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Notes on Digital Signal Processing Practical Recipes for
Design Analysis and Implementation 1st Edition C.
Britton Rorabaugh Digital Instant Download
Author(s): C. Britton Rorabaugh
ISBN(s): 9780131583344, 0131583344
Edition: 1
File Details: PDF, 28.35 MB
Year: 2010
Language: english
Notes on Digital Signal Processing
This page intentionally left blank
Notes on Digital Signal
Processing
Practical Recipes for Design, Analysis,
and Implementation
C. Britton Rorabaugh
The author and publisher have taken care in the preparation of this book, but make no expressed
or implied warranty of any kind and assume no responsibility for errors or omissions. No liability
is assumed for incidental or consequential damages in connection with or arising out of the use of
the information or programs contained herein.
The publisher offers excellent discounts on this book when ordered in quantity for bulk purchases
or special sales, which may include electronic versions and/or custom covers and content
particular to your business, training goals, marketing focus, and branding interests. For more
information, please contact:
International Sales
[email protected]
All rights reserved. Printed in the United States of America. This publication is protected by
copyright, and permission must be obtained from the publisher prior to any prohibited reproduction,
storage in a retrieval system, or transmission in any form or by any means, electronic, mechanical,
photocopying, recording, or likewise. For information regarding permissions, write to:
ISBN-13: 978-0-13-158334-4
ISBN-10: 0-13-158334-4
Text printed in the United States on recycled paper at Edwards Brothers in Ann Arbor, Michigan
First printing, November 2010
To Joyce
This page intentionally left blank
Contents
Preface .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. . xi
About the Author .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. . xiii
vii
viii Contents
S tandard advice for writing a preface tells the author to begin by answering the
question, “Why did you write this book?” The published answers almost always
include an explanation of how something is still missing in the already vast body of
existing literature, and how the book in question represents a valiant attempt to fill the
void at least partially.
This book is no exception. There still is a dearth of good collections of step-by-
step procedures, or recipes, for design and implementation of anything beyond just
the most elementary DSP procedures. This book is an attempt to fill this void—at least
partially. However, the tagline for this book is most definitely not meant to be, “Get
a result without really gaining much understanding along the way.” Here, the focus is
clearly on the recipes, but supporting explanations and mathematical material are also
provided. This supporting material is set off in such a way so that it is easily bypassed
if the reader so desires.
This book provides an opportunity to delve deeper into the nuances of certain in-
teresting topics within DSP. A good alternative title might be Exploring the Nooks and
Crannies of Digital Signal Processing. As with all books, every reader will not resonate
with every topic, but I’m confident that each reader will share an interest in a large sub-
set of the topics presented.
Note 1, Navigating the DSP Landscape, provides diagrams that map the relationships
among all the book’s various topics. One diagram is dedicated to processing techniques
that operate on real-valued digital signals to modify in some way the properties of those
signals while leaving their fundamental real-valued and digital natures intact. A sec-
ond diagram is dedicated to processing techniques that are concerned primarily with
conversion between real-valued digital signals and other entities such as analog signals,
complex-valued signals, and estimated spectra.
Many of the Notes include examples that demonstrate an actual application of the
technique being presented. Most sections use Matlab tools for routine tasks such as
designing the digital filters that are used in the reference designs. When appropriate, the
xi
xii Preface
use of these tools is discussed in the text. The results provided in Note 66, Generating I and Q
Channels Digitally: Generalization of Rader’s Approach, were generated by a modified version of
the PracSim simulation package that is described in Simulating Wireless Communication Sys-
tems (Prentice Hall, 2004). However, the filter coefficients used in the simulation were generated
using Matlab. The examples make heavy use of Matlab as a convenience. However, it is not my
intent to make this a Matlab “workbook” with projects and exercises that require the reader to
use Matlab, because I want the book to remain useful and attractive to readers who do not have
access to Matlab. The m-files for the Matlab programs discussed in the book, as well as for
programs used to generate some of the illustrations, can be found at the website www.informit.
com/ph.
