Digital Signal Processing: A Practical Approach

by Ifeachor, Emmanuel; Jervis, Barrie

• ISBN13: 9780201544138
• ISBN10: 020154413X
• Edition: 2nd
• Format: Hardcover
• Copyright: 2002-01-01
• 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.

Table of Contents

Preface	p. xv
Introduction	p. 1
Digital signal processing and its benefits	p. 1
Application areas	p. 3
Key DSP operations	p. 5
Digital signal processors	p. 13
Overview of real-world applications of DSP	p. 13
Audio applications of DSP	p. 15
Telecommunication applications of DSP	p. 23
Biomedical applications of DSP	p. 29
Summary	p. 35
Problems	p. 35
References	p. 35
Bibliography	p. 36
Analog I/O interface for real-time DSP systems	p. 37
Typical real-time DSP systems	p. 38
Analog-to-digital conversion process	p. 39
Sampling - lowpass and bandpass signals	p. 40
Uniform and non-uniform quantization and encoding	p. 65
Oversampling in A/D conversion	p. 71
Digital-to-analog conversion process: signal recovery	p. 84
The DAC	p. 84
Anti-imaging filtering	p. 86
Oversampling in D/A conversion	p. 86
Constraints of real-time signal processing with analog input/output signals	p. 90
Application examples	p. 91
Summary	p. 92
Problems	p. 92
References	p. 102
Bibliography	p. 102
Discrete transforms	p. 104
Introduction	p. 104
DFT and its inverse	p. 111
Properties of the DFT	p. 118
Computational complexity of the DFT	p. 120
The decimation-in-time fast Fourier transform algorithm	p. 121
Inverse fast Fourier transform	p. 132
Implementation of the FFT	p. 133
Other discrete transforms	p. 135
An application of the DCT: image compression	p. 151
Worked examples	p. 154
Problems	p. 158
References	p. 160
Appendices	p. 161
C language program for direct DFT computation	p. 161
C program for radix-2 decimation-in-time FFT	p. 167
DFT and FFT with MATLAB	p. 170
References for Appendices	p. 171
The z-transform and its applications in signal processing	p. 172
Discrete-time signals and systems	p. 173
The z-transform	p. 174
The inverse z-transform	p. 179
Properties of the z-transform	p. 194
Some applications of the z-transform in signal processing	p. 197
Summary	p. 218
Problems	p. 218
References	p. 223
Bibliography	p. 223
Appendices	p. 223
Recursive algorithm for the inverse z-transform	p. 223
C program for evaluating the inverse z-transform and for cascade-to-parallel structure conversion	p. 225
C program for estimating frequency response	p. 231
z-transform operations with MATLAB	p. 233
References for Appendices	p. 241
Correlation and convolution	p. 242
Introduction	p. 242
Correlation description	p. 243
Convolution description	p. 273
Implementation of correlation and convolution	p. 301
Application examples	p. 302
Summary	p. 310
Problems	p. 311
References	p. 315
Appendix	p. 316
C language program for computing cross- and autocorrelation	p. 316
A framework for digital filter design	p. 317
Introduction to digital filters	p. 318
Types of digital filters: FIR and IIR filters	p. 319
Choosing between FIR and IIR filters	p. 321
Filter design steps	p. 324
Illustrative examples	p. 334
Summary	p. 339
Problems	p. 339
Reference	p. 341
Bibliography	p. 341
Finite impulse response (FIR) filter design	p. 342
Introduction	p. 343
FIR filter design	p. 349
FIR filter specifications	p. 350
FIR coefficient calculation methods	p. 351
Window method	p. 352
The optimal method	p. 367
Frequency sampling method	p. 380
Comparison of the window, optimum and frequency sampling methods	p. 398
Special FIR filter design topics	p. 402
Realization structures for FIR filters	p. 407
Finite wordlength effects in FIR digital filters	p. 411
FIR implementation techniques	p. 420
Design example	p. 422
Summary	p. 425
Application examples of FIR filters	p. 425
Problems	p. 426
References	p. 435
Bibliography	p. 436
Appendices	p. 437
C programs for FIR filter design	p. 437
FIR filter design with MATLAB	p. 440
Design of infinite impulse response (IIR) digital filters	p. 454
Introduction: summary of the basic features of IIR filters	p. 455
Design stages for digital IIR filters	p. 456
Performance specification	p. 457
Coefficient calculation methods for IIR filters	p. 459
