Digital Signal Processing A Practical Approach

by Ifeachor, Emmanuel; Jervis, Barrie

• ISBN13: 9780201596199
• ISBN10: 0201596199
• Edition: 2nd
• Format: Paperback
• Copyright: 2001-09-27
• Publisher: Prentice Hall

Summary

This book clearly explains digital signal processing principles and shows how they can be used to build DSP systems. The aim is to give enough insight and practical guidance to enable an engineer to construct DSP systems. The book's programs are written in C, the language used in DSP.

Author Biography

Emmanuel Ifeachor is Professor of Intelligent Electronic Systems and Director of the Centre for Communications, Networks and Information Systems at the University of Plymouth, UK.

Barrie Jervis is Professor of Electronic Engineering at Sheffield Hallam University, UK.

This book evolved from the authors' extensive experience in teaching practically oriented courses in DSP to both undergraduates and engineers in industry. Their own research in applied DSP has influenced the contents of the book and provided many of the examples and case studies.

Table of Contents

Preface	xv
Introduction	1	(36)
Digital signal processing and its benefits	1	(2)
Application areas	3	(2)
Key DSP operations	5	(8)
Convolution	5	(2)
Correlation	7	(2)
Digital filtering	9	(2)
Discrete transformation	11	(1)
Modulation	11	(2)
Digital signal processors	13	(1)
Overview of real-world applications of DSP	13	(2)
Audio applications of DSP	15	(8)
Digital audio mixing	15	(1)
Speech synthesis and recognition	16	(3)
The compact disc digital audio system	19	(4)
Telecommunication applications of DSP	23	(6)
Digital cellular mobile telephony	23	(4)
Set-top box for digital television reception	27	(1)
Adaptive telephone echo cancellation	28	(1)
Biomedical applications of DSP	29	(6)
Fetal ECG monitoring	30	(3)
DSP-based closed loop controlled anaesthesia	33	(2)
Summary	35	(2)
Problems	35	(1)
References	35	(1)
Bibliography	36	(1)
Analog I/O interface for real-time DSP systems	37	(67)
Typical real-time DSP systems	38	(1)
Analog-to-digital conversion process	39	(1)
Sampling - lowpass and bandpass signals	40	(25)
Sampling lowpass signals	40	(16)
Sampling bandpass signals	56	(9)
Uniform and non-uniform quantization and encoding	65	(6)
Uniform quantization and encoding (linear pulse code modulation (PCM))	66	(2)
Non-uniform quantization and encoding (nonlinear PCM)	68	(3)
Oversampling in A/D conversion	71	(13)
Introduction	71	(1)
Oversampling and anti-aliasing filtering	71	(3)
Oversampling and ADC resolution	74	(4)
An application of oversampling - single-bit (oversampling) ADC	78	(6)
Digital-to-analog conversion process: signal recovery	84	(1)
The DAC	84	(2)
Anti-imaging filtering	86	(1)
Oversampling in D/A conversion	86	(4)
Oversampling D/A conversion in the CD player	87	(3)
Constraints of real-time signal processing with analog input/output signals	90	(1)
Application examples	91	(1)
Summary	92	(12)
Problems	92	(10)
References	102	(1)
Bibliography	102	(2)
Discrete transforms	104	(68)
Introduction	104	(7)
Fourier series	106	(3)
The Fourier transform	109	(2)
DFT and its inverse	111	(7)
Properties of the DFT	118	(2)
Computational complexity of the DFT	120	(1)
The decimation-in-time fast Fourier transform algorithm	121	(11)
The butterfly	127	(1)
Algorithmic development	128	(4)
Computational advantages of the FFT	132	(1)
Inverse fast Fourier transform	132	(1)
Implementation of the FFT	133	(2)
The decimation-in-frequency FFT	134	(1)
Comparison of DIT and DIF algorithms	134	(1)
Modifications for increased speed	134	(1)
Other discrete transforms	135	(16)
Discrete cosine transform	135	(1)
Walsh transform	136	(3)
Hadamard transform	139	(2)
Wavelet transform	141	(3)
Multiresolution analysis by the wavelet method	144	(3)
Signal representation by singularities: the wavelet transform method	147	(4)
An application of the DCT: image compression	151	(3)
The Discrete Cosine transform	152	(1)
2D DCT coefficient quantization	153	(1)
Coding	153	(1)
Worked examples	154	(18)
Problems	158	(2)
References	160	(1)
Appendices	161	(1)
C language program for direct DFT computation	161	(6)
C program for radix-2 decimation-in-time FFT	167	(3)
DFT and FFT with MATLAB	170	(1)
References for Appendices	171	(1)
The z-transform and its applications in signal processing	172	(70)
Discrete-time signals and systems	173	(1)
The z-transform	174	(5)
The inverse z-transform	179	(15)
Power series method	179	(3)
Partial fraction expansion method	182	(6)
Residue method	188	(6)
Comparison of the inverse z-transform methods	194	(1)
Properties of the z-transform	194	(3)
Some applications of the z-transform in signal processing	197	(21)
Pole-zero description of discrete-time systems	197	(3)
Frequency response estimation	200	(1)