This book is not the best choice for a first book from which to learn DSP if you’re start-
ing from scratch. For this task, I recommend Understanding Digital Signal Processing by Rich-
ard Lyons (Prentice Hall, 2004). This book is, however, a good N+1st book for anyone—from
novice to expert—with an interest in DSP. Its contents comprise an assortment of interesting
tidbits, unique insights, alternative viewpoints, and rarely published techniques. The following
are some examples.
t The set of five techniques for generating analytic signals presented in Notes 60 through 64 do
not appear together in any single text.
t The visualization techniques used in Note 22 probably are not discussed anywhere else, be-
cause I came up with them while writing this book. These techniques follow directly from
first principles, but I’ve never seen them explicitly presented elsewhere.
t Natural sampling, as discussed in Note 6, usually can be found only in older texts that cover
traditional (that is, analog) communication theory.
My overarching goal was to write an easy-to-read book loaded with easy-to-access informa-
tion and easy-to-use recipes. I hope I have succeeded.
About the Author
xiii
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Note 1
Window
Functions
N21 - N25
FIR Filters
optimal designs
N32-N34, N37 FIR Filters
designs based
on windows
N35, N36
Real-
Valued
Digital
Signals z Transform
IIR Filters used in IIR
Decimation derived from design &
to decrease analog designs analysis
sample rate
N49 - N51 N44 - N48
N52 - N54
Interpolation
to increase
sample rate
N55 - N57 Analog Filter
Designs
N38 - N43
Figure 1.1 Processing techniques that modify the properties of real-valued digital signals. The
numbers “Nnn” indicate the Notes in which each technique is discussed. Solid paths indicate “run-
time” data connections. Dashed paths indicate “design-time” connections.
1-1
1-2 Notes on Digital Signal Processing
for organizational purposes. Figure 1.1 shows those most of these techniques are easily extended to cor-
techniques that are concerned primarily with op- responding complex cases. Figure 1.2 shows those
erating on real-valued digital signals to modify in techniques that are concerned primarily with con-
some way the properties of those signals while leav- version between real-valued digital signals and
ing their fundamental real-valued and digital na- other entities such as analog signals, complex-
tures intact. Given that complex-valued signals are valued signals, and spectrum estimates.
really just quadrature pairs of real-valued signals,
Natural Sampling
model for analog
signal commutation
N6
Sampling
Analog models for Parametric Modeling of
Signals A/D conversion Discrete-Time Signals
N2 - N5, N58 N67 - N73
Instantaneous Window
Real-
Sampling Functions
Valued
model for Digital N21 - N25
D/A conversion Signals
N7, N8
Generating
Complex & Improved
DFT Sample Periodogram
Analytic Signals Estimate
Spectrum Techniques
N13 - N19 of Signal’s
N60 - N66
N26 - N31 Spectrum
Complex-
Parametric Modeling of
Valued
Discrete-Time Signals
Digital
Signals N67 - N73
Figure 1.2 Processing techniques that convert real-valued digital signals to or from other things such as analog signals,
complex-valued digital signals, or spectral estimates. The numbers “Nnn” indicate the Notes in which each technique is
discussed. Solid paths indicate “run-time” data connections. Dashed paths indicate “design-time” connections.
Note 2
This note discusses the difference between implicit and explicit sam- 2.2 Explicit Sampling Techniques
pling. It introduces three different mathematical models of explicit
sampling techniques—ideal sampling, natural sampling, and in- Unlike implict sampling, in which samples are trig-
stantaneous sampling. gered by some aspect of signal behavior, in explicit
sampling, signal values are measured at specified
x(t) X(f )
t f
2-1
2-2 Notes on Digital Signal Processing
constant long enough to create flat-topped Spectral images, with each image
scaled by a factor that is constant
sample pulses. The output of a digital-to- Non zero width samples with
over the image but varies image
to image
time varying amplitudes that
analog converter (DAC) can be modeled follow the contours of the
analog signal
as the output of an instantaneous sampling
process, often as the limiting case in which
the sample width equals the sampling in-
terval. As discussed in Note 7, the results
of the instantaneous sampling model play
a key role in the specification of the analog
filter used to smooth the DAC output.
Reference
1. H. D. Lüke, “The Origins of the Sampling
Theorem,” IEEE Communications, April 1999,
pp. 106–108.
