[DSP] Digital Signal Processing & Fundamentals and Applications by Li Tan, e-book free download



E-Book Details:

Title:
Digital Signal Processing & Fundamentals and Applications
Publisher:
Elsevier
Author:
Li Tan,DeVry University,Decatur, Georgia
Edition:
Hardcover,  1 edition (July 26 2007)
Format:
pdf
ISBN:
0123740908
EAN:
978-0-12-374090-8
No. of Pages:
840

Book Description:

The book is suitable for a sequence of two-semester courses at the senior level in undergraduate electronics, computer, and biomedical engineering technology programs. Chapters 1 to 8 provide the topics for a one semester course, and a second course can complete the rest of the chapters. This textbook can also be used in an introductory DSP course at the junior level in undergraduate electrical engineering programs at traditional colleges. Additionally, the book should be useful as a reference for undergraduate engineering students, science Students, and practicing engineers. The material has been tested in two consecutive courses in signal processing sequence at DeVry University on the Decatur campus in Georgia. With the background established from this book, students can be well prepared to move forward to take other senior-level courses that deal with digital signals and systems for communications and controls.

ABOUT THE AUTHOR:

Dr. Li Tan is a Professor of Electronics Engineering Technology at DeVry University, Decatur, Georgia. He received his M.S. and Ph.D. degrees in Electrical Engineering from the University of New Mexico. He has extensively
taught analog and digital signal processing and analog and digital communications for many years. Before teaching at DeVry University, Dr. Tan worked in the DSP and communications industry. Dr. Tan is a senior member of the Institute of Electrical and Electronics Engineers (IEEE). His principal technical areas include digital signal processing, adaptive signal processing, and digital communications. He has published a number of papers in these areas

Table of Contents:
The textbook consists of 13 chapters, organized as follows:
Chapter 1 introduces concepts of DSP and presents a general DSP block diagram. Application examples are included.
& Chapter 2 covers the sampling theorem described in time domain and frequency domain and also covers signal reconstruction. Some practical considerations for designing analog anti-aliasing lowpass filters and antiimage
lowpass filters are included. The chapter ends with a section dealing with analog-to-digital conversion (ADC) and digital-to-analog conversion (DAC), as well as signal quantization and encoding. & Chapter 3 introduces digital signals, linear time-invariant system concepts, difference equations, and digital convolutions.
Chapter 4 introduces the discrete Fourier transform (DFT) and digital signal spectral calculations using the DFT. Applying the DFT to estimate the speech spectrum is demonstrated. The chapter ends with a section
dedicated to illustrating fast Fourier transform (FFT) algorithms.
 Chapter 5 is devoted to the z-transform and difference equations.
 Chapter 6 covers digital filtering using difference equations, transfer functions, system stability, digital filter frequency responses, and implementation methods such as the direct form I and direct form II.
 Chapter 7 deals with various methods of finite impulse response (FIR) filter design, including the Fourier transform method for calculating FIR filter coefficients, window method, frequency sampling design, and optimal design.
Chapter 7 also includes applications using FIR filters for noise reduction and digital crossover system design.
Chapter 8 covers various methods of infinite impulse response (IIR) filter design, including the bilinear transformation (BLT) design, impulse invariant design, and pole-zero placement design. Applications using IIR filters include audio equalizer design, biomedical signal enhancement, dual-tone multifrequency (DTMF) tone generation and detection with the Goertzel
algorithm.
 Chapter 9 introduces DSP architectures, software and hardware, and fixed-point and floating-point implementations of digital filters.
 Chapter 10 covers adaptive filters with applications such as noise cancellation, system modeling, line enhancement, cancellation of periodic interferences, echo cancellation, and 60-Hz interference cancellation in biomedical signals.
 Chapter 11 is devoted to speech quantization and compression, including pulse code modulation (PCM) coding, mu-law compression, adaptive differential pulse code modulation (ADPCM) coding, windowed modified discrete cosine transform (W-MDCT) coding, and MPEG audio format, specifically MP3 (MPEG-1, layer 3).
Chapter 12 covers topics pertaining to multirate DSP and applications, as well as principles of oversampling ADC, such as sigma-delta modulation. Undersampling for bandpass signals is also examined.
Finally, Chapter 13 covers image enhancement using histogram equalization and filtering methods, including edge detection. The chapter also explores pseudo-color image generation and detection, two-dimensional spectra, JPEG compression using DCT, and the mixing of two images to create a video sequence. Finally, motion compensation of the video sequence
is explored, which is a key element of video compression used in MPEG.
MATLAB programs are listed wherever they are possible. Therefore, a MATLAB tutorial should be given to students who are new to the MATLAB environment.
 Appendix A serves as a MATLAB tutorial.
Appendix B reviews key fundamentals of analog signal processing. Topics include Fourier series, Fourier transform, Laplace transform, and analog system basics.
 Appendixes C, D, and E overview Butterworth and Chebyshev filters, sinusoidal steady-state responses in digital filters, and derivation of the FIR filter design equation via the frequency sampling method, respectively.
 Appendix F offers general useful mathematical formulas.


Don’t know how to Download ???????????? click here
Posted by creativeworld9 at 5:50 PM
Labels, , ,

0 comments :

Post a Comment

Followers