Geometric evaluation of frequency response	201	(3)
Direct computer evaluation of frequency response	204	(1)
Frequency response estimation via FFT	205	(1)
Frequency units used in discrete-time systems	205	(3)
Stability considerations	208	(1)
Difference equations	209	(2)
Impulse response estimation	211	(2)
Applications in digital filter design	213	(1)
Realization structures for digital filters	213	(5)
Summary	218	(24)
Problems	218	(5)
References	223	(1)
Bibliography	223	(1)
Appendices	223	(1)
Recursive algorithm for the inverse z-transform	223	(2)
C program for evaluating the inverse z-transform and for cascade-to-parallel structure conversion	225	(6)
C program for estimating frequency response	231	(2)
z-transform operations with MATLAB	233	(8)
References for Appendices	241	(1)
Correlation and convolution	242	(75)
Introduction	242	(1)
Correlation description	243	(30)
Cross - and autocorrelation	249	(8)
Applications of correlation	257	(10)
Fast correlation	267	(6)
Convolution description	273	(28)
Properties of convolution	282	(1)
Circular convolution	283	(1)
System identification	283	(2)
Deconvolution	285	(1)
Blind deconvolution	286	(2)
Fast linear convolution	288	(1)
Computational advantages of fast linear convolution	289	(1)
Convolution and correlation by sectioning	290	(2)
Overlap-add method	292	(5)
Overlap-save method	297	(3)
Computational advantages of fast convolution by sectioning	300	(1)
The relationship between convolution and correlation	301	(1)
Implementation of correlation and convolution	301	(1)
Application examples	302	(8)
Correlation	302	(5)
Convolution	307	(3)
Summary	310	(7)
Problems	311	(4)
References	315	(1)
Appendix	316	(1)
C language program for computing cross- and autocorrelation	316	(1)
A framework for digital filter design	317	(25)
Introduction to digital filters	318	(1)
Types of digital filters: FIR and IIR filters	319	(2)
Choosing between FIR and IIR filters	321	(3)
Filter design steps	324	(10)
Specification of the filter requirements	324	(3)
Coefficient calculation	327	(1)
Representation of a filter by a suitable structure (realization)	328	(4)
Analysis of finite wordlength effects	332	(1)
Implementation of a filter	333	(1)
Illustrative examples	334	(5)
Summary	339	(3)
Problems	339	(2)
Reference	341	(1)
Bibliography	341	(1)
Finite impulse response (FIR) filter design	342	(112)
Introduction	343	(6)
Summary of key characteristic features of FIR filters	343	(1)
Linear phase response and its implications	344	(3)
Types of linear phase FIR filters	347	(2)
FIR filter design	349	(1)
FIR filter specifications	350	(1)
FIR coefficient calculation methods	351	(1)
Window method	352	(15)
Some common window functions	354	(4)
Summary of the window method of calculating FIR filter coefficients	358	(8)
Advantages and disadvantages of the window method	366	(1)
The optimal method	367	(13)
Basic concepts	367	(3)
Parameters required to use the optimal program	370	(1)
Relationships for estimating filter length, N	371	(1)
Summary of procedure for calculating filter coefficients by the optimal method	372	(1)
Illustrative examples	373	(7)
Frequency sampling method	380	(18)
Nonrecursive frequency sampling filters	380	(9)
Recursive frequency sampling filters	389	(1)
Frequency sampling filters with simple coefficients	390	(8)
Summary of the frequency sampling method	398	(1)
Comparison of the window, optimum and frequency sampling methods	398	(4)
Special FIR filter design topics	402	(5)
Half-band FIR filters	402	(2)
Frequency transformation	404	(2)
Computationally efficient FIR filters	406	(1)
Realization structures for FIR filters	407	(4)
Transversal structure	407	(1)
Linear phase structure	408	(2)
Other structures	410	(1)
Choosing between structures	410	(1)
Finite wordlength effects in FIR digital filters	411	(9)
Coefficient quantization errors	412	(7)
Roundoff errors	419	(1)
Overflow errors	419	(1)
FIR implementation techniques	420	(2)
Design example	422	(3)
Summary	425	(1)
Application examples of FIR filters	425	(29)
Problems	426	(9)
References	435	(1)
Bibliography	436	(1)
Appendices	437	(1)
C programs for FIR filter design	437	(3)
FIR filter design with MATLAB	440	(14)
Design of infinite impulse response (IIR) digital filters	454	(125)
Introduction: summary of the basic features of IIR filters	455	(1)
Design stages for digital IIR filters	456	(1)
Performance specification	457	(2)
Coefficient calculation methods for IIR filters	459	(1)
Pole-zero placement method of coefficient calculation	459	(4)
Basic concepts and illustrative design examples	459	(4)
Impulse invariant method of coefficient calculation	463	(5)
Basic concepts and illustrative design examples	463	(3)
Summary of the impulse invariant method	466	(1)