Note 3
Ideal Sampling
x(t ) x[n]
x(2T ) x[2]
13T 13
t n
T 1 2
2T
x(13T ) x[13]
a b
p(t ) T
produced by this model shows that sampling in the thereby causing Eq. (3.2) to become
time domain can be expected to create periodic
∞
⎛ m⎞
images of the original signal’s spectrum in the fre-
quency domain. Physical measurements confirm
Xs ( f ) = ∑
m = −∞
X⎜ f − ⎟
⎝ T⎠
that creation of images predicted by the mathemat-
ical model does in fact occur in the real world. The creation of the images is the important idea—
Consider an arbitrary continuous-time signal, the presence or absence of the 1/T scaling factor is
x(t), having a spectrum that is bandlimited to the usually lost in the overall scaling strategy of most
frequency range, ±fH, as shown in Figure 3.3. The real-world sampling implementations.
spectrum for the ideally sampled version of the sig-
nal will consist of copies or images of the spectrum 3.1 Aliasing
for x(t) periodically repeated along the frequency Figure 3.4 shows the case in which fs > 2fH . The
axis with a center-to-center spacing that is equal to original baseband spectrum and the images are all
the sampling rate, as shown in Figure 3.4. The im- distinct, and theoretically the original signal could
age corresponding to the original signal’s spectrum be recovered exactly via ideal lowpass filtering to
is often called the baseband. completely remove all of the images and leave the
The sample signal’s spectrum can be expressed undistorted baseband spectrum. The sampling the-
in terms of the original signal’s spectrum, X( f ), as orem, presented in the next section, provides a for-
mal statement of this property, as well as a formula
1 ∞ ⎛ m⎞
Xs ( f ) = ∑ X⎜ f − ⎟
T m = −∞ ⎝ T⎠
for reconstructing the original signal from the ideal
samples.
(3.2)
∞ In the case where fs < 2fH , the spectral images
= fs ∑ X ( f − mf s )
m = −∞
overlap, as shown in Figure 3.5. This overlap cre-
ates the condition known as aliasing. Components
A few texts (such as [2]) redefine the sampling rela- in the original signal at frequencies greater than
tion of Eq. (3.1) to be fs /2 will appear at frequencies below fs /2 after the
x[n] ← Tx(nT )
X S (f )
Baseband
X (f ) Images Images
A
A T
f
2fs fs fH 0 fH fs 2fs
f
fs fH fs
fH fH 2
fs fH fs fH
Figure 3.3 Bandlimited spectrum of an Figure 3.4 Frequency relationships for sampling-induced
arbitrary analog signal images when fs > 2fH
Ideal Sampling 3-3
X S (f )
sampling is performed. As depicted in Figure 3.6,
the result is as though the portion of the spectrum
above fs /2 folds over the line at f = fs /2 and adds
into the spectrum immediately below fs /2. Based f
2fs fH 0 fH fs 2fs
on this viewpoint, fs /2 is sometimes called the fold- fs
fs fH
ing frequency. fs fH fs fH
Note 4 shows how the concept of ideal sampling is Figure 3.5 Frequency relationships for sampling-induced
applied to real-world signals in practical applications. images when fs < 2fH
f = − fs 2 f = fs 2
This note shows how theoretical results from Note 3 are used
Key Concept 4.1
to develop a practical sampling strategy that minimizes the
effects of aliasing. Aliasing
* n the discussions of aliasing in Note 3, The sampling process creates periodic images of the origi-
nal signal’s spectrum in the frequency domain. Overlap
the frequency fH is portrayed as an abso-
between these images creates a condition called aliasing,
lute upper frequency. Prior to sampling, the in which components in the original signal at frequencies
signal of interest has zero spectral content greater than half the sampling rate will appear, or alias, at
at frequencies greater than fH . Under these frequencies below half the sampling rate in the spectrum of
conditions, sampling at rates greater than the sampled signal.
