Introduction to Communication Systems

Upamanyu Madhow
University of California, Santa Barbara

August 18, 2013

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Contents

Preface 9

Acknowledgements 13

1 Introduction 15
1.1 Analog or Digital? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
1.1.1 Analog communication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
1.1.2 Digital communication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
1.1.3 Why digital? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
1.2 A Technology Perspective . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
1.3 Scope of this Textbook . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
1.4 Concept Inventory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
1.5 Endnotes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25

2 Signals and Systems 27

2.1 Complex Numbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
2.2 Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
2.3 Linear Time Invariant Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
2.3.1 Discrete time convolution . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
2.3.2 Multi-rate systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
2.4 Fourier Series . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
2.4.1 Fourier Series Properties and Applications . . . . . . . . . . . . . . . . . . 49
2.5 Fourier Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
2.5.1 Fourier Transform Properties . . . . . . . . . . . . . . . . . . . . . . . . . 53
2.5.2 Numerical computation using DFT . . . . . . . . . . . . . . . . . . . . . . 57
2.6 Energy Spectral Density and Bandwidth . . . . . . . . . . . . . . . . . . . . . . . 59
2.7 Baseband and Passband Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
2.8 The Structure of a Passband Signal . . . . . . . . . . . . . . . . . . . . . . . . . . 63
2.8.1 Time Domain Relationships . . . . . . . . . . . . . . . . . . . . . . . . . . 63
2.8.2 Frequency Domain Relationships . . . . . . . . . . . . . . . . . . . . . . . 70
2.8.3 Complex baseband equivalent of passband filtering . . . . . . . . . . . . . 74

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 2.8.4 General Comments on Complex Baseband . . . . . . . . . . . . . . . . . . 75
2.9 Wireless Channel Modeling in Complex Baseband . . . . . . . . . . . . . . . . . . 77
2.10 Concept Inventory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
2.11 Endnotes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79

3 Analog Communication Techniques 87

3.1 Terminology and notation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
3.2 Amplitude Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
3.2.1 Double Sideband (DSB) Suppressed Carrier (SC) . . . . . . . . . . . . . . 89
3.2.2 Conventional AM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
3.2.3 Single Sideband Modulation (SSB) . . . . . . . . . . . . . . . . . . . . . . 98
3.2.4 Vestigial Sideband (VSB) Modulation . . . . . . . . . . . . . . . . . . . . . 103
3.2.5 Quadrature Amplitude Modulation . . . . . . . . . . . . . . . . . . . . . . 105
3.2.6 Concept synthesis for AM . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
3.3 Angle Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
3.3.1 Limiter-Discriminator Demodulation . . . . . . . . . . . . . . . . . . . . . 110
3.3.2 FM Spectrum . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
3.3.3 Concept synthesis for FM . . . . . . . . . . . . . . . . . . . . . . . . . . . 114
3.4 The Superheterodyne Receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115
3.5 The Phase Locked Loop . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118
3.5.1 PLL Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
3.5.2 Mathematical Model for the PLL . . . . . . . . . . . . . . . . . . . . . . . 122
3.5.3 PLL Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
3.6 Some Analog Communication Systems . . . . . . . . . . . . . . . . . . . . . . . . 128
3.6.1 FM radio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128
3.6.2 Analog broadcast TV . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
3.7 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131

4 Digital Modulation 141

4.1 Signal Constellations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
4.2 Bandwidth Occupancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
4.2.1 Power Spectral Density . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
4.2.2 PSD of a linearly modulated signal . . . . . . . . . . . . . . . . . . . . . . 147
4.3 Design for Bandlimited Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . 150
4.3.1 Nyquist’s Sampling Theorem and the Sinc Pulse . . . . . . . . . . . . . . . 150
4.3.2 Nyquist Criterion for ISI Avoidance . . . . . . . . . . . . . . . . . . . . . . 153
4.3.3 Bandwidth efficiency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
4.3.4 Power-bandwidth tradeoffs: a sneak preview . . . . . . . . . . . . . . . . . 157
4.3.5 The Nyquist criterion at the link level . . . . . . . . . . . . . . . . . . . . 159

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 4.3.6 Linear modulation as a building block . . . . . . . . . . . . . . . . . . . . 160
4.4 Orthogonal and Biorthogonal Modulation . . . . . . . . . . . . . . . . . . . . . . . 160
4.5 Proofs of the Nyquist theorems . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
4.6 Concept Inventory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166
4.7 Endnotes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167
4.A Power spectral density of a linearly modulated signal . . . . . . . . . . . . . . . . 178
4.B Simulation resource: bandlimited pulses and upsampling . . . . . . . . . . . . . . 180

5 Probability and Random Processes 185

5.1 Probability Basics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 185
5.2 Random Variables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
5.3 Multiple Random Variables, or Random Vectors . . . . . . . . . . . . . . . . . . . 196
5.4 Functions of random variables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 202
5.5 Expectation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 206
5.5.1 Expectation for random vectors . . . . . . . . . . . . . . . . . . . . . . . . 210
5.6 Gaussian Random Variables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 211
5.6.1 Joint Gaussianity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 216
5.7 Random Processes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224
5.7.1 Running example: sinusoid with random amplitude and phase . . . . . . . 224
5.7.2 Basic definitions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 225
5.7.3 Second order statistics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227
5.7.4 Wide Sense Stationarity and Stationarity . . . . . . . . . . . . . . . . . . . 228
5.7.5 Power Spectral Density . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
5.7.6 Gaussian random processes . . . . . . . . . . . . . . . . . . . . . . . . . . 234
5.8 Noise Modeling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 235
5.9 Linear Operations on Random Processes . . . . . . . . . . . . . . . . . . . . . . . 240
5.9.1 Filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 240
5.9.2 Correlation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
5.10 Concept Inventory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
5.11 Endnotes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 247
5.12 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 247
5.A Q function bounds and asymptotics . . . . . . . . . . . . . . . . . . . . . . . . . . 260
5.B Approximations using Limit Theorems . . . . . . . . . . . . . . . . . . . . . . . . 260
5.C Noise Mechanisms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 262
5.D The structure of passband random processes . . . . . . . . . . . . . . . . . . . . . 263
5.D.1 Baseband representation of passband white noise . . . . . . . . . . . . . . 265
5.E SNR Computations for Analog Modulation . . . . . . . . . . . . . . . . . . . . . . 266
5.E.1 Noise Model and SNR Benchmark . . . . . . . . . . . . . . . . . . . . . . . 266
5.E.2 SNR for Amplitude Modulation . . . . . . . . . . . . . . . . . . . . . . . . 266

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 5.E.3 SNR for Angle Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . . 269

6 Optimal Demodulation 277

6.1 Hypothesis Testing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 278
6.1.1 Error probabilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 279
6.1.2 ML and MAP decision rules . . . . . . . . . . . . . . . . . . . . . . . . . . 280
6.1.3 Soft Decisions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
6.2 Signal Space Concepts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
6.2.1 Representing signals as vectors . . . . . . . . . . . . . . . . . . . . . . . . 288
6.2.2 Modeling WGN in signal space . . . . . . . . . . . . . . . . . . . . . . . . 292
6.2.3 Hypothesis testing in signal space . . . . . . . . . . . . . . . . . . . . . . . 293
6.2.4 Optimal Reception in AWGN . . . . . . . . . . . . . . . . . . . . . . . . . 295
6.2.5 Geometry of the ML decision rule . . . . . . . . . . . . . . . . . . . . . . . 298
6.3 Performance Analysis of ML Reception . . . . . . . . . . . . . . . . . . . . . . . . 301
6.3.1 The Geometry of Errors . . . . . . . . . . . . . . . . . . . . . . . . . . . . 301
6.3.2 Performance with binary signaling . . . . . . . . . . . . . . . . . . . . . . . 302
6.3.3 M-ary signaling: scale-invariance and SNR . . . . . . . . . . . . . . . . . . 305
6.3.4 Performance analysis for M-ary signaling . . . . . . . . . . . . . . . . . . . 310
6.3.5 Performance analysis for M-ary orthogonal modulation . . . . . . . . . . . 318
6.4 Bit Error Probability . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
6.5 Link Budget Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 322
6.6 Concept Inventory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 327
6.7 Endnotes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 329
6.A Irrelevance of component orthogonal to signal space . . . . . . . . . . . . . . . . . 347

7 Channel Coding 349

7.1 Motivation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 350
7.2 Model for Channel Coding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 352
7.3 Shannon’s Promise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 354
7.3.1 Design Implications of Shannon Limits . . . . . . . . . . . . . . . . . . . . 360
7.4 Introducing linear codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 361
7.5 Soft decisions and belief propagation . . . . . . . . . . . . . . . . . . . . . . . . . 370
7.6 Concept inventory on channel coding . . . . . . . . . . . . . . . . . . . . . . . . . 374
7.7 Endnotes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 376
7.8 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 376

8 Dispersive Channels and MIMO 387

8.1 Singlecarrier System Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 388
8.1.1 Signal Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 388

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 8.1.2 Noise Model and SNR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 393
8.2 Linear equalization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 394
8.2.1 Adaptive MMSE Equalization . . . . . . . . . . . . . . . . . . . . . . . . . 397
8.2.2 Geometric Interpretation and Analytical Computations . . . . . . . . . . . 400
8.3 Orthogonal Frequency Division Multiplexing . . . . . . . . . . . . . . . . . . . . . 406
8.3.1 DSP-centric implementation . . . . . . . . . . . . . . . . . . . . . . . . . . 408
8.4 MIMO . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 413
8.4.1 The linear array . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 413
8.4.2 Beamsteering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 415
8.4.3 Rich Scattering and MIMO-OFDM . . . . . . . . . . . . . . . . . . . . . . 418
8.4.4 Diversity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 421
8.4.5 Spatial multiplexing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 426
8.5 Concept Inventory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 428
8.6 Endnotes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 430
8.7 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 431

Epilogue 445

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Preface

Progress in telecommunications over the past two decades has been nothing short of revolution-
ary, with communications taken for granted in modern society to the same extent as electricity.
There is therefore a persistent need for engineers who are well-versed in the principles of commu-
nication systems. These principles apply to communication between points in space, as well as
communication between points in time (i.e, storage). Digital systems are fast replacing analog
systems in both domains. This book has been written in response to the following core question:
what is the basic material that an undergraduate student with an interest in communications
should learn, in order to be well prepared for either industry or graduate school? For example, a
number of institutions only teach digital communication, assuming that analog communication
is dead or dying. Is that the right approach? From a purely pedagogical viewpoint, there are
critical questions related to mathematical preparation: how much mathematics must a student
learn to become well-versed in system design, what should be assumed as background, and at
what point should the mathematics that is not in the background be introduced? Classically,
students learn probability and random processes, and then tackle communication. This does not
quite work today: students increasingly (and I believe, rightly) question the applicability of the
material they learn, and are less interested in abstraction for its own sake. On the other hand,
I have found from my own teaching experience that students get truly excited about abstract
concepts when they discover their power in applications, and it is possible to provide the means
for such discovery using software packages such as Matlab. Thus, we have the opportunity to
get a new generation of students excited about this field: by covering abstractions “just in time”
to shed light on engineering design, and by reinforcing concepts immediately using software ex-
periments in addition to conventional pen-and-paper problem solving, we can remove the lag
between learning and application, and ensure that the concepts stick.
This textbook represents my attempt to act upon the preceding observations, and is an out-
growth of my lectures for a two-course undergraduate elective sequence on communication at
UCSB, which is often also taken by some beginning graduate students. Thus, it can be used as
the basis for a two course sequence in communication systems, or a single course on digital com-
munication, at the undergraduate or beginning graduate level. The book also provides a review
or introduction to communication systems for practitioners, easing the path to study of more
advanced graduate texts and the research literature. The prerequisite is a course on signals and
systems, together with an introductory course on probability. The required material on random
processes is included in the text.
A student who masters the material here should be well-prepared for either graduate school
or the telecommunications industry. The student should leave with an understanding of base-
band and passband signals and channels, modulation formats appropriate for these channels,
random processes and noise, a systematic framework for optimum demodulation based on signal
space concepts, performance analysis and power-bandwidth tradeoffs for common modulation
schemes, introduction to communication techniques over dispersive channels, and a hint of the
power of information theory and channel coding. Given the significant ongoing research and
development activity in wireless communication, and the fact that an understanding of wireless
link design provides a sound background for approaching other communication links, material
enabling hands-on discovery of key concepts for wireless system design is interspersed throughout

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the textbook.
The goal of the lecture-style exposition in this book is to clearly articulate a selection of concepts
that I deem fundamental to communication system design, rather than to provide comprehensive
coverage. “Just in time” coverage is provided by organizing and limiting the material so that we
get to core concepts and applications as quickly as possible, and by sometimes asking the reader
to operate with partial information (which is, of course, standard operating procedure in the real
world of engineering design).

Organization
• Chapter 1 provides a perspective on communication systems, including a discussion of the
transition from analog to digital communication and how it colors the selection of material in
this text. Chapter 2 provides a review of signals and systems (biased towards communications
applications), and then discusses the complex baseband representation of passband signals and
systems, emphasizing its critical role in modeling, design and implementation. A software lab
on modeling and undoing phase offsets in complex baseband, while providing a sneak preview of
digital modulation, is included.
• Chapter 2 also includes a section on wireless channel modeling in complex baseband using ray
tracing, reinforced by a software lab which applies these ideas to simulate link time variations
for a lamppost based broadband wireless network.
• Chapter 3 covers analog communication techniques which are relevant even as the world goes
digital, including superheterodyne reception and phase locked loops. Legacy analog modulation
techniques are discussed to illustrate core concepts, as well as in recognition of the fact that
suboptimal analog techniques such as envelope detection and limiter-discriminator detection
may have to be resurrected as we push the limits of digital communication in terms of speed and
power consumption.
• Chapter 4 discusses digital modulation, including linear modulation using constellations such
as Pulse Amplitude Modulation (PAM), Quadrature Amplitude Modulation (QAM), and Phase
Shift Keying (PSK), and orthogonal modulation and its variants. The chapter includes discussion
of the number of degrees of freedom available on a bandlimited channel, the Nyquist criterion
for avoidance of intersymbol interference, and typical choices of Nyquist and square root Nyquist
signaling pulses. We also provide a sneak preview of power-bandwidth tradeoffs (with detailed
discussion postponed until the effect of noise has been modeled in Chapters 5 and 6). A software
lab providing a hands-on feel for Nyquist signaling is included in this chapter.
The material in Chapters 2 through 4 requires only a background in signals and systems.
• Chapter 5 provides a review of basic probability and random variables, and then introduces
random processes. This chapter provides detailed discussion of Gaussian random variables, vec-
tors and processes; this is essential for modeling noise in communication systems. Examples
which provide a preview of receiver operations in communication systems, and computation of
performance measures such as error probability and signal-to-noise ratio (SNR), are provided.
Discussion of circular symmetry of white noise, and noise analysis of analog modulation tech-
niques is placed in an appendix, since this is material that is often skipped in modern courses on
communication systems.
• Chapter 6 covers classical material on optimum demodulation for M-ary signaling in the pres-
ence of additive white Gaussian noise (AWGN). The background on Gaussian random variables,
vectors and processes developed in Chapter 5 is applied to derive optimal receivers, and to analyze
their performance. After discussing error probability computation as a function of SNR, we are
able to combine the materials in Chapters 4 and 6 for a detailed discussion of power-bandwidth
tradeoffs. Chapter 6 concludes with an introduction to link budget analysis, which provides

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guidelines on the choice of physical link parameters such as transmit and receive antenna gains,
and distance between transmitter and receiver, using what we know about the dependence of
error probability as a function of SNR. This chapter includes a software lab which builds on the
Nyquist signaling lab in Chapter 4 by investigating the effect of noise. It also includes another
software lab simulating performance over a time-varying wireless channel, examining the effects
of fading and diversity, and introduces the concept of differential demodulation for avoidance of
explicit channel tracking.
Chapters 2 through 6 provide a systematic lecture-style exposition of what I consider core con-
cepts in communication at an undergraduate level.
• Chapter 7 provides a glimpse of information theory and coding whose goal is to stimulate the
reader to explore further using more advanced resources such as graduate courses and textbooks.
It shows the critical role of channel coding, provides an initial exposure to information-theoretic
performance benchmarks, and discusses belief propagation in detail, reinforcing the basic con-
cepts through a software lab.
• Chapter 8 provides a first exposure to the more advanced topics of communication over dis-
persive channels, and of multiple antenna systems, often termed space-time communication, or
Multiple Input Multiple Output (MIMO) communication. These topics are grouped together be-
cause they use similar signal processing tools. We emphasize lab-style “discovery” in this chapter
using three software labs, one on adaptive linear equalization for singlecarrier modulation, one on
basic OFDM transceiver operations, and one on MIMO signal processing for space-time coding
and spatial multiplexing. The goal is for students to acquire hands-on insight that hopefully
motivates them to undertake a deeper and more systematic investigation.
• Finally, the epilogue contains speculation on future directions in communications research and
technology. The goal is to provide a high-level perspective on where mastery of the introductory
material in this textbook could lead, and to argue that the innovations that this field has already
seen set the stage for many exciting developments to come.
The role of software: Software problems and labs are integrated into the text, while “code frag-
ments” implementing core functionalities provided in the text. While code can be provided
online, separate from the text (and indeed, sample code is made available online for instruc-
tors), code fragments are integrated into the text for two reasons. First, they enable readers to
immediately see the software realization of a key concept as they read the text. Second, I feel
that students would learn more by putting in the work of writing their own code, building on
these code fragments if they wish, rather than using code that is easily available online. The
particular software that we use is Matlab, because of its widespread availability, and because of
its importance in design and performance evaluation in both academia and industry. However,
the code fragments can also be viewed as “pseudocode,” and can be easily implemented using
other software packages or languages. Block-based packages such as Simulink (which builds upon
Matlab) are avoided here, because the use of software here is pedagogical rather than aimed at,
say, designing a complete system by putting together subsystems as one might do in industry.

