Fourier Transforms and their Application to Pulse Amplitude Modulated Signals
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A long text by Phil Lucht of Rimrock Digital Technology, Salt Lake City, last updated July 23, 2013. It covers the Fourier integral, sine and cosine transforms, convolution, pulse trains and Fourier series, sampled signals, digital and Z transforms, and the discrete Fourier transform. Later chapters treat practical topics (FIR filters, D/A oversampling), dispersion relations, and power in pulse trains including autocorrelation and the Wiener-Khintchine theorem.
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1 Fourier Transforms and their Application to Pulse Amplitude Modulated Signals
Phil Lucht
Rimrock Digital Technology, Salt Lake City, Utah 84103
last update: July 23, 2013
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The table of contents has live links.
Overview an d Summary
........................................................................................................... .............. 5
Chapter 1: The Fourier Integral Transform and Related Topics ...................................................... 7
1. The Fourier Integral an d Sine/Cosine Transforms............................................................................ 7
(a) Pulses and Pulse Trains , Periodic and Aperiodic ........................................................................ 7
(b) The Fourier Integral Transform X( ω).......................................................................................... 7
(c) The Fourier Sine and Cosine Transforms X s(ω) and X c(ω)........................................................ 9
2. Proof of the Fourier Integral Transform ..................................................................................... ....12
3. The Convolution Theorem and its Derivation ................................................................................ 15
4. Applications of the Convolution Theorem..................................................................................... .19
(a) General case............................................................................................................... ................ 19
(b) A specific example: the RC filter section.................................................................................. 20
(c) An even simpler example: L t = (d/dt) ....................................................................................... 22
5. Fourier Integral Transform Conventions ...................................................................................... ..24
6. The Generalized Fourier Integral Transform and the Laplace Transform X(s).............................. 25
7. Reflection Rules............................................................................................................ .................. 28
8. Three simple examples of spectra............................................................................................ ....... 28
9. Spectrum of an isolated square pulse........................................................................................ ......31
10. The Area Rules and Parseval's Formulas..................................................................................... .34
11. Differentiation and Integration Rules with Examples................................................................... 35
12. Time translation x(t) causes phase on X(ω ).................................................................................. 37
13. Exponential Sum Rules................................................................................................................. 37
Chapter 2: Pulse Trains and th e Fourier Seri es Connection ............................................................ 40
14. The Spectrum of a Simple Pulse Train ....................................................................................... ..40
(a) Infinite Length Simple Pulse Train ......................................................................................... ..40
(b) Finite Length Simple Pulse Train........................................................................................... ...44
15. Connection with the tr aditional Fourier Series ............................................................................. 46
16. Fourier Series for a positiv e square wave pulse train ................................................................... 49
17. More about positive sq uare-wave pulse trains.............................................................................. 49
18. Non-positive pulse trains .................................................................................................. ............ 54
19. Biphase pulse and pulse train.............................................................................................. .......... 54
2 Chapter 3: Sampled Signal s and Digit al Transforms........................................................................ 57
20. Sampled Signals and their Image Spectra.................................................................................... .57
21. Digital Filters, Image Spectra and Group Delay........................................................................... 60
(a) A Digital Filter as an approximation to an Analog Filter.......................................................... 60
(b) Filter Group Delay ......................................................................................................... ........... 64
22. The Digital Fourier Transform X'( ω) Part I .................................................................................. 67
23. The Digital Fourier Transform X'( ω) Part II................................................................................. 71
(a) Relation between X'( ω) and X(ω ) ............................................................................................. 71
(b) Summary of the Dig ital Fourier Transform............................................................................... 73
24. The Z Transform X"(z) ...................................................................................................... ........... 75
(a) Convolution Theorem................................................................................................................ 77
(b) Unit Impulse............................................................................................................... ............... 77
(c) Time Translation ....................................................................................................................... 78
(d) Derivative Limit ........................................................................................................... ............. 78
(e) Digital RC filter.......................................................................................................... ............... 79
(f) Poles in H"(z) imply feedback and infinite impulse response (IIR) .......................................... 81
(g) The Digital RC filter revisited............................................................................................ ....... 84
(h) Other circuits............................................................................................................. ................ 85
(i) Z Transform Summary........................................................................................................ ....... 86
25. Amplitude Modulated Pulse Trains ........................................................................................... ...87
Example 1: A finite pulse train ............................................................................................... ....... 89
Example 2: The unit impulse and the sinc sum rule ...................................................................... 91
26. A simple application: Aperture Correction.................................................................................. .94
27. The Discrete Fourier Transform ............................................................................................. ......96
(a) The Discrete Fourier Transform for a Simple Pulse Train x(t) ................................................. 96
(b) Proof of the Discrete Fourier Transform for a Simple Pulse Train x(t) .................................... 99
(c) The Discrete Fourier Transform for an Arbitrary Pulse .......................................................... 101
(d) Comments on the Discret e Fourier Transform........................................................................ 103
Chapter 4: Some Practical Topics ............................................................................................... ......107
28. Do FIR filters have linear phase?.......................................................................................... ......107
29. A Simple Digital Low-Pass Filter........................................................................................... ....110
30. Use of Oversampling in a D/A Convert er Design ...................................................................... 115
(a) A very simple D/A converter................................................................................................ ...115
(b) Oversampling just the D/A converter...................................................................................... 117
(c) Add zero-stuffing to reduce aperture....................................................................................... 118
(d) Add an ω 1/2 digital low-pass in terpolation filter..................................................................... 120
Chapter 5: Some Theoretical Topics............................................................................................. ....125
31. Spectral Dispersion Relations .............................................................................................. ....... 125
(a) A simple integral equation for X(ω ) analytic in the upper half plane ..................................... 125
(b) A simple integral equation for X( ω) analytic in the lower half plane ..................................... 126
(c) Dispersion Relations for X( ω)................................................................................................ 128
(d) Dispersion Relations for γ(ω).................................................................................................. 129
(e) Dispersion and Attenuation ................................................................................................. ....131
(f) Application to coaxial cable............................................................................................... ......132
(g) The Dispersion Relation expressed in terms of the Hilbert Transform................................... 133
3 Chapter 6: Power in Pulse Trains ............................................................................................... ......135
32. The Autocorrelation Function..................................................................................................... 135
(a) Autocorrelation function for a Square Pulse ........................................................................... 135
(b) Energy, power and spectral energy density for a finite signal x(t).......................................... 136
(c) The Wiener-Khintchine theorem ............................................................................................. 136
(d) Verification of Wiener-Khintchine for a Square Pulse ........................................................... 137
(e) Cross-correlation, convol ution, and autocorrelation ............................................................... 138
(f) Z Transform Wiener-Khintchin e theorem for a Pulse Train.................................................... 139
33. Spectral power density of a Simple Pulse Train ......................................................................... 140
(a) Infinite Simple Pulse Train................................................................................................ ......140
(b) Finite Simple Pulse Train.................................................................................................. ......142
(c) Spectral Power Density of a Simple Pulse Train..................................................................... 144
(d) Average Power P of a Simple Pulse Train .............................................................................. 147
34. Spectral power density of a General Pulse Train........................................................................ 149
(a) General Pulse Train results and conn ection with the Autocorrelation Function ..................... 149
(b) Spectral power density fo r a General Pu lse Train ................................................................... 150
(c) Pulse Trains with Repeated Sequences.................................................................................... 152
35. Statistical Pulse Trains................................................................................................................ 163
(a) Spectral power density for a Statistical Pulse Train ................................................................ 163
(b) Infinite Statistical Pulse Trai n with Non-Correlated Coefficients........................................... 164
(c) Finite Statistical Pulse Train with Non-Correlated Coefficients ............................................. 166
(d) Statistical Non-Correlated Pul se Trains: Summary and Examples ........................................ 168
(e) A numerical example of a Statistical Pulse Train.................................................................... 169
(f) What role has the Autocorrelation Fu nction played in our development?............................... 172
(g) A paradox and its resolution............................................................................................... .....173
36. Application to some Standard Non-Correlated Pulse Train Types (Line Codes) ....................... 175
(a) Unipolar NRZ line code ..................................................................................................... .....175
(b) Bipolar NRZ line code ...................................................................................................... ......178
(c) Unipolar RZ line code ...................................................................................................... ....... 180
(d) Manchester line code....................................................................................................... ........ 183
(e) Noise, ISI and Eye Patterns ................................................................................................ .....185
37. The AMI Line Code.................................................................................................................... 186
38. The Change/Hold Line Code .................................................................................................. ....196
Example 1: Unipolar NRZI line code .......................................................................................... 206
Example 2: Bipolar NRZI line code ............................................................................................ 208
Appendix A: Delta Function Technology.......................................................................................... 209
(a) Models for Delta Functions and two derivations of (2.1)............................................................ 210
(b) Models for Periodic Delta Functions........................................................................................ ...213
(c) Derivation of (13.2) and (13.3)............................................................................................ ........ 220
(d) Undoing the limit N → ∞ : the meaning of δ (0) ......................................................................... 221
(e) The function Θ (a ≤ x ≤b) and related sums ................................................................................. 223
(f) The product of two delt a functions and more on δ(0).................................................................. 225
Appendix B: Derivation of a Certain Identity.................................................................................. 228
4 Appendix C: The Fourier Transform and its relation to the Hilbert Transform ......................... 230
(a) Fourier Transform Notations................................................................................................ ....... 230
(b) Principal Value Integral s and the Tick Notation ......................................................................... 233
(c) Example: f(u) = 1/u ....................................................................................................... ............. 234
(d) The Pole Avoidance Rule of Complex Integration ..................................................................... 236
(e) Example: f(u) = 1/(u±iε ).............................................................................................................. 238
(f) Example: f(u) = θ(u) using the Generalized Fourier Transform .................................................. 241
(g) Summary of Examples ........................................................................................................ ........ 242
(h) The Hilbert Transform and its relation to the Fourier Transform ............................................... 242
Appendix D: Probability Theory: how α and β are related to μ and σ......................................... 247
(a) Random Variables X and Y................................................................................................... ......247
(b) Application to the Pulse Train Ensemble .................................................................................... 249
(c) Main conclusions for the Pulse Train Ensemble ......................................................................... 250
Appendix E: Table of Transforms................................................................................................ .....252
Appendix F: The Spectrum and Power Density for Repeated-Sequence Pulse Trains ............... 256
(a) Calculation of X( ω) for a Pulse Train with a Repeated Sequence............................................... 257
(b) Calculation of P(ω) for a Pulse Train with a Repeated Sequence............................................... 258
(c) Calculation for an Ensemble of such Pu lse Trains subject to Certain Conditions....................... 261
(d) Limit as P → ∞ of the Ensemble Result...................................................................................... 265
(e) Calculation of P(ω) for a single Pulse Train subject to Certain Conditions............................... 266
(f) Summary of Results of (c) and (e) and an Example: The MLS Sequence.................................. 270
(g) Results for an A,B repeated sequence ....................................................................................... ..273
Detailed Summary of this Document................................................................................................. 276
Chapter 1: The Fourier Integral Tr ansform and Related Topics ( 34 p) ........................................... 276
Chapter 2: Pulse Trains and the Fourier Series Connection ............................................................. 277
Chapter 3: Sampled Signals a nd Digital Transforms (53 p) ............................................................. 278
Chapter 4: Some Practical Topics (19 p) ........................................................................................ ..281
Chapter 5: Some Theoretical Topics (10 p) ...................................................................................... 281
Chapter 6: Power in Pulse Trains (70 p) ....................................................................................... ...282
Appendix A: Delta Function Technology (19 p) .............................................................................. 284
Appendix B: Derivation of a Certain Identity (2p) ........................................................................... 285
Appendix C: The Fourier Transform and its relation to the Hilbert Transform (17 p)..................... 285
Appendix D: Probability Theory: how α and β are related to μ and σ (5 p) .................................... 286
Appendix E: Table of Transforms (4 p)............................................................................................ 286
Appendix F: The Spectrum and Power Density for Repeated-Sequence Pulse Trains.................... 287
References............................................................................................................................................ 288
Overview and Summary
5 Overview and Summary
The Fourier Integral Transform and its
various brethren play a major role in the scientific world. This
monograph develops the analog and digital theory of these transforms and applies that theory to pulse-
amplitude-modulated (PAM) signals referred to as "pulse trains" -- signals formed from a single arbitrary
pulse shape, x(t) = Σnynxpulse (t-nT1). Particular attention is paid to the spectral power density P(ω) of
pulse trains which form a statistical ensemble. When PAM signals are passed through a "linear time-
invariant" circuit or other apparatus, that appara tus may be viewed as a "filter" and, due to the
Convolution Theorem, the behavior of such a filter is most easily understood in the frequency domain.
All calculations are done in line for the reader to see amd perhaps critique. This is done to provide a clear
tracing path for the repair of errors, to demonstrate unusual techniques, and hopefully to remove some of
the mystery associated with Fourier Transform mathem atics. With just a few exceptions, every equation
appearing in this document is derived in this document.
The reader is assumed to be familiar with calculus and complex integration. Equations which are quotes of earlier equations have their equation numbers in italics.
A detailed summary of the material presented appears at the end of this document. Here we provide only
a brief chapter-level summary.
Chapter 1 (Sec 1-13) develops the basic theory of the Four ier Integral Transform and its Sine and Cosine
cousins. This Chapter forms the underpinning of all subsequent Chapters. The Convolution Theorem
receives special attention. The connection is made between the Laplace Transform and the "generalized"
Fourier Transform applied to causal functions. Various "rules" are de rived, and a connection is made
between filter spectra and time- domain Green's Functions.
Chapter 2 (Sec 14-19) examines the Fourier Integral Transform spectrum of a simple pulse train formed
from a general pulse shape. Consideration of pulse trains of infinite length leads to a derivation of the
Fourier Series Transform. The chapter concludes with a discussion of sample pulse trains formed from
box and bi-phase pulses. Chapter 3 (Sec 20-27) deals with various digital forms of the Fourier Transform and their corresponding
convolution theorems and applies these concepts to amplitude-modulated pulse trains (PAM signals) and
to digital filters. Topics include image spectra, aliasi ng and Nyquist rate, group de lay, FIR and IIR filters,
poles, and impulse response. The first digital transform is called the Digital Fourier Transform which is
an ω-domain version of the Z Transform whose variable is z = e
iωΔt . The Z Transform and Discrete
Fourier Transforms are then addressed for both period ic and aperiodic signals. A recurring example is a
simple RC filter section.
Chapter 4 (Sec 28-30) shows that symmetric FIR filters ha ve linear phase and thus constant group delay.
A specific brick wall digital filter is designed and then later used as an oversampling interpolation filter in
the design of a D/A converter output section. This system is then simulated with a simple Maple program.
Overview and Summary
6 Chapter 5 (Sec 31) explores the subject of disper sion relations for the spectral function X( ω) and for
other related functions treated as analyt ic functions of a complex variable.
Chapter 6 (Sec 32-38) derives expressions for the energy and power, and the spectral energy and power
densities of an amplitude-modulated pulse train. This work is carried out with a moderate amount of
mathematical rigor. Correlation and autocorrelation are mentioned. The notion of a statistical pulse train
is presented and the spectral power density of such pulse trains is established. These results are then
applied to various uncorre lated standard line codes including NRZ, RZ and Manchester. Two examples of
correlated pulse trains are then treated -- AMI and Change /Hold -- and the latter is then used to obtain
results for the NRZI line code. Appendix A discusses delta functions at a someone deeper and more practical level than is commonly
found in texts and on the web. Delta function models are constructed and many mathematical identities
are developed which find use in the main text.
Appendix B derives an obscure identity used in the Discrete Fourier Transform discussion of Section 27.
Appendix C further develops Fourier Integral Transform theory beyond the treatment of the main text.
Instead of using the notation f(t) and F( ω), here we use f(t) and f^( ω).
Appendix D reviews some basic probability theory and relates the α and β parameters of the uncorrelated
spectral power density formula to statistical properties of pulse train amplitudes. Appendix E provides a table of all transform pairs appear ing in this document and shows how they are
related to each other.
Appendix F explores the properties of infinite pulse trains which consist of a repeated subsequence of P
elements. A square wave and an MLS sequence are examples.
Chapter 1: The Fourier Integral Transform
7 Chapter 1: The Fourier Integral Transform and Related Topics
The purpose of this chapter is to dem
onstrate the u se of certain mathematical tools associated with the
Fourier Integral transform. Along the way simple examples are considered, with emphasis on the
particular example of an isolated square pulse in th e time domain. If we can make things work out for a
square pulse, we can presumably go on to harder problems with the same tools.
One normally analyzes a square-wave pulse train using a Fourier Series, since such a pulse train is a
periodic function. In Chapter 2 below, we will ma ke the connection between the conventional Fourier
Series, and our Fourier Integral approach.
1. The Fourier Integral and Sine/Cosine Transforms
(a) Pulses and Pulse Trains, Periodic and Aperiodic
Before startin
g, we need to define a few basic terms describing a function x(t) :
x(t) is a "pulse" if x(t) decays to zero at both t = ± ∞. Normally we think of a pulse as having a finite
extent, but it could be something li ke a Gaussian pulse with an infin ite extent. Such pulses fall into the
class of aperiodic (non-periodic) functions. Further restrictions on x(t) will be given below.
x(t) is a "simple pulse train" if it is constructed as a sum of identical pulses each of which is shifted by the
same constant amount T
1 from the previous pulse. Simple pulse tr ains which are infinite in extent then
fall into the class of periodic functions. If finite in extent, they are aperiodic.
x(t) is a "general pulse train" if the pulses are allowed to have arbitrarily different amplitudes. We refer to
this as an amplitude-modulated pulse train. If the pulse tr ain is infinite and the amplitude-modulation is a
repeating pattern (such as in a square wave), the pulse train is periodic. If the amplitudes are random or
are, say, the decimals of π, the pulse train is aperiodic. We do not treat pulse trains composed of pulses
having different shapes, such as would be encounter ed in frequency or phase shift keying, although the
methods presented can be modified to account for such pulse trains.
(b) The Fourier Integral Transform X( ω)
Strictly
speaking, the Fourier Integral Transfor m only applies to aperiodic functions due to the
integrability condition given below, but if we ignore that condition and blindly apply the transform to a
periodic function, the limit of the Fourier Integral Transform becomes the Fourier Series Transform, as
will be demonstrated below.
We now state the Fourier Integral transform. Let x(t) be some function of time. If we define X( ω) to be
the "spectral components" or "spectrum" of x(t) according to (1.1), then the claim is that we can recover x(t) from these spectral components according to (1.2 ): [ conventions are discussed in Section 5 ]
Chapter 1: The Fourier Integral Transform
8 Fourier Integral Transform:
X(ω) = ∫-∞ ∞ dt x(t) e-iωt projection = transform (1.1)
x(t) = (1/2 π) ∫-∞ ∞ dω X(ω) e+iωt expansion = inverse transform (1.2)
Dimensions: If Dim[x(t)] = V, then Dim[X( ω)] = V-sec. Here "V" could by anything or nothing. Often
we shall regard x(t) as dimensionless, in which case V is "nothing", but V does suggest Volts as a typical
unit for x(t) one might encounter in practice. In equivalent language, (1.2) repr esents an expansion of x(t) in terms of the spectral components X( ω).
Equation (1.1) shows how these components are "proj ected out" of the function x(t). Sometimes this
projection (1.1) is called "the transform" and then (1.2) is "the inverse transform" or "inversion formula" or "recovery formula" in the sense that x(t) is recovered from its spectral components. The variables t and ω are referred to as "conjugate variables". In th is document, we shall think of t as time and ω as angular
frequency, but they could be arbitrary conjugate variables. In the theory of waves, they might be position x and wavenumber k. The expansion (1.2) can be rewritten in terms of frequency f = ω/2π (so df = d ω/2π) as follows:
X(f) =
∫-∞ ∞ dt x(t) e-i2πft projection = transform (1.3)
x(t) = ∫-∞ ∞ df X(f) e+i2πft expansion = inverse transform (1.4)
where X(f) = X(ω) = X(2πf). This form gets rid of the (1/2 π) in (1.2), but sticks us with 2 π factors in the
exponents. In general we shall stick with the ω form.
There are restrictions on the function x(t) (or equivalently, on X( ω) going in the other direction). One
restriction is that x(t) must be "piecewise continuous", which allows x(t) to have isolated places where it
is discontinuous such as at the edges of our box pulse considered below. A second restriction is that the
derivative of x(t) must also be piecewise continuous. If t is a discontinuous point (such as an edge of our
box), one must interpret x(t) in (1.2) as lim ε→0 [ x(t+ε) + x(t-ε) ]/2. This is why one often sees the
Heaviside step function θ(t) with the property θ(0) = 1/2, as will be demonstrated later.
Fig 1.1
The Heaviside step function is often denoted by u(t) or H(t), but we shall always use θ(t).
A third condition is that x(t) must be "L 1 integrable" which means this:
Chapter 1: The Fourier Integral Transform
9
∫-∞ ∞ dt |x(t)| < ∞ // that is, this integral must be finite (1.5)
Notice that x(t) = sin(t) is not L1 integrable, although x(t) = sin(t) e-ε|t| is for any tiny ε > 0. Certainly
any finite amplitude pulse of any shape having a finite temporal extent will be L 1 integrable. If we
consider the Fourier Transform of a periodic function in the ε limit sense just stated, then we may apply
the Fourier Transform to periodic functions as well as aperiodic ones. This ε limit sense is directly
associated with the theory of distributions, and that is why lots of delta functions appear in the analysis.
Appendix C provides more detail on the Fourier Integral Tr ansform. It seemed best to keep this material
out of the main document flow, though it is quite important. See also Stakgold Vol. 2 Section 5.6.
There are various names associated with this subject , including Fourier, Riem ann, Lebesgue, Fubini,
Parseval and Plancherel. A special class of functio ns for which Fourier Transforms are guaranteed to
work are the Schwartz Functions. It is a story that goes on and on. For example, the Uncertainty Principle of quantum mechanics is directly associated with the Fourier Integral Transform where conjugate
variables are x and p (positi on and momentum) or t and E= hω (time and energy). If one tries to localize
x(t), X(ω ) spreads out and vice versa. Force light to go through a pinhole and this positional confinement
causes uncertainty in photon momentum and the light be am diffracts out from its original center line path
through the hole.
In what follows, we shall often inte rchange the order of two integrations in an expression, or the order of
two sums, or of one sum and one integral. When fu nctions are reasonable and integration or summation
endpoints are finite, this is always an allowed proc edure. When endpoints are infinite, there is some
danger that the interchange gives wrong results. Essentially , a sum or integral with an infinite endpoint (or
endpoints) is a limiting process, and one is then talk ing about interchanging the order of two limits. This
is a rather technical subject having to do with so-called uniform convergence. A certain Moore-Osgood
Theorem says that interchange is allowed as long as both limits exist and at least one of the limits is
uniformly convergent. The situation is furthe r complicated by what we above called the " ε limit sense" of
distribution theory which in effect makes slightly non-convergent forms be convergent. Suffice it to say
that all our order interchanges are ju stified providing the integrands like x(t) respect the conditions stated
above.
(c) The Fourier Sine and Cosine Transforms X
s(ω) and X c(ω)
Any function x(t) can be decomposed into even and odd parts under t ↔ -t,
x(t) = x
even(t) + xodd(t) = [x(t) + x(-t)]/2 + [x(t) - x(-t)]/2 . (1.6)
For x
even(t), only the cos( ωt) part of (1.1) contributes, and for x odd(t) only the sin( ωt) part. Thus,
Xeven(ω) = 2 ∫0 ∞ dt xeven(t) cos(ωt)
Xodd(ω) = 2 ∫0 ∞ dt xodd(t) sin(ωt)
Chapter 1: The Fourier Integral Transform
10
where the factor of 2 arises from reflecting the nega tive part of the integral to the positive side.
Clearly X even(ω) is even in ω , and Xodd(ω) is odd in ω . Therefore, the inverse transformations can be
restated in terms of cos and sin in this same manner, where now (1/2 π) 2 = (1/π ),
xeven(t) = (1/π) ∫0 ∞ dω Xeven(ω) cos(ωt)
xodd(t) = (1/π) ∫0 ∞ dω Xodd(ω) sin(ωt) .
Another view to take of these transforms is to regard x(t) as an arbitrary starting function which is defined
only for t ≥ 0. One can then by fiat add a left side to the function, thereby making it either even or odd as
desired. For example, if x(t) = exp(-t) for t > 0, one could either "evenize" or "oddize" the function in this
manner
Fig 1.2
Since the projections shown above only make use of data for t ≥ 0, this process is just something the user
does mentally to explain why the following two transforms are valid for an arbitrary x(t) which is defined
only for t ≥ 0. The inverse transforms if examined at t < 0 will produce left sides for x(t) having the
appropriate symmetry as suggested in the above figure.
X
c(ω) = 2 ∫0 ∞ dt x(t) cos(ω t) Fourier Cosine Transform
x(t) = (1/ π) ∫0 ∞ dω Xc(ω) cos(ωt ) ( 1 . 7 )
Xs(ω) = 2 ∫0 ∞ dt x(t) sin( ωt) Fourier Sine Transform
x(t) = (1/ π) ∫0 ∞ dω Xs(ω) sin(ωt ) ( 1 . 8 )
One can add an arbitrary factor A to the projection and 1/A to the inversion which will rescale the multiplicative constants, but the prod uct of the constants must be (2/π ).
As noted, any x(t) defined over all t (-∞,∞) can be decomposed into its even and odd parts, and then one
can use tables of Fourier Sine and Cosine Transforms to compute the complete Fourier Transform:
Chapter 1: The Fourier Integral Transform
11 X(ω) = ∫-∞ ∞ dt x(t) e-iωt = ∫-∞ ∞ dt [ xeven(t) + xodd(t) ] [cos( ωt) - i sin(ωt)]
= ∫-∞ ∞ dt xeven(t) cos(ωt) - i ∫-∞ ∞ dt xodd(t) sin(ωt)
= 2 ∫0 ∞ dt xeven(t) cos(ωt) - 2i ∫0 ∞ dt xodd(t) sin(ωt)
= X even,c (ω) - i Xodd,s (ω) . ( 1 . 9 )
Extensive tables of Fourier Sine and Cosine transforms appear in Erdélyi Vol. 4 (ET I).
Example
: x(t) = e-at with Re(a) > 0 :
Xc(ω) = 2 ∫0 ∞ dt x(t) cos(ω t) = 2 ∫0 ∞ e-at cos(ωt) = 2a/( ω2+a2) FCT
x(t) = (1/ π) ∫0 ∞ dω Xc(ω) cos(ωt) = (2a/π) ∫0 ∞ dω cos(ω t) /(ω2+a2) = e-at
XS(ω) = 2 ∫0 ∞ dt x(t) sin( ωt) = 2 ∫0 ∞ e-at sin(ωt) = 2ω /(ω2+a2) FST
x(t) = (1/ π) ∫0 ∞ dω Xc(ω) sin(ωt) = (2/π ) ∫0 ∞ dω sin(ωt) ω /(ω2+a2) = e-at
Below we shall discuss how the Fourier Integral Tr ansform becomes the Fourier Series Transform for
periodic functions. In the same manner, the Fourier Sine Transform becomes the Fourier Sine Series Transform, and similarly for the Cosine transform, though we shall not explicitly discuss these cases.
Chapter 1: The Fourier Integral Transform
12 2. Proof of the Fourier Integral Transform
A si
mple proof of the Fourier Integral theorem follows from this fact,
∫-∞ ∞ dx e±ikx = 2πδ(k) (2.1)
where δ(k) is a "distribution" or "symbolic function" known as the Dirac delta function. To "prove" (2.1),
we first note that when k = 0, both sides ar e infinite, which seems promising. When k ≠ 0, the usual arm-
waving argument is that the oscillating phasor integrat es to 0 over the long haul, or perhaps one claims
that instead for the cos(kx) + isin(kx) real and imaginary parts. The " ε limit sense" mentioned above
strengthens the arm-waving argument. For example, for k ≠ 0,
limit ε→0 [ ∫0 ∞ cos(kx) e-εx dx ] = limit ε→0 [ε / (ε2+k2) ] = 0 . k ≠ 0
To establish the 2π in (2.1), integrate both sides from k = -a to k = a. The right side gives 2 π since the area
"under" δ (k) = 1. The LHS gives (here is our first order interchange),
∫-a a dk ∫-∞ ∞ dx e±ikx = ∫-∞ ∞ dx ∫-a a dk e±ikx = ∫-∞ ∞ dx [ ∫-a a dk cos(kx)] // sin(kx) is odd
= ∫-∞ ∞ dx [ 2 sin(ax)/x ] = 2 [ ∫-∞ ∞ dx sin(ax)/x ] = 2 [ π ] = 2π .
More serious derivations of (2.1) are presented in Appendix A (a) which the reader is encouraged to
peruse. This Appendix also discusses the meaning of δ(0), a symbol we shall be using in Chapter 6.
One key property of the delta function is its "sifting property",
∫a b dx δ(x-y)f(x) = f(y) θ(b-y)θ(y-a) = f(y) Θ(a≤y≤b) a < b (2.2)
where θ(x) is the Heaviside Step Function noted above, and Θ(a≤y≤b) ≡ θ(b-y)θ (y-a) is a special notation
explained in Appendix A (e) which makes certain mani pulations easier to visualize. These functions
cause the integral to vanish if y lies outside the range (a,b). If y coincides with endpoint b, say, then since
θ(0) = 1/2, the right side becomes f(y)/2, as if the integral were picking up half the area of the delta
function. A special case of the above equation is
∫-∞ ∞ dx δ(x-y)f(x) = f(y) . (2.3)
Accepting (2.1), we can verify the Fourier Integral Transform in both directions. First (and here are more
order interchanges!),
Chapter 1: The Fourier Integral Transform
13 X(ω) = ∫-∞ ∞ dt x(t) e-iωt = ∫-∞ ∞ dt {(1/2π) ∫-∞ ∞ dω' X(ω') e+iω't } e-iωt
= (1/2π) ∫-∞ ∞ dω' X(ω')[ ∫-∞ ∞ dt e+i(ω'-ω)t] = (1/2π) ∫-∞ ∞ dω' X(ω') 2π δ(ω'-ω)
= ∫-∞ ∞ dω' X(ω') δ(ω'-ω) = X(ω) .
Going the other way is similar,
x(t) = (1/2 π)
∫-∞ ∞ dω X(ω ) e+iωt = (1/2π) ∫-∞ ∞ dω { ∫-∞ ∞ dt' x(t') e-iωt' } e+iωt
= ( 1 / 2 π) ∫-∞ ∞ dt' x(t') [ ∫-∞ ∞ dω e+iω(t-t')] = (1/2π) ∫-∞ ∞ dt' x(t') 2π δ(t-t')
= ∫-∞ ∞ dt' x(t') δ(t-t') = x(t) .
Have we "proved" the Fourier Transform by doing th ese verifications? Yes, but we have not proven in
detail that the restrictions stated above on x(t) must be respected. In general, "proving" the viability of a
transform lies in the realm of Sturm-Liouville theory , see following Comments. The main idea is that one
must show that a set of basis functions is "complete" for an interval of interest, which means one must know the full "spectrum" of a certain operator L. Comments
The set of functions eikx/2π form a complete orthonormal set on the interval (- ∞,∞) for functions f(x)
of the restricted class described above (L 1 integrable, etc). Picking one of the signs, we can write (2.1) in
these two ways, where * means complex conjuga tion (perhaps think of x as t, and k as ω )
∫-∞ ∞ dx [eikx/2π ] [eik'x/2π ]* = δ (k-k') // functions eikx/2π are orthonormal
∫-∞ ∞ dk [eikx/2π ] [eikx'/2π ]* = δ (x-x') // functions eikx/2π are complete
In general, every "self-adjoint" linear differential ope rator L on a given interval (a,b) defines a complete
orthonormal set of functions on that interval and an associated transform on that interval. These functions
are the normalized eigenfunctions of the eigenvalue equation Lu λ = λuλ. The combination of L and (a,b)
is said to define a "Sturm-Liouville problem". When the endpoints are finite and L is "regular" at these
endpoints, the spectrum for λ is discrete. When the endpoints are "singular", as for example when one or
both are infinite, the spectrum for λ is usually all continuous, but sometimes there is also a discrete
component.
In the case of the Fourier Transform, which is our only transform family of interest, L = -d2/dx2,
Chapter 1: The Fourier Integral Transform
14 λ = k2, the interval is (- ∞,∞), the eigenvalue equation is -d2uk/dx2 = k2uk and uk = eikx/2π .
There are many other "name-brand" transforms and each is associated with a particular L on a
particular interval. An example is the Legendre Pol ynomial Transform on the interval (-1,1). Another is
the Fourier Series Transform on some (a,b). Just as we expand x(t) on the e-iωt in (1.2) for interval
(-∞,∞), so also can we expand f(z) on P l(z) for z in (-1,1), or f(x) on sin(nπ x/L) for x in (0,L). In the latter
two cases, the spectrum is indicated by l = 1,2,3... or n = 1,2,3.. , while in the former ω = real, a
continuous spectrum. The subject has further extension to functions of more than one variable. For example, a function
f(θ,φ) where θ,φ define points on a sphere may be expanded on the "spherical harmonics" Y
lm(θ,φ) which
are simultaneously eigenfunctions of two sel f-adjoint differential operators called L2 and Lz (angular
momentum). The interval for θ is (0,π) and for φ is (0,2π ).
Stakgold Volume I discusses the spectra of differen tial operators in Chapter 4, and the theory of
distributions (such as the delta function) in Chapter 1. Volume II then extends these ideas to multiple
variables.
These Comments are only for the reader's possible interest and are not "used" anywhere below except
where it is noted that the Fourier Transform functions e-iωt form a complete set on the interval (- ∞.∞).
Chapter 1: The Fourier Integral Transform
15 3. The Convolution Theorem and its Derivation
Suppose thre
e functions of t are relate d as follows (a convolution integral):
a(t) = ∫-∞ ∞ dt' b(t-t')c(t') sometimes written a = b * c (3.1)
Letting t" = t - t' this can also be written
a(t) =
∫-∞ ∞ dt" b(t")c(t-t") = ∫-∞ ∞ dt' b(t')c(t-t') = ∫-∞ ∞ dt' c(t-t') b(t') (3.2)
which just shows that the integral is invariant under the change b ↔ c ( so a = b * c = c * b).
Now assume that Fourier Integral expansions exis t for a(t), b(t) and c(t) so we can write,
a(t) = (1/2 π)
∫-∞ ∞ dω A(ω ) e+iωt A( ω) = ∫-∞ ∞ dt a(t) e-iωt
b(t) = (1/2 π) ∫-∞ ∞ dω B(ω) e+iωt B(ω) = ∫-∞ ∞ dt b(t) e-iωt
c(t) = (1/2 π) ∫-∞ ∞ dω C(ω) e+iωt C(ω) = ∫-∞ ∞ dt c(t) e-iωt (3.3)
Then apply the operation ∫-∞ ∞ dt e-iωt to both sides of (3.1),
∫-∞ ∞ dt e-iωt a(t) = ∫-∞ ∞ dt e-iωt [ ∫-∞ ∞ dt' b(t-t')c(t')]
or
A(ω) = ∫-∞ ∞ dt e-iωt [ ∫-∞ ∞ dt' b(t-t')c(t')] . (3.4)
Next, these expressions follow from (3.3),
b(t-t') = (1/2 π)
∫-∞ ∞ dω" B(ω ") e+iω"(t-t')
c(t') = (1/2 π) ∫-∞ ∞ dω' C(ω') e+iω't' ( 3 . 5 )
and we can install them into (3.4) to get (lots of steps here)
A(ω) = ∫-∞ ∞ dt e-iωt [ ∫-∞ ∞ dt' b(t-t')c(t')]
= ∫-∞ ∞ dt e-iωt ∫-∞ ∞ dt' { (1/2 π) ∫-∞ ∞ dω" B(ω") e+iω"(t-t') } { (1/2π) ∫-∞ ∞ dω' C(ω') e+iω't'}
Chapter 1: The Fourier Integral Transform
16
= (1/2 π)2 ∫-∞ ∞ dω" B(ω ") ∫-∞ ∞ dω' C(ω') ∫-∞ ∞ dt e-iωt ∫-∞ ∞ dt' e+iω"(t-t') e+iω't'
= (1/2 π)2 ∫-∞ ∞ dω" B(ω ") ∫-∞ ∞ dω' C(ω') ∫-∞ ∞ dt ∫-∞ ∞ dt' ei(ω"-ω)t ei(ω'-ω")t'
= (1/2 π)2 ∫-∞ ∞ dω" B(ω ") ∫-∞ ∞ dω' C(ω') [ ∫-∞ ∞ dt ei(ω"-ω)t ] [ ∫-∞ ∞ dt' ei(ω'-ω")t' ]
= (1/2 π)2 ∫-∞ ∞ dω" B(ω ") ∫-∞ ∞ dω' C(ω') [2π δ(ω"-ω)] [2πδ(ω'-ω")]
= ∫-∞ ∞ dω" B(ω ") δ(ω"-ω) [ ∫-∞ ∞ dω' C(ω') δ(ω'-ω") ] = ∫-∞ ∞ dω" B(ω ") δ(ω"-ω) [ C(ω ") ]
= ∫-∞ ∞ dω" B(ω ")C(ω") δ(ω"-ω) = B(ω) C(ω) .
Thus, we have proven that
a(t) =
∫-∞ ∞ dt' b(t-t')c(t') ⇒ A(ω) = B(ω) C(ω ) .
Using the very same method, one can show that ⇐ is also true, and we end up with this very important
theorem:
The Convolution Theorem:
a(t) = ∫-∞ ∞ dt' b(t-t')c(t') ⇔ A(ω) = B(ω) C(ω ) (3.6)
The significance of this result cannot be overstated. It says that, whereas the relationship between a,b,c might be complicated in the time domain as shown on the left, that complication goes away in the
frequency domain on the right, where we have a simple product of functions A = BC. One says that the Fourier Integral Transform "diagona lizes" the convolution integral.
Again using the same method of proof, one can obtain this corresponding theorem:
A(ω) = (1/2π)
∫-∞ ∞ dω' B(ω-ω')C(ω') ⇔ a(t) = b(t) c(t) (3.7)
The extra (1/2 π) factor arises because we started with (1.1) and (1.2) which are not symmetric.
Chapter 1: The Fourier Integral Transform
17 Comments
(1) Dimensions . In the convolution equation in (3.6), we shall think of a(t) and c(t) as having the same
dimensional units we generically call V, because we are going to think of this equation as being a "filter"
where c(t) is the input, a(t) is the output, and b(t) is the "filter kernel". In the examples of this document,
we shall take a(t) and c(t) to be dimensionless, but in some application one might add a dimension of
"volts" or "amperes" to the functions a(t) and c(t). Looking at (3.6), we find that if a(t) and c(t) are
dimensionless or have the same dimensions, then b(t) must have dimensions of inverse time. Looking
then at (1.1), we see that A( ω) and C(ω) have dimensions of time, whereas B( ω) is dimensionless. This
then is how the dimensions work out in A( ω) = B(ω) C(ω ).
(2) Operators
. In the language of linear operators, one can regard the functions a and c in (3.6) as vectors
in an infinite dimensional vector space of functions, and then the left equation of (3.6) is a "matrix
equation" which says a = bc where b is a linear operator. Specifically, it is an "integral operator". Operator
b acts on vector c to produce vector a. One could think of (3.6) as a matrix equation a t = Σt' btt'ct'
where the continuous time variables act as indices. If we similarly write the right side of (3.6) as A ω = Σω'
Bωω'Cω' then we find that the matrix Bωω' = δω,ω'Bω so matrix B is "diagonal", hence the term
"diagonalization". We shall not pursue this language much, but make the reader aware of this
interpretation. For more on this subject see St akgold Chapter 3 on linear integral equations.
(3) Groups . In the more general theory of Fourier Anal ysis on groups, the "projection" and "expansion"
have this form, analogous to (1.1) and (1.2), where σ plays the role of ω and g the role of t,
Fσ
kk' = ∫dg f(g) Dσ
kk'(g-1) // projection, transform
f(g) = Σσ dσ Σk,k' Fσ
kk' Dσ
k'k(g) = Σ σ dσ tr[Fσ Dσ(g)] // expansion, inverse transform
where in the last line tr means trace and F and D are rega rded as square matrices. Here g refers to a set of
group variables like Euler angles ψ,θ,φ for the rotation group. The functions Dσ
k'k(g) are the "matrix
representations" of the group which have some dimension dσ. To say that a set of matrices forms a group
representation means that, when multiplied, the matri ces which represent group elements have the same
multiplicative property had by the abstract group elements themselves,
Σ
k" Dσ(g1)kk" Dσ(g2)k"k' = Dσ(g3)kk' or Dσ(g1) Dσ(g2) = Dσ(g3) "group property"
where g
1g2 = g3. Quantity dg is the "invariant measure" on the group which is d ψd(cosθ)dφ for the
rotation group. In our simple Fourier Transform case, we have dg = dt, the group is the group of
translations along the time axis, and the matrix representations have dimension dσ = 1 and are thus 1x1
matrices, namely, e-iωt. The group property shown above is just e-ω1t e-ω2t = e-(ω1+ω2)t .
In the general case, the convolution equation and its diagonalization are given by this generalized
convolution theorem,
Chapter 1: The Fourier Integral Transform
18 a(g) = ∫dg1 b(g1-1g)c(g1) ⇔ Aσ
kk' = Σk" Bσ
kk" Cσ
k"k'
where the dg 1 integral is over the entire parameter space of the group. The derivation of this theorem
makes use of the group property shown above and the fact that dg 3 = d(g1g2) = dg1 when the integration
is over the full group space, just as in the simple case dt 3 = d(t1+t2) = dt1. This is why dg is referred to as
the "invariant" measure.
In the case of one-dimensional representations, the convolution theorem says Aσ = BσCσ which is our
A(ω) = B(ω)C(ω) with σ = ω. Despite the sum on k", the equation on the right is said to be "diagonalized"
because it is true separately for each value of the label σ. If one writes Bσ
kk" = δσ,σ" Bσk,σ"k", then the
matrix B σk,σ"k" is diagonal in the sense that it is mostly zero but has square matrices of size dσ x dσ on
its diagonal ("block diagonal form"). For more on the subject of Fourier analysis on groups, see Hermann.
Chapter 1: The Fourier Integral Transform
19 4. Applications of the Convolution Theorem
This section
is included because books often do not make the connection between the convolution
theorem, Green's Functions, and the real world of everyday electronics. Often too this discussion is presented in the language of Laplace Transforms, so here we work in terms of the above Fourier
Transform. We shall state the general case, then do specific examples.
(a) General case
The real world see
ms to be described by linear differential equations. Here is a general form:
Lt u(t) = f(t) (4.1)
where L
t contains perhaps first and second order differential operators d/dt and d2/dt2. One would like to
solve this equation for u, given some driving function f. It would be nice if one could find some operator
that is the inverse of L t and apply it to both sides of (4.1); the problem would then be solved. This is
exactly what we are going to do. We firs t define a related equation as follows,
L
t g(t-t') = δ ( t - t ' ) . ( 4 . 2 )
Here g(t-t') is the "impulse response" of the differential equation to the driving impulse term δ (t-t'). If we
can solve (4.2) for g, then we know a solution to (4.1) for u(t) in terms of f and g, namely,
u(t) =
∫-∞ ∞ dt' g(t-t') f(t') . (4.3)
In general one can add to this "particular" solutio n any solution of (4.1) with f(t) = 0. These extra
homogeneous solutions can be tailored to meet required boundary conditions.
Proof:
Lt u(t) = L t { ∫-∞ ∞ dt' g(t-t') f(t')} = ∫-∞ ∞ dt' [Lt g(t-t')] f(t') = ∫-∞ ∞ dt' [δ(t-t') ] f(t') = f(t) .
The function g is called the "Green's Func tion", "propagator", or "kernel" of L t. In (4.3) one is applying
an integral operator G = ∫g to function f to get function u, so u = Gf. Looking at (4.1), this integral
operator G must in some sense be the inverse of the differential operator L t.
Now we come to the main point: equation (4.3) is a convolution equation of the form (3.6)! Therefore, we
can write (4.3) in the ω -domain as follows:
U(ω) = G(ω) F(ω ) . ( 4 . 4 )
Chapter 1: The Fourier Integral Transform
20 (b) A specific example: the RC filter section
Consider a sim
ple unloaded RC filter section with input voltage v i(t) and output voltage v o(t),
Fig 4.1
Here is the differential equati on, derived on the right above,
[ RC d/dt + 1] v o(t) = vi(t) . (4.5)
Define the Green's Function g(t) by [ RC d/dt + 1] g(t) = δ (t) . (4.6)
Then the solution to (4.5) is this:
v
o(t) = ∫-∞ ∞ dt' g(t-t') v i( t ' ) . ( 4 . 7 )
This has the convolution form, so in the frequency domain we get
V
o(ω) = G(ω) Vi(ω) . ( 4 . 8 )
Sometimes this is called "filter theory", where G( ω) is the "transfer function" of the filter -- in our case a
simple RC filter. If we expand g(t) as in (1.2) and δ(t) as in (2.1) then (4.6) says
[ RC d/dt + 1] (1/2 π) ∫-∞ ∞ dω G(ω ) e+iωt = (1/2π) ∫-∞ ∞ dω e+iωt
or
∫-∞ ∞ dω G(ω ) [ RC d/dt + 1] e+iωt = ∫-∞ ∞ dω e+iωt
or
∫-∞ ∞ dω G(ω ) [ RC (iω) + 1] e+iωt = ∫-∞ ∞ dω e+iωt .
Since the basis functions eiωt form a complete set on the interval (- ∞,∞), we conclude that
G(ω) [ RC (iω) + 1] = 1
or G(ω) = 1/ [ 1 + i ωRC]
or
G(ω) = (1/iωC) / [(1/iωC) + R] = (-iX
c)/ [ R +(- iX c)] // X C = capacitive reactance = ( ωC)-1
Chapter 1: The Fourier Integral Transform
21 or
G(ω) = Zc/(R+Zc) . / / Z C = -i XC
In the frequency domain, we see G( ω) as the output of a simple voltage divider where one element has
real impedance R and the other imaginary impedance Z c.
The above series of steps shows that in the frequency domain, one can replace d/dt by i ω.
Let τ ≡ RC and compute g(t) using (1.2),
g(t) = (1/2 π)
∫-∞ ∞ dω G(ω ) e+iωt = (1/2π) ∫-∞ ∞ dω e+iωt / [ 1 + iωτ]
= ( 1 / 2 πiτ) ∫-∞ ∞ dω e+iωt / [ ω - i/τ] .
Thinking of this as a contour integral,
Fig 4.2
For t > 0 we can close in the upper half plane and pick up the residue of the pole sitting at ω = i/τ to get
g(t) = (1/2 πiτ) 2πi ei(i/τ)t = (1/τ ) e-t/τ = (1/RC) e-t/RC .
For t < 0 we close instead in the lower half plan e and pick up nothing, so the result is 0. Thus
g(t) = (1/RC) e-t/RC θ(t)
where θ(t) is the Heaviside step function. To summarize, the transfer function G( ω) and its time-domain
Green's Function g(t) are:
G(ω) = 1/( 1 + i ωRC) = Z
C/( R + ZC) ( 4 . 9 )
g(t) = (1/RC) e-(t/RC) θ(t ) . ( 4 . 1 0 )
If vi(t) = δ (t), then from (1.1) we have V i(ω) = 1. In this case, V o(ω) = G(ω) 1 and it must be that v 0(t) =
g(t). Thus, one always interprets g(t) as the impulse response of the filter. In this case, it is of course a
simple decaying exponential. One then interprets θ(t) as saying that the impulse response only propagates
forward in time, never backward ("causality").
We conclude this section by writing out (4.7), whic h shows the time domain solution of our simple RC
filter:
Chapter 1: The Fourier Integral Transform
22 vo(t) = ∫-∞ ∞ dt' g(t-t') v i(t') = (1/RC) ∫-∞ ∞ dt' e-(t-t')/RC θ(t-t') vi(t')
= (1/RC) ∫-∞ t dt' e-(t-t')/RC vi( t ' ) . ( 4 . 1 1 )
This says that the present response of the system at time t is the cumulative result of the impulse
responses at all past times, weighted by the value of the input function v i(t'). In other disciplines, the
expression (1/RC)e-(t-t')/RC = g(t-t') is called a propagator, si nce it describes exactly how the voltage
amplitude v i(t') at some past time propagates into v o(t) at a some future time. This terminology is more
useful when the integral operator has more th an one variable. If dt' were replaced by dt' d3x', then an
equation like (4.11) would perhaps describe how a wa ve propagates through 3D space. The result is then
the sum of "scattering" at all t' in the past, and all positions x ' in space. In our example, there is no spatial
aspect, and the output voltage is just the input voltage scattered off the RC filter at all times in the past.
(c) An even simpler example: L t = (d/dt)
In this section, we use the same equation numbers as in section b above, but add label c, as in (4.9) c.
Let's start by considering L'
t = RC (d/dt). If RC is regarded as very large, RC >> 1, then we may take
over the results (4.9) and (4.10) of the previous section as follows:
G'(ω) = 1/(iωRC) for L' t = RC (d/dt)
g'(t) = (1/RC) θ (t) .
Then if we rescale so that L t = (1/RC) L' t = (d/dt), we just multiply the above results by RC to get
G(ω) = 1/(iω) for L t = (d/dt). (4.9) c
g(t) = θ(t) .
( 4 . 1 0 ) c
Our starting differential equation is
[d/dt] v o(t) = vi( t ) . ( 4 . 5 ) c
Define the Green's Function g(t) by
[d/dt] g(t) = δ( t ) . ( 4 . 6 ) c
Then the solution to (4.5)
c is this:
vo(t) = ∫-∞ ∞ dt' g(t-t') v i( t ' ) . ( 4 . 7 ) c
Chapter 1: The Fourier Integral Transform
23 This has the convolution form, so in the frequency domain we get
Vo(ω) = G(ω) Vi(ω) = [ 1/(iω)] Vi(ω) . ( 4 . 8 ) c
Inserting g(t-t') = θ (t-t') from (4.10) c we get
vo(t) = ∫-∞ ∞ dt' g(t-t') v i(t') = ∫-∞ t dt' vi( t ' ) . ( 4 . 1 1 ) c
We can differentiate this result to obtain the starting equation (4.5) c.
In this case, the time propagator is simply g(t-t') = θ(t - t'). The forward propagator amplitude is just 1
regardless of how far t and t' are separated, and the pr opagator is 0 if it tries to send something backwards
in time. In other words, causality is built into this propagator, and this was also the case for the RC filter
(4.10). Physically, this example is an RC filter with a very long time constant, so basically all effects from
the recent past propagate to the present with no attenuation. The capacitor is an integrator, just as one uses
in an operational-amplifier-based analog computer design.
There are some subtleties involving the transform pair G( ω) = 1/(iω) and g(t) = θ (t) which have been
swept under the rug in the last few paragraphs , but which are laid bare in Appendix C.
Chapter 1: The Fourier Integral Transform
24 5. Fourier Integral Transform Conventions
(a) Sign
of Phase. The Fourier Transform (1.1) and (1.2) is al so true if one replaces i with -i in both
equations. This follows trivially from (2.1). EE pe ople usually think of the fundamental "spectral
component" time dependence as e+iωt and cos( ωt), so they want to see e+iωt in the expansion (1.2).
Physics people who are often pondering plane waves described by exp[+i( k•r - ωt)] or cos( k•r - ωt) want
to see e-iωt in the expansion (1.2). We have chosen to use the EE convention. If you want to use the
physics convention, you must replace all our i by -i, and also Im[ ] by -Im[ ]. The physics convention is used, for example, by Stakgold Vol. II page 23 equa tion (5.32), a source we sometimes quote below.
(b) j Versus i.
EE texts favor j, physics texts always use i, which is of course the true historical symbol for
-1. The reason is that EE people deal with lumped circuits containing currents labeled "i", whereas
physicists deal with Maxwell's equations which contain current density "j". Each discipline chooses its
symbol for -1 to minimize confusion with these other symbols. We shall use i.
(c) Allocation of 2π
. Our convention has been to put the factor of (1/2 π) into the inversion formula (1.2),
and to have no factor at all in the transform formul a (1.1). We shall describe our motivations for doing
this below.
Sometimes books put a 1/ 2π factor in the transform (1.1), which causes the appearance of an
identical 1/ 2π factor in equation (1.2). This has the a dvantage of making the two equations completely
symmetrical, and reminds us that there is complete symmetry between the conjugate variables t and ω.
We have chosen not to do this in our presentation.
And of course the world would not be complete if some people did not prefer to put a 1/2 π into the
expansion equation (1.1), and have none of it in (1.2).
In general, the product of the two factors must be 1/2 π. This is simply due to the 2 π factor sitting on
the right of (2.1). The main reason we choose to put th e factor entirely in (1.2) is the following. Suppose
we have a constant k ≠1 on the right side of (1.1). Then the transform X( ω) so defined is scaled differently
than our X( ω). If we rescale all terms in the ω-plane part of the convolution theorem (3.6), we must end
up with an extra factor of k hanging around in the ne w version of the right side of (3.6). The other
alternative is to add a k factor into the definition of the convolution integral (3.1). Neither is very nice,
and there is a lot of history behind (3.6) as written. This is why we have done our 2 π factors as shown
above.
(d) Comments
. The conventions discussed above have no real physical significance, they just lead to
different definitions of X( ω), so there are slight variations in (1.1 ) and (1.2). It is important to at least
adopt some convention so one knows what one is talking about. A potential problem comes when one
tries to look up something in a table or handbook; one may be off by a factor of 2 π or 2π if one is not
clear on the conventions (attention people sending sp acecraft to planets). The conventions we have
adopted are consistent with 33.7,8 of the 1968 Sc haum's Mathematical Handbook (now 4th Ed. 2012),
and also with a 1967 printing of the Fourth Edition IT T Reference Data for Radio Engineers (now 9th Ed.
2001). There must be something good about these two publications since they are both alive and well
after half a century.
One other small convention detail is that, with our adopted phase convention, spectra X( ω) are
normally analytic in the lower half ω plane and have poles in the upper ha lf plane. Use of the other phase
sign results in X( ω) which are analytic in the upper half ω plane and have poles in the lower half plane,
Chapter 1: The Fourier Integral Transform
25 since in effect the entire ω plane is reflected in the real ω axis by a change of sign phase. This affects the
form of dispersion relations, as we shall see in Chapter 5.
6. The Generalized Fourier Integral Transform and the Laplace Transform X(s)
In the discus
sion above, the Fourier Integral Transform spectrum X( ω) is defined for ω real and for x(t)
being L1 integrable. One can show that the idea of the Fourier Transform can be extended to allow for
x(t) which are not L 1 integrable, provided one thinks of ω as a complex variable, and one thinks of the
inversion integral contour of (1.2) as being a horizontal line in the complex ω plane which runs below any
possible singularities of X( ω). In this extension of the Fourier Integral Transform, one must use single-
sided functions, and one usually deals with right-sided (causal) functions which vanish for t < 0. Such a
function has the general form x(t) = θ(t)f(t). As an example, suppose
x(t) = θ(t)eαt . ( 6 . 1 )
In this case, (1.1) says
X(ω) =
∫0 ∞ dt eαt e-iωt = ∫0 ∞ dt e(α-iω)t = -1
α-iω = -i
ω-(-iα) (6.2)
which has a pole at ω = -iα. The integral converges because we assume that ω has a sufficiently large
negative imaginary part (perhaps -ic) to ma ke it converge. The inversion formula is then
x(t) = (1/2 π) ∫-ci-∞ -ci+∞ dω X(ω ) e+iωt = (-i/2π ) ∫-ci-∞ -ci+∞ dω e+iωt
ω-(-iα) (6.3)
Fig 6.1
where we position the contour at -ci which we assume lies below the pole. For t < 0, we close the contour
downward, e+iωt decays, the great circle makes no contribution, and we recover that fact that x(t) = 0 for
t < 0. For t > 0 we close upward and wrap the pole to get
x(t) = (-i/2 π) 2πi ei(-iα)t = eαt
which of course is the desired result. So our generalized Fourier Integral Transform may be stated as :
Chapter 1: The Fourier Integral Transform
26 X(ω) = ∫0 ∞ dt x(t) e-iωt projection = transform (6.4)
x(t) = (1/2 π) ∫-ci-∞ -ci+∞ dω X(ω ) e+iωt expansion = inverse transform (6.5)
where -ci lies below all singularities of X( ω).
For a detailed discussion of this subject, see Stakgold Vol 2 pp 23-28. Since Stakgold uses the opposite
phasor sign in his definition of the Fourier Transform, the ω plane contour for him is raised up so it runs
above all poles of X( ω), which he calls x^( ω). Stakgold is interested in the Fourier Transform of a
distribution, but in these pages he talks only about functions.
If we now change variables from ω to s = iω, the above generalized Fourier transform becomes
X(s/i) =
∫0 ∞ dt x(t) e-st ( 6 . 6 )
x(t) = (1/2 πi) ∫c-i∞ c+i∞ ds X(s/i) e+is ( 6 . 7 )
where the contour in the s-plane is as shown here, lying to the right of all singularities of X(s/i),
Fig 6.2
We rotated the previous picture 90o counterclockwise to get the s-plane picture. If we now define
X(s) ≡ L[x(t), s] ≡ X ( s / i ) ( 6 . 8 )
the transform becomes
X(s) =
∫0 ∞ dt x(t) e-st ( 6 . 9 )
x(t) = (1/2 πi) ∫c-i∞ c+i∞ ds X(s) e+is ( 6 . 1 0 )
Chapter 1: The Fourier Integral Transform
27 which is the Laplace Transform and its inverse. The inversion c ontour runs to the right of all
singularities in X (s).
Here then is our conclusion: for the set of functi ons x(t) which are "causal", like our Green's Function
propagator g(t) discussed above, and which therefore vanish at negative time, we can make an exact
identification between the Laplace Transform X(s) and the generalized Fourier Transform X( ω) evaluated
at ω = s/i. If we think of s = real, then we are "analytically continuing" the function X( ω) off its real ω
axis. If we think of ω as real, then we are analytically conti nuing the Laplace Transform to imaginary s.
For non-causal functions x(t), the Fourier and Laplace Transforms do not have this simple relationship. Of course, when considering some ge neral function x(t), we can easily make it causal "by
fiat" by simply multiplying it by θ (t). In this case, our association holds all the time,
L [θ(t)x(t), s] = X(s/i) X( ω) = L [θ(t)x(t), iω] . ( 6 . 1 1 )
This lets us make use of extensive ta bles of Laplace Transforms to look up X( ω) for given x(t), and lets us
also understand that the "general properties" of Lapla ce Transforms also apply to the Fourier Transform,
with the appropriate replacement s = i ω. A very large table (~ 100 pages) of Laplace Transforms appears
in the Bateman Manuscript Project Vol. 4 (see Erdelyi. et. al .).
Example
: The Laplace Transform of x(t) = eat is 1/(s-a), and a trivial "property" of the Laplace
Transform is that kx(t) maps into k L [x(t),s] (the transform is linea r). Thus, for our Green's Function of
(4.10), using a = -1/RC,
L [ g(t), s] = L [ (1/RC) e-t/(RC) θ(t) , s] = (1/RC) [1 / (s + (1/RC))] = 1/(sRC + 1) . (6.12)
Thus we would conclude from (6.12) that G(ω) = 1/(iωR C + 1 ) ( 6 . 1 3 )
which agrees with (4.9) above.
Chapter 1: The Fourier Integral Transform
28 7. Reflection Rules
Nothing in o
ur Section 2 proof of the Fourier Transfor m required that x(t) be real. However, if we do
assume that x(t) is real (for example, a voltage or cu rrent in a real circuit), then from (1.1) the following
fact follows at once (* means complex conjugation),
X(-ω) = [X(ω)]* . // x(t) real (7.1)
Thus, one can think of the mysterious negative frequency spectral components of a real function x(t) as simply being defined in this manner in terms of the positive spectral components. Note that X( ω) is in
general complex, even if x(t) is real, because exp(-i ωt) is complex in (1.1).
Regardless of whether x(t) is real or not, we know from (1.2) that
x(t) ↔ X(ω ) ⇔ x(-t) ↔ X(-ω) // any x(t) (7.2)
This notation, used later, means that if x(t) has spectrum X( ω), then x(-t) has spectrum X(- ω).
Similarly, (1.2) says that x(-t) = [x(t)]*
// X(ω ) real (7.3)
A function having the property f(-x) = f*(x) is called a Hermitian function . So we have shown that if
x(t) is real, then X( ω) is Hermitian, and if X( ω) is real, then x(t) is Hermitian.
If x(t) is real, then (7.1) implies | X(-ω )|
2 = | X(ω)|2 ( 7 . 4 )
and this is why, when dealing with spectral densitie s (as we shall below), many authors simply reflect the
left half of the spectrum to the right side which doubles the right side. This must be done carefully if the spectrum includes a δ(ω) term: the folded spectrum for such a term gets a factor of 1/2. In general we
shall not use such folded spectra.
8. Three simple examples of spectra
(a) The spectr
um of x(t) = 1 :
In this example, x(t) is a constant over all time. Using (1.1) and (2.1), we find:
x(t) = 1 X( ω) = 2π δ(ω) . ( 8 . 1 )
This comes as no surprise. For a DC signal, all the en ergy is concentrated at zero frequency. Of course
x(t) = 1 does not respect the requirement
∫-∞ ∞ dt |x(t)| < 0, which is why the spectrum is a distribution.
Chapter 1: The Fourier Integral Transform
29 (b) The spectrum of x(t) = δ(t - t1) :
Here x(t) is an infinitely narrow pulse of area 1, positioned at t = t 1. Using (1.1), we get the following
Fourier spectrum:
x(t) = δ(t - t1) X( ω) = e-iωt1 . ( 8 . 2 )
The spectrum X( ω) has a constant magnitude 1 for all ω, out to infinite frequency. For such a pulse at t=0,
x(t) = δ(t) X( ω) = 1 ( 8 . 3 )
and here the phase is constant. This result is (8.1) with ω ↔ t and the constant adjusted due to the
asymmetry of the transform in our adopted conventio n. From Appendix C we quote this general rule
FT of x(t) = X( ω) ⇔ FT of X(t) = 2 πx(-ω) (C.5)
which, when applied to (8.3), gives (8.1) since δ (-ω) = δ(ω).
(c) The spectrum of x(t) = θ (t) :
The regular Fourier Integral Transform spectrum of the Heaviside step function θ (t) is the somewhat
peculiar first line following, whereas the generalized Fourier Integral Transform gives the second line
x(t) = θ(t) X( ω) = " 1
iω " = 1
iω+ε = 1
i 1
ω - iε
( 8 . 4 ) X ( ω) = 1
iω
// generalized Fourier Integral Transform of (6.4)
which we now explain. Like x(t) = 1, the Heaviside θ(t) is also not in the class of functions for which the
Fourier Transform is defined ( ∫-∞ ∞ dt |x(t)| < 0). We bring θ (t) into the acceptable class by replacing θ by
θε where,
θε(t) ≡ ⎩⎨⎧ e-εt t > 0
0 t < 0 for some very small ε > 0 . (8.5)
Then
X
ε(ω) = ∫-∞ ∞ dt θε(t) e-iωt = ∫0 ∞ dt e-εt e-iωt = 1
iω+ε
and then X( ω) = lim ε→0 Xε(ω) = (1/iω). But we really have to think of (1/i ω) as meaning the limit of
1
iω+ε . To see why, we now compute x(t) from the inversion formula,
Chapter 1: The Fourier Integral Transform
30
x(t) = (1/2 π) ∫-∞ ∞ dω 1
iω+ε e+iωt = (1/2πi) ∫-∞ ∞ dω 1
ω- iε e+iωt . pole at ω = +iε
For t < 0, close the ω contour down and get 0 since the great circle vanishes. For t > 0 close up and pick
up the pole reside to get x(t) = e-εt. Thus we have recovered (8.5). For t = 0, we let the pole move to the
real axis from above and deflect the contour down
Fig 8.1
In the limit the contour is shrunk around the pole, the contributions from (- ∞,0) and (0,∞ ) cancel. These
two terms are known as a principle value integral and we have
PV ∫-∞ ∞ dω (1/ω) ≡ ∫-- ∞
-∞ dω (1/ω) = 0
since the left and right sides cancel (even range integral of an odd function). All that is left is the half turn around the pole which picks up half the residue at the pole (see Appendix C) so the result is then
x(0) = (1/2 πi)
∫-∞ ∞ dω 1
ω- iε = (1/2πi) (1/2) (2 πi * 1) = 1/2
and we obtain the fact that θ(0) = 1/2 as was shown in Fig 1.1.
If we apply our upcoming differentiation rule (11.1) [ which is to multiply by i ω ] we find that
θ(t) ↔ 1
iω+ε => δ(t) = dθ(t)/dt ↔ iω 1
iω+ε = 1
which agrees with (8.3) above. Using the generalized Fourier Integral Transform stated in (6.4) and (6.5), we can regard X(ω ) = 1/(iω)
without all the ε business since the ω recovery contour in (6.5) runs below all singularities in the ω plane,
which contour, when deformed up, gives Fig 8.1 an d all the results quoted above. Applying our Laplace
equivalence notion (6.8), we would predict from X( ω) = 1/(iω) that
L [θ(t), s] = X(s) = X(s/i) = 1
i(s/i)
= 1
s
which is in agreement with any Laplace table. The above examples are treated in more deta il in Appendix C where the pf pseudofunction is
introduced and the Pole Avoidance Rule is derived and then used.
Chapter 1: The Fourier Integral Transform
31 9. Spectrum of an isolated square pulse
Consider a positive square pulse of amplitude A and width τ whic
h is centered at t=0. We can represent
this using the Heaviside step function,
x(t) = A [ θ(t + τ/2) - θ(t - τ/2) ] // = A rect(t/ τ)
Fig 9.1
This "square" pulse is in general rectangular a nd we often refer to it as a box-shaped pulse.
Apply (1.1) to get the spectrum of this pulse,
X(ω) =
∫-∞ ∞ dt x(t) e-iωt = A ∫-∞ ∞ dt { [ θ(t + τ/2) - θ(t - τ/2) ]} e-iωt
= A [ ∫-τ/2 ∞ - ∫τ/2 ∞ ] dt e-iωt = A ∫-τ/2 τ/2 dt e-iωt = A ∫-τ/2 τ/2 dt cos(ω t) // sin = odd
= 2 A ∫0 τ/2 dt cos(ω t) = 2A sin( ωτ/2)/ω = (Aτ) [sin(ωτ /2)] / (ωτ/2) = (Aτ ) sinc(ωτ /2)
where we use the definition sinc(x) ≡ sin(x)/x (there are other definitions). To summarize:
x(t) = A [ θ(t + τ/2) - θ(t - τ/ 2 ) ] ( 9 . 1 )
X(ω) = (Aτ ) sinc(ωτ / 2 ) . ( 9 . 2 )
Observations:
(a) The spectrum is real (see (7.3) for why), and it is a continuous function of ω .
