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Digital Signal Processing: Instant Access
Digital Signal Processing: Instant Access
Digital Signal Processing: Instant Access
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Digital Signal Processing: Instant Access

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Digital signal processing is commonplace in most electronics including MP3 players, HDTVs, and phones, just to name a few of the applications. The engineers creating these devices are in need of essential information at a moment's notice. The Instant Access Series provides all the critical content that a signal or communications engineer needs in his or her daily work. This book provides an introduction to DSPs as well as succinct overviews of linear systems, digital filters, and digital compression. This book is filled with images, figures, tables, and easy to find tips and tricks for the engineer that needs material fast to complete projects to deadline.
  • Tips and tricks feature that will help engineers get info fast and move on to the next issue
  • Easily searchable content complete with tabs, chapter table of contents, bulleted lists, and boxed features
  • Just the essentials, no need to page through material not needed for the current project
LanguageEnglish
Release dateNov 25, 2008
ISBN9780080560137
Digital Signal Processing: Instant Access

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    Digital Signal Processing - James D. Broesch

    Digital Signal Processing

    Instant Access

    James D. Broesch

    Copyright

    Copyright © 2009, Elsevier Inc. All rights reserved.

    30 Corporate Drive, Suite 400, Burlington, MA 01803, USA

    Linacre House, Jordan Hill, Oxford OX2 8DP, UK

    No part of this publication may be reproduced, stored in a retrieval system, or transmitted in any form or by any means, electronic, mechanical, photocopying, recording, or otherwise, without the prior written permission of the publisher.

    Permissions may be sought directly from Elsevier's Science & Technology Rights Department in Oxford, UK: phone: (+44) 1865 843830, fax: (+44) 1865 853333, E-mail: permissions@elsevier.com. You may also complete your request online via the Elsevier homepage (http://www.elsevier.com), by selecting Support & Contact then Copyright and Permission and then Obtaining Permissions.

    Recognizing the importance of preserving what has been written, Elsevier prints itsbooks on acid-free paper whenever possible.

    Library of Congress Cataloguing-in-Publication Data

    Application submitted.

    British Library Cataloguing-in-Publication Data

    A catalogue record for this book is available from the British Library.

    ISBN: 978-0-7506-8976-2

    For information on all Newnes publications visit our Web site at: www.books.elsevier.com

    Printed in Canada

    08 09 10 11 12 13 10 9 8 7 6 5 4 3 2 1

    Typeset by Charon Tec Ltd., A Macmillan Company (www.macmillansolutions.com)

    Brief Table of Contents

    Copyright

    Brief Table of Contents

    Table of Contents

    List of Figures

    List of Tables

    Chapter 1. Why DSP?

    Chapter 2. The Analog-Digital Interface

    Chapter 3. DSP System General Model

    Chapter 4. The Math of DSP

    Chapter 5. Transforms

    Chapter 6. Digital Filters

    Chapter 7. Applications of DSP

    Chapter 8. Digital Signal Processors

     Glossary

     Bibliography

    Table of Contents

    Copyright

    Brief Table of Contents

    Table of Contents

    List of Figures

    List of Tables

    Chapter 1. Why DSP?