Suggestions for using this book

I view Chapter 2 (complex baseband), Chapter 4 (digital modulation), and Chapter 6 (optimum
demodulation) as core material that must be studied to understand the concepts underlying
modern communication systems. Chapter 6 relies on the probability and random processes
material in Chapter 5, especially the material on jointly Gaussian random variables and WGN,
but the remaining material in Chapter 5 can be covered selectively, depending on the students’
background. Chapter 3 (analog communication techniques) is designed such that it can be
completely skipped if one wishes to focus solely on digital communication. Finally, Chapter

11
7 and Chapter 8 contain glimpses of advanced material that can be sampled according to the
instructor’s discretion. The qualitative discussion in the epilogue is meant to provide the student
with perspective, and is not intended for formal coverage in the classroom.
In my own teaching at UCSB, this material forms the basis for a two-course sequence, with
Chapters 2-4 covered in the first course, and Chapters 5-6 covered in the second course, with the
dispersive channels portion of Chapter 8 providing the basis for the labs in the second course.
The content of these courses are constantly being revised, and it is anticipated that the material
on channel coding and MIMO may displace some of the existing material in the future. UCSB is
on a quarter system, hence the coverage is fast-paced, and many topics are omitted or skimmed.
There is ample material here for a two-semester undergraduate course sequence. For a single
one-semester course, one possible organization is to cover Chapter 4, a selection of Chapter 5,
Chapter 6, and if time permits, Chapter 7.

12
Acknowledgements

This book grew out of lecture notes for an undergraduate elective course sequence in communi-
cations at UCSB, and I am grateful to the succession of students who have used, and provided
encouraging comments on, the evolution of the course sequence and the notes. I would also like
to acknowledge faculty in the communications area at UCSB who were kind enough to give me
a “lock” on these courses over the past few years, as I was developing this textbook.
The first priority in a research university is to run a vibrant research program, hence I must
acknowledge the extraordinarily capable graduate students in my research group over the years
this textbook was developed. They have done superb research with minimal supervision from
me, and the strength of their peer interactions and collaborations is what gave me the mental
space, and time, needed to write this textbook. Current and former group members who have
directly helped with aspects of this book include Andrew Irish, Babak Mamandipoor, Dinesh
Ramasamy, Maryam Eslami Rasekh, Sumit Singh, Sriram Venkateswaran, and Aseem Wadhwa.
I gratefully acknowledge the funding agencies that have provided support for our research group in
recent years, including the National Science Foundation (NSF), the Army Research Office (ARO),
the Defense Advanced Research Projects Agency (DARPA), and the Systems on Nanoscale In-
formation Fabrics (SONIC), a center supported by DARPA and Microelectronics Advanced Re-
search Corporation (MARCO). One of the primary advantages of a research university is that
undergraduate education is influenced, and kept up to date, by cutting edge research. This text-
book embodies this paradigm both in its approach (an emphasis on what one can do with what
one learns) and content (emphasis of concepts that are fundamental background for research in
the area).
I thank Phil Meyler and his colleagues at Cambridge University Press for encouraging me to ini-
tiate this project, and for their blend of patience and persistence in getting me to see it through
despite a host of other commitments. I also thank the anonymous reviewers of the book pro-
posal and sample chapters sent to Cambridge several years back for their encouragement and
constructive comments. I am also grateful to a number of faculty colleagues who have given
encouragement, helpful suggestions, and pointers to alternative pedagogical approaches: Pro-
fessor Soura Dasgupta (University of Iowa), Professor Jerry Gibson (UCSB), Professor Gerhard
Kramer (Technische Universitat Munchen, Munich, Germany), Professor Phil Schniter (Ohio
State University), and Professor Venu Veeravalli (University of Illinois at Urbana-Champaign).
Finally, I would like to thank my wife and children for always being the most enjoyable and
interesting people to spend time with. Recharging my batteries in their company, and that of
our many pets, is what provides me with the energy needed for an active professional life.

13
14
Chapter 1

Introduction

This textbook provides an introduction to the conceptual underpinnings of communication tech-

nologies. Most of us directly experience such technologies daily: browsing (and audio/video
streaming from) the Internet, sending/receiving emails, watching television, or carrying out a
phone conversation. Many of these experiences occur on mobile devices that we carry around
with us, so that we are always connected to the cyberworld of modern communication systems.
In addition, there is a huge amount of machine-to-machine communication that we do not di-
rectly experience, but which are indispensable for the operation of modern society. Examples
include signaling between routers on the Internet, or between processors and memories on any
computing device.
We define communication as the process of information transfer across space or time. Commu-
nication across space is something we have an intuitive understanding of: for example, radio
waves carry our phone conversation between our cell phone and the nearest base station, and
coaxial cables (or optical fiber, or radio waves from a satellite) deliver television from a remote
location to our home. However, a moment’s thought shows that that communication across time,
or storage of information, is also an everyday experience, given our use of storage media such as
compact discs (CDs), digital video discs (DVDs), hard drives and memory sticks. In all of these
instances, the key steps in the operation of a communication link are as follows:
(a) insertion of information into a signal, termed the transmitted signal, compatible with the
physical medium of interest.
(b) propagation of the signal through the physical medium (termed the channel) in space or
time;
(c) extraction of information from the signal (termed the received signal) obtained after propa-
gation through the medium.
In this book, we study the fundamentals of modeling and design for these steps.
Chapter Plan: In Section 1.1, we provide a high-level description of analog and digital com-
munication systems, and discuss why digital communication is the inevitable design choice in
modern systems. In Section 1.2, we briefly provide a technological perspective on recent devel-
opments in communication. We do not attempt to provide a comprehensive discussion of the
fascinating history of communication: thanks to the advances in communication that brought us
the Internet, it is easy to look it up online! A discussion of the scope of this textbook is provided
in Section 1.3.

1.1 Analog or Digital?

Even without defining information formally, we intuitively understand that speech, audio, and
video signals contain information. We use the term message signals for such signals, since these

15
are the messages we wish to convey over a communication system. In their original form–
both during generation and consumption–these message signals are analog: they are continuous
time signals, with the signal values also lying in a continuum. When someone plays the violin,
an analog acoustic signal is generated (often translated to an analog electrical signal using a
microphone). Even when this music is recorded onto a digital storage medium such as a CD (using
the digital communication framework outlined in Section 1.1.2), when we ultimately listen to the
CD being played on an audio system, we hear an analog acoustic signal. The transmitted signals
corresponding to physical communication media are also analog. For example, in both wireless
and optical communication, we employ electromagnetic waves, which correspond to continuous
time electric and magnetic fields taking values in a continuum.

1.1.1 Analog communication

Information Information
Modulator Channel Demodulator
Source consumer

Message Transmitted Received Message

signal signal signal signal

Figure 1.1: Block diagram for an analog communication system. The modulator transforms
the message signal into the transmitted signal. The channel distorts and adds noise to the
transmitted signal. The demodulator extracts an estimate of the message signal from the received
signal arriving from the channel.

Given the analog nature of both the message signal and the communication medium, a natural
design choice is to map the analog message signal (e.g., an audio signal, translated from the
acoustic to electrical domain using a microphone) to an analog transmitted signal (e.g., a radio
wave carrying the audio signal) that is compatible with the physical medium over which we wish
to communicate (e.g., broadcasting audio over the air from an FM radio station). This approach
to communication system design, depicted in Figure 1.1, is termed analog communication. Early
communication systems were all analog: examples include AM (amplitude modulation) and FM
(frequency modulation) radio, analog television, first generation cellular phone technology (based
on FM), vinyl records, audio cassettes, and VHS or beta videocassettes
While analog communication might seem like the most natural option, it is in fact obsolete. Cel-
lular phone technologies from the second generation onwards are digital, vinyl records and audio
cassettes have been supplanted by CDs, and videocassettes by DVDs. Broadcast technologies
such as radio and television are often slower to upgrade because of economic and political factors,
but digital broadcast radio and television technologies are either replacing or sidestepping (e.g.,
via satellite) analog FM/AM radio and television broadcast. Let us now define what we mean by
digital communication, before discussing the reasons for the inexorable trend away from analog
and towards digital communication.

1.1.2 Digital communication

The conceptual basis for digital communication was established in 1948 by Claude Shannon,
when he founded the field of information theory. There are two main threads to this theory:
• Source coding and compression: Any information-bearing signal can be represented ef-
ficiently, to within a desired accuracy of reproduction, by a digital signal (i.e., a discrete time
signal taking values from a discrete set), which in its simplest form is just a sequence of binary

16
digits (zeros or ones), or bits. This is true whether the information source is text, speech, au-
dio or video. Techniques for performing the mapping from the original source signal to a bit
sequence are generically termed source coding. They often involve compression, or removal of
redundancy, in a manner that exploits the properties of the source signal (e.g., the heavy spatial
correlation among adjacent pixels in an image can be exploited to represent it more efficiently
than a pixel-by-pixel representation).
• Digital information transfer: Once the source encoding is done, our communication task re-
duces to reliably transferring the bit sequence at the output of the source encoder across space or
time, without worrying about the original source and the sophisticated tricks that have been used
to encode it. The performance of any communication system depends on the relative strengths
of the signal and noise or interference, and the distortions imposed by the channel. Shannon
showed that, once we fix these operational parameters for any communication channel, there
exists a maximum possible rate of reliable communication, termed the channel capacity. Thus,
given the information bits at the output of the source encoder, in principle, we can transmit them
reliably over a given link as long as the information rate is smaller than the channel capacity,
and we cannot transmit them reliably if the information rate is larger than the channel capac-
ity. This sharp transition between reliable and unreliable communication differs fundamentally
from analog communication, where the quality of the reproduced source signal typically degrades
gradually as the channel conditions get worse.
A block diagram for a typical digital communication system based on these two threads is shown
in Figure 1.2. We now briefly describe the role of each component, together with simplified
examples of its function.

Information Source Information

Source Encoder bits
Message
signal

Information Channel Channel Information

Modulator Channel Demodulator
bits encoder decoder bits

Coded Transmitted Received Estimates

Digital
bits signal signal of coded bits
communication
link

Information Source Information

bits Decoder Consumer

Message
signal

Figure 1.2: Components of a digital communication system.

Source encoder: As already discussed, the source encoder converts the message signal into a
sequence of information bits. The information bit rate depends on the nature of the message
signal (e.g., speech, audio, video) and the application requirements. Even when we fix the class
of message signals, the choice of source encoder is heavily dependent on the setting. For example,
video signals are heavily compressed when they are sent over a cellular link to a mobile device,
but are lightly compressed when sent to an high definition television (HDTV) set. A cellular link

17
can support a much smaller bit rate than, say, the cable connecting a DVD player to an HDTV
set, and a smaller mobile display device requires lower resolution than a large HDTV screen. In
general, the source encoder must be chosen such that the bit rate it generates can be supported
by the digital communication link we wish to transfer information over. Other than this, source
coding can be decoupled entirely from link design (we comment further on this a bit later).
Example: A laptop display may have resolution 1024 × 768 pixels. For a grayscale digital image,
the intensity for each pixel might be represented by 8 bits. Multiplying by the number of
pixels gives us about 6.3 million bits, or about 0.8 Mbyte (a byte equals 8 bits). However,
for a typical image, the intensities for neighboring pixels are heavily correlated, which can be
exploited for significantly reducing the number of bits required to represent the image, without
noticeably distorting it. For example, one could take a two-dimensional Fourier transform, which
concentrates most of the information in the image at lower frequencies and then discard many
of the high frequency coefficients. There are other possible transforms one could use, and also
several more processing stages, but the bottomline is that, for natural images, state of the art
image compression algorithms can provide 10X compression (i.e., reduction in the number of bits
relative to the original uncompressed digital image) with hardly any perceptual degradation. Far
more aggressive compression ratios are possible if we are willing to tolerate more distortion. For
video, in addition to the spatial correlation exploited for image compression, we can also exploit
temporal correlation across successive frames.
Channel encoder: The channel encoder adds redundancy to the information bits obtained
from the source encoder, in order to facilitate error recovery after transmission over the channel.
It might appear that we are putting in too much work, adding redundancy just after the source
encoder has removed it. However, the redundancy added by the channel encoder is tailored to
the channel over which information transfer is to occur, whereas the redundancy in the original
message signal is beyond our control, so that it would be inefficient to keep it when we transmit
the signal over the channel.
Example: The noise and distortion introduced by the channel can cause errors in the bits we
send over it. Consider the following abstraction for a channel: we can send a string of bits (zeros
or ones) over it, and the channel randomly flips each bit with probability 0.01 (i.e., the channel
has a 1% error rate). If we cannot tolerate this error rate, we could repeat each bit that we wish
to send three times, and use a majority rule to decide on its value. Now, we only make an error
if two or more of the three bits are flipped by the channel. It is left as an exercise to calculate
that an error now happens with probability approximately 0.0003 (i.e., the error rate has gone
down to 0.03%). That is, we have improved performance by introducing redundancy. Of course,
there far more sophisticated and efficient techniques for introducing redundancy than the simple
repetition strategy just described; see Chapter 7.
Modulator: The modulator maps the coded bits at the output of the channel encoder to a
transmitted signal to be sent over the channel. For example, we may insist that the transmitted
signal fit within a given frequency band and adhere to stringent power constraints in a wireless
system, where interference between users and between co-existing systems is a major concern.
Unlicensed WiFi transmissions typically occupy 20-40 MHz of bandwidth in the 2.4 or 5 GHz
bands. Transmissions in fourth generation cellular systems may often occupy bandwidths ranging
from 1-20 MHz at frequencies ranging from 700 MHz to 3 GHz. While these signal bandwidths
are being increased in an effort to increase data rates (e.g., up to 160 GHz for emerging WiFi
standards, and up to 100 MHz for emerging cellular standards), and new frequency bands are
being actively explored (see the epilogue for more discussion), the transmitted signal still needs
to be shaped to fit within certain spectral constraints.
Example: Suppose that we send bit value 0 by transmitting the signal s(t), and bit value 1 by
transmitting −s(t). Even for this simple example, we must design the signal s(t) so it fits within
spectral constraints (e.g., two different users may use two different segments of spectrum to avoid
interfering with each other), and we must figure out how to prevent successive bits of the same
user from interfering with each other. For wireless communication, these signals are voltages

18
generated by circuits coupled to antennas, and are ultimately emitted as electromagnetic waves
from the antennas.
The channel encoder and modulator are typically jointly designed, keeping in mind the antici-
pated channel conditions, and the result is termed a coded modulator.
Channel: The channel distorts and adds noise, and possibly interference, to the transmitted sig-
nal. Much of our success in developing communication technologies has resulted from being able
to optimize communication strategies based on accurate mathematical models for the channel.
Such models are typically statistical, and are developed with significant effort using a combi-
nation of measurement and computation. The physical characteristics of the communication
medium vary widely, and hence so do the channel models. Wireline channels are typically well
modeled as linear and time-invariant, while optical fiber channels exhibit nonlinearities. Wireless
mobile channels are particularly challenging because of the time variations caused by mobility,
and due to the potential for interference due to the broadcast nature of the medium. The link
design also depends on system-level characteristics, such as whether or not the transmitter has
feedback regarding the channel, and what strategy is used to manage interference.
Example: Consider communication between a cellular base station and a mobile device. The elec-
tromagnetic waves emitted by the base station can reach the mobile’s antennas through multiple
paths, including bounces off streets and building surfaces. The received signal at the mobile can
be modeled as multiple copies of the transmitted signal with different gains and delays. These
gains and delays change due to mobility, but the rate of change is often slow compared to the
data rate, hence over short intervals, we can get away with modeling the channel as a linear
time-invariant system that the transmitted signal goes through before arriving at the receiver.
Demodulator: The demodulator processes the signal received from the channel to produce bit
estimates to be fed to the channel decoder. It typically performs a number of signal processing
tasks, such as synchronization of phase, frequency and timing, and compensating for distortions
induced by the channel.
Example: Consider the simplest possible channel model, where the channel just adds noise to
the transmitted signal. In our earlier example of sending ±s(t) to send 0 or 1, the demodulator
must guess, based on the noisy received signal, which of these two options is true. It might
make a hard decision (e.g., say that it guess that 0 was sent), or hedge its bets, and make a soft
decision, saying, for example, that it is 80% sure that the transmitted bit is a zero. There are
a host of other functions that we have swept under the rug: before making any decisions, the
demodulator has to perform functions such as synchronization (making sure that the receiver’s
notion of time and frequency is consistent with the transmitter’s) and equalization (compensating
for the distortions due to the channel).
Channel decoder: The channel decoder processes the imperfect bit estimates provided by
the demodulator, and exploits the controlled redundancy introduced by the channel encoder to
estimate the information bits.
Example: The channel decoder takes the guesses from the demodulator and uses the redundancies
in the channel code to clean up the decisions. In our simple example of repeating every bit three
times, it might use a majority rule to make its final decision if the demodulator is putting out
hard decisions. For soft decisions, it might use more sophisticated combining rules with improved
performance.
While we have described the demodulator and decoder as operating separately and in sequence
for simplicity, there can be significant benefits from iterative information exchange between the
two. In addition, for certain coded modulation strategies in which channel coding and modulation
are tightly coupled, the demodulator and channel decoder may be integrated into a single entity.
Source decoder: The source decoder processes the estimated information bits at the output
of the channel decoder to obtain an estimate of the message. The message format may or may
not be the same as that of the original message input to the source encoder: for example, the
source encoder may translate speech to text before encoding into bits, and the source decoder

19
may output a text message to the end user.
Example: For the example of a digital image considered earlier, the compressed image can be
translated back to a pixel-by-pixel representation by taking the inverse spatial Fourier transform
of the coefficients that survived the compression.
We are now ready to compare analog and digital communication, and discuss why the trend
towards digital is inevitable.