Remarks on the impulse invariant method	466	(2)
Matched z-transform (MZT) method of coefficient calculation	468	(3)
Basic concepts and illustrative design examples	468	(2)
Summary of the matched z-transform method	470	(1)
Remarks on the matched z-transform method	471	(1)
Bilinear z-transform (BZT) method of coefficient calculation	471	(11)
Basic concepts and illustrative design examples	471	(2)
Summary of the BZT method of coefficient calculation	473	(5)
Comments on the bilinear transformation method	478	(4)
Use of BZT and classical analog filters to design IIR filters	482	(18)
Characteristic features of classical analog filters	483	(2)
The BZT methodology using classical analog filters	485	(6)
Illustrative design examples (lowpass, highpass, bandpass and bandstop filters)	491	(9)
Calculating IIR filter coefficients by mapping s-plane poles and zeros	500	(8)
Basic concepts	500	(5)
Illustrative examples	505	(3)
Using IIR filter design programs	508	(1)
Choice of coefficient calculation methods for IIR filters	509	(8)
Nyquist effect	510	(7)
Realization structures for IIR digital filters	517	(7)
Practical building blocks for IIR filters	518	(2)
Cascade and parallel realization structures for higher-order IIR filters	520	(4)
Finite wordlength effects in IIR filters	524	(5)
Coefficient quantization errors	526	(3)
Implementation of IIR filters	529	(1)
A detailed design example of an IIR digital filter	530	(5)
Summary	535	(1)
Application examples in digital audio and instrumentation	536	(2)
Digital audio	536	(1)
Digital control	536	(1)
Digital frequency oscillators	536	(2)
Application examples in telecommunication	538	(41)
Touch-tone generation and reception for digital telephones	538	(2)
Digital telephony: dual tone multifrequency (DTMF) detection using the Goertzel algorithm	540	(6)
Clock recovery for data communication	546	(3)
Problems	549	(5)
References	554	(1)
Bibliography	555	(2)
Appendices	557	(1)
C programs for IIR digital filter design	557	(5)
IIR filter design with MATLAB	562	(15)
Evaluation of complex square roots using real arithmetic	577	(2)
Multirate digital signal processing	579	(66)
Introduction	579	(2)
Some current uses of multirate processing in industry	580	(1)
Concepts of multirate signal processing	581	(9)
Sampling rate reduction: decimation by integer factors	582	(2)
Sampling rate increase: interpolation by integer factors	584	(2)
Sampling rate conversion by non-integer factors	586	(3)
Multistage approach to sampling rate conversion	589	(1)
Design of practical sampling rate converters	590	(11)
Filter specification	590	(1)
Filter requirements for individual stages	591	(1)
Determining the number of stages and decimation factors	592	(2)
Illustrative design examples	594	(7)
Software implementation of sampling rate converters-decimators	601	(5)
Program for multistage decimation	602	(2)
Test example for the decimation program	604	(2)
Software implementation of interpolators	606	(6)
Program for multistage interpolation	610	(1)
Test example	610	(2)
Sample rate conversion using polyphase filter structure	612	(5)
Polyphase implementation of interpolators	612	(5)
Application examples	617	(15)
High quality analog-to-digital conversion for digital audio	618	(1)
Efficient digital-to-analog conversion in compact hi-fi systems	618	(2)
Application in the acquisition of high quality data	620	(6)
Multirate narrowband digital filtering	626	(5)
High resolution narrowband spectral analysis	631	(1)
Summary	632	(13)
Problems	633	(4)
References	637	(1)
Bibliography	638	(1)
Appendices	639	(1)
C programs for multirate processing and systems design	639	(1)
Multirate digital signal processing with MATLAB	640	(5)
Adaptive digital filters	645	(36)
When to use adaptive filters and where they have been used	646	(1)
Concepts of adaptive filtering	647	(4)
Adaptive filters as a noise canceller	647	(1)
Other configurations of the adaptive filter	648	(1)
Main components of the adaptive filter	648	(1)
Adaptive algorithms	648	(3)
Basic Wiener filter theory	651	(3)
The basic LMS adaptive algorithm	654	(8)
Implementation of the basic LMS algorithm	655	(3)
Practical limitations of the basic LMS algorithm	658	(3)
Other LMS-based algorithms	661	(1)
Recursive least squares algorithm	662	(4)
Recursive least squares algorithm	663	(1)
Limitations of the recursive least squares algorithm	664	(1)
Factorization algorithms	665	(1)
Application example 1 - adaptive filtering of ocular artefacts from the human EEG	666	(2)
The physiological problem	666	(1)
Artefact processing algorithm	667	(1)
Real-time implementation	668	(1)
Application example 2 - adaptive telephone echo cancellation	668	(2)
Other applications	670	(11)
Loudspeaking telephones	670	(1)
Multipath compensation	670	(1)
