پردازش سیگنال دیجیتال: مبانی و کاربردها (مجموعه پردازش سیگنال دیجیتال)

I think the reason DSP is preceived as difficult to understand, is because traditional book/class tend to spend most of the time solving equations such as DFT, FFT, Z-transform of different signal (pulse, unit step, sinusoidal, square, triangular etc), differential equation to get zero and pole, stability, then bilinear transformation etc. As a result, students might spend all the time doing mathematics, than trying to understand what DSP is and how to apply it to real world. This book reduce the mathematics and proof to minimum when required, rather than throwing you all the equations and explanation at one time. For example, I dont always why other authors always try to explain the details of non-causal system since they usually say real time system is always causal and hence ignore the non-causal DSP design.I particularly like the exercise and MATLAB code. They are incredibly simple, but are very useful to aid understanding and cover almost all aspect of a DSP system, from equalizing ADC quantising noise, through to multirate DSP (which is rarely touched by most of the fundamental DSP books). The MATLAB code allows the reader to design filter and visualise the signal and its frequency spectrum very quickly. I think it is more important for beginner to first understand the concept of DSP using MATLAB, than trying to solve the mathematics, try to write C-code for FFT and plotting graph.As I mentioned, the mathematics is not detailed in this book. This is a double edge sword. Reader should have good knowledge on FFT, Z-transform. I recommend reading "Understanding Digital Signal Processing" by Lyons beforehand, which provide very good explanation on fundamental DSP theory and mathematics.This is the most useful DSP book I have ever read!!!! Digital Signal Processing......Page 4 Copyright Page......Page 5 Contents......Page 6 Preface......Page 14 About the Author......Page 18 1.1 Basic Concepts of Digital Signal Processing......Page 19 1.2.1 Digital Filtering......Page 21 1.2.2 Signal Frequency (Spectrum) Analysis......Page 22 1.3.1 Digital Crossover Audio System......Page 24 1.3.3 Speech Coding and Compression......Page 25 1.3.4 Compact-Disc Recording System......Page 27 1.3.5 Digital Photo Image Enhancement......Page 28 1.4 Digital Signal Processing Applications......Page 29 References......Page 30 2.1 Sampling of Continuous Signal......Page 31 2.2 Signal Reconstruction......Page 38 2.2.1 Practical Considerations for Signal Sampling: Anti-Aliasing Filtering......Page 43 2.2.2 Practical Considerations for Signal Reconstruction: Anti-Image Filter and Equalizer......Page 47 2.3 Analog-to-Digital Conversion, Digital-to-Analog Conversion, and Quantization......Page 53 2.4 Summary......Page 67 2.5 MATLAB Programs......Page 68 2.6 Problems......Page 69 References......Page 74 3.1 Digital Signals......Page 75 3.1.1 Common Digital Sequences......Page 76 3.1.2 Generation of Digital Signals......Page 80 3.2.1 Linearity......Page 82 3.2.2 Time Invariance......Page 83 3.2.3 Causality......Page 85 3.3.1 Format of Difference Equation......Page 86 3.3.2 System Representation Using Its Impulse Response......Page 87 3.4 Bounded-in-and-Bounded-out Stability......Page 90 3.5 Digital Convolution......Page 92 3.6 Summary......Page 100 3.7 Problems......Page 101 4.1 Discrete Fourier Transform......Page 105 4.1.1 Fourier Series Coefficients of Periodic Digital Signals......Page 106 4.1.2 Discrete Fourier Transform Formulas......Page 110 4.2 Amplitude Spectrum and Power Spectrum......Page 116 4.3 Spectral Estimation Using Window Functions......Page 128 4.5 Fast Fourier Transform......Page 138 4.5.1 Method of Decimation-in-Frequency......Page 140 4.5.2 Method of