1.1.3 Why digital?

Comparing the block diagrams for analog and digital communication in Figures 1.1 and 1.2,
respectively, we see that the digital communication system involves far more processing. How-
ever, this is not an obstacle for modern transceiver design, due to the exponential increase in
the computational power of low-cost silicon integrated circuits. Digital communication has the
following key advantages.
Optimality: For a point-to-point link, it is optimal to separately optimize source coding and
channel coding, as long we do not mind the delay and processing incurred in doing so. Due
to this source-channel separation principle, we can leverage the best available source codes and
the best available channel codes in designing a digital communication system, independently
of each other. Efficient source encoders must be highly specialized. For example, state of the
art speech encoders, video compression algorithms, or text compression algorithms are very
different from each other, and are each the result of significant effort over many years by a large
community of researchers. However, once source encoding is performed, the coded modulation
scheme used over the communication link can be engineered to transmit the information bits
reliably, regardless of what kind of source they correspond to, with the bit rate limited only
by the channel and transceiver characteristics. Thus, the design of a digital communication
link is source-independent and channel-optimized. In contrast, the waveform transmitted in an
analog communication system depends on the message signal, which is beyond the control of the
link designer, hence we do not have the freedom to optimize link performance over all possible
communication schemes. This is not just a theoretical observation: in practice, huge performance
gains are obtained from switching from analog to digital communication.
Scalability: While Figure 1.2 shows a single digital communication link between source en-
coder and decoder, under the source-channel separation principle, there is nothing preventing
us from inserting additional links, putting the source encoder and decoder at the end points.
This is because digital communication allows ideal regeneration of the information bits, hence
every time we add a link, we can focus on communicating reliably over that particular link. (Of
course, information bits do not always get through reliably, hence we typically add error recovery
mechanisms such as retransmission, at the level of an individual link or end-to-end.) Another
consequence of the source-channel separation principle is that, since information bits are trans-
ported without interpretation, the same link can be used to carry multiple kinds of messages. A
particularly useful approach is to chop the information bits up into discrete chunks, of packets,
which can then be processed independently on each link. These properties of digital communica-
tion are critical for enabling massively scalable, general purpose, communication networks such
as the Internet. Such networks can have large numbers of digital communication links, possibly
with different characteristics, independently engineered to provide bit pipes that can support
data rates. Messages of various kinds, after source encoding, are reduced to packets, and these
packets are switched along different paths along the network, depending on the identities of the
source and destination nodes, and the loads on different links in the network. None of this would
be possible with analog communication: link performance in an analog communication system
depends on message properties, and successive links incur noise accumulation, which limits the
number of links which can be cascaded.

20
The preceding makes it clear that source-channel separation is crucial in the formation and growth
of modern communication networks. It is worth noting, however, that joint source-channel design
can provide better performance in some settings, especially when there are constraints on delay
or complexity, or if multiple users are being supported simultaneously on a given communication
medium. In practice, this means that “local” violations of the separation principle (e.g., over a
wireless last hop in a communication network) may be a useful design trick.

1.2 A Technology Perspective

We now discuss some technology trends and concepts that have driven the astonishing growth
in communication systems in the past two decades, and that are expected to impact future
developments in this area. Our discussion is structured in terms of big technology “stories.”
Technology story 1: The Internet. Some of the key ingredients that contributed to its
growth and the essential role it plays in our lives are as follows:
• Any kind of message can be chopped up into packets and routed across the network, using an
Internet Protocol (IP) that is simple to implement in software;
• Advances in optical fiber communication and high-speed digital hardware enable a super-fast
“core” of routers connected by very high-speed, long-range links, that enable world-wide coverage;
• The World Wide Web, or web, makes it easy to organize information into interlinked hypertext
documents which can be browsed from anywhere in the world;
• The digitization of content (audio, video, books) means that ultimately “all” information is
expected to be available on the web;
• Search engines enable us to efficiently search for this information;
• Connectivity applications such as email, teleconferencing, videoconferencing and online social
networks have become indispensable in our daily lives.
Technology story 2: Wireless. Cellular mobile networks are everywhere, and are based on
the breakthrough concept that ubiquitous tetherless connectivity can be provided by breaking
the world into cells, with “spatial reuse” of precious spectrum resources in cells that are “far
enough” apart. Base stations serve mobiles in their cells, and hand them off to adjacent base
stations when the mobile moves to another cell. While cellular networks were invented to support
voice calls for mobile users, today’s mobile devices (e.g., “smart phones” and tablet computers)
are actually powerful computers with displays large enough for users to consume video on the go.
Thus, cellular networks must now support seamless access to the Internet. The billions of mobile
devices in use easily outnumber desktop and laptop computers, so that the most important parts
of the Internet today are arguably the cellular networks at its edge. Mobile service providers are
having great difficulty keeping up with the increase in demand resulting from this convergence
of cellular and Internet; by some estimates, the capacity of cellular networks must be scaled up
by several orders of magnitude, at least in densely populated urban areas! As discussed in the
epilogue, a major challenge for the communication researcher and technologist, therefore, is to
come up with the breakthroughs required to deliver such capacity gains.
Another major success in wireless is WiFi, a catchy term for a class of standardized wireless
local area network (WLAN) technologies based on the IEEE 802.11 family of standards. Cur-
rently, WiFi networks use unlicensed spectrum in the 2.4 and 5 GHz bands, and have come into
widespread use in both residential and commercial environments. WiFi transceivers are now
incorporated into almost every computer and mobile device. One way of alleviating the cellular
capacity crunch that was just mentioned is to offload Internet access to the nearest WiFi net-
work. Of course, since different WiFi networks are often controlled by different entities, seamless
switching between cellular and WiFi is not always possible.
It is instructive to devote some thought to the contrast between cellular and WiFi technologies.
Cellular transceivers and networks are far more tightly engineered. They employ spectrum that

21
mobile operators pay a great deal of money to license, hence it is critical to use this spectrum
efficiently. Furthermore, cellular networks must provide robust wide-area coverage in the face of
rapid mobility (e.g., automobiles at highway speeds). In contrast, WiFi uses unlicensed (i.e., free!)
spectrum, must only provide local coverage, and typically handles much slower mobility (e.g.,
pedestrian motion through a home or building). As a result, WiFi can be more loosely engineered
than cellular. It is interesting to note that despite the deployment of many uncoordinated
WiFi networks in an unlicensed setting, WiFi typically provides acceptable performance, partly
because the relatively large amount of unlicensed spectrum (especially in the 5 GHz band) allows
for channel switching when encountering excessive interference, and partly because of naturally
occurring spatial reuse (WiFi networks that are “far enough” do not interfere with each other).
Of course, in densely populated urban environments with many independently deployed WiFi
networks, the performance can deteriorate significantly, a phenomenon sometimes referred to as
a tragedy of the commons (individually selfish behavior leading to poor utilization of a shared
resource). As we briefly discuss in the epilogue, both the cellular and WiFi design paradigms
need to evolve to meet our future needs.
Technology story 3: Moore’s law. Moore’s “law” is actually an empirical observation at-
tributed to Gordon Moore, one of the co-founders of Intel Corporation. It can be paraphrased as
saying that the density of transistors in an integrated circuit, and hence the amount of compu-
tation per unit cost, can be expected to increase exponentially over time. This observation has
become a self-fulfilling prophecy, because it has been taken up by the semiconductor industry
as a growth benchmark driving their technology roadmap. While Moore’s law may be slowing
down somewhat, it has already had a spectacular impact on the communications industry by
drastically lowering the cost and increasing the speed of digital computation. By converting
analog signals to the digital domain as soon as possible, advanced transceiver algorithms can
be implemented in digital signal processing (DSP) using low-cost integrated circuits, so that re-
search breakthroughs in coding and modulation can be quickly transitioned into products. This
leads to economies of scale that have been critical to the growth of mass market products in both
wireless (e.g., cellular and WiFi) and wireline (e.g., cable modems and DSL) communication.

Figure 1.3: The Internet has a core of routers and servers connected by high-speed fiber links,
with wireless networks hanging off the edge (figure courtesy Aseem Wadhwa).

How do these stories come together? The sketch in Figure 1.3 highlights key building blocks
of the Internet today. The core of the network consists of powerful routers that direct packets
of data from an incoming edge to an outgoing edge, and servers (often housed in large data
centers) that serve up content requested by clients such as personal computers and mobile devices.
The elements in the core network are connected by high-speed optical fiber. Wireless can be
viewed as hanging off the edge of the Internet. Wide area cellular networks may have worldwide
coverage, but each base station is typically connected by a high-speed link to the wired Internet.

22
WiFi networks are wireless local area networks, typically deployed indoors (but potentially also
providing outdoor coverage for low-mobility scenarios) in homes and office buildings, connected to
the Internet via last mile links, which might run over copper wires (a legacy of wired telephony,
with transceivers typically upgraded to support broadband Internet access) or coaxial cable
(originally deployed to deliver cable television, but now also providing broadband Internet access).
Some areas have been upgraded to optical fiber to the curb or even to the home, while some
others might be remote enough to require wireless last mile solutions.

Figure 1.4: A segment of a cellular network with idealized hexagonal shapes (figure courtesy
Aseem Wadhwa).

Zooming in now on cellular networks, Figure 1.4 shows three adjacent cells in a cellular network
with hexagonal cells. A working definition of a cell is that it is the area around a base station
where the signal strength is higher than that from other base stations. Of course, under realistic
propagation conditions, cells are never hexagonal, but the concept of spatial reuse still holds: the
interference between distant cells can be neglected, hence they can use the same communication
resources. For example, in Figure 1.4, we might decide to use three different frequency bands
in the three cells shown, but might then reuse these bands in other cells. Figure 1.4 also shows
that a user may be simultaneously in range of multiple base stations when near cell boundaries.
Crossing these boundaries may result in a handoff from one base station to another. In addition,
near cell boundaries, a mobile device may be in communication with multiple base stations
simultaneously, a concept known as soft handoff.
It is useful for a communication system designer to be aware of the preceding “big picture” of
technology trends and network architectures in order to understand how to direct his or her
talents as these systems continue to evolve (the epilogue contains more detailed speculation
regarding this evolution). However, the first order of business is to acquire the fundamentals
required to get going in this field. These are quite simply stated: a communication system
designer must be comfortable with mathematical modeling (in order to understand the state of
the art, as well as to devise new models as required), and with devising and evaluating signal
processing algorithms based on these models. The goal of this textbook is to provide a first
exposure to such a technical background.

23
1.3 Scope of this Textbook
Referring to the block diagram of a digital communication system in Figure 1.2, our focus in
this textbook is to provide an introduction to design of a digital communication link as shown
inside the dashed box. While we are primarily interested in digital communication, circuit de-
signers implementing such systems must deal with analog waveforms, hence we believe that a
rudimentary background in analog communication techniques, as provided in this book, is useful
for the communication system designer. We do not discuss source encoding and decoding in
this book; these topics are highly specialized and technical, and doing them justice requires an
entire textbook of its own at the graduate level. A detailed outline of the book is provided in
the preface, hence we restrict ourselves here to summarizing the roles of the various chapters:
Chapter 2: introduces the signal processing background required for DSP-centric implementa-
tions of communication transceivers;
Chapter 3: provides just enough background on analog communication techniques (can be
skipped if only focused on digital communication);
Chapter 4: discusses digital modulation techniques;
Chapter 5: provides the probability background required for receiver design, including noise
modeling;
Chapter 6: discusses design and performance analysis of demodulators in digital communication
systems for idealized link models;
Chapter 7: provides an initial exposure to channel coding techniques and benchmarks;
Chapter 8: provides an introduction to approaches for handling channel dispersion, and to mul-
tiple antenna communication;
Epilogue: discusses emerging trends shaping research and development in communications.
Chapters 2, 4 and 6 are core material that must be mastered (much of Chapter 5 is also core
material, but some readers may already have enough probability background that they can skip,
or skim, it). Chapter 3 is highly recommended for communication system designers with interest
in radio frequency circuit design, since it highlights, at a high level, some of the ideas and issues
that come up there. Chapters 7 and 8 are independent of each other, and contain more advanced
material that may not always fit within an undergraduate curriculum. They contain “hands-on”
introductions to these topics via code fragments and software labs that hopefully encourage the
reader to explore further.

1.4 Concept Inventory

The goal of this chapter is to provide an intellectual framework and motivation for the rest of
this textbook. Some of the key concepts are as follows.
• Communication refers to information transfer across either space or time, where the latter
refers to storage media.
• Signals carrying information and signals that can be sent over a communication medium are
both inherently analog (i.e., continuous-time, continuous-valued).
• Analog communication corresponds to transforming an analog message waveform directly into
an analog transmitted waveform at the transmitter, and undoing this transformation at the
receiver.
• Digital communication corresponds to first reducing message waveforms to information bits,
and then transporting these bits over the communication channel.
• Digital communication requires the following steps: source encoding and decoding, modulation
and demodulation, channel encoding and decoding.
• While digital communication requires more processing steps than analog communication, it has
the advantages of optimality and scalability, hence there is an unstoppable trend from analog to

24
digital.
• The growth in communication has been driven by major technology stories including the
Internet, wireless and Moore’s law.
• Key components of the communication system designer’s toolbox are mathematical modeling
and signal processing.

1.5 Endnotes
There are a large number of textbooks on communication systems at both the undergraduate and
graduate level. Undergraduate texts include Haykin [1], Proakis and Salehi [2], Pursley [3], and
Ziemer and Tranter [4]. Graduate texts, which typically focus on digital communication include
Barry, Lee and Messerschmitt [5], Benedetto and Biglieri [6], Madhow [7], and Proakis and Salehi
[8]. The first coherent exposition of the modern theory of communication receiver design is in
the classical (graduate level) textbook by Wozencraft and Jacobs [9]. Other important classical
graduate level texts are Viterbi and Omura [10] and Blahut [11]. More specialized references (e.g.,
on signal processing, information theory, channel coding, wireless communication) are mentioned
in later chapters. In addition to these textbooks, an overview of many important topics can be
found in the recently updated mobile communications handbook [12] edited by Gibson.
This book is intended to be accessible to readers who have never been exposed to communication
systems before. It has some overlap with more advanced graduate texts (e.g., Chapters 2, 4, 5
and 6 here overlap heavily with Chapters 2 and 3 in the author’s own graduate text [7]), and
provides the technical background and motivation required to easily access these more advanced
texts. Of course, the best way to continue building expertise in the field is by actually working
in it. Research and development in this field requires study of the research literature, of more
specialized texts (e.g., on information theory, channel coding, synchronization), and of commer-
cial standards. The Institute for Electrical and Electronics Engineers (IEEE) is responsible for
publication of many conference proceedings and journals in communications: major conferences
include IEEE Global Telecommunications Conference (Globecom), IEEE International Com-
munications Conference (ICC), major journals and magazines include IEEE Communications
Magazine, IEEE Transactions on Communications, IEEE Journal on Selected Areas in Commu-
nications. Closely related fields such as information theory and signal processing have their own
conferences, journals and magazines. Major conferences include the IEEE International Sympo-
sium on Information Theory (ISIT) and IEEE International Conference on Acoustics, Speech and
Signal Processing (ICASSP), journals include the IEEE Transactions on Information Theory and
the IEEE Transactions on Signal Processing. The IEEE also publishes a number of standards
online, such as the IEEE 802 family of standards for local area networks.
A useful resource for learning source coding and data compression, which are not discussed in
this text, is the textbook by Sayood [13]. Textbooks on core concepts in communication networks
include Bertsekas and Gallager [14], Kumar, Manjunath and Kuri [15], and Walrand and Varaiya
[16].

25
26
Chapter 2

Signals and Systems

A communication link involves several stages of signal manipulation: the transmitter transforms
the message into a signal that can be sent over a communication channel; the channel distorts
the signal and adds noise to it; and the receiver processes the noisy received signal to extract
the message. Thus, communication systems design must be based on a sound understanding of
signals, and the systems that shape them. In this chapter, we discuss concepts and terminology
from signals and systems, with a focus on how we plan to apply them in our discussion of
communication systems. Much of this chapter is a review of concepts with which the reader
might already be familiar from prior exposure to signals and systems. However, special attention
should be paid to the discussion of baseband and passband signals and systems (Sections 2.7
and 2.8). This material, which is crucial for our purpose, is typically not emphasized in a first
course on signals and systems. Additional material on the geometric relationship between signals
is covered in later chapters, when we discuss digital communication.
Chapter Plan: After a review of complex numbers and complex arithmetic in Section 2.1, we
provide some examples of useful signals in Section 2.2. We then discuss LTI systems and convolu-
tion in Section 2.3. This is followed by Fourier series (Section 2.4) and Fourier transform (Section
2.5). These sections (Sections 2.1 through Section 2.5) correspond to a review of material that
is part of the assumed background for the core content of this textbook. However, even readers
familiar with the material are encouraged to skim through it quickly in order to gain familiarity
with the notation. This gets us to the point where we can classify signals and systems based
on the frequency band they occupy. Specifically, we discuss baseband and passband signals and
systems in Sections 2.7 and 2.8. Messages are typically baseband, while signals sent over channels
(especially radio channels) are typically passband. We discuss methods for going from baseband
to passband and back. We specifically emphasize the fact that a real-valued passband signal is
equivalent (in a mathematically convenient and physically meaningful sense) to a complex-valued
baseband signal, called the complex baseband representation, or complex envelope, of the pass-
band signal. We note that the information carried by a passband signal resides in its complex
envelope, so that modulation (or the process of encoding messages in waveforms that can be
sent over physical channels) consists of mapping information into a complex envelope, and then
converting this complex envelope into a passband signal. We discuss the physical significance
of the rectangular form of the complex envelope, which corresponds to the in-phase (I) and
quadrature (Q) components of the passband signal, and that of the polar form of the complex
envelope, which corresponds to the envelope and phase of the passband signal. We conclude by
discussing the role of complex baseband in transceiver implementations, and by illustrating its
use for wireless channel modeling.

