數(shù)字信號(hào)處理實(shí)踐方法

前言

　　2001年7月間，電子工業(yè)出版社的領(lǐng)導(dǎo)同志邀請(qǐng)各高校十幾位通信領(lǐng)域方面的老師，商量引進(jìn)國(guó)外教材問(wèn)題。與會(huì)同志對(duì)出版社提出的計(jì)劃十分贊同，大家認(rèn)為，這對(duì)我國(guó)通信事業(yè)、特別是對(duì)高等院校通信學(xué)科的教學(xué)工作會(huì)很有好處?！　〗滩慕ㄔO(shè)是高校教學(xué)建設(shè)的主要內(nèi)容之一。編寫、出版一本好的教材，意味著開(kāi)設(shè)了一門好的課程，甚至可能預(yù)示著一個(gè)嶄新學(xué)科的誕生。20世紀(jì)40年代MIT林肯實(shí)驗(yàn)室出版的一套28本雷達(dá)叢書(shū)，對(duì)近代電子學(xué)科、特別是對(duì)雷達(dá)技術(shù)的推動(dòng)作用，就是一個(gè)很好的例子?！　∥覈?guó)領(lǐng)導(dǎo)部門對(duì)教材建設(shè)一直非常重視。20世紀(jì)80年代，在原教委教材編審委員會(huì)的領(lǐng)導(dǎo)下，匯集了高等院校幾百位富有教學(xué)經(jīng)驗(yàn)的專家，編寫、出版了一大批教材；很多院校還根據(jù)學(xué)校的特點(diǎn)和需要，陸續(xù)編寫了大量的講義和參考書(shū)。這些教材對(duì)高校的教學(xué)工作發(fā)揮了極好的作用。近年來(lái)，隨著教學(xué)改革不斷深入和科學(xué)技術(shù)的飛速進(jìn)步，有的教材內(nèi)容已比較陳舊、落后，難以適應(yīng)教學(xué)的要求，特別是在電子學(xué)和通信技術(shù)發(fā)展神速、可以講是日新月異的今天，如何適應(yīng)這種情況，更是一個(gè)必須認(rèn)真考慮的問(wèn)題。解決這個(gè)問(wèn)題，除了依靠高校的老師和專家撰寫新的符合要求的教科書(shū)外，引進(jìn)和出版一些國(guó)外優(yōu)秀電子與通信教材，尤其是有選擇地引進(jìn)一批英文原版教材，是會(huì)有好處的?！　∫荒甓鄟?lái)，電子工業(yè)出版社為此做了很多工作。他們成立了一個(gè)“國(guó)外電子與通信教材系列”項(xiàng)目組，選派了富有經(jīng)驗(yàn)的業(yè)務(wù)骨干負(fù)責(zé)有關(guān)工作，收集了230余種通信教材和參考書(shū)的詳細(xì)資料，調(diào)來(lái)了100余種原版教材樣書(shū)，依靠由20余位專家組成的出版委員會(huì)，從中精選了40多種，內(nèi)容豐富，覆蓋了電路理論與應(yīng)用、信號(hào)與系統(tǒng)、數(shù)字信號(hào)處理、微電子、通信系統(tǒng)、電磁場(chǎng)與微波等方面，既可作為通信專業(yè)本科生和研究生的教學(xué)用書(shū)，也可作為有關(guān)專業(yè)人員的參考材料。此外，這批教材，有的翻譯為中文，還有部分教材直接影印出版，以供教師用英語(yǔ)直接授課。希望這些教材的引進(jìn)和出版對(duì)高校通信教學(xué)和教材改革能起一定作用?！　≡谶@里，我還要感謝參加工作的各位教授、專家、老師與參加翻譯、編輯和出版的同志們。各位專家認(rèn)真負(fù)責(zé)、嚴(yán)謹(jǐn)細(xì)致、不辭辛勞、不怕瑣碎和精益求精的態(tài)度，充分體現(xiàn)了中國(guó)教育工作者和出版工作者的良好美德?！　‰S著我國(guó)經(jīng)濟(jì)建設(shè)的發(fā)展和科學(xué)技術(shù)的不斷進(jìn)步，對(duì)高校教學(xué)工作會(huì)不斷提出新的要求和希望。我想，無(wú)論如何，要做好引進(jìn)國(guó)外教材的工作，一定要聯(lián)系我國(guó)的實(shí)際。教材和學(xué)術(shù)專著不同，既要注意科學(xué)性、學(xué)術(shù)性，也要重視可讀性，要深入淺出，便于讀者自學(xué)；引進(jìn)的教材要適應(yīng)高校教學(xué)改革的需要，針對(duì)目前一些教材內(nèi)容較為陳舊的問(wèn)題，有目的地引進(jìn)一些先進(jìn)的和正在發(fā)展中的交叉學(xué)科的參考書(shū)；要與國(guó)內(nèi)出版的教材相配套，安排好出版英文原版教材和翻譯教材的比例。我們努力使這套教材能盡量滿足上述要求，希望它們能放在學(xué)生們的課桌上，發(fā)揮一定的作用?！　∽詈螅A(yù)?！皣?guó)外電子與通信教材系列”項(xiàng)目取得成功，為我國(guó)電子與通信教學(xué)和通信產(chǎn)業(yè)的發(fā)展培土施肥。也懇切希望讀者能對(duì)這些書(shū)籍的不足之處、特別是翻譯中存在的問(wèn)題，提出意見(jiàn)和建議，以便再版時(shí)更正。

內(nèi)容概要

　　《數(shù)字信號(hào)處理實(shí)踐方法（第二版）》根據(jù)實(shí)際工程應(yīng)用和具體實(shí)例，詳細(xì)介紹了數(shù)字信號(hào)處理（DSP）領(lǐng)域內(nèi)的基本概念和相關(guān)技術(shù)。全書(shū)共分為14章，首先講解了DSP的基本概念及其應(yīng)用，并從實(shí)際的例子出發(fā)，闡述了DSP的一些基本內(nèi)容，如信號(hào)的抽樣、量化及其在實(shí)時(shí)DSP上的內(nèi)涵。然后，作者介紹了離散變換（DFT和FFT），離散時(shí)間信號(hào)與系統(tǒng)分析的工具（z變換），以及DSP的基本運(yùn)算（相關(guān)和卷積），并分析了數(shù)字濾波器設(shè)計(jì)的實(shí)際問(wèn)題。《數(shù)字信號(hào)處理實(shí)踐方法（第二版）》還介紹了多抽樣率數(shù)字信號(hào)處理、自適應(yīng)數(shù)字濾波器、譜估計(jì)及其分析等現(xiàn)代數(shù)字信號(hào)處理理論，最后討論了通用和專用數(shù)字信號(hào)處理器、定點(diǎn)DSP系統(tǒng)有限字長(zhǎng)效應(yīng)分析及DSP的應(yīng)用和設(shè)計(jì)實(shí)例。另外，書(shū)中還提供了有關(guān)范例和實(shí)驗(yàn)的MATLAB實(shí)現(xiàn)方法?！　　稊?shù)字信號(hào)處理實(shí)踐方法（第二版）》可作為通信與電子信息類專業(yè)高年級(jí)本科生和研究生的教材或教學(xué)參考書(shū)，而且對(duì)于相關(guān)學(xué)科的工程技術(shù)人員也具有很好的參考價(jià)值。

作者簡(jiǎn)介

　　Emmanuel C.Ifeachor：智能電子系統(tǒng)方向的教授，英國(guó)普利茅斯大學(xué)通信、網(wǎng)絡(luò)和信息系統(tǒng)研究中心主任?！　arriv W.Jervis：英國(guó)Sheffield Hallam大學(xué)電子工程系教授。

書(shū)籍目錄

