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Digital Signal Processing: A Practical Guide (Part 3)

Authored on: Feb 3, 2010 by Michael Parker

Technical Paper / Book Excerpt

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This book is intended for those who work in or provide components for industries that use digital signal processing (DSP). There is a wide variety of industries that utilize this technology. While the engineers who implement applications using DSP must be very familiar with the technology, there are many others who can benefit from a basic knowledge of its' fundamental principals, which is the goal of this book—to provide a basic tutorial on DSP.

Editor's Note: This extract is Part 3 of a seven-part series from this book.

Read Part 1.
Read Part 2.
Read Part 3.
Read Part 4.
Read Part 5.
Read Part 6.
Read Part 7.

Reproduced from the book Digital Signal Processing: A Practical Guide Copyright © 2010 Newnes Press. Reproduced by permission of Newnes Press. Written permission from Newnes Press is required for all other uses.

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andyx Posted Aug 12, 2010

part 6 is labelled part 7 and I cannot find part 7.

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dspgirl Posted Sep 20, 2010

I can't download part 3.

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dtrainor Posted Sep 21, 2010

Download link for part 3 still seems to be broken.

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Convergent Posted Sep 21, 2010

I can not download Part 3

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fender Posted Dec 14, 2010

I can't download part 3 and part 7

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DSPer Posted Jan 2, 2014

Hi. There seems to be some wildly incorrect misconceptions in Chapter 10 (DFT, FFT). On page 6 the author refers to "a signal's frequency response." Signals do not have frequency responses. A digital filter (or perhaps a digital differentiator) has a frequency response but a signal does not. A signal has a spectrum, not a frequency response. On page 11 the author gives an equation for the frequency response of a digital filter, H(w). But the 2nd equation, on page 11, of the filter's H(w) freq response contains a signal variable (x-sub-i). This implies that a digital filter's freq response depends on the filter's input signal. That is terribly incorrect. A digital filter's freq response depends on the filter's coefficients, not the filter's input signal. On page 12 the text claims the spectrum (X-sub-k) of a time sequence (x-sub-i), is: X-sub-k = H(2*pi*k/N). This implies that the x-sub-i signal sequence's spectrum depends, somehow, on the H(w) freq response of some mysterious filter. I can't explain why in the world would a freq response variable H() be included anywhere in an equation defining the DFT. I'm sorry to say, but I'm compelled to warn DSP beginners, "reader beware." [-Rick-]

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