27
2.1 Complex Numbers

Im(z)
y (x,y)
r
θ
Re(z)
x

Figure 2.1: A complex number z represented in the two-dimensional real plane.

√
A complex number z can be written as z = x+jy, where x and y are real numbers, and j = −1.
We say that x = Re(z) is the real part of z and y = Im(z) is the imaginary part of z. As depicted
in Figure 2.1, it is often advantageous to interpret the complex number z as a two-dimensional
real vector, which can be represented in rectangular form as (x, y) = (Re(z), Im(z)), or in polar
form (r, θ) as p
r = |z| = x2 + y 2
(2.1)
θ = z = tan−1 xy
We can go back from polar form to rectangular form as follows:

x = r cos θ, y = r sin θ (2.2)

Complex conjugation: For a complex number z = x + jy = rejθ , its complex conjugate

z ∗ = x − jy = re−jθ (2.3)

That is,
Re(z ∗ ) = Re(z) , Im(z ∗ ) = −Im(z)
(2.4)
|z ∗ | = |z| , z∗ = − z

The real and imaginary parts of a complex number z can be written in terms of z and z ∗ as
follows:
z + z∗ z − z∗
Re(z) = , Im(z) = (2.5)
2 2j

Euler’s formula: This formula is of fundamental importance in complex analysis, and relates
the rectangular and polar forms of a complex number:

ejθ = cos θ + j sin θ (2.6)

The complex conjugate of ejθ is given by

∗
e−jθ = ejθ = cos θ − j sin θ

28
We can express cosines and sines in terms of ejθ and its complex conjugate as follows:

ejθ + e−jθ
ejθ − e−jθ
Re ejθ = = cos θ , Im ejθ = = sin θ (2.7)
2 2j

Applying Euler’s formula to (2.1), we can write

z = x + jy = r cos θ + jr sin θ = rejθ (2.8)

Being able to go back and forth between the rectangular and polar forms of a complex number
is useful. For example, it is easier to add in the rectangular form, but it is easier to multiply in
the polar form.
Complex Addition: For two complex numbers z1 = x1 + jy1 and z2 = x2 + jy2 ,
z1 + z2 = (x1 + x2 ) + j(y1 + y2 ) (2.9)
That is,
Re(z1 + z2 ) = Re(z1 ) + Re(z2 ) , Im(z1 + z2 ) = Im(z1 ) + Im(z2 ) (2.10)
Complex Multiplication (rectangular form): For two complex numbers z1 = x1 + jy1 and
z2 = x2 + jy2 ,
z1 z2 = (x1 x2 − y1 y2 ) + j(y1 x2 + x1 y2 ) (2.11)
This follows simply by multiplying out, and setting j 2 = −1. We have
Re(z1 z2 ) = Re(z1 )Re(z2 ) − Im(z1 )Im(z2 ) , Im(z1 z2 ) = Im(z1 )Re(z2 ) + Re(z1 )Im(z2 ) (2.12)
Note that, using the rectangular form, a single complex multiplication requires four real multi-
plications.
Complex Multiplication (polar form): Complex multiplication is easier when the numbers
are expressed in polar form. For z1 = r1 ejθ1 , z2 = r2 ejθ2 , we have

z1 z2 = r1 r2 ej(θ1 +θ2 ) (2.13)

That is,
|z1 z2 | = |z1 ||z2 | , z1 z2 = z1 + z2 (2.14)
Division: For two complex numbers z1 = x1 + jy1 = r1 ejθ1 and z2 = x2 + jy2 = r2 ejθ2 (with
z2 6= 0, i.e., r2 > 0), it is easiest to express the result of division in polar form:

z1 /z2 = (r1 /r2 )ej(θ1 −θ2 ) (2.15)

That is,
|z1 /z2 | = |z1 |/|z2 | , z1 /z2 = z1 − z2 (2.16)
In order to divide using rectangular form, it is convenient to multiply numerator and denominator
by z2∗ , which gives
(x1 + jy1 )(x2 − jy2 )
z1 /z2 = z1 z2∗ /(z2 z2∗ ) = z1 z2∗ /|z2 |2 =
x22 + y22
Multiplying out as usual, we get
(x1 x2 + y1 y2 ) + j (−x1 y2 + y1 x2 )
z1 /z2 = (2.17)
x22 + y22

29
Example 2.1.1 (Computations with complex numbers) Consider the complex numbers
z1 = 1 + j and z2 = 2e−jπ/6 . Find z1 + z2 , z1 z2 , and z1 /z2 . Also specify z1∗ , z2∗ .
For complex addition, it is convenient to express both numbers in rectangular form. Thus,
√
z2 = 2 (cos(−π/6) + j sin(−π/6)) = 3 − j

and √ √
z1 + z2 = (1 + j) + ( 3 − j) = 3 + 1
For complex multiplication
√ jπ/4 and division, it is convenient to express both numbers in polar form.
We obtain z1 = 2e by applying (2.1). Now, from (2.11), we have
√ √ √
z1 z2 = 2ejπ/4 2e−jπ/6 = 2 2ej(π/4−π/6) = 2 2ejπ/12

Similarly, √
2ejπ/4 1 1
z1 /z2 = −jπ/6
= √ ej(π/4+π/6) = √ ej5π/12
2e 2 2
Multiplication using the rectangular forms of the complex numbers yields the following:
√ √ √ √  √ 
z1 z2 = (1 + j)( 3 − j) = 3 − j + 3j + 1 = 3+1 +j 3−1
√ √
Note that z1∗ = 1 − j = 2e−jπ/4 and z2∗ = 2ejπ/6 = 3 + j. Division using rectangular forms
gives √ √
∗ 2
√ 2 3−1 3+1
z1 /z2 = z1 z2 /|z2 | = (1 + j)( 3 + j)/2 = +j
4 4

No need to memorize trigonometric identities any more: Once we can do computations

using complex numbers, we can use Euler’s formula to quickly derive well-known trigonometric
identities involving sines and cosines. For example,

cos(θ1 + θ2 ) = Re ej(θ1 +θ2 )

But
ej(θ1 +θ2 ) = ejθ1 ejθ2 = (cos θ1 + j sin θ1 ) (cos θ2 + j sin θ2 )
= (cos θ1 cos θ2 − sin θ1 sin θ2 ) + j (cos θ1 sin θ2 + sin θ1 cos θ2 )
Taking the real part, we can read off the identity

cos(θ1 + θ2 ) = cos θ1 cos θ2 − sin θ1 sin θ2 (2.18)

Moreover, taking the imaginary part, we can read off

sin(θ1 + θ2 ) = cos θ1 sin θ2 + sin θ1 cos θ2 (2.19)

2.2 Signals
Signal: A signal s(t) is a function of time (or some other independent variable, such as fre-
quency, or spatial coordinates) which has an interesting physical interpretation. For example, it
is generated by a transmitter, or processed by a receiver. While physically realizable signals such
as those sent over a wire or over the air must take real values, we shall see that it is extremely

30
useful (and physically meaningful) to consider a pair of real-valued signals, interpreted as the
real and imaginary parts of a complex-valued signal. Thus, in general, we allow signals to take
complex values.
Discrete versus Continuous Time: We generically use the notation x(t) to denote continuous
time signals (t taking real values), and x[n] to denote discrete time signals (n taking integer
values). A continuous time signal x(t) sampled at rate Ts produces discrete time samples x(nTs +
t0 ) (t0 an arbitrary offset), which we often denote as a discrete time signal x[n]. While signals
sent over a physical communication channel are inherently continuous time, implementations at
both the transmitter and receiver make heavy use of discrete time implementations on digitized
samples corresponding to the analog continuous time waveforms of interest.
We now introduce some signals that recur often in this text.
Sinusoid: This is a periodic function of time of the form
s(t) = A cos(2πf0 t + θ) (2.20)
where A > 0 is the amplitude, f0 is the frequency, and θ ∈ [0, 2π] is the phase. By setting θ = 0,
we obtain a pure cosine A cos 2πfc t, and by setting θ = − π2 , we obtain a pure sine A sin 2πfc t.
In general, using (2.18), we can rewrite (2.20) as
s(t) = Ac cos 2πf0 t − As sin 2πf0 t (2.21)
where Ac = A cos θ and As = A sin θ are real numbers. Using Euler’s formula, we can write
Aejθ = Ac + jAs (2.22)
Thus, the parameters of a sinusoid at frequency f0 can be represented by the complex number in
(2.22), with
p (2.20) using the polar form, and (2.21) the rectangular form, of this number. Note
that A = A2c + A2s and θ = tan−1 A s
Ac
.
Clearly, sinusoids with known amplitude, phase and frequency are perfectly predictable, and
hence cannot carry any information. As we shall see, information can be transmitted by making
the complex number Aejθ = Ac + jAs associated with the parameters of sinusoid vary in a way
that depends on the message to be conveyed. Of course, once this is done, the resulting signal
will no longer be a pure sinusoid, and part of the work of the communication system designer is
to decide what shape such a signal should take in the frequency domain.
We now define complex exponentials, which play a key role in understanding signals and systems
in the frequency domain.
Complex exponential: A complex exponential at a frequency f0 is defined as

s(t) = Aej(2πf0 t+θ) = αej2πf0 t (2.23)

where A > 0 is the amplitude, f0 is the frequency, θ ∈ [0, 2π] is the phase, and α = Aejθ is a
complex number that contains both the amplitude and phase information. Let us now make
three observations. First, note the ease with which we handle amplitude and phase for complex
exponentials: they simply combine into a complex number that factors out of the complex
exponential. Second, by Euler’s formula,

Re Aej(2πf0 t+θ) = A cos(2πf0 t + θ)

so that real-valued sinusoids are “contained in” complex exponentials. Third, as we shall soon
see, the set of complex exponentials {ej2πf t }, where f takes values in (−∞, ∞), form a “basis”
for a large class of signals (basically, for all signals that are of interest to us), and the Fourier
transform of a signal is simply its expansion with respect to this basis. Such observations are

31
 1/a
1/a

t t
−a/2 a/2 −a a

Figure 2.2: The impulse function may be viewed as a limit of tall thin pulses (a → 0 in the
examples shown in the figure).

Unit area
p(t)

s(t)

t
t0

t 0− a1 t0+ a 2

Figure 2.3: Multiplying a signal with a tall thin pulse to select its value at t0 .

key to why complex exponentials play such an important role in signals and systems in general,
and in communication systems in particular.
The Delta, or Impulse, Function: Another signal that plays a crucial role in signals and sys-
tems is the delta function, or the unit impulse, which we denote by δ(t). Physically, we can think
of it as a narrow, tall pulse with unit area: examples are shown in Figure 2.2. Mathematically,
we can think of it as a limit of such pulses as the pulse width shrinks (and hence the pulse height
goes to infinity). Such a limit is not physically realizable, but it serves a very useful purpose in
terms of understanding the structure of physically realizable signals. That is, consider a signal
s(t) that varies smoothly, and multiply it with a tall, thin pulse of unit area, centered at time
t0 , as shown in Figure 2.3. If we now integrate the product, we obtain
Z ∞ Z t0 +a2 Z t0 +a1
s(t)p(t)dt = s(t)p(t)dt ≈ s(t0 ) p(t)dt = s(t0 )
−∞ t0 −a1 t0 −a1

That is, the preceding operation “selects” the value of the signal at time t0 . Taking the limit of
the tall thin pulse as its width a1 + a2 → 0, we get a translated version of the delta function,
namely, δ(t − t0 ). Note that the exact shape of the pulse does not matter in the preceding
argument. The delta function is therefore defined by means of the following sifting property: for
any “smooth” function s(t), we have
Z ∞
s(t)δ(t − t0 )dt = s(t0 ) Sifting property of the impulse (2.24)
−∞

Thus, the delta function is defined mathematically by the way it acts on other signals, rather
than as a signal by itself. However, it is also important to keep in mind its intuitive interpretation
as (the limit of) a tall, thin, pulse of unit area.

32
The following function is useful for expressing signals compactly.
Indicator function: We use IA to denote the indicator function of a set A, defined as

1, x ∈ A
IA (x) =
0, otherwise
The indicator function of an interval is a rectangular pulse, as shown in Figure 2.4.
I (x)
[a,b]

x
a b

Figure 2.4: The indicator function of an interval is a rectangular pulse.

v(t)
u(t)
3
2
1
2
t 1 t
−1 1 −1
−1

Figure 2.5: The functions u(t) = 2(1 − |t|)I[−1,1] (t) and v(t) = 3I[−1,0] (t) + I[0,1] (t) − I[1,2] (t) can
be written compactly in terms of indicator functions.

The indicator function can also be used to compactly express more complex signals, as shown in
the examples in Figure 2.5.
Sinc function: The sinc function, plotted in Figure 2.6, is defined as
sin(πx)
sinc(x) =
πx
where the value at x = 0 is defined as the limit as x → 0 to be sinc(0) = 1. Since | sin(πx)| ≤ 1,
1
we have that |sinc(x)| ≤ π|x| , with equality if and only if x is an odd multiple of 1/2. That is,
1
the sinc function exhibits a sinusoidal variation, with an envelope that decays as |x| .
The analogy between signals and vectors: Even though signals can be complicated functions
of time that live in an infinite-dimensional space, the mathematics for manipulating them are
very similar to those for manipulating finite-dimensional vectors, with sums replaced by integrals.
A key building block of communication theory is the relative geometry of the signals used, which
is governed by the inner products between signals. Inner products for continuous-time signals can
be defined in a manner exactly analogous to the corresponding definitions in finite-dimensional
vector space.
Inner Product: The inner product for two m × 1 complex vectors s = (s[1], ..., s[m])T and
r = (r[1], ..., r[m])T is given by
Xm
hs, ri = s[i]r ∗ [i] = rH s (2.25)
i=1

33
 1

0.8

0.6

0.4

sinc(x)
0.2

−0.2

−0.4
−5 −4 −3 −2 −1 0 1 2 3 4 5
x

Figure 2.6: The sinc function.

Similarly, we define the inner product of two (possibly complex-valued) signals s(t) and r(t) as
follows: Z ∞
hs, ri = s(t)r ∗ (t) dt (2.26)
−∞

The inner product obeys the following linearity properties:

ha1 s1 + a2 s2 , ri = a1 hs1 , ri + a2 hs2 , ri

hs, a1 r1 + a2 r2 i = a∗1 hs, r1i + a∗2 hs, r2 i

where a1 , a2 are complex-valued constants, and s, s1 , s2 , r, r1 , r2 are signals (or vectors). The
complex conjugation when we pull out constants from the second argument of the inner product
is something that we need to maintain awareness of when computing inner products for complex-
valued signals.
Energy and Norm: The energy Es of a signal s is defined as its inner product with itself:
Z ∞
2
Es = ||s|| = hs, si = |s(t)|2 dt (2.27)
−∞

where ||s|| denotes the norm of s. If the energy of s is zero, then s must be zero “almost
everywhere” (e.g., s(t) cannot be nonzero over any interval, no matter how small its length).
For continuous-time signals, we take this to be equivalent to being zero everywhere. With this
understanding, ||s|| = 0 implies that s is zero, which is a property that is true for norms in
finite-dimensional vector spaces.

Example 2.2.1 (Energy computations) Consider s(t) = 2I[0,T ] + jI[T /2,2T ] . Writing it out in
more detail, we have 
 2, 0 ≤ t < T /2
s(t) = 2 + j, T /2 ≤ t < T
j, T ≤ t < 2T


so that its energy is given by

Z T /2 Z T Z 2T
2 2 2
||s|| = 2 dt + |2 + j| dt + |j|2 dt = 4(T /2) + 5(T /2) + T = 11T /2
0 T /2 T

34
As another example, consider s(t) = e−3|t|+j2πt , for which the energy is given by
Z ∞ Z ∞ Z ∞
2 −3|t|+j2πt 2 −6|t|
||s|| = |e | dt = e dt = 2 e−6t dt = 1/3
−∞ −∞ 0

Note that the complex phase term j2πt does not affect the energy, since it goes away when we
take the magnitude.

Power: The power of a signal s(t) is defined as the time average of its energy computed over a
large time interval:
Z To
1 2
Ps = lim |s(t)|2 dt (2.28)
To →∞ To − To
2

Finite energy signals, of course, have zero power.

We see from (2.28) that power is defined as a time average. It is useful to introduce a compact
notation for time averages.
Time average: For a function g(t), define the time average as
To
1
Z
2
g = lim g(t)dt (2.29)
To →∞ To − T2o

That is, we compute the time average over an observation interval of length To , and then let
the observation interval get large. We can now rewrite the power computation in (2.28) in this
notation as follows.
Power: The power of a signal s(t) is defined as

Ps = |s(t)|2 (2.30)

Another time average of interest is the DC value of a signal.