1 introduction1.1 digital signal processing and its benefts1.2 application areas1.3 key dsp operations1.3.1 convolution1.3.2 correlation1.3.3 digital fitering1.3.4 discrete transformation1.3.5 modulation1.4 digital signal processors1.5 overview of real-world applications of dsp1.6 audio applications of dsp1.6.1 digital audio mixing1.6.2 speech synthesis and recognition1.6.3 the compact disc digital audio system1.7 telecommunication applications of dsp1.7.1 digital cellular mobile telephony1.7.2.set-top box for digital television reception1.7.3 adaptive telephone echo cancellation1.8 biomedical applications of dsp1.8.1 fetal ecg monitoring1.8.2 dsp-based closed loop controlled anaesthesia1.9 summaryproblemsreferencesbibliography2 analog i/o interface for real-time dsp systems2.1 typical real-time dsp systems2.2 analog-to-digital conversion process2.3 sampling- iowpass and bandpass signals2.3.1 sampling iowpass signals2.3.2 sampling bandpass signals2.4 uniform and non-uniform quantization and encoding2.4.1 uniform quantization and encoding (linear pulse code modulation (pcm))2.4.2 non-uniform quantization and encoding (nonlinear pcm)2.5 oversampling in aid conversion2.5.1 introduction2.5.20versampling and anti-aliasing fltering2.5.30versampling and adc resolution2.5.4 an application of oversampling - single-bit (oversampling) adc2.6 digital-to-analog conversion process: signal recovery2.7 the dac2.8 anti-imaging fltering2.9 oversampling in d/a conversion2.9.10versampling d/a conversion in the cd player2.10 constraints of real-time signal processing with analog input/output signals2.11 application examples2.12 summaryproblemsreferencesbibliography3 discrete transforms3.1 introduction3.1.1 fourier series3.1.2 the fourier transform3.2 dft and its inverse3.3 properties of the dft3.4 computational complexity of the dft3.5 the decimation-in-time fast fourier transform algorithm3.5.1 the butterfly3.5.2 algorithmic development3.5.3 computational advantages of the fft3.6 inverse fast fourier transform3.7 implementation of the fft3.7.1 the decimation-in-frequency fft3.7.2 comparison of dit and dif algorithms3.7.3 modifications for increased speed3.8 other discrete transforms3.8.1 discrete cosine transform3.8.2 walsh transform3.8.3 hadamard transform3.8.4 wavelet transform3.8.5 multiresolution analysis by the wavelet method3.8.6 signal representation by singularities: the wavelet transform method3.9 an application of the dct: image compression3.9.1 the discrete cosine transform3.9.2 2d dct coeffi:ient quantization3.9.3 coding3.10 worked examplesproblemsreferencesappendices3a c language program for direct dft computation3b c program for radix-2 decimation-in-time fft3c dft and fft with matlab references for appendices4 the z-transform and its applications in signal processing4.1 discrete-time signals and systems4.2 the z-transform4.3 the inverse z-transform4.3.1 power series method4.3.2 partial fraction expansion method4.3.3 residue method4.3.4 comparison of the inverse z-transform methods4.4 properties of the z-transform4.5 some applications of the z-transform in signal processing4.5.1 pole-zero description of discrete-time systems4.5.2 frequency