(b) Because sinc(-x) = sinc(x), X( ω) is an even function of ω.
(c) X(ω) has the shape we are all familiar with.. The positive zeros are at x = ( ωτ/2) = nπ for n=1,2,3...
The first zero is at ω = 2π/τ (f = 1/τ). The central peak has height (A τ). Here is a plot of y = sinc(x),
Chapter 1: The Fourier Integral Transform
32
Fig 9.2
(d) Most of the spectral energy is in the central hump, and this represents positive frequency in the range
f=0 to f=1/ τ .
(e) Quantity (A τ) is the Area under the time-domain pulse. If the pulse is made twice as narrow ( τ → τ/2)
and twice as high (A →2A), this area stays constant, but the first zero of X( ω) moves out twice as far (as
do all zeros), so the spectral width doubles.
(f) In the limit τ → 0 with (A τ) = (Area) = fixed, the square pulse x(t) approaches (Area) δ(t). Since
sinc(x) → 1 as x→ 0, we find from (9.2) that X( ω) = (Area) . This is in agreement with (8.3) above. In this
limit, the height of the central ω hump stays fixed, and the zeros move out to infinity, as if a small flat
portion of the central hump has expanded to fill all ω. A delta function has a "white" spectrum since X( ω)
is a constant for all frequencies.
(g) What about the limit τ → ∞ ? If we take this limit with A = fixed, we are converting our pulse to a
constant DC signal x(t) = A. In this limit, as long as ω ≠ 0, the argument of the sinc function oscillates
infinitely fast, giving a function that is zero when averaged over any finite interval. At ω = 0, something
singular happens. The result comes out X( ω) = 2πA δ(ω), in accordance with (8.1) above. In terms of
(9.1), in this limit the central hump gets higher and higher, and the zeros all move in toward ω = 0. As
these zeros get closer together, the oscillation frequency of the tail of sinc(x) becomes infinite and washes
out. To derive X( ω) = 2πA δ(ω) from (9.2), one can use (A.12)
lim
B→∞ δ4(k,B) = lim B→∞ sin(Bk)
πk = limB→∞ B
π sinc(Bk) = δ(k) , (A.12)
which says, with k = ω and B = τ /2,
limτ→∞ τ
2π sinc( τ
2 ω) = δ(ω)
so
lim τ→∞ [(Aτ) sinc(ωτ/2)] = A 2π δ(ω) . ( 9 . 3 )
Chapter 1: The Fourier Integral Transform
33 (h) We can recover the box function from its spectral components using (1.2),
x(t) = (1/2 π) ∫-∞ ∞ dω X(ω ) e+iωt = (1/2π) ∫-∞ ∞ dω (Aτ) sinc(ωτ/2) e+iωt
= (1/2π) (Aτ) ∫-∞ ∞ dω(ωτ/2)-1 sin(ωτ/2) e+iωt
= ( A τ /2π) ∫-∞ ∞ dω(ωτ/2)-1 (1/2i) [ eiωτ/2 - e-iωτ/2] e+iωt
= ( A τ /2π)(2/τ) (1/2i) ∫-∞ ∞ dω (1/ω) [ eiω(t+τ/2) - eiω(t-τ/2)]
= ( A / 2 πi) [ ∫-∞ ∞ dω (1/ω) eiω(t+τ/2) - ∫-∞ ∞ dω (1/ω) eiω(t-τ/2)] ] .
Recall from the generalized Fourier Transform discussion of Section 6 that the ω contours run below the
pole at ω = 0, so the pole is effectively located at ω = +iε. In either integral, if the exponent is positive, the
exponential decays on the upper half great circle, so we close the contour up and pick up the pole residue.
On the other hand, if the exponent is negative, we close down and pick up nothing. Thus
= (A /2 πi) { θ (t+τ/2) 2πi - θ(t- τ/2)2πi }
= A [ θ (t+τ/2) - θ (t- τ/2) ]
which replicates (9.1). A bit more directly, we can compute x(t) on the sides of the box :
x(t) = (1/2 π)
∫-∞ ∞ dω (Aτ) sinc(ωτ/2) e+iωt = (1/2π) ∫-∞ ∞ dω (Aτ) sinc(ωτ /2) cos(ωt)
so that, using x = ωτ/2 so dx = ( τ/2)dω,
x(±τ/2) = (1/2 π) ∫-∞ ∞ dω (Aτ) sinc(ωτ /2) cos(±ωτ/2)
= (A τ/2π)(τ/2) ∫-∞ ∞ sinc(x) cos(x) = (A τ/2π)(τ/2) π/2 = (A/2) ,
supporting the notion of Section 1 that x(t) = lim ε→0 [ x(t+ε) + x(t-ε) ]/2 at a point of discontinuity.
Notice that the single integral has no pole at ω = 0 since sinc(0) = 1, so there is no issue of principle part
integrals involved. The poles only appeared above wh en we split the integral into two integrals.
Chapter 1: The Fourier Integral Transform
34 10. The Area Rules and Parseval's Formulas
Setting ω = 0
in the Fourier Transform (1.1) and then t = 0 in (1.2), one gets
X(0) = ∫-∞ ∞ dt x(t) = [area under x(t) ] (10.1)
x(0) = (1/2 π) ∫-∞ ∞ dω X(ω ) = (1/2π) [area under X( ω) ] . (10.2)
For the box pulse example above, we saw that X(0) = (A τ) from (9.2). In light of (10.1), it is thus not a
coincidence that this is the area under the time-domain box.
From (10.2), we may conclude that the total area under the X( ω) curve (9.2) for our box pulse is 2 πA,
since x(0) = A, the height of our pulse. This is consistent with the fact that
∫-∞ ∞ dx sinc(x) = π . ( 1 0 . 3 )
Another area rule involves the power spectrum. First, it is easy using (1.1), (1.2) and (2.1) to prove this
identity (one of Parseval's),
∫-∞ ∞ dt a(t) b*(t) = (1/2π ) ∫-∞ ∞ dω A(ω ) B*(ω ) . (10.4)
Here * means complex conjugation and is need ed to make things work so one gets δ(ω - ω') in the proof.
Again, the 2 π factor is missing if one uses df in place of d ω.
In the case a = b = x, one gets the energy area rule which says
∫-∞ ∞ dt |x(t)|2 = (1/2π) ∫-∞ ∞ dω |X(ω)|2 = ∫-∞ ∞ df |X(f)|2 . (10.5)
If x(t) is a voltage or current pulse, this says that the total energy (R = 1 Ω) contained in the pulse is the
same no matter which space is used to add it up. The pulse energy density is |x(t)|2 in the time domain, it
is |X(ω)|2/2π in the ω domain, and it is | X(f)|2 in the frequency domain.
For our box pulse, the left side of (10.5) is A2τ. The right hand side gives the same result using (9.2) and
the following fact,
∫-∞ ∞ dx sinc2(x) = π . ( 1 0 . 6 )
It is rather interesting that sinc(x) and sinc2(x) have the exact same area, see (10.3) and (10.6).
Chapter 1: The Fourier Integral Transform
35
F i g 1 0 . 1
There are two other less-well-known Parseval's fo rmulas which we just mention in passing,
∫-∞ ∞ dt a(t) b(t) = (1/2π ) ∫-∞ ∞ dω A(ω ) B(-ω ) (10.7)
∫-∞ ∞ dt A(t) b(t) = ∫-∞ ∞ dω a(ω)B(ω) . ( 1 0 . 8 )
These appear in Stakgold Vol. 2, page 24 and elsewhe re. All these formulas can be proven in the same
manner: just use (1.1), (1.2) and (2.1).
11. Differentiation and Integration Rules with Examples
Below (4.8) and at the end of Section 8 we saw
examples of how d/dt → i ω in ω-space. Here we state the
differentiation rule both ways:
dx(t)/dt ↔ [iωX(ω) ] ( 1 1 . 1 )
dX(ω )/dω ↔ [ – i t x ( t ) ] ( 1 1 . 2 )
To derive the first rule in the general ca se we use (1.2) to expand x(t) so that
dx(t)/dt = d/dt [(1/2 π)
∫-∞ ∞ dω X(ω ) e+iωt] = (1/2π) ∫-∞ ∞ dω X(ω ) d/dt (e+iωt)
= (1/2 π) ∫-∞ ∞ dω X(ω)iω (e+iωt) = (1/2π) ∫-∞ ∞ dω [ iω X(ω)] e+iωt .
Equation (11.2) has a similar derivation with a minus sign due to the sign of the exponent in (1.1).
Chapter 1: The Fourier Integral Transform
36 So, (11.1) says that one gets the spectrum of the derivative of a function by multiplying the original
function's spectrum by i ω. For integration, one must therefore divide by i ω.
(a) Let's apply (11.1) to our square pulse f unction. We have from (9.1) and (9.2),
x(t) = A [ θ(t + τ/2) - θ(t - τ/2) ]
X(ω) = (Aτ ) sinc(ωτ /2) .
Differentiating x(t), we get a pair of opposite signed delta functions separated by distance τ (derivatives of
the box edges),
x
1(t) ≡ dx(t)/dt = A [ δ(t + τ/2) - δ(t - τ/2) ] .
According to (11.1), the spectrum must be,
X1(ω) = iω (Aτ) sinc(ωτ /2) = iω (Aτ)sin(ωτ/2)/ (ωτ/2) = 2iA sin( ωτ/2) .
This agrees with direct calculation,
X
1(ω) = ∫-∞ ∞ dt x1(t) e-iωt = ∫-∞ ∞ dt A [ δ(t + τ/2) - δ (t - τ/2) ] e-iωt
= A [eiωτ/2 - e-iωτ/2] = 2iA sin(ωτ /2) .
As expected, there is no DC component since lim ω→0 X1(ω) = 0. In this drawing,
Fig 11.1
we see x 1(t) on the left in heavy black, and the spectrum X 1(ω) is on the right. The red line and dot show
that there is zero energy at DC, ω = 0. The green dot on the right at the first peak of X 1(ω)/i corresponds
to the green sine curve on the left, which we woul d expect to give a strong component for the double
delta. To understand the sign of the green dot, recall that (1.9) for an odd function x 1(t) states X 1(ω) = - i
X1,s(ω) so that X 1(ω)/i = - X 1,s(ω) = the negative of the Fourier Sine projection.
(b) Now let's apply (11.2) to the following function (multiply box x(t) above by t )
x2(t) ≡ t x(t) = A t [ θ(t + τ/2) - θ(t - τ/2) ].
Chapter 1: The Fourier Integral Transform
37
This represents a doublet sawtooth pulse centered at t=0. According to (11.2) in the ← direction,
X2(ω) = i dX(ω)/dω = i (Aτ)(τ/2) sinc'(ωτ /2) = ( iA τ2/2) sinc'(ωτ /2)
where sinc'(x) = cos(x)/x - sin(x)/x2. Again, X 2(ω) has no DC component, since lim x→0 sinc'(x) = 1/x -
1/x = 0. This is an agreement with the fact that th e sawtooth clearly has a zero integral and this integral
according to (1.1) is just X(0). Here is a picture similar to that shown above,
F i g 1 1 . 2
12. Time translation x(t) causes phase on X( ω).
Assu
me that some x(t) has a spectrum X( ω),
x(t) ↔ X(ω)
Then it follows directly from (1.2) that:
x(t - t
1) ↔ X(ω) e-iωt1 . (12.1)
We saw this happening in the special case of (8.2); he re we see that the result is completely general.
Translation of a signal in time causes the spectrum to gain the phase shown.
According to (1.1), we have this analogous result ,
X(ω-ω
1) ↔ x(t) e+iω1t (12.2)
13. Exponential Sum Rules
For the first ti
me in this document, sums appear. Everything above was integrals only.
In (2.1) stated above, the delta function is written as an infinite integral of an exponential,
Chapter 1: The Fourier Integral Transform
38 ∫-∞ ∞ dx e±ikx = 2πδ( k ) . (2.1)
A similar result involves a summation of exponentials. For k in the range - π to π we claim that:
∑
n = -∞∞
eink = 2πδ(k) -π < k < π . (13.1)
To "prove" this, we argue as we did for (2.1) that for k ≠ 0, the phasors "wash out" in the infinite sum and
we get zero = zero. For k = 0, the summand is 1, so th e result is infinite, and thus the result is proportional
to δ(k). As we did above, we can prove that the factor of 2 π is correct by integrating both sides over k
from -a to +a. The right side gives 2 π. On the left, do the dk integral exactly as done above (2.2) to get
∫-a a dk∑
n = -∞∞
eink = ∑
n = -∞∞
∫-a a dk eink = ∑
n = -∞∞
∫-a a dk cos(nk) // sin(nk) is odd
= ∑
n = -∞∞
[2sin(na)/n ] = 2a + 4 { ∑
n =1∞
[sin(na)/n ] } = 2a + 4 { π-a
2 } = 2π .
In the last few steps, the negative pa rt of the sum is reflected into a positive part since [2sin(na)/n] is even
in n. The sum in curly brackets appears as 1.441.1 on p 46 of Gradshteyn-Ryzhik,
This sum is restricted then to 0 < a < 2 π, but since we had in mind a be ing some small positive number,
this is not a problem, although it does provide a hint of what is to come below.
Since the right side of (13.1) is real, the equation is also valid with e
-ink on the left.
Now we want to generalize (13.1) for k in the range - ∞ to +∞. The result is fairly obvious. Instead of just
a delta function at k=0, we have delta function spikes at each k value for which the summand equals 1.
Thus, spikes will be at k = 0, ± 2 π, ± 4π, and so on,
∑
n = -∞∞
eink = ∑
m = -∞∞
2πδ(k - 2πm) -∞ < k < ∞ . (13.2)
Again, one can prove that the constant is 2 π at each spike by integrating over dk from 2 πm-a to 2πm+a,
for small a. Generally speaking, the right side of (13.2) must have the form shown because the left side is periodic in k with period 2 π, and (13.1) gives that result for - π < k < π .
Replacing n →-n shows that (13.2) is also valid with e
-ink on the left side.
The exponential sum rule (13.2) plays a critical role in the analysis of periodic pulse trains in Chapter 2
below.
Chapter 1: The Fourier Integral Transform
39
Equations (2.1) and (13.2) are more carefully derived in Appendix A. There it is shown that in each case the delta function is the limit of a certain sequence of functions which all have unit area and which, as the limit is taken, become more and more isolated to the neighborhood of the delta function argument. The
Appendix presents the essence of the distribution theory of delta functions.
One result derived in Appendix A (b) is a finite su m version of (13.2) [ see (A.29) and (A.30) ]
∑
n = -NN
eink = 2π { sin[(N+1/2)k]
2π sin(k/2) } ≡ 2π δ5(k,N) . - ∞ < k < ∞ (13.3)
In the limit N →∞, Appendix A shows how the right side of (13.3) approaches the right side of (13.2).
Replacing n →-n shows that (13.3) is also valid with e-ink on the left side.
Poisson Sum Formula : Setting k = 2 πt/α in (13.2) with e-ink gives ( where α is any real number)
∑
n = -∞∞
e-in2πt/α = ∑
m = -∞∞
δ(t/α - m) = | α | ∑
m = -∞∞
δ(t - mα) . (13.4)
Applying ∫-∞ ∞ dt x(t) to both sides and using (1.1) one finds that
∑
n = -∞∞
X(2π n/α) = | α | ∑
m = -∞∞
x(mα) . ( 1 3 . 5 )
This fascinating result appears for example in Sta kgold Vol I (1.23b) and is known as the Poisson Sum
Formula. Sometimes people refer to the underlying equation (13.2) by this name.
Chapter 2: Pulse Trains and Fourier Series
40 Chapter 2: Pulse Trains and the Fourier Series Connection
In this chapte
r we take an arbitrarily shaped pulse and superpose an infinite number of instances of that
pulse spaced by a fixed time interval T 1. This "pulse train" is then a periodic function of period T 1. We
continue with the Fourier Integral notions of Chapter 1 -- such as the spectral components X( ω) -- and we
then make the connection with traditional Fourier Seri es and their coefficients. We show how the Fourier
Integral spectrum becomes discrete for a periodic functi on, and we are able then to relate the Fourier
Series coefficients to the spectral components X pulse (ω) of the pulse used to generate the pulse train.
In the background, and to serve as a vehicle for doing a few calculations, we address a particular problem.
We consider the symmetric zero-DC-offset square -wave pulse train generated from two completely
different methods, one involving adding a negative DC offset to a simple positive square wave pulse train, and the other using biphase pulses. Our main purpose here is to build tools that will be used in more complicated problems. The methods
presented here form the basis for treating amplitude-m odulated pulse trains made from pulses of arbitrary
shape, including the "arbitrariness" of a pul se being statistically present or absent.
14. The Spectrum of a Simple Pulse Train
Recall that a
simple pulse train is just a sequence of identical pulses of shape x pulse (t).
(a) Infinite Length Simple Pulse Train
We assu
me that x pulse (t) is some "reasonable" (non-pathological) function. The pulse train is given by,
x(t) = ∑
n = -∞∞
xpulse (t - tn) t n = nT1 . pulse train (14.1)
Let Xpulse (ω) be the spectrum of x pulse (t). This x pulse (t) does not really have to be a "pulse", but it is
convenient to think of it as such. We imagine that x pulse (t) is a function that is somewhat localized in the
region of t=0, and vanishes for very large positive a nd negative time. The half-width of the pulse can be
larger than T 1 as discussed below, so pulses can overlap. Thus, our x pulse (t) is not itself periodic, and we
thus expect it to have a continuous spectrum.
For example, from (9.2) we already know the spectrum of a single box pulse of height A and width τ
centered at t=0:
X
pulse (ω) = (Aτ ) sinc(ωτ / 2 ) . ( 1 4 . 2 )
Now consider a second pulse which is a copy of our original pulse, but which is translated T 1 units to the
right in time. From (12.1), we know the spectrum of this second pulse:
X(ω, second pulse) = X
pulse (ω) e-iωT1 .
Chapter 2: Pulse Trains and Fourier Series
41
Now construct an infinite periodic wave by superposing pulses at t = 0, ±T 1, ±2T1, ... . We get,
X(ω) = Xpulse (ω) ∑
n = -∞∞
eiωnT1 ( 1 4 . 3 )
According to our exponential sum rule (13.2) with k = ωT1, we can write the exponential sum in (14.3) as
a sum of delta functions to get,
X(ω) = Xpulse (ω)∑
m = -∞∞
2πδ(ωT1 - 2πm ) . ( 1 4 . 4 )
Defining ω1 ≡ 2π/T1 this becomes
X(ω) = (1/T1) Xpulse (ω)∑
m = -∞∞
2πδ(ω - mω1) . (14.5)
which is a standard form. Moving X
pulse (ω) into the sum then gives
X(ω) = ∑
m = -∞∞
(1/T1) Xpulse (ω) 2πδ(ω - mω1)
= ∑
m = -∞∞
(1/T1) Xpulse (mω1) 2πδ(ω - mω1) . (14.6)
Thus, we have a set of evenly spaced delta f unction spikes which occur at these frequencies:
ωm = mω1 m = 0,1,2,3...... (14.7)
In general, one has,
f(ω ) δ(ω - a) = f(a) δ(ω - a) .
Both sides of this last equation are zero when ω ≠ a, and at ω = a, f(a) = f( ω).
Because the quantity (1/T
1)Xpulse (ω) occurs frequently in the following discussion, we define a more
compact notation for it as follows:
c(ω) ≡ (1/T
1)Xpulse (ω) . ( 1 4 . 8 )
Thus, c(ω) is nothing more than our (continuous) pulse spectrum divided by the fundamental period T 1.
We can then rewrite (14.6) as follows:
Chapter 2: Pulse Trains and Fourier Series
42 X(ω) = ∑
m = -∞∞
c(ω) 2π δ(ω - mω1) = ∑
m = -∞∞
c(ωm) 2π δ(ω - mω1) . (14.9)
As was just noted above, we can harmlessly replace ω with ωm = mω1 inside c(ω) in (14.8). This leads us
to define a set of numbers as follows
cm ≡ c(ωm ) = c(mω1) . ( 1 4 . 1 0 )
These numbers are just the values that the function c( ω) takes at our delta spike frequencies. We arrive
then at our final form for the spectrum of an infinite simple pulse train,
X(ω) = ∑
m = -∞∞
cm 2π δ(ω - mω1) . ( 1 4 . 1 1 )
Now we are ready to summarize all these results:
Fourier Integral Transform of an Infinite Simple Pulse Train (14.12)
1. Let x pulse (t) be any reasonable pulse. Construct a pulse train x(t) with spacing T 1:
x ( t ) = ∑
n = -∞∞
xpulse (t - nT1) . (14.1)
By its construction, x(t) is periodic with period T 1, which we can write formally as:
x(t + nT 1) = x(t) . n = any integer
If x(t) is a known periodic function of period T 1, a candidate for x pulse (t) is x(t) over
any one period.
2. Define c( ω) to be the Fourier Integral transform of the pulse, scaled by 1/T 1:
c ( ω) ≡ (1/T1)Xpulse (ω) = (1/T1) ∫-∞ ∞ dt xpulse (t) e-iωt . (14.8) and (1.1)
3. Then the Fourier Integral transform of the Pulse Train is as follows:
X(ω) = ∑
m = -∞∞
c(ω) 2π δ(ω - mω1) = ∑
m = -∞∞
cm 2π δ(ω - mω1) (14.9)
where c m = c(ωm ), ωm = mω1, ω1 = 2π/T1.
4. These c m are the same c m which appear in the next section. That is, they the complex Fourier Series
coefficients.
Chapter 2: Pulse Trains and Fourier Series
43 Item 3 is our main result. It says that the Fourier Tr ansform spectrum of an infinite sequence of pulses is a
sum of equally-spaced delta function spikes whose coe fficients are given by the continuous spectrum of
the central pulse evaluated at the spike frequencies ω = mω1. The pulse spectrum c( ω) = (1/T1)Xpulse (ω)
is normally thought of as the "coefficient envelope", while the equally spaced delta function spikes are the
"lines". In this sample symbolic drawing of a spect rum, the infinitely-high delta function spikes of the
spectrum are represented by finite vertical red lin e segments whose heights are the coefficients c m. The
red lines are the spectrum, and they track the envelope c( ω).
Fig 14.1
It may happen that certain c m vanish, meaning that such lines are not present.
The item 3 sum includes the DC line m=0 having ω
0 = 0. Unless c 0 happens to vanish, the pulse train has
a DC component. As noted in (10.1), X pulse (0) is the area under x pulse (t). Only if this area is zero do we
get c0 = c(0) = (1/T1) Xpulse (0) = 0.
Whereas the Fourier Transform spectrum of a singl e pulse (localized, non-periodic) is continuous in ω,
that of an infinite sequence of pulses is entirely discrete and has no continuous portions. This conforms with the well-known fact that the spectrum of any pe riodic function is discrete. In fact, we have just
proven this to be so. Any periodic signal has to repeat so me pattern, and we just take that pattern to be our
x
pulse (t).
Note on x pulse (t)
In our summary box (14.12) above, we say that if x(t) is some known periodic function, one can take as a
candidate for xpulse (t) the function x(t) restricted to any one period. In this case, the dt integration
endpoints for the projection X pulse (ω) only cover that selected period. If we select the period centered at
t=0, then item 2 in the above summary box becomes perhaps more familiar:
c(ω) = (1/T1)Xpulse (ω) = (1/T1) ∫
-T1/2 T1/2
dt x(t) e-iωt . (14.13)
What is perhaps less obvious is that th ere are many different candidates for x pulse (t) that result in the
same x(t) pulse train. These other choices for x pulse (t) are pulses which slop over into more than one
period T 1. When a pulse train is formed w ith such pulses, the pulses overlap.
To see how this might work, think of a pulse which has a nice gaussian shape and goes about half way
into each neighboring T 1 interval. Draw some of these, then add them up to ma ke the sum curve x(t). In
Chapter 2: Pulse Trains and Fourier Series
44 this case, for a candidate x pulse (t), one can use either the gaussian, which overlaps into several intervals,
or one can use one interval's worth of the sum curve x(t) (shown as the darker curve)
Fig 14.2
We have tried to keep our formulas completely general to allow for pulse trains formed from pulses
which overlap into more than one period. The resu lting spectrum is of course the same no matter which
xpulse (t) is chosen. Later on in the power discussion we shall always regard x pulse (t) as meaning the
shape of x(t) over T 1, as indicated by the black curve above.
To summarize, we can write c( ω) in two equivalent ways
c(ω) = (1/T
1) ∫
-T1/2 T1/2
dt x(t) e-iωt = (1/T1) ∫-∞ ∞ dt xpulse (t) e-iωt . (14.14)
If we evaluate (14.14) at the discrete spike frequencies ω = mω1, we get
cm = (1/T1) ∫
-T1/2 T1/2
dt x(t) e-imω1t = (1/T1) ∫-∞ ∞ dt xpulse (t) e-imω1t . (14.15)
Now since e-imω1(t+T1) = e-imω1t e-imω1T1 = e-imω1t e-im2π = e-imω1t, function e-imω1t is periodic
with period T 1. Since x(t) in the first integral in (14.15) is also assumed periodic with period T 1, the
integration can be over any interval of width T 1, so one usually takes this interval to be (0,T 1). Thus,
cm = (1/T1) ∫0 T1 dt x(t) e-imω1t = (1/T1) ∫-∞ ∞ dt xpulse (t) e-imω1t (14.16)
(b) Finite Length Simple Pulse Train
It is a si
mple matter to modify the above development for a finite length pulse train. We use a finite pulse
train having pulses centered about time t = 0 as in (14.17), then (14.3) becomes (14.18) below :
x(t) = ∑
n = -NN
xpulse (t - tn) t n = nT1 , pulse train (14.17)
X(ω) = Xpulse (ω) ∑
n = -NN
eiωnT1 . (14.18)
Chapter 2: Pulse Trains and Fourier Series
45 The sum is done using (13.3) to give
X(ω) = Xpulse (ω) 2πδ5(ωT1,N) = c( ω) 2π T1δ5(ωT1, N ) ( 1 4 . 1 9 )
where δ5 is a delta function model d escribed in Appendix A. This δ5 is periodic with period 2 π and has
identical peaks separated by 2π . For large N we know that
δ5(k,N) ≈ ∑
m = -∞∞
δ4(k - 2πm, N+1/2) for large N (A.18)
where δ4(x,M) is another delta function model which peaks only near x=0. Thus for large N we can write
X(ω) ≈ c(ω) ∑
m = -∞∞
2π T1δ4(ωT1 - 2πm, N+1/2) . (14.20)
To the extent that these δ 4 peaks are very narrow (large N) we can move c( ω) inside the sum and evaluate
it at ωT1 = 2πm (which means ω = mω1) to get
X(ω) ≈ ∑
m = -∞∞
cm 2π T1δ4(ωT1 - 2πm, N+1/2) . (14.21)
In the limit N → ∞, we get δ4(ωT1 - 2πm, N+1/2) → δ(ωT1 - 2πm) and then
X(ω) = ∑
m = -∞∞
cm 2π T1 δ(ωT1 - 2πm) = ∑
m = -∞∞
cm 2π δ(ω - mω1)
which agrees with (14.11).
Example:
Suppose the pulse is our usual box of width T 1 and height A. Then the pulse train is a constant
DC level x(t) = A. We know that all the c m will vanish except c 0 which is easy to evaluate
c0 = (1/T1) ∫-∞ ∞ dt xpulse (t) = (1/T 1) (AT1) = A .
The spectrum is then
X(ω) =∑
m = -∞∞
cm 2π δ(ω - mω1) = c02πδ(ω) = A 2πδ(ω)
which is a delta line at ω = 0 with factor 2 πA, consistent with (8.1).
Chapter 2: Pulse Trains and Fourier Series
46 15. Connection with the traditional Fourier Series
The pulse train spectrum
(14.11) may be inserted into the Fourier transform expansion (1.2) to get
x(t) = (1/2 π) ∫-∞ ∞ dω X(ω ) e+iωt = (1/2π) ∫-∞ ∞ dω [∑
m = -∞∞
cm 2π δ(ω - mω1)] e+iωt
= ∑
m = -∞∞
cm ∫-∞ ∞ dω δ(ω - mω1) e+iωt = ∑
m = -∞∞
cm e+imω1t (15.1)
= c 0 + ∑
m = 1∞
[ cm e+imω1t + c-me-imω1t] = c0 + ∑
m = 1∞
[ cm e+imω1t + (cm e+imω1t)* ]
= c
0 + ∑
m = 1∞
2 Re [c m e+imω1t] = c0 + 2 Re [ ∑
m = 1∞
cm e+imω1t ] . (15.2)
In the above we have used the reflection rule (7.1) applied to c m ≡ (1/T1)Xpulse (ωm) to find that
c-m = cm*, and then c -me-imω1t = (cm e+imω1t)* .
We know that the c m are in general complex numbers, so make the following two definitions:
am ≡ 2 Re [ c m ] = (2/T 1) Re [ Xpulse (mω1) ]
- bm ≡ 2 Im [ c m ] = (2/T 1) Im [ Xpulse (mω1) ] . (15.3)
Since cm = (1/T1)Xpulse (mω1) it follows that
cm = (1/2) [ a m - ibm ] . ( 1 5 . 4 )
From (14.16) , assuming as we do from now on that x(t) is real, we get
c
m = (1/T1) ∫0 T1 dt x(t) e-imω1t ( 1 5 . 5 )
= ( 1/T 1) ∫0 T1 dt x(t) [ cos(m ω1t) - i sin(m ω1t)]
so that
am = 2 Re [ c m ] = (2/T 1) ∫0 T1 dt x(t) cos(m ω1t) (15.6)
bm = -2 Im [c m ] = (2/T 1) ∫0 T1 dt x(t) sin(m ω1t) . (15.7)
In all the above integrals, we can replace ∫0 T1 dt x(t) by ∫-∞ ∞ dt xpulse (t) as noted in (14.16).
Chapter 2: Pulse Trains and Fourier Series
47 Since x(t) is real, we know from (7.1) that X(-ω ) = X(ω)*, so X(0) must be real. We also know this from
the "area rule" (10.1) -- the area under a real function x(t) had better be real. Thus from (15.3) b 0 = 0 and
from (15.4) c 0 = a0/2 so that
DC component of x(t) = c 0 = (a0/2) = (1/T 1) Xpulse (0) . (15.8)
If we install expression (15.4) for c
m into (15.2) we get this result:
x(t) = c 0 + 2 Re [ ∑
m = 1∞
cm e+imω1t ] = a0/2 + ∑
m = 1∞
Re { [ a m - ibm ] [ cos(m ω1t) + i sin(m ω1t)] }
= a 0/2 + ∑
m = 1∞
am cos(mω1t) + ∑
m = 1∞
bm sin(mω1t) ω1 = 2π/T1 (15.9)
This expansion, along with projections (15.6) and (15.7), is the traditional Fourier Series expansion of a
periodic function of period T 1. Thus, our seemingly uninteresting a m and bm coefficients are exactly the
standard Fourier Series coefficients. Moreover, the DC component of x(t) is equal to c 0 = (a0/2).
For completeness, we write down an alternate form of (15.9),
x(t) = a
0/2 + ∑
m = 1∞
Am cos(mω1t + φm) (15.10)
= a 0/2 + ∑
m = 1∞
Am [cos(mω1t) cos(φm) - sin(m ω1t)sin(φm) ]
= a 0/2 + ∑
m = 1∞
[Am cos(φm)] cos(mω1t) + ∑
m = 1∞
[-Am sin(φm)] sin(mω1t)
Thus,
am = Am cos(φm) A m = am2 + bm2
-bm = Am sin(φm) tan( φm) = -bm/am . (15.11)
So, here is a summary of the above efforts:
Chapter 2: Pulse Trains and Fourier Series
48
Fourier Series Transform (15.12)
1. Let x pulse (t) be any reasonable pulse. Construct a pulse train x(t) with spacing T 1:
x ( t ) = ∑
n = -∞∞
xpulse (t - nT1) (14.1)
By its construction, x(t) is periodic with period T 1, which we can write formally as:
x(t + nT 1) = x(t) n = any integer
If x(t) is a known periodic function of period T 1, a candidate for x pulse (t) is x(t) over any one period.
2. Define the Fourier Series coefficients by these projections = transforms (c m = [ am - ibm ]/2)
c m ≡ (1/T1) ∫-∞ ∞ dt xpulse (t) e-imω1t = (1/T 1) ∫0 T1 dt x(t) e-imω1t (14.16)
a m ≡ (2/T1) ∫-∞ ∞ dt xpulse (t) cos(mω1t) = (2/T 1) ∫0 T1 dt x(t) cos(m ω1t) (15.6)
b m ≡ (2/T1) ∫-∞ ∞ dt xpulse (t) sin(mω1t) = (2/T 1) ∫0 T1 dt x(t) sin(m ω1t) (15.7)
3. The pulse train is then given by these expansions = inverse transforms: (ω 1 = 2π/T1)
x ( t ) = ∑
m = -∞∞
cm e+imω1t = a0/2 + ∑
m = 1∞
am cos(mω1t) + ∑
m = 1∞
bm sin(mω1t) (15.9)
Note that a m, bm, cm and x(t) all have the same dimensions, perhaps volts.
Thus, we have derived the Fourier Series Transform from the Fourier Integral Transform. Recall that it is
not necessary that x pulse (t) be totally contained within a width T 1 We have infinite endpoints on the dt
integrations above, and x pulse (t) is allowed to be any "reasonabl e" function, meaning the projection
integrals must converge.
Chapter 2: Pulse Trains and Fourier Series
49 16. Fourier Series for a positive square wave pulse train
From
equation (9.2) the Fourier Integral spectrum of a positive box pulse of width τ and height A is
Xpulse (ω) = (Aτ ) sinc(ωτ / 2 ) . ( 1 6 . 1 )
From (14.8) and (14.10) the complex Fourier Series coefficients are, using ω
1 = 2π/T1,
c
m = (1/T1) Xpulse (mω1) = (Aτ/T1) sinc(mπτ/T1) . (16.2)
Thus, from (15.3), we know the a and b coefficients as well:
a
m = (2Aτ /T1) sinc(mπτ/T1) m = 0,1,2,3... (16.3)
bm = 0. m = 0,1,2,3...
We have here the Fourier Series coefficients for an infinite pulse train of positive pulses of amplitude A,
width τ, and period T 1, such that the time t=0 occurs in the middle of a positive pulse. If T 1 = τ, the pulse
train is a constant DC level A and c m = δm,0A, a case of minimal interest, so we assume T 1 > τ :
Fig 16.1
The reader is invited to compute the above Fourier Se ries coefficients in the standard manner, using the
conventional formulas in the summary box (15.12). This is done also on page 32-33 of Bennett and
Davey (who use T 1 = T). Their result agrees with the above.
17. More about positive square-wave pulse trains
We can now
summarize what we know about the positiv e square-wave pulse train with pulse height A,
pulse width τ , pulse centered at t = 0, and period T 1 (with ω1 = 2π/T1, see drawing above) :
xpulse (t) = A [ θ(t + τ/2) - θ(t - τ/2) ]. (9.1) (17.1)
c(ω) = (1/T
1) Xpulse (ω) = (Aτ /T1) sinc(ωτ /2) (9.2) and (14.8) (17.2)
x(t) = ∑
n = -∞∞
xpulse (t - nT1) (14.12) (17.3)
X(ω) =∑
m = -∞∞
c(ω) 2πδ(ω - mω1 ) = ∑
m = -∞∞
cm 2πδ(ω - mω1 ) (14.11) (17.4)
cm = (1/T1) Xpulse (mω1) = (Aτ/T1) sinc(mπτ/T1) (17.2) ω = mω1 (17.5)
c0 = (Aτ/T1) = DC component . (15.8) and (14.2) (17.6)
Chapter 2: Pulse Trains and Fourier Series
50
For general τ , all spectral lines are present. Apart from an overall constant, the envelope function c( ω) is
sinc(x), where x = ωτ/2. Here is a linear graph of |sinc(x)| = |sin(x)/x| (red) along with a graph of 1/x
(blue). It is traditional to plot the absolute valu e of the spectrum since the power in each line is
proportional to the square |c m|2 (shown later in (33.27)),
Figure 17.1: Plot of |sinc(x)| function along with 1/x. Fig 17.1 In the case of general pulse width τ (that is, for arbitrary pulse train duty cycle = τ/T
1), one should
imagine the evenly-spaced delta sp ikes superposed on the above pictur e. The spikes are located at x m =
ωmτ/2 = mω1τ/2 = m(πτ/T1), for m = 1,2,3... The spacing between the spikes is dx = (τ/T1)π. Thus, the
number of spikes per hump is (T 1/τ) since each hump is π wide. At low duty cycle, the spacing is small,
and there are many lines for each hump of the |sinc(x)| curve. Here is a rough plot for an ~8% duty cycle,
(T1/τ) = 16:
Figure 17.2. Same |sinc(x)| function with delta spike "lines". Height of each line Fig 17.2
is relative magnitude of the c m coefficient. Plot is for τ = T1/16, duty cycle about 8% .
Chapter 2: Pulse Trains and Fourier Series
51
We shall now examine some special cases as appli cation of what has so far been established.
(a) If τ = T1/2 (50% duty cycle) we get a symmetric positive square wave pulse train,
Fig 17.3
and (17.5) reduces to
c
m = (A /2) sinc(m π/2 ) . ( 1 7 . 7 )
The DC component (the m=0 line) is c 0 = (A/2), which is what we expect. All the other even lines vanish
due to the sinc form. For m = odd integers, we know that
sin(mπ/2) = (-1)(1-m)/2 = (i)1-m = real, since m odd (17.8)
We summarize these facts for our symmetric pulse train with t=0 centered on a positive pulse:
c
m = (A /π) (i)1-m (1/m) m = odd (17.9)
cm = 0 m = even, m ≠ 0
c0 = (A/2)
( If we were to shift the square wave down A/2, the c m would be the same except c 0 = 0. )
For Fig 17.3, if A = 2 we get
c
1 = (2/π) = 0.64 c 3 = - (1/3)(2/π ) = - 0.21 c 5 = (1/5)(2/π ) = 0.13 .
In terms of Figure 17.1, the spacing between the spike positions is π/2. Thus, all the m=even spikes occur
exactly at the zeros of the sinc(x) function, that is why they all vanish. The m=odd spikes occur centered between these zeros, very close to the peaks of the humps. The 1/m drop-off of the Fourier coefficients
seen in (17.9) is reflected in our plot of 1/x in the picture. The 1/x curve intersects the odd spikes at the c
m
coefficient values which are dropping off as 1/m (apart from overall constant). This plot uses A = 2 :
Chapter 2: Pulse Trains and Fourier Series
52
Fig 17.4
Since all c m in (17.9) are real, we know that b n = 0 so there are only Fourier Series cosine contributions to
x(t). This is pretty clear looking at the time domain waveform shown in Fig 17.3 which is even in t.
The pulse train we have constructed above has t=0 oc curring in the middle of a positive pulse. If we were
to shift our entire pulse train to the left by τ/2,
Fig 17.5
so that falling pulse edge lines up with t=0, we would acquire an overall factor eiωτ/2 according to (12.1)
which should then be added as a factor to (17.4). At the lines ω = mω1 this factor becomes
e+imω1τ/2 = e+im(2π/T1)(T1/4) = eimπ/2 = (i)m acting on the c m coefficients. This cancels the phase
shown in (17.9) leaving only a constant i.
So, here are the results for the same pulse train with t=0 occurring at a falling edge:
c
m = i (A /π) (1/m) m = odd (17.10)
cm = 0 m = even, m ≠ 0
c0 = (A/2)
Since all the odd-m c m are now imaginary, we know that the corresponding a m vanish, and only Fourier
sines contribute to the above, as one would expect, since x(t) is now an odd function of t. The magnitudes
of the c m are the same for the original pulse train and the shifted pulse train, so the spectral energy
distribution is unaffected by a time shift of the pulse train.
(b) If τ = T1, we get from (17.5) that c m = Asinc(m π), so now all lines vanish except the line at m=0,
which has a coefficient A. This is again reasonable, since such a pulse train is just a constant DC function
x(t) = A. In terms of Figure 17.1, the zeros spacing is now π, and all the delta spikes align with zeros of
the sinc function, except the DC line spike.
Chapter 2: Pulse Trains and Fourier Series
53
(c) If τ > T1, the theory still applies, but the waveforms ar e a bit strange looking since they overlap. As τ
is continuously increased, the amount of overlap builds up, and the DC coefficient continues to increase,
as shown in (17.6). The black waveform below shows x(t) in the case where T 1/τ = 3/4. The contributing
pulses are drawn alternating red and blue and slightly displaced to make them more visible.
Fig 17.6
(d) If τ → 0, but A τ = area is held fixed, we have x pulse (t) → Aτ δ(t). For area Aτ = 1, xpulse (t) = δ (t)
and in this case x pulse is exactly the first delta function model considered in Appendix A (a). To control
dimensions properly, we instead set area A τ = T1 so that x pulse (t) = T1δ(t) = δ(t/T1) which is
dimensionless. We then find from (1.1) that X pulse (ω) = T1, from (17.2) that c( ω) = 1, and from (17.4)
the spectrum shown below,
Pulse
Pulse Train
xpulse (t) = T1δ(t) x(t) = ∑
n = -∞∞
T1δ(t - nT1)
Xpulse (ω) = T1 X( ω) = ∑
m = -∞∞
2πδ(ω - mω1) . (17.11)
Thus, in the spectrum of a sequence of time-domain delta functions, all "lines" are present and have the same coefficient 2 π. This function is often used as a samp ling function in A/D conversion analysis.
Notice that here, even though the time-domain pulse is a delta function, it's spectrum is still continuous --
being a constant T
1. The infinite pulse train spectrum is discret e, as always. We shall have more to say
later on the implications of (17.11).
(e) We started with a time-domain pulse centered at t = 0. As noted earlier, if this is not the case,
Xpulse (ω) picks up the phase exp(-i ωa) where a is the new time origin of the pulse. Looking at (15.5), we
see that as x(t) → x(t-a), c m → e-imω1a cm so the phasor c m simply rotates in the complex plane. This does
not affect any of our qualitative conclusions above, su ch as lines disappearing in certain cases. Also, the
DC coefficients are unaffected since this exp(-i ωa) = 1 at ω = 0. As one slides the pulse train by varying
point a, the mixture of real and imaginary part of the c m varies. This corresponds to amplitude moving
between the sine and cosine terms of the Fourier series. The energy/power in spectral lines is unaffected
since |cm|2 does not change.
Chapter 2: Pulse Trains and Fourier Series
54 18. Non-positive pulse trains
This is pretty
much a non-issue. We can take any pulse train described by coefficients c m and superpose a
constant DC level of say - B units. This corresponds to Δc0 = -B. Thus, if we choose B = -A/2, we can
cancel out the DC level in our pulse trains of (17.9) or (17.10).
Here are the c m for a pulse train with no DC offset, 50% duty cycle, peak-to-peak amplitude A, and with
falling edge aligned on t=0, taken from (17.10) with cancellation of the DC term :
cm = i (A /π) (1/m) m = odd (18.1)
cm = 0 m = even
Fig 18.1
19. Biphase pulse and pulse train
Define a
biphase pulse as being centered at t=0. The left pulse has width τ and amplitude A/2, the right
pulse has width τ and amplitude -A/2, so the peak-to-peak amplitude is A, and a negative going edge
aligns with t=0.
Fig 19.1
We can analyze this as the superposition of a positive and negative pulse of the square type studied above,
but each pulse has amplitude A/2 instead of A. Also, the positive square pulse is time- shifted to the left
by τ/2 and the negative pulse is shifted to the right by τ/2, so we pick up as corresponding spectral (12.1)
"shift phase" on each contributing pulse. The result is:
x
pulse (t) = SquarePulse(A/2, t+ τ/2) - SquarePulse(A/2, t - τ/2) (19.1)
X
pulse (ω) = (Aτ /2) sinc(ωτ/2) [ e+iωτ/2 - e-iωτ/2]
= (A τ) sinc(ωτ /2) [i sin(ωτ/2) ] = (A τ) sin(ωτ/2)
(ωτ/2) [i sin(ωτ/2) ] = (2iA/ ω) sin2(ωτ/2)
= ( i A τ) sin2(x)/x x = ωτ/ 2 . ( 1 9 . 2 )
Chapter 2: Pulse Trains and Fourier Series
55 This is the same as our envelope (9.2) for the positive pulse train (with pulse centered at t=0), except for
the extra factor [isin( ωτ/2)]. Because of this extra factor, the coef ficient envelope here is quite different
from that for the square pulse. As ω → 0, the envelope function approaches zero -- there is no longer a
central hump. Here is a normalized plot compar ing the box spectrum (9.2) [red] to the biphase pulse
spectrum (19.2) [black], both in absolute value :
Figure 19.2. Same |sinc(x)| and 1/x function, with biphase sin2(x)/x plot added. Fig 19.2
Our biphase pulse train spectrum is given by (17.4) and a new version of (17.5),
X(ω) = ∑
m = -∞∞
cm 2πδ(ω - mω1 ) (17.4)
cm = (1/T1) Xpulse (mω1)= (iAτ/T1) sinc(mπτ/T1) sin(mπτ/T1)
= ( i A / m π) sin
2(mπτ/T1) . ( 1 9 . 3 )
Notice that c
0 = 0 so that all biphase pulse trains have zero DC offset.
Now select the special case τ = T
1/2 to construct a square wave with zero DC component,
Fig 19.3
Chapter 2: Pulse Trains and Fourier Series
56 The expression (19.3) reduces to:
cm = (iA/mπ) sin2(mπ/ 2 ) . ( 1 9 . 4 )
As before, the even lines all vanish. Fo r the odd m lines, the phase factor in (17.8) is now squared, so it is
always 1. Thus, we summarize our results for a ( τ = T1/2) biphase pulse train:
cm = (iA /πm) m = odd
cm = 0 m = e v e n ( 1 9 . 5 )
This result agrees exactly with (18.1) which was obtained by a different process involving three steps: (1) treat a positive symmetric square wave with positive pulse centered at t=0; (2) shift it so that negative going edge aligns with t= 0, thus changing the phase factor; (3) a dd a DC term to cancel the DC offset.
One might wonder how such different spectra envelopes (black and red in Fig 19.1) can yield exactly the
same c
m coefficients for m = 1,2,3... in the case τ = T1/2. The reason is easily understood. When τ = T1/2,
the delta spikes in Figure 19.1 are positioned at x = m π/2, and at these points the two curves have the
same values. Here is a (semi) logarithmic view Fig 19.2. Of course log(0) = - ∞, so the downward spikes of both the red
and black curves really go down infinitely far, but get truncated in the plotting calculation mesh. Spectrum analyzers often allow for such a logarithmic vertical scale.
Figure 19.4. Logarithmic version of Figure 19.2. Fig 19.4
Chapter 3: Sampled Signals and Digital Transforms
57 Chapter 3: Sampled Signals and Digital Transforms
In Sections 20-26, we shal
l used symbols ∆t, T1, tn, and ω1 frequently. We use whichever symbol seems
most convenient at the moment. ∆t is the time spacing between samples of an analog signal. We define T 1
= ∆t to make a connection with Chapter 2 where T 1 was the spacing between pulses superposed to make a
pulse train. As before, ω1 = 2π/T1 . Symbol t n = n ∆t represents the particular times we choose to examine
some signal. So:
T1 = ∆t ω1 = 2π/T1 = 2π/∆t t n = n ∆t = n T 1
20. Sampled Signals and their Image Spectra
As
a specific application of our simple pulse train results boxed in (14.12) we consider the case where
pulse xpulse (t) is a delta function, so we have a pulse train x(t) which is an infinite sequence of these
delta functions spaced by time T 1. This time we let T 1 be the amplitude of each delta function, and as
usual, we use ω1 = 2π/T1. The basic equations for this situation are :
xpulse (t) = T1 δ(t) = δ(t/T1) // dimensionless
X
pulse (ω) = T1 (8.3)
c(ω) = 1 (14.12) item 2
d(t) = ∑
n = -∞∞
T1 δ(t - nT1) (14.12) item 1 (20.1)
D(ω) = ∑
m = -∞∞
2πδ(ω - mω1) (14.12) item 3 (20.2)
where we have renamed our pulse train of delta functions and its spectrum to be d(t) and D( ω), in order to
free up x(t) and X( ω) for new meanings.
If we now multiply the delta function sequence d(t) times some reasonable continuous signal y(t), the
result is a set of delta spikes which are amplitude modulated by the values that y(t) takes at the spike
sampling points t n = nT1. This product we shall call x(t) , it is our "sampled signal", and ω1 is the
radian/sec "sampling rate".
x(t) = y(t) d(t) =
∑
n = -∞∞
y(t)T1δ(t - nT1) =∑
n = -∞∞
y(tn) T1δ(t - nT1) =∑
n = -∞∞
yn T1δ(t - nT1) (20.3)
where y n ≡ y(tn) and tn = n T1. We can apply the "reverse" convolut ion theorem stated in (3.7) to the
leftmost equation in (20.3) to get
Chapter 3: Sampled Signals and Digital Transforms
58 X(ω) = (1/2π) ∫-∞ ∞ dω' Y(ω - ω') D(ω' ) . ( 2 0 . 4 )
Inserting (20.2) into (20.4) quickly yields a famous result
X(ω) = (1/2π)
∫-∞ ∞ dω' Y(ω - ω') D(ω') = (1/2π) ∫-∞ ∞ dω' Y(ω - ω')[ ∑
m = -∞∞
2πδ(ω' - mω1)]
= ∑
m = -∞∞
∫-∞ ∞ dω' Y(ω - ω') δ(ω' - mω1) = ∑
m = -∞∞
Y(ω - mω1)
or
X(ω) = ∑
m = -∞∞
Y(ω - mω1) . ( 2 0 . 5 )
We have therefore shown that,
x(t) = ∑
n = -∞∞
T1 δ(t - tn) y(t) = ∑
n = -∞∞
T1 δ(t - tn) y(tn) (20.6)
X(ω) = Y(ω) + ∑
m ≠ 0
Y(ω - mω1) . ( 2 0 . 7 )
The first term on the right side of (20.7) is the good old Fourier Integral spectrum of y(t). The second term
is a set of identical copies of Y( ω) that are shifted by all possible integer multiples of ω1. Usually these
are called image spectra , and the term Y( ω) is called the main spectrum . Here is a picture,
Figure 20.1. Example of a main spectrum with four of the image spectra. Fig 20.1
Thus, by sampling signal y(t) with delta functions to create the sampled signal x(t), we have picked up an
infinite set of image spectra in addition to the main spectrum Y( ω).
If the spectrum Y( ω) of the "reasonable" original signal y(t) completely cuts off at some ωc below ω1/2
(as shown in Figure 20.1), then the spectra are comple tely disjoint. One can then run signal x(t) through a
low-pass filter that removes all thes e image spectra, ending up with X(ω ) = Y(ω). And from Y( ω), one
can presumably reconstruct y(t). Thus, these image sp ectra can be "dealt with". The larger the gaps
between the spectra, the lower the co st of the low-pass filter required to remove the image spectra.
Chapter 3: Sampled Signals and Digital Transforms
59 Of course if the spectrum of y(t) has ωc > ω1/2, then the spectra in (20.7) and Fig 20.1 overlap, and it is
impossible to recover Y( ω) by itself using a low pass filter. If a low pass filter is placed just above the end
of the Y( ω) spectrum at ωc, the filtered signal will be contamin ated with contributions from the first
image spectrum, an effect loosely known as aliasing , as indicated in this picture,
Figure 20.2. Here the image spectra overlap the main one. This is bad news. Fig 20.2
There is no place one can set a low-pass or band-pass filter to cleanly capture just the main spectrum (or
any of its images) by itself. To avoid this problem, one must select the sampling rate ω 1 > 2ωc. The
quantity 2 ωc is known as the Nyquist rate , so the sampling rate must be larger than the Nyquist rate. This
means that for the highest frequency of interest, one must have at least 2 samples per sine wave. In audio,
one thinks of ωc/2π ≈ 20 KHz, so the Nyquist rate is 2 ωc/2π = 40KHz and typical values for ω 1 are
ω1/2π = 44.1 KHz or 48 KHz. Aliasing in audio sounds like distortion, and in video causes "edge
jaggies" and other artifacts.
Chapter 3: Sampled Signals and Digital Transforms
60 21. Digital Filters, Image Spectra and Group Delay
(a) A Digital Filter as an approximation to an Analog Filter
In Section 3
we derived the convolution theore m stated in (3.6) which we repeat here:
a(t) = ∫-∞ ∞ dt' b(t-t') c(t') sometimes written a = b * c (21.1)
A(ω) = B(ω) C(ω ) . ( 2 1 . 2 )
As demonstrated in Section 4 (b), one can interpret c(t) as an input signal, a(t) as an output signal, and b(t)
as a "filter" which acts on the input to create the output. Equation (21.2) shows the action of such a filter in the frequency domain. B( ω) might be a low-pass filter, a band-pass filter, or some other filter.
When a spectrum like B( ω) is associated with a filter, it is called the transfer function of that filter.
As discussed in the second comme nt after (3.7), the filter (21.1) can be thought of as a = Bc where B is a
linear integral operator, so the filter (21.1) is linear in the usual sense of a linear operator,
B (c
1+c2) = Bc1 + Bc2 and B (αc) = αB (c).
Moreover, by considering the fact that
a(t+Δ) = ∫-∞ ∞ dt' b(t+Δ-t') c(t') = ∫-∞ ∞ dt" b(t-t") c(t"+ Δ) // -t" = Δ - t' (21.3)
one sees that the filter is invariant under a time shift of the input stream. This might not be the case if the
filter kernel had the more general form b(t,t') instead of b(t-t'). Filters of the type (21.1) are therefore referred to as linear time-invariant (LIT) filters.
A time-domain digital filter can only approximate the continuou s integration shown in (21.1). What a
digital (FIR) filter really does is this,
a(t
n) = ∑
m = -∞∞
∆t b(tn - tm) c(tm) where t n = n ∆t . (21.4)
The objects appearing in (21.4) are just numbers -- the values the functions a, b and c take at particular
times, so one could just as well write this as
an = ∑
m = -∞∞
∆t bn-m cm . ( 2 1 . 5 )
Chapter 3: Sampled Signals and Digital Transforms
61 Think of the "digital filter" as a set of numbers b k ; ck is the input signal to the filter, and a n is the output
of the filter. Typically these numbers are represented by one byte in (black & white) video, and by two
bytes in audio. The set of numbers b k is in practice finite. For example, a "5 tap filter" has only these
non-zero values: b -2, b-1, b0, b1, b2 . Therefore the summation in (21.5) in practice is finite. ∆t is the
time between samples, so perhaps 1/ ∆t is 13.5 MHz for digital 601 video or 44.1 KHz for digital audio.
Since in digital practice the convolution integral (21.1) is replaced by the summation (21.5), one is forced to ask oneself: what happens in this case to (21.2)
? We have all the tools needed to answer this question.
Recall the Fourier Integral transform pair (1.1) and (1.2) which we repeat here,
X(ω) =
∫-∞ ∞ dt x(t) e-iωt // projection, transform (21.6)
x(t) = 1
2π ∫-∞ ∞ dω X(ω ) e+iωt . // expansion, inverse transform (21.7)
If we set t = t n = n ∆t , we can rewrite (21.7) as:
x(tn) = 1
2π ∫-∞ ∞ dω X(ω ) e+iωnΔt . ( 2 1 . 8 )
Here now is the set of steps one needs to carry out: (1) write (21.8) for each of the functions a(t
n), b(tn) and c(t n) in terms of A(ω ), B(ω ), and C(ω') :
a(tn) = 1
2π ∫-∞ ∞ dω A(ω) e+iωnΔt
b(tn) = 1
2π ∫-∞ ∞ dω B(ω) e+iωnΔt
c(tm) = 1
2π ∫-∞ ∞ dω' C(ω') e+iω'mΔt ( 2 1 . 9 )
(2) jam these three expansions into (21.4) :
a(t
n) = ∑
m = -∞∞
∆t b(tn - tm) c(tm) (21.4)
1
2π ∫-∞ ∞ dω A(ω ) e+iωnΔt = ∑
m = -∞∞
∆t 1
2π ∫-∞ ∞ dω B(ω) e+iω(n-m)Δt 1
2π ∫-∞ ∞ dω' C(ω') e+iω'mΔt
(3) move the m-summation as far to the right as possi ble, it comes to rest against an exponential,
RHS = 1
2π ∫-∞ ∞ dω B(ω) e+iωnΔt 1
2π ∫-∞ ∞ dω' C(ω') ∑
m = -∞∞
∆t e+i(ω'-ω)mΔt
Chapter 3: Sampled Signals and Digital Transforms
62 (4) do this summation using the exponen tial addition theorem (13.2) with k = ∆t(ω - ω') :
∑
m = -∞∞
eimΔt(ω-ω') = ∑
m = -∞∞
2π δ[ ∆t(ω - ω') - 2πm ] = (2π /∆t) ∑
m = -∞∞
δ( ω - ω' - mω1)
Right here is where the image spectra descr ibed below first appear! Both sides of this equation treated as
a function of ω are periodic with period ω1. If we were to multiply both sides by Δt then take the limit
Δt→0 we would get (since ω1 = 2π/Δt, this means ω1→ ∞ as well)
∫-∞ ∞ dt eit(ω-ω') = 2π δ(ω-ω')
which is just (2.1). The images (δ lines at this point) have run off to infinity and the function is no longer
periodic. So image spectra arise from the fact that a discrete sum of phasor functions eimΔt(ω-ω') ( each
of which is periodic in ω) produces a periodic function, even if that sum is infinite.
(5) kill the d ω' integration against the delta function
RHS = 1
2π ∫-∞ ∞ dω B(ω) e+iωnΔt 1
2π ∫-∞ ∞ dω' C(ω') 2π ∑
m = -∞∞
δ( ω - ω' - m ω1)
= 1
2π ∫-∞ ∞ dω B(ω) e+iωnΔt ∑
m = -∞∞
∫-∞ ∞ dω' C(ω') δ( ω - ω' - m ω1)
= 1
2π ∫-∞ ∞ dω B(ω) e+iωnΔt ∑
m = -∞∞
C(ω-mω1)
so that
1
2π ∫-∞ ∞ dω A(ω ) e+iωnΔt = 1
2π ∫-∞ ∞ dω B(ω) e+iωnΔt ∑
m = -∞∞
C(ω-mω1) .
Since the functions e+iωnΔt form a complete set, we may identify the integrands to obtain
A(ω) = B(ω) ∑
m = -∞∞
C(ω-mω1) . ( 2 1 . 1 0 )
(6) Alternatively, we could kill the d ω integration against the delta functi on. We repeat the first line in (3)
changing the order of integration
RHS = 1
2π ∫-∞ ∞ dω' C(ω') 1
2π ∫-∞ ∞ dω B(ω) e+iωnΔt 2π ∑
m = -∞∞
δ( ω - ω' - m ω1)
= 1
2π ∫-∞ ∞ dω' C(ω') ∑
m = -∞∞
∫-∞ ∞ dω B(ω) e+iωnΔt δ( ω - ω' - m ω1)
Chapter 3: Sampled Signals and Digital Transforms
63 = 1
2π ∫-∞ ∞ dω' C(ω') ∑
m = -∞∞
B(ω' + mω1) e+i(ω'+ mω1n)Δt .
But eimω1nΔt = 1 because m ω1nΔt = mn(2 π/T1)T1 = mn2π. Since the m sum is symmetric, we can
replace m → -m making no difference, and we then replace ω'→ω on the RHS. The result is then
1
2π ∫-∞ ∞ dω A(ω ) e+iω'nΔt = 1
2π ∫-∞ ∞ dω C(ω)eiωΔt ∑
m = -∞∞
B(ω - mω1) .
Again using the completeness of the e+iωnΔt basis functions, we equa te integrands to get
A(ω) = ∑
m = -∞∞
B(ω - mω1) C(ω) . ( 2 1 . 1 1 )
which is the form we shall use below. We noted in (3.2) how the convolution theorem is invariant under
b↔c, and we see this symmetry in the two results just obtained, (21.10) and (21.11). We have then
arrived at this statement of our digital convolution theorem:
a(tn) = ∑
m = -∞∞
∆t b(tn - tm) c(tm) t n = n ∆t (21.12)
A(ω) = [ B (ω) + ∑
m ≠ 0
B(ω - mω1) ] C(ω) . (21.13)
This pair of equations should be compared to th e analog convolution theorem (21.1) and (21.2),
a(t) = ∫-∞ ∞ dt' b(t-t') c(t') sometimes written a = b * c (21.1)
A(ω) = B(ω) C(ω ) (21.2)
We see that there is a penalty for working in the imperfect, discrete world of time-sampled signals like
a(t
n). The penalty is that there are extra ω-space terms in (21.13) that are not present in (21.2).
Recall that B( ω) is the spectrum (transfer function) of a filte r kernel b(t) which we are approximating by a
set of coefficients b
k. In analogy with the spectrum X(ω ) shown in (20.7) of a delta-sampled signal x(t),
the main term B( ω) in (21.13) is called the main spectrum of the filter, while the other terms in the square
bracket are the filter's image spectra passbands. The filter then has a transfer function which looks like the spectrum of Fig 20.1. A perfect analog filter would of course have no such imag e spectra, this is what (21.2) is all about. It has
no such artifacts because it filters at all times t, not just at particular points t
n. The digital filter is "blind"
between sample points, so you can stick it with so me high frequency signals which wiggle an arbitrary
number of wiggles between the sample points. These high frequency signals are what in effect get passed
Chapter 3: Sampled Signals and Digital Transforms
64 through the image pass bands of a digital low-pass fi lter. All the above math should not blind the reader
to this straightforward physical understanding of the image spectra. Here is an example,
Fig 21.1
The red and black signals are treated exactly the same by a digital filter since they have exactly the same
sample values. But the red signal has a ve ry strong frequency component with period Δt = T1 and thus
with frequency ω 1 = 2π/T1 and so the red signal in effect passes through the first image passband of the
filter, giving the same output signal that the black signal would give going through the main passband.
One says that the red signal is an alias of the black signal, or it is aliased into the black signal, giving the
same filter output. Perhaps a violin comes of our filter out sounding like a tuba.
A common use of a digital filter is to remove the im age spectra of digitized signals. These image spectra
are sitting staring us in the face in (20.7). Suppose we construct a digital filter with some set of b n
coefficients to implement a low-pass filter to remove the signal image spectra. But we have just seen that this filter itself has image passbands, so we have to be careful that some of the image spectra of our
sampled signal x(t) don't slip through these im age pass bands of the filter. A standard trick
("oversampling") is to run the filter at a rate ω'
1 which is perhaps 4X or 8X times faster than the rate ω1
of the sampled signal ( ω1 = 2π/Δt). Recall that ω1 > 2ωc, the Nyquist rate. The filter's image spectra now
at mω1' are then pushed away from the central region, cau sing the lower image spectra of the signal x(t)
to be blocked by the filter. Some very high frequency data might get through the image bands of the filter,
but this can be removed by a simple analog filter (p erhaps just a resistor and capacitor) after the D/A
converter which converts the digital signal to analog. An example of this technique is presented in Section
30 using a digital filter implemente d in Section 29 which has the desirable properties of Section 28.
(b) Filter Group Delay
The discussion here is gi
ven in terms of an analog f ilter. The steps stated belo w can be repeated for a
digital filter and one arrives at the same set of conclusions. Recall the convolution theorem from the start of this section,
a(t) =
∫-∞ ∞ dt' b(t-t') c(t') sometimes written a = b * c (21.1)
A(ω) = B(ω) C(ω ) . (21.2)
We interpret this as a filter acting on signal c(t) to produce signal a(t). To assist this interpretation, we
rename signals in this way ( i = input, o = output)
Chapter 3: Sampled Signals and Digital Transforms
65 o(t) = ∫-∞ ∞ dt' b(t-t') i(t') sometimes written o = b * i (21.1)
O(ω) = B(ω) I(ω ) (21.2)
where b and B represent the action of the filter. In general we can write the complex filter spectrum in
terms of its magnitude and phase functions (usi ng a traditional sign convention for said phase)
B(ω) = |B(ω)| e-iφ(ω) . ( 2 1 . 1 4 )
In order to derive the concept of group delay, we assume that our filter is a passband filter of width 2a
centered at some frequency ω2, and having this somewhat idealized spectral shape,
B(ω) = ⎩⎨⎧ |B(ω)| e-iφ(ω) if ω2-a < ω < ω2+a
0 outside this narrow band (21.15)
This could for example be a low-pass filter centered at ω2 = 0 with ω range (-a,a).
Let us assume that i(t) represents a very narrow inpu t pulse whose center lies at t = 0. Since the pulse is
narrow in the time domain, we know (uncertainty pr inciple in Section 1) that it will have a broad
smoothly-varying spectrum X pulse (ω). The ultimate pulse is i(t) = δ(t) which has X pulse (ω) = 1 from
(8.3). From (1.2) the output of the filter can be written as
o(t) = (1/2 π)
∫-∞ ∞ dω O(ω ) e+iωt = (1/2π) ∫-∞ ∞ dω B(ω) I(ω ) e+iωt
= ( 1 / 2 π) ∫-∞ ∞ dω B(ω) Xpulse (ω)e+iωt = (1/2π) ∫ω2-a ω2+a dω |B(ω)| Xpulse (ω)e-iφ(ω)e+iωt
≈ (1/2π) |B(ω2)| Xpulse (ω2) ∫ω2-a ω2+a dω e-iφ(ω)e+iωt .
We have assumed that |B( ω)| is a smooth function near ω = ω2 to make the approximation on the last line.
Similarly, we assume that that filter phase function is also smooth so we can approximate it in this linear
fashion in the neighborhood of ω = ω2,
φ(ω) ≈ φ(ω
2) + (ω-ω2)φ'(ω2) = α + (ω -ω2)β α = φ(ω2) β = φ'(ω2) . (21.16)
Then we find that
o(t) ≈ (1/2π) |B(ω
2)| Xpulse (ω2) e-i(α-βω2) ∫ω2-a ω2+a dω e+iω(t-β) .
Chapter 3: Sampled Signals and Digital Transforms
66 The integral may be evaluated as
∫ω2-a ω2+a dω e+iω(t-β) = [i(t-β)]-1 [ e+i(ω2+a)(t- β) - e+i(ω2-a)(t- β)]
= [ i ( t - β)]-1 eiω2(t-β) 2i sin[a(t- β)] = 2a eiω2(t-β) sin[a(t-β)]/ [a(t-β)]
= 2 a e
iω2(t-β) sinc[a(t- β) ] . ( 2 1 . 1 7 )
Thus, the filter output is
o(t) = (a/ π) |B(ω
2)| Xpulse (ω2) e-i(α-βω2) eiω2(t-β) sinc[a(t-β)]
= [ ( a / π) |B(ω
2)| Xpulse (ω2) e-iα] eiω2t sinc[a(t-β)] . (21.18)
The last two factors show the time dependence of o(t). The e
iω2t represents an oscillation at ω2 which is
the center of the bandpass filter. This is m odulated by an envelope function sinc[a(t-β )] causing the
spectrum of o(t) to fill the pass band, as we also know from O( ω) = B(ω) I(ω) . The initial narrow pulse
i(t) = Xpulse (t) is spread out into a pulse o(t) of width determined by the first zero of the sinc function,
and centered at t = β. Comparing the center of the input and output pulses, one concludes that the pulse
has been delayed by amount β , which is known as the group delay. Recall that β = φ'(ω2) .