DC value: The DC value of s(t) is defined as s(t).
Let us compute these quantities for the simple example of a complex exponential, s(t) =
Aej(2πf0 t+θ) , where A > 0 is the amplitude, θ ∈ [0, 2π] is the phase, and f0 is a real-valued
frequency. Since |s(t)|2 ≡ A2 for all t, we get the same value when we average it. Thus, the
power is given by Ps = s2 (t) = A2 . For nonzero frequency f0 , it is intuitively clear that all the
power in s is concentrated away from DC, since s(t) = Aej(2πf0 t+θ) ↔ S(f ) = Aejθ δ(f − f0 ).
We therefore see that the DC value is zero. While this is a convincing intuitive argument, it is
instructive to prove this starting from the definition (2.29).
Proving that a complex exponential has zero DC value: For s(t) = Aej(2πf0 t+θ) , the
integral over its period (of length 1/f0 ) is zero. As shown in Figure 2.7, the length L of any
interval I can be written as L = K/f0 + ℓ where K is a nonnegative integer and 0 ≤ ℓ < f10 is
the length of the remaining interval Ir . Since the integral over an integer number of periods is
zero, we have Z Z
s(t)dt = s(t)dt
I Ir

Thus,
A
Z Z
| s(t)dt| = | s(t)dt| ≤ ℓ maxt |s(t)| = Aℓ <
I Ir f0

35
 K/f 0
1/f 0 Interval Ir (length l )

[ ]
Interval I

Figure 2.7: The interval I for computing the time average of a periodic function with period
1/f0 can be decomposed into an integer number K of periods, with the remaining interval Ir of
length ℓ < f10 .

since |s(t)| = A. We therefore obtain

Z To
2
| s(t)dt| ≤ (A/f0 )
− T2o

which yields that the DC value s = 0, since

To
1 A
Z
2
|s| = | lim s(t)dt| ≤ lim =0
To →∞ To − T2o To →∞ f0 To

Essentially the same argument implies that, in general, the time average of a periodic signal
equals the average over a single period. We use this fact without further comment henceforth.
Power and DC value of a sinusoid: For a real-valued sinusoid s(t) = A cos(2πf0 t + θ), we
can use the results derived for complex exponentials above. Using Euler’s identity, a real-valued
sinusoid at f0 is a sum of complex exponentials at ±f0 :
A j(2πf0 t+θ) A −j(2πf0 t+θ)
s(t) = e + e
2 2
Since each complex exponential has zero DC value, we obtain

s=0

That is, the DC value of any real-valued sinusoid is zero.

A2 A2 A2
Ps = s2 (t) = A2 cos2 (2πf0 t + θ) = + cos(4πf0 t + 2θ) =
2 2 2
since the DC value of the sinusoid at 2f0 is zero.

2.3 Linear Time Invariant Systems

System: A system takes as input one or more signals, and produces as output one or more
signals. A system is specified once we characterize its input-output relationship; that is, if we
can determine the output, or response, y(t), corresponding to any possible input x(t) in a given
class of signals of interest.

36
Our primary focus here is on linear time invariant (LTI) systems, which provide good models
for filters at the transmitter and receiver, as well as for the distortion induced by a variety of
channels. We shall see that the input-output relationship is particularly easy to characterize for
such systems.
Linear system: Let x1 (t) and x2 (t) denote arbitrary input signals, and let y1 (t) and y2 (t)
denote the corresponding system outputs, respectively. Then, for arbitrary scalars a1 and a2 , the
response of the system to input a1 x1 (t) + a2 x2 (t) is a1 y1 (t) + a2 y2 (t).
Time invariant system: Let y(t) denote the system response to an input x(t). Then the
system response to a time-shifted version of the input, x1 (t) = x(t − t0 ) is y1 (t) = y(t − t0 ). That
is, a time shift in the input causes an identical time shift in the output.

Example 2.3.1 Examples of linear systems It can (and should) be checked that the following
systems are linear. These examples show that linear systems may or may not be time invariant.

y(t) = 2x(t − 1) − jx(t − 2) time invariant

y(t) = (3 − 2j)x(1 − t) time varying

y(t) = x(t) cos(100πt) − x(t − 1) sin(100πt) time varying
Z t+1
y(t) = x(τ )dτ time invariant
t−1

Example 2.3.2 Examples of time invariant systems It can (and should) be checked that
the following systems are time invariant. These examples show that time invariant systems may
or may not be linear.
y(t) = e2x(t−1) nonlinear
Z t
y(t) = x(τ )e−(t−τ ) dτ linear
−∞
Z t+1
y(t) = x2 (τ )dτ nonlinear
t−1

Linear time invariant system: A linear time invariant (LTI) system is (unsurprisingly) defined
to be a system which is both linear and time invariant. What is surprising, however, is how
powerful the LTI property is in terms of dictating what the input-output relationship must look
like. Specifically, if we know the impulse response of an LTI system (i.e., the output signal
when the input signal is the delta function), then we can compute the system response for any
input signal. Before deriving and stating this result, we illustrate the LTI property using an
example; see Figure 2.8. Suppose that the response of an LTI system to the rectangular pulse
p1 (t) = I[− 1 , 1 ] (t) is given by the trapezoidal waveform h1 (t). We can now compute the system
2 2
response to any linear combination of time shifts of the pulse p(t), as illustrated by the example
in the figure. More generally, P using the LTI property,
P we infer that the response to an input
signal of the form x(t) = i ai p1 (t − ti ) is y(t) = i ai h1 (t − ti ).
Can we extend the preceding idea to compute the system response to arbitrary input signals?
The answer is yes: if we know the system response to thinner and thinner pulses, then we
can approximate arbitrary signals better and better using linear combinations of shifts of these
pulses. Consider p∆ (t) = ∆1 I[− ∆ , ∆ ] (t), where ∆ > 0 is getting smaller and smaller. Note that we
2 2
have normalized the area of the pulse to unity, so that the limit of p∆ (t) as ∆ → 0 is the delta

37
 h 1(t)
p (t)
1
1 1
S
t
−0.5 0.5 t 0 1 2 3

2h1 (t)

2
x(t) = 2 p1 (t) − p (t−1)
1 y(t)
2 2

0 3
=
S + 3 4
1.5 t
t 0 1 2
−0.5
−h1 (t−1)
−1 −1
1 4

−1

Figure 2.8: Given that the response of an LTI system S to the pulse p1 (t) is h1 (t), we can use the
LTI property to infer that the response to x(t) = 2p1 (t) − p1 (t − 1) is y(t) = 2h1 (t) − h1 (t − 1).

x(t)

... ...
t

Figure 2.9: A smooth signal can be approximated as a linear combination of shifts of tall thin
pulses.

38
function. Figure 2.9 shows how to approximate a smooth input signal as a linear combination of
shifts of p∆ (t). That is, for ∆ small, we have
∞
X
x(t) ≈ x∆ (t) = x(k∆)∆p∆ (t − k∆) (2.31)
k=−∞

If the system response to p∆ (t) is h∆ (t), then we can use the LTI property to compute the
response y∆ (t) to x∆ (t), and use this to approximate the response y(t) to the input x(t), as
follows: ∞
X
y(t) ≈ y∆ (t) = x(k∆)∆h∆ (t − k∆) (2.32)
k=−∞

As ∆ → 0, the sums above tend to integrals, and the pulse p∆ (t) tends to the delta function δ(t).
The approximation to the input signal in equation (2.31) becomes exact, with the sum tending
to an integral: Z ∞
lim x∆ (t) = x(t) = x(τ )δ(t − τ )dτ
∆→0 −∞

replacing the discrete time shifts k∆ by the continuous variable τ , the discrete increment ∆ by
the infinitesimal dτ , and the sum by an integral. This is just a restatement of the sifting property
of the impulse. That is, an arbitrary input signal can be expressed as a linear combination of
time-shifted versions of the delta function, where we now consider a continuum of time shifts.
In similar fashion, the approximation to the output signal in (2.32) becomes exact, with the sum
reducing to the following convolution integral:
Z ∞
lim y∆ (t) = y(t) = x(τ )h(t − τ )dτ (2.33)
∆→0 −∞

where h(t) denotes the impulse response of the LTI system.

Convolution and its computation: The convolution v(t) of two signals u1 (t) and u2 (t) is
given by Z ∞ Z ∞
v(t) = (u1 ∗ u2 )(t) = u1 (τ )u2 (t − τ ) dτ = u1 (t − τ )u2 (τ ) dτ (2.34)
−∞ −∞

Note that τ is a dummy variable that is integrated out in order to determine the value of the
signal v(t) at each possible time t. The role of u1 and u2 in the integral can be exchanged. This
can be proved using a change of variables, replacing t − τ by τ . We often drop the time variable,
and write v = u1 ∗ u2 = u2 ∗ u1 .
An LTI system is completely characterized by its impulse response: As derived in
(2.33), the output y of an LTI system is the convolution of the input signal u and the system
impulse response h. That is, y = u ∗ h. From (2.34), we realize that the role of the signal and the
system can be exchanged: that is, we would get the same output y if a signal h is sent through
a system with impulse response u.
Flip and slide: Consider the expression for the convolution in (2.34):
Z ∞
v(t) = u1 (τ )u2 (t − τ ) dτ
−∞

Fix a value of time t at which we wish to evaluate v. In order to compute v(t), we must multiply
two functions of a “dummy variable” τ and then integrate over τ . In particular, s2 (τ ) = u2 (−τ )
is the signal u2 (τ ) flipped around the origin, so that u2 (t − τ ) = u2 (−(τ − t)) = s2 (τ − t) is
s2 (τ ) translated to the right by t (if t < 0, translation to the right by t actually corresponds to

39
translation is to the left by |t|). In short, the mechanics of computing the convolution involves
flipping and sliding one of the signals, multiplying with the other signal, and integrating. Pictures
are extremely helpful when doing such computations by hand, as illustrated by the following
example.
u1 (τ ) u 2( τ ) Flip u 2 (−τ )

τ τ τ
5 11 1 3 −3 −1

u 2 (t−τ )
(a) t−1 < 5
τ
t−3 t−1
u 2 (t−τ )
(b) t−3 < 5, t−1 > 5
Slide by t
τ
t−3 t−1 Different ranges of t
u 2 (t−τ ) depicted in (a)−(e)
(c) t−3 > 5, t−1 < 11
τ
t−3 t−1
u 2 (t−τ )
(d) t−3 < 11, t−1 > 11
τ
t−3 t−1
u 2 (t−τ )
(e) t−3 > 11
τ
t−3 t−1

Figure 2.10: Illustrating the flip and slide operation for the convolution of two rectangular pulses.

v(t)

6 8 12 14 t

Figure 2.11: The convolution of the two rectangular pulses in Example 2.3.3 results in a trape-
zoidal pulse.

Example 2.3.3 Convolving rectangular pulses: Consider the rectangular pulses u1 (t) =
I[5,11] (t) and u2 (t) = I[1,3] (t). We wish to compute the convolution
Z ∞
v(t) = (u1 ∗ u2 )(t) = u1 (τ )u2 (t − τ )dτ
−∞

We now draw pictures of the signals involved in these “flip and slide” computations in order to
figure out the limits of integration for different ranges of t. Figure 2.10 shows that there are five
different ranges of interest, and yields the following result:
(a) For t < 6, u1 (τ )u2 (t − τ ) ≡ 0, so that v(t) = 0.
(b) For 6 < t < 8, u1 (τ )u2 (t − τ ) = 1 for 5 < τ < t − 1, so that
Z t−1
v(t) = dτ = t − 6
5

40
(c) For 8 < t < 12, u1 (τ )u2 (t − τ ) = 1 for t − 3 < τ < t − 1, so that
Z t−1
v(t) = dτ = 2
t−3

(d) For 12 < t < 14, u1 (τ )u2 (t − τ ) = 1 for t − 3 < τ < 11, so that
Z 11
v(t) = dτ = 11 − (t − 3) = 14 − t
t−3

(e) For t > 14, u1 (τ )u2 (t − τ ) ≡ 0, so that v(t) = 0.

The result of the convolution is the trapezoidal pulse sketched in Figure 2.11.

1 1 a

* = −(b+a)/2 (b+a)/2
−a/2 a/2 −b/2 b/2 −(b−a)/2 (b−a)/2

1 1 a

* =
−a/2 a/2 −a/2 a/2 −a a

Figure 2.12: Convolution of two rectangular pulses as a function of pulse durations. The trape-
zoidal pulse reduces to a triangular pulse for equal pulse durations.

It is useful to record the general form of the convolution between two rectangular pulses of the
form I[−a/2,a/2] (t) and I[−b/2,b/2] (t), where we take a ≤ b without loss of generality. The result is
a trapezoidal pulse, which reduces to a triangular pulse for a = b, as shown in Figure 2.12. Once
we know this, using the LTI property, we can infer the convolution of any signals which can be
expressed as a linear combination of shifts of rectangular pulses.
Occasional notational sloppiness can be useful: As the preceding example shows, a con-
volution computation as in (2.34) requires a careful distinction between the variable t at which
the convolution is being evaluated, and the dummy variable τ . This is why we make sure that
the dummy variable does not appear in our notation (s ∗ r)(t) for the convolution between sig-
nals s(t) and r(t). However, it is sometimes convenient to abuse notation and use the notation
s(t) ∗ r(t) instead, as long we remain aware of what we are doing. For example, this enables us
to compactly state the following linear time invariance (LTI) property:
(a1 s1 (t − t1 ) + a2 s2 (t − t2 )) ∗ r(t) = a1 (s1 ∗ r)(t − t1 ) + a2 (s2 ∗ r)(t − t2 )
for any complex gains a1 and a2 , and any time offsets t1 and t2 .

Example 2.3.4 (Modeling a multipath channel) We can get a delayed version of a signal
by convolving it with a delayed impulse as follows:
y1 (t) = u(t) ∗ δ(t − t1 ) = u(t − t1 ) (2.35)

41
To see this, compute
Z Z
y1 (t) = u(τ )δ(t − τ − t1 )dτ = u(τ )δ(τ − (t − t1 ))dτ = u(t − t1 )

where we first use the fact that the delta function is even, and then use its sifting property.

Reflector

LOS
TX antenna path

RX antenna
Reflector

Figure 2.13: Multipath channels typical of wireless communication can include line of sight (LOS)
and reflected paths.

Equation (2.35) immediately tells us how to model multipath channels, in which multiple scat-
tered versions of a transmitted signal u(t) combine to give a received signal y(t) which is a
superposition of delayed versions of the transmitted signal, as illustrated in Figure 2.13:

y(t) = α1 u(t − τ1 ) + ... + αm u(t − τm )

(plus noise, which we have not talked about yet). From (2.35), we see that we can write

y(t) = α1 u(t) ∗ δ(t − τ1 ) + ... + αm u(t) ∗ δ(t − τm ) = u(t) ∗ (α1 δ(t − τ1 ) + ... + αm δ(t − τm ))

That is, we can model the received signal as a convolution of the transmitted signal with a
channel impulse response which is a linear combination of time-shifted impulses:

h(t) = α1 δ(t − τ1 ) + ... + αm δ(t − τm ) (2.36)

Figure 2.14 illustrates how a rectangular pulse spreads as it goes through a multipath channel
with impulse response h(t) = δ(t − 1) − 0.5δ(t − 1.5) + 0.5δ(t − 3.5). While the gains {αk } in this
example are real-valued, as we shall soon see (in Section 2.8), we need to allow both the signal
u(t) and the gains {αk } to take complex values in order to model, for example, signals carrying
information over radio channels.

u(t) Multipath Channel y(t) = (u*h) (t)

h(t)
1
1
1 0.5
0.5
1.5
t t 1 3 3.5 5.5 t
0 2 1 3.5
−0.5
Rectangular pulse −0.5
Channel output

Figure 2.14: A rectangular pulse through a multipath channel.

42
 e j2 π f0 t LTI System H(f 0) e j2 π f0 t

Figure 2.15: Complex exponentials are eigenfunctions of LTI systems.

Complex exponential through an LTI system: In order to understand LTI systems in the
frequency domain, let us consider what happens to a complex exponential u(t) = ej2πf0 t when it
goes through an LTI system with impulse response h(t). The output is given by
R∞
y(t) = (u ∗ h)(t) = −∞ h(τ )ej2πf0 (t−τ ) dτ
R∞ (2.37)
= ej2πf0 t −∞ h(τ )e−j2πf0 τ dτ = H(f0 )ej2πf0 t
where Z ∞
H(f0 ) = h(τ )e−j2πf0 τ dτ
−∞
is the Fourier transform of h evaluated at the frequency f0 . We discuss the Fourier transform
and its properties in more detail shortly.
Complex exponentials are eigenfunctions of LTI systems: Recall that an eigenvector of
a matrix H is any vector x that satisfies Hx = λx. That is, the matrix leaves its eigenvectors
unchanged except for a scale factor λ, which is the eigenvalue associated with that eigenvector.
In an entirely analogous fashion, we see that the complex exponential signal ej2πf0 t is an eigen-
function of the LTI system with impulse response h, with eigenvalue H(f0). Since we have not
constrained h, we conclude that complex exponentials are eigenfunctions of any LTI system. We
shall soon see, when we discuss Fourier transforms, that this eigenfunction property allows us
to characterize LTI systems in the frequency domain, which in turn enables powerful frequency
domain design and analysis tools.