response estimation4.5.3 geometric evaluation of frequency response4.5.4 direct computer evaluation of frequency response4.5.5 frequency response estimation via fft4.5.6 frequency units used in discrete-time systems4.5.7 stability considerations4.5.8 difference equations4.5.9 impulse response estimation4.5.10 applications in digital fiter design4.5.11 realization structures for digital fiters4.6 summaryproblemsreferencesbibliographyappendices4a recursive algorithm for the inverse z-transform4b c program for evaluating the inverse z-transform and for cascade-to-parallel structure conversion4c c program for estimating frequency response4d z-transform operations with matlab references for appendices5 correlation and convolution5.1 introduction5.2 correlation description5.2.1 cross- and autocorrelation5.2.2 applications of correlation5.2.3 fast correlation5.3 convolution description5.3.1 properties of convolution5.3.2 circular convolution5.3.3 system identification5.3.4 deconvolution5.3.5 blind deconvolution5.3.6 fast linear convolution5.3.7 computational advantages of fast linear convolution5.3.8 convolution and correlation by sectioning5.3.9 overlap-add method5.3.10 overlap-save method5.3.11 computational advantages of fast convolution by sectioning5.3.12 the relationship between convolution and correlation5.4 implementation of correlation and convolution5.5 application examples5.5.1 correlation5.5.2 convolution5.6 summaryproblemsreferencesappendix5a c language program for computing cross- and autocorrelation6 a framework for digital fiter design6.1 introduction to digital fiters6.2 types of digital fiters: fir and iir fiters6.3 choosing between fir and iir fiters6.4 filter design steps6.4.1 specifcation of the fiter requirements6.4.2 coefficient calculation6.4.3 representation of a fiter by a suitable structure (realization)6.4.4 analysis of fhite wordlength effects6.4.5 implementation of a fiter6.5 illustrative examples6.6 summaryproblemsreferencebibliography7 finite impulse response (fir) fiter design7.1 introduction7.1.1 summary of key characteristic features of fir filters7.1.2 linear phase response and its implications7.1.3 types of linear phase fir flters7.2 fir fiter design7.3 fir fiter specif'cations7.4 fir coefficient calculation methods7.5 window method7.5.1 some common window functions7.5.2 summary of the window method of calculating fir flter coeffi:ients7.5.3 advantages and disadvantages of the window method7.6 the optimal method7.6.1 basic concepts7.6.2 parameters required to use the optimal program7.6.3 relationships for estimating fiter length, n7.6.4 summary of procedure for calculating flter coeffi:ients by the optimal method7.6.5 illustrative examples7.7 frequency sampling method7.7.1 nonrecursive frequency sampling flters7.7.2 recursive frequency sampling flters7.7.3 frequency sampling flters with simple coeffi:ients7.7.4 summary of the frequency sampling method7.8 comparison of the window, optimum and frequency sampling methods7.9 special fir fiter design topics7.9.1 half-band fir fiters7.9.2 frequency transformation7.9.3 computationally efficient fir flters7.10 realization structures for fir fiters7.10.1 