2fH would result in no aliasing. However,
Spectrum of
this ideal situation is impossible to achieve analog signal
in practical applications, where there will al-
ways be some aliasing. In many cases, even
to achieve extremely low levels of aliasing,
Spectrum of
fH would have to be set so high that, in most fH fH sampled signal
cases, sampling at a rate of 2fH would be pro- Sa
m Images
hibitively difficult and expensive. In practi- pl Baseband Images
in
g
cal situations, rather than trying to avoid all
aliasing, the design goal is to minimize the
effects of aliasing while recognizing that they f
can not be completely eliminated. Overlap
Baseband and
creates
Practical sampling is performed at a rate aliasing. first image cross
at half the
Center-to-center spacing
greater than 2fH, where the signal of inter- of images equals the sampling frequency.
sampling frequency.
est is known to have negligible (rather than
zero) spectral content above some upper In practical systems, a lowpass anti-aliasing filter is typically
frequency, fH . The definition of “negligible” used prior to the sampling operation in order to attenuate
varies based on the particular application. components at frequencies greater than half the sampling
In some cases, fH might be set so restrictively rate and thereby minimize the effects of aliasing.
that less than 0.01 percent of the signal’s en-
ergy is at frequencies greater than fH . In less
demanding applications in which the analog-
to-digital converter (ADC) cost may need to
be kept low, fH may be set lower to allow up
to 5 percent of the signal’s energy to be at fre-
quencies greater than fH .
In most applications, the signal is passed
through an anti-aliasing filter prior to being
digitized. The purpose of that filter is to ensure
4-1
Practical Application of Ideal Sampling 4-2
that aliasing is held to tolerably low levels. A t The signal of interest may contain spuri-
value for fH can be chosen based on the prop- ous self-interference caused by nonlinear
erties of the signal of interest and the require- processing in a mixer or a saturated am-
ments of the application, but in real-world plifier prior to sampling. Because nonlin-
situations, it is generally not possible to guar- ear processing creates components at new
antee that the chosen value for fH bounds the frequencies, some of this self-interference
frequency extent of the actual signal that is may occur at frequencies above fH .
presented to the ADC. There are several rea-
The selection of the sampling rate, fs, and
sons why this is so:
the design of the anti-aliasing filter are co-
t The signal of interest may be contaminat- ordinated, observing the guidelines called
ed with wideband additive noise. out in Design Strategy 4.1, so that the filter
produces minimal attenuation or distortion
t The signal of interest may be contami-
for frequencies below fH , but provides severe
nated with an unanticipated interfering
attenuation for all frequencies above fs/2.
signal having a bandwidth that extends
beyond fH .
References
1. A.V. Oppenheim and R.W. Schafer, Discrete-
Time Signal Processing, Prentice Hall, 1989.
2. R.A. Roberts and C.T. Mullis, Digital Signal
Processing, Addison-Wesley, 1987.
Note 5
Sampling Theorem
stated as a fact and accepted without mathe- If the spectrum, X( f ), of a function, x(t ), van-
matical justification. To generate mathemati- ishes beyond an upper frequency of fH Hz,
then x(t ) can be completely determined by its
cal support for the existence of these images,
values at uniform intervals of less than 1/(2 fH).
a more complicated mathematical model of
the sampling process must be adopted. In ad- Reconstruction Formula
dition to correctly predicting the appearance
If a function is sampled at a rate, fs = T -1, that
of spectral images, such a model can also satisfies the sampling theorem, the original
be used to derive the discrete-time Fourier function, x(t ), can be reconstructed from the
transform (DTFT) and the discrete Fourier samples as
transform (DFT) from the “usual” contin-
⎡ ⎛t ⎞⎤
uous-time Fourier transform (CTFT). (See ∞
sin ⎢π ⎜ − n ⎟ ⎥
⎝T ⎠⎦
Math Boxes 12.1 and 13.2.) x(t ) = ∑ x(nT ) ⎣
⎛t ⎞
If the result of ideal sampling is considered n =− ∞
π ⎜ −n⎟
in the continuous-time domain, each sample ⎝ T ⎠ (MB 5.1)
∞
⎛t ⎞
exists for a single instant on the continuous- = ∑ x(nT ) sincπ ⎜ − n ⎟
time axis, and the value of the sampled wave- n =− ∞ ⎝T ⎠
form is zero between sample instants. This
differs from the approach corresponding to
Eq. (3.1) in Note 3 in which the result of ideal
sampling is viewed in the discrete-time do-
main where the result is defined only at the
sample instants—between samples the result
is not zero, it is simply not defined at all.