Adaptive jammer suppression	671	(1)
Radar signal processing	672	(1)
Separation of speech signals from background noise	672	(1)
Fetal monitoring - cancelling of maternal ECG during labour	673	(1)
Problems	674	(1)
References	674	(1)
Bibliography	675	(1)
Appendices	676	(1)
C language programs for adaptive filtering	676	(4)
MATLAB programs for adaptive filtering	680	(1)
Spectrum estimation and analysis	681	(46)
Introduction	682	(2)
Principles of spectrum estimation	684	(3)
Traditional methods	687	(20)
Pitfalls	687	(3)
Windowing	690	(13)
The periodogram method and periodogram properties	703	(1)
Modified periodogram methods	704	(1)
The Blackman-Tukey method	705	(1)
The fast correlation method	706	(1)
Comparison of the power spectral density estimation methods	706	(1)
Modern parametric estimation methods	707	(1)
Autoregressive spectrum estimation	708	(7)
Autoregressive model and filter	708	(1)
Power spectrum density of AR series	709	(1)
Computation of model parameters - Yule-Walker equations	710	(3)
Solution of the Yule-Walker equations	713	(1)
Model order	714	(1)
Comparison of estimation methods	715	(1)
Application examples	715	(6)
Use of spectral analysis by a DFT for differentiating between brain diseases	715	(4)
Spectral analysis of EEGs using autoregressive modelling	719	(2)
Summary	721	(1)
Worked example	721	(6)
Problems	722	(2)
References	724	(1)
Appendix	725	(1)
MATLAB programs for spectrum estimation and analysis	725	(2)
General- and special-purpose digital signal processors	727	(78)
Introduction	728	(1)
Computer architectures for signal processing	728	(18)
Harvard architecture	730	(2)
Pipelining	732	(5)
Hardware multiplier-accumulator	737	(1)
Special instructions	738	(3)
Replication	741	(1)
On-chip memory/cache	742	(1)
Extended parallelism - SIMD, VLIW and static superscalar processing	742	(4)
General-purpose digital signal processors	746	(13)
Fixed-point digital signal processors	747	(9)
Floating-point digital signal processors	756	(3)
Selecting digital signal processors	759	(2)
Implementation of DSP algorithms on general-purpose digital signal processors	761	(26)
FIR digital filtering	761	(9)
IIR digital filtering	770	(7)
FFT processing	777	(5)
Multirate processing	782	(4)
Adaptive filtering	786	(1)
Special-purpose DSP hardware	787	(5)
Hardware digital filters	789	(1)
Hardware FFT processors	790	(2)
Summary	792	(13)
Problems	793	(3)
References	796	(1)
Bibliography	797	(1)
Appendix	798	(1)
TMS320 assembly language programs for real-time signal processing and a C language program for constant geometry radix-2 FFT	798	(7)
Analysis of finite wordlength effects in fixed-point DSP systems	805	(68)
Introduction	805	(1)
DSP arithmetic	806	(9)
Fixed-point arithmetic	808	(4)
Floating-point arithmetic	812	(3)
ADC quantization noise and signal quality	815	(2)
Finite wordlength effects in IIR digital filters	817	(43)
Influence of filter structure on finite wordlength effects	818	(4)
Coefficient quantization errors in IIR digital filters	822	(1)
Coefficient wordlength requirements for stability and desired frequency response	823	(5)
Addition overflow errors and their effects	828	(1)
Principles of scaling	829	(3)
Scaling in cascade realization	832	(2)
Scaling in parallel realization	834	(1)
Output overflow detection and prevention	835	(1)
Product roundoff errors in IIR digital filters	836	(1)
Effects of roundoff errors on filter performance	837	(4)
Roundoff noise in cascade and parallel realizations	841	(4)
Effects of product roundoff noise in modern DSP systems	845	(1)
Roundoff noise reduction schemes	846	(7)
Determining practical values for error feedback coefficients	853	(4)
Limit cycles due to product roundoff errors	857	(2)
Other nonlinear phenomena	859	(1)
Finite wordlength effects in FFT algorithms	860	(4)
Roundoff errors in FFT	860	(2)
Overflow errors and scaling in FFT	862	(2)
Coefficient quantization in FFT	864	(1)
Summary	864	(9)
Problems	865	(3)
References	868	(1)
Bibliography	868	(2)
Appendices	870	(1)
Finite wordlength analysis program for IIR filters	870	(1)
L2 scaling factor equations	870	(3)
Applications and design studies	873	(52)
Evaluation boards for real-time signal processing	874	(3)
Background	874	(1)
TMS320C10 target board	874	(2)
DSP56002 evaluation module for real-time DSP	876	(1)
TMS320C54 and D5P56300 evaluation boards	876	(1)
DSP applications	877	(27)
Detection of fetal heartbeats during labour	877	(8)
Adaptive removal of ocular artefacts from human EEGs	885	(16)
Equalization of digital audio signals	901	(3)
Design studies	904	(7)
Computer-based multiple choice DSP questions	911	(9)
Summary	920	(5)
Problems	921	(1)
References	921	(2)
Bibliography	923	(1)
Appendix	923	(1)
The modified UD factorization algorithm	923	(2)
Index	925

