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內容簡介

《國外電子與通信教材係列：數字語音處理理論與應用（英文版）》是作者繼1978年版經典教材digital processing of speech signals之後的又一著作，《國外電子與通信教材係列：數字語音處理理論與應用（英文版）》除有簡練精闢的基礎知識介紹外，係統介紹瞭近30年來語音信號處理的新理論、新方法和在應用上的新進展。《國外電子與通信教材係列：數字語音處理理論與應用（英文版）》共14章，分四部分：一部分介紹語音信號處理基礎知識，主要包括數字信號處理基礎、語音産生機理、（人的）聽覺和聽感知機理和聲道中的聲傳播原理；第二部分介紹語音信號的時頻域錶示和分析；第三部分介紹語音參數估計算法；第四部分介紹語音信號處理的應用，主要包括語音編碼、語音和音頻信號的頻域編、語音閤成、語音識彆和自然語言理解。
《國外電子與通信教材係列：數字語音處理理論與應用（英文版）》可供高等院校通信、電子、信息、計算機等專業作為研究生和本科生教材，也可以供有關科研和工程技術人員參考，是一本既有係統的基礎理論講解、又有新研究前沿介紹並密切結閤應用發展的教材。

作者簡介

Lawrence R．Rabiner，美國工程院和美國科學院院士，美國聲學學會、IEEE、Bell實驗室、AT&T;會士，以及Eta Kappa Nu、Sigma Xi、Tau Beta Pi等榮譽學會會員。曾擔任美國聲學學會副主席、IEEE Trans．ASSP主編和IEEE Proceedings編委會成員。其主要研究方嚮包括：通信、控製與信號處理、數字信號處理、數字語音處理、多媒體通信、多模態處理等。Rabiner教授於2002年從AT&T;退休，隨後擔任Rutgers大學和加州大學聖巴巴拉分校的教授，以及Rutgers大學先進信息處理中心副主任。

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

前言/序言

國外電子與通信教材係列：數字語音處理理論與應用（英文版） 下載 mobi epub pdf txt 電子書

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1.1 The SpeechSignal

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書中對語音參數方麵說的很詳細

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作者：(美國)勞倫斯 R.拉比納(Lawrence R. Rabiner) (美國)羅納德 W.謝弗(Ronald W. Schafer)

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英文版

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大師的著作，一切評價都是蒼白的

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語音領域大牛的權威著作。

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朋友推薦的，剛到手，還沒看，看完再評！！！！！！！！！

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1.5 Summary

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Problems

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