數(shù)字語音處理理論與應(yīng)用

內(nèi)容概要

　　本書是作者繼1978年版經(jīng)典教材digital processing of speech signals之后的又一著作，本書除有簡(jiǎn)練精辟的基礎(chǔ)知識(shí)介紹外，系統(tǒng)介紹了近30年來語音信號(hào)處理的新理論、新方法和在應(yīng)用上的新進(jìn)展。本書共14章，分四部分：第一部分介紹語音信號(hào)處理基礎(chǔ)知識(shí)，主要包括數(shù)字信號(hào)處理基礎(chǔ)、語音產(chǎn)生機(jī)理、(人的)聽覺和聽感知機(jī)理和聲道中的聲傳播原理；第二部分介紹語音信號(hào)的時(shí)頻域表示和分析；第三部分介紹語音參數(shù)估計(jì)算法；第四部分介紹語音信號(hào)處理的應(yīng)用，主要包括語音編碼、語音和音頻信號(hào)的頻域編?、語音合成、語音識(shí)別和自然語言理解。　　本書可供高等院校通信、電子、信息、計(jì)算機(jī)等專業(yè)作為研究生和本科生教材，也可以供有關(guān)科研和工程技術(shù)人員參考，是一本既有系統(tǒng)的基礎(chǔ)理論講解、又有最新研究前沿介紹并密切結(jié)合應(yīng)用發(fā)展的教材。

作者簡(jiǎn)介

　　Lawrence R．Rabiner，美國(guó)工程院和美國(guó)科學(xué)院院士，美國(guó)聲學(xué)學(xué)會(huì)、IEEE、Bell實(shí)驗(yàn)室、AT&T會(huì)士，以及Eta Kappa Nu、Sigma Xi、Tau Beta Pi等榮譽(yù)學(xué)會(huì)會(huì)員。曾擔(dān)任美國(guó)聲學(xué)學(xué)會(huì)副主席、IEEE Trans．ASSP主編和IEEE Proceedings編委會(huì)成員。其主要研究方向包括：通信、控制與信號(hào)處理、數(shù)字信號(hào)處理、數(shù)字語音處理、多媒體通信、多模態(tài)處理等。Rabiner教授于2002年從AT&T退休，隨后擔(dān)任Rutgers大學(xué)和加州大學(xué)圣巴巴拉分校的教授，以及Rutgers大學(xué)先進(jìn)信息處理中心副主任。

書籍目錄

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

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