Decimation-in-Time......Page 145 4.7 Problems......Page 149 References......Page 152 5.1 Definition......Page 153 5.2 Properties of the z-Transform......Page 157 5.3 Inverse z-Transform......Page 160 5.3.1 Partial Fraction Expansion Using MATLAB......Page 166 5.4 Solution of Difference Equations Using the z-Transform......Page 169 5.5 Summary......Page 173 5.6 Problems......Page 174 Reference......Page 176 6.1 The Difference Equation and Digital Filtering......Page 177 6.2 Difference Equation and Transfer Function......Page 183 6.2.1 Impulse Response, Step Response, and System Response......Page 187 6.3 The z-Plane Pole-Zero Plot and Stability......Page 189 6.4 Digital Filter Frequency Response......Page 197 6.5 Basic Types of Filtering......Page 206 6.6.1 Direct-Form I Realization......Page 213 6.6.2 Direct-Form II Realization......Page 214 6.6.3 Cascade (Series) Realization......Page 215 6.6.4 Parallel Realization......Page 216 6.7.1 Pre-Emphasis of Speech......Page 220 6.7.2 Bandpass Filtering of Speech......Page 223 6.8 Summary......Page 226 6.9 Problems......Page 227 Reference......Page 232 7.1 Finite Impulse Response Filter Format......Page 233 7.2 Fourier Transform Design......Page 235 7.3 Window Method......Page 247 7.4.1 Noise Reduction......Page 271 7.4.2 Speech Noise Reduction......Page 273 7.4.3 Two-Band Digital Crossover......Page 274 7.5 Frequency Sampling Design Method......Page 278 7.6 Optimal Design Method......Page 286 7.7.1 Transversal Form......Page 298 7.7.2 Linear Phase Form......Page 300 7.8 Coefficient Accuracy Effects on Finite Impulse Response Filters......Page 301 7.9 Summary of Finite Impulse Response (FIR) Design Procedures and Selection of FIR Filter Design Methods in Practice......Page 305 7.10 Summary......Page 308 7.11 MATLAB Programs......Page 309 7.12 Problems......Page 312 References......Page 319 8.1 Infinite Impulse Response Filter Format......Page 321 8.2 Bilinear Transformation Design Method......Page 323 8.2.1 Analog Filters Using Lowpass Prototype Transformation......Page 324 8.2.2 Bilinear Transformation and Frequency Warping......Page 328 8.2.3 Bilinear Transformation Design Procedure......Page 335 8.3.1 Lowpass Prototype Function and Its Order......Page 340 8.3.2 Lowpass and Highpass Filter Design Examples......Page 344 8.3.3 Bandpass and Bandstop Filter Design Examples......Page 354 8.4 Higher-Order Infinite Impulse Response Filter Design Using the Cascade Method......Page 361 8.5 Application: Digital Audio Equalizer......Page 364 8.6 Impulse Invariant Design Method......Page 368 8.7 Polo-Zero Placement Method for Simple Infinite Impulse Response Filters......Page 376 8.7.1 Second-Order Bandpass Filter Design......Page 377 8.7.2 Second-Order Bandstop (Notch) Filter Design......Page 378 8.7.3 First-Order Lowpass Filter Design......Page 380 8.7.4 First-Order Highpass Filter Design......Page 382 8.8 Realization Structures of Infinite Impulse Response Filters......Page 383 8.8.1 Realization of Infinite Impulse Response Filters in Direct-Form I and Direct-Form II......Page 384 8.8.2 Realization of Higher-Order Infinite Impulse Response Filters via the Cascade Form......Page 386 8.9 Application: 60-Hz Hum Eliminator and Heart Rate Detection Using Electrocardiography......Page 388 8.10 Coefficient Accuracy Effects on Infinite Impulse Response Filters......Page 395 8.11 Application: Generation and Detection of Dual-Tone Multifrequency Tones Using Goertzel Algorithm......Page 399 8.11.1 Single-Tone Generator......Page 400 8.11.2 Dual-Tone Multifrequency Tone Generator......Page 402 8.11.3 Goertzel Algorithm......Page 404 8.11.4 Dual-Tone Multifrequency Tone Detection Using the Modified Goertzel Algorithm......Page 409 8.12 Summary of Infinite Impulse Response (IIR) Design