2.3.1 Discrete time convolution

DSP-based implementations of convolutions are inherent discrete time. For two discrete time
sequences {u1 [n]} and {u2 [n]}, their convolution y = u1 ∗ u2 is defined analogous to continuous
time convolution, replacing integration by summation:
X
y[n] = u1 [k]u2 [n − k] (2.38)
k

Matlab implements this using the “conv” function. This can be interpreted as u1 being the input
to a system with impulse response u2 , where a discrete time impulse is simply a one, followed by
all zeros.
Continuous time convolution between u1 (t) and u2 (t) can be approximated using discrete time
convolutions between the corresponding sampled signals. For example, for samples at rate 1/Ts ,
the infinitesimal dt is replaced by the sampling interval Ts as follows:
Z X
y(t) = (u1 ∗ u2 )(t) = u1 (τ )u2 (t − τ )dτ ≈ u1 (kTs )u2 (t − kTs )Ts
k

Evaluating at a sampling time t = nTs , we have

X
y(nTs ) = Ts u1 (kTs )u2 (nTs − kTs )
k

43
Letting x[n] = x(nTs ) denote the discrete time waveform corresponding to the nth sample for
each of the preceding waveforms, we have
X
y(nTs ) = y[n] ≈ Ts u1 [k]u2 [n − k] = Ts (u1 ∗ u2 )[n] (2.39)
k

which shows us how to implement continuous time convolution using discrete time operations.

1
u1
0.9 u2
y
0.8

0.7

0.6

0.5

0.4

0.3

0.2

0.1

0
−2 0 2 4 6 8 10 12

Figure 2.16: Two signals and their continuous time convolution, computed in discrete time using
Code Fragment 2.3.1.

The following Matlab code provides an example of a continuous time convolution approximated
numerically using discrete time convolution, and then plotted against the original continuous
time index t, as shown in Figure 2.16 (cosmetic touches not included in the code below). The
two waveforms convolved are u1 (t) = t2 I[−1,1] (t) and u2 (t) = e−(t+1) I[−1,∞) (the latter is truncated
in our discrete time implementation).

Code Fragment 2.3.1 (Discrete time computation of continuous time convolution)

dt=0.01; %sampling interval T_s

%%FIRST SIGNAL
u1start=-1; u1end = 1; %start and end times for first signal
t1=u1start:dt:u1end; %sampling times for first signal
u1=t1.^2; %discrete time version of first signal
%%SECOND SIGNAL (exponential truncated when it gets small)
u2start=-1; u2end = 10;
t2=u2start:dt:u2end;
u2=exp(-(t2+1));
%%APPROXIMATION OF CONTINUOUS TIME CONVOLUTION
y=dt*conv(u1,u2);
%%PLOT OF SIGNALS AND THEIR CONVOLUTION
ystart=u1start+u2start; %start time for convolution output
time_axis = ystart:dt:ystart+dt*(length(y)-1);

44
%%PLOT u1, u2 and y
plot(t1,u1,’r-.’);
hold on;
plot(t2,u2,’r:’);
plot(time_axis,y);
legend(’u1’,’u2’,’y’,’Location’,’NorthEast’);
hold off;

2.3.2 Multi-rate systems

While continuous time signals can be converted to discrete time by sampling “fast enough,” it is
often required that we operate at multiple sampling rates. For example, in digital communication,
we may send a string of symbols {b[n]} (think of these as taking values +1 or -1 for now) by
modulating them onto shifted versions of a pulse p(t) as follows:
X
u(t) = b[n]p(t − nT ) (2.40)
n

where 1/T is the rate at which symbols are generated (termed the symbol rate). In order to
represent the analog pulse p(t) as discrete time samples, we may sample it at rate 1/Ts , typically
chosen to be an integer multiple of the symbol rate, so that T = mTs , where m is a positive
integer. Typical values employed in transmitter DSP modules might be m = 4 or m = 8. Thus,
the system we are interested is multi-rate: waveforms are sampled at rate 1/Ts = m/T , but
the input is at rate 1/T . Set u[k] = u(kTs ) and p[k] = p(kTs ) as the discrete time signals
corresponding to samples of the transmitted waveform u(t) and the pulse p(t), respectively. We
can write the sampled version of (2.40) as
X X
u[k] = b[n]p(kTs − nT ) = b[n]p[k − nm] (2.41)
n n

The preceding almost has the form of a discrete time convolution, but the key difference is
that the successive symbols {b[n]} are spaced by time T , which corresponds to m > 1 samples
at the sampling rate 1/Ts . Thus, in order to implement this system using convolution at rate
1/Ts , we must space out the input symbols by inserting m − 1 zeros between successive symbols
b[n], thus converting a rate 1/T signal to a rate 1/Ts = m/T signal. This process is termed
upsampling. While the upsampling function is available in certain Matlab toolboxes, we provide
a self-contained code fragment below that illustrates its use for digital modulation, and plots
the waveform obtained for symbol sequence −1, +1, +1, −1. The modulating pulse is a sine
pulse: p(t) = sin(πt/T )I[0,T ] , and our convention is to set T = 1 without loss of generality
(or, equivalently, to replace t by t/T ). We set the oversampling factor M = 16 in order to
obtain smooth plots, even though typical implementations in communication transmitters may
use smaller values.

Code Fragment 2.3.2 (Upsampling for digital modulation)

m=16; %sampling rate as multiple of symbol rate

%discrete time representation of sine pulse
time_p = 0:1/m:1; %sampling times over duration of pulse
p = sin(pi*time_p); %samples of the pulse
%symbols to be modulated
symbols = [-1;1;1;-1];

45
 1

0.8

0.6

0.4

0.2

u(t)
0

−0.2

−0.4

−0.6

−0.8

−1
0 0.5 1 1.5 2 2.5 3 3.5 4
t/T

Figure 2.17: Digitally modulated waveform obtained using Code Fragment 2.3.2.

%UPSAMPLE BY m
nsymbols = length(symbols);%length of original symbol sequence
nsymbols_upsampled = 1+(nsymbols-1)*m;%length of upsampled symbol sequence
symbols_upsampled = zeros(nsymbols_upsampled,1);%
symbols_upsampled(1:m:nsymbols_upsampled)=symbols;%insert symbols with spacing M
%GENERATE MODULATED SIGNAL BY DISCRETE TIME CONVOLUTION
u=conv(symbols_upsampled,p);
%PLOT MODULATED SIGNAL
time_u = 0:1/m:(length(u)-1)/m; %unit of time = symbol time T
plot(time_u,u);
xlabel(’t/T’);

2.4 Fourier Series

Fourier series represent periodic signals in terms of sinusoids or complex exponentials. A signal
u(t) is periodic with period T if u(t + T ) = u(t) for all t. Note that, if u is periodic with period
T , then it is also periodic with period nT , where n is any positive integer. The smallest time
interval for which u(t) is periodic is termed the fundamental period. Let us denote this by T0 , and
define the corresponding fundamental frequency f0 = 1/T0 (measured in Hertz if T0 is measured
in seconds). It is easy to show that if u(t) is periodic with period T , then T must be an integer
multiple of T0 . In the following, we often simply refer to the fundamental period as “period.”
Using mathematical machinery beyond our current scope, it can be shown that any periodic signal
with period T0 (subject to mild technical conditions) can be expressed as a linear combination
of complex exponentials

ψm (t) = ej2πmf0 t = ej2πmt/T0 , m = 0, ±1, ±2, ...

whose frequencies are integer multiples of the fundamental frequency f0 . That is, we can write
∞
X ∞
X
u(t) = un ψn (t) = un ej2πnf0 t (2.42)
n=−∞ n=−∞

46
The coefficients {un } are in general complex-valued, and are called the Fourier series for u(t).
They can be computed as follows:

1
Z
uk = u(t)e−j2πkf0 t dt (2.43)
T0 T0
R
where T0 denotes an integral over any interval of length T0 .
Let us now derive (2.43). For m a nonzero integer, consider an arbitrary interval of length T0 , of
the form [D, D + T0 ], where the offset D is free to take on any real value. Then, for any nonzero
integer m 6= 0, we have
R D+T0 j2πmf t D+T0
ej2πmf0 t 
D
e 0
dt = j2πmf0 
D
(2.44)
ej2πf0 mD −ej(2πmf0 D+2πm)
= j2πmf0
=0
since ej2πm = 1. Thus, when we multiply both sides of (2.42) by e−j2πkf0 t and integrate over a
period, all terms corresponding to n 6= k drop out by virtue of (2.44), and we are left only with
the n = k term:
R D+T0 −j2πkf0 t
R D+T0 P∞ j2πnf0 t
 −j2πkf t
D
u(t)e dt = D n=−∞ u n e e 0
dt
R D+T0 R D+T0
ej2πkf0 t e−j2πkf0 t dt + ej2π(n−k)f0 t dt = uk T0 + 0
P
= uk D n6=k un D

which proves (2.43).

We denote the Fourier series relationship (2.42)-(2.43) as u(t) ↔ {un }. It is useful to keep
in mind the geometric meaning of this relationship. The space of periodic signals with period
T0 = f10 can be thought of in the same way as the finite-dimensional vector spaces we are familiar
with, except that the inner product between two periodic signals is given by
Z
hu, viT0 = u(t)v ∗ (t)dt
T0

The energy over a period for a signal u is given by ||u||2T0 = hu, uiT0 , where ||u||T0 denotes the
norm computed over a period. We have assumed that the Fourier basis {ψn (t)} spans this vector
space, and have computed the Fourier series after showing that the basis is orthogonal:

hψn , ψm iT0 = 0 , n 6= m

and equal energy:

||ψn ||2T0 = hψn , ψn iT0 = T0
The computation of the expression for the Fourier series {uk } can be rewritten in these vector
space terms as follows. A periodic signal u(t) can be expanded in terms of the Fourier basis as
∞
X
u(t) = un ψn (t) (2.45)
n=−∞

Using the orthogonality of the basis functions, we have

X
hu, ψk iT0 = un hψn , ψk iT0 = uk ||ψk ||2
n

47
That is,
hu, ψk iT0 hu, ψk iT0
uk = 2
= (2.46)
||ψk || T0
In general, the Fourier series of an arbitrary periodic signal may have an infinite number of terms.
In practice, one might truncate the Fourier series at a finite number of terms, with the number
of terms required to provide a good approximation to the signal depending on the nature of the
signal.
T0
A max

... ...

A min

Figure 2.18: Square wave with period T0 .

Example 2.4.1 Fourier series of a square wave: Consider the periodic waveform u(t) as
shown in Figure 2.18. For k = 0, we get the DC value u0 = Amax +A
2
min
. For k 6= 0, we have,
using (2.43), that

1
R0 1
R T0
uk = T0 −
T0 Amin e−j2πkt/T0 dt + T0 0
2
Amax e−j2πkt/T0 dt
2

0  T0
Amin e−j2πkt/T0  Amax e−j2πkt/T0  2
= T0 −j2πk/T0  T0
+ T0 −j2πk/T0 
− 2 0

Amin (1−ejπk )+Amax (e−jπk −1)

= −j2πk

For k even, ejπk = e−jπk = 1, which yields uk = 0. That is, there are no even harmonics. For k
odd, ejπk = e−jπk = −1, which yields uk = Amaxjπk
−Amin
. We therefore obtain

0, k even
uk = Amax −Amin
jπk
, k odd

Combining the terms for positive and negative k, we obtain

Amax + Amin X 2(Amax − Amin )

u(t) = + sin 2πkt/T0
2 k odd
πk

Example 2.4.2 Fourier series of an impulse train: Even though the delta function is not
physically realizable, the Fourier series of an impulse train, as shown in Figure 2.19 turns out to
be extremely useful in theoretical development and in computations. Specifically, consider
∞
X
u(t) = δ(t − nT0 )
n=−∞

48
 ... ...
−T0 0 T0

Figure 2.19: An impulse train of period T0 .

By integrating over an interval of length T0 centered around the origin, we obtain

T0 T0
1 1 1
Z Z
2 2
−j2πkf0 t
uk = u(t)e dt = δ(t)e−j2πkf0t dt =
T0 −
T0
2
T0 −
T0
2
T0

using the sifting property of the impulse. That is, the delta function has equal frequency content
at all harmonics. This is yet another manifestation of the physical unrealizability of the impulse:
for well-behaved signals, the Fourier series should decay as the frequency increases.

While we have considered signals which are periodic functions of time, the concept of Fourier
series applies to periodic functions in general, whatever the physical interpretation of the argu-
ment of the function. In particular, as we shall see when we discuss the effect of time domain
sampling in the context of digital communication, the time domain samples of a waveform can
be interpreted as the Fourier series for a particular periodic function of frequency.

2.4.1 Fourier Series Properties and Applications

We now state some Fourier series properties which are helpful both for computation and for
developing intuition. The derivations are omitted, since they follow in a straightforward manner
from (2.42)-(2.43), and are included in any standard text on signals and systems. In the following,
u(t), v(t) denote periodic waveforms of period T0 and Fourier series {uk }, {vk } respectively.
Linearity: For arbitrary complex numbers α, β,
αu(t) + βv(t) ↔ {αuk + βvk }

Time delay corresponds to linear phase in frequency domain:

u(t − d) ↔ {uk e−j2πkf0 d = uk e−j2πkd/T0 }

The Fourier series of a real-valued signal is conjugate symmetric: If u(t) is real-valued,

then uk = u∗−k .
Harmonic structure of real-valued periodic signals: While both the Fourier series coef-
ficients and the complex exponential basis functions are complex-valued, for real-valued u(t),
the linear combination on the right-hand side of (2.42) must be real-valued. In particular, as
we show below, the terms corresponding to uk and u−k (k ≥ 1) combine together into a real-
valued sinusoid which we term the kth harmonic. Specifically, writing uk = Ak ejφk in polar
form, we invoke the conjugate symmetry of the Fourier series for real-valued u(t) to infer that
u−k = u∗k = Ak e−jφk . The Fourier series can therefore be written as
∞
X ∞
X
j2πkf0 t −j2πkf0 t
Ak ejφk ej2πkf0 t + Ak e−jφk e−j2πkf0 t


u(t) = u0 + uk e + u−k e = u0 +
k=1 k=1

49
This yields the following Fourier series in terms of real-valued sinusoids:
∞
X ∞
X
u(t) = u0 + 2Ak cos(2πkf0 t + φk ) = u0 + 2|uk | cos (2πkf0 t + uk ) (2.47)
k=1 k=1

Differentiation amplifies higher frequencies:

d
x(t) = u(t) ↔ xk = j2πkf0 uk (2.48)
dt
Note that differentiation kills the DC term, i,.e, x0 = 0. However, the information at all other
frequencies is preserved. That is, if we know {xk } then we can recover {uk , k 6= 0} as follows:
xk
uk = , k 6= 0 (2.49)
j2πf0 k
This is a useful property, since differentiation often makes Fourier series easier to compute.
T0
A max

... ...

A min

d/dt
A max −A min

... T0/2
...
0 T0
... ...
−(A max −A min )

Figure 2.20: The derivative of a square wave is two interleaved impulse trains.

Example 2.4.1 redone (using differentiation to simplify Fourier series computation):

Differentiating the square wave in Figure 2.18 gives us two interleaved impulse trains, one cor-
responding to the upward edges of the rectangular pulses, and the other to the downward edges
of the rectangular pulses, as shown in Figure 2.20.
!
d X X
x(t) = u(t) = (Amax − Amin ) δ(t − kT0 ) − δ(t − kT0 − T0 /2)
dt k k

Compared to the impulse train in Example 2.4.2, the first impulse train above is offset by 0,
while the second is offset by T0 /2 (and inverted). We can therefore infer their Fourier series
using the time delay property, and add them up by linearity, to obtain
Amax − Amin Amax − Amin −j2πf0 kT0 /2 Amax − Amin
1 − e−jπk , , k 6= 0


xk = − e =
T0 T0 T0

50
Using the differentiation property, we can therefore infer that
xk Amax −Amin


uk = j2πf0 k
= −j2πkf0 T0
1 − e−jπk

which gives us the same result as before. Note that the DC term u0 cannot be obtained using
this approach, since it vanishes upon differentiation. But it is easy to compute, since it is just
the average value of u(t), which can be seen to be u0 = (Amax + Amin )/2 by inspection.
In addition to simplifying computation for waveforms which can be described (or approximated)
as polynomial functions of time (so that enough differentiation ultimately reduces them to im-
pulse trains), the differentiation method explicitly reveals how the harmonic structure (i.e., the
strength and location of the harmonics) of a periodic waveform is related to its transitions in
the time domain. Once we understand the harmonic structure, we can shape it by appropriate
filtering. For example, if we wish to generate a sinusoid of frequency 300 MHz using a digital
circuit capable of generating symmetric square waves of frequency 100 MHz, we can choose a
filter to isolate the third harmonic. However, we cannot generate a sinusoid of frequency 200
MHz (unless we make the square wave suitably asymmetric), since the even harmonics do not
exist for a symmetric square wave (i.e., a square wave whose high and low durations are the
same).
Parseval’s identity (periodic inner product/power can be computed in either time
or frequency domain): Using the orthogonality of complex exponentials over a period, it can
be shown that
Z X∞
∗
hu, viT0 = u(t)v (t)dt = T0 uk vk∗ (2.50)
T0 k=−∞

Setting v = u, and dividing both sides by T0 , the preceding specializes to an expression for signal
power (which can be computed for a periodic signal by averaging over a period):
∞
1
Z X
2
|u(t)| dt = |uk |2 (2.51)
T0 T0 k=−∞

2.5 Fourier Transform

We define the Fourier transform U(f ) for a aperiodic, finite energy waveform u(t) as
Z ∞
U(f ) = u(t)e−j2πf t dt , −∞<f <∞ Fourier Transform (2.52)
−∞

The inverse Fourier transform is given by

Z ∞
u(t) = U(f )ej2πf t df , − ∞ < t < ∞ Inverse Fourier Transform (2.53)
−∞

The inverse Fourier transform tells us that any finite energy signal can be written as a linear com-
bination of a continuum of complex exponentials, with the coefficients of the linear combination
given by the Fourier transform U(f ).
Notation: We call a signal and its Fourier transform a Fourier transform pair, and denote them
as u(t) ↔ U(f ). We also denote the Fourier transform operation by F , so that U(f ) = F (u(t)).

