国声学学会、IEEE、Bell实验室、AT&T会士，以及Eta Kappa Nu、Sigma Xi、Tau Beta Pi等荣誉学会会员。曾担任美国声学学会副主席、IEEE Trans.ASSP主编和IEEE Proceedings编委会成员。其主要研究方向包括：通信、控制与信号处理、数字信号处理、数字语音处理、多媒体通信、多模态处理等。Rabiner教授于2002年从AT&T退休，随后担任Rutgers大学和加州大学圣巴巴拉分校的教授，以及Rutgers大学先进信息处理中心副主任。
内容简介
《数字语音处理理论与应用(英文版)》是作者继1978年版经典教材《语音信号数字处理》之后的又一著作，全书除有简练精辟的基础知识介绍外，系统介绍了近30年来语音信号处理的新理论、新方法和在应用上的新进展。全书共14章，分四部分：第一部分介绍语音信号处理基础知识，主要包括数字信号处理基础、语音产生机理、(人的)听觉和听感知机理和声道中的声传播原理；第二部分介绍语音信号的时频域表示和分析；第三部分介绍语音参数估计算法；第四部分介绍语音信号处理的应用，主要包括语音编码、语音和音频信号的频域编码、语音合成、语音识别和自然语言理解。
目录
Preface
CHAPTER 1 Introduction to Digital SpeechProcessing
1.1 The SpeechSignal
1.2 The SpeechStack
1.3 Applicationsof Digital SpeechProcessing 
1.4 Commentonthe References 
1.5 Summary 
CHAPTER 2 Reviewof Fundamentalsof DigitalSignalProcessing 
2.1 Introduction 
2.2 Discrete-Time Signals and Systems 
2.3 Transform Representation of Signals and Systems 
2.4 Fundamentalsof DigitalFilters 
2.5 Sampling 
2.6 Summary 
Problems 
CHAPTER 3 Fundamentalsof Human SpeechProduction 
3.1 Introduction 
3.2 The ProcessofSpeechProduction 
3.3 Short-TimeFourierRepresentationofSpeech 
3.4 AcousticPhonetics 
3.5 DistinctiveFeaturesof thePhonemesof American English 
3.6 Summary 
Problems 
CHAPTER 4 Hearing,Auditory Models,and SpeechPerception 
4.1 Introduction 
4.2 The SpeechChain 
4.3 Anatomy andFunctionof theEar 
4.4 The Perception of Sound 
4.5 Auditory Models 
4.6 Human SpeechPerceptionExperiments 
4.7 MeasurementofSpeechQualityand Intelligibility 
4.8 Summary 
Problems 
CHAPTER 5 Sound Propagationinthe HumanVocalTract 
5.1 The AcousticTheoryofSpeechProduction 
5.2 LosslessTube Models 
5.3 Digital Models forSampled SpeechSignals 
5.4 Summary 
Problems 
CHAPTER 6 Time-DomainMethods for SpeechProcessing 
6.1 Introduction 
6.2 Short-TimeAnalysisofSpeech 
6.3 Short-TimeEnergyand Short-TimeMagnitude 
6.4 Short-TimeZero-Crossing Rate 
6.5 The Short-TimeAutocorrelation Function 
6.6 The Modied Short-TimeAutocorrelation Function 
6.7 The Short-TimeAverage Magnitude DifferenceFunction 
6.8 Summary 
Problems 
CHAPTER 7 Frequency-DomainRepresentations 
7.1 Introduction 
7.2 Discrete-TimeFourierAnalysis 
7.3 Short-TimeFourierAnalysis 
7.4 SpectrographicDisplays 
7.5 OverlapAddition Methodof Synthesis 
7.6 Filter Bank SummationMethodof Synthesis 
7.7 Time-DecimatedFilter Banks 
7.8 Two-ChannelFilter Banks 
7.9 Implementationof theFBS Method Usingthe FFT 
7.10 OLARevisited 
7.11 Modicationsof theSTFT 
7.12 Summary 
Problems 
CHAPTER 8 TheCepstrumand Homomorphic SpeechProcessing 
8.1 Introduction 
8.2 HomomorphicSystems forConvolution 
8.3 HomomorphicAnalysisofthe SpeechModel 
8.4 Computingthe Short-TimeCepstrumand ComplexCepstrum of Speech 