An ideal sampling process that produces
5.1 Dirac Delta Function
a continuous-time result can be constructed
using a generalized function: The impulse is As far as mathematicians are concerned,
sometimes called the delta function (due to the Dirac delta function is not a function in
its usual notation) or the Dirac delta func- the usual sense, and its use in engineering is
tion, in honor of the English physicist Paul sometimes controversial. Nevertheless, the
Dirac (1902–1984), who made extensive use delta function provides a convenient way to
of impulse functions in his work on quantum relate the spectra of ideal sampling, natural
mechanics.
5-1
Delta Functions and the Sampling Theorem 5-2
to unity. ∫ δ (t − t ) f (t )dt = f (t )
−∞
0 0 (MB 5.3)
The impulse function is usually denoted
as δ(t) and is depicted as a vertical arrow at 1
the origin. The rigorous definition of δ(t), δ (at ) = δ (t ) (MB 5.4)
a
introduced in 1950 by Laurent Schwartz
[1], rejects the notion that the impulse is
δ (t − t 0 ) f (t ) = δ (t − t 0 ) f (t 0 ) (MB 5.5)
an ordinary function and instead defines
it as a distribution. The rigorous definition
FT
notwithstanding, most engineers are more δ (t ) ←→ 1 (MB 5.6)
comfortable defining the impulse in an op-
erational sense. Specifically, δ(t) is taken as
that function which exhibits the so-called
sifting property:
∞
∫ δ (t ) f (t )dt = f (0)
−∞
(5.1)
∑ δ (t − kT )
k = −∞
(5.2)
Other texts refer to this function as the shah
function and denote it using the Cyrillic letter
shah, which somewhat resembles the graph
The DSP literature is about evenly split be-
of the function
tween two different names and notations for
the function represented by Eq. (5.2). Many ∞
T (t )x ( ) ∑ kt (t kT )
(MB 5.7)
x s (t ) = x a (t )δ T (t ) k = −∞
where ∞ ∞
δT (t ) ∑ δ (t − nT )
n=−∞
δ T (t ) ⊗ x(t ) = ∑ x(t − kT )
k = −∞
(MB 5.8)
∞
δ ωS (ω ) ∑ δ (ω − mω )
m = −∞
s
ωs ∞ T
X s (ω ) ∑ δ (ω − mωs )
2 πm = −∞ Figure 5.1 Continuous-to-discrete converter with two
separate processing steps
Delta Functions and the Sampling Theorem 5-4
References
1. L. Schwartz, Théorie des distributions, Herman &
Cie, Paris, 1950.
2. R. N. Bracewell, The Fourier Transform and Its
Applications, 3rd ed., McGraw-Hill, 2000.
3. H. D. Lüke, “The Origins of the Sampling
Theorem,” IEEE Communications, April 1999,
pp. 106–108.
4. H. Nyquist, “Certain topics in telegraph
transmission theory,” Trans. AIEE, vol. 47, Apr.
1928, pp. 617–644, republished in Proc. IEEE,
vol. 90, no. 2, Feb. 2002, pp. 280–305.
5. C. E. Shannon, “Communications in the
presence of noise,” Proc. IRE, vol. 37, 1949, pp.
10–21.
6. V. A. Kotelnikov, “O propusknoj sposobnosti
‘efira’ i provoloki v elektrosvjazi,” (“On the
transmission capacity of ‘ether’ and wire in
electro-communications”), First All-Union
Conference on Questions of Communication,
Jan. 14, 1933.
7. H. Raabe, “Untersuchungen an der
wechselzeitigen Mehrfachubertragung
(Multiplexubertragung),” Elektrische
Nachrichtentechnik, vol. 16, 1939, pp. 213–228.
8. W. R. Bennett, “Time division multiplex
systems,” Bell Syst. Tech. J., vol. 20, 1941, pp.
199–221.