transversal structure7.10.2 linear phase structure7.10.3 other structures7.10.4 choosing between structures7.11 finite wordlength effects in fir digital fiters7.11.1 coefficient quantization errors7.11.2 roundoff errors7.11.3 overfbw errors7.12 fir implementation techniques7.13 design example7.14 summary7.15 application examples of fir fltersproblemsreferencesbibliographyappendices7a c programs for fir flter design7b fir fiter design with matlab8 design of infhite impulse response (iir) digital flters8.1 introduction: summary of the basic features of iir fiters8.2 design stages for digital iir fiters8.3 performance specification8.4 coeff'cient calculation methods for iir fiters8.5 pole-zero placement method of coeffcient calculation8.5.1 basic concepts and illustrative design examples8.6 impulse invariant method of coefficient calculation8.6.1 basic concepts and illustrative design examples8.6.2 summary of the impulse invariant method8.6.3 remarks on the impulse invariant method8.7 matched z-transform (mzt) method of coeffcient calculation8.7.1 basic concepts and illustrative design examples8.7.2 summary of the matched z-transform method8.7.3 remarks on the matched z-transform method8.8 bilinear z-transform (bzt) method of coeffcient calculation8.8.1 basic concepts and illustrative design examples8.8.2 summary of the bzt method of coeff'cient calculation8.8.3 comments on the bilinear transformation method8.9 use of bzt and classical analog fiters to design iir fiters8.9.1 characteristic features of classical analog flters8.9.2 the bzt methodology using classical analog fiters8.9.3 illustrative design examples (iowpass, highpass, bandpass and bandstop fiters)8.10 calculating iir fiter coefircients by mapping s-plane poles and zeros8.10.1 basic concepts8.10.2 illustrative examples8.11 using iir flter design programs8.12 choice of coeffcient calculation methods for iir flters8.12.1 nyquist effect8.13 realization structures for iir digital fiters8.13.1 practical building blocks for iir fiters8.13.2 cascade and parallel realization structures for higher-order iir fiters8.14 finite wordlength effects in iir fiters8.14.1 coefficient quantization errors8.15 implementation of iir fiters8.16 a detailed design example of an iir digital flter8.17 summary8.18 application examples in digital audio and instrumentation8.18.1 digital audio8.18.2 digital control8.18.3 digital frequency oscillators8.19 application examples in telecommunication8.19.1 touch-tone generation and reception for digital telephones8.19.2 digital telephony: dual tone multifrequency (dtmf) detection using the goertzel algorithm8.19.3 clock recovery for data communicationproblemsreferencesbibliographyappendices8a c programs for iir digital fiter design8b iir flter design with matlab8c evaluation of complex square roots using real arithmetic9 multirate digital signal processing9.1 introduction9.1.1 some current uses of multirate processing in industry9.2 concepts of multirate signal processing9.2.1 sampling rate reduction: decimation by integer factors9.2.2 sampling rate increase: interpolation by integer factors9.2.3 sampling rate conversion by non-integer factors9.2.4 