We have therefore proven the following theorem: Group Delay Theorem. When a narrow time-domain pulse is passed through a bandpass filter, the
output pulse is delayed approximately by an amount τ
d = dφ/dω evaluated at the bandpass center
frequency. This delay is called the gr oup delay of the filter.
( 2 1 . 1 9 ) Corollary. If the phase function of a filter φ(ω) is linear in ω , then the phase approximation made in
(21.16) is exact, so the theorem just stated has a group delay which is a constant throughout the passband
of the filter. That is to say, d φ/dω is a constant for a filter with linear phase. The implication is that
different pulse shapes, each having slightly different spectra, will all pass through the filter with the same
delay, regardless of where in the bandpass band these pul se spectra hit. The result is good "fidelity" of a
time varying signal such as an audio signal or a radar pulse stream. ( 2 1 . 2 0 )
Chapter 3: Sampled Signals and Digital Transforms
67 22. The Digital Fourier Transform X'( ω) Part I
As a reminder from
the opening paragraph of this Chapter,
T1 = ∆t ω1 = 2π/T1 = 2π/∆t t n = n ∆t = n T 1 .
Recalling from (1.1) that X( ω) = ∫-∞ ∞ dt x(t) e-iωt, we can write down the following non-equation:
X(ω) ≠ ∑
n = -∞∞
∆t x(tn) e-iωnΔt projection = transform (22.1)
X(ω) is the genuine Fourier Integral transform of x(t). Only in the limit ∆t → 0 are the two sides equal,
and we then reproduce (1.1). So let' s define something new called X' that is equal for any finite ∆t:
X'(ω) ≡ ∑
n = -∞∞
∆t x(tn) e-iωnΔt projection = transform (22.2)
Dimensions: If Dim[x(t n)] = V, then Dim[X'( ω)] = V-sec, the same as Dim[X( ω)] .
Although X( ω) can have any shape we want, the new spectrum X' (ω) is periodic with period ω1 ,
X'(ω - mω1 ) = ∑
n = -∞∞
∆t x(tn) e-i(ω-mnω1)Δt = ∑
n = -∞∞
∆t x(tn) e-iωnΔt = X'(ω)
where, as earlier, eimω1nΔt = 1 because m ω1nΔt = mn(2π/T1) T1 = mn2π. So X'(ω) is periodic:
X'(ω - mω1 ) = X'(ω) m = any integer . (22.3)
We now claim (to be shown below) that the inverse of (22.2) is the following:
x(t
m) = 1
2π ∫
-ω1/2 ω1/2
dω X'(ω) e+iωmΔt expansion = inversion (22.4)
This looks like to (1.2) except the integration endpoints are here finite. Thus, we have a different
projection formula, and a correspondingly different expansion formula. For want of a better name, let us
call this new transform the Digital Fourier Transform pair, as opposed to the Fourier Integral Transform
pair given in (21.6) and (21.7). We shall now verify that (22.4) is correct "in bot h directions". We do this in full detail to give the
reader a chance to "practice" using many results presented earlier. First
, insert (22.4) into the right side of (22.2) to get ,
Chapter 3: Sampled Signals and Digital Transforms
68
∑
n = -∞∞
∆t x(tn) e-iωnΔt = ∑
n = -∞∞
∆t [1
2π ∫
-ω1/2 ω1/2
dω' X'(ω') e+iω'nΔt] e-iωnΔt
= 1
2π ∫
-ω1/2 ω1/2
dω' X'(ω) Δt ∑
n = -∞∞
e+inΔt(ω'-ω) // sliding n sum to the right
= 1
2π ∫
-ω1/2 ω1/2
dω' X'(ω) Δt ∑
m = -∞∞
2πδ(Δt(ω'-ω) - 2πm) // (13.2) with k = Δt(ω'-ω)
= ∑
m = -∞∞
∫
-ω1/2 ω1/2
dω' X'(ω') δ(ω'-ω-mω1) // δ(ax) = (1/a) δ(x)
= ∑
m = -∞∞
X'(ω+mω1) Θ( -ω1/2 ≤ ω+mω1 ≤ ω1/2) // (2.2) with special Θ notation
= X ' ( ω) ∑
m = -∞∞
Θ( -ω1/2 ≤ ω+mω1 ≤ ω1/2) // (22.3) that X'( ω + mω1 ) = X'(ω)
= X ' ( ω) . // (A.50), see Appendix A (e). (22.5)
In the second last step, we used (22.3) that X'( ω+mω1) = X'(ω), allowing X'( ω) to be extracted from the
sum on m. Then in the last step we use (A.50) with α = ω1 and x = ω. The sum Σm Θ = 1 basically says
that one partitions the curve f( ω) = 1 into little sections of length ω1, with attention paid to what happens
at the boundaries of these little sections.
Second , insert (22.2) into the right side of (22.4) to get
1
2π ∫
-ω1/2 ω1/2
dω X'(ω) e+iωmΔt = 1
2π ∫
-ω1/2 ω1/2
dω [∑
n = -∞∞
∆t x(tn) e-iωnΔt] e+iωmΔt
= 1
2π ∑
n = -∞∞
∆t x(tn) ∫
-ω1/2 ω1/2
dω e-iω(n-m)Δt = 1
π ∑
n = -∞∞
∆t x(tn) ∫0 ω1/2 dω cos[(n-m) Δtω]
= 1
π ∑
n = -∞∞
∆t x(tn) δn,m (π/Δt) = ∑
n = -∞∞
x(tn) δn,m = x(tm)
Here we have used ∫0 ω1/2 dω cos[(n-m) Δtω] = δn,m (π/Δt) since
Chapter 3: Sampled Signals and Digital Transforms
69 ∫0 ω1/2 dω cos[(n-m) Δtω] = 1
(n-m)Δ t sin[(n-m) Δt (ω1/2) ] = 0 n ≠ m
∫0 ω1/2 dω cos[(n-m) Δtω] = ∫0 ω1/2 dω = (ω1/2) = (π/Δt) n = m
One should keep in mind that this new transform, the Digital Fourier Transform, is dependent on the
constant ∆t = T1. Changing this constant changes the transfor m. The Fourier Integral Transform contains
no such constant. In effect, T 1 = 0.
We use the term "digital" in Digital Fourier Tran sform only because in the time domain the function x(t)
is represented by a sequence of evenly spaced samples x(t n) and we say nothing about what x(t) might be
doing between these sample times. The term Digital Four ier Transform is just our unofficial name for this
transform, and the official name will appear later in Section 24. Notice that both the Fourier Integral Transform and the Digital Fourier Transform are (at first) used to
analyze time-domain functions (or sequences) that are of limited temporal extent, so we think of x(t) or
x(t
n) more or less as some kind of pulse. Technically, x(t) or x(t n) are non-periodic (aperiodic). Here is a
side by side comparison of these two transforms: Fourier Integral Transform
X(ω) = ∫-∞ ∞ dt x(t) e-iωt projection = transform (1.1)
x(t) = (1/2 π) ∫-∞ ∞ dω X(ω ) e+iωt expansion = inverse transform (1.2)
Digital Fourier Transform
X'(ω) ≡ ∑
n = -∞∞
∆t x(tn) e-iωnΔt projection = transform (22.2)
x(tm) = 1
2π ∫
-ω1/2 ω1/2
dω X'(ω) e+iωmΔt expansion = inversion (22.4)
For an aperiodic temporal function x(t) or sequence x(t n), the spectra X( ω) and X'(ω) are both continuous
spectra, even though (1.1) shows X( ω) as an integral and (22.2) shows X'( ω) as a sum of functions which
are continuous in ω. In the next section, we shall see the fascinating relationship between X( ω) and X'(ω).
In the limit Δ t→0, X'(ω) → X(ω) and ω
1→ ∞, so the Digital Fourier Transform is where the Fourier
Integral Transform ends up if the continuum of time is divided into discrete chunks.
Now let's go back to our discrete convolution relation (21.4),
a(tn) = ∑
m = -∞∞
∆t b(tn - tm) c(tm) t n = n ∆t . (22.6)
Dimensions: dim(b) = sec-1, dim(Δt b) = 1, so dim(a) = dim(c).
Chapter 3: Sampled Signals and Digital Transforms
70
What does this look like in the frequency domain? To find out, we insert into (22.6) expansions of the form (22.4) for the functions a, b and c. We just di d this in Section 21 above. The steps (1),(2),(3),(4) are
exactly the same except our dω and dω' integration endpoints are (- ω
1/2, ω1/2) instead of (- ∞,∞). The first
new feature occurs in step (5) where we pick up the analysis:
1
2π ∫
-ω1/2 ω1/2
dω A'(ω) e+iωnΔt = 1
2π ∫
-ω1/2 ω1/2
dω B'(ω) e+iωnΔt 1
2π ∫
-ω1/2 ω1/2
dω' C'(ω') 2π ∑
m = -∞∞
δ( ω - ω' - m ω1)
= 1
2π ∫
-ω1/2 ω1/2
dω B'(ω) e+iωnΔt∑
m = -∞∞
∫
-ω1/2 ω1/2
dω' C'(ω') δ( ω - ω' - m ω1)
= 1
2π ∫
-ω1/2 ω1/2
dω B'(ω) e+iωnΔt ∑
m = -∞∞
C'(ω-mω1) Θ( -ω1/2 ≤ ω-mω1 ≤ ω1/2) // (2.2)
= 1
2π ∫
-ω1/2 ω1/2
dω B'(ω) e+iωnΔt C'(ω) ∑
m = -∞∞
Θ( -ω1/2 ≤ ω-mω1 ≤ ω1/2) // (22.3) for C'( ω)
= 1
2π ∫
-ω1/2 ω1/2
dω B'(ω) e+iωnΔt C'(ω) // (A.50), see Appendix A (e).
Since e+iωnΔt forms a complete set on the interval (- ω1/2, ω1/2), we may equate integrands to find
A'(ω) = B'(ω)C'(ω) . ( 2 2 . 7 )
Thus, our new Digital Fourier transform yields this simple diagonalized result with none of those extra
image terms. Of course we must remain aware that A'( ω) is not the genuine spectrum of a(t), it is some
new thing. Just because we defined a new animal and got (22.7) does not mean that the image spectra go away in (20.7) and (21.10) and (21.11). Here then is the Digital Fourier Transform convolution theorem
in comparison with that for the Fourier Integral Transform:
a(t
n) = ∑
m = -∞∞
∆t b(tn - tm) c(tm) ⇔ A'(ω) = B'(ω)C'(ω) t n = nΔt (22.8)
a(t) = ∫-∞ ∞ dt' b(t-t')c(t') ⇔ A(ω) = B(ω) C(ω ) (3.6)
Chapter 3: Sampled Signals and Digital Transforms
71 23. The Digital Fourier Transform X'( ω) Part II
(a) Relation between X'( ω) and X(ω)
The next pro
blem is to figure out how the genuine Fourier Integral spectrum X( ω) is related to our new
Digital Fourier Transform X'( ω). Start with the Fourier Integral expansion (1.2) ,
x(t) = (1/2 π) ∫-∞ ∞ dω X(ω ) e+iωt expansion = inverse transform . (1.2)
Partition the integration into a set of little ranges of width ω1 :
x(t) = 1
2π ∑
m = -∞∞
∫
mω1-ω1/2 mω1+ω1/2
dω X(ω ) e+iωt .
Change integration variable to ω' = ω - mω1,
x(t) = 1
2π ∑
m = -∞∞
∫
-ω1/2 ω1/2
dω' X(ω'+mω1) e+i(ω'+mtω1)
= 1
2π ∫
-ω1/2 ω1/2
dω' [ ∑
m = -∞∞
X(ω'+mω1) e+imω1t ] e+iω't .
Next, set t = t n = nT1 on both sides. This makes the exponential inside the square bracket equal 1, so
x(tn) = 1
2π ∫
-ω1/2 ω1/2
dω' [ ∑
m = -∞∞
X(ω'-mω1) ] e+iω'nΔt .
Now compare this to the Digital Fourier expansion defined in (22.4) which we duplicate here, changing ω
to ω' and m to n:
x(tn) = 1
2π ∫
-ω1/2 ω1/2
dω' X'(ω') e+iω'nΔt . expansion = inversion (22.4)
Since the functions e+iω'nΔt form a complete basis on the interval (- ω1/2,ω1/2), equate integrands of the
last two equations to get,
Chapter 3: Sampled Signals and Digital Transforms
72 X'(ω) = ∑
m = -∞∞
X(ω- mω1) = [ X( ω) + ∑
m ≠ 0
X(ω - mω1) ] . (23.1)
This is that thing that keeps popping up everywhere -- the main spectrum plus all the image spectra. Thus,
we have shown that this combination is precisely the Digital Fourier Transform spectrum. We can
therefore go back and reexamine some of our earlier results with this new knowledge:
Consider (20.7):
X(ω) =
∑
m = -∞∞
Y(ω - mω1) = [ Y( ω) + ∑
m ≠ 0
Y(ω - mω1) ] = Y'(ω) . (23.2)
This says that the Fourier Integral spectrum of a "reasonable" signal y(t) multiplied by a sequence of delta
functions is exactly the Digital Fourier Transform spectrum Y'( ω). Of course Y'( ω) is computed from
(22.2) from a knowledge of y(t) only at the sample points t n.
Next, we realize that our digital f ilter equations (21. 10) and (21.11) ,
A(ω) = B(ω)
∑
m = -∞∞
C(ω-mω1) = ∑
m = -∞∞
B(ω - mω1) C(ω) ,
become
A(ω) = B(ω) C'(ω )
A(ω) = B'(ω) C(ω) ( 2 3 . 3 )
The second equation is our low-pass digital filter B w ith input C and output A. The filter with all its
image pass bands is now conveniently represented by B'(ω ). A(ω) and C(ω) are still the Fourier Integral
spectra of a and c. However, we already know from ( 22.7) that (23.3) is true with primes on A and C as
well,
A'(ω) = B'(ω) C'(ω) . ( 2 3 . 4 )
It might seem unusual that (23.3) and (23.4) can all be true. They are all true, and we can now present a
much more compact derivation of (23.4) by making use of (23.3) :
A(ω) = B'(ω) C(ω) // (23.3) which is really just (21.11)
A(ω - mω
1) = B'(ω - mω1) C(ω - mω1) // set ω → ω - mω1,
= B'(ω) C(ω - mω1) // (22.3) for B'( ω)
Then:
∑
m = -∞∞
A(ω - mω1) = B'(ω) ∑
m = -∞∞
C(ω - mω1)
or
A'(ω) = B'(ω) C'(ω)
Chapter 3: Sampled Signals and Digital Transforms
73
(b) Summary of the Digital Fourier Transform
We now summarize what
we know about the Dig ital Fourier Transform (this box takes 2 pages)
Digital Fourier Transform ( 2 3 . 5 )
1. Let x(t) be any reasonable function.
2. Divide up the time axis into steps t n = n ∆t; let T 1 = ∆t and ω1 = 2π/T1.
3. We can think of samples x n = x(tn) for the above x(t). Alternatively, we can think of the
x n as some given sequence, and one could then construct an infinite number of functions
x(t) for which x n = x(tn).
4. In terms of x(t), the Digital Fourie r Transform and its inverse are given by
X ' ( ω) ≡ ∑
n = -∞∞
∆t x(tn) e-iωtn projection = transform (22.2) // V-sec
x ( t n) = 1
2π ∫
-ω1/2 ω1/2
dω X'(ω) e+iωtn expansion = inversion (22.4) // V
More generally, dispensing now with x(t) and writing t n = nT1 in the exponential,
X ' ( ω) ≡ T1∑
n = -∞∞
xn e-iωnT1 projection = transform
x n = 1
2π ∫
-ω1/2 ω1/2
dω X'(ω) e+iωnT1 expansion = inversion
If Dim(x n) = V, then Dim(X') = V-sec.
5. By its definition (and t n = n ∆t), X'(ω) is periodic in ω with period ω1:
X ' ( ω - mω1) = X'(ω) m = any integer (22.3)
6. The relation between X'( ω) and the Fourier Integral spectrum X( ω) of x(t) is given by:
X ' ( ω) = ∑
m = -∞∞
X(ω- mω1) = [ X( ω) + ∑
m ≠ 0
X(ω- mω1) ] (23.1)
Chapter 3: Sampled Signals and Digital Transforms
74
7. The Digital Fourier Transform diagonalizes any convolution sum:
a ( t n) = ∑
m = -∞∞
∆t b(tn - tm) c(tm) t n = n ∆t (22.6)
A ' ( ω) = B' (ω) C'(ω) (22.7)
8. It is also true that
A ( ω) = B' (ω) C(ω) = B(ω)C'(ω) (23.3)
Chapter 3: Sampled Signals and Digital Transforms
75 24. The Z Transform X"(z)
The Digital Fourier Transform de scribed in Sections 22 and 23 is really
the Z Transform times Δt. We
have concealed this fact up till now because the ω-space version, called X'( ω) in Section 23, allows direct
comparison to the Fourier Integral spectrum X( ω). We have already drawn the major conclusions. Here
we just change the clothing. Change variables from ω to dimensionless z, [ ω
1= 2π/Δt so Δ t = 2π/ω1 = π/(ω1/2) = T1 ]
z ≡ eiωΔt = eiπ[ω/(ω1/2)] dz = z i ∆t dω d ω = dz
iz Δt . (24.1)
Note that as ω runs over its range - ω1/2 to +ω1/2 , phasor z runs from - π to π on a unit circle.
Now define the Z Transform X"(z) in term s of the Digital Fourier Transform X'( ω),
X"(z) ≡ 1
∆t X'(ω( z ) ) . ( 2 4 . 2 )
Dimensions : If Dim(x n) = V, then Dim(X') = V-sec so Dim(X") = V, the same as x n, see also (24.3).
With this substitution, and letting x
n = x(tn) = x(n∆t),
the above Digital Fourier Transform formulas (22. 2) and (22.4) (see box above) immediately become:
X"(z) =
∑
n = -∞∞
xn z-n projection = transform (24.3)
xn = 1
2πi ∫
C
dz X"(z) zn-1 expansion = inversion (24.4)
where C is a contour doing one counterclockwise trav ersal of the unit circle in the z plane. These two
equations are the Z Transform and its inverse.
Limit Comment : Since X'( ω) → X(ω) as Δt→0, it follows that lim Δt→0 [ Δt X"(z)] = X( ω). So this is
how one could get from the Z Transform to the Fourier Integral Transform.
Mapping Comment: One can think of z = eiΔtω as describing an analytic mapping (conformal map) from
the complex ω-plane to the z-plane. Here is a picture of that mapping :
Chapter 3: Sampled Signals and Digital Transforms
76
Fig 24.1
The infinite gray vertical strip of the ω-plane with - ω
1/2 ≤ Re(ω) ≤ ω1/2 maps into the entire z plane. The
real axis in the red range - ω1/2 ≤ ω ≤ ω1/2 maps into the unit circle in the z plane as shown. The upper
half of the strip in the ω-plane maps into the interior of the unit ci rcle, and the lower half of the strip maps
into the exterior of the unit circle. The blue a nd green arrows map as shown. Generally, horizontal
segments in ω map into origin-centered circles in z, and vertical lines in ω map into rays in z.
The z plane shows the principle Riemann sheet of the mapping with a black branch cut going off to
the left from the z plane origin. If in the ω plane one continues the red arrow into the next strip to the
right, Re( ω) > ω1/2, one dives through the branch cut on the right and arrives on the next sheet in z.
For some general function f(ω ) define F(z) ≡ f(ω(z)). The function F(z) would have an induced
branch cut as shown on the right, with some discontinui ty across it. In this case, the red circle would not
represent a closed integration contour, so the usual rules of complex integration around closed loops
would not apply. However, if f( ω) were periodic with period ω1, there would be no discontinuity across
the branch cut in F(z) because f( ω) would take the same value on the two vertical edges of the grey strip
in the ω plane, and therefore F(z) would have the same value on the two sides of the cut, which means
there is no cut. In this case, the red circle does represent a closed contour. According to (22.3), the Digital Fourier Transform X'(ω ) is periodic with period ω
1. Therefore the Z
Transform X"(z) has no branch cut and the red circle is a closed contour, as used in the examples below.
This is why the Z Transform is so useful. The redunda nt information in the infinite number of vertical
strips in the ω plane is reduced to non-redundant information in the z plane. The mapping is completely
analytic, introducing no poles or branch cuts. If a function F(z) = (z-a)
-1 has a pole in the z plane at a, as s hown in Fig 24.1 by the x on the right,
then f(ω) has a pole at ωa = ln(a)/(iΔ t) as shown by the x on the left, ln(a) < 0. To show this, consider ω in
the neighborhood of ω a:
f(z(ω )) = 1
z-a = 1
eiΔt(ω-ωa) eiΔtωa - a = 1
eiΔt(ω-ωa)a- a = 1
a(eiΔt(ω-ωa)- 1) ≈ 1
aiΔt(ω-ωa) .
The bottom line is that in going from the Digital Four ier Transform to the Z Transform, we are simply
making a change of variable and removing redundant information. There is nothing dramatically new
introduced by doing this. As we shall see belo w, there is a notational economy in writing z-1 for a delay
of Δt in place of e-iΔtω .
Chapter 3: Sampled Signals and Digital Transforms
77 So far in these notes we have run into the Fourier Inte gral Transform and its Sine and Cosine cousins, the
Laplace Transform, the Fourier Series Transform, the Digital Fourier Transform, and the Z Transform.
They are all variations on the same theme, and we have shown how they are all related to each other. They all have an analogous set of "basic results" an d "rules". We now peruse these results and rules for
the Z Transform.
(a) Convolution Theorem
According to the definiti
on X"(z) ≡ X'(ω(z))/Δ t, if we transcribe the convolution result A'( ω) = B'(ω)
C'(ω) of (23.4) , we pick up an extra factor of ∆t . Thus we compare convolution theorems :
an = ∑
m = -∞∞
∆t bn-m cm ⇔ A'(ω) = B'(ω)C'(ω) (22.8)
an = ∑
m = -∞∞
∆t bn-m cm ⇔ A"(z) = ∆t B"(z) C"(z) (24.5)
Dimensions : Dim(a,c,A",C") = V, Dim(b,B") = sec-1.
In (24.5), it is convenient to absorb the Δt factor into the b
n-m coefficients. To do this, we define
h
n ≡ ∆t bn ( 2 4 . 6 )
so that (24.5) may be written in this simpler and more traditional form, where H"(z) is the Z Transform of
hn,
an = ∑
m = -∞∞
hn-m cm ⇔ A"(z) = H"(z) C"(z) . (24.7)
Dimensions : Dim(a,c,A",C") = V, Dim(h,H") = 1.
Equation (24.7) is the digital convolution theo rem stated in terms of the Z Transform.
(b) Unit Impulse
The analog of the unit i
mpulse δ(t-a) at t=a must be a sequence of numbers x n which are all zero except
the one say at some integer m. We might write this as
xn = δm(n) ≡ δn,m "unit impulse" n = all integers, the sequence index (24.8)
If we stuff this into the Z-Transform projection (24.3) , we get
X"(z) = z-m . " unit impulse response" (24.9)
This looks a lot like our Fourier Integral result (8.2) (setting t 1 = tm)
Chapter 3: Sampled Signals and Digital Transforms
78
x(t) = δ(t - tm)
X(ω) = e-iωtm = e-iωΔtm = z-m . (8.2)
The expression z-m on the right is the same in both cases. The Fourier Integral Transform does to its
appropriate "unit impulse" just what the Z Transform does to its appropriate "unit impulse". The unit
impulses are different.
In a filter with input I and output O we have O"(z) = H"(z) I"(z), where H"(z) is the filter transfer
function. If I is taken to be a unit impulse at time t=0, then from the above I"(z) = z
0 = 1. Thus, quantity
H"(z) is the z-domain response of the filter to an impulse at t=0. From (24.4) one can then get the time
domain impulse response h n. We shall do this below for an "RC" filter.
(c) Time Translation
Above we
show a unit impulse δm(n) at time m and its Z transform z-m. A unit impulse one step later in
time would be δm+1(n), and its Z transform would be z-m-1 = z-m z-1 . This suggests that if a signal is
delayed by one time step, its Z transform acquires a factor z-1. Advancing a signal one step means
multiply by z+1. These facts are true for an arbitrary signal; they follow immediately from (24.4):
xn+1 = 1
2πi ∫
C
dz X"(z) zn-1 z+1 // advance one step x n+1 ↔ z+1X"(z) (24.10)
xn-1 = 1
2πi ∫
C
dz X"(z) zn-1 z-1 // delay one step x n-1 ↔ z-1X"(z) . (24.11)
A very similar thing happens in the Fourier Integral Transform world, where time translation generates a
multiplicative phase as shown in (12.1): x(t - t 1) ↔ X(ω) e-iωt1 .
In general, one can delay a digital signal one step in time by running it through a D flip-flop having clock
period Δt, so this is why such f lip-flops are associated with z
-1 in a digital filter. There is no analogous
device to associate with z+1. It would have to be a causality-violating device.
(d) Derivative Limit
Based on the preceding subsection, we know that
the following difference of two sequences has this
transform,
x
n – xn-1
∆t ↔ X"(z) [ 1 - z-1
∆t] . (24.12)
Chapter 3: Sampled Signals and Digital Transforms
79 This is just a simple super position. If we take the limit ∆t → 0, the LHS becomes dx/dt, and the RHS
becomes X"(z) i ω, since z-1 = exp(-i ω∆t) ≈ 1 - iω∆t. Since limΔt→0 [ Δt X"(z)] = X( ω), we are not
surprised to find that the i ω rule (11.1) applies to both X"(z) and X( ω).
(e) Digital RC filter
This filter was treated in
terms of the Fourier Transform in Section 4 (b) where we wrote its analog
description in (4.5),
RC dv
o(t)/dt + v o(t) = vi(t) . (4.5) (24.13)
Undoing the limit as just described above, we write this in digital form as
RC v
o(tn) - vo(tn-1)
∆t + vo(tn) = vi(tn) . (24.14)
For any finite Δt, this equation is of course different from (24.13) but for small Δt we expect it to be a
good approximation for the system described by (24.13).
Letting o n ≡ vo(tn) and in ≡ vi(tn) this reads
RC on - on-1
∆t + on = in . ( 2 4 . 1 5 )
Z Transform each of the four terms shown and use (24.11) on o n-1 to get
RC O"(z) [ 1 - z-1
∆t] + O"(z) = I"(z)
or
[ α (1-z-1) + 1] O"(z) = I"(z) . α ≡ (RC/Δt) = dimensionless
If we want to interpret this circuit as a digital filter, we write, as in (24.7), O"(z) = H"(z) I"(z) . (24.16) The filter transfer function is then
H"(z) = 1
α(1-z
-1) + 1 = z
α(z-1) + z = z
(α+1)z -α = z/(1+α)
z - α/(1+α) . (24.17)
We may now use (24.4) to recover the time domain signal,
h
n = 1
2πi ∫
C
dz H"(z) zn-1 = 1
2πi(1+α) ∫
C
dz zn
z - α/(1+α) . (24.18)
Chapter 3: Sampled Signals and Digital Transforms
80 For n ≥ 0, the integrand has a single pole at location z = α/(1+α) where α = (RC/Δt). As α ranges from 0
to ∞, the pole location moves from 0 to 1, so it is always inside the unit circle contour,
Fig 24.2
The integral may be evaluated as 2 πi times the residue at this pole :
hn = 1
2πi(1+α) 2πi (α
1+α )n = 1
α (α
1+α )n+1 = Δt
RC (α
1+α )n+1 .
The samples h n are dimensionless, but to compare with our earlier RC work we write as in (24.6) that h n
= Δt gn to get
gn = 1
RC (α
1+α )n+1 . ( 2 4 . 1 9 )
In the case n < 0, in addition to the pole just mentioned, there is a pole or order n at z = 0. But when n < 0,
we can expand the contour out to a Gr eat Circle at infinity and the z-|n| factor then causes the integral to
vanish. The reason is that in this limit we have, with z = Reiθ,
∫GC dz z-|n|-1 = ∫0 2π Ri eiθ e-iθ(|n|+1) R-(|n|+1) = R-|n| { ∫0 2π e-iθ|n| } . (24.20)
But the integral {..} is finite, and as R →∞, R-|n| → 0 for n = -1,-2... so the GC integral is 0.
Thus, in terms of the Heaviside Step function,
g
n = g(tn) = (1/RC) (1+ α-1)-n-1 θ(n+ε) α = (RC/Δ t) (24.21)
where ε > 0 is any quantity less than 1 so we avoid the fact that θ(0) = 1/2. We can compare this to the
analog output of the true RC filter (4.10)
g(t) = (1/RC) e
-(t/RC) θ(t ) . (4.10)
Both the analog and digital filters demonstrate causality with the θ factors shown. However, the analog
filter decays in an exponential fashion, whereas the di gital decays in a geometric manner. We can rewrite
the digital result in this manner
gn = g(tn) = (1/RC) e-(n+1)ln(1+1/ α) θ(n+ε) . (24.22)
In the limit Δ t << RC we have α << 1 and then ln(1+1/ α) ≈ 1/α = Δt/(RC) and this result becomes
Chapter 3: Sampled Signals and Digital Transforms
81
g(tn) = (1/RC) e-(n+1)Δt/(RC) θ(n+ε) = (1/RC) e-tn+1/(RC) θ(n+ε) (24.23)
and, since t n+1 = tn + Δt ≈ tn, this replicates the analog result (4.10) in the small Δt limit.
So we learn that, in order to make our digita l RC filter produce the same results as an analog RC
filter, we must take Δt << RC, which is no big surprise since this was assumed at the start going from
(24.13) to the difference equation (24.14).
(f) Poles in H"(z) imply feedback and infinite impul
se response (IIR)
A general form for an implementable dimensionless transfer function H"(z) is a ratio of polynomials in z. We saw an example in the RC filter (24.17) above. So consider this general form,
H"(z) = [ Σ
n=0M anzn ]/ [Σn=0N bnzn] . (24.24)
In the special case that the denominator has the form Σ
n=1N bnzn = zk for some integer k ≥ M, we have
H"(z) = [ Σn=0M anzn] / (zk) = Σn=0M an zn-k = Σn=0M an (z-1)k-n .
Since k ≥ M, the exponents on (z-1)k-n are all non-negative. In this case we can write
H"(z) = a
0(z-1)k + a1(z-1)k-1 + ..... + a M (z-1)k-M
= a
0z-k + a1z-k+1 + ..... + a M z-k+M k ≥ M (24.25)
where all the (z
-1) exponents are non-negative integers. As we shall show by example below, since a
filter transfer function having this gene ral form has only positive powers of (z-1), it can be implemented
by hardware which has no feedback loops, and whic h therefore has an output which dies out some finite
number of clocks after the input dies out. If this filter is given an impulse as input, the output dies out
after a certain number of clocks. Thus, the filter has a finite impulse response and is then called a Finite
Impulse Response or FIR filter (example below).
Notice that in our special case H"(z) has a pole of order k at z = 0. When we later claim that transfer functions having poles must be implemented in hardwa re with feedback giving an infinite impulse
response, we are referring to poles not located at z = 0.
For our example we shall assume k = M = 2. Then,
H"(z) = a
0z-2 + a1z-1 + a2 = A + Bz-1 + Cz-2. (24.26)
Then if I"(z) and O"(z) are the input and output of our filter, O"(z) = H"(z) I"(z), (24.16)
we have
Chapter 3: Sampled Signals and Digital Transforms
82 O"(z) = [A + Bz-1 + Cz-2] I"(z) = A I"(z) + Bz-1 I"(z) + Cz-2 I"(z) (24.27)
Using the shift rule (24.11) we can translate the above equation into the time domain to get
on = A in + B in-1 + C in-2 . ( 2 4 . 2 8 )
These last two equations can be represented by these diagrams:
F i g 2 4 . 3 The time-domain diagram represents a piece of "har dware" wherein the output is developed with two D
flip-flop registers (clock period Δt) and three constant multipliers and two adders. This hardware circuit
has no "feedback" because no flip-flop output is ever involved in determining a flip-flop input.
Sometimes authors combine these two pictures, drawing the time domain register elements as boxes with
z
-1 labels inside. This convention appears in the following wiki picture (left),
http://en.wikipedia.org/wiki/Finite_impulse_response Fig 24.4
On the right we show the usual notation for attaching a clock to a register. In these pictures, each line (but not the clock) represents a "bus" of however many bits n is used to represent a digital sample. The triangle symbol for a multiplier suggests an "amplifier" whic h scales a signal. Later we shall use a simple X
placed on a bus to indicate multiplication by a constant . The flip-flops are clocked by a square-wave clock
pulse train having period Δt. At each positive edge of the clock signal, the value which the register input
has just before that edge is loaded into the register. The register output then holds that value constant until
the next positive clock edge.
Chapter 3: Sampled Signals and Digital Transforms
83 Here is the response of the circuit of Fig 24.3 to a unit impulse aligned with i 1 :
Fig 24.5
and it seems pretty clear that the impulse response is finite. We now consider a different example with poles. Suppose H"(z) is the inverse of that of the previous example,
H"(z) = z
2
Az2+Bz+C = 1
A+Bz-1+Cz-2 (24.29)
so now H"(z) has some non-zero poles (poles not at z=0). Then we get I and O swapped, so
I"(z) = [ A + Bz-1 + Cz-2 ] O " ( z ) . ( 2 4 . 3 0 )
Solve this for O"(z) in the following manner (solve for A O"(z) then divide by A),
O"(z) = (1/A) I"(z) + (- B/A) z
-1 O"(z) + (- C/A) z-2 O"(z) . (24.31)
Translating this to the time domain using rule (24.11) gives,
on = (1/A) i n + (- B/A) o n-1 + (- C/A) o n-2 . (24.32)
The corresponding drawings are these :
F i g 2 4 . 6 Here one can see the feedback: register inputs are dependent on register outputs. This is an Infinite
Impulse Response (IIR) filter, since the impulse response h
n carries on forever due to the feedback. See
Chapter 3: Sampled Signals and Digital Transforms
84 hn in (24.19) as another example: geometric decay wh ich never goes away completely. In that example
the transfer function H(z") has a pole as shown in ( 24.17). [ The response might go away in a real digital
circuit with a finite number of quantization bits. ]
(g) The Digital RC filter revisited
We can now draw up the
RC filter discussed above. We had in (24.16) and (24.17),
O"(z) = [H"(z) ]I"(z) = z/(1+α)
z - α/(1+α) I"(z) = 1/(1+α)
1 - z-1α/(1+α) I"(z) (24.33)
or
O"(z) – α
1+α z-1 O"(z) = 1
1+α I"(z)
or
O"(z) = α
1+α z-1 O"(z) + 1
1+α I"(z) (24.34)
Fig 24.7
As just noted above, the presence of a non-zero pole in H"(z) gives feedback.
Defining β ≡ 1/α = (Δt/RC), then in the regime in which the filter is accurate α >> 1 so β << 1, and then
1
1+α = 1/α
1+1/α = β
1+β ≈ β(1-β) ≈ β
( 2 4 . 3 5 )
α
1+α = 1
1+1/α = 1
1+β ≈ (1-β) ≈ e-β ,
which yields another form which sometimes a ppears in textbooks (Lam pages 509 and 503)
Fig 24.8
If we use these constants in (24.17) we get the rational polynomial transfer function
Chapter 3: Sampled Signals and Digital Transforms
85 H"(z) = zβ
z - e-β
and then from (24.18) with h n = Δt g(tn),
g(tn) = 1
RC e-nΔt/RC = 1
RC e-tn/RC θ(n+ε) . (24.36)
This is an accurate result even when α is not large ( β not small), so although this is not the design that
emerged from our small- Δt analysis in section (e) above, it is certainly a better design for a digital RC
filter. The two designs produce the same output for Δt << RC.
(h) Other circuits
Typical examples of IIR
filters having feedback are se rial scramblers and CRC ge nerators. One thinks of
the input sequence I"(z) as a huge polynomial in z, where the presence or absence of each power
represents a 1 or a 0. That is, think of the incomi ng stream as a superposition of unit impulses with
weights equal to the binary digits of the data str eam. The transfer function H"(z) = 1/polynomial, so we
write O"(z) = H"(z) I"(z) = I"(z)/polynomial. The out put stream is then the quotient of polynomial
division. One way to interpret the above discussion is as follows:
no poles → polynomial multiplication → no feedback → FIR
poles with no zeros → polynomial division → feedback → IIR
poles and zeros → simultaneous polynomial multiplication and division → feedback → IIR
This subject will be pursued more in a separate document.
(Z Transform summary box on next page)
Chapter 3: Sampled Signals and Digital Transforms
86 (i) Z Transform Summary
Z Transform ( 2 4 . 3 7 )
1. Let x(t) be any reasonable function.
2. Divide up the time axis into steps t n = n ∆t; let T 1 = ∆t and ω1 = 2π/T1.
3. We can think of samples x n = x(tn) for the above x(t). Alternatively, we can think of the
x n as some given sequence, and one could then construct an infinite number of functions
x(t) for which x n = x(tn).
4. In terms of x(t), the Z Transform and its inverse are given by
X"(z) = ∑
n = -∞∞
x(tn) z-n (24.3)
x ( t n) = 1
2πi ∫C dz X"(z) zn-1 . (24.4)
More generally, dispensing now with x(t) and using just x n,
X"(z) = ∑
n = -∞∞
xn z-n (24.3)
x n = 1
2πi ∫C dz X"(z) zn-1 (24.4)
The contour C goes once counterclockwise around the unit circle in the z-plane.
5. The Z Transform diagonalizes any convolution sum :
a n = ∑
m = -∞∞
∆t bn-m cm ⇔ A"(z) = ∆t B"(z) C"(z) (24.5)
a n = ∑
m = -∞∞
hn-m cm ⇔ A"(z) = H"(z) C"(z) (24.7)
where h n ≡ ∆t bn and H"(z) = ∆t B"(z)
6. The Z transform is related to the Digital Fourier Transform of box (23.5) by
X"(z) ≡ 1
∆t X'(ω) where z = eiωΔt (24.2)
Chapter 3: Sampled Signals and Digital Transforms
87
25. Amplitude Modulated Pulse Trains
In Section
14 we studied the spectrum of a simple pulse train made by superposing equal pulses x pulse (t)
with spacing T 1. We found that the Fourier Integral spectrum X( ω) of such a pulse train was given by an
infinite sequence of delta functi on spikes with amplitudes determin ed by an envelope function c( ω) which
is just a multiple of the spectrum of the pulse (summary box 14.12).
x(t) = ∑
n = -∞∞
xpulse (t - tn) t n = nT1 pulse train (14.1)
X(ω) = ∑
m = -∞∞
c(ω) 2π δ(ω - mω1) = ∑
m = -∞∞
cm 2π δ(ω - mω1) spectrum (14.9)
where c( ω) ≡ (1/T1)Xpulse (ω) = (1/T1) ∫-∞ ∞ dt xpulse (t) e-iωt (14.8) and (1.1)
The numbers c m = c(mω1) turned out to be exactly the complex Fourier Series coefficients.
Later in Section 20 we studied an am plitude-modulated pulse train in which x
pulse (t) = T1δ(t), and we
took note of the continuous spectrum of such a pulse train,
x(t) = y(t) d(t) = ∑
n = -∞∞
yn T1δ(t - nT1) y n = y(nT1) (20.3)
X(ω) = Y'(ω) = ∑
m = -∞∞
Y(ω - mω1) = Y(ω) + ∑
m ≠ 0
Y(ω - mω1) . (20.7)
where Y'( ω) was the Digital Fourier Transform of y(t) shown in items 3 and 4 of box (25.3).
In this section we shall combine both these ideas to obtain an amplitude modulated pulse train with an
arbitrary pulse shape x pulse (t). We continue to denote the amplitude modulated pulse train by x(t),
x(t) = ∑
n = -∞∞
yn xpulse (t -tn) . ( 2 5 . 1 )
What is the spectrum of this new pulse train? To find out, we insert (25.1) into (1.1),
X(ω) =
∫-∞ ∞ dt x(t) e-iωt = ∫-∞ ∞ dt [∑
n = -∞∞
yn xpulse (t - tn)] e-iωt
= ∑
n = -∞∞
yn ∫-∞ ∞ dt xpulse (t - tn)] e-iωt =∑
n = -∞∞
yn [ ∫-∞ ∞ dt' xpulse (t') e-iωt'] e+iωtn // t' = t-t n
= ∑
n = -∞∞
yn Xpulse (ω) e+iωtn = Xpulse (ω) ∑
n = -∞∞
yn e+iωtn
Chapter 3: Sampled Signals and Digital Transforms
88 where we used (1.1) to recognize X pulse (ω). Recall now the Digital Fourier Transform as summarized in
the box (23.5). From item 4 in that box, we can interpret the n sum above in this way ( Δt = T1)
∑
n = -∞∞
yn e+iωtn = 1
T1 Y'(ω) = Y " ( z ) ( 2 5 . 2 )
which says, apart from a constant factor, this sum is th e Digital Fourier Transform of y(t). [In this and the
following equations, we will try to show results in terms of both the Digital Fourier Transform Y'( ω) and
the Z Transform Y"(z) = Y'(ω )/T1. ]
From item 6 in that same box, we know that Y'( ω) = [ Y(ω) + ∑
m ≠ 0
Y(ω- mω1) ]. We conclude the above
calculation of the spectrum X(ω ) to find that, using the definition (14.8) of c( ω),
X(ω) = Xpulse (ω) 1
T1 Y'(ω) = c(ω) Y'(ω) = Xpulse (ω) Y"(z) . (25.3)
Comparing this to (20.7) quoted just above, we see that the spectral effect of replacing the T 1δ(t) pulse by
xpulse (t) is the addition of the pulse spectrum c( ω) = Xpulse (ω)/T1 as an overall factor. We then recover
the delta function result as a special case where c( ω) = T1/T1 = 1 as at the start of Section 20.
Thus we arrive at this very significant result which deserves its own box:
Amplitude Modulated Pulse Train ( 2 5 . 4 )
x(t) = ∑
n = -∞∞
yn xpulse (t -tn) (25.1)
X(ω) = c(ω) Y'(ω) = c(ω) [ Y(ω ) + ∑
m ≠ 0
Y(ω - mω1)] = Xpulse (ω) Y"(z) (25.3)
c ( ω) = (1/T1)Xpulse (ω) = (1/T1) ∫-∞ ∞ dt xpulse (t) e-iωt (14.8) and (1.1)
Y'( ω) = T1Y"(z) = T 1∑
n = -∞∞
yn e-iωnT1 projection = transform (23.5)
The boxed result above is one of the holy grails of th e spectral analysis of digital signals. Notice that by
selecting y n to vanish outside some range, the box applies to both infinite and finite pulse trains.
Chapter 3: Sampled Signals and Digital Transforms
89 Example 1: A finite pulse train
Consider this
finite sequence of y n samples
Fig 25.1
We can regard the red outline curve as an amplitude modulated pulse train whose pulse shape is a box of
height A = 1 and width τ = T1. The Fourier Integral Transform spectrum of this box from (9.2) is
Xpulse (ω) = T1 sinc(ωT1/ 2 ) . (9.2)
From box (25.4) the Fourier Integral Transform spectrum X( ω) of the pulse train is given by
X(ω) = (1/T 1)Xpulse (ω) { Y' (ω) } = Xpulse (ω) [ Y'(ω)/T1]
= sinc( ωT1/2) { Y' (ω) } = sinc(ω T1/2) { T1∑
n = -∞∞
yn e-iωnT1 } (25.5)
where Y' (ω) is the Digital Fourier Transform of the sequence y n as shown in (23.5).
Here is a Maple plot of | Y' (ω) | with T 1 = 1.
Fig 25.2
where we see the expected image spectra at N ω1 = N2π. There is a strong DC component at ω = 0 and the
peak there is the sum 19 of the y n values (all of which are positive) so X(0) = Y'(0) = 19 .
Chapter 3: Sampled Signals and Digital Transforms
90 Next we show in red a plot of | X( ω) | from (25.5) for the finite pulse train,
Fig 25.3
where the blue curve provides an outline of X(0) |sinc( ωT1/2)| . The red curve is the spectrum of the
physical red analog signal shown in Fig 25.1 and ther e are no image spectra. The sinc function in this
example crushes out the image spectra with its zeros.
Now we shall attempt some reconstructions.
First, we reconstruct the y
n from Y'(ω) using the inversion formula in box (23.5),
y k = 1
2π ∫
-ω1/2 ω1/2
dω Y'(ω) e+iωkT1 expansion = inversion (23.5)
which are in fact the y n we started with.
Second, we reconstruct the pulse train x(t) from X( ω) using the inversion formula (1.2),
x(t) = (1/2 π) ∫-∞ ∞ dω X(ω ) e+iωt expansion = inverse transform (1.2)
Chapter 3: Sampled Signals and Digital Transforms
91
Fig 25.4
which replicates our starting figure. The reason for Re (s) is that that s has a tiny imaginary part ~ 10-9
due to calculational error, and the plot routine requires a real function. Maple does the integral
analytically as shown (see (C.14) for signum ).
Example 2: The unit impulse and the sinc sum rule
Even the simplest cas
e is interesting. The y n sequence is taken as a unit impulse scaled by y 0
Fig 25.5
Xpulse (ω) = T1 sinc(ωT1/2) . // for box of unit height (9.2)
From box (25.4) the Fourier Integral Transform spectrum X( ω) of this "pulse train" is given by
X(ω) = (1/T 1)Xpulse (ω) { Y'(ω) }
= sinc( ωT1/2) { T1∑
n = -∞∞
yn e-iωnT1 }
= sinc( ωT1/2) { T1y0 } = sinc[π (ω/ω1)] { T1y0 } // = Y( ω) (25.6)
so in this example Y' (ω) = T1y0. If we regard the red plot as y(t), then X( ω) = Y(ω), the Fourier Integral
Transform of y(t). They are the same since this pulse train has only one pulse.
A plot of |X( ω)| = |Y(ω)| has a familiar look (T 1= 1, ω1= 2π, y0 = 1) :
Chapter 3: Sampled Signals and Digital Transforms
92
Fig 25.6
We have just noted that Y' (ω) ≡ y0T1 = 1. How exactly does the image spectrum equation in box (23.5)
item 6 , namely
Y'(ω) = [ Y(ω) + Σm≠0Y(ω- mω1) ] , ( 2 5 . 7 )
work out? First, we plot the right side of (25.7) limiting the sum range to m = -100 to 100 :
Fig 25.7
It appears that the shifted sinc functions are adding up to produce the constant Y' (ω) = 1. In terms of the
math, this must mean that
Y'(ω) = ∑
m = -∞∞
Y(ω - mω1) = yo T1∑
m = -∞∞
sinc[π (ω/ω1 - m)] = y oT1 (25.8)
which implies the following unusual su m rule, valid for any real x :
∑
m = -∞∞
sinc[π (x-m)] = 1 . (25.9)
Chapter 3: Sampled Signals and Digital Transforms
93 This result is sometimes quoted with x = 0 but it is in fact valid for any x. We have in effect proven the
sum rule with the above analysis, but as usual we woul d like to find verification. First process the sum as
follows,
∑
m = -∞∞
sinc[π (x-m)] = ∑
m = -∞∞
sin[π(x-m)]
π(x-m) = (1/π ) sin[πx]∑
m = -∞∞
(-1)m
x-m .
The sum on the right can be further processed,
∑
m = -∞∞
(-1)m
x-m = 1
x + [ ∑
m = -∞-1
+∑
m = 1∞
] (-1)m
x-m = 1
x + 2x ∑
m = 1∞
(-1)m
x2-m2 .
According to GR 142.3 (page 44),
we may replace
1
x
+ 2x ∑
m = 1∞
(-1)m
x2-m2 = π csc(πx)
and then we find that
∑
m = -∞∞
sinc[π (x-m)] = (1/π ) sin[πx] π csc(πx) = 1
which then verifies the sum rule (25.9).
Chapter 3: Sampled Signals and Digital Transforms
94 26. A simple application: Aperture Correction
The output of
a digital system usually involves a D/A converter followed by an analog post-filter. Let us
assume that this system is attempting to reproduce some reasonable analog waveform y(t). Assume that
each converted value is held as charge on a capacitor for some portion τ of the conversion period T 1, and
then the capacitor charge is instantly dumped to ground for the remainder of the period. Period τ is called
the aperture. In this way, we produce a signal x(t) that is a sequence of square pulses modulated by the
values y(t n) :
Figure 26.1. The smooth curve is y(t), the pulse train is x(t). Spacing is ∆t = T1. Width Fig 26.1
of each pulse is τ (the aperture), so duty cycle is τ/T1.
What is the spectrum of x(t)? It is an amplitude modul ated pulse train, so according to (25.3) the spectrum
of x(t) is
X(ω) = (1/T 1) Xpulse (ω) Y'(ω) . (26.1)
The pulse x pulse (t) is a square pulse of unit height, width τ , and with its left edge aligned with t=0. We
know the spectrum of this pulse from (9.2), but by (12.1) we must add a time-shift phase exp(-i ωτ/2)
because we are translating our earlier pulse τ/2 units to the right to make the left edge line up at t=0. Thus,
from (9.2) and (12.1),
X
pulse (ω) = τ sinc(ωτ/2) e-iωτ/2 ( 2 6 . 2 )
so the spectrum of x(t) is
X(ω) = (τ/T
1) sinc(ωτ /2) exp(-i ωτ/2) Y' (ω) . (26.3)
The magnitude of X( ω) is the product of two functions which we now illustrate, where the grey humps
represent Y( ω) and its images which combine to make Y' (ω), whatever it might be,
Figure 26.2. The humps are |Y'( ω)| and the red curve is ( τ/T1) |sinc(ωτ/2)|. Fig 26.2
Chapter 3: Sampled Signals and Digital Transforms
95
Presumably our low pass analog post-filter is going to select out the portion of the spectrum indicated by
the dotted lines in Figure 26.2. In this region we have,
X(ω) = (τ/T
1) sinc(ωτ/2) exp(-i ωτ/2) Y (ω) , ( 2 6 . 4 )
where Y( ω) is the main spectrum of Y'( ω). The factor ( τ/T1)sinc(ωτ/2) represents an undesired magnitude
distortion of the spectrum X( ω) due to the aperture τ. The phase φ(ω) = -iωτ/2 is a harmless linear phase
which just means the whole signal is delayed by time τ/2, as shown in Section 21 (b).
The distortion is at its worst when the aperture τ fills the entire period T
1, in which case the signal in
Figure 26.1 looks like a traditional stepwise fit to y(t). The distortion is worst because the zeros of sinc(ωτ /2) are at ω
m = m (2π/τ), so they are moved in as close as possible when τ is as large as possible, τ
= T1.
The distortion can be reduced by making τ as small as practicable. In this case, the zeros move out, and
the central hump of sinc( ωτ/2) is broad, so its drop-off during Y(ω ) is minimized. Of course the amplitude
(τ/T1) of X(ω ) also drops off as τ is made small, so there is a tradeoff.
In any event, there is still some distortion represented by sinc( ωτ/2) varying in the dotted region in Figure
26.2. Usually one attempts to correct for this apertu re distortion by building into the analog post-filter an
exactly compensating boost at frequencies near the cutoff region of the filter. Thus, if the post-filter would normally be some F( ω) cutting off in the region of the second dotted line in
Figure 26.2, a correcting filter would have the spectrum F( ω)/sinc(ωτ /2). This filter needs to know the
aperture time τ in addition to the cutoff frequency. Such a filte r is said to have "sine x over x correction".
Chapter 3: Sampled Signals and Digital Transforms
96 27. The Discrete Fourier Transform
Up to this point, x pulse (t) has always been considered a continuous function of time t. A simple pulse
train was formed as x(t) = Σ n xpulse (t-tn) and an amplitude modulated pulse train as Σn yn xpulse (t-tn)
where tn = nT1. Now for the first time we wish to consider a digital approximation to x pulse (t). That is
the main subject of this section, and we shall develo p it in analogy to Section 22 for the Digital Fourier
Transform.
In a slight reversal of our normal order of doing things, in section (a) we shall develop the Discrete
Fourier Transform (DFT) for a pulse train, then in section (b) we develop the Discrete Fourier Transform for an isolated pulse, this latter being the traditional form of the DFT.
(a) The Discrete Fourier Transform f
or a Simple Pulse Train x(t)
Recall from Section 15 the discussion of the Fourier Series Transform with complex coefficients c
m. This
was summarized in box (15.12) from which we quote,
Fourier Series Transform:
(complex form)
cm ≡ (1/T1) ∫-∞ ∞ dt xpulse (t) e-imω1t = (1/T1) ∫0 T1 dt x(t) e-imω1t (14.16) (27.1)
x(t) = ∑
n = -∞∞
xpulse (t - nT1) = ∑
m = -∞∞
cm e+imω1t (14.1) + (15.9) (27.2)
Here, x(t) is an infinite pulse train created by superposing pulses x pulse (t) at spacing T 1. Thus, x(t) is a
periodic function with period T 1.
We wish now to redefine our concept of interval ∆t. In our previous discussion, we set ∆t = T
1. Here we
wish instead to break up each interval T 1 into N pieces of size ∆t, so now we have:
∆t = T
1/N ω1 = (2π/T1) = 2π/(N∆t)
T1 = N ∆t (2 π/N) = ω1Δt
tn = n∆t ω1tn = n ω1Δt = n (2π/N) . (27.3)
Here t
n = n∆t represents a sequence of sample times for x( t). We still have our same periodic pulse train
as in (14.1) which we then evaluate at discrete times t = t n to get (27.5).
x(t) = ∑
m = -∞∞
xpulse (t-mT1) . (14.1) (27.4)
x(tn) = ∑
m = -∞∞
xpulse (tn-mT1) . ( 2 7 . 5 )
Chapter 3: Sampled Signals and Digital Transforms
97 There are N points t n per period T 1. If we rewrite (27.2) evaluated at these points, taking ω1 from (27.3)
and setting t = t n = n∆t , we get
x(tn) = ∑
m = -∞∞
cm e+imn(2π/N) . (27.6)
So far we haven't really done anything excep t examine x(t) at some sample points.
Following an approach similar to that of Section 22, consider now the following non-equation,
cm ≠ (1/T1) ∑
n = -∞∞
∆t xpulse (tn) e-imω1tn . (27.7)
Only in the limit ∆t → 0 does (27.7) become an equality, since it then reproduces (27.1). So let's define
something new called c' m that is equal to the right side of (27.7) for a specific finite Δ t = T1/N :
c'm ≡ (1/T1) ∑
n = -∞∞
∆t xpulse (tn) e-imω1tn . (27.8)
Using (27.3) this becomes
c'm ≡ (1/N) ∑
n = -∞∞
xpulse (tn) e-imn(2π/N) . projection = transform (27.9)
Recall that x pulse (t) is usually taken to be a pulse which vanishes outside a range of width T 1, and in this
case the sum in (27.9) has only N non-vanishing terms.
In the Fourier Series world, one can have any number of unique c m coefficients. For the c' m in (27.9) this
is no longer true. There are in fact only N unique values of c' m because they keep re peating. This is
because (27.9) implies that
c'm+kN = c'm for any integer k (27.10)
due to the fact that e-i(kN)n(2 π/N) = e-ikn(2π) = 1.
So c'm is a periodic digital sequence of period N.
We claim now (to be proven in section (b) below) that the correct expansion of pulse train x(t) to accompany projection (27.9) is the following:
x(t
n) = ∑
m = 0N-1
c'm e+imn(2π/N) . expansion = inverse transform (27.11)
Chapter 3: Sampled Signals and Digital Transforms
98
Due to the periodicity of c' m shown in (27.10), the expansi on (27.11) can also be written as
x(tn) =
⎩⎪⎨⎪⎧ ∑
m = -N/2N/2-1
c'm e+imn(2π/N) N even
∑
m = -(N-1)/2(N-1)/2
c'm e+imn(2π/N) N odd (27.12)
Proof of (27.12): For N even write the claimed result separa ting off the negative part of the series,
∑
m = -N/2N/2-1
c'm e+imn(2π/N) = ∑
m = -N/2-1
c'm e+imn(2π/N) + ∑
m = 0N/2-1
c'm e+imn(2π/N) .
In the first term replace m by m' = m+N to get
∑
m = -N/2-1
c'm e+imn(2π/N) = ∑
m' = N/2N-1
c'm'-N e+i(m'-N)n(2 π/N) = ∑
m' = N/2N-1
c'm' e+i(m')n(2 π/N)
where we have used (27.10) to say c' m'-N = c'm' and e+i(-N)n(2 π/N) = 1. Changing m' →m we then write
the our two-term sum as
∑
m = -N/2N/2-1
c'm e+imn(2π/N) = ∑
m = N/2N-1
c'm e+imn(2π/N) + ∑
m = 0N/2-1
c'm e+imn(2π/N) = ∑
m = 0N-1
c'm e+imn(2π/N) .
But this the sum in (27.11), so we have proven (27.12 ) for N even. The proof for odd N is similar and it is
left to the reader. We can now verify that our new transform approaches the Fourier Series Transform (27.1) and (27.2) in
the limit N →∞. The first line below is the large N (small Δt) limit of (27.9), while the second line is the
limit of (27.12) for even or odd N where N>>1 ( us e is made of the relati ons in (27.3) and lim
N→∞ tn =
limN→∞ (nΔt) = t) :
limN→∞ c'm = limN→∞ {(1/T1)∑
n = -∞∞
∆t xpulse (tn) e-imω1tn} = (1/T 1) ∫-∞ ∞ xpulse (t) e-imω1t = cm
limN→∞ x(tn) = limN→∞ { ∑
m = -N/2N/2
c'm e+imω1tn} = ∑
m = -∞∞
cm e+imω1t = x(t) ,
This new transform pair (27.9) and (27.11) we shall call the Discrete Fourier Transform of a Pulse
Train , as opposed to the Fourier Series Transform pair gi ven in (27.1) and (27.2). We reserve the term
Discrete Fourier Transform to refer to the transform of an isolated pulse. As we shall see in section (c) below, for this normal DFT, the su m in (27.9) becomes finite.
Chapter 3: Sampled Signals and Digital Transforms
99 In the Discrete Fourier Transform, time is "discrete", only t n appear, and therefore we only see functions
evaluated at these discrete time points,
x n ≡ x(tn)
x pulse,n ≡ xpulse (tn)
In contrast, the Fourier Series Transform has a continuous time variable t and functions x
pulse (t) and
pulse train x(t) appear.
Both transforms have discrete spectra as indicated by c m and c'm and this is because in both cases the pulse
train is a periodic function.
Just as a reminder, with the Digital Fourier Transform we dealt with sample sequences like y n = y(tn) =
y(nT1) where T 1 was the spacing between pulses composing a pulse train. In such a sequence, there is
only one sample per T 1 period. In contrast, with our current Discrete Fourier Transform discussion, y n =
y(tn) = y(nT 1/N) and there are N samples per T 1 time period. In both cases one could argue that the
sequence is a set of digital or discrete values, so the transform names are somewhat arbitrary.
(b) Proof of the Discrete Fourier Transform for a Simple Pulse Train x(t)
To show this
transform really works, we insert (27.9) for c' m into (27.11),
x(tn) = ∑
m = 0N-1
c'm e+imn(2π/N) = ∑
m = 0N-1
[(1/N) ∑
k = -∞∞
xpulse (tk) e-imk(2π/N)] e+imn(2π/N)
= ( 1 / N ) ∑
k = -∞∞
xpulse (tk) ∑
m = 0N-1
e+im(2π/N)(n-k) . (27.13)
We now quote an obscure identity proven in Appendix B which says
∑
m = 0N-1
e+ims(2π/N) = N ∑
m = -∞∞
δs,mN N > 0 s = integer . (B.1)
This is a discrete version (s = integer) of ( 13.2) (s = real) which we quote for comparison
∑
m = -∞∞
eims = ∑
m = -∞∞
2πδ(s - 2πm) - ∞ < s < ∞ . (13.2)
Setting s = n-k, (B.1) says [ since δn-k,mN = δk,n-mN ]
∑
m = 0N-1
e+im(2π/N)(n-k) = N ∑
m = -∞∞
δk,n-mN . (27.14)
Chapter 3: Sampled Signals and Digital Transforms
100
Then we find that
x(t
n) = (1/N) ∑
k = -∞∞
xpulse (tk) N ∑
m = -∞∞
δk,n-mN = ∑
m = -∞∞
∑
k = -∞∞
xpulse (tk) δk,n-mN
= ∑
m = -∞∞
xpulse (tn-mN) = ∑
m = -∞∞
xpulse (tn - mT1) . (27.15)
Since this reproduces the pulse train (27.5), we concl ude that indeed (27.11) is the expansion that
accompanies the projection (27.9).
At this point, we make a box to summarize the Discr ete Fourier Transform of a simple pulse train:
Discrete Fourier Transform of a Simple Pulse Train (27.16)
1. Let x pulse (t) be any reasonable pulse. Construct a simple pulse train x(t) with spacing T 1:
x ( t ) = ∑
n = -∞∞
xpulse (t - nT1) ⇒ x(t + mT 1) = x(t) (27.5)
By its construction, x(t) is periodic with period T 1. If x(t) is a known periodic function of
period T 1, a candidate for x pulse (t) is x(t) over any one period (and zero elsewhere).
2. Break up each T 1 interval into N steps of width ∆t = T1/N. Let t n = n∆t = (n/N)T 1 .
3. Define the Discrete Fourier Transform coefficients c' m by this projection = transform:
c ' m ≡ (1/N) ∑
n = -∞∞
xpulse (tn) e-imn(2π/N) m = integer (27.9)
Only N of these are unique because c' m is periodic in index m with period N:
c ' [m+nN] = c'm n = any integer (27.10)
4. The pulse train at sample points t n is then given by this expansion = inversion:
x ( t n) = ∑
m = 0N-1
c'm e+imn(2π/N) (27.11) but see also (27.12)
From this last result we may confirm that x(t n + mT1) = x(tn) so x is indeed periodic.
Chapter 3: Sampled Signals and Digital Transforms
101
(c) The Discrete Fourier Transform for an Arbitrary Pulse
The results of box (
27.16) apply to any sampled pulse x pulse (t) . We could consider, for example, the set
of xpulse (tn) functions which are non-zero only for t n = nΔt lying inside some limited temporal range
indicated by A ≤ n ≤ B. For such functions (27.8) will have the form,
c'm ≡ (1/N) ∑
n = AB
xpulse (tn) e-imn(2π/N) . (27.17)
The inversion formula continues to be (27.11)
x(t
n) = ∑
m = 0N-1
c'm e+imn(2π/N) . (27.11)
If the range (A-B) Δt > T1, the above stated transfor m is valid, but the pulse x pulse (tn) cannot in this case
have an arbitrary shape. This is because (27.11) forces x(t n + T1) = x(tn), meaning x(t) is periodic. For
example, if (A-B) Δt ≈ 1.3 T1, then the portion of x pulse (tn) in (T1, 1.3T1) must be a replication of the
portion of x pulse (tn) in (0, 0.3T 1). If we want a DFT pulse transform th at allows for arbitrary pulse shape,
we must restrict A and B so that (A-B) Δt ≤ T1. We can of course consider pulses which are restricted to
(A-B)Δt < T1 to be special cases of pulses defined on (A-B) Δt = T1, where we just add zero padding to
arrive at the interval T 1.
Therefore we restrict A,B so that (A-B) Δt = T1 or (A-B) = T 1/Δt = N.
In this way, we arrive at this special case of the transform of the box (27.16) which applies to an
arbitrary pulse of width T 1 (that is, a pulse having only N discrete values)
c'm ≡ (1/N) ∑
n = 0N-1
xpulse (tn) e-imn(2π/N) m = 0,1...N-1 (27.18)
/ / ( 2 π/N) = ω1Δt
xpulse (tn) = ∑
m = 0N-1
c'm e+imn(2π/N) n = 0,1,...N-1 (27.11) (27.19)
This is the official Discrete Fourier Transform (DFT) . Due to the periodicity property (27.10), the sum
in (27.17) could be taken over any set of N adjacent steps, and without loss of generality we take these N steps to be 0,1,...N-1. If one were to regard the pulse as being translated to some other set of N steps like n
= -3,-2,-1,0,1,... N-4, the coefficients c'
m would be exactly the same apart from a simple m-dependent
phase. For example, let x' pulse be the translated pulse. Then,
Chapter 3: Sampled Signals and Digital Transforms
102 d'm ≡ (1/N) ∑
n = -3N-4
x'pulse (tn) e-imn(2π/N) = (1/N) ∑
n = -3N-4
xpulse (tn-3) e-imn(2π/N)
= (1/N) ∑
n' = 0N-1
xpulse (tn') e-im(n'-3)(2 π/N) // n' = n+3
= e+i3m(2π/N) { (1/N) ∑
n' = 0N-1
xpulse (tn') e-imn(2π/N)} = e+i3m(2π/N) c'm
= e+i3(mω1)Δt c'm ( 2 7 . 2 0 )
This result is a reflection in the current context of the time-shift rule (12.1) which we restate here as
x(t + 3 Δt) ↔ e+i3ωΔt X(ω ) (12.1)
Note that c'
m refers to the spectral frequency m ω1.
We now summarize the DFT in a box:
Discrete Fourier Transform for an Arbitrary Pulse (27.21)
1. Let x pulse (t) be an arbitrary reasonable pulse defined for t in (0,T 1).
2. Break up T 1 into N steps of width ∆t = T 1/N. These relationships hold
∆t = T 1/N ω1 ≡ (2π/T1) = 2π/(N∆t)
T 1 = N ∆t (2 π/N) = ω1Δt
t n = n∆t ω1tn = n ω1Δt = n (2π/N) (27.3)
Thus, the sequence values of interest are x pulse (tn) for n = 0,1,2...N-1.
3. Define the Discrete Fourier coefficients c' m by this projection = transform:
c ' m ≡ (1/N) ∑
n = 0N-1
xpulse (tn) e-imn(2π/N) m = 0,1...N-1 (27.18)
4. The accompanying expansion = inverse transform is given by
x pulse (tn) = ∑
m = 0N-1
c'm e+imn(2π/N) n = 0,1,...N-1 (27.19)
Chapter 3: Sampled Signals and Digital Transforms
103 In the above box one could of course replace x pulse (tn) by some generic function x(t n) defined on (0,T 1),
and one could go further and write x(t n) as xn to get this more common statement of the DFT:
c'm ≡ (A/N) ∑
n = 0N-1
xn e-imn(2π/N) m = 0,1...N-1 projection = transform
xn = (1/A) ∑
m = 0N-1
c'm e+imn(2π/N) n = 0,1,...N-1 expansion = inverse transform (27.22)
Here we have added an arbitrary constant A to th e first equation and 1/A to the second which maintains
the validity of the DFT transform pair. If one takes A=N, the 1/N factor moves to the second equation.