Note 6
Natural Sampling
6-1
Natural Sampling 6-2
Time Frequency
T1 Domain Domain F1
Rectangular pulse
IJ
CTFT Spectrum of pulse
°1, t
p(t ) ® 2 P(f ) IJ sinc Ⱥ (f IJ )
°¯0, otherwise
T2 F2
Multiply
Convolve
F3
T3
CTFT Spectrum of pulse train
Rectangular pulse train P(f ) ƩT (f )
F4
T4
CTFT Spectrum of original signal
Original signal X (f )
x(t )
Frequency Convolve
Multiply Time
Domain Domain
T5 F5
Figure 6.2 Natural sampling is mathematically equivalent to multiplying the original spectrum with a train of rectangular
sampling pulses. Therefore, the spectrum of a naturally sampled signal can be determined by convolving the original
signal’s spectrum with the spectrum of the train of sampling pulses. (The multiply operation that creates spectrum F3
is not mathematically rigorous, but it is consistent with typical engineering use of Dirac delta functions as discussed in
Note 5. “CTFT” indicates the continuous-time Fourier transform.)
τ
T
1 1
τ τ
1
fs T
1
fs T
0
Figure 6.3 Magnitude spectrum of rectangular pulse train (corresponds to block F3
in Figure 6.2)
6-3 Notes on Digital Signal Processing
The magnitude spectrum for a naturally sam- corresponding to the center of the image. In other
pled sinusoid is shown in Figure 6.4. The spacing words, the image centered at f = nfs is scaled by
of the images is equal to the reciprocal of the sam- τT -1|sincπ( nfsτ)|. This factor changes from image to
pling interval, and each image is amplitude scaled image, but remains constant across the width of each
by the value of τT -1|sincπ( fτ)| at the frequency image.
f =0
IJ
T
IJ
sinc Ⱥ (f IJ )
T
IJ 1 fx Tx 1 IJ 1
fs T 1
fs T 1
Reference
1. W. R. Bennett, “Time division multiplex
systems,” Bell Syst. Tech. J., vol. 20, 1941,
pp. 199–221.
Note 7
Instantaneous Sampling
7-1
7-2 Notes on Digital Signal Processing
Time Frequency
T1 Domain Domain F1
Multiply X (f )
T3
Convolve
Ideally sampled signal CTFT
F3
Spectrum of pulse
P(f ) IJ sinc Ⱥ (f IJ )
Convolve
Multiply
Time Frequency
Domain Domain
T5 F5
Figure 7.2 Instantaneous sampling is mathematically equivalent to convolving a single rectangular sam-
pling pulse with an ideally sampled version of the original signal. Therefore, the spectrum of an instanta-
neously sampled signal can be determined by multiplying the ideally sampled signal’s spectrum with the
spectrum of a single sample pulse. (“CTFT” indicates the continuous-time Fourier transform.)
Instantaneous Sampling 7-3
IJ
T
f
1
IJ 1 IJ
f=0 1
fs T
1
TP
Figure 7.4 Magnitude spectrum of instantaneously sampled sinusoid (corresponds to block F5 in Figure 7.2)
Reference
1. W. R. Bennett, “Time division multiplex
systems,” Bell Syst. Tech. J., vol. 20, 1941,
pp. 199–221.
Note 8
Digital Analog
samples Digital-to- output
Reconstruction
analog
filter
converter
The spectrum of the DAC output signal is distorted by a sinc envelope, with the nulls
of the sinc envelope falling in the center of each image other than the baseband.
Baseband
Images Images
Spectrum of
digital signal
f
−T −1 0 T −1
Re
con
str
uc tio
n
Higher frequencies
in baseband image Spectrum of
are attenuated by DAC output
sinc envelope. signal
Images are
sinc envelope
severely distorted
by sinc envelope.
f
−2T −1 −T −1 0 T −1 2T −1
The reconstruction filter must remove the images and correct for the sinc distor-
tion in the baseband. The close spacing between the baseband and the adjacent
images can make the design of this filter relatively difficult.