Procedures and Selection of the IIR Filter Design Methods in Practice......Page 414 8.13 Summary......Page 419 8.14 Problems......Page 420 References......Page 430 9.1 Digital Signal Processor Architecture......Page 431 9.2.1 Multiplier and Accumulator......Page 434 9.2.2 Shifters......Page 435 9.2.3 Address Generators......Page 436 9.3 Digital Signal Processors and Manufactures......Page 437 9.4.1 Fixed-Point Format......Page 438 9.4.2 Floating-Point Format......Page 447 9.4.3 IEEE Floating-Point Formats......Page 452 9.4.5 Fixed-Point Digital Signal Processors......Page 455 9.4.6 Floating-Point Processors......Page 457 9.5 Finite Impulse Response and Infinite Impulse Response Filter Implementation in Fixed-Point Systems......Page 459 9.6.1 Overview of TMS320C67x DSK......Page 465 9.6.2 Concept of Real-Time Processing......Page 469 9.6.3 Linear Buffering......Page 470 9.6.4 Sample C Programs......Page 473 9.7 Summary......Page 478 9.8 Problems......Page 479 References......Page 480 10.1 Introduction to Least Mean Square Adaptive Finite Impulse Response Filters......Page 481 10.2 Basic Wiener Filter Theory and Least Mean Square Algorithm......Page 485 10.3.1 Noise Cancellation......Page 491 10.3.2 System Modeling......Page 497 10.3.3 Line Enhancement Using Linear Prediction......Page 502 10.4 Other Application Examples......Page 504 10.4.1 Canceling Periodic Interferences Using Linear Prediction......Page 505 10.4.2 Electrocardiography Interference Cancellation......Page 506 10.4.3 Echo Cancellation in Long-Distance Telephone Circuits......Page 507 10.6 Problems......Page 509 References......Page 514 11.1 Linear Midtread Quantization......Page 515 11.2.1 Analog mu-Law Companding......Page 519 11.2.2 Digital mu-Law Companding......Page 524 11.3.1 Examples of Differential Pulse Code Modulation and Delta Modulation......Page 528 11.3.2 Adaptive Differential Pulse Code Modulation G.721......Page 533 11.4.1 Discrete Cosine Transform......Page 540 11.4.2 Modified Discrete Cosine Transform......Page 543 11.4.3 Transform Coding in MPEG Audio......Page 548 11.5 Summary......Page 551 11.6 MATLAB Programs......Page 552 11.7 Problems......Page 568 References......Page 573 12.1 Multirate Digital Signal Processing Basics......Page 575 12.1.1 Sampling Rate Reduction by an Integer Factor......Page 576 12.1.2 Sampling Rate Increase by an Integer Factor......Page 582 12.1.3 Changing Sampling Rate by a Non-Integer Factor L/M......Page 588 12.1.4 Application: CD Audio Player......Page 593 12.1.5 Multistage Decimation......Page 596 12.2 Polyphase Filter Structure and Implementation......Page 601 12.3 Oversampling of Analog-to-Digital Conversion......Page 607 12.3.1 Oversampling and Analog-to-Digital Conversion Resolution......Page 608 12.3.2 Sigma-DeltaModulation Analog-to-Digital Conversion......Page 611 12.4 Application Example: CD Player......Page 617 12.5 Undersampling of Bandpass Signals......Page 619 12.6 Summary......Page 627 12.7 Problems......Page 628 References......Page 632 13.1 Image Processing Notation and Data Formats......Page 635 13.1.1 8-Bit Gray Level Images......Page 636 13.1.2 24-Bit Color Images......Page 637 13.1.3 8-Bit Color Images......Page 638 13.1.4 Intensity Images......Page 639 13.1.5 Red, Green, Blue Components and Grayscale Conversion......Page 640 13.1.6 MATLAB Functions for Format Conversion......Page 642 13.2.1 Grayscale Histogram and Equalization......Page 643 13.2.2 24-Bit Color Image Equalization......Page 650 13.2.3 8-Bit Indexed Color Image Equalization......Page 651 13.2.4 MATLAB Functions for Equalization......Page 654 13.3 Image Level Adjustment and Contrast......Page 655 13.3.1 Linear Level Adjustment......Page 656 13.3.2 Adjusting the Level for Display......Page 659 13.4 Image Filtering