Another choice is A = N to make the two equations symmetrical. The transform pair is also valid if the
phase signs are switched, just as with the Fourier Inte gral Transform. This would be associated with a
version of (B.1) having the opposite phase sign obtained by just complex conjugating (B.1).
(d) Comments on the Discrete Fourier Transform
One
might wonder about the purpose of the Discrete Fourier Transform of a Pulse Train, and its relation
to earlier transforms. This can be illuminated by a simple set of pictures.
First, go back to the Section 2 analysis of a making a pulse train x(t) by superposing shifted copies of
pulse x
pulse (t). Imagine, as in the discussion at the end of Section 14, that x pulse (t) is a Gaussian which
of necessity extends beyond the domain of one period T 1. Here is a picture of this pulse:
Figure 27.1. The pulse x pulse (t). Bars are distance T 1 apart. Fig 27.1
If we now superpose these pulses, we get the followi ng pulse train x(t), as was shown in Fig 14.2,
Figure 27.2. The function x(t) is the heavy curve. It is the sum of the gaussians. Fig 27.2
This picture gives us a chance to repeat a point made earlier, namely that x pulse (t) is not unique. One
could use instead a portion of the heavy curve between any adjacent pair of bars.
Chapter 3: Sampled Signals and Digital Transforms
104 The heavy curve is our pulse train, and we have cho sen the most complicated case, that where the pulses
overlap. Typically they do not overlap.
Now, the heavy curve is a periodic continuous function of time x(t), and it has in pr inciple an infinite set
of Fourier Series coefficients c
m. These coefficients are really dete rmined from the underlying pulse
xpulse (t). In general, it takes an infinite number of time points to represent the smooth function x pulse (t),
so there are an infinite number of coefficients c m in the "transformed space" wh ere these coefficients live.
We know that the transform of the Gaussian x pulse (t) is a Gaussian X pulse (ω), and we know that the
Fourier coefficients are given by
cm = (1/T1) Xpulse (mω1) where ω1 = 2π/T1 .
One can visualize (see Fig 14.1 ) the infinite set of the c
m as tracing the envelope of this gaussian
Xpulse (ω). Of course it may happen that many of the c m vanish if x pulse (t) has simple harmonic content.
The point is that there could be an infinite number of c m.
This infinitude matches the infinitude of real points along the pulse x
pulse (t). If we consider the regular
Fourier integral spectrum X pulse (ω) in its own right, we again have an infinitude of complex numbers
needed to describe the pulse in th e transformed space, subject to X(- ω) = [ X(ω)]* of (7.1) which knocks
down this complex double infinity to a single infinity , balancing the time side of the transform.
Having said all this, we are now ready to move from analog to digital. Consider the same pulse x
pulse (t)
evaluated only at the discrete points t n, so the pulse is now represented by this sequence of numbers:
x
pulse (tn) t n = n ∆t
and we assume that there are N sample points in each period T
1. In our figures below, N = 8, and we
approximate the tail of the Gaussian with a few extra points.
Here then is a picture of the set of numbers x pulse (tn) which describe our pulse:
Figure 27.3. Pulse is now a set of 18 numbers x n(pulse) . N = 8 Fig 27.3
Now as before, build a digital pulse train by superposing pulses:
Chapter 3: Sampled Signals and Digital Transforms
105
Figure 27.4. Digital pulse tr ain represented by the fat ha tched bars. Fig 27.4
In this figure, the thin dark bars are the numbers which describe the pulse. The fat hatched bars represent
the sum of the thin bars -- remember that we have overlap here. Note that the resulting sequence -- the fat bars -- fo rm a periodic sequence, just as we had a periodic
function given by the heavy curve in Figure 27.2. Note also that again we could have used an "equivalent
pulse sequence" here consisting of just th e set of 8 fat bars in one interval.
Thus, although our original pulse contained 18 number s, the minimal pulse contains only 8 numbers. If
we now compute the Discrete Fourier Transform coefficients c
m' according to the formula in box (26.15),
c'm ≡ (1/N) ∑
n = -∞∞
xpulse (tn) e-imn(2π/N)
we find that only 8 of the c' m are unique because of the translation rule shown in the same box. Select
those with m = 0,1,2,3,4,5,6,7. We should be happy to find that it takes only 8 numbers in the transform
space (where the c m' live) to represent the 8 numbers in the time domain which represented our pulse in its
minimal representation -- the 8 fat bars in period T 1. For this minimal pulse, there are only 8 non-
vanishing terms in the above sum. Thus, the c m' are related to the eight fat bar heights by a set of numbers
which form an 8x8 symmetric matrix, namely
M
mn = (1/N) e-imn(2π/N)
in terms of which we have
c'm = Σm Mmn xpulse (tn)
or c' = M x
pulse (27.18)
xpulse = M-1 c' = N M* c ' (27.19)
Of the 64 matrix elements of M, only 8 are unique, a nd these 8 elements lie equally spaced on a circle of
radius (1/N) in the complex plane.
As a possible application of the Discrete Fourier Transf orm (DFT), consider some sort of digital circuit
that puts out a set of numbers that repeat after every N numbers. Perhaps this is what a scrambler does with a constant input. In this case, one can think of the set of N numbers as tracing the envelope of a pulse
Chapter 3: Sampled Signals and Digital Transforms
106 xpulse (t). The appropriate "frequency domain" transform of this repeating sequence of numbers is the
DFT. In the frequency domain we get a finite set of N numbers c m' as the transform.
Just as with regular Fourier series coefficients, the DFT coefficients c' m are a measure of the frequency
content of the signal. Recall that the pulse train x(t) is mapped out by:
x(tn) = ∑
m = 0N-1
c'm e+imn(2π/N) = ∑
m = 0N-1
c'm e+imω1tn ω1 = (2π/T1)
One should think of n taking lots of values and trac ing out the envelope of the function x(t). Clearly,
coefficient c' m is the weight of frequency component m ω1. So c'0 measures the DC component, and c' 1
measures the amount of frequency component ω1 and so on.
There is a limit on how high a frequency component on e can have. Consider a sine wave with period ∆t =
the sample spacing. It would have the same value at every sample point, and would thus show up in the
DC component. The frequency corresponding to period ∆t is ωN = Nω1. This is why c' N = c'0. In a similar
fashion, potential frequencies ω n with n>N are also "aliased" down into lower frequencies according to
the translation rule for the c' m. Thus, the highest frequency we can really have is (N-1) ω1.
So this gives a reasonable "Fourier explanation" of why there are a finite number of distinct c' m
coefficients involved in the spectral expansion above for x(t n).
We repeat one more time an important fact stress ed earlier: as N (the number of sample points per T
1
interval) increases, the number of DFT coefficients c' m increases as well, and these c' m becomes closer and
closer to the Fourier Series coefficients c m. In the limit N → ∞, c'm = cm, and the DFT and the Fourier
Series exactly align.
For finite N, the c' m differ from the c m in exactly the same way th at the area under a stepwise
approximated curve differs from the area under the smoot h curve. This fact follows directly from the
definitions of the c' m and cm.
Chapter 4: Some Practical Topics
107 Chapter 4: Some Practical Topics
This chapter
applies the results of earlier chapters to a few simple test situations and applications. By
providing some wordy discussion of seemingly mundane topics, we attempt to prop up our so-far mostly mathematical approach to Fourier analysis. It is in matters like these that one's understanding is really put
to the test.
28. Do FIR filters have linear phase?
We shall
show in two different ways that a FI R filter has linear phase provided it has symmetric
coefficients. The first method is direct, the second more intuitive. Method 1
We saw in (21.5) how a digital filter is represented by a set of numbers b n. Recall the Z transform
projection,
B"(z) = ∑
n = -∞∞
bn z-n . // FIR filter
Since z lies on the unit circle, we may represent it as z = eiθ as in (24.1). Thus,
B"(eiθ) =∑
n = -∞∞
bn e-inθ . // FIR filter
Assume that b n is a finite set b 0, b1, b2....bN. Then,
B"(z) = ∑
n = 0N-1
bn z-n = b0 + b1 z-1 + b2 z-2 + ... bN-1z-(N-1) . (28.1)
Assume next that the set of b n is "symmetric" such that b 0 = bN-1, b1 = bN-2 and so on.
It is not hard to obtain our conclusion using gene ral N, but it is a lot easier to see what is
going on if we pick some sample N values. Let N = 5. Then we have
B"(z) =
∑
n = 04
bn z-n = b0 + b1 z-1 + b2 z-2 + b3z-3 + b4z-4
= b 0 + b1 z-1 + b2 z-2 + b1z-3 + b0z-4 // assume symmetric
= z-2 (b0z2 + b1 z + b2 + b1z-1 + b0z-2)
= z-2 [b0(z2 + z-2) + b1(z +z-1) + b2] .
Chapter 4: Some Practical Topics
108 Now set z = eiθ and continue along,
= e-2iθ [b0(e2iθ + e-2iθ) + b1(eiθ + e-iθ) + b2]
= 2e-2iθ [b0cos(2θ) + b1cos(θ) + b2] . 2 = (N-1)/2
The phase of this filter is -2 θ. If we used N = 7, a repeat of the above analysis would give
B"(z) = 2e-3iθ [b0cos(3θ) + b1cos(2θ) + b2cos(θ) + b3] 3 = (N-1)/2
with a phase of -3 θ. For a general odd value of N, the z phase comes out being -i[(N-1)/2] θ .
Now consider even values of N. For N = 4 we have
B"(z) =
∑
n = 03
bn z-n = b0 + b1 z-1 + b1 z-2 + b0z-3
= z-1.5 ( b0z1.5 + b1 z.5 + b1 z-.5 + b0z-1.5)
= z-1.5 [ b0(z1.5 + z-1.5) + b1 (z.5 + z-.5)]
= 2 e-i1.5θ[ b0cos(1.5θ) + b1cos(0.5θ)] . 1.5 = (N-1)/2
For N = 6 the result would be = 2 e
-i2.5θ[ b0cos(2.5θ) + b1cos(1.5θ) + b2 cos(0.5θ)] 2.5 = (N-1)/2 .
For a general even value of N, the phase comes out being -[(N-1)/2] θ which is the same as the phase for
the general odd N value. Thus we have shown that, for general N, and using θ = ωΔt from (24.1),
B"(z) = 2 e
-iωΔt(N-1)/2 [ real sum of cosine terms ] (28.2)
According to the definition of "filter phase" in (21.14), our filter B"(z) has
phase = + [(N-1)/2] Δt ω . ( 2 8 . 3 )
Since this phase is linear in ω, our symmetric-coefficient digital FIR filter has "linear phase". Using the
same definition of group delay used for an analog filter in (21.19), the group de lay for such a linear phase
digital filter is
τ
d = d(phase)/d ω = Δt(N-1)/2 = a constant (28.4)
so we expect a symmetric FIR filter to exhibit good fidelity when it acts on an input signal.
Chapter 4: Some Practical Topics
109 Method 2
Consider the Fourier integral spectrum X( ω) of a real-valued pulse x(t) that is symmetrical and centered at
t=0. Since x(-t) = x(t), we can fold the negative porti on of the dt integration in (1.1) over to the positive
side. Doing this gives x(t) times e-iωt + e+iωt = 2cos(ωt). Thus, everything is real, and X(ω) must
therefore be real. We have alr eady seen several examples of this: δ(t) gives X( ω) = 1, a square pulse gives
(Aτ) sinc(ωτ /2).
If we displace the pulse to the right by some amount of time M ∆t, then X( ω) is no longer real, it picks
up the usual shift phase from (12.1) which here would be exp(-i ωM∆t).
We now construct a digital filter O"(z) = B"(z)I"(z). The input will be a unit pulse at time 0 so i n(t) =
δn,0 and therefore I"(z) = 1 from (24.8) and (24.9) w ith m=0. The output of the filter is then O"(z) =
B"(z). Consider an N=3 filter with B"(z) = a + bz-1 + az-2. This corresponds to output signal o(0) = a,
o(∆t) = b and o(2 ∆t) = a. Since this is a symmetric pulse centered at t = Δt, we know from the previous
paragraph that the spectrum of this pulse has a phase e-iΔt relative to the real spectrum of a similar pulse
centered at t = 0. Next, consider an N=4 filter with coefficients a,b,b,a. The output will be sequence
a,b,b,a centered at t = (3/2) Δt, so its spectral phase will be e-i(3/2) Δt relative to that of similar pulse
centered at t = 0. In both cases, we see that the output pulse is centered at t = [(N-1)/2] Δt, so this must be
the group delay of a symmetric filter with N coefficients. The filter phase must then be the function φ =
[(N-1)/2] Δt ω, which is linear in ω, in agreement with the result of the more formal Method 1.
Chapter 4: Some Practical Topics
110 29. A Simple Digital Low-Pass Filter
This section
describes a particular implementation of a digital low-pass filter. There is a whole world of
such filters, and this design is meant only as an illustra tion. In Section 30 the filte r described here will be
used as a 4x oversampling interpolation filter for a D/A converter output design.
The ideal "brick wall" filter has this spectrum,
Fig 29.1
Recall our box-shaped pulse in the time domain (height 1, width τ) and its spectrum
x(t) = [ θ (t + τ/2) - θ(t - τ/ 2 ) ] (9.1)
X(ω) = τ sinc(ωτ/ 2 ) . (9.2)
For this x(t), (1.1) gives the X( ω) shown. If we try X( ω) = [ θ (ω + ω
c) - θ(ω - ωc) ] in (1.2), we know the
result will be (1/2π ) * 2ωc sinc(tωc), just swapping the variables t ↔ω and τ/2→ωc. We thus obtain the
following brick wall filter B( ω) of Fig 29.1 and it associated time-domain pulse shape b(t),
B(ω) = [ θ (ω + ω
c) - θ(ω - ωc) ] ( 2 9 . 1 )
b(t) = ( π/ω
c) sinc(ωct) . // b(t) is even in t (29.2)
As mentioned in Comment (1) at the end of Section 3, since b(t) is a filter kernel, it has dimensions of
inverse time and the filter spectrum (transfer function) B( ω) is dimensionless.
Recall the Convolution theorem (3.6),
o(t) =
∫-∞ ∞ dt' b(t-t')i(t') ⇔ O(ω) = B(ω) I(ω ) (3.6)
where i(t) is the input to a filter and o(t) the output. Using i(t) = δ(t), we get o(t) = b(t), so b(t) is the
impulse response of the filter. [In this special situa tion, we are not following our Section 3 Comment (1)
convention since this i(t) has dimensions of inverse time instead of being dimensionless.]
A digital filter approximating (3.6) has this form,
o(t
n) = ∑
m = -∞∞
∆t b(tn - tm) i(tm) where t n = n ∆t . (21.4)
or
Chapter 4: Some Practical Topics
111 on = ∑
m = -∞∞
∆t bn-m im . ( 2 9 . 3 )
If we set the input to a digital unit impulse at t = 0, i(t m) = im = δm,0, then the output is
on = ∆t bn ( 2 9 . 4 )
so ∆t b
n is the unit impulse response of the digital f ilter. Recalling from (3.2) the symmetry of the
convolution equation (or just set m' ≡ n-m) we can write (29.3) instead as
on = ∑
m = -∞∞
∆t in-m bm . ( 2 9 . 5 )
In equations (29.3) and (29.5) we think of i n and on as being dimensionless, and b n as having dimensions
of inverse time so ∆t bn is dimensionless. In (24.6) we used h n ≡ ∆t bn but here we stick with b n.
Example: We assume these parameters, since the resulting filter will be useful later on :
T1 = 1
ω1 = 2π/T1 = 2π
ωc = ω1/2 = π // Nyquist rate, see end of Section 20
Δt = T1/4 = 1/4 // filter will be clocked at 4x rate (29.6)
Maple computes the b
m from (29.2) as follows (tiny offset added to avoid divide by zero in the hand-made
sinc function)
( 2 9 . 7 )
The filter has symmetric coefficients b -n = bn since b(t) in (29.2) is even in t, so it will exhibit a linear
phase response as discussed in Section 28. This in turn means a constant group delay as in (28.4).
Chapter 4: Some Practical Topics
112
We can verify the locations of these points on the sinc curve,
Fig 29.2
The digital filter implementation usin g (29.5) is just this equation ( Δt = 1/4),
4o
n = ∑
m = -1010
in-m bm = inb0 + ∑
m =110
in-m bm + ∑
m =-1-10
in-m bm
= i nb0 + ∑
m =110
in-m bm + ∑
m =110
in+m bm // since b -m = bm
= i nb0 + ∑
m =110
[in-m + in+m ] bm
= i nb0 + [in-1 + in+1 ] b1 + [in-2 + in+2 ] b2 + .... + [i n-10 + in+10 ] b1 . (29.8)
This equation is implemented in the following piece of hardware,
Chapter 4: Some Practical Topics
113 F i g 2 9 . 3
Notice that the registers on the top march samples left to right, while those on the bottom go right to left.
The clock lines are not drawn; all registers are clocked with period Δt = T
1/4 = 1/4. The registers (D flip-
flops) sometimes appear as boxes containing z-1 as in Fig 24.4. If a register input is i n+1 in the middle of
a clock period, that register's output is the previously clocked sample i n. Usually the clock is a square
wave signal and the registers transfer input to outpu t on the positive clock edges of the square wave. The
circled plus signs are adders, while lines marked with an X indicate multiplication by the constant
appearing next to the X. All lines indicate busses cont aining some number of bits used to represent the
digital signals, perhaps 8, 10 or 12.
An actual design might be done a bit differently using pipelining registers to avoid the large
combinatoric delay built up through the long string of adders at the bottom (if speed is an issue).
We now wish to compute the spectrum ("transfer functi on") of this digital filter to see how close it comes
to being a "brick wall" with cutoff at ω
c. Basically, we want to compute the Digital Fourier Transform
spectrum associated with the finite sequence of samples b m. Recall that Δt bm is the dimensionless impulse
response of the filter.
b
m = (π/ωc)sinc(ωcmΔt) m = -10 to 10, else 0 21 "taps" (29.9)
We know that if we include terms from m=- ∞ to m=+∞, we shall obtain for the Digital Fourier Transform
spectrum B'( ω) an exact brick wall box-shaped filter with image boxes going off to the left and right. But
for m limited to the range (-10,10), which makes use of 21 b m coefficients (a 21 "tap" filter), we expect to
get only an approximation to Fig 29.1. From box (23.5),
B'(ω) ≡ (T
1/4)∑
n = -1010
bn e-iωnT1/4 . (29.3)
Here is a plot of the central peak for a filter of 21 taps (blue) compared with one of 101 taps (red). Notice
that the cutoff frequency is at ωc = π, and as usual plots are of | B'( ω) |, so the ringing on both sides of the
peak is "rectified",
Fig 29.4
Chapter 4: Some Practical Topics
114
The 21-tap blue curve "brick wall", though not perfect, is pretty good, giving a fairly steep edge while
maintaining a linear phase characteristic. Below is the same plot with a wider range of ω (called w in the Maple code ), showing the two nearest
image spectra. Because we have selected Δ t = T
1/4 = 1/4 (4x oversampling) for this filter, the first image
spectrum on the right is centered at 4 ω1 = 4(2π) ≈ 25 .
F i g 2 9 . 5 In this Section we have described a particular exam ple of a low-pass filter. One problem that is evident
from the plots above is that there is ringing in the spectra, known as "the Gibbs phenomenon". This can be reduced by multiplying the filter coefficients by a symmetric Gaussian-like weighting or "window"
function. There are many proposed window functions associated with names Bartlett, Hann, Kaiser,
Hamming, Blackman, etc.
Chapter 4: Some Practical Topics
115 30. Use of Oversampling in a D/A Converter Design
Here
we discuss the design of an output circuit wh ich contains a D/A converter. We are not concerned
with the internal design of the actual D/A converter "chip" itself.
(a) A very simple D/A converter
Consider the following finite-length digital signal, which we assu
me has time spacing T 1,
y
n (n=-5..5) = { 1,2,3,3,2,1,1/5,-1,-2,-2,-1}
Fig 30.1
All samples other than those shown are 0.
Using an arbitrary pulse shape x pulse (t), we can construct an amplitude-modulated pulse train x(t)
whose spectrum is X( ω), as shown in summary box (25.4),
x(t) = ∑
n = -∞∞
yn xpulse (t -tn) t n = n T1 y n = y(tn) (30.1)
X(ω) = (1/T1)Xpulse (ω) Y'(ω) , ( 3 0 . 2 )
where from summary box (23.5),
Y'(ω) ≡ T
1∑
n = -∞∞
yn e-iωnT1 . // Y"(z) = Y'(ω )/T1 (30.3)
If we were to select x pulse (t) to be a box of height 1 and width T 1, then we could regard the stair-step
outline function shown in Fig 30.1 as a candidate analog signal y(t) whose samples are the y n. In this
special case, y(t) = x(t) so Y( ω) = X(ω). From (9.2) and (12.1) we have
Xpulse (ω) = Xbox(ω,T1) = e-iωT1/2 T1 sinc(ω T1/2) (30.4)
Chapter 4: Some Practical Topics
116 where the phase arises since the box (0,T 1) = (0,1) is shifted T 1/2 to the right of the position of the
symmetric box used in Section 9. Then using (30.2),
Y(ω) = X(ω) = e-iωT1/2 sinc(ωT1/2) Y'(ω) = e-iωT1/2 sinc(ω T1/2) T1∑
n = -∞∞
yn e-iωnT1 . (30.5)
Y(ω) is the Fourier Integral Transform spectrum of the stair-step analog signal in Fig 30.1. Here are plots
first of |Y'( ω)| from (30.3) using Fig 30.1 data, and second (red) of |Y( ω)| using (30.5).
| Y ' ( ω)| Fig 30.2
| Y ( ω)| Fig 30.3
We see the expected image spectra in Y'( ω) in the first plot, but these spectra are quite suppressed in the
second plot due to the taming effect of the sinc function zeros in (30.5), as shown in blue.
The above discussion describes the output of the following simple n-bit D/A converter design.
Fig 30.4
Chapter 4: Some Practical Topics
117 The purpose of the register on the left is to provide a stable signal on bus B to the D/A converter. We
assume that the D/A converter is "glitch free" on its output, and just does what it should do.
(b) Oversampling just the D/A converter
We now trivially
modify the above design by cha nging the D/A clock from clk1x to clk4x which runs 4X
faster than clk1x,
Fig 30.5
The D/A converter is now "4x oversampling" the da ta on the B bus. Besides making the D/A converter
work harder, it seems clear that the output signal x(t) will be exactly the same as shown in Fig 30.1. Thus
the plots of | Y'( ω) | and | Y( ω) | = | X(ω) | shown above apply to this design as well as that of Fig 30.4.
It is useful, nevertheless, to think of the output of the oversampled design as follows, where the nonvanishing y
n amplitudes are numbered n = -20 to + 23,
Fig 30.6
Now the output rectangles are 1/4 as wide because th e D/A is clocking 4X faster. The analog outline is
the same, but our analysis will be different. We shall now compute X( ω) in terms of the thin rectangles of
Fig 30.6. Looking at the four y 0 = 1 samples to the right of the verti cal axis, those four boxes will make
this contribution to the spectrum
X(ω) = X
box(ω,T1/4) [... + y 0 + y0 e-iω(T1/4) + y0 e-2iω(T1/4) + y0 e-3iω(T1/4) + .... ] (30.6)
Each thin box has a phase e-iω(T1/4) relative to the box to its left due to (12.1). Since the pulse is 4x
narrower than before, X box(ω,T1/4) is given by (30.4) with T 1→T1/4.
We can write the square bracket in (30.6) as
Chapter 4: Some Practical Topics
118 [ 1 + e-iω(T1/4) + e-2iω(T1/4) + e-3iω(T1/4) ] y0 ≡ F(ω) y0 . (30.7)
Every group of four terms will ha ve this same common factor F( ω), so we can factor it out of the entire
sum. The sum now looks like this:
X(ω) = Xpulse (ω,T1/4) F(ω ) { ..... y0 + y1 e-4iω(T1/4) + y2 e-8iω(T1/4) + ..... } . (30.8)
But now the expression in curly brackets is exactly Y'( ω)/T1 of (30.3), our original Digital Fourier
Transform spectrum of y(t). Thus we conclude that
X(ω) = (1/T 1) Xpulse (ω,T1/4) F(ω ) Y'(ω)
= [ e-iωT1/8 (1/4) sinc(ω T1/8) ] F(ω) Y'(ω ) . (30.9)
So this is X( ω) as computed in terms of the thin boxes of Fig 30.6. But we already argued that
oversampling the D/A does not change the analog output signal x(t) or its spectrum X( ω), so somehow the
expressions in (30.9) and ( 30.5) must be the same. This can only be true if
e-iωT1/2 sinc(ωT1/2) = [ e-iωT1/8(1/4) sinc(ω T1/8) ] F(ω ) ?
or, writing out the sinc functions,
e
-iωT1/2 sin(ωT1/2)(2/ωT1) = [ e-iωT1/8(1/4) sin(ωT1/8) (8/ω T1) ] F(ω ) ?
or e
-iωT1/2 sin(ωT1/2) = [ e-iωT1/8
sin(ωT1/8) ] F(ω ) ?
To verify this fact, we define z ≡ e
-iωT1/4 ( variable for the Z transform). The above then reads
z2 (z-2-z2) = [z1/2
(z-1/2 - z1/2) ] [ 1 + z + z2 + z3 ] ?
or (1-z
4) = (1 - z)( 1 + z + z2 + z3) ?
But this is a standard factorization so we find that X( ω) is indeed the same either way we compute it. For
some other oversampling factor like 6x or 8x, the ve rification is similar. We have just shown that
X
box(ω,T1) = F(ω) Xbox(ω,T1/4)
which we can think of as saying the product of two filters on the right gives the one on the left.
(c) Add zero-stuffing to reduce aperture
We now add a m
ultiplexor to our previous D/A co nverter design which causes the first sample in each
group of four samples to pass through, but "grounds" the last three samples:
Chapter 4: Some Practical Topics
119
Fig 30.7
The output of this design is the following analog signal with pulse amplitudes y n ( where n = -20 to +23
as before, but three of every four samples in the region of interest are zero),
Fig 30.8
Due to the zero stuffing, we have in effect reduced the aperture of the signal from 100% to 25%. We saw
in Section 26 how this broadens the sinc function envelope (narrower box ⇒ broader sinc) which in turn
reduces the sinc distortion of the main spectrum. That same effect appears below.
Versions of equations (30.1,2,3,4) which apply to Figure 30.8 are (x pulse is now the thin box)
x(t) = ∑
n = -∞∞
yn xpulse (t -tn) t n = n (T1/4) y n = y(tn) (30.1)
X(ω) = (4/T1)Xpulse (ω) Y'(ω) , (30.2)
Y'(ω) ≡ (T1/4)∑
n = -∞∞
yn e-iωnT1/4 . (30.3)
Xpulse (ω) = Xbox(ω,T1/4) = e-iωT1/8 (T1/4) sinc(ω T1/8) . (30.4)
Combining the pieces gives,
X(ω) = e-iωT1/8 sinc(ω T1/8) [ (T1/4) ∑
n = -∞∞
yn e-iωnT1/4 ]
= e-iωT1/8 sinc(ω T1/8) [Y'(ω)]
Chapter 4: Some Practical Topics
120 The plot of |Y'( ω)| is the same as shown in Fig 30.2 but with 1/4 the amplitude ( the amplitudes y n shown
in Fig 30.8 are stored in array yz[n] as will be shown later),
F i g 3 0 . 9 and the X( ω) plot is this,
F i g 3 0 . 1 0 The good news is that the blue sinc distortion is smoother near the central main spectrum (compare to Fig 30.3). The bad news is that there are lots of high -amplitude image spectra the must be dealt with.
(d) Add an ω
1/2 digital low-pass interpolation filter
The new D/A design is this,
F i g 3 0 . 1 1 where B'( ω) is the transfer function of a low-pass filter wh ich is clocked at the faster clk4x rate. We add
low-cost registers at each stage in the pipeline to provide a stable input to the next stage.
Chapter 4: Some Practical Topics
121 In Section 29 we constructed an approxima te brick-wall filter with this spectrum B'( ω),
F i g 3 0 . 1 2
The output spectrum of the Fig 30.11 design which includes this filter is then
X
new(ω) = B(ω) X(ω) ,
where X( ω) was plotted in red in Fig 30.10. Here then is a plot of X
new(ω) :
F i g 3 0 . 1 3
The effect of the digital filter is to remove the image spectra from Fig 30.10 , a process sometimes called
alias-rejection. Since this low-pass filter is not a perf ect brick wall, there is some small distortion of the
central spectrum. On the other hand, the aperture reduction due to oversampling with zero-stuffing has
broadened the sinc hump perhaps alleviating the need for a sin(x)/x post-filter (Section 26). Residual high
frequency data in the signal can be removed by a lo w-cost analog filter located to the right of the D/A
converter in Fig 30.11. Since we never specified the original signal y(t) for wh ich Fig 30.1 is the sampled version, it is difficult to
compare the spectrum of that y(t) with the output of the Fig 30.11 design. Nevertheless, plotting the time-
domain output of the Figure 30.11 design is quite interesting.
It was noted that the thin bar amplitudes shown in Fig 30.8 (3/4 of which are 0) are stored in array
yz[n]. Here is how that array is computed from the original amplitudes y[n] of Fig 30.1:
Chapter 4: Some Practical Topics
122
In terms of these yz amplitudes, the convolution sum in (29.5) which defines the action of our intepolation filter appears as
(y
out)n = (1/4) ∑
m = -∞∞
(yz)n-m bm (29.5)
where (y out)n is the output of the filter which becomes, after two 4x clock delays, the output of the design
of Fig 30.11.
Notice that the aperture duty cycle of 25% reduces th e amplitude of the the filter output by 1/4, and
there will be a corresponding reduction in the time-domain filter output signal. As aperture goes to 0, the
entire signal eventually goes away. This fact, glossed over in Section 29, is one cost of using a small
aperture. For comparison purposes, we shall omit the 1/4 factor in the following code which computes
(yout)n for our 21 tap filter,
Here is a plot of the output sample values y out[n],
F i g 3 0 . 1 4
Chapter 4: Some Practical Topics
123 Since the D/A converter holds each sample for duration T 1/4, the actual analog output will have the
following form, which we obtain using our ancient Maple V's primitive histogram routine (which we have
been painfully using to make all the bar plots above),
F i g 3 0 . 1 5 This may be compared with our starting digital signal of Fig 30.1,
Fig 30.1
The output Fig 30.15 seems a little "ratty". If we increase the filter from 21 taps to 41 taps, things improve significantly, though there is still some ringi ng before and after the output pulse of interest
(recall the comment about Gibbs phenomena at the end of Section 29).
Chapter 4: Some Practical Topics
124
F i g 3 0 . 1 6
In the literature of oversampling, our oversampled dig ital low-pass filter is usually referred to as a digital
interpolation filter, for obvious reasons comparing Fig 30.1 and Fig 30.16. In this application, since 3 out
of every 4 incoming samples are zero from the zero-stuffing logic, it is possible to implement the filter more efficiently that we show in Fig 29.3 using polyphase techniques.
Chapter 5: Dispersion Relations
125 Chapter 5: Some Theoretical Topics
In this chapte
r we wander off on a more theoretical topic before returning in Chapter 6 to the practical
computation of the power spectra of specific pulse trai n line codes. The material in this chapter is not
used in that computation and the uninterested reader would do well to skip Section 31.
31. Spectral Dispersion Relations
(a) A simple integral equation for X( ω) analytic in the upper half plane
Let X(ω) be
some arbitrary function of the complex variable ω which has two properties: (1) X( ω) is
analytic in the upper half ω plane; (2) On any ray to ∞ in this upper half plane, X(ω ) → X(∞), a constant
which could be 0. Later we shall consider the case where "upper" ↔ "lower".
Consider then the following vanishing contour integral, where ω is real,
∫C dω' X(ω')
ω'-ω = 0 . ( 3 1 . 1 )
Here, C is a counterclockwise cont our which goes around the upper half ω' plane, but which detours
infinitesimally around and above the pole at ω' = ω. The integral vanishes as usual since we can shrink
the contour away to nothing.
Fig 31.1
We can regard this integral as being made of three pieces:
(1) infinite semicircle. The contribution here is, using ω' = Re
iθ and thinking R → ∞,
∫SC dω' X(ω')
ω'-ω = ∫0 π (Reiθidθ) X(Reiθ)
Reiθ -ω ≈ i ∫0 π dθ X(Reiθ) = X(∞ ) iπ
(2) tiny semicircular detour around the pole at ω = ω'. This gives minus one half the pole residue since the
path goes half way around this pole the wrong way, so the contribution is – i π X(ω) (see Appendix C text
below Fig C.1, the partial residue rule).
(3) the two pieces (- ∞,ω-ε) and (ω+ε,+∞) along the real ω' axis as ε→0. This is basically the integral
along the real axis but missing the single point ω = ω'. As noted in Appendix C, this is called a Cauchy
Chapter 5: Dispersion Relations
126 principle part (principle value) integral, and sometimes people (including us) denote it with a little tick
mark through the integral, ∫--, while others use the notation P ∫ or p.v.∫. The principle part integral is a
limit just as are the previous two pieces of the contour C.
Thus, we can rewrite (31.1) as follows:
X(∞) iπ – iπ X(ω ) + ∫-- ∞
-∞ dω' X(ω')
ω'-ω = 0
or X(ω) = X(∞) + (1/iπ)
∫-- ∞
-∞ dω' X(ω')
ω'-ω ( 3 1 . 2 )
where X( ω) is analytic in the upper half plane and in this half plane on any ray X(ω ) → X(∞)
This is an integral equation for X( ω) which involves a principle value integral of X(ω ).
(b) A simple integral equation for X( ω) analytic in the lower
half plane
We now repeat the previous section with upper → lower.
Let X(ω) be some arbitrary function of the complex variable ω which has two properties: (1) X( ω) is
analytic in the lower half ω plane; (2) On any ray to ∞ in this lower half plane, X( ω) → X(∞), a constant
which could be 0.
The new contour of interest is this
Fig 31.2
The differences now are that the great circle contour is now clockwise instead of counterclockwise, and
that the little detour contour now goes "t he right way" around the pole at ω' = ω . As a result, the
contributions from both these terms get negated, but th e principle part integral is unchanged. Equation
(31.1) is as before but C is now this new contour
∫C dω' X(ω')
ω'-ω = 0 . (31.1)'
Chapter 5: Dispersion Relations
127
so that
–X(∞ ) iπ + iπ X(ω ) + ∫-- ∞
-∞ dω' X(ω')
ω'-ω = 0
or X(ω) = X(∞) – (1/iπ)
∫-- ∞
-∞ dω' X(ω')
ω'-ω ( 3 1 . 3 )
X(ω) is analytic in the lower half plan e and in this half plane on any ray X( ω) → X(∞ )
Comparing to (31.2) one sees that the only difference is the sign of the integral. One way to understand this result is to think of reflecting the ω plane through the real axis, so th at the imaginary axis is reflected
and this is accounted for by taking i → - i in (31.2) to get (31.3). If we now define σ by
σ ≡
⎩⎨⎧ +1 if X(ω) is analytic in the upper half plane with ray limit X( ∞)
–1 if X(ω) is analytic in the lower half plane with ray limit X( ∞) (31.4)
We can combine our two results as follows :
X(ω) = X(∞) + σ (1/iπ ) ∫-- ∞
-∞ dω' X(ω')
ω'-ω . ( 3 1 . 5 )
With our adopted phase sign convention e-iωt in the Fourier Integral Transform (1.1), we normally obtain
spectral functions X(ω ) which are analytic in the lower half plane and so σ = -1. For other sources (such
as Stakgold) which use the reverse sign e+iωt of the Fourier transform phase, one would use σ = + 1.
Example
We considered an RC filter in Section 4 (b) and found there the following transfer function,
G ( ω) = 1
1 + iωRC = (-i/RC)
ω - i/RC = (-i/τ)
ω - i/τ . τ ≡ RC (31.6)
This function has a pole in the upper half plane at ω = i/RC and is analytic in the lower half plane with a
ray limit there G( ω) → 0. For this function, (31.3) or (31.5) with σ = -1 claims that
G(ω) = – (1/iπ ) ∫-- ∞
-∞ dω' G(ω')
ω'-ω ( 3 1 . 7 )
or 1
1 + iωRC
= – (1/iπ ) ∫-- ∞
-∞ dω' 1
ω'-ω (-i/τ)
ω' - i/τ = – (1/i π)(-i/τ ) ∫-- ∞
-∞ dω' 1
(ω'-ω)(ω'-i/τ)
= (1/ πτ)
∫-- ∞
-∞ dω" 1
ω"(ω" +ω - i/τ) = (1/πτ ) ∫-- ∞
-∞ dx 1
x(x +a) a ≡ ω-i/τ, τ = RC .
Chapter 5: Dispersion Relations
128 As a principle value integral exercise, we now evaluate the integral shown to verify that it really comes
out being G(ω ) :
∫-- ∞
-∞ dx 1
x(x +a) = lim ε→0 [ ∫-∞ -ε + ∫ε ∞ ] dx
x(x +a)
= (1/a) lim ε→0 { ln(x
x+a )|-ε
-∞ + ln(x
x+a )|∞
ε ]
= (1/a) lim
ε→0 { ln(-ε
-ε+a ) – ln(1) + ln(1) – ln(ε
ε+a ) } = (1/a) lim ε→0 { ln(-ε
-ε+a ) – ln(ε
ε+a ) }
= (1/a) lim
ε→0 { ln(-ε
-ε+a ε+a
ε )} = (1/a) lim ε→0 { ln(ε+a
ε-a )} = (1/a) lim ε→0 { ln(- 1-iε ') }
= (1/a) lim ε→0 { ln(e-iπ) } = (1/a) (-i π) .
The term -i ε' represents the fact that the log argument has a small negative imaginary part, which takes a
bit of work to show:
ε+a
ε-a = – a+ε
a-ε a*-ε
a*-ε = -|a|2 + 2iεIm(a) - ε2
|a|2 - 2εRe(a) + ε2 = -|a|2 + 2iε(-1/τ) - ε2
|a|2 - 2εω + ε2 .
We have then shown that the right side of (31.7) evaluates to the left side,
– (1/i π) ∫-- ∞
-∞ dω' G(ω')
ω'-ω = (1/πτ) ∫-- ∞
-∞ dx 1
x(x +a) = (1/πτ) (1/a) (-iπ) = - i/τ
a = -i/τ
ω-i/τ = G(ω) .
(c) Dispersion Relations for X( ω)
Notice in (31.5) the ver
y important factor of (1/i). If we now break X( ω) into its real and imaginary parts
and then write down the real and im aginary parts of equation (31.5), we find that Re(X) and Im(X) are
related to each other by the following two equations:
Re[X( ω)] = Re[X( ∞)] + σ(1/π) ∫-- ∞
-∞ dω' Im[X(ω ')]
ω'-ω (31.8a)
Im[X(ω )] = Im[X( ∞)] – σ(1/π) ∫-- ∞
-∞ dω' Re[X(ω')]
ω'-ω . ( 3 1 . 8 b )
Basically, this says that the real part of X( ω) along the real axis completely determines the imaginary part,
and vice versa. One cannot arbitrarily set the real a nd imaginary parts independently. This is a general
fact about analytic functions X( ω).
Next, let us assume in addition that X( ω) is the spectrum of a real function x(t). As we saw in Section 7,
this implies the reflection rule X( −ω) = X(ω)*, which says X( ω) is Hermitian. Thus, we can fold the
Chapter 5: Dispersion Relations
129 negative portions of the above inte grations over to the positive side. First (we omit principle value tick
marks just for a while),
∫-∞ 0 dω' Im[X(ω ')]
ω'-ω = ∫+∞ 0 [-dω"] Im[X(-ω ")]
-ω"-ω = ∫+∞ 0 [-dω"] Im[X(ω ")*]
-ω"-ω
= ∫+∞ 0 [-dω"] -Im[X(ω")]
-ω"-ω = ∫0 ∞ [dω"] -Im[X(ω")]
-ω"-ω = ∫0 ∞ dω" Im[X(ω ")]
ω"+ω
and therefore
∫-∞ ∞ dω' Im[X(ω ')]
ω'-ω = ∫0 ∞ dω' Im[X(ω ')]
ω'+ω + ∫0 ∞ dω' Im[X(ω ')]
ω'-ω
= ∫0 ∞ dω' Im(X(ω')] [1
ω'+ω + 1
ω'-ω ] = 2 ∫0 ∞ dω' ω'Im[X(ω')]
ω'2-ω2 .
The other integral can be folded in a similar manner,
∫-∞ 0 dω' Re[X(ω')]
ω'-ω = .... = – ∫0 ∞ dω' Re[X(ω')]
ω'+ω
∫-∞ ∞ dω' Re[X(ω')]
ω'-ω = ∫0 ∞ dω' Re(X(ω')] [- 1
ω'+ω + 1
ω'-ω ] = 2ω ∫0 ∞ dω' Re[X(ω')]
ω'2-ω2
We then rewrite (31.5) as , valid for X( −ω) = X(ω)* (tick marks restored),
Re[X( ω)] = Re[X( ∞)] + σ (2/π) ∫--∞
0 dω'ω' Im[X(ω ')]
ω'2-ω2 ( 3 1 . 9 a )
Im[X(ω )] = Im[X( ∞)] – σ(2/π) ∫--∞0 dω' Re[X(ω')]
ω'2-ω2 . (31.9b)
These two equations are completely general, given the assumptions we have made. They are associated with the names Kramers and Kronig who wrote similar equations in 1926 for certain functions
connected with the index of refraction and the dispersion of light (hence "dispersion relations").
(d) Dispersion Relations for γ(ω)
In filter theory, one thinks
of X(ω) as the "transfer function" of a filter. It is usually easier to think in
terms of the function γ(ω) which we define as
γ(ω) ≡ - ln [X(ω )] = α(ω) + i β (ω) . ( 3 1 . 1 0 )
Then we get
Chapter 5: Dispersion Relations
130 X(ω) = e-γ(ω) = e-α(ω) e-iβ(ω) . ( 3 1 . 1 1 )
Notice that we defined γ(ω) with a minus sign, so both exponents have minus signs. The real quantities
α(ω) and β(ω) are the attenuation and phase functions of the filter.
Can we apply the dispersion relations to the function γ(ω) instead of X( ω) ? Yes, provided γ(ω) meets the
same requirements assumed for X( ω). If X(ω) has a pole in the upper half ω plane, as in (31.6), then γ(ω)
has a branch cut singularity in the upper half plane starting at the pole and going off to the left. No problem since γ(ω) is still analytic in the lower half plane. However, consider:
γ(ω) ≡ - ln [X(ω)] = + ln[ 1/X( ω)] .
This says that a zero in X( ω) is just as bad as a pole from γ (ω)'s point of view. A zero of X( ω) in the
lower half plane means γ(ω) has a branch cut in the lower half plane starting at this zero location and
going off to the left, and th is invalidates our conditions ( σ = -1).
Thus, we must now assume that X(ω ) has neither zeros nor poles in the lower half plane, which maps into
the Z Transform H"(z) having neither zeros nor poles outside the unit circle in Fig 24.1. A filter satisfying this condition is called a minimum phase filter .
Since we have assumed X(ω ) goes to X( ∞) on the great circle at infinity, we know that γ (ω) goes to γ(∞)
= -ln[ X( ∞)], so no extra assumption is needed here. If X( ∞) = 0, then γ(∞) =
-∞, which is a little
inconvenient. It just says that the attenuation of our filter is infinite as ω → ∞.
So now we can write the dispersion relations analogous to (31.8) for γ(ω) instead of X( ω), assuming now
that X(ω) has neither poles nor zeros in the lower half ω plane. We must choose σ = -1 in (31.4), and note
that Re[γ(ω)] = α(ω) and Im[γ(ω)] = β(ω) :
α(ω) = α (∞) + σ (1/π) ∫-- ∞
-∞ dω' β(ω')
ω'-ω ( 3 1 . 1 2 a )
β(ω) = β (∞) – σ (1/π) ∫-- ∞
-∞ dω' α(ω')
ω'-ω ( 3 1 . 1 2 b )
Now if x(t) is real so that X( ω) = X(-ω)* , we find that γ(ω) is also Hermitian since X( ω) = e-γ(ω) :
X(-ω) = X(ω)* ⇒ γ(-ω) = γ(ω) * ( 3 1 . 1 3 )
⇒ α(-ω) = α (ω) and β(-ω) = - β (ω)
Since γ(ω) is Hermitian, we can process (31.12) just as we did processed (31.8) to get
α(ω) = α (∞) + σ (2/π)
∫--∞
0 dω'ω' β(ω')
ω'2-ω2 ( 3 1 . 1 4 a )
β(ω) = β (∞) – σ (2/π) ω ∫--∞0 dω' α(ω')
ω'2-ω2 (31.14b)
Chapter 5: Dispersion Relations
131
The main point of the above is that the phase of a mi nimum phase filter is completely determined by its
attenuation, and vice versa. Even for general filters there will be some relation like the above, but it will
include terms to describe the zeros of X( ω) in the lower half plane. The conclusion that the phase and
attenuation cannot be independe ntly set is unavoidable.
(e) Dispersion and Attenuation
We showed in (21.19) that
the gr oup delay of a filter is given by
τd = dφ/dω where B( ω) = |B(ω)| e-iφ(ω) .
Translating that into our current context, we get
τ(ω) = dβ(ω)/dω where X( ω) = e-γ(ω) = e-α(ω) e-iβ(ω) . (31.15)
If we define the integral appearing in (31.14b) as K( ω), including the (2/π ),
K(ω) ≡ (2/π) ∫--∞
0 dω' α(ω')
ω'2-ω2 ( 3 1 . 1 6 )
then we find that τ(ω) = dβ(ω)/dω = d [ ω K(ω) ] /dω .
If the integral K( ω) were somehow a constant κ, we would conclude that τ(ω) = κ, and the filter would be
"non-dispersive". As discussed in Section 21 (b), this means the filter is "linear phase" and the group delay is a constant independent of ω. All frequency components of a pulse packet would then traverse the
filter in the same time, so the pulse does not spread out (disperse) in time. Obviously K( ω) cannot really be independent of ω, so a non-dispersive minimum phase filter does not
exist. However, over certain ranges of ω where K( ω) is very slowly varying, such a filter can be
reasonably non-dispersive. This would be a region of ω far away from any region where the attenuation
α(ω) strongly varies.
Crude Proof:
Suppose α(ω) is very smoothly varying near some ω. In the integral (31.16) for K( ω)
near ω we expect the main contribution to come from ω ' close to ω, (ω-a,ω+a) for small a, since the
denominator is 0 and therefore amplifies the numerator there. The denominator is roughly 2 ω(ω'- ω)
which is a pole. If we assume that α(ω) has some linear form α(ω') ≈ α(ω) + (ω-ω')α'(ω) near ω , then
this integration region which would normally be high ly amplifying in fact yields the following,
K(ω) ≡ (2/π) ∫0 ∞ dω' α(ω')
ω'2-ω2 ≈ (2/π) ∫ω-a ω+a dω' α(ω) + (ω-ω')α'(ω)
ω'2-ω2
≈ (2/π) α(ω) 1
2ω { ∫ω-a ω+a dω'
ω-ω' } + (2/π) α'(ω) 1
2ω { ∫ω-a ω+a (ω-ω')dω'
ω-ω' }
Chapter 5: Dispersion Relations
132
= (2/ π) α(ω) 1
2ω { 0 } + (2/ π) α'(ω) 1
2ω 2a ≈ (2a/π) α'(ω)
ω ≈ 0 since α'(ω) is small
The rest of the integration region which is far from ω yields a contribution to K( ω) which is weakly
dependent on ω and which we can regard as roughly constant K( ω) ≈ κ for some band Δω . We then
get our desired approximate linear phase and cons tant group delay, and therefore very small
dispersion, τ(ω) = dβ/dω = d [ ω K(ω) ] /dω ≈ d [ ω κ ] /dω = κ .
However, if α (ω) varies significantly near ω, our linear term with α'(ω) might be large and there will
likely be additional higher terms in the Taylor expansion of α(ω) so K(ω) might then vary strongly
with ω due to the integration contribution from region ( ω-a,ω+a). In this case, we get non-linear phase
and dispersion.
To repeat the claim above: For ω far from regions where attenuation α(ω) significantly varies, we expect
K(ω) to be roughly constant and so we have nearly linear phase, nearly constant group velocity, and small
dispersion. Attenuation and dispersion are intertwined. You can't have one without the other. This is a general fact
one learns from the dispersion relations, without any specific filter in mind.
(f) Application to coaxial cable
Consider an infinitel
y long coaxial cable driven at its left end at z = 0. A coaxial cable acts as a filter
G(ω). In the frequency domain, if we drive the cable with I( ω), the output O( ω,z) at z is
O(ω, z) = G(ω, z) I(ω) G( ω) = e-γ(ω)z γ(ω) = α (ω) + iβ(ω) . (31.17)
For a coaxial cable one has
γ = (R+iωL)(G+iωC ) ( 3 1 . 1 8 )
where R,L,G and C are resistance, inductance, conducta nce (across the dielectric) and capacitance all per
unit length of the cable. If we ignore ohmic losses in both the conductor and the dielectric, then R = G = 0 and we get
γ = iω
LC .
The inductance L is generally independent of ω , but C = ε(ω)2πε0/ln(b/a) = k 1 ε(ω), where ε(ω) is the
dielectric "constant", which in general is not constant as a function of ω. Thus we have
γ(ω) = iω Lk1 ε(ω) .
Chapter 5: Dispersion Relations
133 From Maxwell's equations one knows that the inde x of refraction of a medium is given by
n(ω) = με = ε(ω) /k2 .
So we then have,
γ(ω) = iωLk1 k2 n(ω) = iωk3 n(ω)
= i ωk3[ Re(n) + i Im(n) ] = - ωk3Im(n) + iωk3Re(n) = α (ω) + iβ(ω)
so
α(ω) = -ωk
3Im[n(ω)] β(ω) = ωk3Re[n(ω)] .
If it were true that Re[n( ω)] were inde pendent of ω, then β(ω) would have linear phase and we would then
have constant group delay and no dispersion in the coaxial cable.
It turns out that the index n( ω) has the right properties for the dispersion relations (31.8) to be valid, so
Re[n(ω )] = Re[n( ∞)] + σ(1/π) ∫-- ∞
-∞ dω' Im[n(ω')]
ω'-ω (31.19a)
Im[n( ω)] = Im[n( ∞)] – σ(1/π) ∫-- ∞
-∞ dω' Re[n(ω ')]
ω'-ω (31.19b)
where usually Re[n( ∞)] = 1 and Im[n( ∞)] = 0. If it happened that Im[n( ω)] were very small, then
(31.19a) says that Re[n( ω)] = Re[n( ∞)] = constant, just what we want to get no cable dispersion.
In a non-polar dielectric, like polyethylene or teflon, n( ω) does in fact have a very small imaginary part
for frequencies below the electromagnetic resonances of the medium. Thus, if we could ignore ohmic
losses in the conductors, coaxial cables using these materi als as dielectrics would be non-dispersive up to
infrared frequencies -- where vibrationa l and rotational resonances set in.
Dispersion relations are often written for othe r functions such as the dielectric constant ε(ω). In this case
one can regard the relationship between electric displacement D and electric field E D(ω) = e(ω) E(ω) ( 3 1 . 2 0 )
as a "filter", where everything is evaluated at the same point in space.
(g) The Dispersion Relation expressed in terms of the Hilbert Transform
Recall the integral equation which is the
starting point for the dispersion relation discussion above,
X(ω) = X(∞) + i σ (1/π) ∫-- ∞
-∞ dω' X(ω')
ω-ω' (31.5)
Chapter 5: Dispersion Relations
134 σ ≡ ⎩⎨⎧ +1 if X(ω) is analytic in the upper half plane with ray limit X( ∞)
–1 if X(ω) is analytic in the lower half plane with ray limit X( ∞) (31.4)
where we have introduced some offsetting minus signs . The integral appearing here is in fact another
transform in our growing cornucopia of transforms, this one being the Hilbert Transform ,
Xh(ω) ≡ (1/π) ∫-- ∞
-∞ dω' X(ω')
ω-ω' . // projection = transform (31.21)
Thus, it happens that the dispersi on relation (31.5) can be written
X(ω) = X(∞) + i σ Xh(ω) . ( 3 1 . 2 2 )
If X(∞) = 0 and σ = +1, any solution to the dispersion relation (31.5) must be a function X( ω) which is
equal to its own Hilbert transform times i σ. In Appendix C we find that (with β = 1)
X(ω) = eiω ⇔ Xh(ω) = -i eiω . (C.43)
This X(ω) is then a particular solution to the dispersion relation since i X h(ω) = eiω = X(ω) and in the
upper half plane X( ∞) ~ ei(+i∞) = e-∞ = 0. In (31.7) we found another solution for the case σ = -1 where
X(∞) = 0 in the lower half plane.
The general study of the integral equation (31.5)
X(ω) = X(∞) + σ (1/iπ ) ∫-- ∞
-∞ dω' 1
ω'-ω X(ω' ) (31.5)
and its possible solutions is complicated because the kernel 1/( ω'-ω) (called in this case a Cauchy kernel)
is singular (it blows up when ω = ω'). See for example Polyanin and Manzhirov Chapter 15. It is more
useful to think of the dispersion relation as an integral condition which a spectrum X( ω) must satisfy
rather than an integral equation which completely determines that spectrum.
Chapter 6: Power in Pulse Trains
135 Chapter 6: Power in Pulse Trains
32. The Autocorrelation Function
Here w
e deal with some preliminary matters befo re studying the power spectra of pulse trains.
Start with a reasonable function x(t). Define the autocorrelation function of x(t) as follows:
r
x(t) ≡ ∫-∞ ∞ dt' x(t') x(t' + t) . (32.1)
The integrand is the function evaluated at time t' times the same function evaluated at later time t'+t.
Some sources define r x(t) with an extra overall "normalizing" constant factor. For example, if one
were to define a x(t) ≡ rx(t)/ T where T is the duration of a pulse train, then a x(t) and r x(t) have different
dimensions. Below we show that our r x(t) has dimensions of energy, so a x(t) would have dimensions of
power. We prefer defining autocorrelation as show n in (32.1) with no normalizing constant.
A simple reflection property follows from th e above definition (use t" = t' + t ):
r
x(-t) = rx( t ) . ( 3 2 . 2 )
Since r
x(t) is an even function of t, in (32.1) we coul d put either +t or -t in the last parentheses.
Although we have not yet mentioned statistics and randomness, one could easily imagine the following
situation. Suppose x(t') is some sort of random function ("noise") that takes values in the range -1 to 1. It
seems likely that for a value of the separation t that is larger than some small value, one might get r x(t) =
0. The vague argument would be that there is no "co rrelation" between x(t') and x(t'+t), so the product of
these two functions ought to be pretty random, and a su m of random numbers in the range -1 to 1 ought to
be zero. Even in this case, we can see that the result is not zero if t = 0, since we are then summing a positive quantity. In fact, r
x(0) is the area under x(t)2.
(a) Autocorrelation function for a Square Pulse
Before going any
further, let us compute the auto correlation function for some simple case we are
familiar with. A good candidate is x(t) = a square pulse of width τ and amplitude A. As in (9.1),
xpulse (t) = A [ θ(t + τ/2) - θ(t - τ/ 2 ) ] . (9.1)
One can easily do the above integral (32.1) to get th e answer, but it is very obvious what the answer is.
We are multiplying a box times a box shifted by t. Where they overlap, the integrand is A2. The boxes
only overlap if the absolute value of shift t is less than the width τ of the pulse. If this is so, the size of the
overlap is τ - |t|. If |t| is larger than τ, there is no overlap, so the integral is 0. Thus,
rpulse (t) = A2(τ - |t|) θ(τ - |t|) = (A2τ) [ 1 - |t|/τ ] θ(τ - |t|) (32.3)
Chapter 6: Power in Pulse Trains
136 and we find that the autocorrelation function is a triangle whose base is twice the pulse width,
Fig 32.1
(b) Energy, power and spectral energy density for a finite signal x(t)
The total energy
in a finite duration signal x(t) can be computed in either the t-domain or the ω-domain
using Parseval's formula (10.5), to which we add 1/R to each side,
E =
∫-∞ ∞ dt |x(t)|2/R = ∫-∞ ∞ dω |X(ω)|2
2πR . ( 3 2 . 4 )
If we think of x(t) as the voltage across a resistor R, then dt x2(t)/R is the energy delivered to the resistor
in time dt, and the integral on the left is the total en ergy in signal x(t). The dimensions on the right are,
looking at (1.1),
dω |X(ω)|2
2πR = sec-1 (volt-sec)2/ohms = (volt2/ohms)sec = watts-sec = joules = energy . (32.5)
Setting R = 1 Ω, we can write this as
E = ∫-∞ ∞ dt p(t) = ∫-∞ ∞ dω E(ω) ( 3 2 . 6 )
p(t) ≡ |x(t)|2 = energy density in the t-domain (joule/sec = watt)
p(t)dt = energy in dt (joules)
E(ω) ≡ |X(ω)|2/2π = energy density in the ω-domain (joule-sec)
E(ω)dω = energy in d ω ( j o u l e s )
We shall refer to E(ω) as the spectral energy density of signal x(t) whose Fourier Transform is X( ω).
The isolated single pulse x
pulse (t) has a corresponding Epulse (ω). The function p(t) is the temporal
energy density.
(c) The Wiener-Khintchine theorem
Changing t
o t" = -t', we can trivially rewrite the definition (32.1) as follows:
rx(t) = ∫-∞ ∞ dt"x(t - t") x(-t") . // energy units (32.7)
From now on, we assume x(t) is a real valued f unction. Recall now the convolution theorem (3.6),
Chapter 6: Power in Pulse Trains
137
a(t) = ∫-∞ ∞ dt" b(t-t") c(t") ⇔ A(ω) = B(ω) C(ω ) . (3.6)
We see that (32.7) has the standard convolution equation form where we select
b(t) = x(t) ↔ B( ω) = X(ω)
c(t) = x(-t) ↔ C( ω) = X( −ω) = X(ω)* // from (7.1) and (7.2)
Thus, the diagonalized frequency domain form A( ω) = B(ω) C(ω) is
R
x(ω) = |X(ω)|2 . ( 3 2 . 8 )
Dividing by 2 π we find that
E(ω) = (1/2π) Rx(ω) . ( 3 2 . 9 )
This says that the spectral energy density E(ω) of signal x(t) is 1/2 π times the Fourier Integral Transform
Rx(ω) of the autocorrelation function r x(t) of the signal x(t). This result is sometimes called the Wiener-
Khintchine [Khintchin] theorem.
Note also from (32.1) that r x(t) evaluated at t = 0 gives the total energy in signal x(t),
rx(0) = ∫-∞ ∞ dt x2(t) ≡ E = total energy in signal x(t) (32.10)
For us, the significance of (32.9) is that we can "inject statistics" into a computation of the autocorrelation
function, and then we will know the power spectrum of our statistical signal from (32.9). All we have to do is Fourier transform the autocorrelation function r
x(t). Examples will follow.
(d) Verification of Wiener-Khintchine for a Square Pulse
Equation (
32.3) gives the autocorrelation function r x(t) for a square pulse. One can insert this into the
Fourier transform (1.1) to compute R x(ω),
Rx(ω) = ∫-τ τ dt (A2τ) [ 1 - | t | / τ ] e-iωt = 2(A2τ) ∫0 τ dt (1-t/τ) cos(ωt)
= 2(A2τ) (1-cos(ωτ ))/(ω2τ) = 4(A2τ) sin2(ωτ/2)/(ω2τ) = (A2τ2) sin2(ωτ/2)/(ωτ/2)2
= (A τ)2 [ sinc(ωτ/2) ]2 , ( 3 2 . 1 1 )
and this is recognized from (9.2) to be |X( ω)|
2 for the square pulse, in agreement with (32.8) .
Chapter 6: Power in Pulse Trains
138 (e) Cross-correlation, convolution, and autocorrelation
Notation : a* means complex conjugation, b ∗ c means convolution, b ⋆ c means cross-correlation .
The cross-correlation of two functions b and c is defined this way
(b⋆c)(t) ≡ ∫-∞ ∞ dt' b*(t')c(t+t') = (c ⋆b)*(-t) (32.12)
where the right equality is easy to show setting t+t' = t". So in general, b ⋆c ≠ c⋆b .
In Section 7 we noted that a function is Hermitian if f*(-t) = f(t).
Fact : If b and c are both Hermitian, then b
⋆c = c⋆b. (32.13)
Proof: b⋆c = ∫-∞ ∞ dt' b*(t')c(t+t') = ∫-∞ ∞ dt' b(-t')c*(-t-t') = ∫-∞ ∞ dt' b(t"+t)c*(t") = c ⋆b
If b and c are both real and both even, they are both Hermitian so again, b ⋆c = c⋆b .
The convolution of two functions b and c we saw from the (3.1) and (3.2) was this
(b∗c)(t) = ∫-∞ ∞ dt' b(t')c(t-t') = (c ∗b)(t) .
In order to relate these two operations, we need to show more detail in the notation. Thus, using t" = -t',
[b(t)⋆c(t)](t) = ∫-∞ ∞ dt' b*(t')c(t+t') = ∫-∞ ∞ dt" b*(-t")c(t-t") = [b*(-t) ∗ c(t)](t) .
If b(t) is a Hermitian function so b*(-t) = b(t), then we have shown that :
Fact : If b is Hermitian, then b
⋆c = b∗c. (32.14)
If b = c = real, then we have from (32.1) and (32.2),
rb(t) = ∫-∞ ∞ dt' b(t')b(t'+t) = ∫-∞ ∞ dt' b(t')b(t'-t)
so we have just proven this fact :
Fact: If b is real, then r
b = b⋆b = b∗b. (32.15)
Chapter 6: Power in Pulse Trains
139
Thus, for a real function b, the autocorrelation functi on is the cross-correlation function of b with itself.
This then gives some motivation for the name associated with r b. The autocorrelation function is also the
convolution of b with itself.
(f) Z Transform Wiener-Khintchine theorem for a Pulse Train
The Wiener-Khintchine theorem
was stated above in section (c) as
Rx(ω) = |X(ω)|2 (32.8)
where R
x(ω) is the Fourier Integral transfor m of the autocorrelation function r x(t)
rx(t) ≡ ∫-∞ ∞ dt' x(t') x(t' + t) . (32.1)
From the amplitudes y n of an infinite pulse train one can define an autocorrelation sequence in analogy
with the autocorrelation function,
rs ≡ limN→∞ [1
(2N+1) ∑
n = -NN
yn yn+s ] ≡ <yn yn+s> 1 . (32.16)
Although r x(t) (our particular definition) has no normalization factor, we have added 1
(2N+1) to the
definition of r s in order to obtain the finite result r s = <yn yn+s>1 . The subscript 1 is used as a reminder
that this is an expectation value for a single sequence { y n }. Later we shall use <y n yn+s> without the 1
subscript to indicate an "ensemble average" whic h involves an ensemble of many pulse trains.
It is convenient to use this shorthand notation for r s,
rs = 1
(2N+1) ∑
n = -∞∞
yn yn+s = T1
T ∑
n = -∞∞
yn yn+s (32.17)
where T = (2N+1)T 1 is the duration of the pulse train. We know that this infinite T is going to cancel
another T in any "application" so we allow it to exist temporarily, as in (33.22) where T = [2 πδ(0)]T1 .
The Z transform of r s is given by
R"(z) ≡ ∑
s = -∞∞
rs z-s = ∑
s = -∞∞
{ T1
T ∑
n = -∞∞
yn yn+s } z-s = T1
T ∑
n = -∞∞
∑
s = -∞∞
[yn zn ] [yn+s z-(n+s)]
= T
1
T ∑
n = -∞∞
∑
m = -∞∞
[yn zn ] [ym z-m] m ≡ n+2
Chapter 6: Power in Pulse Trains
140
= T1
T [ ∑
n = -∞∞
yn zn] [∑
m = -∞∞
ym z-m ] = T1
T Y"(z)* Y"(z)
so we have obtained this Z Transform version of th e Wiener-Khintchine theorem, which we compare to
the regular version,
R"(z) = T
1
T | Y"(z) |2 Z Transform Wiener-Khintchine (32.18)
Rx(ω) = |X(ω)|2 . regular Wiener-Khintchine (32.8)
Here X(ω) is the Fourier Integral Transform of the pulse train x(t),
x(t) = ∑
n = -∞∞
yn xpulse (t -tn) . (25.1)
while Y"(z) is the Z transform of the sequence of pulse train amplitudes.
33. Spectral power density of a Simple Pulse Train
In subsections (a) through (d) we deal
only with si mple pulse trains. Along the way, certain facts are
developed which apply to general as well as simple pulse trains. In subsection (e) we gather together these
general facts, and then show how quantities of intere st can be related to the autocorrelation function.
Comment
: Some texts use the phrase "power spectral dens ity" (PSD). Although this wins on Google by a
ratio of 3 to 1, we still prefer the phrase "spectra l power density", and the same for "spectral energy
density".
(a) Infinite Simple Pulse Train
This is the fir
st section in which we use the δ(0) notation of Appendix A which may make the reader feel
a bit uncomfortable. In subsection (b) we shall repeat everything for a finite pulse train and then take the
limit N→∞ to obtain the same results without using δ(0). In both sections we shall include the Z
Transform in passing, but our main work is in the ω variable, not the z variable.
From Section 14 (a) we know the spectrum of an infinite pulse train formed from pulses x
pulse (t)
separated by time T 1 ,
x(t) = ∑
n = -∞∞
xpulse (t - nT1) (14.1)
X(ω) = Xpulse (ω) ∑
m = -∞∞
2π δ(ωT1 - 2πm ) . (14.4)
Chapter 6: Power in Pulse Trains
141
We can obtain the same expressions from box (25.4) which summarizes amplitude modulated pulse trains
by setting all amplitudes to y n = 1,
x(t) = ∑
n = -∞∞
yn xpulse (t -tn) = ∑
n = -∞∞
xpulse (t -tn) (33.1)
X(ω) = (1/T 1)Xpulse (ω) Y'(ω) = Xpulse (ω) Y"(z) (33.2)
Y"(z) = Y' (ω)/T1 = ∑
n = -∞∞
yn e-iωnT1 = ∑
n = -∞∞
e-iωnT1 . (33.3)
Y'(ω) is the Digital Fourier Transform of y n = 1, and Y"(z) is the Z Transform, where z = eiωT1 .