51
Example 2.5.1 Rectangular pulse and sinc function form a Fourier transform pair:
Consider the rectangular pulse u(t) = I[−T /2,T /2] (t) of duration T . Its Fourier transform is given
by
R∞ R T /2
U(f ) = −∞ u(t)e−j2πf t dt = −T /2 e−j2πf t dt

e−j2πf t T /2 e−jπf T −ejπf T
= −j2πf −T /2
= −j2πf

sin(πf T )
= πf
= T sinc(f T )
We denote this as
I[−T /2,T /2] (t) ↔ T sinc(f T )

Duality: Given the similarity of the form of the Fourier transform (2.52) and inverse Fourier
transform (2.53), we can see that the roles of time and frequency can be switched simply by
negating one of the arguments. In particular, suppose that u(t) ↔ U(f ). Define the time
domain signal s(t) = U(t), replacing f by t. Then the Fourier transform of s(t) is given by
S(f ) = u(−f ), replacing t by −f . Since negating the argument corresponds to reflection around
the origin, we can simply switch time and frequency for signals which are symmetric around the
origin. Applying duality to the Example 2.5.1, we infer that a signal that is ideally bandlimited
in frequency corresponds to a sinc function in time:
I[−W/2,W/2] (f ) ↔ W sinc(W t)

Application to infinite energy signals: In engineering applications, we routinely apply the

Fourier and inverse Fourier transform to infinite energy signals, even though its derivation as the
limit of a Fourier series is based on the assumption that the signal has finite energy. While infinite
energy signals are not physically realizable, they are useful approximations of finite energy signals,
often simplifying mathematical manipulations. For example, instead of considering a sinusoid
over a large time interval, we can consider a sinusoid of infinite duration. As we shall see, this
leads to an impulsive function in the frequency domain. As another example, delta functions
in the time domain are useful in modeling the impulse response of wireless multipath channels.
Basically, once we are willing to work with impulses, we can use the Fourier transform on a very
broad class of signals.

Example 2.5.2 The delta function and the constant function form a Fourier trans-
form pair: For u(t) = δ(t), we have
Z ∞
U(f ) = δ(t)e−j2πf t dt = 1
−∞

for all f . That is,

δ(t) ↔ I(−∞,∞) (f )

Now that we have seen both the Fourier series and the Fourier transform, it is worth commenting
on the following frequently asked questions.
What do negative frequencies mean? Why do we need them? Consider a real-valued
sinusoid A cos(2πf0 t + θ), where f0 > 0. If we now replace f0 by −f0 , we obtain A cos(−2πf0 t +
θ) = A cos(2πf0 t−θ), using the fact that cosine is an even function. Thus, we do not need negative
frequencies when working with real-valued sinusoids. However, unlike complex exponentials, real-
valued sinusoids are not eigenfunctions of LTI systems: we can pass a cosine through an LTI
system and get a sine, for example. Thus, once we decide to work with a basis formed by complex

52
exponentials, we do need both positive and negative frequencies in order to describe all signals
of interest. For example, a real-valued sinusoid can be written in terms of complex exponentials
as
A j(2πf0 t+θ)
A A
A cos(2πf0 t + θ) = e + e−j(2πf0 t+θ) = ejθ ej2πf0 t + e−jθ e−j2πf0 t
2 2 2
so that we need complex exponentials at both +f0 and −f0 to describe a real-valued sinusoid
at frequency f0 . Of course, the coefficients multiplying these two complex exponentials are not
arbitrary: they are complex conjugates of each other. More generally, as we have already seen,
such conjugate symmetry holds for both Fourier series and Fourier transforms of real-valued
signals. We can therefore state the following:
(a) We do need both positive and negative frequencies to form a complete basis using complex
exponentials;
(b) For real-valued (i.e., physically realizable) signals, the expansion in terms of a complex
exponential basis, whether it is the Fourier series or the Fourier transform, exhibits conjugate
symmetry. Hence, we only need to know the Fourier series or Fourier transform of a real-valued
signal for positive frequencies.

2.5.1 Fourier Transform Properties

The Fourier transform can be obtained by taking the limit of the Fourier series as the period
gets large, with T0 → ∞ and f0 → 0 (think of an aperiodic signal as periodic with infinite
period). We do not provide details, but sketch the process of taking this limit: T0 uk tends to
U(f ), where f = kf0 , and the Fourier series sum in (2.42) become the inverse Fourier transform
integral in (2.53), with f0 becoming df . Not surprisingly, therefore, the Fourier transform exhibits
properties entirely analogous to those for Fourier series. However, the Fourier transform applies
to a broader class of signals, and we can take advantage of time-frequency duality more easily,
because both time and frequency are now continuous-valued variables.
We now state some key properties. In the following, u(t), v(t) denote signals with Fourier
transforms U(f ), V (f ), respectively.
Linearity: For arbitrary complex numbers α, β,
αu(t) + βv(t) ↔ αU(f ) + βV (f )

Time delay corresponds to linear phase in frequency domain:

u(t − t0 ) ↔ U(f )e−j2πf t0

Frequency shift corresponds to modulation by complex exponential:

U(f − f0 ) ↔ u(t)ej2πf0 t

The Fourier transform of a real-valued signal is conjugate symmetric: If u(t) is real-

valued, then U(f ) = U ∗ (−f ).
Differentiation in the time domain amplifies higher frequencies:
d
x(t) = u(t) ↔ X(f ) = j2πf U(f )
dt
As for Fourier series, differentiation kills the DC term, i,.e, X(0) = 0. However, the information
at all other frequencies is preserved. Thus, if we know X(f ) then we can recover U(f ) for f 6= 0
as follows:
X(f )
U(f ) = , f 6= 0 (2.54)
j2πf

53
This specifies the Fourier transform almost everywhere (except at DC: f = 0). If U(f ) is finite
everywhere, then we do not need to worry about its value at a particular point, and can leave
U(0) unspecified, or define it as the limit of (2.54) as f → 0 (and if this limit does not exist,
we can set U(0) to be the left limit, or the right limit, or any number in between). In short, we
can simply adopt (2.54) as the expression for U(f ) for all f , when U(0) is finite. However, the
DC term does matter when u(t) has a nonzero average value, in which case we get an impulse
at DC. The average value of u(t) is given by
T
1
Z
2
ū = lim u(t)dt
T →∞ T − T2

and has Fourier transform given by ū(t) ≡ ū ↔ ūδ(f ). Thus, we can write the overall Fourier
transform as
X(f )
U(f ) = + ūδ(f ) (2.55)
j2πf
We illustrate this via the following example.

Example 2.5.3 (Fourier transform of a step function) Let us use differentiation to com-
pute the Fourier transform of the unit step function

0, t < 0
u(t) =
1, t ≥ 0

Its DC value is given by

ū = 1/2
and its derivative is the delta function (see Figure 2.21):
d
x(t) = u(t) = δ(t) ↔ X(f ) ≡ 1
dt
Applying (2.55), we obtain that the Fourier transform of the unit step function is given by

du/dt
u(t)
1

0 t 0 t

Figure 2.21: The unit step function and its derivative, the delta function.

1 1
U(f ) = + δ(f )
j2πf 2

Parseval’s identity (inner product/energy can be computed in either time or fre-

quency domain): Z ∞ Z ∞
∗
hu, vi = u(t)v (t)dt = U(f )V ∗ (f )df
−∞ −∞

54
Setting v = u, we get an expression for the energy of a signal:
Z ∞ Z ∞
2 2
||u|| = |u(t)| dt = |U(f )|2 df
−∞ −∞

Next, we discuss the significance of the Fourier transform in understanding the effect of LTI
systems.
Transfer function for an LTI system: The transfer function H(f ) of an LTI system is
defined to be the Fourier transform of its impulse response h(t). That is, H(f ) = F (h(t)). We
now discuss its significance.
From (2.37), we know that, when the input to an LTI system is the complex exponential ej2πf0 t ,
the output is given by H(f0 )ej2πf0 t . From the inverse Fourier transform (2.53), we know that
any input u(t) can be expressed as a linear combination of complex exponentials. Thus, the
corresponding response, which we know is given by y(t) = (u ∗ h)(t) must be a linear combination
of the responses to these complex exponentials. Thus, we have
Z ∞
y(t) = U(f )H(f )ej2πf t df
−∞

We recognize that the preceding function is in the form of an inverse Fourier transform, and
read off Y (f ) = U(f )H(f ). That is, the Fourier transform of the output is simply the product
of the Fourier transform of the input and the system transfer function. This is because complex
exponentials at different frequencies propagate through an LTI system without mixing with each
other, with a complex exponential at frequency f passing through with a scaling of H(f ).
Of course, we have also derived an expression for y(t) in terms of a convolution of the input
signal with the system impulse response: y(t) = (u ∗ h)(t). We can now infer the following key
property.
Convolution in the time domain corresponds to multiplication in the frequency do-
main
y(t) = (u ∗ h)(t) ↔ Y (f ) = U(f )H(f ) (2.56)

We can also infer the following dual property, either by using duality or by directly deriving it
from first principles.
Multiplication in the time domain corresponds to convolution in the frequency do-
main
y(t) = u(t)v(t) ↔ Y (f ) = (U ∗ V )(f ) (2.57)

LTI system response to real-valued sinusoidal signals: For a sinusoidal input u(t) =
cos(2πf0 t + θ), the response of an LTI system h is given by

y(t) = (u ∗ h)(t) = |H(f0 )| cos (2πf0 t + θ + H(f0 ))

This can be inferred from what we know about the response for complex exponentials, thanks
to Euler’s formula. Specifically, we have
1 j(2πf0 t+θ)
1 1
u(t) = e + e−j(2πf0 t+θ) = ejθ ej2πf0 t + e−jθ e−j2πf0 t
2 2 2
When u goes through an LTI system with transfer function H(f ), the output is given by
1 1
y(t) = ejθ H(f0 )ej2πf0 t + e−jθ H(−f0 )e−j2πf0 t
2 2

55
If the system is physically realizable, the impulse response h(t) is real-valued, and the transfer
function is conjugate symmetric. Thus, if H(f0 ) = Gejφ (G ≥ 0), then H(−f0 ) = H ∗ (f0 ) =
Ge−jφ . Substituting, we obtain
G j(2πf0 t+θ+φ) G −j(2πf0 t+θ+φ)
y(t) = e + e = G cos(2πf0 t + θ + φ)
2 2
This yields the well-known result that the sinusoid gets scaled by the magnitude of the transfer
function G = |H(f0 )|, and gets phase shifted by the phase of the transfer function φ = H(f0 ).

Example 2.5.4 (Delay spread, coherence bandwidth, and fading for a multipath
channel) The transfer function of a multipath channel as in (2.36) is given by

H(f ) = α1 e−j2πf τ1 + ... + αm e−j2πf τm (2.58)

Thus, the channel transfer function is a linear combination of complex exponentials in the fre-
quency domain. As with any sinusoids, these can interfere constructively or destructively, leading
to significant fluctuations in H(f ) as f varies. For wireless channels, this phenomenon is called
frequency-selective fading. Let us examine the structure of the fading a little further. Suppose,
without loss of generality, that the delays are in increasing order (i.e., τ1 < τ2 < ... < τm ). We
can then rewrite the transfer function as
m
X
H(f ) = e−j2πf τ1 αk e−j2πf (τk −τ1 )
k=1

The first term e−j2πf τ1 corresponds simply to a pure delay τ1 (seen by all frequencies), and can
be dropped (taking τ1 as our time origin, without loss of generality), so that the transfer function
can be rewritten as m
X
H(f ) = α1 + αk e−j2πf (τk −τ1 ) (2.59)
k=2

The period of the kth sinusoid above (k ≥ 2) is 1/(τk − τ1 ), so that, the smallest period, and
hence the fastest fluctuations as a function of f , occurs because of the largest delay difference
τd = τm − τ1 , which we call the channel delay spread. Thus, for a frequency interval which is
significantly smaller than 1/τd , the variation of |H(f )| over the interval is small. We define the
channel coherence bandwidth as the inverse of the delay spread, i.e., as Bc = 1/(τm − τ1 ) =
1/τd (this definition is not unique, but in general, the coherence bandwidth is defined to be
inversely proportional to some appropriately defined measure of the channel delay spread). As
we have noted, H(f ) can be well modeled as constant over intervals significantly smaller than
the coherence bandwidth.
Let us apply this to the example in Figure 2.14, where we have a multipath channel with impulse
response h(t) = δ(t − 1) − 0.5δ(t − 1.5) + 0.5δ(t − 3.5). Dropping the first delay as before, we
have
H(f ) = 1 − 0.5e−jπf + 0.5e−j5πf
For concreteness, suppose that time is measured in microseconds (typical numbers for an outdoor
wireless cellular link), so that frequency is measured in MHz. The delay spread is 2.5µs, hence
the coherence bandwidth is 400KHz. We therefore ballpark the size of the frequency interval
over which H(f ) can be approximated as constant to about 40KHz (i.e., of size 10% of the
coherence bandwidth). Note that this is a very fuzzy estimate: if the larger delays occur with
smaller relative amplitudes, as is typical, then they have a smaller effect on H(f ), and we could
potentially approximate H(f ) as constant over a larger fraction of the coherence bandwidth.
Figure 2.22 depicts the fluctuations in H(f ) first on a linear scale, and then on a log scale. A

56
plot of the transfer function magnitude is shown in Figure 2.22(a). This is the amplitude gain on
a linear scale, and shows significant variations as a function of f (while we do not show it here,
zooming in to 40 KHz bands shows relatively small fluctuations). The amount of fluctuation
becomes even more apparent on a log scale. Interpreting the gain at the smallest delay (α1 = 1
in our case) as that of a nominal channel, the fading gain is defined as the power gain relative
to this nominal, and is given by 20 log10 (|H(f )|/|α1|) in decibels (dB). This is shown in Figure
2.22(b). Note that the fading gain can dip below -18 dB in our example, which we term a fade
of depth 18 dB. If we are using a “narrowband” signal which has a bandwidth small compared to
the coherence bandwidth, and happen to get hit by such a fade, then we can expect much poorer
performance than nominal. To combat this, one must use diversity. For example, a “wideband”
signal whose bandwidth is larger than the coherence bandwidth provides frequency diversity,
while, if we are constrained to use narrowband signals, we may need to introduce other forms of
diversity (e.g., antenna diversity as in Software Lab 2.2).
2 10

1.8

5
1.6
Magnitude of transfer function

1.4
0

Fading gain (dB)

1.2

1 −5

0.8

−10
0.6

0.4
−15

0.2

0 −20
−1 −0.8 −0.6 −0.4 −0.2 0 0.2 0.4 0.6 0.8 1 −1 −0.8 −0.6 −0.4 −0.2 0 0.2 0.4 0.6 0.8 1
Frequency (MHz) Frequency (MHz)

(a) Transfer Function Magnitude (linear scale) (b) Frequency-selective fading (dB)

Figure 2.22: Multipath propagation causes severe frequency-selective fading.

2.5.2 Numerical computation using DFT

In many practical settings, we do not have nice analytical expressions for the Fourier or in-
verse Fourier transforms, and must resort to numerical computation, typically using the discrete
Fourier transform (DFT). The DFT of a discrete time sequence {u[n], n = 0, ..., N − 1} of length
N is given by
N
X −1
U[m] = u[n]e−j2πmn/N , m = 0, 1, ..., N − 1. (2.60)
n=0

Matlab is good at doing DFTs. When N is a power of 2, the DFT can be computed very
efficiently, and this procedure is called a Fast Fourier Transform (FFT). Comparing (2.60) with
the Fourier transform expression
Z ∞
U(f ) = u(t)e−j2πf t dt (2.61)
−∞

we can view the sum in the DFT (2.60) as an approximation for the integral in (2.61) under the
right set of conditions. Let us first assume that u(t) = 0 for t < 0: any waveform which can be

57
truncated so that most of its energy falls in a finite interval can be shifted so that this is true.
Next, suppose that we sample the waveform with spacing ts to get
u[n] = u(nts )
Now, suppose we want to compute the Fourier transform U(f ) for f = mfs , where fs is the
desired frequency resolution. We can approximate the integral for the Fourier transform by a
sum, using ts -spaced time samples as follows:
Z ∞ X
U(mfs ) = u(t)e−j2πmfs t dt ≈ u(nts )e−j2πmfs nts ts
−∞ n

(dt in the integral is replaced by the sample spacing ts .) Since u[n] = u(nts ), the approximation
can be computed using the DFT formula (2.60) as follows:
U(mfs ) ≈ ts U[m] (2.62)
1
as long as fs ts =N
. That is, using a DFT of length N, we can get a frequency granularity of
fs = N1ts . This implies that if we choose the time samples close together (in order to represent
u(t) accurately), then we must also use a large N to get a desired frequency granularity. Often
this means that we must pad the time domain samples with zeros.
Another important observation is that, while the DFT in (2.60) ranges from m = 0, ..., N − 1, it
actually computes the Fourier transform for both positive and negative frequencies. Noting that
ej2πmn/N = ej2π(−N +m)n/N , we realize that the DFT values for m = N/2, ..., N − 1 correspond
to the Fourier transform evaluated at frequencies (m − N)fs = −N/2fs , ..., −fs . The DFT
values for m = 0, ..., N/2 − 1 correspond to the Fourier transform evaluated at frequencies
0, fs , ..., (N/2 − 1)fs . Thus, we should swap the left and right halves of the DFT output in order
to represent positive and negative frequencies, with DC falling in the middle. Matlab actually
has a function, fftshift, that does this.
Note that the DFT (2.60) is periodic with period N, so that the Fourier transform approximation
(2.62) is periodic with period Nfs = t1s . We typically limit the range of frequencies over which
we use the DFT to compute the Fourier transform to the fundamental period (− 2t1s , 2t1s ). This is
consistent with the sampling theorem, which says that the sampling rate 1/ts must be at least as
large as the size of the frequency band of interest. (The sampling theorem is reviewed in Chapter
4, when we discuss digital modulation.)

