 [内容简介]
[内容简介]
本书是作者继1978年版经典教材digital processing of speech signals之后的又一著作,本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。本书共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?、语音合成、语音识别和自然语言理解。 
  本书可供高等院校通信、电子、信息、计算机等专业作为研究生和本科生教材,也可以供有关科研和工程技术人员参考,是一本既有系统的基础理论讲解、又有最新研究前沿介绍并密切结合应用发展的教材。
[目次]
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 
 9.7 somepropertiesofthe lpcpolynomial a(z) 
 9.8 relationoflinear predictive analysisto losslesstube models 
 9.9 alternative representationsof thelpparameters 
 9.10 summary 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 
 10.7 summary 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 
 11.14 summary 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 
 12.8 summary 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 
 14.13 summary 984problems 
appendices 
 a speechandaudioprocessing demonstrations 
 b solutionoffrequency-domaindifferentialequations 
bibliography 
index