In the last line we then use (13.2)
∑
n = -∞∞
e±ink = ∑
m = -∞∞
2πδ(k - 2πm) -∞ < k < ∞ (13.2)
so that (33.3) becomes
Y"(z) = Y' (ω)/T
1 = ∑
n = -∞∞
e-iωnT1 = ∑
m = -∞∞
2πδ(ωT1 - 2πm) (33.4)
and then (33.2) says
X(ω) = (1/T 1)Xpulse (ω) Y'(ω) = X pulse (ω) ∑
m = -∞∞
2πδ(ωT1 - 2πm) (33.5)
in agreement with (14.4) quoted just above (33.1).
To find the power spectrum of a signal x(t), our first task is to compute | X( ω) |
2. From (33.2) we get
|X(ω)|2 = |Xpulse (ω)|2 (1/T1)2 |Y'(ω)|2 = |Xpulse (ω)|2 | Y"(z) |2 . z = eiωT1 (33.6)
We therefore must deal with the following object, using (33.4),
| Y"(z) |
2 = (1/T 1)2 |Y'(ω)|2 = [ ∑
m = -∞∞
2πδ(ωT1 - 2πm) ] 2 , (33.7)
and we are now faced with the issue of squaring de lta functions. Formally these objects don't exist in the
realm of distribution theory, but (as discussed in Appendix A) we can deal with them in an ad hoc way
which proves to be useful. Consider,
Chapter 6: Power in Pulse Trains
142
[ ∑
m = -∞∞
2πδ(ωT1 - 2πm) ] 2 = ∑
m = -∞∞
2πδ(ωT1 - 2πm) ∑
n = -∞∞
2πδ(ωT1 - 2πn)
=
∑
m = -∞∞
∑
n = -∞∞
2πδ(ωT1 - 2πm) 2πδ(ωT1 - 2πn) .
Looking at the product of the two delta functions, there can be no contribution to the double sum unless m = n, so we continue
=
∑
m = -∞∞
2πδ(ωT1 - 2πm) 2πδ(0) = [2 πδ(0)] ∑
m = -∞∞
2πδ(ωT1 - 2πm)
so that [
∑
m = -∞∞
2πδ(ωT1 - 2πm) ] 2 = [2πδ(0)] ∑
m = -∞∞
2πδ(ωT1 - 2πm) . (33.8)
The object δ(0) is formally undefined, but in Appendix A we ascribe the meaning that 2 πδ(0) = 2N+1 in
the limit that N → ∞ and we can always "undo the limit" when necessary. We shall firm up this idea in
section (b) directly below. So we have shown then that
| Y"(z) |
2 = (1/T 1)2 |Y'(ω)|2 = [2πδ(0)] ∑
m = -∞∞
2πδ(ωT1 - 2πm) (33.9)
or |Y"(z)|
2
[2πδ(0)] = (1/T 1)2 |Y'(ω)|2
[2πδ(0)] = ∑
m = -∞∞
2πδ(ωT1 - 2πm) . (33.10)
Then from (33.6)
|X(ω)|
2
[2πδ(0)] = |Xpulse (ω)|2 |Y"(z)|2
[2πδ(0)] = |Xpulse (ω)|2 (1/T1)2|Y'(ω)|2
[2πδ(0)]
= |X pulse (ω)|2 ∑
m = -∞∞
2πδ(ωT1 - 2πm) . (33.11)
We shall now repeat the above set of steps for a finite pulse train.
(b) Finite Simple Pulse Train
Our finite pulse train alw
ays has pulses ra nging from n = -N to N instead of from n = - ∞ to ∞ . We start
off exactly as in the previous section but with limited sums,
Chapter 6: Power in Pulse Trains
143 x(t) = ∑
n = -NN
yn xpulse (t -tn) = ∑
n = -NN
xpulse (t -tn) (33.12)
X(ω) = (1/T 1)Xpulse (ω) Y'(ω) = Xpulse (ω) Y"(z) (33.13)
Y"(z) = Y' (ω)/T1 = ∑
n = -NN
yn e-iωnT1 = ∑
n = -NN
e-iωnT1 . (33.14)
In the last line we then use (13.3),
∑
n = -NN
eink = 2π { sin[(N+1/2)k]
2π sin(k/2) } ≡ 2π δ5(k,N) -∞ < k < ∞ (13.3)
where δ5 is a periodic delta function model discussed in Appendix A (b). Equation (33.14) becomes
Y"(z) = Y' (ω)/T1 = ∑
n = -NN
e-iωnT1 = 2π δ5(ωT1,N) (33.15)
and then equation (33.13) says
X(ω) = Xpulse (ω) 2π δ5(ωT1, N ) . ( 3 3 . 1 6 )
This δ
5 is periodic with period 2 π and has identical peaks separated by 2π . For finite N, these are peaks of
finite width and height. Squaring, we find
| X(ω) |2 = | Xpulse (ω) |2 [2π δ5(ωT1,N)]2 . (33.17)
Recalling the definition of the δ6 delta function model from Appendix A (A.20),
2π δ6(k,N) ≡ [2πδ5(k,N)]2
(2N+1) (A.20)
we obtain
|X(ω)|2
(2N+1) = | Xpulse (ω) |2 2π δ6(ωT1,N) (33.18)
and this is the finite pulse train result. We can then take the limit N →∞ and make use of (A.21),
limN→∞ δ6(ωT1,N) = ∑
m = -∞∞
δ(ωT1 - 2πm ) (A.21)
Chapter 6: Power in Pulse Trains
144 to find that
limN→∞ [|X(ω)|2
(2N+1)] = | X pulse (ω) |2 ∑
m = -∞∞
2π δ(ωT1 - 2πm) (33.19)
and this replicates (33.11) with the promised connection 2 πδ(0) = lim N→∞ (2N+1).
It is useful now to provide some side-by-side comparisons of results :
Y"(z) = Y' (ω)/T
1 = ∑
n = -∞∞
e-iωnT1 = ∑
m = -∞∞
2π δ(ωT1 - 2πm) infinite (33.4)
Y"(z) = Y' (ω)/T1 = ∑
n = -NN
e-iωnT1 = 2π δ5(ωT1,N) finite (33.15)
X(ω) = X
pulse (ω) ∑
m = -∞∞
2π δ(ωT1 - 2πm) infinite (33.5)
X(ω) = Xpulse (ω) 2π δ5(ωT1,N) . finite (33.16)
|X(ω)|2
[2πδ(0)] = |Xpulse (ω)|2 ∑
m = -∞∞
2π δ(ωT1 - 2πm) infinite (33.11)
|X(ω)|2
(2N+1) = | Xpulse (ω) |2 2π δ6(ωT1, N ) f i n i t e (33.18)
One can interpret |X(ω)|2
[2πδ(0)] as the value of |X( ω)|2 per pulse in an infinite pulse train.
(c) Spectral Power Density of a Simpl e Pulse Train
In this section, everything is in the frequ ency domain, nothing is in the time domain.
Recall from (32.6) that
E(ω) ≡ |X(ω)|
2/2π = energy density in the ω-domain (joule-sec) (32.6)
E(ω)dω = energy in d ω ( j o u l e s )
where E(ω) is the spectral energy density of signal x(t) whose Fourier Transform is X( ω).
If we divide E (ω) by 2N+1 or 2 πδ(0) we obtain the pulse train's average spectral energy density per
pulse, which is the same as the energy density of an average pulse in the pulse train. Using our two
expressions above for infinite and finite pulse trains, we then find
Chapter 6: Power in Pulse Trains
145 E(ω)
[2πδ(0)] = |X(ω)|2
2π[2πδ(0)] = 1
2π |Xpulse (ω)|2 ∑
m = -∞∞
2πδ(ωT1 - 2πm)
E(ω)
(2N+1) = |X(ω)|2
2π(2N+1) = 1
2π | Xpulse (ω) |2 2π δ6(ωT1,N) . (33.20)
If we divide the spectral energy density of the average pulse by T 1, we obtain the spectral power density
of an average pulse, and this is the same as the aver age spectral power density of the pulse train. Thus,
P(ω) ≡ E(ω)
T1[2πδ(0)] = |X(ω)|2
2πT1[2πδ(0)] = 1
2π |Xpulse (ω)|2 (1/T1)∑
m = -∞∞
2πδ(ωT1 - 2πm)
P(ω) ≡ E(ω)
T1[2N+1] = |X(ω)|2
2πT1[2N+1] = 1
2π |Xpulse (ω)|2 (1/T1) 2π δ6(ωT1,N) . (33.21)
If is perhaps helpful to define
T ≡
⎩⎨⎧ [2πδ(0)]T1 infinite pulse train
(2N+1)T 1 finite pulse train . (33.22)
Then (33.21) maybe be restated
P(ω) ≡ E(ω)
T = |X(ω)|2
2πT = 1
2π |Xpulse (ω)|2 (1/T1)∑
m = -∞∞
2πδ(ωT1 - 2πm)
P(ω) ≡ E(ω)
T = |X(ω)|2
2πT = 1
2π |Xpulse (ω)|2 (1/T1) 2π δ6(ωT1,N) . (33.23)
Meanwhile, our x pulse (t) which lasts only for duration T 1 itself has an energy and power density,
Ppulse (ω) ≡ Epulse (ω)
T1 = |Xpulse (ω)|2
2πT1 . (33.24)
Note : Even if x pulse (t) is wider than T 1 as the gaussians are in Fig 14.2, the pulse is associated with the
interval of width T 1, and Xpulse (ω) involves the time integral over all of x pulse (t). It is convenient to
think of the pulse as the black curve in Fig 14. 2, in which case the pulse really fits within T 1.
We can then write (33.23) in the following compact form
P(ω) ≡ P
pulse (ω)∑
m = -∞∞
2πδ(ωT1 - 2πm) joules infinite
P(ω) ≡ Ppulse (ω) 2π δ6(ωT1,N) joules finite (33.25)
Notice that δ(ωT1 - 2πm) and δ6(ωT1,N) are both dimensionless, so the dimensions in each equation
trivially match. A power density P(ω) has dimensions of energy = joules, so that P(ω)dω then has the
Chapter 6: Power in Pulse Trains
146 dimensions of joules/sec = watts, and this is the pulse train power contained in interval d ω of the
spectrum.
For the infinite pulse train, we are always allowed to write 2π δ(ωT
1 - 2πm) = (2π/T1) δ(ω - m(2π/T1)) = ω1 δ(ω - mω1) ω1 ≡ 2π/T1
to get
P(ω) ≡ P
pulse (ω) ω1∑
m = -∞∞
δ(ω - mω1) infinite (33.26)
which shows more explicitly that th e power lines occur at the harmonics ω = mω1. Recall now these
earlier facts,
Chapter 6: Power in Pulse Trains
147 c(ω) ≡ (1/T1)Xpulse (ω) (14.14)
cm ≡ c(mω1) = (1/T1)Xpulse (mω1) . (14.10), (14.8)
For the infinite pulse train we can then write, using (33.24),
P(ω) ≡ P
pulse (ω) ω1∑
m = -∞∞
δ(ω - mω1) = |Xpulse (ω)|2
2πT1 2π
T1 ∑
m = -∞∞
δ(ω - mω1)
= |X
pulse (ω)|2
T12 ∑
m = -∞∞
δ(ω - mω1) = |c(ω)|2 ∑
m = -∞∞
δ(ω - mω1) = ∑
m = -∞∞
|c(mω1)|2 δ(ω - mω1)
so that P(ω) =
∑
m = -∞∞
|cm|2 δ(ω - mω1) . ( 3 3 . 2 7 )
This gives the power spectrum of a simple pulse train in terms of the complex Fourier Series coefficients
cm.
Recall from box (15.12) that Dim(c m) = Dim[x(t)] = volts (say), so |c m|2 = watts into a 1 Ω resistor, and
since δ(ω - mω1) has dimensions sec, |c m|2 δ(ω - mω1) then has dimensions watt-sec = joules, as befits
any P (ω) object.
If we assume x(t) is a real pulse train, then X( ω) is Hermitian, X(- ω) = [ X(ω)]* by (7.1), which
means
c
-m = (1/T1)Xpulse (-mω) = (1/T1)[Xpulse (mω)]* = cm*
so |c
-m|2 = |cm|2 x(t) real (33.28)
and then we can fold the negative part of the sum in (33.27) to get
P(ω) =
∑
m = -∞∞
|cm|2 δ(ω - mω1) = |c0|2 δ(ω) + 2 ∑
m =1∞
|cm|2 δ(ω - mω1) . (33.29)
Finally, recalling from our Fourier Series box (15.2) that c m = [ am - ibm ]/2 and b 0= 0, we get
P(ω) = a02 δ(ω) + (1/2) ∑
m =1∞
(am2+bm2) δ(ω - mω1) . (33.30)
(d) Average Power P of a Simple Pulse Train
We se
ek an expression for the average power P in a general pulse train. This is of course a time domain
quantity, not a frequency domain quantity.
Chapter 6: Power in Pulse Trains
148 P = [ total energy in pulse train / time duration of pulse train ] = average pulse train power
= ( 1 / T ) ∫-∞ ∞ dt |x(t)|2 = ∫-∞ ∞ dω |X(ω)|2
2πT // from (32.4) with R = 1 Ω
so
P = ∫-∞ ∞ dω P(ω) . // from (33.23) (33.31)
This is certainly reasonable since P (ω) is the average spectral power density of the pulse train.
For the special case of an infinite si mple pulse train, we found above that
P(ω) = ∑
m = -∞∞
|cm|2 δ(ω - mω1) = a02 δ(ω) + (1/2) ∑
m =1∞
(am2+bm2) δ(ω - mω1) .
Therefore the power in a simple pulse train is given by (33.31) as
P =
∑
m = -∞∞
|cm|2 = a02 + (1/2) ∑
m =1∞
(am2+bm2) . (33.32)
Chapter 6: Power in Pulse Trains
149 34. Spectral power density of a General Pulse Train
(a) General Pulse Train results and connect ion with the Autocorrelation Function
Certain result
s of the previous section apply to general pulse trains. They are gathered here:
E(ω) ≡ |X(ω)|2/2π = energy density in the ω-domain (joule-sec) (32.6)
T ≡ ⎩⎨⎧ [2πδ(0)]T1 infinite pulse train
(2N+1)T 1 finite pulse train (33.22)
P (ω) ≡ E(ω)
T = |X(ω)|2
2πT (33.23)
Ppulse (ω) ≡ Epulse (ω)
T1 = |Xpulse (ω)|2
2πT1 (33.24)
P = ∫-∞ ∞ dω P(ω) . (33.31)
We now bring the autocorrelation func tion into the discussion, but only in passing. Let x(t) be an arbitrary
but real pulse train, and recall that
rx(t) ≡ ∫-∞ ∞ dt' x(t') x(t' + t) . (32.1)
Then, using T from (33.22),
r
x(0) ≡ ∫-∞ ∞ dt' x(t')2 = P T = E = total energy in the pulse train (34.1)
so the average pulse train power maybe written in terms of the autocorrelation function evaluated at t = 0,
P = r x( 0 ) / T . ( 3 4 . 2 )
We diagonalized (32.1) treated as a convolution equation to obtain
|X(ω)|
2 = Rx(ω),
called the Wiener-Khintchine theorem. Here R
x(ω) is the Fourier Integral Transform of the
autocorrelation function r x(t) of x(t). Then from (33.23) that P(ω) = |X(ω)|2
2πT we get
P(ω) = Rx(ω)/ (2πT ) . ( 3 4 . 3 )
Chapter 6: Power in Pulse Trains
150 In this way, both P and P(ω) can be expressed in terms of the autocorrelation function. Thus, one
approach to finding P and P(ω) for a pulse train is to try and determine r x(t).
Here then is a box summarizing all the general pulse train results:
Energy and Power Properties of a General Pulse Train (34.4)
E(ω) ≡ |X(ω)|2/2π = energy density in the ω-domain (joule-sec) (32.6)
T ≡ ⎩⎨⎧ [2πδ(0)]T1 infinite pulse train
(2N+1)T 1 finite pulse train (33.22)
P (ω) ≡ E(ω)
T = |X(ω)|2
2πT = Rx(ω)/ (2πT) joules (33.23) and (34.3)
Ppulse (ω) ≡ Epulse (ω)
T1 = |Xpulse (ω)|2
2πT1 (33.24)
P = ∫-∞ ∞ dω P(ω) = rx(0)/T watts (33.31) and (34.2)
If P(f)df = P(ω )dω = P(ω) 2πdf , then P(f) = 2 πP(ω).
(b) Spectral power density for a General Pulse Train
In Section 33 (d) we dealt with simple pulse trains. Here
we consider the more general amplitude
modulated pulse train. In all equations, one can replace ∑
n = -∞∞
by ∑
n = -NN
to adapt the equation to a finite
pulse train instead of an infinite one. We start then with
x(t) = ∑
n = -∞∞
yn xpulse (t -tn) (25.1) (34.5)
X(ω) = (1/T 1)Xpulse (ω) Y'(ω) = Xpulse (ω) Y"(z) (25.3) (34.6)
Y"(z) = Y' (ω)/T1 = ∑
n = -∞∞
yn e-iωnT1 . (24.2) (34.7)
To find the frequency-domain power spectrum of a signal x(t), our first task is to compute | X( ω) |2. From
(34.6) we get
|X(ω)|2 = |Xpulse (ω)|2 (1/T1)2 |Y'(ω)|2 = |Xpulse (ω)|2 | Y"(z) |2 z = eiωT1 (34.8)
Chapter 6: Power in Pulse Trains
151 We therefore must deal with the following object, using (34.7),
| Y"(z) |2 = (1/T 1)2 |Y'(ω)|2 = | ∑
n = -∞∞
yn e-iωnT1 | 2
=
∑
n = -∞∞
yn e-iωnT1 ∑
m = -∞∞
ym* e+iωmT1 = ∑
n = -∞∞
∑
m = -∞∞
ym* yn eiω(m-n)T1 (34.9)
so that | X(ω) |
2 = | Xpulse (ω) |2 ∑
n = -∞∞
∑
m = -∞∞
ym* yn eiω(m-n)T1 . (34.10)
From box (34.4) we then have
E(ω) ≡ |X(ω)|
2/2π = (1/2π) | Xpulse (ω) |2 | Y"(z) |2
= (1/2 π) | Xpulse (ω) |2 ∑
n = -∞∞
∑
m = -∞∞
ym* yn eiω(m-n)T1 (34.11)
P(ω) ≡ E(ω)
T = (1/2 πT) | Xpulse (ω) |2| Y"(z) |2
= (1/2π T) | Xpulse (ω) |2∑
n = -∞∞
∑
m = -∞∞
ym* yn eiω(m-n)T1 (34.12)
which we can write as (again using box (34.4) results)
E(ω) = T
1 Ppulse (ω) | Y"(z) |2 = T1 Ppulse (ω) ∑
n = -∞∞
∑
m = -∞∞
ym* yn eiω(m-n)T1 (34.13)
P(ω) = Ppulse (ω) T1
T | Y"(z) |2 = Ppulse (ω) T1
T ∑
n = -∞∞
∑
m = -∞∞
ym* yn eiω(m-n)T1 (34.14)
P = ∫-∞ ∞ dω P(ω) . (33.31) (34.15)
Using the Z Transform Wiener-Khintch ine theorem (32.18) that R"(z) = T1
T | Y"(z) |2 we get
E(ω) = T1 Ppulse (ω) | Y"(z) |2 = T Ppulse (ω) R"(z) (34.13a)
P(ω) = Ppulse (ω) T1
T | Y"(z) |2 = Ppulse (ω) R"(z) (34.14a)
where R"(z) is the Z transform of the autocorrelation sequence r s obtained from the y n.
Not knowing details of the y n there is not much else we can do in these expressions.
Chapter 6: Power in Pulse Trains
152
(c) Pulse Trains with Repeated Sequences
Consider a pulse train com
posed of some general pulse shape x pulse (t) whose amplitudes are repeated
sequences of A,B. We shall compute the spectrum X( ω) and spectral power density P(ω) by two different
methods.
The first method is more or less by brute force, and it reveals a potential pitfall in using the δ(0)
notation and shows a clean way to avoid the pitfall. The second method, much simpler, is to use the Fourier Series results in box (15.12) applied to the repeating sequence.
We then state X( ω) and P(ω) for a few special cases including various square waves.
Appendix F treats the general case of a repeated sequence {A,B,C,D......}.
Method 1: Brute Force Approach
Our starting point is (34.8) with (34.7), where we assume N is large and later we will take N →∞ :
X(ω) = (1/T
1)Xpulse (ω) Y'(ω) = Xpulse (ω) Y"(z) (34.6)
|X(ω)|2 = |Xpulse (ω)|2 (1/T1)2 |Y'(ω)|2 = |Xpulse (ω)|2 | Y"(z) |2 z = eiωT1 (34.8)
Y"(z) = Y' (ω)/T1 = ∑
n = -NN
yn e-iωnT1. (34.7)
The main problem is to compute Y"(z) and then square it. We have
∑
n = -NN
yn e-iωnT1 = A ∑
n even
e-iωnT1 + B ∑
n odd
e-iωnT1 .
Now process the sums as follows, where
∑
n even
e-iωnT1 = ∑
m = -N/2N/2
e-iω(2m)T1 where we used n = 2m
∑
n odd
e-iωnT1 = ∑
n = (-N-1)/2(N-1)/2
e-iω(2m+1)T1 where we used n = 2m + 1 .
We assume N is very large, so we regard (N±1)/2 ≈ N/2 . We then find
Y"(z) = Y' (ω)/T1 = ∑
n = -NN
yn e-iωnT1 = [ A + B e-iωT1]∑
m= -N/2N/2
e-iω(2m)T1 .
Chapter 6: Power in Pulse Trains
153 We now use (13.3),
∑
n = -NN
eink = 2π { sin[(N+1/2)k]
2π sin(k/2) } ≡ 2π δ5(k,N) , - ∞ < k < ∞ (13.3)
to write
∑
m= -N/2N/2
e-iω(2m)T1 = 2π δ5(2ωT1,N/2)
where δ5 and δ6 to come are explained in Appendix A (b). Therefore,
Y"(z) = [ A + B e-iωT1] 2πδ5(2ωT1,N/2) . (34.16)
Using this Appendix A result,
limN→∞ δ5(k,N) = ∑
m = -∞∞
δ(k-2πm ) (A.19)
we obtain the N →∞ limit for our spectrum
Y"(z) = [ A + B e-iωT1] 2π∑
m = -∞∞
δ(2ωT1-2πm)
= (1/2)[ A + B e-iωT1] (1/T1) 2π ∑
m = -∞∞
δ(ω - mω1/2)
= (1/2) ω 1 ∑
m = -∞∞
[ A + B (-1)m] δ(ω-mω1/2) (34.17)
and correspondingly
X(ω) = Xpulse (ω) (1/2) ω1∑
m = -∞∞
[ A + B(-1)m ] δ(ω - mω1/2) . (34.18)
There is a certain logic to the [ A + B e-iωT1] factor. If we set A = K and B = 0 we get one result, and if
we set A = 0 and B = K we get the same result multiplied by e-iωnT1 . The second pulse train is just the
first pulse train shifted T 1 units to the right, and this adds phase e-iωnT1 as in (12.1).
If we were to square (34.18) and use our usual 2πδ (0) = 2N+1 association, we get a result that is off by a
factor of 2. The reason is that our pre-limit sums are going from -N/2 to N/2, so we would get the right answer if we were to adjust and say 2 πδ(0) = N+1. Rather than make an arm-waving argument to this
Chapter 6: Power in Pulse Trains
154 effect, it is safer to continue along with our pre-limit expressions, having paused to take the limit for the
spectrum X( ω) as in (34.18).
So, backing off again from limit, we square (34.16) to get
|Y"(z)|
2 = |A + Be-iωT1|2 [2π δ5(2ωT1,N/2)]2 .
Then from (A.20) applied with N → N/2
δ6(k,N/2) ≡ 1
2π [2πδ5(k,N/2)]2
(N+1) (A.20)
we get
|Y"(z)|
2 = |A + Be-iωT1|2 [2π δ5(2ωT1,N/2)]2
or
|Y"(z)|2
2π(N+1) = |A + Be-iωT1|2 { 1
2π [2πδ5(2ωT1,N/2)]2
(N+1) } = |A + Be-iωT1|2 δ6(2ωT1,N/2) .
Now for large N we ignore the difference between N and N + 1 and so on, so we divide both sides by 2 to
get,
|Y"(z)|
2
2π(2N+1) = (1/2) |A + Be-iωT1|2 δ6(2ωT1, N / 2 )
Notice that a very important factor of 1/2 appears on the right in the last step. We now insert the squared pulse spectrum to get
|X(ω)|
2
2π(2N+1) = |Xpulse (ω)|2 |Y"(z)|2
2π(2N+1) = |Xpulse (ω)|2 (1/2)|A + Be-iωT1|2 δ6(2ωT1,N/2) .
If we divide both sides by T 1 the left side is |X(ω)|2
2πT where T is the length of the pulse train and this in
turn equals P(ω), all as shown in box (34.4). So for large N we have shown that
P(ω) = |Xpulse (ω)|2 (1/T1)(1/2) |A + Be-iωT1|2 δ6(2ωT1, N / 2 )
= P
pulse (ω)(1/2) |A + Be-iωT1|2 2π δ6(2ωT1,N/2) . (34.19)
Now at last we take the limit N →∞ and use
limN→∞ δ6(k,N) = ∑
m = -∞∞
δ(k-2πm ) (A.21)
Chapter 6: Power in Pulse Trains
155
to get our desired infinite pulse train result
P(ω) = Ppulse (ω)(1/2) |A + Be-iωT1|2 ∑
m = -∞∞
2π δ(2ωT1 - 2πm)
= Ppulse (ω)(1/4) |A + Be-iωT1|2 (1/T1)∑
m = -∞∞
2π δ(ω - mω1/ 2 )
= Ppulse (ω)(1/4) (1/T 1)∑
m = -∞∞
|A + B(-1)m |2 2π δ(ω - mω1/2)
= Ppulse (ω)(1/4) ω1∑
m = -∞∞
{ |A|2 + |B|2 + 2Re(AB*)(-1)m } δ(ω - mω1/2) . (34.20)
Summarizing the key results:
Fig 34.1
X(ω) = Xpulse (ω) (1/2) ω1∑
m = -∞∞
[ A + B(-1)m ] δ(ω - mω1/2) . (34.18)
P (ω) = Ppulse (ω) (1/4) ω1∑
m = -∞∞
{ |A|2 + |B|2 + 2Re(AB*)(-1)m } δ(ω - mω1/2) (34.20)
Method 2: Fourier Series Approach
We regard our A,B pair of pulses as a single pulse x
PULSE (t) of width 2T 1.
xPULSE (t) = ⎩⎨⎧ A xpulse (t) 0 < t < T 1
B xpulse (t-T1) T1 < t < 2T 1 .
We then apply the Fourier Series results of box (15.12) but with T 1→ 2T1 (so ω1 → ω1/2)
x(t) = ∑
n = -∞∞
xPULSE (t - n2T1)
Cm = (1/2T 1) ∫0 2T1 dt xPULSE (t) e-imω1t/2 .
We then calculate C m as follows
Cm = (1/2T 1) ∫0 2T1 dt xPULSE (t) e-imω1t/2
Chapter 6: Power in Pulse Trains
156 = (1/2T 1) A ∫0 T1 dt xpulse (t) e-imω1t/2 + (1/2T 1) B ∫T1 2T1 dt xpulse (t-T1) e-imω1t/2
= (1/2T 1) A ∫0 T1 dt xpulse (t) e-imω1t/2 + (1/2T 1) Be-imω1(T1/2) ∫0 T1 dt' xpulse (t') e-imω1t'/2
= (1/2) [ A + B (-1)m ] (1/T1) ∫0 T1 dt xpulse (t) e-imω1t/2
= (1/2) [ A + B (-1)m ] cm/2
so that C
m = (1/2) [ A + B (-1)m ] cm/2 ,
where cn are the Fourier coefficients for a simple pulse train made from x pulse (t) pulses. The spectrum
can then be read from box (14.12) item 3, where we continue to replace T 1 → 2T1 ,
X(ω) = ∑
m = -∞∞
Cm 2π δ(ω - mω1/2) .
But from (14.10) and (14.8) we know that
c
m/2 = (1/T1)Xpulse (mω1/2)
so that
C
m = (1/2) [ A + B (-1)m ] (1/T1)Xpulse (mω1/2) .
Then
X(ω) =∑
m = -∞∞
{ (1/2) [ A + B (-1)m ] (1/T1)Xpulse (mω1/2)} 2π δ(ω - mω1/2)
= X pulse (ω) (1/2) ω1∑
m = -∞∞
[ A + B (-1)m ] δ(ω - mω1/2)
which agrees with our Method 1 result (34.18).
Then from (33.29) we get
P(ω) =
∑
m = -∞∞
|Cm|2 δ(ω - mω1/2)
= ∑
m = -∞∞
| (1/T1)Xpulse (mω1/2) (1/2) [ A + B(-1)m ] |2 δ(ω - mω1/2)
Chapter 6: Power in Pulse Trains
157 = ( 1 / T 1)2 | Xpulse (ω) |2 (1/4) ∑
m = -∞∞
| [ A + B(-1)m ] |2 δ(ω - mω1/2) // now use (33.24)
= Ppulse (ω) (1/4) ω1 ∑
m = -∞∞
{ |A|2 + |B|2 + 2Re(AB*)(-1)m } δ(ω - mω1/2)
and this agrees with our Method 1 result (34.20).
Chapter 6: Power in Pulse Trains
158 Special Case Group 1
Suppose A = 1 and B = -1. Then
{ |A|
2 + |B|2 + 2Re(AB*)(-1)m } = { 1 + 1 - 2(-1)m} = 2 [ 1 - (-1)m ] .
Our general results are these
X(ω) = Xpulse (ω) (1/2) ω1∑
m = -∞∞
[ A + B(-1)m ] δ(ω - mω1/2) . (34.18)
P (ω) = Ppulse (ω) (1/4) ω1∑
m = -∞∞
{ |A|2 + |B|2 + 2Re(AB*)(-1)m } δ(ω - mω1/2) (34.20)
which become X(ω) = X
pulse (ω) (1/2) ω1∑
m = -∞∞
[ 1 - (-1)m ] δ(ω-mω1/2)
P (ω) = Ppulse (ω) (1/2) ω1∑
m = -∞∞
{ 1 – (-1)m } δ(ω - mω1/2)
or
Fig 34.2
X(ω) = Xpulse (ω) ω1∑
m = ±odd
δ(ω - mω1/2)
P (ω) = Ppulse (ω) ω1 ∑
m = ±odd
δ(ω - mω1/2) . (34.21)
Square wave pulse train with peak-to-peak = 2 units and period 2T 1
X
pulse (ω) = T1 sinc(ωT1/ 2 ) (9.2)
Ppulse (ω) = |Xpulse (ω)|2/ (2πT1) = (1/ω1) sinc2(ωT1/2) . (34.22)
Evaluated at ω = mω1/2, ω T1/2 = mπ/2 and sin(m π/2) = 0 for m even and (-1)(m-1)/2 for m odd.
Therefore,
Xpulse (mω1/2) = T1 (-1)(m-1)/2 / (mπ/2) = (2/π) T1 (-1)(m-1)/2 (1/m)
Ppulse (mω1/2) = (1/ω1) (-1)(m-1)/ (mπ/2)2 = (2/π)2(1/ω1) (-1)(m-1)(1/m2) .
For m odd, (-1)(m-1) = 1, so we get
Chapter 6: Power in Pulse Trains
159
Fig 34.3
X(ω) = 4 ∑
m = ±odd
(-1)(m-1)/2 (1/m)δ(ω - mω1/2)
P (ω) = (2/π)2∑
m = ±odd
(1/m2) δ(ω - mω1/2) . (34.23)
Square wave pulse train with peak-to-peak = 1 units and period 2T 1
In (34.23) X goes to 1/2 and P goes to 1/4 :
Fig 34.4
X(ω) = 2 ∑
m = ±odd
(-1)(m-1)/2 (1/m)δ(ω - mω1/2)
P (ω) = (1/π)2∑
m = ±odd
(1/m2) δ(ω - mω1/2) . (34.24)
Square wave pulse train with peak-to-peak = 2 units and period T 1 :
In (34.23) replace ω1 → 2ω1 :
Fig 34.5
X(ω) = 4 ∑
m = ±odd
(-1)(m-1)/2 (1/m)δ(ω - mω1)
P (ω) = (2/π)2∑
m = ±odd
(1/m2) δ(ω - mω1) . (34.25)
Square wave pulse train with peak-to-peak = 1 unit and period T 1 :
In (34.25) X goes to 1/2 and P goes to 1/4 :
Chapter 6: Power in Pulse Trains
160
Fig 34.6
X(ω) = 2 ∑
m = ±odd
(-1)(m-1)/2 (1/m)δ(ω - mω1)
P (ω) = (1/π)2∑
m = ±odd
(1/m2) δ(ω - mω1) . (34.26)
In (17.9) we showed that for the above square wave c m = (1/πm) (i)1-m for odd m. Then (17.4) says
X(ω) = ∑
m = ±odd
cm 2πδ(ω - mω1 ) = ∑
m = ±odd
(1/πm) (i)1-m 2πδ(ω - mω1 )
= 2 ∑
m = ±odd
(1/m) (-1)(m-1)/2 δ(ω - mω1 ) . // agrees with (34.26)
The power density from (33.27) is
P(ω) =
∑
m = -∞∞
|cm|2 δ(ω - mω1) = ∑
m = ±odd
(1/πm)2 δ(ω - mω1) . // agrees with (34.26)
Special Case 2
Here A = 1 and B = 0. Our general results (34.18) and (34.20) become
Fig 34.7
X(ω) = Xpulse (ω) (1/2) ω1∑
m = -∞∞
δ(ω - mω1/2)
P (ω) = Ppulse (ω) (1/4) ω1∑
m = -∞∞
δ(ω - mω1/2) . (34.27)
Inserting the same square wave pulse shown in (34.22) this becomes
X(ω) = π
∑
m = -∞∞
sinc(mπ/2) δ(ω - mω1/2)
P (ω) = (1/4) ∑
m = -∞∞
sinc2(mπ/2) δ (ω - mω1/2) .
Separating out the m = 0 term and using sin(m π/2) = (-1)(m-1)/2 we get
Chapter 6: Power in Pulse Trains
161
Fig 34.8
X(ω) = 2 ∑
m = ±odd
(1/m) (-1)(m-1)/2 δ(ω - mω1/2) + (1/2) 2 πδ(ω)
P (ω) = (1/π2)∑
m = ±odd
(1/m2) δ(ω - mω1/2) + (1/4) δ(ω) (34.28)
These match (34.24) but here we have DC terms.
Special Case 3 : Recovering the Simple Pulse Train Spectra
Here A = 1 and B = 1. Our general results (34.18) and (34.20) become
X(ω) = X
pulse (ω) (1/2) ω1∑
m = -∞∞
[ 1 + (-1)m ] δ(ω - mω1/ 2 ) (34.18)
P (ω) = Ppulse (ω) (1/2) ω1∑
m = -∞∞
{ 1 + (-1)m } δ(ω - mω1/ 2 ) (34.20)
or X(ω) = X
pulse (ω) ω1∑
m = ±even
δ(ω - mω1/2)
P (ω) = Ppulse (ω) ω1∑
m = ±even
δ(ω - mω1/2)
or
Fig 34.9
X(ω) = Xpulse (ω) ω1∑
m = -∞∞
δ(ω - mω1) // agrees with (14.5) for simple pulse train
P (ω) = Ppulse (ω) ω1∑
m = -∞∞
δ(ω - mω1) // agrees with (33.26) for simple pulse train
Chapter 6: Power in Pulse Trains
162 Exercises for the Reader:
(a) If the repeating amplitude sequence is A,B,C show that
X ( ω) = X
pulse (ω) (1/3)[ A + B e-iωT1 + C e-i2ωT1 ] ω1 ∑
m = -∞∞
δ(ω - mω1/3)
P (ω) = Ppulse (ω) (1/3)2 |A + Be-iωT1 + C e-i2ωT1|2 ω1∑
m = -∞∞
δ(ω - mω1/3) . (34.29)
(b) If the repeating sequence is y 0,y1....yP-1 show that
X ( ω) = Xpulse (ω) 1
P [ ∑
k = 0P-1
yke-ikωT1 ] ω1 ∑
m = -∞∞
δ(ω - mω1/P)
P (ω) = Ppulse (ω) 1
P2 | ∑
k = 0P-1
yke-ikωT1 |2 ω1∑
m = -∞∞
δ(ω - mω1/P) . (34.30)
These results can be expressed in terms of the Z Transform Y P"(z) = Σk=0P-1 yk z-k of the repeated
sequence, where z = eikωT1 :
X ( ω) = Xpulse (ω) 1
P YP"(z) ω1 ∑
m = -∞∞
δ(ω - mω1/P)
P (ω) = Ppulse (ω) 1
P2 | YP"(z)|2 ω1∑
m = -∞∞
δ(ω - mω1/P) (34.31)
Hint: These two equations are derived in Appendix F as (F.12) and (F.23).
Chapter 6: Power in Pulse Trains
163 35. Statistical Pulse Trains
(a) Spectral power density for a Statistical Pulse Train
We i
magine now a large ensemble of I pulse trains in which a particular pulse train is labeled by index i.
This pulse train has coefficients y n(i) . We are interested in the ensemble averages of E(ω) and P(ω) and
P. We indicate the ensemble average of some quantit y Q by <Q>. By averaging equations (34.13) through
(34.15) we obtain
< E(ω)> = T
1 Ppulse (ω) ∑
n = -∞∞
∑
m = -∞∞
<ym* yn> eiω(m-n)T1 (35.1)
< P(ω)> = T1 Ppulse (ω) (1/T) ∑
n = -∞∞
∑
m = -∞∞
<ym* yn> eiω(m-n)T1 (35.2)
<P> = ∫-∞ ∞ dω <P(ω ) > (35.3)
where
<ym* yn> = (1/I) ∑
i = 1I
ym(i)* yn(i) . ( 3 5 . 4 )
Up to this point we have been general, allowing the y m(i) to be complex coefficients, but from now on we
consider them to be real, so we can dele te the asterisks in the above equations.
In general one can consider statistical pulse trains whose amplitudes are ra ndomly selected in some
manner from a discrete set of possible amplitudes {A,B ,C....}, or perhaps even from a continuous set. We
shall limit our interest to the binary discrete set {A,B}, but the methods outlined below can be generalized for signals having more than two encoded amplitudes. A very simple case to consider is this, where A and B are real,
y
m = A probability p
ym = B probability 1-p . (35.5)
Remember that index m labels pulse location m in the pulse train. If probabilities of a pulse value at
locations m and at n are uncorrelated (they are independent of each other), then we can write the
averaged coefficients as follows, first for two different locations m ≠ n, then for the same location m = n :
n ≠ m α ≡ <ymyn> = [pp] AA + [p(1-p)] AB + [(1-p)p]BA + [(1-p)(1-p)]BB
n = m β ≡ <y
m2> = [p]AA + [(1-p)] BB . (35.6)
Chapter 6: Power in Pulse Trains
164 In the square brackets [..] we indicate the probability of some case occurring, and this is multiplied by the
value that the quantity in question takes in that case. The reader must now stop reading and stare at the
above equations until they make complete sense. Notice that the second line is quite distinct from the first
line. In the double summation in (35.2), there ar e both diagonal terms where m = n, and off-diagonal
terms where m ≠n. These groupings must be treated separately according to the above. We have defined
new symbols α and β to emphasize that, for the situations shown, there is no dependence on the n and m
indices for these quantities. In a later sections, we sha ll encounter situations wher e the above equations do
not apply because there is correlation between positions m and n.
Inserting (35.6) into (35.2) gives a fundamental result [ T as in (34.3) ]
< P(ω)> = T
1 Ppulse (ω) (1/T) { α∑
n = -∞∞
∑
m ≠ n
[ eiω(m-n)T1] + β∑
n = -∞∞
∑
m = n
[1] } . (35.7)
Appendix D shows how α and β may be written in terms of the mean μ and variance σ2 of the values of
the amplitudes y m(i) for the pulses at position m in the pulse train ensemble. There we show that
<y
m> = μ = [p]A + [1-p]B
n ≠ m α ≡ <y
myn> = <ym>2 = μ2 = { [p]A + [1-p]B }2 = [pp] AA + [p(1-p)] 2AB + [(1-p)(1-)]BB
n = m β ≡ <y
m2> = [p]AA + [(1-p)] BB = σ2 + μ2 = σ2 + α (D.30)
σ
2 = [ p(1-p)] (A-B)2 . (D.32)
The key relations of interest are these,
α = μ2
β = μ2 + σ2 = α + σ2
σ2 = (β -α) . (D.31)
Thus we can write (35.7) as
< P(ω)> = T1 Ppulse (ω) (1/T) { μ2∑
n = -∞∞
∑
m ≠ n
[ eiω(m-n)T1] + (μ2 + σ2)∑
n = -∞∞
∑
m = n
[1] }
(35.7a)
In general we shall contin ue to use the parameters α and β, occasionally referring to μ and σ.
(b) Infinite Statistical Pulse Train with Non-Correlated Coefficients
Recall now equation (
35.7),
< P(ω)> = T1 Ppulse (ω) (1/T) { α∑
n = -∞∞
∑
m ≠ n
[ eiω(m-n)T1] + β∑
n = -∞∞
∑
m = n
[1] } . (35.7)
Chapter 6: Power in Pulse Trains
165 To evaluate the first double sum, we write it as
∑
n = -∞∞
∑
m ≠ n
[ eiω(m-n)T1] = ∑
n = -∞∞
{ ∑
m = -∞∞
[ eiω(m-n)T1] – 1 }
= (
∑
n = -∞∞
e+iωnT1 ) (∑
m = -∞∞
e-iωmT1 ) – ∑
n = -∞∞
1 = | ∑
n = -∞∞
e+iωnT1 |2 - ∑
n = -∞∞
1 . (35.8)
Note that each sum is real, according to (13.2), so we can replace | |2 by { }2.
We now quote two results from Appendix A,
∑
n = -∞∞
1 = [ 2 π δ( 0 ) ] (A.36)
{ ∑
n = -∞∞
einωT1 }2 = [ 2πδ (0) ] { ∑
m = -∞∞
2π δ(ωT1- 2πm) } , (A.39)
so the first double sum in (35.7) becomes
∑
n = -∞∞
∑
m ≠ n
[ eiω(m-n)T1] = [ 2πδ(0) ] { ∑
m = -∞∞
2π δ(ωT1- 2πm) - 1} . (35.9)
From (A.36) the second double sum in (35.7) is just [ Σ m=n 1 for fixed n has just one term which is 1 ]
∑
n = -∞∞
∑
m = n
[1] = ∑
n = -∞∞
[1] = [ 2π δ(0)] , (35.10)
and therefore we may write (35.7) as
< P(ω)> = T
1 Ppulse (ω) (1/T) { α [ 2πδ(0) ] [ ∑
m = -∞∞
2π δ(ωT1- 2πm) - 1] + β [ 2π δ(0)] }
= Ppulse (ω) (T1 [ 2πδ(0) ]/T) { α [ ∑
m = -∞∞
2π δ(ωT1- 2πm) - 1] + β }
or < P(ω)> = P
pulse (ω) { (β -α) + α∑
m = -∞∞
2π δ(ωT1- 2πm) } (35.11)
where we used T = [2 πδ(0)]T1 from (33.22). This is a famous result which will be applied below.
In Appendix D (D.31) we show that ( β-α) = σ2 and α = μ2 where μ and σ2 are the mean and variance
of the pulse train amplitudes y m(i) which occur at position m in the pulse train. Thus (35.11) may be
written,
Chapter 6: Power in Pulse Trains
166 < P(ω)> = Ppulse (ω) { σ2 + μ2∑
m = -∞∞
2π δ(ωT1-2πm) } . (35.11a)
In this form we see that the continuous part of the spectrum arises from the variance σ2 in the ensemble
values of the pulse train amplitudes y m(i), while the discrete part of the spectrum is present only if the
pulse amplitudes have a non-vanishing mean μ .
Verification
To find external verification for (35.11), we write things in terms of f where ω = 2πf,
P
pulse (ω) = |Xpulse (ω)|2
2πT1 = |Xpulse (f)|2
2πT1 // (34.4) and text after (1.4)
P(f) = 2 πP(ω) // (34.4) last item
2π δ(ωT1-2πm) = 2π δ(2πfT1-2πm) = (1/T 1) δ( f - m/T 1) .
Then (35.11a) becomes
1
2π <P(f)> = |Xpulse (f)|2
2πT1 { σ2 + μ2
T1 ∑
m = -∞∞
δ( f - m/T 1) }
or
<P(f)> = |Xpulse (f)|2
T1 { σ2 + μ2
T1 ∑
m = -∞∞
δ( f - m/T 1) } (35.11b)
This may be compared to result (A.17) in Appendix A of Xiong which we quote,
(c) Finite Statistical Pulse Train with Non-Correlated Coefficients
Recall again
equation (35.7) but assume now a fi nite pulse train so the sums are different
< P(ω)> = T1 Ppulse (ω) (1/T) { α∑
n = -NN
∑
m ≠ n
[ eiω(m-n)T1] + β∑
n = -NN
∑
m = n
[1] } . (35.12)
To evaluate the first double sum, we write it as
Chapter 6: Power in Pulse Trains
167 ∑
n = -NN
∑
m ≠ n
[ eiω(m-n)T1] = ∑
n = -NN
{ ∑
m = -NN
[ eiω(m-n)T1] – 1 }
= ( ∑
n = -NN
e+iωnT1 ) (∑
m = -NN
e-iωmT1 ) – ∑
n = -NN
1 = | ∑
n = -NN
e+iωnT1 |2 - (2N+1) . (35.13)
where | |2 = { }2 since (13.3) says each sum is real. From Appendix A we have
∑
n = -NN
eink = 2π { sin[(N+1/2)k]
2π sin(k/2) } = 2πδ 5(k,N) , (A.30)
so using (35.13) the first double sum in (35.12) becomes
∑
n = -NN
∑
m ≠ n
[ eiω(m-n)T1] = [2πδ5(ωT1,N)]2 - (2N+1) . (35.14)
The second double sum in (35.12) is just
∑
n = -NN
∑
m = n
[1] = ∑
n = -NN
[1] = (2N+1) (35.15)
and therefore we may write (35.12) as
< P(ω)> = T
1 Ppulse (ω) (1/T) { α { [2πδ5(ωT1,N)]2 - (2N+1)} + β(2N+1) }
= Ppulse (ω) ((2N+1) T 1/T) { α { [2πδ5(ωT1,N)]2
(2N+1) - 1)} + β }
= Ppulse (ω) { (β-α) + α [2πδ5(ωT1,N)]2
(2N+1) }
where this time we used T = (2N+1)T 1 from (33.22). The factor multiplying α can be replaced using
(A.20),
δ6(k,N) ≡ 1
2π [2πδ5(ωT1,N)]2
(2N+1) = 1
2N+1 sin2[(N+1/2)k]
2π sin2(k/2) (A.20)
to give
< P(ω)> = P
pulse (ω) { (β-α) + α 2π δ6(ωT1,N) } (35.16)
which is the finite pulse train version of (35.11). In the limit N → ∞ we know from (A.21)
Chapter 6: Power in Pulse Trains
168 limN→∞ δ6(k,N) = ∑
m = -∞∞
δ(k-2πm ) (A.21)
so that (35.16) becomes
< P(ω)> = Ppulse (ω) { (β -α) + α∑
m = -∞∞
2π δ(ωT1-2πm) }
which reproduces (35.11).
(d) Statistical Non-Correlated Pulse Trains: Summary and Examples
Here then is a brief summary
of the above results:
Spectral power density of a Non-Correlated Pulse Train (35.17)
x(t) = ∑
n = -∞∞
yn xpulse (t - nT1) // or ∑
n = -NN
for a finite pulse train
< P(ω)> = Ppulse (ω) [ (β -α) + α ∑
m = -∞∞
2π δ(ωT1- 2πm) ] // infinite (35.11)
< P(ω)> = Ppulse (ω) [ (β -α) + α 2π δ6(ωT1,N) ] // finite (35.16)
α = [pp] AA + [p(1-p)] AB + [(1-p)p]BA + [(1-p)(1-p)]BB // = <y myn> when n≠m
(35.6)
β = [p]AA + [(1-p)] BB // = <y myn> when n=m
A pulse has probably p of having amplitude A, and probability 1-p of having amplitude B.
A=1 and B=0 => α = p2 β = p (β - α) = p(1-p)
Example 1 : Assuming A = 1 and B = 0, setting p = 1 in the above box gives α = 1 and ( β-α) = 0. In this
limit, our statistical average pulse train spectrum becom es the same as that for the simple pulse train of
(33.25)
P(ω) ≡ P
pulse (ω)∑
m = -∞∞
2πδ(ωT1 - 2πm) joules
P(ω) ≡ Ppulse (ω) 2π δ6(ωT1, N ) j o u l e s (33.25)
This is because with p = 1 every pu lse in the pulse train has the same amplitude A=1, which is how a
simple pulse train is defined. As we reduce p be low p = 1, the line spectra are scaled down by α = p2 < 1,
Chapter 6: Power in Pulse Trains
169 and a continuous spectrum starts to appear with ( β - α) = p(1-p). The randomness (variance σ2) of the
ensemble creates a continuous component in the spectral power density P(ω) as per (35.11a) .
Example 2 : Let p=0 with A=1 and B=0. Then every pulse has B = 0, x(t) ≡ 0, α = β = 0, and both terms in
P(ω) vanish so there is zero spectral energy density.
Example 3: Let p=1/2 with A=1 and B=0, so α = p2 = 1/4 and β = p = 1/2. This pulse train then has an
equal probably of 1's and 0's. We find from box (35.17) that
< P(ω)> = Ppulse (ω) [(1/4) + (1/4) ∑
m = -∞∞
2π δ(ωT1- 2πm) ] infinite (35.18)
< P(ω)> = Ppulse (ω) [(1/4) + (1/4) 2 π δ6(ωT1,N) ] finite (35.19)
In (35.18) the discrete spectrum has been reduced to 1/4 of its full strength, and a continuous spectrum
exists with coefficient 1/4 as shown.
(e) A numerical example of a Statistical Pulse Train
Recall fro
m box (34.4) that for a finite pulse train,
P(ω) = |X(ω)|2
2πT Ppulse (ω) = |Xpulse (ω)|2
2πT1 T = (2N+1)T 1. (35.20)
Therefore we can write our p = 1/2 Example 3 result (35.19) as
<|X(ω)|
2>
(2N+1) = |X pulse (ω)|2 [(1/4) + (1/4) 2 π δ6(ωT1,N) ] . (35.21)
We shall use for x pulse (t) a square pulse of height A=1 and width τ = T1 = 1 so that, from (9.2),
|X
pulse (ω) | = sinc( ω/ 2 ) . ( 3 5 . 2 2 )
Since our pulse train will be fairly short (N = 20 pulses) and since we shall only average a small number
of pulse trains (M = 10), we know our result will not exactly match (35.21). Still, we hope to see in our
result some kind of continuous background spectrum which approximates the curve (1/4)|X pulse (ω)|2 =
(1/4) sinc2(ω/2), and we expect to see delta-function-like peaks which, since 2 πδ6(0,N) = (2N+1), have a
peak value of about (1/4)41 = 10.25. Since this will be added to the conti nuous background, the peak
should have a height of 10.25 + .25 = 10.5. However, for our small ensemble, we won't have exactly p =
1/2, so the delta peak won't be exactly 10.5 units high.
We know that δ
6 has identical peaks spaced by 2 π, but we expect the non-central peaks to be suppressed
by the sinc2(ω/2) zeros which occur at ω = n(2π).
Chapter 6: Power in Pulse Trains
170 Here is the self-documented Ma ple program which generates <|X( ω)|2> . The program also generates the
quantity <X( ω)> upon which we shall comment in Section 36 below.
At this point, before X pulse (ω) is added to the result, we plot <|X( ω)|2>. As expected, we see the peaks of
δ6 spaced by 2 π and having height around 10 units,
F i g 3 5 . 1
We now insert copies of X
pulse (ω) as appropriate,
Chapter 6: Power in Pulse Trains
171
and then we can plot <|X(ω)|2>
(2N+1) for ω in the same range (-10,10)
F i g 3 5 . 2
We see that the zeros of the sinc functi on have killed off the adjacent peaks.
Next, we restrict the plot height to be 0.8 units to view the detail, chopping off the δ
6 peak,
F i g 3 5 . 3
The spectrum <|X(ω)|2>
(2N+1) is seen to have a continuous com ponent which very well approximates one
quarter of the sinc2 curve, as we hoped it would. This tracking also occurs away from the central peak.
Here is a blow-up of the above plot for ω in the range (5, 30)
Chapter 6: Power in Pulse Trains
172
F i g 3 5 . 4
It might be noted that Maple does this work analytically, so that Was-av is a function of ω having a large
number of trigonometric terms. For the reader's interest, we show Was-av( ω) for a typical program run :
In more serious work with larger numbers, one would of course do this in a more numeric fashion, but we
are able to confirm the basic results even with this small experiment.
(f) What role has the Autocorrelation Function played in our development?
In Section 32 the autocorrelation function was first introduced as r x(t) for x(t), and was computed for the
simple case of a box pulse. Treating the definition of r x(t) as a convolution equation, the Wiener-
Khintchine Relation R x(ω) = |X(ω )|2 was trivially derived. It was then shown that the spectral energy
density of a pulse train is E(ω) = (1/2π) Rx(ω) due to this relation. Finally, it was shown that r x(0) = E, the
total energy in the pulse train. We then commented on the origin of the name, showing that the auto-
correlation function is the cross-correlation function of a function with itself when that function is real.
Chapter 6: Power in Pulse Trains
173 In Section 34 it was noted again that P = r x(0)/T since P = E/T, and that P(ω) = Rx(ω)/ (2πT) since R x(ω)
= |X(ω)|2.
Sections 33 and 35 made no reference at all to the autocorrelation function.
This leads us to make several comments: (1) The autocorrelation function played no role whatsoev er in our development of key equations such as
< P(ω)> = P
pulse (ω) [ (β -α) + α ∑
m = -∞∞
2π δ(ωT1- 2πm) ] // infinite (35.11)
(2) In some textbooks, one gets the impression that the autocorrelation function is somehow crucial for
the development of such equations. It is not.
(3) Nevertheless, since P(ω) = R
x(ω)/ (2πT), one can start with a description of x(i)(t) of pulse train i in
a statistical ensemble, compute from it r x(i)(t) and from that R x(i)(ω) . One could then do a statistical
average to obtain < P(ω)> = (2πT)-1(1/I) Σi=1I Rx(i)(ω). So it is possible to take a pathway to deriving
equations like (35.11) which does pass through the land of the autocorrelation function.
(4) Our main reason for even bringing it up is that au tocorrelation is closely related to what we are doing,
and in other applications such as those involving noise (and pseudo-noise PN) it becomes more
significant. In such applications, the definition of r
x(t) might include a normalizing factor so it is then
autocorrelation per unit time, or per pulse (per chip in the PN world).
(5) In (34.14a) we noted that P(ω) = Ppulse (ω) R"(z) where R"(z) is the Z transform the autocorrelation
sequence rs defined in (32.17). Then in Appendix F (e) we compute R"(z) for a sequence which respects
certain simple conditions met by a Ma ximum Length Sequence (MLS). The resulting P(ω) is not easily
computed in any other manner, so in this case the pa thway through the autocorrelation route is extremely
valuable.
(g) A paradox and its resolution
Now while here, we
can back up and apply our statistical average directly to the spectrum in (34.6) using
(34.7). The result is
<X(ω)> = X
pulse (ω) ∑
n = -∞∞
<yn> e-inωT1 . (34.6,7)
We could then argue that
<y
n> = [p] A + [1-p] B = p if A=1, B=0 . (35.23)
Chapter 6: Power in Pulse Trains
174 In this case, we can extract p from the above sum, wh ich then collapses to form the usual (13.2) delta
function sum. The result is then the same as the regular pulse train result (14.4) with an overall factor of p
out front, namely,
<X(ω)> = p ∑
m = -∞∞
Xpulse (mω1) 2π δ( ωT1 - 2πm) . (35.24)
Thus says that our average spectrum is 100% discrete , there is no continuous part! If p = 1/2, the average
spectrum is just 1/2 times our discrete unit amplitude pulse train spectrum (14.4). How can this be true, if
we just showed in (35.11) that the average spectral density <|X( ω)|2> has a continuous spectral
component?
The answer lies in the fact that <ab> ≠ <a><b>, where < > is our averaging operation. Thus
<|X(ω)|
2> ≠ <X(ω)><X(ω ) * > . ( 3 5 . 2 5 )
There is no reason in the world why the average of a product should be the product of the averages,
( 1
N Σi=1N AiBi) ≠ ( 1
N Σi=1N Ai) ( 1
N Σi=1N Bi) .
In the numerical example presented in Section 35 (f), we computed <X( ω)> for a small ensemble of pulse
trains. Here are plots of the real and imaginary part of <X( ω)>,
F i g 3 5 . 5
We can see that, apart from the noise of our low statisti cs, there is only the central peak and no continuous
spectrum component. A blow-up of the central region follows,
Chapter 6: Power in Pulse Trains
175
F i g 3 5 . 6
36. Application to some Standard Non- Correlated Pulse Train Types (Line Codes)
Here we apply our boxed results (35.17) to random pulse trains of various types. When a pulse train is in
fact a voltage on a pair of wires (transmission line, such as a telephone "line"), the way in which signals are encoded in the pulse train is called a line code . One could consider a random speed Morse code signal
going down a wire as a line code, but the term usually refers to a sequence of equally spaced amplitude
modulated pulses, meaning a pulse train. Often line codes get modulated onto an RF carrier, in which case the line code is thought of as the baseband signal prior to modulation. For this reason, line codes are often discussed in the "baseband chapter" of any digital communications text.
The line code names are a little strange due to thei r history. Here are the pulse shapes used for RZ and
NRZ lines codes. In either case a 1 is (is coded as) a pulse and a 0 is no pulse.
F i g 3 6 . 1
On the left, since the signal returns to zero inside the pulse period, it is called a "return to zero" code RZ.
Since this does not happen on the right, that is a "non return to zero" code, NRZ.
(a) Unipolar NRZ line code
Pulse Shape
. The pulse is a box of amplitude V and width τ = T1,
Fig 36.2
From (9.2) we know that
Chapter 6: Power in Pulse Trains
176
Xpulse (ω) = (VT1) sinc(ω T1/ 2 )
( 3 6 . 1 )
Ppulse (ω) = |Xpulse (ω)|2
2πT1 = (VT 1)2 sinc2(ωT1/2)/(2πT1) = (1/2π) V2T1 sinc2(ωT1/2)
Coding : NRZ is a normal binary signal, high for period T 1 to indicate a 1, and low for T 1 to indicate a 0.
Sometimes this is called unipolar NRZ since the signal never goes negative.
Fig 36.3
Coefficients α and β: Looking at summary box (35.17), since A = 1 and B = 0 we have α = p2 and β = p.
Spectrum: The average spectral power density for the NRZ line code is :
< P(ω)> = Ppulse (ω) [(β-α) + α ∑
m = -∞∞
2πδ(ωT1- 2πm ) ] (35.11)
= (1/2 π) V2T1 sinc2(ωT1/2) [ p(1-p) + p2∑
m = -∞∞
2π δ(ωT1- 2πm) ]
= (V2/ω1) sinc2(π ω
ω1 ) [ p(1-p) + p2∑
m = -∞∞
δ( ω
ω1 - m) ] .
But for all m ≠0, the sinc function vanishes, so the discrete part of the spectrum collapses to a single term
and we get
< P(ω)> = (V
2/ω1) sinc2(π ω
ω1 ) [ p(1-p) + p2 δ( ω
ω1 ) ]
From now on we shall express < P(ω)> in terms of a dimensionless frequency x,
x ≡ ω
ω1 ω1 ≡ 2π/T1 => πx = (ωT1/2) (36.2)
so the above becomes
< P(ω)> = (V
2/ω1) sinc2(πx) [ p(1-p) + p2 δ(x) ]
= (V2/ω1) [p(1-p)sinc2(πx) + p2 δ(x)] // unipolar NRZ (36.3)
Chapter 6: Power in Pulse Trains
177 In order to express this (and later) results in the frequency domain, we use these relations
P(ω) = P(f)/2π // (34.4) bottom line
1/ω1 = T1/2π
x = ω/ω1 = fT1 // ω = 2πf
δ(x) = δ(f)/T1
to obtain
<P(f)> = V
2 [p(1-p) T 1sinc2(πfT1) + p2δ(f)] // unipolar NRZ, f (36.3a)
<P(f)> = (V
2/4)[ T1 sinc2(πfT1) + δ(f) ] // unipolar NRZ , f, p=1/2 (36.3b)
This last result agrees with Xiong (2.25).
Plot: Ignoring the overall factor (V2/ω1) we make this plot of < P(ω)> given by (36.3):
Fig 36.4
The red curve should be scaled by the red factor show n on the left, and the blue delta line should be
scaled by the blue factor on the right.
Power Partition: The total power in the continuous part of the spectrum is:
AC power = ∫-∞ ∞ ω1dx<P(xω1)> = p(1-p) V2 ∫-∞ ∞ dx sinc2(πx) = p(1-p) V2 .
The total power in the DC line at ω = 0 is
DC power = ∫-∞ ∞ ω1dx<P(xω1)> = V2 ∫-∞ ∞ dx p2 δ(x) = p2V2
Chapter 6: Power in Pulse Trains
178 Thus we find that
total power = p2 V2 + p(1-p) V2 = pV2 ( 3 6 . 4 )
DC AC
If p = 1/2, then
total power = (1/4) V2 + (1/4) V2 = (1/2)V2
DC AC
so half the power is in the DC line and half in the AC signal. The DC term is certainly reasonable since
we know that with p = 1/2, the average voltage is (V/2). If one wanted to reduce wasted power, it would be goo d to give this signal a DC offset of -V/2 and
then there would be no DC line. This is in fact the next example if one takes V → V/2.
(b) Bipolar NRZ line code
Pulse Shape
. The pulse shape is the same as for Unipolar NRZ
Fig 36.2
Ppulse (ω) = (1/2π) V2T1 sinc2(ωT1/2) = (V2/ω1) sinc2(ωT1/2) same as for unipolar NRZ (36.1)
Coding : 1 is coded as a positive box with amplitude V, and a 0 as a negative box having amplitude -V.
Fig 36.5
Coefficients α and β: From (35.17) we have
α = [pp] AA + [p(1-p)] AB + [(1-p)p]BA + [(1-p)(1-p)]BB
= [pp] (1)(1) + [p(1-p)] (1)(-1) + [(1-p)p] (-1)(1) + [(1-p)(1-p)] (-1)(-1)
= p
2 - 2p(1-p) + (1-p)2 = 4p2 - 4p + 1 = (1-2p)2 // = μ2
β = [p]AA + [(1-p)] BB = [p] (1)(1) + [(1-p)] (-1)(-1) = p + 1 - p = 1
Chapter 6: Power in Pulse Trains
179 (β - α) = 1 - [4p(p-1)+1] = 4p(1-p) // = σ2 (36.5)
Spectrum: The average spectral power density for the NRZ line code is :
< P(xω1)> = Ppulse (ω) [ (β-α) + α∑
m = -∞∞
δ(x - m) ] (35.11)
= (V2/ω1) sinc2(πx) [4p(1-p) + (1-2p)2∑
m = -∞∞
δ(x - m) ] .
As before the sinc function kills all the δ peaks except the DC peak, so in fact
< P(ω)> = (V2/ω1) sinc2(πx) [ 4p(1-p) + (1-2p)2δ(x) ] // bipolar NRZ (36.6)
<P(f)> = (V2T1) sinc2(πfT1) [ 4p(1-p) + (1-2p)2δ(f)/T1 ] // bipolar NRZ, f (36.6a)
< P(ω)> = (V2/ω1) sinc2(πx) = Ppulse (ω) // bipolar NRZ, p=1/2 (36.6b)
<P(f)> = (V2T1) sinc2(πfT1) // bipolar NRZ, f, p = 1/2 (36.6c)
This last result agrees with Xiong (2.20). The spectrum is all continuous when p = 1/2 since then the DC portion is killed off. As noted in the comment afte r (35.11a), a discrete spectrum cannot exist if the
waveform amplitudes have zero mean μ = 0.
Notice that for p = 1/2, bipolar NRZ has < P(ω)> = P
pulse (ω), so the statistical pulse train spectrum
is the same as that of the underlying pulse.
Plot:
Ignoring the overall factor (V2/ω1) we make this plot of < P(xω1)> given by (36.6)
Fig 36.6
which is the same as the spectrum for unipolar NRZ except for the two scaling factors.
Chapter 6: Power in Pulse Trains
180
Power Partition: We can again compute the DC and AC power.
AC power = unipolar NRZ with p(1-p) → 4p(1-p), so AC = 4p(1-p) V2
DC power = unipolar NRZ with p2 → (2p-1)2, so DC = (2p-1)2 V2
total power = (2p-1)2V2 + 4p(1-p)V2 = V2 , independent of p. (36.7)
DC AC
The total power is independent of p because a pulse has the same AC power if it goes up or down. For p = 1/2 we get
total power = 0 + V
2 = V2 // p = 1/2
DC AC
and now no power is wasted pushing DC through a line. If we take V →V/2 to have a comparable peak-to-
peak amplitude, we find
AC power = (V/2)
2
which is the same as the AC power in (36.5); it is not affected by a DC offset of -V/2.
(c) Unipolar RZ line code
Pulse Shape
. Here the basic pulse is a box that fills only half the time interval T 1.
Fig 36.7
We can use result (36.1) with T 1→ T1/2 since the pulse only fills half the T 1 period,
X
pulse (ω) = (VT1/2) sinc(ωT1/ 4 )
Ppulse (ω) = |Xpulse (ω)|2
2πT1 = (VT 1/2)2 sinc2(ωT1/4)/(2πT1) = (1/2π) (V/2)2T1 sinc2(ωT1/4)
= ( V / 2 )2 (1/ω1) sinc2( π
2 ω
ω1 ) ( 3 6 . 8 )
Coding : 1 is coded as the presence of the pulse, 0 is coded as the absence of a pulse.
Chapter 6: Power in Pulse Trains
181
Fig 36.8
Coefficients α and β: Looking at summary box (35.17), since A = 1 and B = 0 we have α = p2 and β = p.
Spectrum: The average spectral power density for the unipolar RZ line code is :
< P(ω)> = Ppulse (ω) [(β-α) + α ∑
m = -∞∞
2πδ(ωT1- 2πm) ]
= (V/2)2 (1/ω1) sinc2( π
2 ω
ω1 ) [ p(1-p) + p2∑
m = -∞∞
δ(ω
ω1 - m) ]
= (V/2)2 (1/ω1) sinc2( π2 x) [ p(1-p) + p2∑
m = -∞∞
δ(x - m) ] .