multistage approach to sampling rate conversion9.3 design of practical sampling rate converters9.3.1 filter specircation9.3.2 filter requirements for individual stages9.3.3 determining the number of stages and decimation factors9.3.4 illustrative design examples9.4 software implementation of sampling rate converters-decimators9.4.1 program for multistage decimation9.4.2 test example for the decimation program9.5 software implementation of interpolators9.5.1 program for multistage interpolation9.5.2 test example9.6 sample rate conversion using polyphase flter structure9.6.1 polyphase implementation of interpolators9.7 application examples9.7.1 high quality analog-to-digital conversion for digital audio9.7.2 effcient digital-to-analog conversion in compact hi-fisystems9.7.3 application in the acquisition of high quality data9.7.4 multirate narrowband digital fitering9.7.5 high resolution narrowband spectral analysis9.8 summaryproblemsreferencesbibliographyappendices9a c programs for multirate processing and systems design9b multirate digital signal processing with matlab10 adaptive digital fiters10.1 when to use adaptive fiters and where they have been used10.2 concepts of adaptive fitering10.2.1 adaptive fiters as a noise canceller10.2.2 other configurations of the adaptive flter10.2.3 main components of the adaptive fiter10.2.4 adaptive algorithms10.3 basic wiener fiter theory10.4 the basic lms adaptive algorithm10.4.1 implementation of the basic lms algorithm10.4.2 practical limitations of the basic lms algorithm10.4.3 other lms-based algorithms10.5 recursive least squares algorithm10.5.1 recursive least squares algorithm10.5.2 limitations of the recursive least squares algorithm10.5.3 factorization algorithms10.6 application example 1 - adaptive fltering of ocular artefacts from the human eeg10.6.1 the physiological problem10.6.2 artefact processing algorithm10.6.3 real-time implementation10.7 application example 2 - adaptive telephone echo cancellation10.8 other applications10.8.1 loudspeaking telephones10.8.2 multipath compensation10.8.3 adaptive jammer suppression10.8.4 radar signal processing10.8.5 separation of speech signals from background noise10.8.6 fetal monitoring - cancelling of matemal ecg during labourproblemsreferencesbibliographyappendices10a c language programs for adaptive fltering10b matlab programs for adaptive fitering11 spectrum estimation and analysis11.1 introduction11.2 principles of spectrum estimation11.3 traditional methods11.3.1 pitfalls11.3.2 windowing11.3.3 the periodogram method and periodogram properties11.3.4 modified periodogram methods11.3.5 the blackman-tukey method11.3.6 the fast correlation method11.3.7 comparison of the power spectral density estimation methods11.4 modern parametric estimation methods11.5 autoregressive spectrum estimation11.5.1 autoregressive model and flter11.5.2 power spectrum density of ar series11.5.3 computation of model parameters - yule-walker equations11.5.4 solution of the yule-walker equations11.5.5 model order11.6 comparison of estimation methods11.7 application examples11.7.1 use of spectral analysis by a dft for differentiating between brain diseases11.7.2 