    The Need for DSP

    Advantages of DSP

    Learning Digital Signal Processing Techniques

    Instant Summary

    Chapter 2. The Analog-Digital Interface

    Sampling and Reconstruction

    Quantization

    Encoding and Modulation

    Technology Trade-offs

    Number Representation

    Digital-to-Analog Conversion

    Types of DACs

    Analog-to-Digital Conversion

    Types of ADCs

    Successive Approximation ADCs

    Instant Summary

    Chapter 3. DSP System General Model

    The Big Picture

    Input

    Signal-conditioning Circuit

    Anti-aliasing Filter

    Analog-to-Digital Converter

    Processor

    Program Store, Data Store

    Data Transmission

    Display and User Input

    Digital-to-Analog Converter

    Output Smoothing Filter

    Output Amplifier

    Output Transducer

    Signal Acquisition

    More on Sampling Theory

    Sampling Resolution

    Instant Summary

    Chapter 4. The Math of DSP

    Functions

    Limits

    Integration

    Oscillatory Motion

    Complex Numbers

    Example Applications

    Filters

    Example 1

    Example 2

    Causality

    Convolution

    The Fourier Series

    The Nyquist Theorem Completed

    Orthogonality

    Continuous Functions vs. Discrete Sequences

    Orthogonality Continued

    Quadrature

    Instant Summary

    Chapter 5. Transforms

    Background

    The z-transform and DFT

    Application of The DFT

    The Fourier Transform

    Properties of The Fourier Transform

    The Laplace Transform

    Fast Fourier Transform (FFT)

    Instant Summary

    Chapter 6. Digital Filters

    Fir Filters

    What Is an FIR Filter?

    Stability of FIR Filters

    Cost of Implementation

    FIR Filter Design Methodology

    FIR Design Example

    The IIR

    Instant Summary

    Chapter 7. Applications of DSP

    Measurements and Analysis

    Telecommunications

    Audio and Television

    Household Appliances and Toys

    Automotive

    Nonlinear Applications

    Median Filter

    Artificial Neural Networks

    Fuzzy Logic

    Instant Summary

    Chapter 8. Digital Signal Processors

    System Considerations

    Applications and Requirements

    Hardware Implementation

    Digital Signal Processors versus Microprocessors

    Conventional Microprocessors

    Digital Signal Processors

    The future

    Instant Summary

     Glossary

     Bibliography

    References

    List of Figures

    FIGURE 1.1. Analog and digital systems

    Figure 2.1. Sampling viewed as a multiplication process

    Figure 2.2. Example, a byte (96H) encoded (weights in parenthesis) using PCM in parallel mode (parallel bus, 8 bits, eight wires) and in serial mode as an 8-bit pulse train (over one wire)

    Figure 2.3. Different modulation schemes for serial mode data communication, PAM, PPM, PNM, PWM and PDM

    Figure 2.4. A generic multiplying DAC using current sources controlled by the bits in the digital input code

    Figure 2.5. An example integrating (counting) DAC. (In a real world implementation, additional control and synchronization circuits are needed.)

    Figure 2.6. An example 2-bit flash ADC

    Figure 2.7. An example SAR ADC (simplified block diagram)

