Digital Signal Processing Using MATLAB

✍ Scribed by Vinay K. Ingle, John G. Proakis

• Publisher: Cengage Learning
• Year: 2011
• Tongue: English
• Leaves: 671
• Edition: 3
• Category: Library

✦ Synopsis

In this supplementary text, MATLAB is used as a computing tool to explore traditional DSP topics and solve problems to gain insight. This greatly expands the range and complexity of problems that students can effectively study in the course. Since DSP applications are primarily algorithms implemented on a DSP processor or software, a fair amount of programming is required. Using interactive software such as MATLAB makes it possible to place more emphasis on learning new and difficult concepts than on programming algorithms. Interesting practical examples are discussed and useful problems are explored.

✦ Table of Contents

Cover Page......Page 1
Half-Title Page......Page 2
Title Page......Page 4
Copyright Page......Page 6
Contents......Page 8
Preface......Page 14
ORGANIZATION OF THE BOOK......Page 15
ABOUT THE SOFTWARE......Page 17
ACKNOWLEDGMENTS......Page 18
CHAPTER 1: INTRODUCTION......Page 20
1.1: Overview of Digital Signal Processing......Page 21
1.2: A Brief Introduction to MATLAB......Page 24
1.3: Applications of Digital Signal Processing......Page 36
1.4: Brief Overview of the Book......Page 39
2.1: Discrete-time Signals......Page 41
2.2: Discrete Systems......Page 55
2.3: Convolution......Page 59
2.4: Difference Equations......Page 66
2.5: Problems......Page 72
3.1: The Discrete-time Fourier Transform (DTFT)......Page 78
3.2: The Properties of the DTFT......Page 86
3.3: The Frequency Domain Representation of LTI Systems......Page 93
3.4: Sampling and Reconstruction of Analog Signals......Page 99
3.5: Problems......Page 116
4.1: The Bilateral z-Transform......Page 122
4.2: Important Properties of the z-Transform......Page 126
4.3: Inversion of the z-Transform......Page 131
4.4: System Representation in the z-Domain......Page 137
4.5: Solutions of the Difference Equations......Page 147
4.6: Problems......Page 153
CHAPTER 5: THE DISCRETE FOURIER TRANSFORM......Page 160
5.1: The Discrete Fourier Series......Page 161
5.2: Sampling and Reconstruction in the z-Domain......Page 168
5.3: The Discrete Fourier Transform......Page 173
5.4: Properties of the Discrete Fourier Transform......Page 184
5.5: Linear Convolution Using the DFT......Page 199
5.6: The Fast Fourier Transform......Page 206
5.7: Problems......Page 219
CHAPTER 6: IMPLEMENTATION OF DISCRETE-TIME FILTERS......Page 231
6.1: Basic Elements......Page 232
6.2: IIR Filter Structures......Page 233
6.3: FIR Filter Structures......Page 247
6.4: Lattice Filter Structures......Page 258
6.5: Overview of Finite-Precision Numerical Effects......Page 269
6.6: Representation of Numbers......Page 270
6.7: The Process of Quantization and Error Characterizations......Page 285
6.8: Quantization of Filter Coefficients......Page 292
6.9: Problems......Page 307
CHAPTER 7: FIR FILTER DESIGN......Page 322
7.1: Preliminaries......Page 323
7.2: Properties of Linear-phase FIR Filters......Page 326
7.3: Window Design Techniques......Page 341
7.4: Frequency Sampling Design Techniques......Page 363
7.5: Optimal Equiripple Design Technique......Page 377
7.6: Problems......Page 394
CHAPTER 8: IIR FILTER DESIGN......Page 405
8.1: Some Preliminaries......Page 406
8.2: Some Special Filter Types......Page 409
8.3: Characteristics of Prototype Analog Filters......Page 419
8.4: Analog-to-Digital Filter Transformations......Page 442
8.5: Lowpass Filter Design Using MATLAB......Page 462
8.6: Frequency-band Transformations......Page 467
8.7: Problems......Page 480
CHAPTER 9: SAMPLING RATE CONVERSION......Page 493
9.1: Introduction......Page 494
9.2: Decimation by a Factor D......Page 496
9.3: Interpolation by a Factor I......Page 505
9.4: Sampling Rate Conversion by a Rational Factor I/D......Page 512
9.5: FIR Filter Designs for Sampling Rate Conversion......Page 517
9.6: FIR Filter Structures for Sampling Rate Conversion......Page 539
9.7: Problems......Page 549
10.1: Analysis of A/D Quantization Noise......Page 557
10.2: Round-off Effects in IIR Digital Filters......Page 569
10.3: Round-off Effects in FIR Digital Filters......Page 597
10.4: Problems......Page 609
CHAPTER 11: APPLICATIONS IN ADAPTIVE FILTERING......Page 613
11.1: LMS Algorithm for Coefficient Adjustment......Page 615
11.2: System Identification or System Modeling......Page 618
11.3: Suppression of Narrowband Interference in a Wideband Signal......Page 619
11.5: Adaptive Channel Equalization......Page 622
12.1: Pulse-Code Modulation......Page 626
12.2: Differential PCM (DPCM)......Page 630
12.3: Adaptive PCM and DPCM (ADPCM)......Page 633
12.4: Delta Modulation (DM)......Page 637
12.5: Linear Predictive Coding (LPC) of Speech......Page 641
12.6: Dual-tone Multifrequency (DTMF) Signals......Page 645
12.7: Binary Digital Communications......Page 649
12.8: Spread-Spectrum Communications......Page 651
BIBLIOGRAPHY......Page 654
A......Page 656
C......Page 657
D......Page 658
E......Page 660
F......Page 661
I......Page 663
L......Page 664
M......Page 665
O......Page 666
R......Page 667
S......Page 668
V......Page 670
Z......Page 671