8-1
T he mathematical signal reconstruction tech-
niques presented in Note 5 do have practical
uses, but these techniques are not really suited for
interval. The DAC output is a special case in that the
DAC typically holds each output value for an entire
sampling interval, thereby generating a “stair-step”
converting a sequence of digital signal values back signal, such as the one shown in Figure 8.2, in which
into a continuously time-varying voltage that can the sample width equals the sampling interval.
be used to drive a speaker or a pair of headphones. The spectrum of the DAC output contains im-
Reconstruction of a physical analog signal is usually ages and is multiplied by a (sin x) /x envelope, as dis-
accomplished using a digital-to-analog converter cussed in Note 7. However, the illustrations in Note
(DAC). The input to the DAC is a sequence of digi- 7 depict the case in which the sampling interval, T, is
tal words, and the output is a time-varying voltage several times larger than the sample width, τ. When
that is proportional to the sequence of values rep- the sample width is equal or nearly equal to the sam-
resented by the input words. Each output voltage is pling interval, the distortion effects caused by the
held constant until the input value changes. (sin x) /x envelope are much more severe.
The output of the DAC can be viewed as a special Assume that the ideally sampled signal inside
case of the instantaneously sampled signal described the processing computer has a simple trapezoi-
in Note 7. In Note 7, each voltage pulse is depicted dal baseband spectrum, as depicted in Figure 8.3.
as being significantly narrower than the sampling The corresponding DAC output has a spectrum,
digital stepped
smooth
signal analog
analog T IJ
values digital -to- waveform
analog signal
analog
reconstruction
converter
filter
(DAC )
Figure 8.2 Output of DAC modeled as the limiting case of
Figure 8.1 Block diagram of the signal reconstruction process instantaneous sampling
b b b a b b b
f
−1 −1 −1 −1
−2T −T 0 T 2T
Figure 8.3 Idealized trapezoidal spectrum for a sampled signal, showing (a) the baseband
spectrum of the original signal, and (b) spectral images created by the sampling process
8-2
8-3 Notes on Digital Signal Processing
as shown in Figure 8.4, with the main lobe of the output signal in a way that separates the spectral im-
(sin x) /x envelope having a null-to-null width of ages, as shown in Figure 8.5. In many practical sys-
just twice the sampling rate. All of the images are tems, the sample rate is already significantly higher
severely distorted by the side lobes. Because it oc- than twice the signal bandwidth, which causes the
cupies such a large portion of the main lobe, the images to be spread farther apart and simplifies the
baseband component of the spectrum also experi- filter design task. In other applications, such as au-
ences distortion from the (sin x) /x envelope. dio CD players, it is necessary to take explicit steps
The reconstruction filter that follows the DAC to make the reconstruction problem easier to man-
needs to have both a stopband response that se- age by increasing the Nyquist bandwidth (which is
verely attenuates the spectral images and a pass- equal to one-half the sample rate) without increas-
band response that is designed to correct the ing the utilized bandwidth, as shown in Figure 8.5.
(sin x) /x distortion present on the baseband spec- Audio CD players use a sample rate of 44.1 kHz
trum. When the sample rate equals exactly twice to support a utilized bandwidth of about 20 kHz,
the highest frequency component in the signal’s so there would be a gap of about 4 kHz between
original spectrum, the signal is said to be critically images in the DAC output. Design of an acceptable
sampled. When the signal is critically sampled, as reconstruction filter is not impossible, and there
in the case depicted in Figure 8.4, the images are were many early CD players built that used direct
close together, thus making it almost impossible to reconstruction of the 44.1 kHz sample stream.
design a filter that can both remove the images and However, CD players are consumer products, and
compensate for (sin x) /x distortion. Most filter de- there is constant pressure to make them smaller,
signs that can remove the images under these con- lighter, cheaper, and better-sounding. Increasing
ditions are likely to introduce phase distortion into the CD sample rate would make it easier to build
the baseband signal. cheaper reconstruction filters with good perfor-
Design and implementation of the reconstruc- mance, but an increased sample rate for the re-
tion filter can be made easier by modifying the DAC corded signal would require more samples for each
f
1 1 1
2T T 0 T 1 2T
Figure 8.4 Spectrum at output of DAC (solid trace). Shown for comparison are the
undistorted images of the trapezoidal spectrum (dashed trace) and sinc envelope
(dotted trace).