    Figure 2.8. A simplified integrating ADC

    Figure 2.9. A simplified (a) delta modulator and (b) demodulator

    Figure 2.10. A simplified, oversampled bitstream sigma–delta ADC

    Figure 3.1. The general model for a DSP system

    Figure 3.2. Aliasing

    Figure 4.1. Geometric interpretation of integration

    Figure 4.2. Polar and rectangular coordinates

    Figure 4.3. Ideal LC circuit showing voltage and current relationships

    Figure 4.4. Sample points on a sine wave

    Figure 4.5. C listing for generating a sine wave

    Figure 4.6. DAC output for a sine wave

    Figure 4.7. Two cycles of a sine wave

    Figure 4.8. The basic low-pass filter

    Figure 4.9. A noise pop on a sine wave

    Figure 4.10. Effects of a moving average filter

    Figure 4.11. Frequency response of a simple filter

    Figure 4.12. Effects of a difference filter

    Figure 4.13. Creating a square wave from a series of sine waves

    Figure 4.14. The average area under a sine wave is zero

    Figure 4.15. The average of the square of a sine wave is greater than zero

    Figure 4.16. Orthogonality: imaginary part in phase

    Figure 4.17. Orthogonality: real part in phase

    Figure 4.18. Orthogonality: quadrature

    Figure 5.1. Spectrum analysis example

    Figure 5.2. Transform of a square wave

    Figure 5.3. The z-plane

    Figure 5.4. Composite waveform

    Figure 5.5. Spectrum for the signal in Figure 5.4

    Figure 5.6. Damped LRC circuit

    Figure 6.1. Three standard filters

    Figure 6.2. Standard architecture for an FIR filter

    Figure 6.3. Filter design process for an FIR filter

    Figure 6.4. Block diagram for low-pass filter example

    Figure 6.5. Sample waveform for the low-pass filter example

    Figure 6.6. Resulting spectrum for the sample signal

    Figure 6.7. Desired filter shape

    Figure 6.8. Pole/zero plot for the low-pass filter example

    Figure 6.9. Electronic worksheet for designing the low-pass filter

    Figure 6.10. Table of the computations used in Figure 6.9

    Figure 7.1. Median filtering of a three-valued signal by threshold decomposition, M=3

    Figure 7.2. Example of a simple fuzzy logic system

    Figure 8.1. Stream processing and block processing

    Figure 8.2. von Neumann architecture, program code and data share memory

    Figure 8.3. Harvard architecture, separate buses and memories. I/O, data and instructions can be accessed simultaneously

    List of Tables

    Table 2.1. Some fixed-point binary number formats

    Table 2.2. Decimation filter example

    Table 3.1. Binary mapping

    Table 4.1. Most frequently used integrals (where c and a are constants and u and v are functions of x)

    Table 4.2. Result of applying averaging routine to signal in Figure 4.9

    Table 4.3. Frequency response

    Table 4.4. Results of applying Eq. 4.40 to data in Figure 4.9

    Table 4.5. Results of convolution example

    Table 5.1. Signal generation

    Table 5.2. DFT with frequency = 0

    Table 5.3. Caption not provide

    Table 5.4. Caption not provide

    Table 5.5. Some properties of the Fourier transform

    Chapter 1. Why DSP?

    DSP Definitions

    The acronym DSP is used for two terms, digital signal processing and digital signal processor, both of which are covered in this book. Digital signal processing is performing signal processing using digital techniques with the aid of digital hardware and/or some kind of computing device. Signal processing can of course be analog as well, but, for a variety of reasons, we may prefer to handle the processing digitally. A digital computer or processor that is designed especially for signal processing applications is called a digital signal processor.

    The Need for DSP

    To understand the relative merits of analog and digital processing, it is convenient to compare the two techniques in a common application. Figure 1.1 shows two approaches to recording sounds such as music or speech. Figure 1.1a is the analog approach. It works like this:

    Sound waves impact the microphone, where they are converted to electrical impulses.

    These electrical signals are amplified, then converted to magnetic fields by the recording head.

    As the magnetic tape moves under the head, the intensity of the magnetic fields is stored on the tape.

    FIGURE 1.1. Analog and digital systems

    The playback process is just the inverse of the recording process:

    As the magnetic tape moves under the playback head, the magnetic field on the tape is converted to an electrical signal.

    The signal is then amplified and sent to the speaker. The speaker converts the amplified signal back to sound waves.

    The advantage of the analog process is twofold: first, it is conceptually quite simple. Second, by definition, an analog signal can take on virtually an infinite number of values within the signal's dynamic range. Unfortunately, this analog process is inherently unstable. The amplifiers are subject to gain variation over temperature, humidity, and time. The magnetic tape stretches and shrinks, thus distorting the recorded signal. The magnetic fields themselves will, over time, lose some of their strength. Variations in the speed of the motor driving the tape cause additional distortion. All of these factors combine to ensure that the output signal will be considerably lower in quality than the input signal Each time the signal is passed on to another analog process, these adverse effects are multiplied. It is rare for an analog system to be able to make more than two or three generations of copies.

    Now let's look at the digital process as shown in Figure 1.1b:

    As in the analog case, the sound waves impact the microphone and are converted to electrical signals. These electrical signals are then amplified to a usable level.

    The electrical signals are measured or, in other words, they are converted to numbers.

    These numbers can now be stored or manipulated by a computer just as any other numbers are.