Pole-zero placement method of coefficient calculation	p. 459
Impulse invariant method of coefficient calculation	p. 463
Matched z-transform (MZT) method of coefficient calculation	p. 468
Bilinear z-transform (BZT) method of coefficient calculation	p. 471
Use of BZT and classical analog filters to design IIR filters	p. 482
Calculating IIR filter coefficients by mapping s-plane poles and zeros	p. 500
Using IIR filter design programs	p. 508
Choice of coefficient calculation methods for IIR filters	p. 509
Realization structures for IIR digital filters	p. 517
Finite wordlength effects in IIR filters	p. 524
Implementation of IIR filters	p. 529
A detailed design example of an IIR digital filter	p. 530
Summary	p. 535
Application examples in digital audio and instrumentation	p. 536
Application examples in telecommunication	p. 538
Problems	p. 549
References	p. 554
Bibliography	p. 555
Appendices	p. 557
C programs for IIR digital filter design	p. 557
IIR filter design with MATLAB	p. 562
Evaluation of complex square roots using real arithmetic	p. 577
Multirate digital signal processing	p. 579
Introduction	p. 579
Concepts of multirate signal processing	p. 581
Design of practical sampling rate converters	p. 590
Software implementation of sampling rate converters--decimators	p. 601
Software implementation of interpolators	p. 606
Sample rate conversion using polyphase filter structure	p. 612
Application examples	p. 617
Summary	p. 632
Problems	p. 633
References	p. 637
Bibliography	p. 638
Appendices	p. 639
C programs for multirate processing and systems design	p. 639
Multirate digital signal processing with MATLAB	p. 640
Adaptive digital filters	p. 645
When to use adaptive filters and where they have been used	p. 646
Concepts of adaptive filtering	p. 647
Basic Wiener filter theory	p. 651
The basic LMS adaptive algorithm	p. 654
Recursive least squares algorithm	p. 662
Application example 1 - adaptive filtering of ocular artefacts from the human EEG	p. 666
Application example 2 - adaptive telephone echo cancellation	p. 668
Other applications	p. 670
Problems	p. 674
References	p. 674
Bibliography	p. 675
Appendices	p. 676
C language programs for adaptive filtering	p. 676
MATLAB programs for adaptive filtering	p. 680
Spectrum estimation and analysis	p. 681
Introduction	p. 682
Principles of spectrum estimation	p. 684
Traditional methods	p. 687
Modern parametric estimation methods	p. 707
Autoregressive spectrum estimation	p. 708
Comparison of estimation methods	p. 715
Application examples	p. 715
Summary	p. 721
Worked example	p. 721
Problems	p. 722
References	p. 724
Appendix	p. 725
MATLAB programs for spectrum estimation and analysis	p. 725
General- and special-purpose digital signal processors	p. 727
Introduction	p. 728
Computer architectures for signal processing	p. 728
General-purpose digital signal processors	p. 746
Selecting digital signal processors	p. 759
Implementation of DSP algorithms on general-purpose digital signal processors	p. 761
Special-purpose DSP hardware	p. 787
Summary	p. 792
Problems	p. 793
References	p. 796
Bibliography	p. 797
Appendix	p. 798
TMS320 assembly language programs for real-time signal processing and a C language program for constant geometry radix-2 FFT	p. 798
Analysis of finite wordlength effects in fixed-point DSP systems	p. 805
Introduction	p. 805
DSP arithmetic	p. 806
ADC quantization noise and signal quality	p. 815
Finite wordlength effects in IIR digital filters	p. 817
Finite wordlength effects in FFT algorithms	p. 860
Summary	p. 864
Problems	p. 865
References	p. 868
Bibliography	p. 868
Appendices	p. 870
Finite wordlength analysis program for IIR filters	p. 870
L[subscript 2] scaling factor equations	p. 870
Applications and design studies	p. 873
Evaluation boards for real-time signal processing	p. 874
DSP applications	p. 877
Design studies	p. 904
Computer-based multiple choice DSP questions	p. 911
Summary	p. 920
Problems	p. 921
References	p. 921
Bibliography	p. 923
Appendix	p. 923
The modified UD factorization algorithm	p. 923
Index	p. 925
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