Excerpts

Purpose of this book In the last five years, digital signal processing (DSP) has continued to have a major and increasing impact in many key areas of technology, including telecommunication, digital television and media, biomedicine, digital audio and instrumentation. DSP is now at the core of many new and emerging digital products and applications in the information society (e.g. digital cellular phones, digital cameras and TV, and digital audio systems). The need and expectations for electronic, computer and communication engineers to be competent in DSP have grown even stronger since the first edition. DSP is now a core subject in most electronic/computer/communication engineering curricula. This second edition of the book has been modernized by including additional topics of increasing importance, by providing MATLAB-based problems, and by offering a companion handbook and a home page on the web. These additions have been made in response to software developments, the wider availability of information technology, developments in the teaching of signal processing, and reader demand. Universities are increasingly making use of web-based materials and signal processing software tools such as MATLAB. We have consequently found a demand amongst our readers for MATLAB-based material. This high-level language permits sophisticated signal processing and the immediate display of results with relatively few commands. It is possible to have fun in developing signal processing and the solutions to problems without the distraction of having to produce detailed programs. We believe that the MATLAB examples and exercises in the book will enhance the learning experience of the student and increase the teaching resource available to the instructor. As in the first edition, the second edition aims to bridge the gap between theory and practice. Thus, we have retained the main features of the book, namely coverage of modern topics and the provision of practical examples and applications. As in the first edition, we have mixed practical examples and systems with theory, to keep students' interest and motivation high in order to enhance learning. Many of the chapters have been extensively revised, to bring the materials up to date and to improve clarity. End-of-chapter problems have been extended to test, reinforce and extend understanding. In revising the book, we have drawn from our experience and feedback from readers, worldwide, over the last eight years since the first edition was published. The new topics introduced include oversampling and bandpass sampling techniques in analog/digital conversions to exploit the advantages that DSP offers; wavelet transforms used for time-frequency representation and resolution of signals; blind signal deconvolution for identifying input signals from the output of an unknown system; parametric spectrum estimation for greater resolution applicable to shorter signals and with fewer pitfalls; architectures of new DSP processors and practical schemes for round-off noise reduction in fixed-point DSP systems; and computer-based multi-choice questions to aid revision. Throughout the book, MATI;AB-based examples and exercises are provided. This book was born out of our experience in teaching practically oriented courses in digital signal processing to undergraduate students at the University of Plymouth and the Sheffield Hallam University, and to application engineers in industry for many years. It appeared to us that many of the available textbooks were either too elementary or too theoretical to be of practical use for undergraduates or application engineers in industry. As most readers will know from experience, the gap between learning the fundamentals in any subject and actually applying them is quite wide. We therefore decided to write this book which we believe undergraduates will understand and appreciate and which will equip them to undertake practical digital si