8.5 HomomorphicFilteringofNatural Speech 
8.6 CepstrumAnalysisofAll-Pole Models 
8.7 CepstrumDistanceMeasures 
8.8 Summary 
Problems 
CHAPTER 9 Linear Predictive Analysisof SpeechSignals 
9.1 Introduction 
9.2 Basic Principles of Linear Predictive Analysis 
9.3 Computationofthe Gainfor theModel 
9.4 FrequencyDomainInterpretationsof Linear PredictiveAnalysis 
9.5 Solutionofthe LPCEquations 
9.6 The Prediction ErrorSignal 
9.7SomePropertiesofthe LPCPolynomial A（z） 
9.8 RelationofLinear Predictive Analysisto LosslessTube Models 
9.9 Alternative Representationsof theLPParameters 
9.10Summary 560Problems 
CHAPTER 10 Algorithms for Estimating SpeechParameters 
10.1 Introduction 
10.2 MedianSmoothing and SpeechProcessing 
10.3 Speech-Background/SilenceDiscrimination 
10.4 ABayesianApproach toVoiced/Unvoiced/Silence Detection 
10.5 Pitch Period Estimation（Pitch Detection） 
10.6 Formant Estimation 
10.7Summary 645Problems 
CHAPTER 11 DigitalCodingof SpeechSignals 
11.1 Introduction 
11.2 Sampling SpeechSignals 
11.3 AStatisticalModelfor Speech 
11.4 Instantaneous Quantization 
11.5 AdaptiveQuantization 
11.6 QuantizingofSpeechModelParameters 
11.7 GeneralTheoryof DifferentialQuantization 
11.8 Delta Modulation 
11.9 DifferentialPCM （DPCM） 
11.10 Enhancements forADPCM Coders 
11.11 Analysis-by-Synthesis SpeechCoders 
11.12 Open-Loop SpeechCoders 
11.13 Applicationsof SpeechCoders 
11.14Summary 819Problems 
CHAPTER 12 Frequency-DomainCodingof SpeechandAudio 
12.1 Introduction 
12.2 HistoricalPerspective 
12.3 Subband Coding 
12.4 AdaptiveTransform Coding 
12.5 APerception ModelforAudioCoding 
12.6 MPEG-1AudioCoding Standard 
12.7 OtherAudioCoding Standards 
12.8Summary 894Problems 
CHAPTER 13 Text-to-SpeechSynthesis Methods 
13.1 Introduction 
13.2 Text Analysis 
13.3 Evolutionof SpeechSynthesis Methods 
13.4 Early SpeechSynthesis Approaches 
13.5 UnitSelection Methods 
13.6 TTS Future Needs 
13.7 Visual TTS 
13.8Summary 947Problems 
CHAPTER 14 Automatic SpeechRecognition andNatural Language Understanding 
14.1 Introduction 
14.2 Basic ASRFormulation 
14.3 Overall SpeechRecognition Process 
14.4 Buildinga SpeechRecognition System 
14.5 The DecisionProcessesinASR 
14.6 Step3:The Search Problem 
14.7 SimpleASR System: IsolatedDigit Recognition 
14.8 Performance Evaluationof SpeechRecognizers 
14.9 SpokenLanguage Understanding 
14.10 Dialog Managementand SpokenLanguage Generation 
14.11 User Interfaces 
14.12 MultimodalUserInterfaces 
14.13Summary 984Problems 
Appendices
A Speechand Audio Processing Demonstrations 
B Solution of Frequency-Domain Differential Equations 
Bibliography 
Index