We see here that the sinc function now kills off all m= even lines for m ≠0, so with x = ω /ω1,
< P(ω)> = (V/2)2 (1/ω1) sinc2( π
2 x) [ p(1-p) + p2 {δ(x) + ∑
m odd
δ(x - m)} ] (36.9)
<P(f)> = (V/2)2T1 sinc2( π
2 fT1) [ p(1-p) + (p2/T1) {δ(f) + ∑
m odd
δ(f - m/T1) } ] (36.9a)
where the sum includes positive and negative odd values of m. As noted above, one can formally include
the even m terms but they vanish since sinc(m π/2) = 0 for even m ≠ 0. For p = 1/2 one gets
<P(f)> = (V/4)2T1 sinc2( π
2 fT1) [ 1 + (1/T 1) {δ(f) + ∑
m odd
δ(f - m/T1) } ]
= (V/4)2T1 sinc2( π
2 fT1) [ 1 + (1/T 1) ∑
m = -∞ ∞
δ(f - m/T1) ] // p = 1/2 (36.9b)
This last result agrees with Xiong (2.31) in which R b ≡ 1/T = our 1/T 1.
Plot:
Ignoring now the overall factor (V/2)2 (1/ω1) we get this power spectrum from (36.9),
Chapter 6: Power in Pulse Trains
182
Fig 36.9
We now have three pieces: a continuous part, the DC line, and a set of lines at odd m. The main peak is
twice as wide as the NRZ peak since the underlying pulse is half as wide.
Power Partition: Once again, we can compute the tota l power for each of these three pieces.
odd lines power = ∫-∞ ∞ ω1dx<P(xω1)> = ω1 (V/2)2 (1/ω1) p2 ∫-∞ ∞ dx sinc2( π
2 x) 2 ∑
m = 1 3 5...∞
δ(x - m)
= (V/2)2 2p2 ∑
m = 1 3 5...∞
sinc2( π
2 m) = (V/2)2 2p2∑
m = 1 3 5...∞
sin2( π2 m)
( π
2 m)2 = (V/2)2 2p2 (2/π)2 ∑
m = 1 3 5...∞
1
m2
= (V/2)2 2p2 (2/π)2 (π2/8) = p2(V/2)2
where the sum Σodd(1/m2) = π2/8 from GR 0.234.2. Then
DC power = ∫-∞ ∞ ω1dx<P(xω1)> = (V/2)2p2 ∫-∞ ∞ dx sinc2( π
2 x) δ(x) = p2 (V/2)2
which is the same as the odd lines power. Finally,
continuum power = ∫-∞ ∞ ω1dx<P(xω1)> = (V/2)2 p(1-p) ∫-∞ ∞ dx sinc2( π
2 x) = 2p(1-p) (V/2)2
So the power partitioning is
total power = p
2(V/2)2 + p2 (V/2)2 + 2p(1-p) (V/2)2
DC other lines continuum
= p2(V/2)2 + p(2-p) (V/2)2 = 2p(V/2)2 = (p/2)V2 . (36.10)
DC AC
Chapter 6: Power in Pulse Trains
183 This is half of the total power of unipolar NRZ ( 36.4), which seems reasonable since the pulses here are
half as long.
Exercise for the Reader:
Compute everything for the bipolar RZ waveform:
For p = 1/2, Xiong (2.30) gives <P(f)> = (V/2)2T1 sinc2( π
2 fT1).
(d) Manchester line code
This line code was develo
ped at the University of Manchester probably in the World War II era. At that
time Tom Kilburn, Alan Turing and others were building the world's first stored-program computer. Pulse Shape:
The pulse shape here is the biphase (biphasic, diphase) pulse,
Fig 36.10
We already computed X pulse (ω) for this pulse in (19.2), so we now set τ = T1/2 and A/2 = V to get
X
pulse (ω) = (4iV/ ω) sin2(ωT1/4) = (4iV/ω ) sin(ωT1/4) [sin(ω T1/4) / (ωT1/4 )] (ωT1/4 )
= ( i V T
1) sin(ωT1/4) sinc(ωT1/4)
Ppulse (ω) = |Xpulse (ω)|2
2πT1 = (1/2π) V2 T1 sin2(ωT1/4) sinc2(ωT1/4) . (36.11)
This spectral pulse density is 4 sin2(ωT1/4) times that of the RZ pulse shown in (36.8). This extra factor
kills off the spectrum near ω = 0.
Coding: 1 is coded as the above pulse, 0 is coded as the negative of the pulse (but some sources use the
opposite polarity),
Chapter 6: Power in Pulse Trains
184
Fig 36.11
Coefficients α and β: From box (35.17) with A = 1 and B = -1,
α = [pp] AA + [p(1-p)] AB + [(1-p)p]BA + [(1-p)(1-p)]BB
= [pp] (1)(1) + [p(1-p)] (1)(-1 ) + [(1-p)p] (-1)(1) + [(1-p)(1-p)] (-1)(-1)
= p
2 - 2p(1-p) + (1-p)(1-p) = (2p-1)2
β = [p]AA + [(1-p)] BB = [p] (-1)(-1) + [(1-p)] (1)(1) = 1
(β-α) = 1 - (2p-1)2 = 4p(1-p) (36.12)
Comment : Notice that the mean value of the waveform in Fig 36.11 is 0 regardless of p, whereas the
mean value μ of the amplitudes y n is given by μ2 = α = (2p-1)2 as in (D.31).
Spectrum:
The average spectral power density for the Manchester line code is :
< P(ω)> = Ppulse (ω) [(β-α) + α ∑
m = -∞∞
2πδ(ωT1- 2πm) ]
= ( 1 / 2 π) V2 T1 sin2(ωT1/4) sinc2(ωT1/4) [4p(1-p) + (2p-1)2∑
m = -∞∞
2πδ(ωT1- 2πm)
= V2 (1/ω1) T1 sin2( π
2 ω
ω1 ) sinc2( π2 ω
ω1 ) [4p(1-p) + (2p-1)2∑
m = -∞∞
δ( ω
ω1 - m) ]
= V2 (1/ω1) sin2( π2 x) sinc2( π2 x) [4p(1-p) + (2p-1)2∑
m = -∞∞
δ(x - m) ]
so that
< P(ω)>= V2 (1/ω1) sin2( π
2 x) sinc2( π
2 x) [4p(1-p) + (2p-1)2∑
m = ±odd
δ(x - m) ] (36.13)
<P(f)> = V2T1 sin2( π
2 fT1) sinc2( π2 fT1) [4p(1-p) + (2p-1)2(1/T1)∑
m = ±odd
δ(f - m/T1 ] (36.13a)
<P(f)> = V2T1 sin2( π
2 fT1) sinc2( π2 fT1) // p = 1/2 (36.13b)
For p≠1/2, the even lines are killed off by the sin2 factor including the DC line m = 0. For p = 1/2 the
spectrum is fully continuous. The l ast result agrees with Xiong (2.38). Xiong refers to this Manchester
code as Bi-Φ -L.
Chapter 6: Power in Pulse Trains
185 Plot: Ignoring the leading factor V2 (1/ω1) the spectrum for (36.13) has this plot,
Fig 36.12
Power Partition :
lines power = ∫-∞ ∞ ω1dx<P(xω1)> = V2 (2p-1)2 ∫-∞ ∞ dx sin2( π
2 x) sinc2( π2 x) ∑
m = ±odd
δ(x - m)
= V2 (2p-1)2∑
m = ±odd
sin2( π
2 m) sinc2( π2 m) = V2 (2p-1)2∑
m = ±odd
sin4( π
2 m) ( π2 m)-2
= V2 (2p-1)2(2/π)2∑
m = ±odd
1 /m2 = V2 (2p-1)2(2/π)2 2∑
m = 1 3 5...∞
1 /m2
= V2 (2p-1)2(2/π)2 2 (π2/8) = (2p-1)2 V2
continuum power = ∫-∞ ∞ ω1dx<P(xω1)> = V2 4p(1-p) ∫-∞ ∞ dx sin2( π
2 x) sinc2( π
2 x) = V2 4p(1-p)
since the integral is just 1. Therefore,
total power = 0 + (2p-1)
2 V2 + 4p(1-p) V2 = V2 (36.14)
DC lines continuum AC
In the case p = 1/2, the lines power vanishes leaving only continuum power = V
2.
Since the power is kept away from DC, Manchester coding is useful for AC-coupled transmission lines,
such as lines incorporating transformers. The down side compared to NRZ is that the first spectral hump
goes out to ω = 2ω1, which reflects the fact that the minimum pulse width is T 1/2 whereas in NRZ it is
T1. So a transmission line must then have twice th e bandwidth for Manchester relative to NRZ.
(e) Noise, ISI and Eye Patterns
In general, if
some spectral components are filtered aw ay in a transmission line (or in some general signal
pathway), the corresponding pulse (by inverse Fourier Tr ansform) has curved corn ers, meaning the pulse
Chapter 6: Power in Pulse Trains
186 gets rounded and spread out. This e ffect along with noise can result in inter-symbol interference (ISI) .
The superposition of such pulses on an o scilloscope (triggered on a recovered T 1 clock) for a random
pulse train is called an eye pattern. This pattern must have a central clear area to allow the two (or more
for some line codes) pulse levels to be distinguished by a receiving circuit. Here is a marginal eye pattern
for NRZ on the left, and a better one for AMI on the right (see Section 37).
Fig 36.13
37. The AMI Line Code
Pulse Shape
. The pulse shape is the same as for unipolar NRZ ,
Fig 36.2
Xpulse (ω) = (VT 1) sinc(ω T1/2)
(36.1)
Ppulse (ω) = (1/2π) V2T1 sinc2(ωT1/2)
However, we shall do the analysis below for a general x
pulse (t) and insert the box shape at the end.
Coding
: Alternate Mark Inversion (AMI) means that a 0 is encoded as a zero (for duration T 1) and a 1 is
encoded as a pulse (of duration T 1) of either plus or minus polarity. As each 1 is encountered in the data,
the pulse polarity is the negative of that used for the previous encoded 1 pulse, so the 1 polarities are
alternated, as in this example
Fig 37.1
Chapter 6: Power in Pulse Trains
187 Coefficients αm,n and β: 0 is coded with amplitude B = 0, but a 1 is coded with either A = +1 or A = -1,
so we have A = ± 1, B = 0. Because there is now correlation between different locations m and n in the
pulse train, we can no longer use the simple results of box (35.17). We have to back up to an earlier point
in the development. Our starting point will be equation (35.2) which we repeat here, averaging over the
statistical ensemble (y n real),
< P(ω)> = T1 Ppulse (ω) (1/T) ∑
n = -∞∞
∑
m = -∞∞
αn,m eiω(m-n)T1 . (35.2)
where αn,m ≡ <ym yn> and β ≡ <yn2> = αn,n
Breaking the double sum into two terms as done in Secti on 35, we obtain this new version of (35.7), valid
when there is correlation between pulse train positions m and n,
< P(ω)> = T
1 Ppulse (ω) (1/T) { ∑
n = -∞∞
∑
m ≠ n
αn,m [ eiω(m-n)T1] + β∑
n = -∞∞
∑
m = n
[1] } . (37.1)
We develop things as usual for a general pulse shape, but in the end we will use a square pulse as shown
in the figure above. The double sum in (35.2) is < | Y"(z) |2> where Y"(z) is the Z Transform of y n from
box (24.37). Consider now the expressions given in (35.6) for the statistical averages <y
n2> and <y myn>. We
assume that p is the probability of a 1 being coded, so 1-p is the probability of a 0 being coded. Then we
have
<yn2> = [p]AA + [(1-p)] BB = [p]AA = [p] (±1) (±1) = p (37.2)
That was the easy one.
For m ≠ n, we have a much harder problem. Consider
<y
myn> = [pp] AA' + [p(1-p)] AB' + [(1-p)p]BA' + [(1-p)(1-p)]BB'
= [pp] σ σ' + [p(1-p)] σ 0 + [(1-p)p] 0 σ' + [(1-p)(1-p)] 0 0
= p2 σ σ'
where σ = ±1 and σ ' = ±1. Here p
2 is the probability that both slot positions y m and yn are coded for 1.
This can happen in four differe nt ways, as illustrated here,
Chapter 6: Power in Pulse Trains
188
Fig 37.2
By symmetry, the probability of cases 1 and 2 is th e same, and the probability of cases 3 and 4 is the
same. This is perhaps not totally obvious, but the reason will become clear below when we talk about
legal pulse patterns. Given that both m and n are coded for 1, let X/2 be the total probability fo r the case 1, and Y/2 be the total
for the case 3. Then we can write
<y
myn> = (+1)(+1) p2X/2 + (-1)(-1) p2X/2 + (+1)(-1) p2Y/2 + (-1)(+1) p2Y/2
= p2(X-Y) .
Given that both m and n are coded for a 1, since we have enumerated all the cases, we must have
X + Y = 1 // probability of getting any of the four cases. Our task then is to compute probabilities X and Y. For cases 1 and 2 taken together, X is the probability th at the gap between the coded 1's is filled with a
legal sequence of pulses. This is the key statemen t and the reader may want to ponder the previous
sentence thinking about probability as the number of legal ways divided by the total number of ways. Only the legal ways can show up in a statistical ensemble. If the gap is "legal", there must be an odd number of coded 1's in the gap, due to the AMI alternation
coding rule. Similarly, Y is the probability that there are an even number of coded 1's in the gap. Define,
k = |m-n| - 1 = size of gap
and think of X and Y as depending on k, so we write X
k and Yk.
Chapter 6: Power in Pulse Trains
189
Note that Y = Y k = (1-Xk) = probability that gap has even number of coded 1's. So far, we add k labels to
our results shown above,
<ymyn> = p2(Xk-Yk) = p2 (2Xk - 1) k = |m-n| - 1 . (37.3)
Assume we have a gap of size k and there exists some X
k and Yk we don't yet know. What can be said
about X and Y if the gap is increased to size k+1 by adding one more pulse period in between? Claim:
X
k+1 = Yk p + Xk (1-p) = probability of having an odd number of coded 1's in gap k+1
Explanation:
• Y
k is the probability the k gap had an even number of coded 1's. In order to make the k+1 gap have an
odd number of coded 1's we have to put a coded 1 in the new space, which has probability p.
This gives the first term Y k p .
• X
k is the probability the k gap had an odd number of coded 1's. In order to make the k+1 gap have an
odd number of coded 1's we have to put a coded 0 in the new space, which has probability (p-1).
This gives the second term X k (1-p) .
Since this exhausts the ways we can get from k to k+1, X
k+1 has the probability shown above. We could
write a similar expression for Y k+1 but it is not needed. Since Y k = 1-Xk we then have
Xk+1 = (1-Xk) p + Xk (1-p) = p - pX k + Xk- pXk = (1-2p)X k + p . (37.4)
Now define, a ≡ (1-2p) => p = (1-a)/2 and 1-p = (1+a)/2
Then the above reads,
X
k+1 = aXk + p . ( 3 7 . 5 )
This is a difference equation (recurrence re lation) which we want to solve for X k. If there is no gap at all
(k=0), we have two adjacent identical pulses which is illegal so X 0 = 0. If the gap is k = 1, then the
middle element must be different from the two ends, so X 1 = p, consistent with (37.5). We now examine
the recurrence relations:
X
0 = 0
X1 = p
X2 = a(p) + p = p(a+1)
X3 = a[p(a+1]+ p = p(a2+a +1)
....
Chapter 6: Power in Pulse Trains
190 Xk = p (ak-1 + ..... + a2 + a + 1)
The geometric series can be summed in the usual manner and yields
X
k = p (1 - ak)/(1-a) = p (1 - ak)/2p = (1 - ak)/2 . (37.6)
The same result can be obtained from Maple in this manner :
Inserting this result into (37.3) gives
<y
myn> = p2 (2Xk - 1) = p2 (2[(1 - ak)/2] - 1) = p2 ( (1 - ak) - 1) = - p2ak
= - p
2 a[|m-n| - 1] = (-p2/a) a|m-n| . (37.7)
Therefore, we have our final results for the coefficients
α
m,n ≡ <ymyn> = (-p2/a) a|m-n| m ≠ n // a ≡ (1-2p)
β ≡ <yn2> = p . ( 3 7 . 8 )
Spectrum. The starting point is (37.1) which we repeat here,
< P(ω)> = T1 Ppulse (ω) (1/T) { ∑
n = -∞∞
∑
m ≠ n
αn,m [ eiω(m-n)T1] + β∑
n = -∞∞
∑
m = n
[1] } . (37.1)
The second double sum is the same as in the uncorrelated case,
∑
n = -∞∞
∑
m = n
[1] = ∑
n = -∞∞
∑
m = -∞∞
δm,n [1] = ∑
n = -∞∞
[1]
which we set later to [2 πδ(0)] or (2N+1) depending on whether we ha ve an infinite or finite pulse train.
The first double sum is different and we write it using (37.8),
∑
n = -∞∞
∑
m ≠ n
αn,m [ eiω(m-n)T1] = ∑
n = -∞∞
∑
m ≠ n
(-p2/a) a|m-n| b(m-n) b ≡ eiωT1 .
Ignoring for the moment the (-p2/a) factor, we painfully process the double sum in many steps,
Chapter 6: Power in Pulse Trains
191 = ∑
n = -∞∞
∑
m ≠ n
a|m-n| b(m-n) ( 3 7 . 9 )
= ∑
n = -∞∞
[∑
m = -∞n-1
(b/a)m-n +∑
m = n+1∞
(ab)m-n ] s = m-n+1 r = m-n-1
= ∑
n = -∞∞
[∑
s = -∞0
(b/a)s-1 + ∑
r = 0∞
(ab)r+1 ] = ∑
n = -∞∞
[∑
s = 0∞
(b/a)–s-1 + ∑
r = 0∞
(ab)r+1 ]
= ∑
n = -∞∞
[∑
s = 0∞
(a/b)s+1 + ∑
r = 0∞
(ab)r+1 ] = [(a/b) ∑
s = 0∞
(a/b)s + (ab) ∑
r = 0∞
(ab)r ] ∑
n = -∞∞
[1]
= [ (a/b) 1
1-a/b + (ab) 1
1-ab ] ∑
n = -∞∞
[1] = [ a
b-a + ab
1-ab ] ∑
n = -∞∞
[1] = a(1+b2-2ab)
(b-a)(1-ab) ∑
n = -∞∞
[1] .
We note that this double sum is valid for |a| < 1 sin ce |b| = 1. In the complex a-plane, the circle of
convergence is |a| = 1. The sum is valid in the limit a →-1, but as a →+1, if b = 1 (ω = 0 or n2 π), a branch
point at a = 1 is encountered and the sum is invalid . For example, if the result above is evaluated for ω = 0
and a = 1, the final ratio shown is -1, which s uggests that adding positive quantities gives a negative
result! The correct way to do all this work is with finite N and then limit N →∞, and this will be done in
more detail for the Change/Hold line code treated in the next section.
We let Maple reduce the ratio where we use b = e
iα = eiωT1 (ignore right side below for now)
Thus we conclude that
∑
n = -∞∞
∑
m ≠ n
αn,m [ eiω(m-n)T1] = (-p2/a) 2a[cos(ωT1)-a]
1+a2-2acos(ωT1) ∑
n = -∞∞
[1] .
The curly bracket in (37.9) is then, since β = p,
{ ... } = [ -p2 2 [cos(ωT1)-a]
1+a2-2acos(ωT1) + p] ∑
n = -∞∞
[1] .
The square bracket is evaluated by Maple as shown on the right above, so using p = (1-a)/2 we get
Chapter 6: Power in Pulse Trains
192 { ... } = (1/2)(1-a2) [1 - cos(ω T1)]
1+a2-2acos(ωT1) ∑
n = -∞∞
[1] = (1-a2) [sin2(ωT1/2)]
1+a2-2acos(ωT1) ∑
n = -∞∞
[1]
where one could write (1-a2) = (1-a)(1+a) = 4p(1-p) in the numerator. So (37.9) now reads
< P(ω)> = Ppulse (ω) (T1/T) (1-a2) [sin2(ωT1/2)]
1+a2-2acos(ωT1) ∑
n = -∞∞
[1] a ≡ (1-2p) .
The sum is either 2 πδ(0) for the infinite series, or (2N+1) for th e finite series. In either case we write the
sum as T/T 1 as in box (34.4), and the final result for our AMI line code is
< P(ω) > = Ppulse (ω) (1-a2) [sin2(ωT1/2)]
1+a2-2acos(ωT1) a = (1-2p) (1-a2) = 4p(1-p) . (37.10)
Using x = ω/ω1 = fT1, this can be written in these alternate forms (the last form uses (33.24)),
< P(ω) > = Ppulse (ω) (1-a2) [sin2(πx)]
1+a2-2acos(2πx) a = (1-2p) (1-a2) = 4p(1-p) (37.11)
<P(f) > = 2π Ppulse (f) (1-a2) [sin2(πx)]
1+a2-2acos(2πx) a = (1-2p) (1-a2) = 4p(1-p) (37.11a)
<P(f) > = 4p(1-p) | Xpulse (f)|2 (1/T1) sin2(πfT1)
1+(1-2p)2-2(1-2p)cos(2 πfT1) (37.11b)
where we recall from the text after (1.4) that X( ω) = X(f). This last result agrees with Bennet and Davey
(19-123) but they have a leading factor 8 instead of 4. Perhaps this is because they regard the frequency
range for f as (0, ∞) instead of (-∞ ,∞) so the left part of the spectrum is folded over to the right side giving
them an extra factor of 2.
Note that the AMI spectrum is comple tely continuous, there is no discrete part at all. The discrete part
previously arose from the Σ
nΣm≠n double summation as in (35.9), but here we see no such discrete
spectrum generated. It was dispersed into the cont inuum by the correlation e ffect between legal bit
patterns. Setting p = 1/2 gives a = 0 and therefore
< P(ω) > = P
pulse (ω) sin2(πx) p = 1/2 . (37.12)
Selecting a box of height V and width T 1 we have from (36.1)
Ppulse (ω) = |Xpulse (ω)|2
2πT1 = (T1V)2 sinc2(πx)
2πT1 = V2 (T1/2π) sinc2(πx)
Chapter 6: Power in Pulse Trains
193
so that
< P(ω) > = V
2(1/ω1) sinc2(πx) sin2(πx) p = 1/2 x = ω /ω1 (37.13)
<P(f) > = V
2T1sinc2(πfT1) sin2(πfT1) p = 1/2 x = fT 1 (37.13a)
This last result agrees with Xiong (2.34) where the code is called AMI-NRZ.
Plot: Ignoring now the overall factor (V2/ω1) [ or V2T1] we get this AMI p = 1/2 power spectrum
Fig 37.3
This shape is the same as the continuous part of the Manchester spectrum, but the first zero is at 1 instead
of 2 since the pulse AMI pulse is twice as wide as the Manchester pulse. The AMI Limit as p → 0 (a → +1
)
In this limit, we know that our pulse train is just the constant value 0 so < P(ω) > = 0. As noted earlier,
our formula is invalid in this limit at ω = 0 or 2π n. Still, it is interesting to see what it says:
< P(ω) > = Ppulse (ω) (1-a2) [sin2(ωT1/2)]
1+a2-2acos(ωT1) = Ppulse (ω) [sin2(ωT1/2)] (1-a2)
1+a2-2acos(ωT1)
Using this limit from Appendix A
lim
a→+1 (1/π) (1-a2)
1+a2-2acos(2k) = ∑
m = -∞∞
δ(k - mπ) (A.23c)
we find that
Chapter 6: Power in Pulse Trains
194 < P(ω) > = Ppulse (ω) [sin2(ωT1/2)] π ∑
m = -∞∞
δ(ωT1/2 - mπ)
= Ppulse (ω) π ∑
m = -∞∞
[sin2(mπ)] δ(ωT1/2 - mπ) = 0
and our expression for < P(ω) > happens to give the correct answer. The correct expression for < P(ω) > in
this limit of a = +1 (which includes delta spikes including at ω = 0) gives this same answer due to the sin2
factor. This subject comes up again with the Change/Hold line code in Section 38.
Chapter 6: Power in Pulse Trains
195 The AMI Limit as p → 1 (a → -1 )
First of all, we can see that in this limit the AMI waveform has alternating-sign pulses. With V = 1, this
waveform matches that shown in (34.21),
P(ω) = P
pulse (ω) ∑
n = ±odd
δ(x - m/2) x = ω/ω1 . (34.21)
Somehow in this limit, the all-continuous AMI spec trum becomes all-discrete! How exactly does this
happen? Consider again our continuous AMI result,
< P(ω) > = P
pulse (ω) (1-a2) [sin2(πx)]
1+a2-2acos(2πx) a = (1-2p) . (37.14)
It seems possible that this becomes discret e because when a = -1, (1-a2) = 0 and < P(ω) > = 0 except
possibly at singular points where the denominator vanishes. In Appendix A it is shown that
lima→-1 δ8(k, a) = lim a→-1 (1/π) (1-a2) [sin2(k)]
1+a2–2acos(2k) = ∑
m = ±odd
δ(k-mπ/2) . (A.25a)
Therefore we may write
< P(ω) > = P
pulse (ω) π δ8(πx,a)
→ Ppulse (ω) π∑
m = ±odd
δ(πx-mπ/2) = Ppulse (ω)∑
m = ±odd
δ(x-m/2)
and this agrees with our expected result shown just above.
The coefficient averaging done in this Section for the AMI line code spectrum is based on the excellent
discussion of Bennett and Davey p338-341.
Chapter 6: Power in Pulse Trains
196 38. The Change/Hold Line Code
There are surely
more efficient ways to compute the results given below, but the method shown provides
a good exercise in directly calculating <y myn>. The method is very similar to that used for the AMI line
code in Section 37. At the end, we a pply the results to the NRZI line code.
Pulse Shape
:
The pulse shape is the same as for unipolar NRZ,
Fig 36.2
Xpulse (ω) = (VT 1) sinc(ω T1/2)
(36.1)
Ppulse (ω) = V2(1/ω1) sinc2(ωT1/2) .
In place of amplitude V we will have A and B as described below. Coding
:
The coding uses two amplitudes A and B. Hold or Change coding means that a 0 is encoded as no change in the pulse amplitude (it remains what it was), while a 1 is encoded as a change A ↔B. Here is an
example starting with an A pulse:
data = [ 1 0 1 1 0 0 1]
encode = [ A B B A B B B A]
Fig 38.1
Again we shall assume an arbitrary pulse shape and insert the box-pulse at the end.
Coefficients α
m,n and β:
The spectrum has the same form give n in (37.1) for the AMI code,
Chapter 6: Power in Pulse Trains
197 < P(ω)> = T1 Ppulse (ω) (1/T) { ∑
n = -∞∞
∑
m ≠ n
αn,m [ eiω(m-n)T1] + β∑
n = -∞∞
∑
m = n
[1] } . (37.1)
where αn,m ≡ <ym yn> and β ≡ <yn2> = αm,m
and our task is now to compute α
n,m and β for the Hold/Change line code.
First consider
<y
n2> = [q]A2 + [(1-q)] B2 ,
where q is the probability that slot n has y
n= A. In this code, since only change and hold are coded, there
is no preference for either amplitude, so q = 1/2 and
<yn2> = (A2+B2) / 2 . ( 3 8 . 1 )
Turning to <y
nym>, consider this picture similar to that used for the AMI case, where the gap is kT 1units.
Here we arbitrarily drawn A > 0 and B < 0 and we dr aw the pulse as square, but it could be any shape and
A and B can have any signs.
Fig 38.2
Denote the four probabilities as p(AA)
k and so on. Since slot m and slot n must each be filled with either
an A or a B, this picture shows the only four possib ilities, so (different scaling relative to AMI analysis)
p
(AA)
k + p(AB)
k + p(BA)
k + p(BB)
k = 1 .
Chapter 6: Power in Pulse Trains
198 Note that p(AA)
k is the probability of slot m and slot n both having amplitude A in the statistical pulse
train. With these probabilities, we will have
<ymyn> = p(AA)
k AA + p(AB)
k AB + p(BA)
k BA + p(BB)
k BB . (38.2)
In case 1, there are a certain number of holds and chang es during the gap such that the overall effect is a
hold. The number of changes must have been even. Bu t this same statement can be made about case 4, so
cases 1 and 4 have the same probabilit y of existing in the pulse train. Similarly, cases 2 and 3 have the
same probability and in those cases the number of changes must be odd. So now we have two variables to
worry about and they add to 1/2 :
p
(AA)
k + p(AB)
k = 1 / 2 ( 3 8 . 3 )
<y
myn> = p(AA)
k AA + p(AB)
k AB + p(AB)
k BA + p(AA)
k BB
= p
(AA)
k ( AA + BB) + p(AB)
k (AB + BA)
so <y
myn> = p(AA)
k( A2 + B2) + p(AB)
k 2AB . (38.4)
If the gap is zero, what is the probability of having an adjacent AA in the pulse stream? The probability of
having the left A is 1/2, and the probability for an A being followed by a A is 1-p. Therefore
p
(AA)
0 = (1/2)(1-p) . (38.5)
Consider now the gap as shown at value k. We claim that
p
(AA)
k+1 = p(AA)
k (1-p) + p(AB)
k p . ( 3 8 . 6 )
Proof: If it was an AA to start with gap k, then to be AA with gap k+1 we have to add another A which has probability (1-p) since this is a hold. Conversely, if it was an AB we have to add an A which is a change, which has probability p. Then from (38.3),
p
(AA)
k+1 = p(AA)
k (1-p) + (1/2 - p(AA)
k ) p . (38.7)
To simplify notation, let X
k ≡ p(AA)
k so that p(AB)
k = 1/2 - X k . Then (38.4) and (38.7) become
<y
myn> = Xk (A2 + B2) + (1/2 - X k)2AB = (A-B)2Xk + AB (38.8)
X
k+1 = Xk (1-p) + (1/2-X k) p = (1-2p)X k + p/2
= aX k + p / 2 a ≡ 1-2p . (38.9)
Maple solves this recursion equation as follows, using (38.5) that X 0 = (1/2)(1-p),
Chapter 6: Power in Pulse Trains
199
so we find that
Xk = (1 + ak+1)/4 = p(AA)
k . ( 3 8 . 1 0 )
As a check, suppose p = 0 so there can be no changes. Then a = 1 and p
(AA)
k = 1/2. We can now have
only case 1 or case 4, so we know p(AB)
k = 0, and that is consistent with p(AA)
k + p(AB)
k = 1/2 .
Continuing from (38.8),
<ymyn> = (A-B)2Xk + AB = (A-B)2(1 + ak+1)/4 + AB
= [ ( A - B )
2/4] ak+1 + (A-B)2/4 + AB
= [ ( A - B )
2/4] ak+1 + [(A+B)2/4]
= (1/4) [ (A-B)
2 ak+1 + (A+B)2 ] (38.11)
and we note that the result is indeed symmetric under A ↔ B. Again for p = 0 (a=1) we find that
<ymyn> = (A2+B2)/2 which is the same then as <y n2>. For a constant pulse train, the amount of slot
separation makes no difference. Since k = |m-n| - 1 in general, we get these final results,
α
n,m = <ymyn> = (1/4) [ a|m-n| (A-B)2 + (A+B)2 ] a ≡ 1-2p
β = <yn2> = (A2+B2)/2 . (38.12)
To save space, define
c ≡ (A-B)
2/4 d ≡ (A+B)2/ 4 ( 3 8 . 1 3 )
so then
α
n,m = <ymyn> = c a|m-n| + d . (38.14)
For later use, notice that β - d = c . ( 3 8 . 1 5 )
Chapter 6: Power in Pulse Trains
200 Spectrum:
The spectrum is determined by (37.1) q uoted above which we repeat here ,
< P(ω)> = T1 Ppulse (ω) (1/T) { ∑
n = -∞∞
∑
m ≠ n
αn,m [ eiω(m-n)T1] + β∑
n = -∞∞
∑
m = n
[1] } . (37.1)
As in the AMI case, the β term in {...} is just
β∑
n = -∞∞
∑
m = n
[1] = β ∑
n = -∞∞
[1] . (38.16)
Since αn,m = c a|m-n| + d , the first double sum has two terms. Again looking at the AMI case, we find
that the first term is given by, again using b ≡ eiωT1,
c ∑
n = -∞∞
∑
m ≠ n
a|m-n| b(m-n) = c a(1+b2-2ab)
(b-a)(1-ab) ∑
n = -∞∞
[1] . |a| < 1 (38.17)
To get the second term in the first double sum, we cannot just replace c by d and then set a = 1 to get
d a(1+b2-2ab)
(b-a)(1-ab) ∑
n = -∞∞
[1] (with a = 1) = d (1+b2-2b)
(b-1)(1-b) ∑
n = -∞∞
[1] = -d ∑
n = -∞∞
[1] . // wrong
This is because the result is not valid at a = 1 for ω = 2πn. So we have to do this d sum separately:
d ∑
n = -∞∞
∑
m ≠ n
[ eiω(m-n)T1] = d ∑
n = -∞∞
∑
m ≠ n
bm-n .
We now omit the d for a while to evaluate this double sum
∑
n = -∞∞
∑
m ≠ n
bm-n = ∑
n = -∞∞
[ ∑
m = -∞∞
bm-n – ∑
m = n
bm-n ]
= ( ∑
n = -∞∞
b-n ) ( ∑
m = -∞∞
bm ) – ∑
n = -∞∞
[ 1 ] . (38.18)
In the first factor, since b ≡ eiωT1 , we are facing squared de lta functions, so we have to back off to finite
N in our processing and later take N →∞. We continue, doing this making use Appendix A,
= ( 2 πδ5(ωT1,N) )2 - (2N+1) // from (A.30)
Chapter 6: Power in Pulse Trains
201 = (2N+1) [ ( 2 πδ5(ωT1,N) )2/ (2N+1) - 1 ]
= ( 2 N + 1 ) [ ( 2 πδ6(ωT1,N) - 1 ] . // from (A.20)
Then we return to N = ∞ and use
limN→∞ δ6(k,N) = ∑
m = -∞∞
δ(k-2πm ) (A.21)
to get
∑
n = -∞∞
∑
m ≠ n
[ eiω(m-n)T1] = [ ∑
m = -∞∞
2π δ(ωT1-2πm) - 1 ] ∑
n = -∞∞
[ 1 ] . (38.19)
The -1 is what one gets from the limit a→1 of a(1+b2-2ab)
(b-a)(1-ab) , but we see that there is more.
We can now reinstall d and assemble the pieces to get
< P(ω)> = P
pulse (ω) (T1/T) { c a(1+b2-2ab)
(b-a)(1-ab) + β - d + d ∑
m = -∞∞
2π δ(ωT1-2πm) } ∑
n = -∞∞
[ 1 ] .
(38.17) (38.16) (38.19) times d
Recalling (38.15) that β - d = c,
< P(ω)> = P
pulse (ω) (T1/T) { c [ a(1+b2-2ab)
(b-a)(1-ab) +1] + d ∑
m = -∞∞
2π δ(ωT1-2πm) } ∑
n = -∞∞
[ 1 ] .
Using the fact (33.22) that Σn=-∞∞[1] = T/T 1 for both the infinite and finite pulse train cases, we get
< P(ω) > = Ppulse (ω) { c [ a(1+b2-2ab)
(b-a)(1-ab) +1] + d ∑
m = -∞∞
2π δ(ωT1-2πm) } (38.20)
a = (1-2p) b = eiωT1 c = (A-B)2/4 d = (A+B)2/4 .
Maple now computes the square-bracketed (sb) expression:
Chapter 6: Power in Pulse Trains
202
so that
[ a(1+b2-2ab)
(b-a)(1-ab) +1] = (1-a2)
1+a2 -2acos(ωT1)
and then here is the final result for the Change/Hold line code :
< P(ω) > = Ppulse (ω) { [ (A-B)
2 ]2 (1-a2)
1+a2 -2acos(ωT1) + [ (A+B)
2 ]2 ∑
m = -∞∞
2π δ(ωT1-2πm) }
( 3 8 . 2 1 )
where a = (2p-1) and p is the probability of a ch ange while 1-p is the probability of a hold.
We shall now investigate various limits of this result (a confidence-building measure).
Limit A → B
: Setting A = B in (38.21) gives
< P(ω) > = Ppulse (ω) { A2∑
m = -∞∞
2π δ(ωT1-2πm) } . (38.22)
In the case that the pulse is a box of unit height we use (34.22),
Ppulse (ω) = (1/ω1) sinc2(ωT1/ 2 ) ( 3 8 . 2 3 )
to get
<P(ω) > = A2 Ppulse (ω) ∑
m = -∞∞
2π δ(ωT1-2πm)
= A2∑
m = -∞∞
2π δ(ωT1-2πm) (1/ω1) sinc2(πm) = A2 2π δ(ωT1) (1/ω1)
= A2 δ(ω) . ( 3 8 . 2 4 )
This is exactly what we expect when A = B, since the pulse train is then just a constant value A ! Limit p→1
( a → -1) :
In this limit we expect to get a square-wave pulse trai n with alternating values A and B. We make use of
this limit from Appendix A with 2k = ωT
1,
Chapter 6: Power in Pulse Trains
203
lima→-1 (1-a2)
1+a2 -2acos(ωT1) = π∑
m = ±odd
δ(ωT1/2-mπ/2) = ∑
m = ±odd
2π δ(ωT1-mπ) (A.25b)
and then the Change/Hold spectral power density (38.21) becomes
< P(ω) > = P
pulse (ω) { [ (A-B)
2 ]2 ∑
m = ±odd
2π δ(ωT1-mπ) + [ (A+B)
2 ]2 ∑
m = -∞∞
2π δ(ωT1-2πm) } .
Installing from (34.22) the unit-height box pulse shape Ppulse (ω) = (1/ω1) sinc2(ωT1/2) and using
(38.24) the second term becomes just [ (A+B)
2 ]2 δ(ω) while the first term is
[ (A-B)
2 ]2 ∑
m = ±odd
2π δ(ωT1-mπ) (1/ω1) sinc2(mπ/2)
= [ (A-B)
2 ]2 2π ∑
m = ±odd
δ(ωT1-mπ) (1/ω1) (mπ/2)-2
= (A-B)2 (1/π2) ∑
m = ±odd
(1/m2) δ(ω - mω1/2)
giving a final result,
< P(ω) > = [ (A-B)
π ]2 ∑
m = ±odd
(1/m2) δ(ω - mω1/2) + [ (A+B)
2 ]2 δ(ω) . (38.25)
Comparing the first term with (34.23), we see that it is the spectrum of a square wave whose peak-to-peak
amplitude is (A-B), which is exactly what it shoul d be. The second term then correctly accounts for the
expected average DC level of (A+B)/2. Limit p→0
( a → +1) :
In this case for a square wave we expect to get a result appropriate for an ensemble of pulse trains half of
which have constant value A and the other have constant value B, since all pulse trains are in a permanent
hold state with p = 0; nothing changes. This time we use this limit (A.23c) with 2k = ωT
1,
lima→+1 (1-a2)
1+a2-2acos(ωT1) = π ∑
m = -∞∞
δ(ωT1/2 - mπ) = ∑
m = -∞∞
2πδ(ωT1 - m2π) (A.23c)
to get from (38.21),
< P(ω) > = P
pulse (ω) { [ (A-B)
2 ]2 ∑
m = -∞∞
2π δ(ωT1 - 2mπ) + [ (A+B)
2 ]2 ∑
m = -∞∞
2π δ(ωT1-2πm) }
Chapter 6: Power in Pulse Trains
204 = A2+B2
2 Ppulse (ω)∑
m = -∞∞
2π δ(ωT1 - 2mπ) . (38.26)
Ignoring the leading factor, this agrees with the first line of (33.25) which was developed for a simple
pulse train with unit amplitudes y n = 1. This result then describes the average spectral power of an
ensemble in which 50% of the pulse trains ha ve amplitude A and the rest amplitude B.
Installing the square pulse spectrum (38.23) P pulse (ω) = (1/ω1) sinc2(ωT1/2) gives
< P(ω) > = A2+B2
2 (1/ω1) sinc2(ωT1/2) ∑
m = -∞∞
2π δ(ωT1 - 2mπ)
= A2+B2
2 (1/ω1) 2π δ(ωT1) = A2+B2
2 (T1ω1)-1 2π δ(ω)
= A2+B2
2 δ(ω)
which describes an ensemble of constant pulse trains 50% of which are x(t) = A and the rest x(t) = B.
Limit p→1/2
( a → 0)
Recall again the general result (38.21),
< P(ω) > = P
pulse (ω) { [ (A-B)
2 ]2 (1-a2)
1+a2 -2acos(ωT1) + [ (A+B)
2 ]2 ∑
m = -∞∞
2π δ(ωT1-2πm) } .
(38.21)
The ratio becomes unity so the spectral power density is then
< P(ω) > = Ppulse (ω) { [ (A-B)
2 ]2 + [ (A+B)
2 ]2 ∑
m = -∞∞
2π δ(ωT1-2πm) } . (38.27)
Box-Shaped Pulse for general p: Start as just above with (38.21) and insert Ppulse (ω) for the box,
< P(ω) > = Ppulse (ω) { [ (A-B)
2 ]2 (1-a2)
1+a2 -2acos(ωT1) + [ (A+B)
2 ]2 ∑
m = -∞∞
2π δ(ωT1-2πm) }
Ppulse (ω) = (1/ω1) sinc2(ωT1/2) .
As usual, the second term becomes [ (A+B)
2 ]2 δ(ω) , so the result is [ a = 1-2p, x = ω/ω1 = fT1 ]
< P(ω) > = [ (A-B)
2 ]2 (1/ω1) sinc2(ωT1/2) (1-a2)
1+a2 -2acos(ωT1) + (1/ω 1)[ (A+B)
2 ]2 δ(ω/ω1) (38.28)
Chapter 6: Power in Pulse Trains
205 <P(f) > = [ (A-B)
2 ]2 T1 sinc2(πfT1) (1-a2)
1+a2 -2acos(2πfT1) + [ (A+B)
2 ]2 δ(f) . (38.28a)
Box-Shaped Pulse for p = 1/2 (a = 0):
< P(ω) > = [ (A-B)
2 ]2 (1/ω1) sinc2(ωT1/2) + [ (A+B)
2 ]2 δ(ω)
( 3 8 . 2 9 )
= (1/ ω1) { [ (A-B)
2 ]2 sinc2(πx) + [ (A+B)
2 ]2 δ(x) } x ≡ ω
ω1 = f T1
<P(f) > = T 1 [ (A-B)
2 ]2 sinc2(πfT1) + [ (A+B)
2 ]2 δ(f) . (38.29a)
Ignoring the factor (1/ω 1) in (38.29), we make this plot of < P(ω) >, which is the same as for unipolar
NRZ but with different scaling factors for the two terms,
Fig 38.3
Chapter 6: Power in Pulse Trains
206 Example 1: Unipolar NRZI line code
Coding: This is a special case of Change/Hold encoding where A = 1 and B = 0.
data = [ 1 0 1 1 0 0 1]
encode = [ 1 0 0 1 0 0 0 1]
Fig 38.4
NRZI means NRZ Invert-on-1, where NRZ means non -return to zero (see comments at the start of
Section 36). NRZI does not mean "NRZ inverted". Some other sources use A = 0 and B = 1 so then transitions happen on 0 instead of 1, as in the standa rd for USB (Universal Serial Bus). The Change/Hold
spectra are symmetric in A↔B, so the NRZI spectra are the same for either convention.
Coefficients α
m,n and β:
αn,m = <ymyn> = (1/4) [ a|m-n| (A-B)2 + (A+B)2 ] = (1/4) [ a|m-n| + 1 ]
β = <y
n2> = (A2+B2)/2 = 1/2 (38.30)
Spectrum: From (38.21), and with a = (1-2p),
< P(ω) > = Ppulse (ω) { [ (A-B)
2 ]2 (1-a2)
1+a2 -2acos(ωT1) + [ (A+B)
2 ]2 ∑
m = -∞∞
2π δ(ωT1-2πm) }
= Ppulse (ω) { 1
4 (1-a2)
1+a2 -2acos(ωT1) + 1
4 ∑
m = -∞∞
2π δ(ωT1-2πm) } . (38.31)
Box-Shaped Pulse for general p: From (38.28),
< P(ω) > = 1
4 (1/ω1) sinc2(ωT1/2) (1-a2)
1+a2 -2acos(ωT1) + 14 δ(ω) . (38.32)
Box-Shaped Pulse for p = 1/2 (a = 0): From (38.29),
< P(ω) > = (1/ ω1) { 1
4 sinc2(πx) + 1
4 δ(x) } x ≡ ω
ω1 = fT1 (38.33)
<P(f) > = T 1 1
4 sinc2(πx) + 14 δ(f) . (38.33a)
The plot is that of Fig 38.3, but with each factor being 1/4.
Chapter 6: Power in Pulse Trains
207
This spectrum is exactly the same as that for unipolar NRZ shown in (36.3b) with V = 1. One way to
understand this fact is that for every NRZ sequence y n there is a NRZI sequence y' n given by (36.3).
y'
n = yn – yn-1 . // mod-2 math
This equation can be solved for y
n in terms of y' n (assume y 0= 0),
y
n = Σm=1n y'm n = 1,2,3....
Consider the space of all random sequences of 1's and 0's ( random pulse trains p = 1/2). Since we just
showed that the relation {y n} ↔ {y'n} is one-to-one, the mapping f: {y n}→{y'n} just reorders the set of
random sequences in the ensemble used to compute the spectral power density, so that density cannot
change.
In contrast, for p ≠ 1/2, the NRZI power spectrum (38.32) is quite different from the NRZ power
spectrum (36.3),
< P(ω) > = 1
4 (1/ω1) [ sinc2(πx) (1-a2)
1+a2 -2acos(ωT1) + δ(x)] // unipolar NRZI (38.32)
a = ( 2 p - 1 )
< P(ω)> == 1
4 (1/ω1) [sinc2(πx) (1-a2) + (2p)2 δ(x)] // unipolar NRZ (36.3)
Chapter 6: Power in Pulse Trains
208 Example 2: Bipolar NRZI line code
data = [ 1 0 1 1 0 0 1]
encode = [ 1 -1 -1 1 -1 -1 -1 1]
In this case A = 1 and B = -1 so the general NRZI spectrum (38.21) becomes
< P(ω) > = P
pulse (ω) (1-a2)
1+a2 -2acos(ωT1) a = (2p-1) // bipolar NRZI (38.34)
and for p = 1/2 (a=0) we obtain,
< P(ω) > = P
pulse (ω) . // = (1/2 π) T1 sinc2(ωT1/2) for the box pulse (36.1) (38.35)
In contrast, the Bipolar NRZ spectrum is Ppulse (ω) [(β-α) + α ∑
m = -∞∞
2πδ(ωT1- 2πm) ] where
α = (1-2p)2, β = 1, and ( β-α) = 4p(1-p) = 1-a2 so it's spectrum is
< P(ω)> = Ppulse (ω) [(1-a2) + (1-2p)2 ∑
m = -∞∞
2πδ(ωT1- 2πm) ] // bipolar NRZ (38.36)
If p = 1/2 (a=0) then the Bipolar NRZ spectrum becomes
< P(ω)> = P
pulse (ω) ( 3 8 . 3 7 )
which is the same as for Bipolar NRZI.
Appendix A: Delta Function Technology
209 Appendix A: Delta Function Technology
The delta fun
ction is a mathematical tool that simplifies the description of mathematical relationships and
allows one to consider certain idealized situations wh ich cannot really exist in practice. The down side is
that delta functions also serve to confuse readers who are not used to working with them. As later sections
of this monograph were written, it became apparent that much of the development leans fairly heavily on
"delta function technology", and one must understand delta functions at a somewhat deeper level than
presented in Section 1. Of particular importance for spectral power density is our special meaning for the
symbol δ(0) which appears in Chapter 6.
Early authors were squeamish about using the Dir ac delta function (for example, Smythe). The theory
of distributions was made rigorous in the 1935-1940 er a by Sobolev and then Schwartz. The theory makes
use of a class of extremely smooth functions and ope rators called linear functionals, and the reader can
find a good presentation in Stakgold Chapters 1 and 5. In brief, a functional T is just a mapping from some Hilbert Space to the real numbers. Usually that
Hilbert Space is a set of reasonable functions define d on some interval. One can write a functional T in
this notation: <T,f> = real number, some function of T and f. A functional is linear if it does the usual
things like <T,f
1+f2> = <T,f 1> + <T,f 2>.
A distribution t is any linear functional which acts on a certain subset of all possible functions f. It acts (theoretically) only on the subset of smooth functions φ called test functions. Thus, a distribution t is
represented as <t, φ> = real number, a function of t and φ.
For example, every reasonable real function f defin es a linear functional and thus a distribution in this
manner, where we happen to use the interval (-∞ ,∞),
<f,φ> =
∫-∞ ∞ dx f(x)φ(x) .
Obviously the integral of real functions is some real number. The function f(x) must be reasonable enough so that the integral is well behaved (it is "locally integrable"). In this context, the linear functional which defines the distribution known as the delta function is
given as
<δ
ξ,φ> = φ (ξ) special case: < δ,φ> = φ (0)
or
∫-∞ ∞ dx δ(x-ξ)φ(x) = φ(ξ) ∫-∞ ∞ dx δ(x)φ(x) = φ(0)
which we normally think of as the sifting property (2.3). The thing δξ is the distribution (the linear
functional name), while the thing written as δ(x-ξ) is a "symbolic function" or "generalized function"
associated with the distribution δξ. But loosely speaking, one refers to δ(x-ξ) as the distribution.
The general idea is that functions like δ(x) or δ'(x) have a meaning only when they are inside an
integral. When they appear standing alone, they are "symbolic functions", such as in (4.6)
[ RC d/dt + 1] g(t) = δ(t). This is a symbolic or distributiona l equation which acquires meaning when both
sides are placed inside the same integration. Formally that integration should be against a test function,
but we can think of it as being against any reasonable function; the main idea is that everything must
converge.
Appendix A: Delta Function Technology
210 Although the above discussion is expressed in terms of one dimensional integrals, the concept applies
in any number of dimensions. For example, δ(3)(r - a) is a delta function in 3D space.
The reader does not have to go learn the theory of distributions in order to follow this appendix, but it
is good to know that such a theory exists and justifies the manipulations done all the time with delta functions.
(a) Models for Delta Functions and two derivations of (2.1)
By
"model" we mean a sequence of smooth functions which all have unit area and which in some limit
become isolated about a certain point on the real axis. The essential requirements for a delta function candidate are these:
lim
ε→0 ∫-ε ε dk δ(k) = 1 δ(k) = 0 for any k>0 and for any k<0 (A.1)
These equations say that the area under δ(k) is 1 and the function δ(k) is isolated to an infinitely small
neighborhood of k = 0. There are an infinite numbe r of possible delta functi on models, and we shall
consider several in this appendix, some of which are used in the main text.
Our first candidate delta function mode l is the pulse shown in (9.1) with τ/2 set to 1/(2A),
δ
1(k,A) ≡ A [ θ (k + 1
2A ) - θ (k - 1
2A ) ]
( A . 2 )
F i g A . 1
In the limit A →∞, the box becomes very tall and very localized and maintains area 1, so
limA→∞ δ1(k,A) = δ(k) . (A.3)
Here are two candidates δ2(k,A) and δ2'(k,A) for which we just show engineering drawings and no
equations,
Appendix A: Delta Function Technology
211
F i g A . 2
Both these candidates meet our requirements (A.1). Notice that both have unit area, and both get horizontally compressed around k = 0 as A → ∞ and both get "tall". Thus we can say
lim
A→∞ δ2(k,A) = δ(k) (A.4)
limA→∞ δ2'(k,A) = δ (k) . (A.5)
For our first model δ1 we get δ (0) = +∞ while for the last two models we get δ(0) = -∞ and δ(0) = 0. It is
in fact possible to construct a model in which δ(0) comes out being any desired real number. This number
δ(0) has no significance because the point k = 0 is singular and the limit of δ(k) approaching this point
from either direction does not exist. This is a mu ch more serious matter than both limits existing and
being different, which is the case for the Heaviside θ function which has a simple discontinuity at t = 0,
Fig A.3
From the left, the limit is 0, from the right, the limit is 1, and the Fourier-correct value at the discontinuous point is θ(0) = 1/2 (as shown in Section 8 (c)).
Our next model of interest is this,
δ
3(k,A) ≡ 1
4πA exp( -k2/ 4 A ) ( A . 6 )
which is a Gaussian centered at k = 0. This function has area = 1 for any A > 0,
The half-width of (A.6) occurs roughly when k2/4A = 1, so the full width is then Δ k ≈ 4A . Here is a
plot with A = .05 for which Δk ≈ 4 .05 = 0.9,
Appendix A: Delta Function Technology
212
Fig A.4
As A→0, the Gaussian becomes taller and narrower and we then have
limA→0 δ3(k,A) = lim A→0 1
4πA exp( -k2/4A) = δ(k). (A.7)
Now consider this standard integral ( GR 3.323.2 where a = p2 and b = q ),
∫-∞ ∞ dx exp[ - ( ax2 + bx ) ] = π
a exp[ (b2/4a) ] . (A.8)
Setting a = A, b = -ik gives
∫-∞ ∞ dx eikx exp(-Ax2) = π
A exp(-k2/4A) = 2 π { 1
4πA exp( -k2/4A) }
or
∫-∞ ∞ dx eikx exp(-Ax2) = 2π δ3(k,A). (A.9)
Taking the limit A →0 of both sides gives
∫-∞ ∞ dx eikx = 2πδ( k ) . ( A . 1 0 )
This then is our first distribution theory derivation of (2.1).
Here is another candidate delta function model :
δ4(k,B) ≡ sin(Bk)
πk = B
π sin(Bk)
Bk = B
π sinc(Bk) (A.11)
The area under (B/ π) sinc(Bk) is unity,
Appendix A: Delta Function Technology
213
The half-width of (A.11) is determined roughl y by the first zero of sin(Bk) = 0 so Bk = π.
The full width of the peak is then Δk = 2π/B. Here is a plot for B = 10 with Δ k ≈ 2π/10 = 0.6.
Fig A.5
As B gets large, the function shrink s in around k = 0, and we then have
limB→∞ δ4(k,B) = lim B→∞ sin(Bk)
πk = limB→∞ B
π sinc(Bk) = δ(k) . (A.12)
Now consider this simple integral
∫-B B dx eikx = 2 ∫0 B dx cos(kx) = 2 sin(kB)
k = 2π sin(kB)
πk = 2π δ4(k,B) (A.13)
Taking the limit as B →0 we find
∫-∞ ∞ dx eikx = 2 ∫0 ∞ dx cos(kx) = 2 π δ(k) (A.14)
and we have a second derivation of (2.1).
(b) Models for Periodic Delta Functions
We start here
with the follo wing delta function model,
δ5(k,N) ≡ 1
2π sin[(N+1/2)k]
sin (k/2) ( A . 1 5 )
and we shall be interested in the limit N →∞. This particular candidate is periodic in k with period 2 π, as
we now show:
Appendix A: Delta Function Technology
214 sin[(N+1/2)(k+2 π)] = sin [(N+1/2)k + 2 π(N+1/2) ] = sin [(N+1/2)k + π ] = - sin [(N+1/2)k]
sin[(k+2 π)/2] = sin(k/2 + π ) = -sin(k/2)
=> sin[(N+1/2)(k+2 π)]
sin[(k+2π)/2] = sin[(N+1/2)k]
sin (k/2) => δ5(k+2π,N) = δ5(k,N) (A.16)
Looking at the peak at k = 0 for large N, the half wi dth occurs at the first zero of sin[(N+1/2)k] so Nk ≈ π.
The full width is then Δk ≈ 2π/N. Here is a plot of the central peak for N = 60 for which Δk ≈ 0.1
Fig A.6
The periodicity of δ5 is apparent if we plot 2 π δ5(k,N) for N = 60,
Fig A.7
Appendix A: Delta Function Technology
215 where the peaks are separated by 2 π. We are going to show that
limN→∞ δ5(k,N) = δ(k) + δ(k-2π) + δ(k+2π) + ..... = ∑
m = -∞∞
δ(k - 2πm) . (A.17)
As N gets large, each red region of activity becomes isolated more and more to the location of the putative delta function peak. The function approaches 0 anywhere between the peaks in the following
sense (which has a very distributional flavor). As N → ∞, the numerator of δ
5(k,N) oscillates faster and
faster. When averaged over any tiny region of width ε, we can make N large enough to make this average
be arbitrarily small. For example, when averaged ove r 1 nanometer of the above plot near k = 2, we can
find a sufficiently large N to make this average be smaller than 10-100. This is the basic idea of
something "washing out". Thus, we realize our delta function requirement that δ(k) = 0 for k ≠ n2π. The
other requirement is that we must show th at the area under each delta peak is unity.
Consider a close neighborhood of the central peak in Fi g A.7. For large N, only a small region of k near
the central peak |k| < Δk ≈ π/N contributes to δ
5 as the Figures above show, since sin[(N+1/2)k] oscillates
so fast beyond this region. In this region we can approximate sin(k/2) by (k/2) so that
δ5(k,N) ≡ 1
2π sin[(N+1/2)k]
sin (k/2) ≈ 1
2π sin[(N+1/2)k]
(k/2) = (N+1/2)k
2π(k/2) sinc[(N+1/2)k]
= (N+1/2)
π
sinc[(N+1/2)k] = δ4(k,N+1/2) // using (A.11)
We already know that the area under δ4 is 1, and that it is a viable delta function model. But since δ5 is
periodic, all its peaks must look like δ4. Broadening our scope to the en tire k axis, we conclude that
δ5(k,N) ≈ ∑
m = -∞∞
δ4(k - 2πm, N+1/2) for large N (A.18)
As N → ∞ we then get
limN→∞ δ5(k,N) = lim N→∞ 1
2π sin[(N+1/2)k]
sin (k/2) = ∑
m = -∞∞
δ(k-2πm) (A.19)
Our next multi-peak delta function candidate is the following,
δ
6(k,N) ≡ 1
2π [2πδ5(k,N)]2
(2N+1) = 1
2N+1 sin2[(N+1/2)k]
2π sin2(k/2) . (A.20)
This δ6 has the same periodicity of δ5 so has identical peaks spaced by 2 π in k. Here is the central peak
for N = 20
Appendix A: Delta Function Technology
216
Fig A.8
and the width is Δ k ≈ 2π/N, the same as for δ5. In the region of contribution, we again set sin(k/2) ≈ k/2
so that
δ6(k,N) ≈ 1
2N+1 sin2[(N+1/2)k]
2π (k/2)2 = [(N+1/2)k]2
2π(2N+1)(k/2)2 sinc2[(N+1/2)k] = (2N+1)
2π sinc2[(N+1/2)k]
= (B/ π) sinc2(Bk) B = N+1/2
The area under δ
6 for any N is unity,
Since δ6 has the same periodicity as δ5, it has the same limit as N →∞
limN→∞ δ6(k,N) = lim N→∞ sin2[(N+1/2)k]
(2N+1)2π (k/2)2 = ∑
m = -∞∞
δ(k-2πm) (A.21)
Our interest in δ6 is that it is a delta function model formed by squaring another delta function model.
Consider next the following ca ndidate delta function model,
δ7(k, a) ≡ (1/π) (1-a2) [cos2(k)]
1+a2+2acos(2k) ( A . 2 2 )
We are interested in this model as a → -1. Here is a motivating plot of δ7 for a = -0.9 :
Appendix A: Delta Function Technology
217
Fig A.9
First of all, we can see that δ 7 is periodic in k having period π, since both cos2(k) and cos(2k) have period
π. Near a = -1, the denominator approaches 2 -2cos(2k) = 2(1-cos(2k)) = 4sin2(k) which vanishes at k =
mπ for m integer. As long as we avoid these points, we can see that lim a→-1 δ7(k, a) = 0 for k ≠ mπ due
to the (1-a2) factor in the numerator. We need only show that the integral if δ7 is 1 when taken in a small
region around one of the delta peaks. As before we co nsider the peak at k = 0 and compute the integral
lim ε→0 ∫-ε ε dk [lima→-1δ7(k, a)] = lim ε→0 lima→-1 ∫-ε ε dk {(1/π) (1-a2) [cos2(k)]
1+a2+2acos(2k) }
= l i m a→-1 limε→0 ∫-ε ε dk {(1/π) (1-a2) [cos2(k)]
1+a2+2acos(2k) } // interchange limit order
= l i m a→-1 limε→0{(1-a2) ∫-ε ε dk 1
1+a2+2acos(2k) } // cos(k) ≈ 1 for ε << 1
= ( 1 / π) limε→0 lima→-1 {(1-a2) ∫-ε ε dk 1
1+a2+2acos(2k) } // interchanged limit order again
In passing, note that the cos2(k) numerator factor really plays no role in things and we could have set it to
1 in δ7, but we kept it since it appears there in our AMI application. We now have Maple do the integral
as follows,
Appendix A: Delta Function Technology
218
We now continue the above evaluation,
= (1/ π) limε→0 lima→-1{(1-a2) [ 2 1
a2-1 tan-1( a-1
a+1 tanε ) ]
= -(2/ π) limε→0 lima→-1{ tan-1( a-1
a+1 tanε ) }
= - (2/ π) limε→0 { tan-1(- ∞) } // argument of tan-1 is (-1-1
+0 tanε) = (-∞)
= - (2/ π) limε→0 { -π/2 } = - (2/π )(-π/2) = 1 .
In this model, the area under δ7 is not unity for any value of a, so we have to deal with both limits in our
evaluation. There are Moore-Osgood theorem subtleties involving the limit order interchanges which
could be reviewed, but we omit that level of detail.
Therefore we have shown that in the region of the central peak.
lim
a→-1δ7(k, a) = δ(k) -π < k < π
Since δ7 is periodic with period π, we know that the full result for all k is this
lima→-1δ7(k, a) = lim a→-1 (1/π) (1-a2) [cos2(k)]
1+a2+2acos(2k) = ∑
m = -∞∞
δ(k - mπ) (A.23a)
If we remove [cos2(k)] from the limit and then divide both sides of the right equation above by this value,
we get on the right that [cos2(k)] = cos2(mπ) = 1, so the following is also true
lima→-1 (1/π) (1-a2)
1+a2+2acos(2k) = ∑
m = -∞∞
δ(k - mπ) (A.23b)
which is really a more fundamental result. Changing a →-a this can also be written
lima→+1 (1/π) (1-a2)
1+a2-2acos(2k) = ∑
m = -∞∞
δ(k - mπ) (A.23c)
Appendix A: Delta Function Technology
219
Going back to (A.23a), if we change from k to k' = k + π/2, then
cos(k) = cos(k'- π/2) = sin(k')
cos(2k) = cos(2k'-π ) = - cos(2k') .
Then δ
7 may be written
δ7(k, a) ≡ (1/π) (1-a2) [cos2(k)]
1+a2+2acos(2k) => δ7(k'-π/2, a) ≡ (1/π) (1-a2) [sin2(k')]
1+a2–2acos(2k')
Then from (A.23a)
lim
a→-1δ7(k'-π/2, a) = ∑
m = -∞∞
δ(k'-π/2 - mπ) = ∑
m = -∞∞
δ(k'- [m+1/2] π) .
Change the summation index to n = 2m+1, so that m+1/2 = n/2. The n sum then includes only odd
integers from -∞ to ∞ , so
lima→-1δ7(k'-π/2, a) = ∑
n = ±odd
δ(k'-nπ/2) .
Thus we have arrived at a new multi-delta function model δ8(k',a) ≡ δ7(k'-π/2, a) to get
δ8(k, a) ≡ (1/π) (1-a2) [sin2(k)]
1+a2–2acos(2k) ( A . 2 4 )
lima→-1 δ8(k, a) = lim a→-1 (1/π) (1-a2) [sin2(k)]
1+a2–2acos(2k) = ∑
m = ±odd
δ(k-mπ/2) (A.25a)
This is the delta model needed to take the p→ 1 limit of our AMI spectrum in Section 37.
If we extract [sin2(k)] from the limit and divide both sides of the rightmost equation by this quantity
and note that sin2(mπ/2) = 1 for all odd values of m, we get these alternate forms:
lima→-1 (1/π) (1-a2)
1+a2–2acos(2k) = ∑
m = ±odd
δ(k-mπ/2) (A.25b)
lim
a→+1 (1/π) (1-a2)
1+a2+2acos(2k) = ∑
m = ±odd
δ(k-mπ/2) (A.25c)
Appendix A: Delta Function Technology
220 (c) Derivation of (13.2) and (13.3)
The first goal here is to derive this equation,
∑
n = -∞∞
eink = ∑
m = -∞∞
2πδ(k - 2πm) - ∞ < k < ∞ . (13.2)
Consider this finite version of the sum appearing on the left side of (13.2). We add and subtract 1 to
obtain the sum as the these three terms,
∑
n = -NN
eink = (1 + eik + ei2k + ... + eiNk) + (1 +e-ik + e-i2k + ... + e-iNk) - 1 . (A.26)
The following is the standard formula fo r summing N terms of a geometric series,
1 + x + x
2 + ... + xN = (1 - xN+1)/(1-x), (A.27)
Using this formula first for x = e
ik and then for x = e-ik, we may write (A.26) as
∑
n = -NN
eink = (1 - eik(N+1))/(1-eik) + (1 - e-ik(N+1))/(1-e-ik) - 1 (A.28)
In the first term multiply top and bottom by e-ik/2 to get
first term = (e
-ik/2 - eik(N+1/2))/ ( e-ik/2- eik/2) = (e-ik/2 - eik(N+1/2)) / [-2isin(k/2)] .
Since the second term is the complex conjugate of the first, we get second term = (e
ik/2 - e-ik(N+1/2)) / [2isin(k/2)]
Therefore
∑
n = -NN
eink = second term + first term - 1
= [ eik/2 - e-ik(N+1/2) – e-ik/2 + eik(N+1/2)] / [2isin(k/2)] - 1
= [ 2i sin(k/2) + 2isin[k(N+1)/2] ] / [2isin(k/2)] - 1 = [sin(k/2) + sin[k(N+1)/2] ] / [sin(k/2)] - 1 = sin[k(N+1)/2] / sin(k/2) // see also Gradshteyn and Ryzhik, p 37, 1.342.2
Appendix A: Delta Function Technology
221 and we arrive at this result (valid for any positive integer N):
∑
n = -NN
eink = 2π { sin[(N+1/2)k]
2π sin(k/2) } ( A . 2 9 )
From (A.15) we recognize {...} in (A.29) as δ5(k,N), so (A.29) says
∑
n = -NN
eink = 2π δ5( k , N ) . ( A . 3 0 )
We have thus derived equation (13.3).
We then take the limit N →∞ of both sides of this equation. The right side is given by (A.19) so we
get
∑
n = -∞∞
eink = ∑
m = -∞∞
2πδ(k - 2πm) - ∞ < k < ∞ (A.31)
and we have derived (13.2) as promised. Bo th sides are visibly periodic with period 2π .