Reconstructing Physical Signals 8-4
second of audio, thus resulting in reduced playing coincides with a more central, flatter portion of the
time for a disc having a given total bit capacity. main lobe, thus lessening the severity of the distor-
Most newer CD players advertise 4⫻, 8⫻, or tion that the reconstruction filter must correct.
even 16⫻ oversampling, but the increased sample If oversampling is carried throughout the digi-
rate is not used for the recorded signal. Instead, the tal processing and if the utilized bandwidth has
digital signal is interpolated to create new sample not already been limited by this processing, then
values in between the sample values that are actu- it may be prudent to perform digital filtering to
ally read from the disc. This type of oversampling limit the utilized bandwidth just prior to sending
does not increase the utilized bandwidth; that the signal to the DAC. Because of the increased
is, it does not increase the bandwidth of the re- Nyquist bandwidth (due to the higher sampling
corded signal or the reconstructed signal. What it rate), earlier processing may have inadvertantly in-
does do is increase the Nyquist bandwidth, which troduced components outside of the intended sig-
moves the spectral images farther apart so that it nal bandwidth and thereby increased the utilized
becomes relatively easy to design a reconstruction bandwidth, reducing the effectiveness of the over-
filter that will reject all of the non-baseband im- sampling strategy. On the other hand, if the over-
ages while simultaneously compensating for the sampling is introduced by an interpolation process
(sin x) /x distortion in the baseband spectral com- just prior to sending the signal to the DAC, there
ponent. As shown in Figure 8.5, this oversampling is no opportunity to inadvertently increase the uti-
will also have the effect of widening the main lobe lized bandwidth, and additional filtering would not
of the sinc function so that the utilized bandwidth be necessary.
Utilized bandwidth
Nyquist bandwidth
f
T 1 fS 0 fS T 1
2 2
Figure 8.5 Spectrum at output of DAC with 2× oversampling
Note 9
Fourier Series
components over a range of different fre- t Signal: periodic function of continuous
quencies. The collection of amplitudes time
and phases for the sinusoids needed to
t Spectrum: nonperiodic function of
completely represent the signal is usually discrete frequency
called the spectrum of the signal. Depend-
ing upon the nature of the signal being t Detailed in Note 10
analyzed, the spectrum can span either
Fourier Transform
a continuum of frequencies or a count-
t Signal: nonperiodic function of continu-
able (but possibly infinite) set of discrete ous time
frequencies.
t Spectrum: nonperiodic function of con-
tinuous frequency
9.1 Fourier Series
t Detailed in Note 11
Periodic continuous-time signals have fi-
nite power (but infinite energy), and can Discrete-Time Fourier Transform
be analyzed using the Fourier series (FS) t Signal: nonperiodic function of discrete
defined by time
9-1
Overview of Fourier Analysis 9-2
II
Avicenna’s Dream
But all these books—for him—were living thoughts,
Clues to the darker Book of Nature’s law;
For, when he climbed, a goat-foot boy, in Spring
Up through the savage Hissar range, he saw
A hundred gorges thundering at his feet
With snow-fed cataracts; torrents whose fierce flight
Uprooted forests, tore great boulders down,
Ground the huge rocks together; and every year
Channelled raw gullies and swept old scars away;
So that the wildered eagle beating up
To seek his last year’s eyry, found that all
Was new and strange; and even the tuft of pines
That used to guide him to his last year’s nest
Had vanished from the crags he knew no more.
Giulio.
The Stranger.
The Stranger.
Giulio.
The Stranger.
A jewel of a sort;
But it may take a thousand years to trace it
Back to its rightful owner.
Giulio (laughing).
Giulio.
The Stranger.
Giulio.
The Stranger.
Giulio.
Obviously enough,
The sea being where it is, it was the Flood
That left them here.
The Stranger.
Giulio.
I must confess
I always feel a pang, sir, when I see
A man of talent wasting his fine powers
On this blind road.
The Stranger.
Giulio.
The Stranger.
Giulio.
Genius leaps
Like lightning to that mark, sir, and can waive
These pains and labours.
The Stranger.
O, I have no doubt
That you are right. I speak with diffidence,
And as a mere spectator; one who likes
To know, and seizes on this happy chance
Of learning what an artist really thinks.
Giulio.
The Stranger.
Giulio.
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