An Introduction to Digital Signal Processing by John H. Karl (Auth.)

By John H. Karl (Auth.)

An advent to electronic sign Processing is written when you have to comprehend and use electronic sign processing and but don't desire to go through a multi-semester path series. utilizing merely calculus-level arithmetic, this booklet progresses speedily in the course of the basics to complicated themes corresponding to iterative least squares layout of IIR filters, inverse filters, strength spectral estimation, and multidimensional applications--all in a single concise volume.
This booklet emphasizes either the basic rules and their smooth desktop implementation. It provides and demonstrates how basic the particular laptop code is for complex smooth algorithms utilized in DSP. result of those courses, which the reader can effectively reproduction and use on a laptop, are provided in lots of genuine machine drawn plots.

Key Features
* assumes no past wisdom of sign processing yet leads as much as very complex techniques
combines exposition of basic ideas with useful applications
* contains issues of each one chapter
* offers intimately the right computing device algorithums for fixing difficulties

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Extra resources for An Introduction to Digital Signal Processing

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Is there a correct result? 13 Find the inverse of A = (1 - aZ) by expanding it in the geometric series for \a \ < 1. T h e n , write a computer program that convolves A with a truncated version of A~ . C o m p u t e A* A~ for various values of a and for various lengths of A~ . 13 for values of \a \ > 1. A n o t h e r way of digitizing an analog system is called step invariance. 15 1/(1-Z) = 1+ Z + Z +--. 2 for the digital form of the step function. 19 t/T digitize the system by computing the ratio of the digitized step function response to the digitized step function input.

O n the other h a n d , our naive guesses for differentiation operators did not lead to using 2(1 - Z ) / ( l + Z ) as suggested by the bilinear transform. W e would expect that the infinite impulse response, indicated by the (1 + Z ) in the denomi­ nator, would yield results superior to our previous 2-term operators. In­ d e e d , of the three difference operators that we have considered so far, all gave us an uneasy feeling because we would have really liked to have considered unavailable data values at t + \ and t - \ in order to center the difference at the time t.

T w o fundamental properties of the frequency response must be immediately emphasized from E q . 1): H(oo) is a continuous function of frequency and it is completely determined by the system's impulse re­ sponse function. F u r t h e r m o r e , in Chapter 7, we will see that E q . 1) can be uniquely inverted to give the impulse response directly from the fre­ quency response; the system can be completely specified by either. These are the two pictures promised in the introduction: any LSI system (or sequence) can be completely described by its time representation h or equivalently by its frequency representation H(oo).

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