(d) Undoing the limit N → ∞ : the mea ning of δ(0)
Go back now to (A.29) and evaluate both sides at k = 0:
∑
n = -NN
1 = 2π limk→0{ sin[(N+1/2)k]
2π sin(k/2) = lim k→0{ sin[(N+1/2)k]
(k/2) } (A.32)
= (2N+1) lim k→0 sinc[ (N+1/2)k] = (2N+1).
But this is exactly the sum shown on the left, so we find that our k = 0 limit of (A.29) happily says
(2N+1) = (2N+1) . Taking the limit N → ∞ of (A.30) and then the limit k → 0 we find that
∑
n = -∞∞
1 = [ 2πδ( 0 ) ] . ( A . 3 3 )
We noted earlier that δ(0) is model dependent and for this model δ5 we get δ (0) = +∞ . There are times
when we want to "undo the limit" N →∞, to get
∑
n = -NN
1 = [ 2 πδ(0)]undone = (2N+1) . (A.34)
Appendix A: Delta Function Technology
222
In this "undoing" we have to be careful not to use the delta function property δ (x/a) = a δ(x) since this is
not valid in the "pre-limit". Example 1
When we deal with pulse trains, we will be adding exponentials of the form exp(in ωT1). From (A.31) we
therefore have,
∑
n = -∞∞
einωT1 = ∑
m = -∞∞
2π δ(ωT1- 2πm) . (A.35)
If we evaluate the above formula at ω = 0, we get
∑
n = -∞∞
1 = ∑
m = -∞∞
2πδ(- 2πm)
We now observe that δ(-2πm) = δm,0 δ(0), so we then get, as in (A.33),
∑
n = -∞∞
1 = [ 2 π δ( 0 ) ] ( A . 3 6 )
We understand this as a symbolic limit. We can now undo the limit by replacing the sum endpoints with -
N and N, and replace 2 πδ(0) with (2N+1), and the result is consistent. Had we rescaled the delta function
by saying for example δ(-2πm) = (1/2 π) δ(-m), we would end up with a contradiction when we tried to
"undo the limit". Example 2
Consider the square of the sum shown in Example 1,
{
∑
n = -∞∞
einωT1 }2 = { ∑
m = -∞∞
2π δ(ωT1- 2πm) } 2 ( A . 3 7 )
We write the RHS using m and k for our two summation indices, then we move both summations to the left. Inside this double sum we get
δ(ωT
1- 2πm)δ(ωT1- 2πk) .
Since the first delta function will "pin" ωT1to the values 2 πm, we can replace ωT1 with 2πm in the
second delta function and not change a thing, obtaining
δ(ωT1- 2πm)δ(2πm- 2πk) = δ(ωT1- 2πm) δm,n δ(0) . (A.38)
Appendix A: Delta Function Technology
223
The Kronecker delta δm,n now removes one of the summations , and we end up with this result:
{ ∑
n = -∞∞
einωT1 }2 = [ 2πδ (0) ] { ∑
m = -∞∞
2π δ(ωT1- 2πm) } . (A.39)
Evaluating the above at ω = 0 yields
{ ∑
n = -∞∞
1 }2 = [ 2πδ(0) ] [ 2πδ( 0 ) ] . ( A . 4 0 )
And if we now undo the limit, we get this self-consistent result,
{ ∑
n = -NN
1 }2 = [ 2πδ(0) ] [ 2πδ(0) ] = (2N+1)2 . ( A . 4 1 )
Once again, δ(0) is really undefined and model dependent, but for our δ5 model we use 2 πδ(0) just as a
shorthand for the limit of (2N+1) as N → ∞. The number (2N+1) will be the number of pulses in a pulse
train and that pulse train becomes infinitely long as N →∞. In such a limit, quantities like average energy
per pulse remain finite.
(e) The function Θ(a ≤ x ≤b) and related sums
In equation (
2.2) we noted that [ θ(x) is the Heaviside step function of Fig 1, sometimes written as H(x) ]
∫a b dx δ(x-y)f(x) = f(y) θ(b-y)θ( y - a ) a < b (2.2)
which we write here as
∫a b dx' δ(x'-x)f(x') = f(x) θ(b-x)θ(x-a) a < b (A.42)
The theta functions just set the result to 0 if the delta function hit lies outside the interval (a,b).
Function θ(x) is the Heaviside step function having these properties:
θ(x) =
⎩⎨⎧ 1 if x>0
1/2 if x=0
0 if x<1 . ( A . 4 3 )
These theta functions are always a bit confusing, and sometimes things are clearer using a different notation. Suppose we define the following new function,
Appendix A: Delta Function Technology
224 Θ(a ≤ x ≤ b) ≡ Θ(a,x,b) ≡
⎩⎨⎧ 1 if a<x<b
1/2 if x=a or x=b
0 if x<a or x>b a < b (A.44)
The notation Θ(a ≤ x ≤b) is merely a suggestive way to write the function Θ(a,x,b).
We shall now state and prove a few simp le "theorem" regarding this function Θ.
Fact: Θ(a ≤ x ≤ b) = θ (b-x)θ(x-a) a < b (A.45)
Proof.
We just exhaust all cases:
x < a: θ(b-x)θ(x-a) = θ(b-x) * 0 = 0
x = a: θ(b-x)θ(x-a) = θ(b-a)θ (a-a) = 1 * 1/2 = 1/2
a < x < b: θ(b-x)θ(x-a) = 1*1 = 1
x=b: θ(b-x)θ(x-a) = θ(b-b)θ(b-a) = 1/2 * 1 = 1/2
x > b: θ(b-x)θ(x-a) = 0* θ(x-a) = 0
Therefore, we may write (A.42) in this more friendly manner,
∫a b dx' δ(x'-x)f(x') = f(x) Θ(a ≤ x ≤ b) a < b (A.46)
Fact: Θ(a ≤ x+c ≤ b) = Θ (a-c ≤ x ≤ b - c ) ( A . 4 7 )
Proof:
The inequality notation makes this fact seem co mpletely obvious, but we will just make sure by
writing out both sides of the equation:
Θ(a ≤ x+c ≤ b) ≡ Θ(a,x+c,b) ≡
⎩⎨⎧ 1 if a< x+c <b
1/2 if x+c =a or x+c =b 0 if x+c <a or x+c >b a < b
Θ(a-c ≤ x ≤ b-c) ≡ Θ(a-c,x,b-c) ≡
⎩⎨⎧ 1 if a-c <x<b-c
1/2 if x= a-c or x= b-c 0 if x<a-c or x>b-c a-c < b-c
The next fact is not quite so obvious but is very useful:
Fact:
∑
m = -∞∞
Θ(mα-α/2 ≤ x ≤ mα+α/2) = 1 α > 0 -∞ < x < ∞ (A.48)
Proof: Let's write out some of the terms in this su m. Here we show terms for m = -1, 0, and 1
.... + Θ(-α-α/2 ≤ x ≤ -α+α/2) + Θ (0-α/2 ≤ x ≤ 0+α/2) + Θ (α-α/2 ≤ x ≤ α+α/2) + ...
or
Appendix A: Delta Function Technology
225 .... + Θ ( -(3/2)α ≤ x ≤ -(1/2)α ) + Θ( -(1/2)α ≤ x ≤ (1/2)α) + Θ((1/2)α ≤ x ≤ (3/2)α) + ...
The terms of the m sum therefore partition the real x axis into intervals of width α. If x falls within one
of these intervals, then only the single term covering that interval contributes in the m sum and the sum is then 1. If x happens to fall exactly on the boundary between two interv als, then each of those intervals
contributes 1/2 to the sum, and the sum is again 1. For example, if x = (1/2) α, then each of the rightmost
two terms shown above contributes 1/2. Therefore the sum is 1 for all possible values of x.
Fact:
∑
m = -∞∞
Θ(-α/2 ≤ x-mα ≤ +α/2) = 1 α > 0 -∞ < x < ∞ (A.49)
Proof : Apply (A.47) to the Θ function shown in the sum :
Θ (a ≤ x+c ≤ b) = Θ( a-c ≤ x ≤ b-c ) a = - α/2, c = -m α, b = α/2
Θ(-α/2 ≤ x-mα ≤ α/2) = Θ (-α/2+mα ≤ x ≤ α/2+mα)
Therefore, summing both sides and using (A.48),
∑
m = -∞∞
Θ(-α/2 ≤ x-mα ≤ α/2) = ∑
m = -∞∞
Θ(-α/2+mα ≤ x ≤ α/2+mα) = 1
Corollary:
∑
m = -∞∞
Θ(-α/2 ≤ x + mα ≤ +α/2) = 1 α > 0 -∞ < x < ∞ (A.50)
Proof: For any summand f(m) it is clear that ∑
m = -∞∞
f(m) = ∑
m = -∞∞
f(-m), so (A.50) is the same as (A.49).
(f) The product of two delta functions and more on δ(0)
This subsecti
on is a sort of coda on the subject of δ(0) where we further attemp t to justify the use of δ(0)
even though δ(0) is formally undefined. We continue the distribution discussion begun at the start of this
Appendix.
Consider a voltage pulse v(t) with a shape correspondi ng to one our delta functi on models. If this voltage
is placed across a resistor of value R = 1, the energy in the pulse is given by
E =
∫-∞ ∞ dt v2(t) . energy = time integral of power
If we take the limit v(t) →δ(t), what happens? One is tempted to say
E = ∫-∞ ∞ dt δ2(t) = ∫-∞ ∞ dt δ(t) δ(t) = δ(0) ∫-∞ ∞ dt δ(t) = δ (0) * 1 = δ(0) = +∞
Appendix A: Delta Function Technology
226 and one concludes (correctly) that the energy in a delta function pulse is infinite and positive. There are
several problems with the this analysis.
First, we have already seen that with different δ models, we can get δ(0) to be any number we want,
including + ∞. -∞ and 0. Very embarrassing.
Second, the distribution < δξ2, φ> = δ (ξ)φ(ξ) is not a sensible distribution since this linear functional
maps into a real number which is undefined at ξ = 0. In general the product of two distributions (in the
sense we use it here) is not even defined in the realm of distribution theory, unless one or both distributions are regular functions. In that case we could have, for example,
< f δ
ξ, φ > = < δξ, fφ> = ∫-∞ ∞ dx δ(x-ξ)f(x)φ (x) = f(ξ)φ(ξ) = well defined .
Some people have tried to incorporate products of sin gular distributions into distribution theory, but it is
not clear how their results apply in our current context (see for example Colombeau 1990). Note that there are other meanings of the prod uct of two distributions. One is called a convolution
product which is like f(g(x)) for functions, while the other involves multiple variables like δ
(2)(r-r') =
δ(x-x')δ(y-y'). Neither of these products involves products of symbolic functions in the same variable such
as δ(t)δ(t).
So, how might we compute the energy in a delta function voltage pulse? The only reasonable thing to do
is to back δ(t) off to one of its models and see wh at happens. For example, suppose we take δ
1(t,A) as
stated in (A.2), which is the simple box model of height A and width 1/A. Then
E = lim A→∞ { ∫-∞ ∞ dt [δ1(t,A)]2 }
Now [δ1(t,A)]2 is a box of width 1/A and height A2 so its area is A. Then
E = lim A→∞ { A } = + ∞ // recall δ(0) = +∞
Suppose we take our deviant delta function model (A.4) δ2(k,A) shown in Fig A.2 left. We get
E = lim A→∞ { ∫-∞ ∞ dt [δ2(t,A)]2 }
Since the pulse is squared, the area under [ δ2(t,A)]2 is 3A and we then get
E = lim A→∞ { 3A } = + ∞ // recall δ(0) = -∞
Finally, for the our second deviant model (A.5) δ2'(k,A) shown in Fig A.2 right we get
E = lim A→∞{ A/2 } = + ∞ // recall δ(0) = 0
Thus, we get E = + ∞ regardless of the value of δ(0) in the model.
Appendix A: Delta Function Technology
227 Despite this discussion which shows that δ(0) is undefined, we nevertheless use δ(0) with a particular
model in mind because it allows us to avoid dealing with specific boundaries in equations.
In Example 2 of the previous subsection, we saw the use of 2 πδ(0) as meaning 2N+1 for a very long
pulse train starting at -N in the distant past and ending at +N in far future. We would rather think in terms
of an infinite pulse train and 2 πδ(0), but the physical meaning is a very long pulse train and 2N+1. It is
just a convenient notation.
Another common example has to do with "box norma lization" which in one dimension is the model
presented as (A.13),
∫-L/2 L/2 dx eikx = 2π sin(kL/2)
πk = 2π δ4(k,L/2)
∫-L/2 L/2 dx ei0x = L = 2 πδ(0) .
In this case, we use 2 πδ(0) to represent the length of a box which is some very large L. Rather than carry
the large but finite L along in all equations, we can use 2 πδ(0) as needed represent this long length.
The conclusion here is that we can use 2 πδ(0) as a notational device provided we are very careful as to the
meaning of that device. We must always know how to undo the limit, and that implies a specific delta function model for a specific application.
Appendix B: A Certain Identity
228 Appendix B: Derivation of a Certain Identity
Theore
m: For N and s both integers (N > 0)
∑
m = 0N-1
e+ims(2π/N) = N ∑
m = -∞∞
δs,mN . ( B . 1 )
This identity is used in Section 27 (b) on the Discrete Fourier Transform.
To prove this relation, we shall first show that each side is periodic in s with period N, then we shall
verify that the relation is true for s = 0,1,2...N-1. We will have then shown that the relation is true for all
integer values of s. First, give names to the two sides of (B.1)
f(s) ≡
∑
m = 0N-1
e+ims(2π/N) g ( s ) ≡ N∑
m = -∞∞
δs,mN .
Function f(s) is periodic as claimed because
f(s+kN) = ∑
m = 0N-1
e+im(s+kN)(2 π/N) = ∑
m = 0N-1
e+ims(2π/N) eimkN(2π/N) = ∑
m = 0N-1
e+ims(2π/N) = f(s)
Function g(s) is periodic as claimed because ( m' ≡ m-k )
g(s+kN) ≡ N ∑
m = -∞∞
δs+kN,mN = N ∑
m = -∞∞
δs,(m-k)N = N ∑
m' = -∞∞
δs,(m')N = g(s)
Thus, each side of (B.1) is periodic in s with period N. Now, consider the proposed equality:
∑
m = 0N-1
e+ims(2π/N) = N ∑
m = -∞∞
δs,mN . (B.1)
For s = 0, the above claims
∑
m = 0N-1
1 = N ∑
m = -∞∞
δ0,mN = N δ0,0 = N .
But this is true since the left side sum is obviously N as well. Thus, (B.4) is valid for s = 0.
Appendix B: A Certain Identity
229 For s = 1,2,3....N-1, we claim that on the left side of (B.1) we are adding N equally spaced points around a
circle in the complex phasor plane, and therefore the sum on the left side is zero. Meanwhile, the right
side is also zero because for any of these s values, ther e is no integer m such that s = mN hence δs,mN = 0.
Thus, if one accepts the circle argument, one finds that for all these s values both sides of (B.1) vanish.
To avoid the circle construction, we can simply compute the left side of (B.1) using the formula
∑
m = 0N-1
e+ims(2π/N) = 1 + x + x2 + ... + xN-1 = (xN- 1)/(x-1) with x = e+is(2π/N)
For s = 1,2,3....N-1 the phasor x = e+ims(2π/N) ≠ 1, so denominator (x-1) ≠ 0. Meanwhile,
x
N = e+is(2π/N)N = e+is2π = 1 for any integer s
Therefore numerator (xN- 1) = 0 and the sum thus vanishes for these values of s.
Thus we have shown that (B.1) is valid for s = 0,1,2...N-1, and since both sides of (B.1) are periodic in s with period N, it must be that (B.1) is valid for all integers s.
Appendix C: The Fourier and Hilbert Transforms
230
Appendix C: The Fourier Transform and its relation to the Hilbert Transform
In this Appendix we refer to the Fourier
Integral Transform simply as the Fourier transform. We develop
more "facts" about Fourier transforms, including new notations, and present a set of closely related examples. The pf pseudofunction and pr incipal part integrals are introduced in the context of what we call
"the pole avoidance rule". The connection between the Fourier and Hilbert transforms is then used as an
exercise in applying the developed methods.
(a) Fourier Transform Notations
Recall fro
m Section 1 the statement of the F ourier transform, derived in Section 2,
X(ω) = ∫-∞ ∞ dt x(t) e-iωt projection = transform (1.1)
x(t) = (1/2 π) ∫-∞ ∞ dω X(ω ) e+iωt expansion = inverse transform (1.2)
We used lower case for a function of time lik e x(t), and upper case for the spectral components X( ω).
Although convenient in many situations , this notation is a bit limiting fo r more general use, so we replace
X(ω) with x^(ω ). The above can then be written as,
x^(ω) = k ∫-∞ ∞ dt x(t) e-iωt projection = transform
x(t) = (1/2 πk) ∫-∞ ∞ dω x^(ω) e+iωt expansion = inverse transform
Here we have added an arbitrary constant k to a llow for other scalings of the Fourier Transform. The
projection has k, the inversion has k-1 as shown. For us, k = 1, but other sources might have k = (2 π)-1 or
perhaps k = 1/ 2π to make the two equations symmetric.
The first line defines the Fourier transform of some arbitrary function x(t), but the second line acts only
on a function x^( ω) which is already a Fourier transform. We would like to have the second line act on an
arbitrary function as well. To do this, we replace x^( ω) by f(ω) and treat the second line as an operation
one applies to some arbitrary function f( ω),
f^
-1(t) = (1/2πk) ∫-∞ ∞ dω f(ω) e+iωt inverse transform
This then defines an operation performed on f(ω ) to generate f^-1(t) which is, by definition, the inverse
Fourier transform of f(ω ). So changing the dummy integrati on variable names both to u we get
f^(ω ) = k ∫-∞ ∞ du f(u) e-iωu Fourier transform of f(u)
f^-1(t) = (1/2πk) ∫-∞ ∞ du f(u) e+iut inverse Fourier transform of f(u) (C.1)
Appendix C: The Fourier and Hilbert Transforms
231
We think of both these equations as defining certain operations on an arbitrary function f(u). The function f(u) must be in the class of functions described in S ection 1(b) in order that the integral converge to a
function, though the class rules may be violated if one allows the transform and/or its inverse to be a
distribution.
From the first line of (C.1) we find that
f^(-ω ) = k
∫-∞ ∞ du f(u) e+iωu = k ∫-∞ ∞ du f(-u) e-iωu = [f(-u)]^(ω)
from which we obtain Fact 0: [f(-u)]^(- ω) = f^(ω ) ( C . 2 )
From the second line of (C.1) we find
f^
-1(-ω) = (1/2πk) ∫-∞ ∞ du f(u) e-iuω = (1/2πk2) k ∫-∞ ∞ du f(u) e-iuω = (1/2πk2) f^(ω ) .
We are dealing here with three dis tinct functions: f(u), f^(u) and f^-1(u), but we just showed that
Fact 1: f^
-1(u) = (1/2 πk2) f^(-u) (C.3)
so two of these three functions have a simple relationship. Notice in Fact 1 that one cannot simply suppress the argument u on both sides since u appears on the
left and –u appears on the right. When an argument is the same on both sides of an equation, or when an
argument is not needed, it can be suppressed to reduce clutter.
The fact that the Fourier transform is valid means that the inverse Fourier transform of the Fourier transform of a function is that function. Similarly, the Fourier transform of the inverse Fourier transform
of a function is that function. This was clear in (1.1) a nd (1.2) and in the new notation this becomes
Fact 2: [f^( ω)]^
-1(t) = [f^-1(ω)]^(t) = f(t)
or [ f ^ ] ^
-1 = [f^-1] ^ = f ( C . 4 )
This provides an example of suppressing arguments to declutter a simple equation. Notice that the constant k does not appear in Fact 2. Cons ider then this statement of Fact 2,
(f^)^
-1(t) = f(t)
Fact 1 applied to f →f^ says
(f^)^-1(t) = (1/2 πk2) (f^)^(-t)
so that
Appendix C: The Fourier and Hilbert Transforms
232
(1/2πk2) (f^)^(-t) = f(t)
or (f^)^(t) = 2 πk
2 f(-t)
which then gives
Fact 3: f^^(t) = 2 πk2 f(-t) (C.5)
FT of f(t) = f^(t) ⇔ FT of f^(t) = 2 πk2f(-t)
The first line says that applying the Fourie r transform twice to a function gives 2 πk2 times the function of
negated argument (see Stakgold Vol II (5.55) with k = 1). Nothing new is happening here, it is all just
notation. When k = 1/ 2π the factor 2 πk2 = 1 on the second line which is a strong motivation for that
scaling, but we had other motivations for k = 1 as outlined in Section 5.
The three Facts just stated are independent of the sign of the phase in the Fourier transform definition.
Operator Notation. An alternative notation similar to that u sed for Laplace transforms is the following:
f^(ω ) = F [f(u),ω] = F [f,ω] = F f(ω)
or f^ = F f ( C . 6 )
and for the inverse Fourier transform,
f^
-1(t) = F-1[f(u),t] = F-1[f,t] = F-1f(t)
or f^
-1 = F-1f . ( C . 7 )
The idea here is that F f = g is a new function obtained by acting upon function f with the Fourier
transform operator F. Similarly F-1f = h is a new function obtained by acting upon function f with the
inverse Fourier transform operator F-1. The three facts stated above can th en be translated into this new
notation.
Fact 1: f^-1(u) = (1/2 πk2) f^(-u) → F-1[f(s),u] = (1/2 πk2) F[f(s),-u]
F-1[f,u] = (1/2π k2) F[f,-u]
F-1f(u) = (1/2 πk2) F f(-u) (C.3)
Fact 2: [f^(ω)]^-1(t) = [f^-1(ω)]^(t) = f(t) → F-1[F[f(ω),s],t] = F[F-1[f(ω),s],t] = f(t)
[ f ^ ] ^-1 = [f^-1]^ = f → F-1F f = FF-1f = f (C.4)
Fact 3: f^^(t) = 2 πk f(-t) → F[F[f(ω),s],t] = 2 πk
2f(-t)
F [F[f,s],t] = 2 πk2f(-t)
F2f(t) = 2 πk2f(-t) (C.5)
Appendix C: The Fourier and Hilbert Transforms
233
The last line of Fact 2 has a particular appeal, since F-1F = FF-1 = 1 , the identity operator.
(b) Principal Value Integrals and the Tick Notation
Consider the following i
ntegral,
∫-∞ ∞ dx 1
x-a f(x) .
If f(x) is non-vanishing at x = a, and if a is real, this integral runs right through a pole at x = a. If a were complex , then the integral would run above or below the pole and one would be less concerned. If the
intention really is to run the integr ation right through the pole, it is u seful to make that fact very clear
using some kind of notation. One defines the notion of "going through the pole" as the following limiting operation,
∫-∞ ∞ dx 1
x-a f(x) = lim ε→0 [ ∫-∞ a-ε dx + ∫a+ε ∞ dx] 1
x-a f(x) ≡ ∫-- ∞
-∞ dx 1
x-a f(x) . (C.8)
Such an integration is referred to as a Cauchy Principal Value (or Principal Part) Integral. We defer to
section (d) below the pf notation whic h formalizes the above definition as
∫-∞ ∞ dx pf( 1
x-a ) f(x) ≡ ∫-- ∞
-∞ dx 1
x-a f(x) ≡ limε→0 [ ∫-∞ a-ε dx + ∫a+ε ∞ dx] 1
x-a f(x)
As an example, consider this case where a = 0 and f(x) = 1,
∫-2 2 dx 1
x = 0 .
Although 1/x "blows up" at x = 0, this integral as defi ned above is exactly 0. A ll contributions to this
integral are real, since 1/x is real, so the integral has no imaginary part. A simple argument for result zero
is that the integration range is ev en while the integrand is odd under x → -x, so contributions from the left
side of x=0 exactly cancel those from the right side of x=0. If the integrand contained some f(x) which was even under x → -x, the same argument would apply and the integral would be 0, but for general f(x)
the integral would not be 0.
Comment
: If one tries to evaluate the integral us ing ln(x), one gets a pre-limit result ln[a-ε
a+ε ] + ln[2
-2 ].
After the limit, the first term gives ln[1] = 0 while th e second term seems to give ln[-1] which one thinks
of as ±iπ and we get a contradictory result that the real in tegral of a real integrand has an imaginary part.
The problem is that this is a singular integral and the normal rules do not apply. The ±i π reflects the fact
that the integral is trying to avoid the pole by going above it or below it, whereas we really want to go
right through it. Stakgold Vol. I Exercise 1.23 shows how this contradiction is resolved using a redefined
log function, and the subject reappears below in our Pole Avoidance Rule discussion.
Appendix C: The Fourier and Hilbert Transforms
234 Here then are two notations used for integrals intended to run through poles,
∫-- ∞
-∞ dx 1
x-a f(x) ≡ P.V. ∫-∞ ∞ dx 1
x-a f(x) ≡ limε→0 [ ∫-∞ a-ε dx + ∫a+ε ∞ dx] 1
x-a f(x) . (C.9)
The tick mark on the integration symbol suggests the idea of running through the pole as the ε limit from
the two sides, but is not easy to typeset, so one often sees the letters P.V, PV, p.v., v.p. , P or some other
set of letters to indicate the principle part integrati on. We shall use the tick mark notation. Our example is
then
∫-- 2
-2 dx 1
x = 0 . ( C . 1 0 )
In the next several sections we shall examine some closely related examples of Fourier transforms, some of which require use of the Principal Value integral. The examples are later summarized in section (g).
(c) Example: f(u) = 1/u
Projection/Transform
:
Let f(u) = 1/u. Using the regular Fourier transform re quires that the integral go right through the pole, so
we have
(1/u)^( ω) = k
∫-∞ ∞ du(1/u) e-iωu = k ∫-- ∞
-∞ du(1/u) e-iωu . (C.11)
For ω ≠ 0 we can evaluate the integral this way,
∫-- ∞
-∞ du(1/u) e-iωu = ∫-- ∞
-∞ du(1/u) [-isin( ωu)] = (-i) 2 ∫0 ∞ du sin(ωu)/u
= (-i) 2 sign( ω) { ∫0 ∞ du sin(|ω|u)/u } = (-i) 2 sign( ω) { ∫0 ∞ dx sinc(x) } // x = | ω|u
= (-i) 2 sign( ω) { π/2 }
= - i π sign(ω) . ( C . 1 2 )
In the first step cos(ω u) was discarded since (1/u) cos( ωu) is an odd function of u. Once that is done, since
sinc(x) = 1 at x = 0, there is no longer a pole at u = 0 so we have just a regular integral. The residual
integral is half of (10.3).
If ω = 0, the integral is just
∫-- ∞
-∞ du(1/u) = 0 based on the discussion of the previous section. One can
combine these results by writing
∫-- ∞
-∞ du(1/u) e-iωu = -iπ sgn(ω) ( C . 1 3 )
Appendix C: The Fourier and Hilbert Transforms
235
where
sgn(ω) ≡
⎩⎨⎧ +1 ω > 0
0 ω = 0
-1 ω < 0 // sometimes called signum( ω) (C.14)
This sgn( ω) function is related to the Heaviside step function by
sgn(ω) = 2θ(ω) – 1 ( C . 1 5 )
where in particular sgn(0) = 2 θ(0) – 1 = 2(1/2)-1 = 0.
Our conclusion is that the Fourier transform of 1/u is given by
(1/u)^( ω) = -iπk sgn(ω) . ( C . 1 6 )
If we treat sgn( ω) like any other function, we can suppress the ω argument to write
(1/u)^ = -i πk s g n . ( C . 1 7 )
In the general case of f^(u) → f^ we could suppress the u, but once the function is stated (such as 1/u), it
is difficult to suppress the u and still know what the f unction is. Note that the u in 1/u is just a dummy
variable and we could just as well write (1/t)^(ω ) = -iπk sgn(ω)
(1/t)^ = -i πk s g n . ( C . 1 8 )
Inversion/Recovery
:
How does the recovery work?
(1/2πk)
∫-∞ ∞ dω (1/t)^(ω) e+iωt = (1/2πk) ∫-∞ ∞ dω [-iπksgn(ω)] e+iωt
= (1/2 π)(-iπ) ∫-∞ ∞ dω sgn(ω) e+iωt = (-i/2) ∫-∞ ∞ dω sgn(ω) [i sin(ωt]
= (-i/2) i 2 ∫0 ∞ dω sin(ωt) = ∫0 ∞ dω sin(ωt) = -(1/t)cos( ωt)|∞
0 = -(1/t) [ cos( ∞t) - cos(0)]
= ( 1 / t ) ( C . 1 9 )
The distributional trick is to set cos( ∞t) = 0. In more detail,
∫0 ∞ dω sin(ωt) = lim ε→0 [ ∫0 ∞ dω sin(ωt) e-εω ] = lim ε→0 t
t2+ε2 = 1
t . (C.20)
Appendix C: The Fourier and Hilbert Transforms
236
Interchange:
We have just shown that
(1/u)^( ω) = -iπ k sgn(ω) .
We can Fourier transform both sides to get [(1/u)^(ω )]^(t) = -i πk [sgn(ω)]^(t)
so that [sgn(ω)]^(t) = (1/-iπ k) [(1/u)^( ω)]^(t) .
But according to Fact 3,
[(1/u)^(ω )]^(t) = [(1/u)^]^(t) = (1/u)^^(t) = 2 πk
2 (1/-t) = -2 πk2/t .
Therefore
[sgn(ω)]^(t) = (1/-iπ k) (-2πk
2/t) = 2k/(it) = -2ik(1/t) . (C.21)
Thus we have learned the Fourier transform of the function sgn(ω ). We could have computed this directly
as follows:
[sgn(ω)]^(t) = k
∫-∞ ∞ du sgn(u) e-itu = k ∫-∞ ∞ du sgn(u) [-i sin(tu)] = (-i) 2k ∫0 ∞ du sin(tu)
= (-2ik)(-1/t)cos(tu)|∞
0 = (2ik/t)(0 - 1) = -2ik(1/t) .
(d) The Pole Avoidance Rule of Complex Integration
The upper pi
cture on the left shows a real-axis integration contour in the ω-plane which passes just below
a pole located at ω = +iε. We are interested in what happens as ε → 0. The upper picture on the right
shows the contour passing just above a pole at ω = -iε and we have a similar interest there as ε → 0. The
lower contour pictures show a certain contour deformation, and the pair of equa tions under each pair of
drawings will be discussed below.
Appendix C: The Fourier and Hilbert Transforms
237
∫-∞ ∞ dω f(ω) 1
ω – iε = ∫-- ∞
-∞ dω f(ω) (1/ω ) + iπ f(0) ∫-∞ ∞ dω f(ω) 1
ω + iε = ∫-- ∞
-∞ dω f(ω)(1/ω ) – iπ f(0)
1
ω – iε = pf(1/ ω) + iπδ(ω) 1
ω + iε = pf(1/ ω) – iπδ(ω)
F i g C . 1 We assume that f( ω) is such that the integrals converge and f( ω) is well defined at ω = 0. We are
interested only in the limit ε→0.
Left Side.
Consider the top left red arrow contour. If we try to take ε→ 0, the pole moves down and hits
the contour, which is a poorly defined concept in co mplex integration. To prevent this from happening,
we first make a tiny semi-circular defo rmation of the contour so it goes around ω = 0. One is certainly
allowed to deform a contour and not change an integral, as long as the deformation hits no singularities. After doing this "for free" deformation, we then let the ε→ 0 so the pole moves down to the real axis. If
we now evaluate the integral, we get two famous pi eces: The first piece is the principle value integral
discussed in section (b) above, namely,
∫-- ∞
-∞ dω f(ω) (1/ω ) ≡ limα→0 [ ∫-∞ -α + ∫α ∞ ] dω f(ω) (1/ω ) .
The second piece is a half-circle counterclockwise contour around the pole which gives one half the pole
residue which result is then (1/2) 2 πi f(0) = iπf(0). In general, if a contour goes some percentage around a
pole, it picks up that percentage of the reside, which we now demonstrate, letting ω = Reiθ ,
∫{dω
ω = ∫{Reiθidθ
Reiθ = i ∫θ1 θ2 dθ = i(θ2-θ1) . // partial residue rule
If we go half way around the pole, then i( θ2-θ1) = iπ. So we have now derived the top left equation in Fig
C.1.
Recall now from Appendix A that a distributiona l equation is one which gains its meaning when
placed inside an integral. So consider
∫-∞ ∞ dω f(ω) 1
ω – iε = ∫-∞ ∞ dω f(ω) [ pf(1/ω ) + iπδ(ω) ] = ∫-∞ ∞ dω f(ω) pf(1/ω ) + iπ f(0) .
Appendix C: The Fourier and Hilbert Transforms
238 Here both pf(1/ ω) and δ(ω) are symbolic functions as discuss ed in Appendix A. The meaning of δ(ω)
seems clear (sifting property) while the meaning of pf(1/ ω) is precisely this:
∫-∞ ∞ dω f(ω) pf(1/ω ) ≡ ∫-- ∞
-∞ dω f(ω) (1/ω ) .
The letters pf stand for pseudofunction. Officially f( ω) should be a distribution theory "test function", but
we just take it to be any reasonable function as described above. So we have now derived both equations
on the left of Fig C.1. Right Side.
This is the same idea, but the required contour deformation is different, and since the
semicircle then goes clockwise around the pole, we pick up minus half the residue which is – i πf(0). Now
both equations on the right are derived. We have then proven the following distributional equation:
The Pole Avoidance Rule:
lim ε→0 1
ω ∓ iε = pf(1/ω) ± iπδ(ω) ( C . 2 2 )
Notice that this rule has no connection with phase sign conventions of the Fourier transform. It really has nothing at all to do with the Fourier transform in fact. This "rule" seems to have no official name, so we
have made one up. For more discussion of this subject see Stakgold Vol. I page 50 (1.27) and previous
pages.
(e) Example: f(u) = 1/(u±i ε)
Here we alway
s imply the limit ε→0.
Projection/Transform:
Using the Pole Avoidance Rule (C.21) above, we ma y compute the Fourier tran sform of this f(u) as
follows:
(1
u±iε )^(ω) = k ∫-∞ ∞ du1
u±iε e-iωu = k ∫-- ∞
-∞ du(1/u) e-iωu ∓ iπk
The principal value integral was found in (C.13) to be -i π sgn(ω) , so we find that
(1
u±iε )^(ω) = –iπ k sgn(ω) ∓ iπk = -iπk (sgn(ω) ± 1) .
Assume first the upper signs,
(1
u+iε )^(ω) = –iπ k sgn(ω) – iπk = -iπk (sgn(ω) + 1)
Appendix C: The Fourier and Hilbert Transforms
239 If ω > 0, the result is -2 πik, and if ω < 0 the result is 0. So
(1
u+iε )^(ω) = -2πikθ(ω) .
Now assume the lower signs
(1
u-iε )^(ω) = –iπ k sgn(ω) + iπk = -iπk (sgn(ω) - 1)
If ω > 0, the result is 0. If ω < 0, the result is +2 πik. So
(1
u-iε )^(ω) = 2πikθ(-ω) .
Combining these results we get the Fourier transform of 1
u±iε
:
(1
u±iε
)^(ω) = ∓ 2πik θ(±ω) . ( C . 2 3 )
Inversion/Recovery:
(1/2πk) ∫-∞ ∞ dω (1
u±iε )^(ω) e+iωt = (1/2π) ∫-∞ ∞ dω [∓2πi θ(±ω)] e+iωt
= ( 1 / 2 π)(∓2πi) ∫-∞ ∞ dω θ(±ω) e+iωt = (∓ i) ∫-∞ ∞ dω θ(±ω) e+iωt
First take the upper sign
= ( – i)
∫0 ∞ dω e+iωt .
If t has a small positive imaginary part, the integral converges to (-1/it) to give
= ( – i) (-1/it) = 1/t
but we write t as t+i ε to show that it has this small positive imaginary part, so the result is
(1/2π)
∫-∞ ∞ dω (1
u+iε )^(ω) e+iωt = 1
t+iε
as desired. Now we look at the lower sign
Appendix C: The Fourier and Hilbert Transforms
240 ( +i) ∫-∞ 0 dω e+iωt = (+i) ∫0 ∞ dω e-iωt
If t has a small negative imaginary part, th e integral converges to (+1/it) to give
= ( +i) (+1/it) = 1/t = 1
t-iε
which is again the desired result.
Interchange:
We have just shown that
(1
u±iε
)^(ω) = ∓ 2πik θ(±ω)
where it is understood that ε → 0 on the left side, and either set of signs is valid.
We can Fourier transform both sides to get
[(1
u±iε
)^(ω)]^(t) = ∓ 2πik [θ(±ω)]^(t) .
But according to Fact 3,
[(1
u±iε )^(ω)]^(t) = [(1
u±iε )^]^(t) = (1
u±iε )^^(t) = 2 πk2 1
-t±iε .
Therefore
[θ(±ω)]^(t) = (∓ 2πik)
-1 2πk2 1
-t±iε = k
∓i 1
-t±iε = k
±it+ε
= ± i k 1
-t±iε = ±ik [ pf(-1/t) ∓ iπδ(-t) ] = ±ik [ -pf(1/t) ∓ iπδ(t) ] = ∓ i k pf(1/t) + π k δ(t)
and so we learn the Fourier transform of the Heaviside step function with either sign argument. The result with the + sign is verified immediately below. A more standard naming of the arguments yields
[θ(±t)]^(ω) = ±i k 1
-ω±iε = ∓ i k pf(1/ω) + π k δ(ω) = k pf( 1
±iω ) + π k δ(ω) . (C.24)
Appendix C: The Fourier and Hilbert Transforms
241 (f) Example: f(u) = θ(u) using the Gen eralized Fourier Transform
We include this example only because it is related to the previous examples.
Projection/Transform:
From the first line of (C.1) the projection (Fourier transform) is given by
[θ(u)]^(ω ) = k
∫-∞ ∞ du θ(u) e-iωu = k ∫0 ∞ du e-iωu .
The generalized Fourier transform was defined in Section 6. Recall that the recovery contour in the inversion formula passes below all singularities of f(u), so for this example that contour runs just below
the real axis so as to put the pole at u = 0 above the contour. Thus, we are really interested in the
projection evaluated at ω-iε, so we then have a convergent integral,
[θ(u)]^(ω -iε) = k
∫0 ∞ du e-i(ω-iε)u = k ∫0 ∞ du e-(iω+ε)u = k 1
iω+ε .
Taking the limit ε→0 we obtain the generalized Fourier transform of θ(u) valid for general complex ω,
[θ(u)]^(ω ) = k
iω ( C . 2 5 )
Inversion/Recovery:
We now evaluate the generalized Fourier inve rsion formula using the above projection,
(1/2πk)
∫-∞-iε ∞-iε dω f(ω) e+iωt = (1/2πk) ∫-∞-iε ∞-iε dω k
iω e+iωt
= ( 1 / 2 πi)
∫-∞-iε ∞-iε dω 1
ω e+iωt = θ(t) as explained below :
For t < 0, we close the contour down and pick up nothing giving 0. For t > 0, we close the contour up and pick up the full pole residue to get (1/2 πi)2πi= 1.
Appendix C: The Fourier and Hilbert Transforms
242 (g) Summary of Examples
Our conventi
on uses k = 1, but see (C.1) for other conventions.
Function/Distribution
Fourier transform Comment
x(t) X( ω) main text notation (k=1)
f(t) fh(ω) Appendix C notation
1/t -i πk sgn(ω) (C.16)
sgn(t) 2k
iω (C.21 )
limε→0 1
t±iε = pf(1/t) ∓ iπδ(t) ∓ 2πik θ(±ω) (C.23)
θ(±t) k lim ε→01
±iω+ε = k pf( 1
±iω ) + πk δ(ω) (C.24)
θ(t) k
iω (generalized FT) (C.25)
Comment : Looking back at (C.11) in light of the pf notation, the formally correct version of (C.11)
would be this
[pf(1/u)]^( ω) = k ∫-∞ ∞ du pf(1/u) e-iωu = k ∫-- ∞
-∞ du(1/u) e-iωu (C.11)
and then one would have this version of (C.18). [pf(1/t)]^( ω) = -iπk sgn(ω) . (C.18)
Most tables of Fourier transforms omit the pf formalit y since things are clear without such notation.
(h) The Hilbert Transform and its relation to the Fourier Transform
The Hilbert Transfor
m is defined as follows
fh(t) ≡ (1/π) ∫-- ∞
-∞ dω f(ω)
t-ω ≡ H[ f(x),t] = H[f,t] . (C.26)
Again we show several different nota tions, though we shall mainly use fh(t). Changing the integration
variable from ω to t' gives
fh(t) = ∫-- ∞
-∞ dt' (1/π)
t-t' f(t') = ∫-∞ ∞ dt' pf((1/π)
t-t' ) f(t') (C.27)
where we use the pf symbolic function introdu ced earlier. We recognize this as having the usual
convolution form (3.1),
Appendix C: The Fourier and Hilbert Transforms
243 a(t) = ∫-∞ ∞ dt' b(t-t')c(t') sometimes written a = b * c (3.1)
where a = b * c becomes,
f
h = pf( 1
πt ) * f . ( C . 2 8 )
The Convolution Theorem was stated in (3.6)
a(t) =
∫-∞ ∞ dt' b(t-t')c(t') ⇔ A(ω) = B(ω) C(ω ) (3.6)
or in our new notation, where for example a^( ω) = kA(ω),
a(t) = ∫-∞ ∞ dt' b(t-t')c(t') ⇔ a^(ω) = k-1 b^(ω) c^(ω) . (3.6)
Therefore we obtain this diagonalized form of (C.27) in ω-space,
[fh]^(ω) = k-1 ( 1
πt )^(ω) f^(ω) ( C . 2 9 )
But from (C.16) we know that (see Comment at the end of section g above)
( 1
πt )^(ω) = -i k sgn( ω) ( C . 3 0 )
Therefore (C.29) becomes (the scaling factor is now gone since both sides are Fourier transforms),
(fh)^(ω) = -i sgn( ω) f^(ω ) . ( C . 3 1 )
This well-known result relates the Fourier transform of the Hilbert transform of a function directly to the Fourier transform of that function. We now make immediate use of this equation. Since (C.31) can be applied to any function f (always assuming the Hilbert integral converges), we
apply it first to g,
[(g)
h]^(ω) = -i sgn( ω) (g)^(ω ) ( C . 3 2 )
and then we set g = f
h to get
[(f
h)h]^(ω) = -i sgn( ω) (fh)^(ω) . ( C . 3 3 )
Installing (C.31) into the right side gives
[f
hh]^(ω) = -i sgn( ω) [ -i sgn( ω) f^(ω )] = - f^(ω ) . ( C . 3 4 )
Appendix C: The Fourier and Hilbert Transforms
244
Applying the inverse Fourier transform to both sides then gives
f
hh(t) = -f(t)
or
fhh = - f ( C . 3 5 )
or in operator notation,
H H f = - f . ( C . 3 6 )
This says that application of the Hilbert transform tw ice to a function gives that function preceded by a
minus sign (compare to Fact 3). We can apply H
-1 to both sides to get
H-1 f = - H f ( C . 3 7 )
and, in so doing, we have discovered the formula for the inverse Hilbert transform,
H-1f(t) = - (1/π ) ∫-- ∞
-∞ dω f(ω)
t-ω ≡ fh-1(t) . (C.38)
We can then apply this last equation to fh instead of f to get
H-1fh(t) = - (1/π ) ∫-- ∞
-∞ dω fh(ω)
t-ω = (fh)h-1(t) = f(t) (C.39)
which gives us this Hilbert transform pair (in which the two members differ only by a sign),
f
h(t) = (1/π ) ∫-- ∞
-∞ dω f(ω)
t-ω // projection = transform
f(t) = - (1/π ) ∫-- ∞
-∞ dω fh(ω)
t-ω // inversion = recovery . (C.40)
We now make these replacements ω→ω ', f → X, fh → Xh, t→ω to get
Xh(ω) = (1/π ) ∫-- ∞
-∞ dω' X(ω')
ω-ω' // projection = transform
X(ω) = - (1/π ) ∫-- ∞
-∞ dω' Xh(ω')
ω-ω' // inversion = recovery (C.41)
and this is a more familiar statement of the Hilbert tr ansform pair. We have proved that this transform is
valid by making use of its connection to the Fourier transform shown in (C.31).
About 13 pages of Hilbert transforms appear in Erdélyi ET2.
Appendix C: The Fourier and Hilbert Transforms
245 Example 1 : Compute the Hilbert Transform of f( ω) = eiβω :
fh(t) = -(1/π ) ∫-- ∞
-∞ dω 1
ω-t eiβω = -(1/π) ∫-- ∞
-∞ dω' 1
ω' eiβ(ω'+t)
= -(1/ π) eiβt ∫-- ∞
-∞ dω' 1
ω' eiβω' = -(1/π) eiβt ∫-- ∞
-∞ dω' 1
ω' (i) sin(βω') // x = βω'
= -(i/ π) eiβt 2 sgn(β) ∫0 ∞ dx sinc(x) = -(i/ π) eiβt 2 sgn(β) { π/2 } // as in (C.12)
= -i sgn( β) eiβt
Therefore
f(ω ) = e
iβω ⇔ fh(t) = -i sgn( β)eiβt ( C . 4 2 )
or in the notation of the main text
X(ω) = e
iβω ⇔ Xh(ω) = -i sgn( β)eiβω ( C . 4 3 )
Example 2 : Equation (C.42) is valid for any β, so it is valid for - β. Since the Hilbert transform is linear
we can then superpose exponentials to get the following correspondences,
f(ω ) = k Σ β gβ e-iβω ⇔ fh(t) = -ik Σ β sgn(-β ) gβ e-iβt
f(ω ) = k ∫-∞ ∞ dβ g(β) e-iβω ⇔ fh(t) = -ik ∫-∞ ∞ dβ sgn(-β) g(β) e-iβt
The last line can be written
f(ω ) = g^(ω) ⇔ fh(t) = +i [ sgn( β)g(β) ]^(t)
which then says
[g^(ω)]
h(t) = +i [ sgn( β)g(β) ]^(t) .
If we suppress ω, then change t to ω, then β to t, and replace function name g by f, this says
(f^)h(ω) = +i [ sgn(t)f(t) ]^( ω) ( C . 4 4 )
which we can compare with (C.31) which has the transforms in the reverse order
(f
h)^(ω) = -i sgn( ω) f^(ω ) . (C.31)
Appendix C: The Fourier and Hilbert Transforms
246 Alternate derivation of (C.44).
Start with (C.26) applied to f^ ,
(f^)
h(t) ≡ (1/π) ∫-- ∞
-∞ dω f^(ω)
t-ω
or
(f^)h = pf( 1
πt ) * f^
The diagonalized version is then, using (C.30), a^(ω) = k
-1 b^(ω) c^(ω)
or
((f^)h)^ = k-1 [-i k sgn( ω)] f^^(ω) = -i sgn( ω) f^^(ω ) .
Use Fact 3 that f^^( ω) = 2πk
2 f(-ω) to get
((f^)
h)^(ω) = -i sgn( ω) 2πk2 f(-ω) = 2π k2 i sgn(-ω) f(-ω )
Now take the Fourier transform of both sides
((f^)h)^^(ω ) = 2πk2i [sgn(-s) f(-s)]^( ω)
Use Fact 3 that ((f^)
h)^^(ω ) = 2πk2((f^)h)(-ω) to get
2πk
2((f^)h)(-ω) = 2πk2i [sgn(-s) f(-s)]^(ω )
or ((f^)
h)(-ω) = i[sgn(-s) f(-s)]^( ω)
or ((f^)
h)(ω) = i [sgn(-s) f(-s)]^(- ω)
Finally we get to use Fact 0 that [f(-u)]^(- ω) = f^(ω ) to obtain the final result
((f^)h)(ω) = i [sgn(s) f(s)]^( ω)
which is (C.44).
Appendix D: Probability Theory
247 Appendix D: Probability Theory: how α and β are related to μ and σ
We first review
some of the basics of random variab les X and Y: expectation values, means, covariance,
variance and standard deviation. We do this in a cont ext in which each variable takes on a set of discrete
values rather than continuous values. We then translate our various facts into the language of X,Y → Ym
Yn where the latter are random variables associated with the amplitudes at positions m and n of a pulse
train. The labels m and n add a notational complexity that is a little difficult to comprehend unless things are written out explicitly, which is done below.
(a) Random Variables X and Y
If X and Y are rando
m variables which can take discrete values x k and yk, one can express the mean
value of each variable as
μ
x = E(X) = Σk xk px(xk)
μy = E(Y) = Σk' yk' py(yk') ( D . 1 )
where E(X) means the expectation value of X. The sum Σk is over all values in the set of values that X
can take, and Σk' is over all values that Y can take. Probability p x(xk) is the probability that X takes the
value xk and similarly for p y(yk'). If we have a very large ensemble of I systems we can measure the
above quantities in this manner,
μx = E(X) = (1/I) Σi=1I x(i)
μy = E(Y) = (1/I) Σi=1I y(i) ( D . 2 )
where i labels a particular system in the ensemble, and x
(i) and y(i) are the values X and Y take in
ensemble system i. The probability functions like p x(xk) do not appear in (D.2), but they are present in
spirit, being embedded in the "experimental data" like x(i).
The expectation value E(XY) is defined by
E(XY) = ΣkΣk' xk yk' pxy(xk,yk') ( D . 3 )
where now p
xy(xk,yk') is the joint probability that X has value x k when Y has the value y k'. One would
measure E(XY) in this manner,
E(XY) = (1/I) Σ i=1I x(i) y(i) . ( D . 4 )
If the random variables X and Y are independent (no correlation), then
p
xy(xk,yk') = px(xk) py(yk') ( D . 5 )
and then it is obvious from the above definitions of E(XY), E(X) and E(Y) that E ( X Y ) = E ( X ) E ( Y ) . ( D . 6 )
Appendix D: Probability Theory
248 The covariance of X and Y is defined as,
cov(X,Y) = E( [X - μx] [Y - μy] ) = ΣkΣk' (xk - μx) (yk'- μy) pxy(xk,yk') (D.7)
and it could be measured for an ensemble in this way
cov(X,Y) = (1/I) Σi=1I (x(i)
- μx) (y(i)- μy) (D.8)
where μ
x and μy would be the measured values of E(X) and E(Y) as shown above. If X and Y are
independent variables then we use (D.5) in (D.7) to find that
cov(X,Y) = E( [X - μx] [Y - μy] ) = E(X - μx) E(Y - μy) . (D.9)
In the special case that X and Y are the same variable, we write cov(X,Y) = var(X), the variance of X, which is also defined as the squa re of the standard deviation σ
x,
var(X) ≡ σx2 ≡ cov(X,X) = ΣkΣk' (xk - μx) (xk'- μx) pxx(xk,xk') . (D.10)
But p
xx(xk,xk') = δk.k'px(xk) because if X has value x k, it cannot also have a different value x k', so
p
xx(xk,xk') = δk,k' px(xk) ( D . 1 1 )
and then
var(X) ≡ σx2 ≡ cov(X,X) = Σk (xk - μx)2 px(xk) ( D . 1 2 )
with experimental measurement
var(X) ≡ σ
x2 ≡ cov(X,X) = (1/I) Σi=1I (x(i)
- μx)2 . (D.13)
Finally, we note that cov(X,Y) = E( [X - μ
x] [Y - μy]) = E(XY) - μ xE(Y) - μyE(X) + μ xμyE(1)
= E(XY) - μxμy - μyμx + μxμy
= E(XY) - μ
xμy . ( D . 1 4 )
When X and Y are the same random variable this says
var(X) ≡ σ
x2 ≡ cov(X,X) = E(X2) - μx2 . ( D . 1 5 )
Appendix D: Probability Theory
249 (b) Application to the Pulse Train Ensemble
In the above
we set
X = Y m = random variable associated with the amplitudes y m(i) at pulse location m in pulse train
Y = Y n = random variable associated with the amplitudes y n(i) at pulse location n in pulse train
Things are a little complicated due to the extra labels m and n, but we can nevertheless carefully translate
all the X and Y equations above. Also, we shall add the following extra notation for an expectation value
to be consistent with our earlier work,
<x> ≡ E(X) <xy> ≡ E(XY) <x
2> ≡ E(X2) . (D.16)
The first pair of equations above translate into the following:
μym = E(Ym) = <ym> = Σk (ym)k pym[(ym)k] = Σk (ym)k p[(ym)k] = μ
μyn = E(Yn) = <yn> = Σk (yn)k pyn[(yn)k] = Σk (ym)k p[(ym)k] = μ . (D.17)
Here (y
m)k is the kth value in the set of all possible amplitudes that position m in the pulse train can have.
For example, we might have (y m)1 = A and (y m)2 = B. For the pulse train, the set of allowed amplitude
values is the same for any pulse in the train, and the probability distribution is also the same, yielding the simplifications shown above. In particular, the means μ
ym and μyn don't depend on position m and n, so
we can just call both these means μ. Here is how the above means would be measured for an ensemble,
μ = μ
ym = E(Ym) = <ym>= (1/I) Σ i=1I ym(i)
μ = μyn = E(Yn) = <yn>= (1/I) Σ i=1I yn(i) . (D.18)
Next, for E(XY) we translate (D.3) to get ,
E(Y
mYn) = <ymyn> = ΣkΣk' (ym)k (yn)k' pym,yn [(ym)k, (yn)k']
E(Y
mYn) = <ymyn> = (1/I) Σi=1I ym(i) yn(i) . ( D . 1 9 )
If Y
m and Yn are uncorrelated (which requires that m ≠n ) then in analogy with (D.5),
p
ym,yn [(ym)k, (yn)k'] = pym[(ym)k] pyn[(yn)k] = p[(y m)k] p[(yn)k] (D.20)
and as in (D.6) we find that, in the case of no correlation,
E(Y
mYn) = <ymyn> = E(Y m) E(Yn) = <ym><yn> . m ≠ n (D.21)
Obviously if m = n, then Y m and Yn cannot be "uncorrelated" since they are the same random variable. If
m = n, then we have instead this version of (D.11),
p
ym,ym [(ym)k, (ym)k'] = δk,k' pym[(ym)k] = δk,k' p[(ym)k] ( D . 2 2 )
Appendix D: Probability Theory
250
and then
E(Y
m2) = <ym2> = Σk [(ym)k]2 p[(ym)k] ( D . 2 3 )
which in experiment (over the ensemble) would be measured as
E(Y m2) = <ym2> = (1/I) Σi=1I (ym(i))2 . ( D . 2 4 )
The covariance equations translat ed from (D.7) and (D.8) become
cov(Y m, Yn) = E( [Y m - μ] [Yn - μ] ) = ΣkΣk' ((ym)k - μ) ((ym)k'- μ) pym,yn [(ym)k, (yn)k']
cov(Y
m, Yn) = (1/I) Σi=1I (ym(i)
- μ) (yn(i)- μ) . ( D . 2 5 )
The variance (and standard deviation) transl ations from (D.12) and (D.13) are.
var(Y m) ≡ σ2 ≡ cov(Ym, Ym) = Σk [(ym)k - μ]2 p[(ym)k]
var(Y
m) ≡ σ2 ≡ cov(Ym, Ym) = (1/I) Σi=1I (ym(i) - μ)2 . (D.26)
Finally, (D.14) and (D.15) become, cov(Y
m, Yn) = E(Y m Yn) - μyxμyn = E(Ym Yn) - μ2
var(Y m) ≡ σ2 ≡ cov(Ym, Ym) = E(Y m2) - μ2 = <ym2> - μ2 => <y m2> = σ2+μ2 . (D.27)
(c) Main conclusions for the Pulse Train Ensemble
The following quantities are
measured from the pulse train ensemble as follows,
<ym> = (1/I) Σi=1I ym(i) = μ (D.18)
<y
myn> = (1/I) Σ i=1I ym(i) yn(i) any m and n (D.19)
<y
m2> = (1/I) Σi=1I [ym(i]]2 = σ2 + μ2 (D.27) (D.28)
where we show how each quantity is related to symbols μ and σ. When the amplitudes at different
locations in the pulse train are uncorrelated, we can further write
<y
myn> = <ym><yn> = μ2 m ≠n . (D.21) (D.29)
We can now restate equation (35.6) in th is more statistically-oriented manner,
<ym> = μ = [p]A + [1-p]B
Appendix D: Probability Theory
251
n ≠ m α ≡ <ymyn> = μ2 = { [p]A + [1-p]B }2 = [pp] AA + [p(1-p)] 2AB + [(1-p)(1-p)]BB
n = m β ≡ <y
m2> = [p]AA + [(1-p)] BB = σ2 + μ2 ( D . 3 0 )
which is to say
α = μ2
β = μ2 + σ2 = α + σ2
σ2 = (β -α) . ( D . 3 1 )
Using our expressions for α and β we can compute σ
2:
which says
σ2 = variance = [ p(1-p)] (A-B)2 . ( D . 3 2 )
If p = 0 or p = 1, this variance vanishes as one would expect.
Appendix E: Table of Transforms
252 Appendix E: Table of Transforms
The following transforms appear in this docum
ent:
Fourier Integral Transform: x(t) aperiodic and continuous, X( ω) continuous, no image spectra
X(ω) =
∫-∞ ∞ dt x(t) e-iωt projection = transform (1.1)
x(t) = (1/2 π) ∫-∞ ∞ dω X(ω) e+iωt expansion = inverse transform (1.2)
Generalized Fourier Integral Transform: recovery contour passes below all singularities of X( ω)
X(ω) = ∫0 ∞ dt x(t) e-iωt projection = transform (6.4)
x(t) = (1/2 π) ∫-ci-∞ -ci+∞ dω X(ω ) e+iωt expansion = inverse transform (6.5)
Fourier Cosine Transform:
Xc(ω) = 2 ∫0 ∞ dt x(t) cos(ω t) projection = transform
x(t) = (1/ π) ∫0 ∞ dω Xc(ω) cos(ωt) expansion = inverse transform (1.7)
Fourier Sine Transform:
Xs(ω) = 2 ∫0 ∞ dt x(t) sin( ωt) projection = transform
x(t) = (1/ π) ∫0 ∞ dω Xs(ω) sin(ωt) expansion = inverse transform (1.8)
Fourier Integral Transform in Appendix C notation : k is an arbitrary convention constant
x^(ω) = k ∫-∞ ∞ dt x(t) e-iωt projection = transform
x(t) = (1/2 πk) ∫-∞ ∞ dω x^(ω) e+iωt expansion = inverse transform
f^(ω ) = k ∫-∞ ∞ du f(u) e-iωu Fourier transform of f(u)
f^-1(t) = (1/2πk) ∫-∞ ∞ du f(u) e+iut inverse Fourier transform of f(u) (C.1)
Appendix E: Table of Transforms
253 Fourier Series Transform: x(t) periodic and continuous with period T 1, spectrum is discrete
x(t) = ∑
n = -∞∞
xpulse (t - nT1) (14.1)
Complex form:
cm ≡ (1/T1) ∫-∞ ∞ dt xpulse (t) e-imω1t = (1/T 1) ∫0 T1 dt x(t) e-imω1t (14.16)
x(t) = ∑
m = -∞∞
cm e+imω1t (15.1)
cm = c(mω1) = (1/T1)Xpulse (mω1) (14.8) and (14.10)
Real form
:
a m ≡ (2/T1) ∫-∞ ∞ dt xpulse (t) cos(mω1t) = (2/T 1) ∫0 T1 dt x(t) cos(m ω1t) (15.6)
b m ≡ (2/T1) ∫-∞ ∞ dt xpulse (t) sin(mω1t) = (2/T 1) ∫0 T1 dt x(t) sin(m ω1t) (15.7)
x ( t ) = a 0/2 + ∑
m = 1∞
am cos(mω1t) + ∑
m = 1∞
bm sin(mω1t) (15.9)
Laplace Transform: Right-sided (causal) x(t) vanishes for t < 0, x(t) and X(s) are continuous
X(s) = ∫0 ∞ dt x(t) e-st projection = transform (6.9)
x(t) = (1/2 πi) ∫c-i∞ c+i∞ ds X(s) e+is expansion = inverse transform (6.10)
Relation to the Fourier Integral Transform:
X(s) = X(s/i) X(ω ) = X(iω) (6.8)
Appendix E: Table of Transforms
254 Digital Fourier Transform: x(tn) aperiodic, spectrum X'( ω) is continuous and contains image spectra
X'(ω) ≡ ∑
n = -∞∞
∆t x(tn) e-iωtn projection = transform (22.2)
x(tn) = 1
2π ∫
-ω1/2 ω1/2
dω X'(ω) e+iωtn expansion = inversion (22.4)
or X'(ω) ≡ T
1∑
n = -∞∞
xn e-iωnT1 projection = transform x n = x(tn).
xn = 1
2π ∫
-ω1/2 ω1/2
dω X'(ω) e+iωnT1 expansion = inversion
where: T 1 = ∆t ω1 = 2π/T1 = 2π/∆t t n = n ∆t = n T 1 .
Relation to the Fourier Integral Transform:
X'(ω) =
∑
m = -∞∞
X(ω - mω1) = [ X( ω) + ∑
m ≠ 0
X(ω - mω1) ] // image spectra (23.1)
Z Transform: x(tn) = xn is aperiodic, spectrum X"(z) is continuous and contains image spectra
X"(z) = ∑
n = -∞∞
xn z-n projection = transform (24.3)
xn = 1
2πi ∫C dz X"(z) zn-1 expansion = inversion (24.4)
where contour C goes once counterclockwise around the unit circle in the z-plane. Relation to the Digital Fourier Transform and the Fourier Integral Transform:
X"(z) ≡ 1
T
1 X'(ω) = 1
T1 ∑
m = -∞∞
X(ω - mω1) z = eiωT1 (24.1), (24.2)
Appendix E: Table of Transforms
255 Discrete Fourier Transform of a Pulse Train : x(tn) periodic, spectrum discrete, m integer
x(tm) = ∑
n = -∞∞
xpulse (tm - nT1) sampled pulse train, t m = (m/N)T 1
c'm ≡ (1/N) ∑
n = -∞∞
xpulse (tn) e-imn(2π/N) projection = transform (27.9)
x(tn) = ∑
m = 0N-1
c'm e+imn(2π/N) expansion = inverse transform (27.11), (27.12)
Discrete Fourier Transform (DFT) : A is an arbitrary convention constant
c'm ≡ (A/N) ∑
n = 0N-1
xn e-imn(2π/N) m = 0,1...N-1 projection = transform
xn = (1/A) ∑
m = 0N-1
c'm e+imn(2π/N) n = 0,1,...N-1 expansion = inverse transform (27.22)
Hilbert Transform:
Xh(ω) = (1/π ) ∫-- ∞
-∞ dω' X(ω')
ω-ω' projection = transform
X(ω) = - (1/π ) ∫-- ∞
-∞ dω' Xh(ω')
ω-ω' expansion = inverse transform (C.41)
Appendix F: Repeating Subsequences
256 Appendix F: The Spectrum and Power Density for Repeated-Sequence Pulse Trains
Overview: F
or infinite pulse trains composed of repeats of some length-P subsequence:
(a) computes X( ω) in (F.12)
(b) computes P(ω) in (F.23)
(c) computes < P (ω)> for an ensemble of pulse trains which respect the special condition
< y
m* yn> = α for m ≠ n
< y m* yn> = β for m = n s < |m-n| (F.25)
The result for < P(ω)> is stated in (F.33) in several different forms.
(d) takes the P →∞ limit of the section (c) result for < P(ω)>
(e) computes P(ω) for a single pulse train which respects the special condition
< y myn>1 = α for m ≠ n + NP N = any integer
< y myn>1 = β f o r m = n + N P (F.43)
where <y myn>1 is a horizontal average across the single sequence (autocorrelation).
The result for P(ω) is stated in (F.52).
It is noted that the results for < P(ω)> of (c) and P(ω) of (e) are exactly the same in terms of their
respectively defined α and β constants.
(f) restates the results of (c) and (e) as Fact 1 (F.53) and Fact 2 (F.54).
A graphical representation is drawn for the spectrum in general, and for a box pulse.
It is shown that the MLS sequence is a candidate for application of Fact 2. A simple ensemble is then constructed to which Fact 1 maybe applied, and the result is then interpreted. (g) treats the P = 2 repeated subsequence A, B using the general formulas of (a) and (b)
We start with this collection of equations:
Y"(z) =
∑
n = -∞∞
yn e-iωnT1 Z Transform of y n (24.2) (F.1)
| Y"(z) |2 = ∑
n = -∞∞
∑
m = -∞∞
ym* yn eiω(m-n)T1 ( F . 2 )
X(ω) = Xpulse (ω) Y"(z) (25.3) (F.3)
P(ω) = Ppulse (ω) T1
T | Y"(z) |2 (34.14) (F.4)
z = eiωT1 (24.1)
ω1 ≡ 2π/T1
In (F.4) T is the duration of the infinite pulse trai n, as in (33.22). The pulse train amplitudes are the y
n.
Since sequence y m is composed of subsequences of length P that repeat, we have this periodicity property
of the yn
Appendix F: Repeating Subsequences
257
ym+IP = ym for any integer I (F.5)
(a) Calculation of X( ω) for a Pulse Train with a R epeated Sequence
Consider first the sum in (F.1). Let n = IP + n' and write this sum as
∑
n = -∞∞
yn e-iωnT1 = ∑
I = -∞∞
∑
n' = 0P-1
yIP+n' e-iω(n'+IP)T 1 = ∑
I = -∞∞
e-iωIPT1 ∑
n' = 0P-1
yn' e-iωn'T1
= (
∑
I = -∞∞
e-iωIPT1) (∑
n = 0P-1
yn e-iωnT1) . (F.6)
We see that the sum factors into the product of two sums. The first sum we evaluate using
∑
n = -∞∞
e-ink = ∑
m = -∞∞
2πδ(k - 2πm) . - ∞ < k < ∞ (A.31)
Setting k = ωPT1 we find
( ∑
I = -∞∞
e-iωIPT1) = ∑
m = -∞∞
2πδ(ωPT1 - 2πm ) . ( F . 7 )
The second sum we give the name Y P"(z) which is the Z transform of the subsequence { y 0, y1.....yP-1}.
YP"(z) ≡ ∑
n = 0P-1
yn e-iωnT1 . ( F . 8 )
Thus we have shown that
Y"(z) =
∑
n = -∞∞
yn e-iωnT1 = YP"(z) ∑
m = -∞∞
2πδ(ωPT1 - 2πm) , (F.9)
and then from (F.3).