Enhancement......Page 660 13.4.1 Lowpass Noise Filtering......Page 661 13.4.2 Median Filtering......Page 664 13.4.3 Edge Detection......Page 669 13.4.4 MATLAB Functions for Image Filtering......Page 673 13.5 Image Pseudo-Color Generation and Detection......Page 675 13.6 Image Spectra......Page 679 13.7 Image Compression by Discrete Cosine Transform......Page 682 13.7.1 Two-Dimensional Discrete Cosine Transform......Page 684 13.7.2 Two-Dimensional JPEG Grayscale Image Compression Example......Page 687 13.7.3 JPEG Color Image Compression......Page 689 13.9 Video Signal Basics......Page 695 13.9.1 Analog Video......Page 696 13.9.2 Digital Video......Page 703 13.10 Motion Estimation in Video......Page 705 13.11 Summary......Page 708 13.12 Problems......Page 710 References......Page 716 A.1 Basic Commands and Syntax......Page 717 A.2 MATLAB Array and Indexing......Page 721 A.4 MATLAB Script Files......Page 722 A.5 MATLAB Functions......Page 723 References......Page 725 B.1.1 Sine-Cosine Form......Page 727 B.1.2 Amplitude-Phase Form......Page 728 B.1.3 Complex Exponential Form......Page 729 B.1.4 Spectral Plots......Page 732 B.1.5 Fourier Transform......Page 739 B.2.1 Laplace Transform and Its Table......Page 744 B.2.2 Solving Differential Equations Using Laplace Transform......Page 745 B.2.3 Transfer Function......Page 748 B.3.1 Poles, Zeros, and Stability......Page 749 B.3.2 Convolution......Page 751 B.3.3 Sinusoidal Steady-State Response......Page 753 B.4 Problems......Page 754 References......Page 758 C.1 Normalized Butterworth Function......Page 759 C.2 Normalized Chebyshev Function......Page 762 D.1 Sinusoidal Steady-State Response......Page 767 D.2 Properties of the Sinusoidal Steady-State Response......Page 769 Appendix E: Finite Impulse Response Filter Design Equations by the Frequency Sampling Design Method......Page 771 Appendix F: Some Useful Mathematical Formulas......Page 775 Bibliography......Page 779 Answers to Selected Problems......Page 783 Index......Page 809 Color Plates......Page 835 This book will enable electrical engineers and technicians in the fields of the biomedical, computer, and electronics engineering, to master the essential fundamentals of DSP principles and practice. Coverage includes DSP principles, applications, and hardware issues with an emphasis on applications. Many instructive worked examples are used to illustrate the material and the use of mathematics is minimized for easier grasp of concepts.
In addition to introducing commercial DSP hardware and software, and industry standards that apply to DSP concepts and algorithms, topics covered include adaptive filtering with noise reduction and echo cancellations; speech compression; signal sampling, digital filter realizations; filter design; multimedia applications; over-sampling, etc. More advanced topics are also covered, such as adaptive filters, speech compression such as PCM, u-law, ADPCM, and multi-rate DSP and over-sampling ADC.

*Covers DSP principles and hardware issues with emphasis on applications and many worked examples
*Website with MATLAB programs for simulation and C programs for real-time DSP
*End of chapter problems are helpful in ensuring retention and understanding of what was just read "This textbook present digital signal processing (DSP) principles, applications, and hardware implementation issues, emphasizing achievable results and conclusions through the presentation of numerous worked examples, while reducing the use of mathematics for an easier grasp of the concepts." "This text gives students in electronics, computer engineering, and bioengineering an understanding of essential DSP principles and implementation, demonstrating how the subject is fundamental to engineering as practiced today."--Jacket