    To play back the signal, the numbers are simply converted back to electrical signals. As in the analog case, these signals are then used to drive a speaker.

    There are two distinct disadvantages to the digital process: first, it is far more complicated than the analog process; second, computers can only handle numbers of finite resolution. Thus, the (potentially) infinite resolution of the analog signal is lost.

    Insider Info

    The first major contribution in the area of digital filter synthesis was made by Kaiser at Bell Laboratories. His work showed how to design useful filters using the bilinear transform. Further, in about 1965 the famous paper by Cooley and Turkey was published. In this paper, FFT (fast Fourier transform), an efficient and fast way of performing the DFT (discrete Fourier transform) was demonstrated.

    Advantages of DSP

    Obviously, there must be some compensating benefits of the digital process, and indeed there are. First, once converted to numbers, the signal is unconditionally stable. Using techniques such as error detection and correction, it is possible to store, transmit, and reproduce numbers with no corruption. The twentieth generation of recording is therefore just as accurate as the first generation.

    Insider Info

    The problems with analog signal reproduction have some interesting implications. Future generations will never really know what the Beatles sounded like, for example. The commercial analog technology of the 1960s was simply not able to accurately record and reproduce the signals. Several generations of analog signals were needed to reproduce the sound: First, a master tape would be recorded, and then mixed and edited; from this, a metal master record would be produced, from which would come a plastic impression. Each step of the process was a new generation of recording, and each generation acted on the signal like a filter, reducing the frequency content and skewing the phase. As with the paintings in the Sistine Chapel, the true colors and brilliance of the original art is lost to history. Things are different for today s musicians. A thousand years from now historians will be able to accurately play back the digitally mastered CDs of today. The discs themselves may well deteriorate, but before they do, the digital numbers on them can be copied with perfect accuracy. Signals stored digitally are really just large arrays of numbers. As such, they are immune to the physical limitations of analog signals.

    There are other significant advantages to processing signals digitally. Geophysicists were one of the first groups to apply the techniques of signal processing. The seismic signals of interest to them are often of very low frequency, from 0.01 Hz to 10 Hz. It is difficult to build analog filters that work at these low frequencies. Component values must be so large that physically implementing the filter may well be impossible. Once the signals have been converted to digital numbers, however, it is a straightforward process to program a computer to perform the filtering.

    Other advantages to digital signals abound. For example, DSP can allow large bandwidth signals to be sent over narrow bandwidth channels. A 20-kHz signal can be digitized and then sent over a 5-kHz channel. The signal may take four times as long to get through the narrower bandwidth channel, but when it comes out the other side it can be reconstructed to its full 20-kHz bandwidth.

    In the same way, communications security can be greatly improved through DSP. Since the signal is sent as numbers, it can be easily encrypted. When received, the numbers are decrypted and then reproduced as the original signal. Modern secure telephone DSP systems allow this processing to be done with no detectable effect on the conversation.

    Technology Trade-offs

    DSP has several major advantages over analog signal processing techniques, including:

    Essentially perfect reproducibility

    Guaranteed accuracy (no individual tuning and pruning needed)

    Well-suited for volume production

    Learning Digital Signal Processing Techniques

    The most important first step of studying any subject is to grasp the overall picture and to understand the basics before diving into the depths. With that in mind, the goal of this book is to provide a broad introduction and overview of DSP techniques and applications. The authors seek to bring an intuitive understanding of the concepts and systems involved in the field of DSP engineering.

    Only a few years ago, DSP techniques were considered advanced and esoteric subjects, their use limited to research labs or advanced applications such as radar identification. Today, the technology has found its way into virtually every segment of electronics. Computer graphics, mobile entertainment and communication devices, and automobiles are just a few of the common examples.

    The rapid acceptance and commercialization of this technology has presented the modern design engineer with a serious challenge: either gain a working knowledge of these techniques or risk obsolescence. Traditionally, engineers have had two options for acquiring new skills: go back to school, or turn to vendors’ technical

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