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Digital Signal Processing Using MATLAB by Ingle, Vinay K.; Proakis, John G.

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Item specifics

Condition
Good: A book that has been read but is in good condition. Very minimal damage to the cover including ...
Binding
Paperback
Product Group
Book
Weight
2 lbs
IsTextBook
Yes
ISBN
9780495073116
Subject Area
Computers, Technology & Engineering
Publication Name
Digital Signal Processing with Matlab
Publisher
Course Technology
Item Length
9.2 in
Subject
Engineering (General), Signals & Signal Processing, Mathematical & Statistical Software, Electrical, Computer Engineering
Publication Year
2006
Type
Textbook
Format
Perfect
Language
English
Item Height
1 in
Author
Vinay K. Ingle, John G. Proakis
Item Weight
32.1 Oz
Item Width
7.4 in
Number of Pages
512 Pages

About this product

Product Information

This supplement to any standard DSP text is one of the first books to successfully integrate the use of MATLAB in the study of DSP concepts. In this book, 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. This updated second edition includes new homework problems and revises the scripts in the book, available functions, and m-files to MATLAB V7.

Product Identifiers

Publisher
Course Technology
ISBN-10
0495073113
ISBN-13
9780495073116
eBay Product ID (ePID)
102920342

Product Key Features

Number of Pages
512 Pages
Language
English
Publication Name
Digital Signal Processing with Matlab
Publication Year
2006
Subject
Engineering (General), Signals & Signal Processing, Mathematical & Statistical Software, Electrical, Computer Engineering
Type
Textbook
Subject Area
Computers, Technology & Engineering
Author
Vinay K. Ingle, John G. Proakis
Format
Perfect

Dimensions

Item Height
1 in
Item Weight
32.1 Oz
Item Length
9.2 in
Item Width
7.4 in

Additional Product Features

Edition Number
2
Dewey Edition
22
Target Audience
College Audience
Illustrated
Yes
Dewey Decimal
621.3/822/028555
Table of Content
Chapter 1 - Introduction1.1 Overview of Digital Signal Processing1.2 A Few Words about MATLAB®Chapter 2 - Discrete-time Signals and Systems2.1 Discrete-time Signals2.2 Discrete Systems2.3 Convolution2.4 Difference Equations2.5 ProblemsChapter 3 - The Discrete-time Fourier Analysis 3.1 The discrete-time Fourier Transform3.2 The Properties of the DTFT3.3 The Frequency Domain Representation of LSI Systems3.4 Sampling and Reconstruction of Analog Signals3.5 ProblemsChapter 4 - The z-Transform4.1 The Bilateral z-Transform4.2 Important Properties of the z-Transform4.3 Inversion of the z-Transform4.4 System Representation in the z-Domain4.5 Solutions of the Difference Equations4.6 ProblemsChapter 5 - The Discrete Fourier Transform5.1 The Discrete Fourier Series5.2 Sampling and Reconstruction in the z-Domain5.3 The Discrete Fourier Transform5.4 Properties of the Discrete Fourier Transform5.5 Linear Convolution using the DFT5.6 Circulant matrices5.7 The Fast Fourier Transform5.8 ProblemsChapter 6 - Digital Filter Structures6.1 Basic Elements6.2 IIR Filter Structures6.3 FIR Filter Structures6.4 Lattice Structures6.5 ProblemsChapter 7 - FIR Filter Design7.1 Preliminaries7.2 Properties of Linear Phase FIR Filters7.3 Window Design Techniques7.4 Frequency Sampling Design Techniques7.5 Optimal Equiripple Design Technique7.6 ProblemsChapter 8 - IIR Filter Design8.1 Some preliminaries8.2 Characteristics of Prototype Analog Filters8.3 Analog-to-Digital Filter Transformations8.4 Lowpass Filter Design using MATLAB 8.5 Frequency-band Transformations8.6 Comparison of FIR vs. IIR Filters8.7 ProblemsChapter 9 - Finite Word-Length Effects9.1 Overview9.2 Representation of Numbers9.3 The Process of Quantization and Error Characterization9.4 Quantization of Filter Coefficients9.5 Analysis of A/D Quantization Noise9.6 Round-Off Effects in IIR Digital Filters9.7 Round-Off Noise in FIR Filter Realizations9.8 Summary9.9 ProblemsChapter 10 - Sampling Rate Conversion10.1 Introduction10.2 Decimation by a Factor D10.3 Interpolation by a Factor10.4 Sampling Rate Conversion by a Rational Factor I/D10.5 FIR Filter Designs for Sample Rate Conversion10.6 FIR Filter Structures for Sampling-Rate Conversion10.7 Summary10.8 ProblemsChapter 11 - Applications in Adaptive Filtering11.1 LMS Algorithm for Coefficient Adjustment11.2 System Identification or System Modeling11.3 Suppression of Narrowband Interference in a Wideband Signal 11.4 Adaptive Line Enhancement11.5 Adaptive Channel Equalization11.6 SummaryChapter 12 - Applications in Communications12.1 Pulse Code Modulation12.2 Differential PCM (DPCM)12.3 Adaptive PCM and DPCM (ADPCM)12.4 Delta Modulation (DM)12.5 Linear Predictive Coding (LPC) of Speech12.6 Dual-Tone Multifrequency (DTMF) Signals12.7 Binary Digital Communications12.8 Spread Spectrum Communications12.9 Summary
Copyright Date
2007

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