spectral analysis of eegs using autoregressive modelling11.8 summary11.9 worked exampleproblemsreferencesappendix11a matlab programs for spectrum estimation and analysis12 general- and special-purpose digital signal processors12.1 introduction12.2 computer architectures for signal processing12.2.1 harvard architecture12.2.2 pipelining12.2.3 hardware multiplier-accumulator12.2.4 special instructions12.2.5 replication12.2.6 on-chip memory/cache12.2.7 extended parallelism - simd, vliw and static superscalar processing12.3 general-purpose digital signal processors12.3.1 fixed-point digital signal processors12.3.2 floating-point digital signal processors12.4 selecting digital signal processors12.5 implementation of dsp algorithms on general-purpose digital signal processors12.5.1 fir digital fltering12.5.2 iir digital fltering12.5.3 fft processing12.5.4 multirate processing12.5.5 adaptive fltering12.6 special-purpose dsp hardware12.6.1 hardware digital fiters12.6.2 hardware fft processors12.7 summaryproblemsreferencesbibliographyappendix12a tms320 assembly language programs for real-time signal processing and a c language program for constant geometry radix-2 fft13 analysis of fhite wordlength effects in fixed-point dsp systems13.1 introduction13.2 dsp arithmetic13.2.1 fixed-point arithmetic13.2.2 floating-point arithmetic13.3 adc quantization noise and signal quality13.4 finite wordlength effects in iir digital flters13.4.1 infljence of fiter structure on fnite wordlensth effects13.4.2 coeffcient quantization errors in iir digital flters13.4.3 coeffcient wordlength requirements for stability and desired frequency response13.4.4 addition overfbw errors and their effects13.4.5 principles of scaling13.4.6 scaling in cascade realization13.4.7 scaling in parallel realization13.4.8 output overflow detection and prevention15.4.9 product roundoff errors in iir digital flters13.4.10 effects of roundoff errors on flter performance13,4.11 roundoff noise in cascade and parallel realizations13.4,12 effects of product roundoff noise in modern dsp systems13.4.13 rouncloff noise reduction schemes13.4,14 determining practical values for error feedback coefficients13.4.15 limit cycles clue to product roundoff errors13.4.16 other nonlinear phenomena13.5 finite wordlength effects in fft algorithms13.5.1 roundoff errors in fft13.5.2 overfbw errors and scaling in fft13.5.3 coeffi:ient quantization in fft13.6 summaryproblemsreferencesbibliographyappendices13a finite wordlength analysis program for iir flters13b l2 scaling factor equations14 applications and design studies14.1 evaluation boards for real-time signal processing14.1.] backsround14.1.2 tms320c 10 target board14.1.3 dsp56002 evaluation module for real-time dsp14.1.4 tms320c54 and dsp56300 evaluation boards14.2 dsp applications14.2.1 detection of fetal heartbeats during labour14.2.2 adaptive removal of ocular artefacts from human ergs14.2.3 equalization of digital audio signals14.3 design studies14.4 computer-based multiple choice dsp questions14.5 summaryproblemsreferencesbibliographyappendix14a the modified ud factorization algorithmindex

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