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数字语音处理理论与应用(英文版)

数字语音处理理论与应用(英文版) 

出版时间:2011年版 

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  

 9.7 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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