X(ω) = X
pulse (ω) Y"(z) = X pulse (ω) YP"(z) ∑
m = -∞∞
2πδ(ωPT1 - 2πm) . (F.10)
It is convenient to write
2πδ(ωPT1 - 2πm) = 2π(PT1)-1δ(ω - 2πm/PT1) = (1/P) ω1 δ(ω - mω1/P) (F.11)
Appendix F: Repeating Subsequences
258 and then
X(ω) = Xpulse (ω) ω1 (1/P) YP"(z) ∑
m = -∞∞
δ(ω - mω1/ P ) ( F . 1 2 )
where Y P"(z) ≡ ∑
n = 0P-1
yn e-iωnT1 . (F.8)
X(ω) is the Fourier Transform Spectrum of an infin ite pulse train composed of a repeating P-length
subsequence. Since the pulse train is periodic, the spectrum is entirely discrete with lines at
ωm = (m/P)ω1. ( F . 1 3 )
(b) Calculation of P (ω) for a Pulse Train with a Repeated Sequence
We can reorganize the dou
ble sum in (F.2) into a quadruple sum by defining :
n = IP + n' m = JP + m' . Then the double sum above becomes,
∑
n = -∞∞
∑
m = -∞∞
ym* yn eiω(m-n)T1
= ∑
I = -∞∞
∑
J = -∞∞
∑
n' = 0P-1
∑
m' = 0P-1
(yJP + m' )* (yIP +n' ) eiωP(I-J)T1 eiω(m'-n')T 1
= ∑
I = -∞∞
∑
J = -∞∞
∑
n' = 0P-1
∑
m' = 0P-1
(ym')* (yn') eiωP(I-J)T1 eiω(m'-n')T 1 , (F.14)
where in the last line we have u sed the periodicity (F.5) of the y n. The following illustration shows how,
in this reorganization, we first sum over a square grid patch with n' and m', and then we sum over an array
of those patches with I and J.
Appendix F: Repeating Subsequences
259
Fig F.1
In order to regulate things, we shall assume that the I and J sums range from -N to N rather than from - ∞
to ∞. This means we are assuming that the sequence is (2 N+1) repeated periods in length and not infinite.
Then of course we can say
T = (2N+1)PT
1 . ( F . 1 5 )
As usual, we keep N finite as long as possible, and then take N → ∞ in the end.
We now rewrite (F.2) by removing the primes from summation indices and reordering the factors
| Y"(z) |
2 = ∑
n = -∞∞
∑
m = -∞∞
ym* yn eiω(m-n)T1 →
= [ ∑
I = -NN
∑
J = -NN
eiωP(I-J)T1 ] [ ∑
n = 0P-1
∑
m = 0P-1
ym* yn eiω(m-n)T1 ]
= [ ∑
I = -NN
∑
J = -NN
eiωP(I-J)T1 ] | YP"(z) |2 ( F . 1 6 )
As happened in (F.6) with a single sum, our double sum factors into a product of two double sums. The
second double sum we recognize from (F.8) as | Y P"(z) |2. The first double sum is
[ ∑
I = -NN
∑
J = -NN
eiωP(I-J)T1 ] = | ∑
I = -NN
eiωPIT1 | 2 ( F . 1 7 )
Then apply (A.30)
Appendix F: Repeating Subsequences
260 ∑
n = -NN
eink = 2π δ5(k,N) = 2 π { sin[(N+1/2)k]
2π sin(k/2) } (A.30)
with k = ωPT1 to get
[ ∑
I = -NN
∑
J = -NN
eiωP(I-J)T1 ] = | 2π δ5(ωPT1,N) | 2 = ( 2π δ5(ωPT1,N) ) 2 . (F.18)
Then from (A.20),
δ
6(k,N) ≡ 1
2π [2πδ5(k,N)]2
(2N+1) = 1
2N+1 sin2[(N+1/2)k]
2π sin2(k/2) , (A.20)
we can write the first factor of (F.16) as
[
∑
I = -NN
∑
J = -NN
eiωP(I-J)T1 ] = (2N+1) 2π δ6(ωPT1,N) = (1/P) T
T1 2π δ6(ωPT1,N) (F.19)
where T is from (F.15). At this point we have, looking at (F.16) and (F.19),
| Y"(z) |2 = [ ∑
I = -NN
∑
J = -NN
eiωP(I-J)T1 ] [ ∑
n = 0P-1
∑
m = 0P-1
ym* yn eiω(m-n)T1 ]
= [ (1/P) T
T1 2π δ6(ωPT1,N) ] | Y P"(z) |2 . ( F . 2 0 )
Now finally we take N→∞ using (A.21)
limN→∞ δ6(k,N) = ∑
m = -∞∞
δ(k-2πm ) (A.21 )
or lim
N→∞ δ6(ωPT1,N) = ∑
m = -∞∞
δ(ωPT1-2πm ) ( F . 2 1 )
with this result
| Y"(z) |2 = [ (1/P) T
T1 2π∑
m = -∞∞
δ(ωPT1-2πm) ] | Y P"(z) |2
or T
1
T | Y"(z) | = (1/P) ∑
m = -∞∞
2π δ(ωPT1-2πm) | YP"(z) |2
Appendix F: Repeating Subsequences
261 Using (F.11) we can rewrite this as
T1
T | Y"(z) | = ω 1 (1/P)2 ∑
m = -∞∞
δ(ω - mω1/P) | YP"(z) |2 . ( F . 2 2 )
Installing this into (F.4) then gives
P(ω) = P
pulse (ω) ω1 (1/P)2 ∑
m = -∞∞
δ(ω - mω1/P) | YP"(z) |2 ( F . 2 3 )
where Y P"(z) ≡ ∑
n = 0P-1
yn e-iωnT1 . (F.8)
P(ω) is the Spectral Power Density of an infinite pulse train composed of a repeating P-length
subsequence. The power density is discrete with lines at ωm = (m/P)ω1, the same lines observed in the
spectrum X( ω) of (F.12).
(c) Calculation for an Ensemble of such Pulse Trains subject to Certain Conditions
We now ima
gine an ensemble of P-length subsequences s i. We then create a corresponding ensemble of
infinite length sequences S i according to S i = {.......s i, si, si, si, ...}. This just an arbitrary ensemble,
not a random ensemble or any other special kind ensemble. Then we apply the ensemble average <..> to (F.23) to get
< P(ω)> = P
pulse (ω) ω1 ∑
m = -∞∞
δ(ω - ω1m/P) 1
P2 ∑
n = 0P-1
∑
m = 0P-1
<ym* yn> eiω(m-n)T1 (F.24)
At this point, suppose it happens that
<ym* yn> = α for m ≠ n
<y
m* yn> = β f o r m = n ( F . 2 5 )
where α and β are independent of the indices shown. Notice in the (F.24) sum that max(n-m) = P-1, so we
don't have to worry about these indices differing by an integral multiple of P. We have now restricted our
interest to the sequence {y 0,y1, ...yP-1}.
Comment
: In the case of a random ensemble, (F.25) would be true if the random variables Y m and Yn
associated with pulse train locations m and n were uncorrelated, but saying that α and β are independent
of the indices covers a larger class of possibilities, including some in which Y m and Yn are in fact
correlated. In other words, saying that α and β as shown above don't depend on the indices m and n does
not imply that <y m* yn> = <ym*>< yn> which is the definition of being uncorrelated.
Appendix F: Repeating Subsequences
262 Then we can write
∑
n = 0P-1
∑
m = 0P-1
<ym* yn> eiω(m-n)T1 = α ∑
n = 0P-1
∑
m≠n eiω(m-n)T1 + β ∑
n = 0P-1
1
= α ∑
n = 0P-1
∑
m≠n eiω(m-n)T1 + βP . ( F . 2 6 )
To evaluate the double sum, we write it as
∑
n = 0P-1
∑
m≠n eiω(m-n)T1 = ∑
n = 0P-1
∑
m = 0P-1
eiω(m-n)T1 – ∑
n = 0P-1
∑
m=n eiω(m-n)T1
= | ∑
n = 0P-1
e-iωnT1 |2 – ∑
n = 0P-1
1 = | ∑
n = 0P-1
e-iωnT1 |2 – P . (F.27)
Now let ω T1 = k and consider
∑
n = 0P-1
e-ink = ∑
n = 0P-1
(e-ik)n = ∑
n = 0P-1
xn = 1 + x + ... + xP-1 = 1-xP
1-x
= 1-e-ikP
1-e-ik . ( F . 2 8 )
With a Maple assist,
Appendix F: Repeating Subsequences
263 we see that
| ∑
n = 0P-1
e-ink |2 = | 1-e-ikP
1-e-ik |2 = sin2(kP/2)
sin2(k/2) . ( F . 2 9 )
But from (A.20)
δ
6(k,N) ≡ 1
2π [2πδ5(k,N)]2
(2N+1) = 1
2N+1 sin2[(N+1/2)k]
2π sin2(k/2) . (A.20)
setting P = 2N+1 we recognize sin
2(kP/2)
sin2(k/2) as one of our multiple delta function models, so then
| ∑
n = 0P-1
e-ink |2 = | 1-e-ikP
1-e-ik |2 = sin2(kP/2)
sin2(k/2) = 2π P δ6(k, P-1
2 ) . (F.30)
If we set k = ωT1, we can insert (F.30) into (F.27) with this result
∑
n = 0P-1
∑
m≠n eiω(m-n)T1 = | ∑
n = 0P-1
e-iωnT1 |2 – P = 2 π P δ6(ωT1, P-1
2 ) - P
= P [ 2 π δ6(ωT1, P-1
2 ) - 1] . (F.31)
We now go back to (F.24)
< P(ω)> = P
pulse (ω) ω1 ∑
m = -∞∞
δ(ω - ω1m/P) 1
P2 ∑
n = 0P-1
∑
m = 0P-1
<ym* yn> eiω(m-n)T1 (F.13)
and replace the double sum using (F.26) to get
= Ppulse (ω) ω1 ∑
m = -∞∞
δ(ω - ω1m/P) 1
P2 [α { ∑
n = 0P-1
∑
m≠n eiω(m-n)T1 } + βP ] . (F.32)
Then we use (F.31) for {...} to get
= Ppulse (ω) ω1 ∑
m = -∞∞
δ(ω - ω1m/P) 1
P2 [α { P [2π δ6(ωT1, P-1
2 ) - 1] } + βP ]
= Ppulse (ω) ω1 ∑
m = -∞∞
δ(ω - ω1m/P) 1
P [α { [2π δ6(ωT1, P-1
2 ) - 1] } + β ]
or
Appendix F: Repeating Subsequences
264 < P(ω)> = Ppulse (ω) ω1 ∑
m = -∞∞
δ(ω - ω1m/P) 1
P [(β-α) + α 2π δ6(ωT1, P-1
2 ) ] (F.33a)
where
δ 6(k,N) ≡ 1
2N+1 sin2[(N+1/2)k]
2π sin2(k/2) => 2 π δ6(ωT1, P-1
2 ) = 1
P sin2(PωT1/2)
sin2(ωT1/2) .
Evaluating at the delta function hit values ω = ω1m/P we find
2π δ6 = 1
P sin2(mπ)
sin2(mπ/P) = ⎩⎨⎧ 0 m ≠ NP
P m = NP for N = any integer .
To get alternate forms for (F.33a), we first express each term as a separate sum,
< P(ω)> = P
pulse (ω) ω1 1
P [(β-α)∑
m = -∞∞
δ(ω - ω1m/P) + α ∑
m = -∞∞
δ(ω - ω1m/P) 2πδ6(ωT1, P-1
2 ) ] .
Since only those m which are multiples of P contribute to the second sum in (F.33a), we may rewrite that
second sum as follows,
<P(ω)> = P
pulse (ω) ω1 1
P [(β-α)∑
m = -∞∞
δ(ω - ω1m/P) + α ∑
N = -∞ ∞
δ(ω - ω1N) P ] // m = NP
= Ppulse (ω) ω1 [(β-α) 1
P ∑
m = -∞∞
δ(ω - ω1m/P) + α ∑
m = -∞∞
δ(ω - ω1m) ] (F.33b)
= Ppulse (ω) ω1 ∑
m = -∞∞
[ (β-α) 1
P δ(ω - ω1m/P) + α δ(ω - ω1m) ] (F.33c)
= Ppulse (ω) ω1 [ (β-α) 1
P ∑
m ≠ 0
δ(ω - ω1m/P) + α ∑
m ≠ 0
δ(ω - ω1m) + { (β-α) /P + α } δ(ω) ] .
(F.33d)
If xpulse (t) is real, then by (7.4) Ppulse (ω) is an even function of ω, and we can then reflect the negative
part of the sum to the positive side to get,
= Ppulse (ω) ω1 [ (β-α) 2
P ∑
m = 1 ∞
δ(ω - ω1m/P) + 2α ∑
m = 1 ∞
δ(ω - ω1m) + { (β-α) /P + α } δ(ω) ] .
( F . 3 3 e )
In all forms of (F.33) we have: α = <ym* yn> for m≠n β = <yn* yn> .
These slightly different forms of < P(ω)> are useful for different purposes. All results are valid for any
finite integer P. Since each sequen ce in the ensemble is periodic with the same period P, the ensemble
average spectrum is entirely discrete. The m ≠0 sums include positive and negative integers.
Appendix F: Repeating Subsequences
265 (d) Limit as P → ∞ of the Ensembl e Result
We would now like to take the limit of the above as P→∞ . Write (F.33b) as
< P(ω)> = Ppulse (ω) [(β -α) { ω1
P ∑
m = -∞∞
δ(ω - ω1m/P) } + ω1 α ∑
m = -∞∞
δ(ω - ω1m) ] . (F.34)
Then define
fP(ω) ≡ ω1
P ∑
m = -∞∞
δ(ω - ω1m / P ) . ( F . 3 5 )
As P→∞, the spacing of the δ lines becomes closer and closer, while the amplitude ω1
P of each δ line
becomes less and less. Perhaps we can argue that in the limit this becomes some continuous function.
In line with the distribution theory approach to symbolic functions noted in Appendix A, suppose we integrate this function from some a to a+ ε for small ε, where a is an arbitrary real number,
∫a a+ε fP(ω) dω = ω1
P ∑
m = -∞∞
∫a a+ε δ(ω - ω1m/P)
= ω1
P ∑
m = -∞∞
Θ(a ≤ω1m/P ≤ a+ε) ( F . 3 6 )
where we use the notation of Appendix A (e) for the Θ function which takes value 1 if the inequality
argument is valid, meaning there is a delta hit. If P is a large integer, how many non-zero terms does this
Σm have? The inequality argument reads
Pa/ω1 ≤ m ≤ Pa/ω1 + Pε/ω1 .
Since P is large, we round each term in this equation to the nearest integer, making little error. We select a
very small ε first, and then we make sure P is large enough so P ε/ω1 is still a reasonably large integer
when rounded. Then we have
∫a a+ε fP(ω) dω = ω1
P ∑
m =Pa/ω1 Pa/ω1 + Pε/ω1
Θ(a ≤ω1m/P ≤ a+ε) = ω1
P ∑
m =Pa/ω1 Pa/ω1 + Pε/ω1
1
= ω1
P ( Pε /ω1) = ε . ( F . 3 7 )
Since we then have (for very large P) that ∫a a+ε fP(ω) dω = ε for any real a and for ε as small as we
like, and since the integral over range ε is proportional to ε , the function f P(ω) is equivalent to the
constant function 1. Thus we have shown that,
Appendix F: Repeating Subsequences
266 limP→∞ fP(ω) = limP→∞ [ ω1
P ∑
m = -∞∞
δ(ω - ω1m/P)] = 1. (F.38)
We then obtain this P →∞ limit of (F.34),
< P(ω)> = Ppulse (ω) [(β -α) + ω1 α ∑
m = -∞∞
δ(ω - ω1m ) ] ( F . 3 9 )
= Ppulse (ω) [(β-α) + (1/T 1) α ∑
m = -∞∞
2π δ(ω - ω1m) ] // ω1 = 2π/T1
= Ppulse (ω) [(β-α) + α ∑
m = -∞∞
2π δ(ωT1 - 2πm) ] .
This limit agrees with our result (35.11) for a random ensemble of infinite sequences for which <a man>
does not depend on the values of m and n,
< P(ω)> = Ppulse (ω) { (β -α) + α∑
m = -∞∞
2π δ(ωT1- 2πm ) } (35.11)
In the limit P →∞ , the discrete lines δ(ω - ω1m/P) shown in (F.34) have coalesced into a continuous
function. See Fig F.2 below for a graphical view.
(e) Calculation of P(ω) for a single Pulse Train su bject to Certain Conditions
Recall this expression for P(ω), which incorporates the discrete Wiener-Khintchine theorem,
P(ω) = Ppulse (ω) T1
T | Y"(z) |2 = Ppulse (ω) R"(z) (34.14a) (F.40)
where R"(z) is the Z transform of the autocorrelation sequence r s obtained from the y n.
Our first task is to find r s, which we defined this way,
rs ≡ limN→∞ [1
(2N+1) ∑
n = -NN
yn yn+s ] = <yn yn+s> 1 . (32.16) (F.41)
From now on we assume that the y m are real. Back in (F.25) we made the assumption that
<ymyn> = α for m ≠ n s < | m-n |
<ymyn> = β f o r m = n (F.25)
where <...> were ensemble averages. Here we are going to make a completely different assumption, and
this assumption applies to a single sequence:
Appendix F: Repeating Subsequences
267
<ymyn>1 = α for m ≠ n s < | m-n |
<ymyn>1 = β f o r m = n ( F . 4 2 )
Comment: If we imagine drawing each sequence of an ensemble as a row in a set of rows
... * * y n * y n+2 ... sequence #1
... * * y' n * y' n+2 ... sequence #2
... * * y" n * y" n+2 ... sequence #3
then any ensemble average like <y
nyn+2> is a vertical average through the ensemble, whereas an average
like <ynyn+2>1 is a horizontal average across one particular sequence row and <y nyn+2>1 never depends
on n as (F.41) shows. These two averages are unrelated and (F.42) being true does not imply that (F.25) is
true. As an example, one might take 10 infinite sequences each of which satisfies (F.42) and put them
down as a set of rows, and then each row is shifted hori zontally some amount to cause a 0 to be in column
n. For the resulting 10 row ensemble, one would find that <y n2> = 0 for column n, whereas <y n2>1 = β .
Since we require r s for arbitrary s in order to compute R"(z), we extend (F.42) in this manner
<y
myn>1 = α for m ≠ n + NP N = any integer
<ymyn>1 = β f o r m = n + N P ( F . 4 3 )
which is to say, for s being any integer whatsoever,
<yn yn+s>1 = α s ≠ NP
<yn yn+s>1 = β s = N P . ( F . 4 4 )
The reason of course is that {y
m} is periodic with period P,
y
m+NP = ym for any integer N (F.5)
so we must have for example <y n yn+P>1 = <yn yn>1 = β.
Thus we have arrived at our character ization of the autocorrelation sequence r s for our specific infinite
periodic sequence with the assumption (F.44),
rs = <yn yn+s>1 = ⎩⎨⎧ α s = NP
β s ≠ NP s = any integer in (F.45)
Our next step is to compute its Z transform R"(z),
Appendix F: Repeating Subsequences
268 R"(z) = ∑
n = -∞∞
rn z-n = β ∑
n =NP
z-n + α ∑
n ≠NP
z-n
= β ∑
n =NP
z-n + α ( ∑
n = -∞∞
z-n – ∑
n =NP
z-n )
= ( β-α) ∑
n =NP
z-n + α ∑
n = -∞∞
z-n . ( F . 4 6 )
The first sum is over n = 0, ±P, ±2P and so on. We can replace summation index n by index N,
∑
n =NP
z-n = ∑
N =-∞ ∞
z-NP = ∑
n = -∞∞
z-nP . ( F . 4 7 )
From (24.1) we know that z lies on the unit circle in the z-plane and is related to ω by
z = eiωT1 (24.1)
where T 1 is the duration of a pulse of the pulse train. We then have
∑
n =NP
z-n = ∑
n = -∞∞
(eiωT1)-nP = ∑
n = -∞∞
e-iωT1nP . ( F . 4 8 )
According to (A.31),
∑
n = -∞∞
eink = ∑
m = -∞∞
2πδ(k - 2πm) - ∞ < k < ∞ (A.31)
so setting k = - ωT1P we get
∑
n =NP
z-n = ∑
m = -∞∞
2πδ(-ωT1P - 2πm) = ∑
m = -∞∞
2πδ(ωT1P + 2πm) = ∑
m = -∞∞
2πδ(ωT1P - 2πm) (F.49)
where we use δ(x) = δ(-x) and in the last step take m → -m.
Meanwhile, our other sum of interest in (F.46) is this one,
∑
n = -∞∞
z-n = ∑
n = -∞∞
(eiωT1)-n = ∑
n = -∞∞
e-iωT1n
which is just the previous sum without the P. Thus,
Appendix F: Repeating Subsequences
269 ∑
n = -∞∞
z-n = ∑
m = -∞∞
2πδ(ωT1 - 2πm ) . ( F . 5 0 )
Inserting (F.48) and (F.49) into (F.46) gives
R"(z) = ( β-α) ∑
n =NP
z-n + α ∑
n = -∞∞
z-n
= ( β-α)
∑
m = -∞∞
2πδ(ωT1P - 2πm) + α ∑
m = -∞∞
2πδ(ωT1 - 2πm)
= ( β-α) (T
1P)-1 ∑
m = -∞∞
2πδ(ω - 2πm/[T1P]) + α (T1)-1 ∑
m = -∞∞
2πδ(ω - 2πm/T1)
= ( 2 π/T
1) { (β -α) (1/P) ∑
m = -∞∞
δ(ω - 2πm/[T1P]) + α ∑
m = -∞∞
δ(ω - 2πm/T1) }
= ω
1 { (β-α) (1/P) ∑
m = -∞∞
δ(ω - mω1/P) + α ∑
m = -∞∞
δ(ω - mω1) } (F.51)
and this concludes our calculation of the Z Tr ansform R"(z) of the autocorrelation sequence r s. It only
remains to install this into the Z Transfor m Wiener-Khintchine theorem (34.14a) which says
P(ω) = Ppulse (ω) R " ( z ) (34.14a)
so then
P(ω) = Ppulse (ω) ω1 { (β-α) 1
P ∑
m = -∞∞
δ(ω - mω1/P) + α ∑
m = -∞∞
δ(ω - mω1) }
= Ppulse (ω) ω1 ∑
m = -∞∞
[ (β-α) 1
P δ(ω - ω1m/P) + α δ(ω - ω1m) ] (F.52c)
We may compare this to the ensemble result of the section (c) above,
< P(ω)> = Ppulse (ω) ω1 ∑
m = -∞∞
[ (β-α) 1
P δ(ω - ω1m/P) + α δ(ω - ω1m) ] . (F.33c)
The expressions are exactly the same, but the meaning of symbols α and β is not the same! Since the
expressions have the exact same form, we can rewrite (F.52c) in the same alternate ways that (F.33c) was written:
Appendix F: Repeating Subsequences
270 P(ω) = Ppulse (ω) ω1 ∑
m = -∞∞
δ(ω - ω1m/P) 1
P [(β-α) + α 2π δ6(ωT1, P-1
2 ) ] (F.52a)
= Ppulse (ω) ω1 1
P [(β-α)∑
m = -∞∞
δ(ω - ω1m/P) + α ∑
N = -∞ ∞
δ(ω - ω1N) P ] // m = NP
= Ppulse (ω) ω1 [(β-α) 1
P ∑
m = -∞∞
δ(ω - ω1m/P) + α ∑
m = -∞∞
δ(ω - ω1m) ] (F.52b)
= Ppulse (ω) ω1 ∑
m = -∞∞
[ (β-α) 1
P δ(ω - ω1m/P) + α δ(ω - ω1m) ] (F.52c)
= Ppulse (ω) ω1 [ (β-α) 1
P ∑
m ≠ 0
δ(ω - ω1m/P) + α ∑
m ≠ 0
δ(ω - ω1m) + { (β-α) /P + α } δ(ω) ]
(F.52d)
(f) Summary of Results of (c) and (e) and an Example: The MLS Sequence
It is useful to summari
ze both results more completely:
_________________________________________________________________________________ Fact 1: (for an ensemble of infinite periodic sequences) If (F.53)
<y
myn> = α for m ≠ n s < | m-n |
<ymyn> = β f o r m = n (F.25)
t h e n
< P(ω)> = Ppulse (ω) ω1 ∑
m = -∞∞
[ (β-α) 1
P δ(ω - ω1m/P) + α δ(ω - ω1m) ] (F.33c)
_________________________________________________________________________________ Fact 2: (for a single infinite periodic sequence) If (F.54)
<y
myn>1 = α for m ≠ n + NP N = any integer
<ymyn>1 = β f o r m = n + N P (F.43)
t h e n
P(ω) = Ppulse (ω) ω1 ∑
m = -∞∞
[ (β-α) 1
P δ(ω - ω1m/P) + α δ(ω - ω1m) ] (F.52c)
_________________________________________________________________________________
Here are the two results (F.33d) and below that (F.52d), where as noted both expressions are the same,
Appendix F: Repeating Subsequences
271 < P(ω)> = Ppulse (ω) ω1 [ (β-α) 1
P ∑
m ≠ 0
δ(ω - ω1m/P) + α ∑
m ≠ 0
δ(ω - ω1m) + { (β-α) /P + α } δ(ω) ]
P(ω) = Ppulse (ω) ω1 [ (β-α) 1
P ∑
m ≠ 0
δ(ω - ω1m/P) + α ∑
m ≠ 0
δ(ω - ω1m) + { (β-α) /P + α } δ(ω) ]
Then here is a graphical representati on the above spectral power densities :
F i g F . 2 Each vertical arrow represents a spectral δ line. The height of the arrow is the value of the red envelope
curve times the quantity shown. The red curve P
pulse (ω) ω1 will in general have an infinite extent, an
example being Ppulse (ω) ω1 = sinc2(ωT1/2) = sinc2(πω/ω1) for a box shaped pulse as in (9.2). Once a
particular Ppulse (ω) is specified, some of the spectral lines ma y be quenched. For the box example, lines
are quenched when ω = Nω1 for N = ±1,±2 .... In this case, all the lines of the central image go away and
the corresponding lines in the left image also vanish, this being every Pth line in that image:
F i g F . 3 As P → ∞, we showed in section (d) that either of our two spectra above become
P
pulse (ω) ω1 [ (β-α) + α ∑
m ≠ 0
δ(ω - ω1m) + α δ(ω) ]
Regarding the three images of Fig F.2, we see that
• the left set of dense arrows coal esces into the continuous function ( β-α) Ppulse (ω) ω1
• the middle set of arrows stays exactly the same
• the amplitude of the DC line becomes α
Appendix F: Repeating Subsequences
272
Example: The MLS Sequence
A Maximum Length Sequence (MLS) (Lucht Polynomial Multipliers... ) which is created by a shift
register generator has the property (F.44), specifically,
<yn yn+s>1 = α = (1/4)(1 + 1/P) s ≠ NP
<yn yn+s>1 = β = (1/2)(1 + 1/P) s = NP (F.44)
where P must be one of the special values P = 2k-1 for k = 1,2,3.... Therefore, by Fact 2 (F.54) the
spectrum P (ω) of an MLS sequence is given by
P(ω) = Ppulse (ω) ω1 ∑
m = -∞∞
[ (β-α) 1
P δ(ω - ω1m/P) + α δ(ω - ω1m) ] . (F.52)
The horizontal average <a n an+s>1 of a sequence {a i} is seen in this picture made for P = 7 and s = 2 :
Fig F.4
The double-arrow pointer points to a 2 and a4 and their product is a 2a4. The pointer is rigidly slid
horizontally so the left arrow points to a 1 through a 7 in turn. The sum of the pointed-to products is added
up and the result divided by P, and that is <a n an+s>1 .
Now we can find a role for our ensemble Fact 1 as follows. Suppose we make a special ensemble of P sequences in which each is shifted one bit to the left of the one above. Then consider this picture of the
entire ensemble (clipped horizontally of course since each sequence is infinite),
Appendix F: Repeating Subsequences
273 F i g F . 5
We now leave the double-arrow pointer horizontally fi xed and apply it vertically to each row one at a
time. For the first row we get the product a
2a4 and for the second row a 3a5. We add up the products and
divide by P. This process generates the vertical average <a n an+s>. But clearly <a n an+s> = <an an+s>1
for this special ensemble since the sums are exactly the same. Thus, (F.43) implies (F.25) with the same α
and β and we can then apply Fact 1 (F.53) to find
< P(ω)> = Ppulse (ω) ω1 ∑
m = -∞∞
[ (β-α) 1
P δ(ω - ω1m/P) + α δ(ω - ω1m) ]
Comparing our results, we have for this ensemble that < P(ω)> = P(ω) of any sequence in the ensemble.
this says that P(ω) is the same for every sequence. This conclusi on is in agreement with the fact that when
an infinite sequence is shifted some number of bit positions, its spectral power density cannot change.
(g) Results for an A,B repeated sequence
This subject is treated in Section 34 (c)
using a "bru te force" approach and a "Fourier Series" approach.
Here we duplicate the main results of that section by applying our general formulas to a sequence where
the repeated subsequence is just P = 2, {y 0,y1} = {A,B}, and we allow A and B to be complex. The fact
that the results here agree with Section 34 lends some confidence to all three methods of computation.
Our general expressions for X(ω ) and P(ω) are,
X(ω) = Xpulse (ω) ω1 (1/P) YP"(z) ∑
m = -∞∞
δ(ω - mω1/ P ) (F.12)
P(ω) = Ppulse (ω) ω1 (1/P)2 ∑
m = -∞∞
δ(ω - mω1/P) | YP"(z) |2 (F.23)
where Y P"(z) ≡ ∑
n = 0P-1
yn e-iωnT1 . (F.8)
For a repeated A,B sequence P = 2 we have
Y
P"(z) = A + B e-iωT1
| YP"(z) |2 = | A + B e-iωT1 |2 = |A|2 + |B|2 + 2 Re{A*B e-iωT1}
Then,
X(ω)AB = Xpulse (ω) ω1 (1/2) [A + B e-iωT1]∑
m = -∞∞
δ(ω - mω1/2)
P (ω)AB = Ppulse (ω) ω1 (1/4) [ |A|2 + |B|2 + 2 Re{A*B e-iωT1} ]∑
m = -∞∞
δ(ω - mω1/2) .
Appendix F: Repeating Subsequences
274
We can slide the square-bracketed f actors inside the sum and then use
ωT1 = (mω1/2)T1 = π m(ω1/2π)T1 = πm => e-iωT1 = (-1)m
and the results simplify to
X(ω)
AB = Xpulse (ω) ω1 (1/2) ∑
m = -∞∞
[A + B (-1)m] δ(ω - mω1/2)
P (ω)AB = Ppulse (ω) ω1 (1/4) ∑
m = -∞∞
[ |A|2 + |B|2 + 2 Re(A*B) (-1)m ] δ(ω - mω1/2) .
These results agree with (34.18) and (34.20), see just below Fig 34.1 . For a box pulse of height 1 and width T
1 we know that
Xpulse (ω) = T1 sinc(ωT1/2) = (1/ω 1) 2π sinc(ωT1/ 2 ) (9.2)
Ppulse (ω) = |Xpulse (ω)|2
2πT1 = (1/ω1) sinc2(ωT1/ 2 ) . (33.24)
Then for a square wave with alternating A,B amplitudes we get
X(ω)AB = 2π sinc(ω T1/2) (1/2) ∑
m = -∞∞
[A + B (-1)m] δ(ω - mω1/2)
P (ω)AB = sinc2(ωT1/2) (1/4) ∑
m = -∞∞
[ |A|2 + |B|2 + 2 Re(A*B) (-1)m ] δ(ω - mω1/2) .
When the sinc functions are moved inside the sum, ωT1/2 = π m/2, so
X(ω)AB = 2π (1/2) ∑
m = -∞∞
[A + B (-1)m] sinc(πm/2) δ(ω - mω1/2)
P (ω)AB = (1/4) ∑
m = -∞∞
[ |A|2 + |B|2 + 2 Re(A*B) (-1)m ] sinc2(πm/2)δ(ω - mω1/2)
But
Appendix F: Repeating Subsequences
275
sinc(π m/2) = (2/ πm) sin(πm/2) = 1 for m = 0
= 0 f o r m e v e n
= (2/ πm) (-1)(m-1)/2 for m odd
so we get
X(ω)AB = 2π (1/2) ∑
m =odd
[A – B (-1)m] (2/πm) (-1)(m-1)/2 δ(ω - mω1/2) + 2π (1/2) [A+B] δ (ω)
P (ω)AB = (1/4) ∑
m =odd
[ |A|2 + |B|2 + 2 Re(A*B) (-1)m ] (2/πm)2 (-1)(m-1) δ(ω - mω1/2)
+ ω1 (1/4) [ |A|2 + |B|2 + 2 Re(A*B) ] δ(ω)
or X(ω)
AB = 2 ∑
m =odd
[A – B] (1/m) (-1)(m-1)/2 δ(ω - mω1/2) + π [A+B] δ(ω)
P (ω)AB = (1/π2) ∑
m =odd
[ |A|2 + |B|2 – 2 Re(A*B) ] (1/m)2 δ(ω - mω1/2)
+ ω1 (1/4) [ |A|2 + |B|2 + 2 Re(A*B) ] δ(ω)
For a standard-issue square wave with B = - A we have
[A – B] = 2A [ |A|
2 + |B|2 – 2 Re(A*B) ] = 4 |A|2
[A + B] = 0 [ |A|2 + |B|2 + 2 Re(A*B) ] = 0
and therefore
X(ω)A,-A = 4A ∑
m =odd
(1/m) (-1)(m-1)/2 δ(ω - mω1/2)
P (ω)A,-A = 4A2
(1/π2) ∑
m =odd
(1/m)2 δ(ω - mω1/2)
in agreement with (34.23).
Document Summary
276 Detailed Summary of this Document
A brief summary
may be found in the opening section be fore Chapter 1. The logical flow of the document
is fairly complicated because so many interrelated t opics are addressed, and because everything is derived
from scratch for the reader to see. Hopefully this 12- page detailed summary can provide an intermediate
level "map" of what is going on in the document. ____________________________________________________________________________________
Chapter 1: The Fourier Integral Transform and Related Topics ( 34 p)
Chapter 1 develops the basic theory
of the Fourier Integral Transform and its Si ne and Cosine cousins.
This Chapter forms the underpinning of all subseq uent Chapters. The Convolution Theorem receives
special attention. The connection is made between the Laplace Transform and the "generalized" Fourier
Transform applied to causal functi ons. Various "rules" are derived, and a connection is made between
filter spectra and time-domain Green's Functions.
Section 1 introduces the Fourier Integral Transform, which in our notation appears as
X(ω) = ∫-∞ ∞ dt x(t) e-iωt projection = transform (1.1)
x(t) = (1/2 π) ∫-∞ ∞ dω X(ω) e+iωt , expansion = inverse transform (1.2)
and discusses restrictions on functions for which the tr ansform applies and is meaningful. The notions of
a "pulse" and a "pulse train" are mentioned in p assing. The connection is made between the Fourier
Integral Transform and the Fourier Sine and Cosine Integral Transforms. This relationship provides a
method of using large published tables of Sine and Cosine transforms to obtain Fourier transforms.
Section 2 provides a traditional arm-waving proof of the Fourier Transform, while a more substantial
proof using bare-bones distribution theory is pr esented in Appendix A. The Fourier Transform is
discussed as a particular instance of Sturm-Liouville theory and the important claim is made that the
exponential functions form a complete orthogonal basis on the interval (- ∞,∞).
Section 3 derives the Convolution Theorem,
a(t) = ∫-∞ ∞ dt' b(t-t')c(t') ⇔ A(ω) = B(ω) C(ω ) (3.6)
with comments about dimensions of the various func tions, integral operators, diagonalization, and the
generalization to Fourier analysis on continuous groups.
Section 4 essentially applies the Convolution Theorem to obt ain the theory of filters and their transfer
functions in the ω domain. The connection is made between the time-domain differential equation
describing a filter and the Green's Function or propagator which is seen to be the response of the filter to a time domain δ impulse. A simple RC filter is treated in detail, then a second example is the same filter in
the limit that RC is very large. Section 5 discusses the set of convention choices we made in stating the Fourier Integral Transform,
and why those choices were made: sign of the expone ntial phase, i versus j, and allocation of the 2 π.
Section 6 presents a version of the Fourier Transform and Inversion which we loosely refer to as "the
generalized Fourier Transform". In the inversion formula, the ω contour is displaced in such a way as to
Document Summary
277 increase the size of the class of causal functions for wh ich Fourier Transforms exist. The main feature
here is the analytic continuation of a spectrum X( ω) off the real axis in the complex ω plane. When the
variable substitution s ≡ iω is made, the generalized Fourier Tr ansform applied to causal functions
becomes the traditional Laplace Transform,
X(s) ≡ L[x(t), s] ≡ X ( s / i ) (6.8)
X(s) = ∫0 ∞ dt x(t) e-st (6.9)
x(t) = (1/2 πi) ∫c-i∞ c+i∞ ds X(s) e+is (6.10)
Section 7 states various reflection rules involving the Fourier transform and defines a Hermitian
function.
Section 8 presents several simple examples of Fourier transforms which show how violation of the
Fourier requirements causes transforms to be dist ributions instead of functions. The last example
computes the spectrum of the Heaviside step function θ(t) (a distribution) an d briefly introduces the
notions of principal part integration and spectra pr esented in the form of limits. Details appear in
Appendix C.
Section 9 considers a simple box-shaped x(t) and its Fourier transform X( ω). This particular example
plays a major role in many later Sections.
Section 10 derives the various Parseval's Formulas and shows how one of these formulas relates to
the total energy in a pulse. Section 11 derives the differentiation rule dx(t)/dt ↔ [iωX(ω)] and presents a few examples.
Section 12 derives the time translation rule x(t - t
1) ↔ X(ω) e-iωt.
Section 13 derives these two "exponential sum rules",
∑
n = -∞∞
eink = ∑
m = -∞∞
2πδ(k - 2πm) -∞ < k < ∞ (13.2)
∑
n = -NN
eink = 2π { sin[(N+1/2)k]
2π sin(k/2) } ≡ 2π δ5(k,N) -∞ < k < ∞ (13.3 )
which are explored in more detail in Appendix A. These rules are important tools used later in processing expressions involving the power spectra of pulse trains. ____________________________________________________________________________________
Chapter 2: Pulse Trains and the Fourier Series Connection
Chapter 2 ex
amines the Fourier Transform spectrum of a simple pulse train formed from a general pulse
shape. Consideration of pulse trains of infinite length leads to a derivation of the Fourier Series
Transform. The chapter concludes with a discussion of sample pulse trains formed from box and bi-phase
pulses.
Section 14 computes the Fourier Transform spectrum of a simple pulse train (y n = 1).
Subsection (a) does this for an infinite pulse train. Although the spectrum of x pulse (t) is continuous,
that of the pulse train is entirely discrete and c onsists of a set of delta function spectral lines.
Document Summary
278 Subsection (b) repeats the calculation for a finite simple pulse train, and this time the spectrum is
found to be continuous. One can see exactly how this spectrum approaches the delta function lines as the
length of the pulse train becomes infinite.
Section 15 in effect derives the traditional Fourier Series Transform from the Fourier Integral
Transform of Chapter 1.
Section 16 obtains the spectrum of an infinite simple pulse train constructed from identical box pulses
of width τ with rising edges separated by T 1. As noted above, the spectrum is entirely discrete.
Section 17 develops some graphical aids to help un derstand pulse train spectra. The discrete delta
function spikes track a certain envelope function which is in fact the pulse spectrum |X pulse (ω)|. In
certain simple cases, some of the delta spikes are missing because these spikes align with zeros of the
pulse spectrum.
Section 18 discusses superposing a negative DC offset onto a positive box pulse train.
Section 19 constructs pulse trains using a certain "biphase pulse" instead of the box pulse. The
spectra of these two pulse train types are compared. In a certain limit, the two become the same. ____________________________________________________________________________________
Chapter 3: Sampled Signals and Digital Transforms (53 p)
Chapter 3 deals with vario
us digital forms of the Fourier Transform and their corresponding convolution
theorems and applies these concepts to amplitude-m odulated pulse trains (PAM signals) and to digital
filters. Topics include image spectra, aliasing and N yquist rate, group delay, FIR and IIR filters, poles,
and impulse response. The first digital transform is called the Digital Fourier Transform which is an ω-
domain version of the Z Transform whose variable is z = eiωΔt . The Z Transform and Discrete Fourier
Transforms are then addressed for both periodic and aperiodic signals. A recurring example is a simple
RC filter section.
Section 20 considers an amplitude-modulated pulse train x(t) = Σn y(t) T1δ(t - nT1). The spectrum of the
pulse train is found to be X( ω) = Σm Y(ω - mω1), and image spectra arrive on the scene. The notions of
aliasing and the Nyquist rate are descri bed with some traditional drawings.
Section 21 addresses two filter topics: digital filter image spectra, and analog filter group delay.
Subsection (a) considers a digital filter as an approximation to a standard "LIT" analog filter. What happens to the convolution theorem in this approximation? The answer is given in (21.13) and we find
that such a digital filter has image spectra. The notion of aliasing is illustrated in Fig 21.1.
Subsection (b) shows that, when a narrow pulse passe s through an analog bandpass filter, it
experiences group delay τ
d = dφ/dω where φ is the filter phase. It then fo llows that filters with linear
phase have a constant group delay, which is desirable to maintain the fidelity of pulses passing through
the filter. It is claimed that the same notion applies to digital filters. Section 22 is the first of two sections devoted to what we call the Digital Fourier Transform X'( ω). This
transform is stated above (22.6) and is compared side-by-side with the Fourier Integral Transform X( ω).
The time domain digital convolution sum a = b * c of (22.6) (a digital filter) is found to have the simple
diagonalized appearance A'( ω) = B'(ω)C'(ω ) in the ω domain.
Section 23 concludes the presentation of the Digital Fourier Transform X'(ω ).
Document Summary
279 Subsection (a) shows that the relation between X'( ω) and X(ω ) is X'(ω) = Σm X(ω- mω1), which is
exactly the form of main spectrum plus image spectr a which occurred in both Sections 20 and 21. We
then realize that the pulse train spectrum of Section 20 is just X( ω) = Y'(ω), where Y'( ω) is the Digital
Fourier Transform of the signal y(t) whose samples y n are the amplitudes of an amplitude-modulated
pulse train whose pulses are delta functions. Similarly, we realize that A( ω) = B'(ω)C(ω) is a compact
way to write the frequency domain digital filter equation where B'( ω) has image spectra.
Subsection (b) contains a "summary box" for the Digital Fourier Transform.
Section 24 shows that the Digital Fourier Transform is really the Z Transform in disguise. Specifically,
the Z Transform X"(z) = (1/ Δt) X'(ω) where z ≡ eiωΔt. This last equation defines an analytic mapping
between the ω plane and the z plane, as shown in Fig 24. 1, which removes the redundancy present in the
ω-plane picture. Various aspects of the Z Transform are then considered:
Subsection (a) states the digital convolution theorem in terms of the Z Transform.
Subsection (b) discusses the digital "unit impulse" signal in Z Transform notation.
Subsection (c) expresses time translation in Z Transfor m language, and here appears the famous
notion that when a time domain signal is delayed one sample period, its Z Transform is multiplied by z-1.
Subsection (d) addresses the notion of a digital time derivative and its Z Transform.
Subsection (e) uses this derivative idea to analyze a digital RC filter. The impulse response of such a
filter is compared with that of the analog RC filter studied back in Section 4. The responses are shown to be the same in the limit Δt→0.
Subsection (f) considers a general polynomial ratio form for a digital filter transfer function H"(z). It
is shown that, in a certain class of cases, the absence of (non-zero) poles in this ratio results in a filter whose impulse response decays in a finite amount of digital time. Conversely, the presence of such poles
results in a never-ending impulse response. These are the FIR and IIR filters and it is shown exactly how
these are represented in z space and the time domain, wher e these filters are implemented with registers,
constant multipliers and adders. Wh en H"(z) has poles, the filter is II R and has feedback. When H"(z) has
no poles, the filter is FIR and has no feedback. [ FIR/IIR mean Finite/Infinite Impulse Response] Subsection (g) revisits the digital RC filter considered in (e), and draws the z domain circuit in the
standard IIR form Fig 24.7. An im proved version of the digital RC filter is then trivially obtained.
Subsection (h) very briefly connects the idea of FIR and IIR filters with polynomial multipliers and
dividers. Such circuits appear in scramblers and cycl ic code error detection and correction algorithms.
Subsection (i) presents a "summary box" for the Z Transform
Section 25 considers the amplitude-modulated pulse train x(t) = Σ
n yn xpulse (t -tn) with an arbitrary pulse
shape. It is shown that the Fourier Transform spectrum of such a pulse train is X( ω) = Xpulse (ω) Y"(z)
where X pulse (ω) is the Fourier Transform spectrum of the pulse, while Y"(z) is the Z Transform of the
sequence of pulse amplitudes y n, where Y"(z) = (1/ Δt) Y'(ω). A summary box appears as equation (25.4).
Example 1 is a specific pulse train having a short sequence of amplitudes y n with a box-shaped pulse.
The Digital Fourier Transform |Y'( ω)| is plotted using Maple, then |X pulse (ω)| and |X( ω)| are plotted as
well. The amplitudes y n are explicitly recovered from the Digita l Fourier Transform inversion formula,
and the entire pulse train x(t) is explicitly recovered from the computed X( ω) using (1.2). This Example
shows that the Digital Fourier Transform is more a ppropriate for making frequency domain plots than the
Z Transform and really does have a role to play in digital signal analysis. Example 2 considers a pulse train having just a single box pulse of amplitude y
0 at t = 0. The formula
Y'(ω) = Σm Y(ω- mω1) is examined in light of the fact that Y'( ω) = y0T1 = a constant. We find an unusual
Document Summary
280 sum rule for the sinc function which states that Σm sinc[π(x-m)] = 1 for any real value of x, and this result
is verified using an obscure summation formula found in Gradshteyn and Ryzhik.
Section 26 considers a y n amplitude-modulated pulse train whose box-shaped pulses fill only a portion τ
of the sample time T 1 , the so called "aperture" of a D/A converter . It is shown that for any finite aperture
size the D/A converter output pulse train spectrum X( ω) differs from the desired signal spectrum Y(ω ) not
just in terms of image spectra (which can be filtered away), but also due to a sinc function distortion of
the spectrum: X( ω) ~ sinc(ωτ/2)Y(ω). This sinc distortion must be compensated by implementation of a
"sine x over x" filter somewhere in the sy stem, sometimes in an analog post-filter.
Section 27 gives a derivation of yet another transform, th e Discrete Fourier Transform or DTF. The new
feature here is that the pulse x
pulse (t) used to create a pulse train is approximated by a sequence of N
digital samples. Up to this point, our pulse trai ns have always been constructed from analog x pulse (t)
functions having discrete amplitudes y n.
Subsection (a) first reviews the Fourier Series Transform for an infinite simple pulse train built from
analog pulses, x(t) = Σn xpulse (t-mT1). When this pulse train is considered only at discrete times t n where
there are N values of t n within each T 1 period, we end up with what we call the Discrete Fourier
Transform of a Simple Pulse Train where the infinite set of Fourier Series c m coefficients are replaced
with a set of c' m coefficients (the DFT coefficients) which have the property c' m+N = c'm, which means the
sequence of c' m coefficients is periodic and contains only N distinct values. The two halves of this pulse
train transform are given in equations (27.9) and (27.11),
c'
m ≡ (1/N) ∑
n = -∞∞
xpulse (tn) e-imn(2π/N) . projection = transform (27.9)
x(tn) = ∑
m = 0N-1
c'm e+imn(2π/N) . expansion = inverse transform (27.11)
Subsection (b) is a proof that the above DFT transform is correct. In this brut e-force proof, a rather
strange identity is required which is obtained in Appendix B. A summary box for the Discrete Fourier
Transform of a Simple Pulse Train is then given as (27.16). Subsection (c) shows how the results of subsection (a) ma y be specialized to describe the Discrete
Fourier Transform is a signal which is a single digitized pulse having N samples x
n and time duration T 1.
This results in the traditional form of the Discr ete Fourier Transform as summarized in box (27.21).
Allowing for an arbitrary convention factor A, this DFT can be stated as
c'm ≡ (A/N) ∑
n = 0N-1
xn e-imn(2π/N) m = 0,1...N-1 projection = transform
xn = (1/A) ∑
m = 0N-1
c'm e+imn(2π/N) n = 0,1,...N-1 expansion = inverse transform (27.22)
Commonly appearing values of A are 1, N and N .
Document Summary
281 Subsection (d) provides a graphical view of the DFT situation for a pulse train. One can consider the
N pulse sample values as a vector xpulse and the N coefficients as another vector c', and these vectors are
then related by an NxN symmetric matrix -- just another way to think about the above transform.
____________________________________________________________________________________
Chapter 4: Some Practical Topics (19 p)
Chapter 4 shows that sy
mmetric FIR filters have lin ear phase and thus constant group delay. A specific
brick wall digital filter is designed and then later used as an oversampling interpolation filter in the design
of a D/A converter output section. This system is then simulated with a simple Maple program. Section 28 shows that FIR filters have linear phase and thus constant group delay if the filter coefficients
are symmetric. The result is demonstrated by two different methods. Section 29 describes a specific design for a digital brick-wall filter with a 4x clock rate. A hardware
implementation is shown in some detail, and the filter spectrum is computed and plotted. It is shown how
101 taps gives a better brick wall than 21 taps , though both exhibit the Gibbs phenomenon.
Section 30 considers a set of increasingly complicated D/A converter output section designs.
Subsection (a) describes a very simple D/A converter output section. A certain specific digital signal
y
n is assumed in Fig 30.1. Both the Di gital Fourier Transform spectrum |Y'( ω)| and Fourier Transform
spectrum |X( ω)| of the stepwise analog output are computed a nd plotted. The first plot shows the expected
image spectra, while the second shows a large sinc di stortion of the output with the image spectra damped
down significantly.
Subsection (b) clocks the D/A converter of the previous d esign at a 4x rate. This has no effect on the
analog output waveform, but allows for a reinterpretation of the output in terms of a 4x rate analysis. This
section is mainly an exercise in computing the output spectrum two different ways.
Subsection (c) adds a zero-stuffing input section to the converter design, which results in a new
sampled signal with a 25% aperture as shown in Fig 30.8. Both |Y'( ω)| and |X( ω)| are computed for this
new signal. The sinc distortion is much reduced, but se veral image spectra appear in the output spectrum.
Subsection (d) adds the 4x-rate brick-wall filter of Secti on 29 between the zero-stuffing input circuit
and the D/A converter output circuit. The new X( ω) spectrum of the output shows the desirable features
of a reduced sinc distortion and image spectra rejection. The system is then simulated with simple Maple code and the time-domain output plotted first for a 21-tap filter and th en for a 41-tap filter. It is seen why
this kind of filter is called an interpola tion filter and an alias-rejection filter.
____________________________________________________________________________________
Chapter 5: Some Theoretical Topics (10 p)
Chapter 5 contains onl
y Section 31 which explores the subject of dispersion relations for the spectral
function X(ω ) and for other related functions treated as an alytic functions of a complex variable.
Subsection (a) derives an integral equation (32.1) which X( ω) must satisfy if X(ω ) is analytic in the
upper half ω plane. The integral in this equation is a "princ iple value" integral which has a tick mark. This
type of integral is described more in Appendix C.
Document Summary
282 Subsection (b) derives a similar equation (31.3) for X(ω ) being analytic in the lower half ω plane.
These two equations are then written in (31.5) as one equation with a parameter σ = ±1 to distinguish the
cases. The RC filter spectrum G( ω) found in Section 4 (b) is shown to satisfy this integral equation.
Subsection (c) rewrites the dispersion relation in terms of the real and imaginary parts of X( ω),
shown in (31.8), so now there are two equations. In the case that X(ω ) is associated with a real signal x(t)
the negative frequency half of the integral can be fold ed into the positive half to give a new form (31.9)
which is associated with the names Kramers and Kronig.
Subsection (d) shows how the dispersion relations are also valid for γ(ω) where X( ω) = e-γ(ω), with
certain assumptions. The real and imaginary parts of γ(ω) = α (ω) + i β (ω) are the attenuation and phase of
the spectrum (or transfer function) X( ω). When those assumptions are met, such a filter is a minimal
phase filter. It is shown that the dispersion relations mix together the attenuation and phase, so neither can
be set independently of the other. Subsection (e) demonstrates that a minimal phase filter has approximate linear phase in any region of
ω that is far away from regions where the attenuation α(ω) significantly varies.
Subsection (f) applies this idea to a zero resistance coax ial cable and argues that such a cable has
linear phase up to infrared frequencies. Linear phase m eans constant group delay which means the cable
is non-dispersive. Real cables of course don't have zero resistance and exhibit skin effect. Subsection (g) rewrites the dispersion relations in terms of the Hilbert Transform. The dispersion
relation becomes roughly the statement that X( ω) must be ±i times its own Hilbert Transform. It is
emphasized that the dispersion relation is only a condition on X( ω) and does not fully determine X( ω).
___________________________________________________________________________
Chapter 6: Power in Pulse Trains (70 p)
Chapter 6 derives expressions for the en
ergy and power , and the spectral energy and power densities of an
amplitude-modulated pulse train. This work is carried out with a moderate amount of mathematical rigor.
Correlation and autocorrelation are mentioned. The notio n of a statistical pulse train is presented and the
spectral power density of such pulse trains is estab lished. These results are then applied to various
uncorrelated standard line codes including NRZ, RZ and Manchester. Two examples of correlated pulse trains are then treated -- AMI and Change/Hold -- and the latter is then used to get results for the NRZI
line code.
Section 32 introduces the autocorrelation function r
x(t) of function x(t) and discusses energy and power.
Subsection (a) computes r x(t) for a square pulse.
Subsection (b) defines quantities E, p(t) and E (ω) which are the total energy, instantaneous power,
and spectral energy density of a pulse train.
Subsection (c) shows that the Fourier Integral Transform R x(ω) of rx(t) is directly related to E(ω), a
fact known as the Wiener-Khintchine theorem.
Subsection (d) verifies this theorem for the square pulse.
Subsection (e) digresses on the subject of cross-correlation and relates this to auto-correlation.
Section 33 computes the power spectral density of a simple pulse train with arbitrary x pulse (t).
Subsection (a) computes |X( ω)|2 for an infinite simple pulse train.
Subsection (b) repeats this calculation for a finite simple pulse train.
Document Summary
283 Subsection (c) defines the spectral power density P (ω) of a pulse train and displays the result for the
infinite and finite pulse trains based on the |X( ω)|2 computations of the previous two subsections. As
expected, the infinite pulse train has a discrete po wer spectrum while the finite one has a continuous
spectrum. In the infinite case, P(ω) is related to the Fourier Series coefficients c m , am and bm.
Subsection (d) defines the average power P of a pulse train and relates that to c m , am and bm .
Section 34 computes the spectral power density of a general (PAM) pulse train with arbitrary x pulse (t).
Subsection (a) gathers up in a summary box a set of energy and power facts already derived which
apply to general pulse trains, not just simple ones.
Subsection (b) computes the spectral energy density E(ω) and power density P(ω) for a general pulse
train. The results are stated for infinite pulse trains , with the understanding that all sums are replaced by
finite sums for finite pulse trains. Subsection (c) uses two methods to compute the spectrum X( ω) and power spectral density P(ω) of a
general pulse train in which the amplitudes ha ve the form A,B,A,B.... with arbitrary x
pulse (t). The first
method does a brute force calculation of the result for a finite pulse train, and then takes the limit N →∞ to
obtain the result for an infinite pulse train. The second method treats the pair of pulses A,B as a single
pulse of period 2T 1 and obtains the same results using a Fourier Series analysis. The results are then
applied to a short catalog of standard cases (such as A = 1, B = 0) and displayed in Figures 34.2 through
34.9. At the very end results are stated for repeat sequences A,B,C and A 0,A1....AM-1 with derivations
left as an exercise for the reader.
Section 35 treats statistical pulse trains in which the amplitudes take random va lues. For example, a
square wave pulse train might have probability p for amplitude 1 and 1-p for amplitude 0.
Subsection (a) defines the notion of a statistical ensemble of general pulse trains. The ensemble
contains I pulse trains and each is la beled by index i and has amplitudes y
n(i). The ensemble average is
indicated by <..> and quantities < E(ω)>, < P(ω)> and <P> are stated in terms of sums which contain
amplitude averages <y myn> . This average is computed for an ensemble of pulse trains whose amplitudes
are non-correlated and < P(ω)> is shown to be expression (35.7) where α = <ymyn> and β ≡ <ym2>. It is
noted that α = μ2 and β = μ2 + σ2 where μ and σ2 are the mean and variance of the y n(i) values.
Subsection (b) examines expression (35.7) for < P(ω)> and evaluates its two double sums in the case
of an infinite pulse train and arrives at the following well-known result for the spectral power density of a
non-correlated statistical pulse train, where Ppulse (ω) is the spectrum of x pulse (t) :
< P(ω)> = Ppulse (ω) { σ2 + μ2∑
m = -∞∞
2π δ(ωT1- 2πm ) } (35.11a)
The above spectrum has both continuous and discrete components. Subsection (c) repeats the calculation for a finite pulse train and then takes the N →∞ limit.
Subsection (d) summarizes the results for non-correlated pulse trains and gives a few examples.
Subsection (e) presents a numerical simulation of an ensemble of random pulse trains. A simple
Maple program computes and plots the average spectral power density < P(ω)> of the ensemble, and one
clearly sees the continuous and discrete pieces of the spectrum. Subsection (f) observes that all the results obtained for statistical pulse trains could be obtained
without mention or use of the autocorrelation function.
Document Summary
284 Subsection (g) considers and then resolves an interesting paradox: the simulation shows a power
spectrum < P(ω)> having a continuous component, whereas the simulated <X( ω)> has no continuous
component. This seems strange since P (ω) ~ |X(ω)|2.
Section 36 computes the spectral power density for several standard uncorrelated line codes.
Section 37 computes the spectral power density for the Alte rnate Mark Inversion (AMI) line code. In this
line code, the locations in the pulse train are correlated with each other which makes the calculation of <y
myn> more difficult, one cannot claim <y myn> = <ym><yn> .
Section 38 computes the power spectral density (38.21) fo r what we call a Change/Hold line code. The
calculation is similar to that done for the AMI line code. Since we could not find external verification of the result, various limits are taken and checked against known results. The unipolar and bipolar NRZI line codes are then treated as special cases of the Change/Hold line code. ____________________________________________________________________________________
Appendix A: Delta Function Technology (19 p)
Appendix A discusses
"delta functions" at a someone deeper and more practical level than one commonly
finds in texts and on the web. Delta function models are constructed a nd many mathematical identities are
developed which find use in the main text. It seemed b est to isolate this material into an appendix rather
than derive its results in line.
The opening text gives a very minimalist outline of "distribution theory", with a mention of the meaning of "symbolic functions " like the delta function.
Subsection (a) develops four different "models" ( δ
1 through δ4) for the delta function. Each model is
a sequence of functions which approaches the true δ in the limit of some parameter. As part of this
development, two different proofs are given for equation (2.1),
∫-∞ ∞ dx e±ikx = 2πδ( k ) . (2.1)
Subsection (b) develops models δ 5, δ6 and δ7 for what we call "periodic delta functions".
Subsection (c) derives the following two "exponential sum rules"
∑
n = -∞∞
eink = ∑
m = -∞∞
2πδ(k - 2πm ) - ∞ < k < ∞ (13.2)
∑
n = -NN
eink = 2π { sin[(N+1/2)k]
2π sin(k/2) } ≡ 2π δ5(k,N) -∞ < k < ∞ (13.3)
and it is shown how the first is a limit of the second. Equation (13.2) is then used to derive the Poisson
Sum Formula. Both sum rules are used many times in the main text. Subsection (d) notes that the number δ(0) is undefined in distribution theory, but is meaningful for a
specific delta function model. It is just a notation to allow us to handle certain infinite sums that occur many times in the main text. A related notion is ca lled "undoing the limit", or "backing off". For example,
one can undo the limit of (13.2) above to arrive at (13.3) in which the right side is a perfectly normal
Document Summary
285 function. Sometimes it is necessary to undo the limit in this way, do some calculations, and then take the
limit N→∞ again.
Subsection (e) considers the delta function sifting property (2 .2) and pays particular attention to what
happens at the two integration endpoints. A special notation Θ(a ≤ x ≤ b) is introduced to incorporate the
endpoint effects. Several properties of the Θ "function" are stated.
Subsection (f) addresses the tricky subject of the product of two delta functions of the same variable.
This concept is undefined in distribution theory, but one must face such products when computing objects
like the spectral power density of a pulse train: the spectrum includes delta functions, and the power
density is proportional to the spectrum squared. The so lution as mentioned just above is to back off from
the N=∞ limit, figure out what is happening, and then resume the limit again. It is in this process that the
delta function models δ5 and δ6 find their use.
____________________________________________________________________________________
Appendix B: Derivation of a Certain Identity (2p)
Appendix B
derives this rather obscure identity, wh ere N and s are arbitrary integers with N > 0,
∑
m = 0N-1
e+ims(2π/N) = N ∑
m = -∞∞
δs,mN (B.1)
____________________________________________________________________________________
Appendix C: The Fourier Transform and its relation to the Hilbert Transform (17 p)
Appendix C
further develops Fourier Integral Transf orm theory beyond the treatment of the main text.
Instead of using the notation f(t) and F( ω), here we use f(t) and f^( ω).
Subsection (a) introduces this ^ notation for the Fourier tran sform and allows for a general constant k
in the transform definition to allow comparison of r esults with sources using other conventions. In this
new notation, some "Facts" are derived, such as
[f(-u)]^(- ω) = f^(ω ) f^-1(u) = (1/2 πk2) f^(-u) f^^(t) = 2 πk2 f(-t) .
An operator notation is introduced which is similar to that commonly used for the Laplace Transform.
Subsection (b) introduces the notion of a Cauchy Principle Value integral with a preliminary look at
its associated symbolic function pf(x). We use the tick notation ∫-- for such an integral.
Subsection (c) computes the "regular" Fourier transform of the function f(u) = 1/u, which is not quite
as simple as it seems. It is shown that (1/u)^( ω) = -iπk sgn(ω) where sgn(ω ) = 2θ(ω) – 1 and where θ(ω) is
the Heaviside step function. The inverse Fourier transfor m is then done to verify that the original function
1/u is recovered. A direc tly related result is [sgn( ω)]^(t) = -2ik(1/t).
Subsection (d) gives a derivation of this fact involving complex integration ( ω is real) ,
lim ε→0 1
ω ∓ iε = pf(1/ω) ± iπδ(ω) ( C . 2 2 )
which for want of a better name we call The Pole A voidance Rule. A certain contour avoids hitting a pole
by deflecting one way or the other around the pole. Along the way, pf(x) is defined as a symbolic function.
Document Summary
286 Subsection (e) computes the Fourier transform of the function lim ε→0 [1/(u±iε)] which seems very
close to the function 1/u explored in subsection (c). One finds that (1
u±iε )^(ω) = ∓ 2πik θ(±ω). The
original function is then recovered using the inverse F ourier transform. A directly related result is the fact
that [θ(±t)]^(ω) = k pf( 1
±iω ) + π k δ(ω) (Fourier transform of the Heaviside step function).
Subsection (f) then computes [ θ(u)]^(ω) using the "generalized" Fourier transform (in which the
recovery contour is vertically shifted). The result is [ θ(u)]^(ω) = k
iω .
Subsection (g) summarizes all the examples considered in this appendix.
Subsection (h) defines the Hilbert Transform fh(t) of a function f(t). This transform has an intimate
connection with the Fourier Transform which is la id out in a series of unusual facts such as
[fh]^(ω) = k-1 ( 1
πt )^(ω) f^(ω) = -i sgn( ω) f^(ω ) fhh(t) = -f(t) .
The inversion formula for the Hilbert transform is de rived and the transform is then computed for some
simple examples. The Hilbert Transform appears in the dispersion relations of Section 31 (g).
____________________________________________________________________________________
Appendix D: Probability Theory: how α and β are related to μ and σ (5
p)
Appendix D reviews some basic probability theory and relates the α and β parameters of the uncorrelated
spectral power density formula to statistical properties of pulse train amplitudes.
Subsection (a) discusses random variables X and Y and their expectation values like μ
x = E(X) =
<x> and E(XY). Expressions are given both in terms of probability dens ity functions and in terms of
ensemble measurements. The variance σ2(X) and covariance cov(X,Y) functions are defined and certain
relations obtained. The concept of X and Y being independent (uncorrelated) random variables is
explored. Subsection (b) translates the results of subsecti on (a) into the context where X = Y
m and Y = Y n
where Y m is the random variable associated with positi on m in a pulse train, and whose values are the
pulse amplitudes y m .
Subsection (c) shows that for an uncorrelated pulse tr ain the following facts are true for the
coefficients α and β used in Chapter 6 on power spectra,
α = μ
2 β = μ2 + σ2 (β-α) = σ2 μ ≡ <ym> σ2 = <ym2> - <ym>2
These relations provide a useful interpretation of the parameters α and β in terms of the statistics of the
pulse train amplitudes. ____________________________________________________________________________________
Appendix E: Table of Transforms (4 p)
Lists all transform
pairs which appear in this document and shows how they are related.
Document Summary
287 ____________________________________________________________________________________
Appendix F: The Spectrum and Power Density for Repeated-Sequence Pulse Trains
For infinite
pulse trains composed of repeats of some length-P subsequence:
Subsection (a) computes X( ω) in (F.12)
Subsection (b) computes P(ω) in (F.23)
Subsection (c) computes <P (ω)> for an ensemble of pulse trains which respect the special condition
< y m* yn> = α for m ≠ n
< y m* yn> = β for m = n s < |m-n| (F.25)
The result for < P(ω)> is stated in (F.33) in several different forms.
Subsection (d) takes the P →∞ limit of the section (c) result for < P(ω)>
Subsection (e) computes P(ω) for a single pulse train which respects the special condition
< y myn>1 = α for m ≠ n + NP N = any integer
< y myn>1 = β f o r m = n + N P (F.43)
where <y myn>1 is a horizontal average across the single sequence (autocorrelation).
The result for P(ω) is stated in (F.52).
It is noted that the results for < P(ω)> of (c) and P(ω) of (e) are exactly the same in terms of their
respectively defined α and β constants.
Subsection (f) restates the results of (c) and (e) as Fact 1 (F.53) and Fact 2 (F.54).
A graphical representation is drawn for the spectrum in general, and for a box pulse. It is shown that the MLS sequence is a candidate for application of Fact 2. A simple ensemble is then constructed to which Fact 1 maybe applied, and the result is then interpreted. Subsection (g) treats the P = 2 repeated subsequence A, B using the general formulas of (a) and (b)
References
288 References
W.R. Bennett and J.R. Davey
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J.F. Colombeau, "Multiplication of Distributions", Bulle tin of the American Mathematical Society, Vol.
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contain Tables of Integral Transforms. As with many out-of-print books, these volumes can perhaps be
found as pdf or djvu files on the web. English and Ru ssian editions exist. The notations EH and ET are
how these works are referenced from Gr adshteyn and Ryzhik (see below).
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joined for the 2008 3rd Ed. Not to be confused with a watered-down "Easy Outline" version. P. Lucht, Polynomial Multipliers and Dividers, Shift Register Generators and Scramblers (2013),
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