Example 2.5.5 (DFT-based Fourier transform computation) Suppose that we want to

compute the Fourier transform of the sine pulse u(t) = sin πtI[0,1] (t). The Fourier transform for
this can be computed analytically (see Problem 2.9) to be
2 cos πf −jπf
U(f ) = e (2.63)
π(1 − 4f 2 )
Note that U(f ) has a 0/0 form at f = 1/2, but using L’Hospital’s rule, we can show that
U(1/2) 6= 0. Thus, the first zeros of U(f ) are at f = ±3/2. This is a timelimited pulse and
hence cannot be bandlimited, but U(f ) decays as 1/f 2 for f large, so we can capture most of
the energy of the pulse within a suitably chosen finite frequency interval. Let us use the DFT to
compute U(f ) over f ∈ (−8, 8). This means that we set 1/(2ts ) = 8, or ts = 1/16, which yields
about 16 samples over the interval [0, 1] over which the signal u(t) has support. Suppose now
that we want the frequency granularity to be at least fs = 1/160. Then we must use a DFT
with N ≥ ts1fs = 2560 = Nmin . In order to efficiently compute the DFT using the FFT, we
choose N = 4096, the next power of 2 at least as large as Nmin . Code fragment 2.5.1 performs
and plots this DFT. The resulting plot (with cosmetic touches not included in the code below)
is displayed in Figure 2.23. It is useful to compare this with a plot obtained from the analytical
formula (2.63), and we leave that as an exercise.

58
 0.7

0.6

0.5

Magnitude Spectrum
0.4

0.3

0.2

0.1

0
−8 −6 −4 −2 0 2 4 6 8
Frequency

Figure 2.23: Plot of magnitude spectrum of sine pulse in Example 2.5.5 obtained numerically
using the DFT.

Code Fragment 2.5.1 Numerical computation of Fourier transform using FFT

ts=1/16; %sampling interval

time_interval = 0:ts:1; %sampling time instants
%%time domain signal evaluated at sampling instants
signal_timedomain = sin(pi*time_interval); %sinusoidal pulse in our example
fs_desired = 1/160; %desired frequency granularity
Nmin = ceil(1/(fs_desired*ts)); %minimum length DFT for desired frequency granularity
%for efficient computation, choose FFT size to be power of 2
Nfft = 2^(nextpow2(Nmin)) %FFT size = the next power of 2 at least as big as Nmin
%Alternatively, one could also use DFT size equal to the minimum length
%Nfft=Nmin;
%note: fft function in Matlab is just the DFT when Nfft is not a power of 2
%freq domain signal computed using DFT
%fft function of size Nfft automatically zeropads as needed
signal_freqdomain = ts*fft(signal_timedomain,Nfft);
%fftshift function shifts DC to center of spectrum
signal_freqdomain_centered = fftshift(signal_freqdomain);
fs=1/(Nfft*ts); %actual frequency resolution attained
%set of frequencies for which Fourier transform has been computed using DFT
freqs = ((1:Nfft)-1-Nfft/2)*fs;
%plot the magnitude spectrum
plot(freqs,abs(signal_freqdomain_centered));
xlabel(’Frequency’);
ylabel(’Magnitude Spectrum’);

2.6 Energy Spectral Density and Bandwidth

Communication channels have frequency-dependent characteristics, hence it is useful to appro-
priately shape the frequency domain characteristics of the signals sent over them. Furthermore,
for wireless communication systems, frequency spectrum is a particularly precious commodity,
since wireless is a broadcast medium to be shared by multiple signals. It is therefore important

59
to quantify the frequency occupancy of communication signals. We provide a first exposure to
these concepts here via the notion of energy spectral density for finite energy signals. These
ideas are extended to finite power signals, for which we can define the analogous concept of
power spectral density, in Chapter 4, “just in time” for our discussion of the spectral occupancy
of digitally modulated signals. Once we know the energy or power spectral density of a signal,
we shall see that there are a number of possible definitions of bandwidth, which is a measure of
the size of the frequency interval occupied by the signal.

H(f)
∆f
1
Energy E u( f*) ∆ f
u(t)
Meter

f*

Figure 2.24: Operational definition of energy spectral density.

Energy Spectral Density: The energy spectral density Eu (f ) of a signal u(t) can be defined
operationally as shown in Figure 2.24. Pass the signal u(t) through an ideal narrowband filter
with transfer function as follows:
1, f ∗ − ∆f < f < f ∗ + ∆f

Hf ∗ (f ) = 2 2
0, else

The energy spectral density Eu (f ∗ ) is defined to be the energy at the output of the filter, divided
by the width ∆f (in the limit as ∆f → 0). That is, the energy at the output of the filter is
approximately Eu (f ∗ )∆f . But the Fourier transform of the filter output is

U(f ), f ∗ − ∆f < f < f ∗ + ∆f


Y (f ) = U(f )H(f ) = 2 2
0, else
By Parseval’s identity, the energy at the output of the filter is
Z ∞ Z f ∗ + ∆f
2
2
|Y (f )| df = |U(f )|2 df ≈ |U(f ∗ )|2 ∆f
−∞ f ∗ − ∆f
2

assuming that U(f ) varies smoothly and ∆f is small enough. We can now infer that the energy
spectral density is simply the magnitude squared of the Fourier transform:
Eu (f ) = |U(f )|2 (2.64)
The integral of the energy spectral density equals the signal energy, which is consistent with
Parseval’s identity.
The inverse Fourier transform of the energy spectral density has a nice intuitive interpretation.
Noting that |U(f )|2 = U(f )U ∗ (f ) and U ∗ (f ) ↔ u∗ (−t), let us define uM F (t) = u∗ (−t) as (the
impulse response of) the matched filter for u(t), where the reasons for this term will be clarified
later. Then
|U(f )|2 = U(f ∗
R
)U (f ) ↔ (u ∗ u M F )(τ ) = u(t)uM F (τ − t)dt
(2.65)
= u(t)u∗ (t − τ )dt
R

where t is a dummy variable for the integration, and the convolution is evaluated at the time
variable τ , which denotes the delay between the two versions of u being correlated: the extreme

60
right-hand side is simply the correlation of u with itself (after complex conjugation), evaluated
at different delays τ . We call this the autocorrelation function of the signal u. We have therefore
shown the following.
For a finite energy signal, the energy spectral density and the autocorrelation function form a
Fourier transform pair.
Bandwidth: The bandwidth of a signal u(t) is loosely defined to be the size of the band
of frequencies occupied by U(f ). The definition is “loose” because the concept of occupancy
can vary, depending on the application, since signals are seldom strictly bandlimited. One
possibility is to consider the band over which |U(f )|2 is within some fraction of its peak value
(setting the fraction equal to 12 corresponds to the 3 dB bandwidth). Alternatively, we might
be interested in energy containment bandwidth, which is the size of the smallest band which
contains a specified fraction of the signal energy (for a finite power signal, we define analogously
the power containment bandwidth).
Only positive frequencies count when computing bandwidth for physical (real-valued)
signals: For physically realizable (i.e., real-valued) signals, bandwidth is defined as its occupancy
of positive frequencies, because conjugate symmetry implies that the information at negative fre-
quencies is redundant.
While physically realizable time domain signals are real-valued, we shall soon introduce complex-
valued signals that have useful physical interpretation, in the sense that they have a well-defined
mapping to physically realizable signals. Conjugate symmetry in the frequency domain does not
hold for complex-valued time domain signals, with different information contained in positive
and negative frequencies in general. Thus, the bandwidth for a complex-valued signal is defined
as the size of the frequency band it occupies over both positive and negative frequencies. The
justification for this convention becomes apparent later in this chapter.

Example 2.6.1 Some bandwidth computations

(a) Consider u(t) = sinc(2t), where the unit of time is microseconds. Then the unit of frequency
is MHz, and U(f ) = 12 I[−1,1] (f ) is strictly bandlimited with 2 MHz.
(b) Now, consider the timelimited waveform u(t) = I[2,4] (t), where the unit of time is microsec-
onds. Then U(f ) = 2sinc(2f )e−j6πf , which is not bandlimited. The 99% energy containment
bandwidth W is defined by the equation
Z W Z ∞ Z ∞ Z 4
2 2 2
|U(f )| df = 0.99 |U(f )| df = 0.99 |u(t)| dt = 0.99 12 dt = 1.98
−W −∞ −∞ 2

where we use Parseval’s identity to simplify computation for timelimited waveforms. Using the
fact that |U(f )| is even, we obtain that
Z W Z W
2
1.98 = 2 |U(f )| df = 2 4sinc2 (2f )df
0 0

We can now solve numerically to obtain W ≈ 5.1 MHz.

2.7 Baseband and Passband Signals

Baseband: A signal u(t) is said to be baseband if the signal energy is concentrated in a band
around DC, and
U(f ) ≈ 0, |f | > W (2.66)
for some W > 0. Similarly, a channel modeled as a linear time invariant system is said to be
baseband if its transfer function H(f ) has support concentrated around DC, and satisfies (2.66).

61
 Re(U(f))

f
−W 0 W

Im(U(f))

f
−W W
−1

Figure 2.25: Example of the spectrum U(f ) for a real-valued baseband signal. The bandwidth
of the signal is W .

Re(Up (f))
W

f
−f c fc

Im(Up (f))

−f c
f
fc

Figure 2.26: Example of the spectrum U(f ) for a real-valued passband signal. The bandwidth
of the signal is W . The figure shows an arbitrarily chosen frequency fc within the band in which
U(f ) is nonzero. Typically, fc is much larger than the signal bandwidth W .

62
A signal u(t) is said to be passband if its energy is concentrated in a band away from DC, with
U(f ) ≈ 0, |f ± fc | > W (2.67)
where fc > W > 0. A channel modeled as a linear time invariant system is said to be passband
if its transfer function H(f ) satisfies (2.67).
Examples of baseband and passband signals are shown in Figures 2.25 and 2.26, respectively.
Physically realizable signals must be real-valued in the time domain, which means that their
Fourier transforms, which can be complex-valued, must be conjugate symmetric: U(−f ) =
U ∗ (f ). As discussed earlier, the bandwidth B for a real-valued signal u(t) is the size of the
frequency interval (counting only positive frequencies) occupied by U(f ).
Information sources typically emit baseband signals. For example, an analog audio signal has
significant frequency content ranging from DC to around 20 KHz. A digital signal in which zeros
and ones are represented by pulses is also a baseband signal, with the frequency content governed
by the shape of the pulse (as we shall see in more detail in Chapter 4). Even when the pulse is
timelimited, and hence not strictly bandlimited, most of the energy is concentrated in a band
around DC.
Wired channels (e.g., telephone lines, USB connectors) are typically modeled as baseband: the
attenuation over the wire increases with frequency, so that it makes sense to design the transmit-
ted signal to utilize a frequency band around DC. An example of passband communication over
a wire is Digital Subscriber Line (DSL), where high speed data transmission using frequencies
above 25 KHz co-exists with voice transmission in the band from 0-4 KHz. The design and use
of passband signals for communication is particularly important for wireless communication, in
which the transmitted signals must fit within frequency bands dictated by regulatory agencies,
such as the Federal Communication Commission (FCC) in the United States. For example, an
amplitude modulation (AM) radio signal typically occupies a frequency interval of length 10 KHz
somewhere in the 540-1600 KHz band allocated for AM radio. Thus, the baseband audio mes-
sage signal must be transformed into a passband signal before it can be sent over the passband
channel spanning the desired band. As another example, a transmitted signal in a WiFi network
may be designed to fit within a 20 MHz frequency interval in the 2.4 GHz unlicensed band, so
that digital messages to be sent over WiFi must be encoded onto passband signals occupying the
designated spectral band.

2.8 The Structure of a Passband Signal

In order to employ a passband channel for communication, we need to understand how to design
a passband transmitted signal to carry information, and how to recover this information from a
passband received signal. We also need to understand how the transmitted signal is affected by
a passband channel.

2.8.1 Time Domain Relationships

Let us start by considering a real-valued baseband message signal m(t) of bandwidth W , to be
sent over a passband channel centered around fc . As illustrated in Figure 2.27, we can translate
the message to passband simply by multiplying it by a sinusoid at fc :
1
up (t) = m(t) cos 2πfc t ↔ Up (f ) =(M(f − fc ) + M(f + fc ))
2
We use the term carrier frequency for fc , and the term carrier for a sinusoid at the carrier
frequency, since the modulated sinusoid is “carrying” the message information over a passband

63
 |M(f))| |U(f)|,|V(f)|
Modulation

f
−W 0 W −fc f c −W fc f

Baseband Passband

Figure 2.27: A baseband message of bandwidth W is translated to passband by multiplying by

a sinusoid at frequency fc , as long as fc > W .

channel. Instead of a cosine, we could also use a sine:

1
vp (t) = m(t) sin 2πfc t ↔ Vp (f ) = (M(f − fc ) − M(f + fc ))
2j
Note that |Up (f )| and |Vp (f )| have frequency content in a band around fc , and are passband
signals (i.e., living in a band not containing DC) as long as fc > W .
I and Q components: If we use both the cosine and sine carriers, we can construct a passband
signal of the form
up (t) = uc (t) cos 2πfc t − us (t) sin 2πfc t (2.68)
where uc and us are real baseband signals of bandwidth at most W , with fc > W . The signal
uc (t) is called the in-phase (or I) component, and us (t) is called the quadrature (or Q) component.
The negative sign for the Q term is a standard convention. Since the sinusoidal terms are entirely
predictable once we specify fc , all information in the passband signal up must be contained in
the I and Q components. Modulation for a passband channel therefore corresponds to choosing
a method of encoding information into the I and Q components of the transmitted signal, while
demodulation corresponds to extracting this information from the received passband signal. In
order to accomplish modulation and demodulation, we must be able to upconvert from baseband
to passband, and downconvert from passband to baseband, as follows.

u c (t) Lowpass u c (t)

Filter

cos 2 π fc t 2cos 2 π fc t
u p (t) u p (t)
−sin 2π fc t −2sin 2π fc t

Lowpass u s (t)
u s (t)
Filter

Upconversion Downconversion
(baseband to passband) (passband to baseband)

Figure 2.28: Upconversion from baseband to passband, and downconversion from passband to
baseband.

Upconversion and downconversion: Equation (2.68) immediately tells us how to upconvert

from baseband to passband. To downconvert from passband to baseband, consider

2up (t) cos(2πfc t) = 2uc (t) cos2 2πfc t − 2us (t) sin 2πfc t cos 2πfc t
= uc (t) + uc (t) cos 4πfc t − us (t) sin 4πfc t

64
The first term on the extreme right-hand side is the I component uc (t), a baseband signal. The
second and third terms are passband signals at 2fc , which we can get rid of by lowpass filtering.
Similarly, we can obtain the Q component us (t) by lowpass filtering −2up (t) sin 2πfc t. Block
diagrams for upconversion and downconversion are depicted in Figure 2.28. Implementation of
these operations could, in practice, be done in multiple stages, and requires careful analog circuit
design.
We now dig deeper into the structure of a passband signal. First, can we choose the I and Q
components freely, independent of each other? The answer is yes: the I and Q components
provide two parallel, orthogonal “channels” for encoding information, as we show next.
Orthogonality of I and Q channels: The passband waveform ap (t) = uc (t) cos 2πfc t corre-
sponding to the I component, and the passband waveform bp (t) = us (t) sin 2πfc t corresponding
to the Q component, are orthogonal. That is,

hap , bp i = 0 (2.69)

Let
1
x(t) = ap (t)bp (t) = uc (t)us (t) cos 2πfc t sin 2πfc t = uc (t)us (t) sin 4πfc t
2
We prove the desired result by showing that x(t) is a passband signal at 2fc , so that its DC
component is zero. That is, Z ∞
x(t)dt = X(0) = 0
−∞

which is the desired result. To show this, note that

1 1
p(t) = uc (t)us (t) ↔ (Uc ∗ Us )(f )
2 2
is a baseband signal: if Uc (f ) is baseband with bandwidth W1 and Us (f ) is baseband with
bandwidth W2 , then their convolution has bandwidth at most W1 + W2 . In order for ap to be
passband, we must have fc > W1 , and in order for bp to be passband, we must have fc > W2 .
Thus, 2fc > W1 + W2 , which means that x(t) = p(t) sin 4πfc t is passband around 2fc , and is
therefore zero at DC. This completes the derivation.

Example 2.8.1 (Passband signal): The signal

up (t) = I[0,1] (t) cos 300πt − (1 − |t|)I[−1,1] (t) sin 300πt

is a passband signal with I component uc (t) = I[0,1] (t) and Q component us (t) = (1 − |t|)I[−1,1] (t).
This example illustrates that we do not require strict bandwidth limitations in our definitions
of passband and baseband: the I and Q components are timelimited, and hence cannot be
bandlimited. However, they are termed baseband signals because most of their energy lies in
baseband. Similarly, up (t) is termed a passband signal, since most of its frequency content lies
in a small band around 150 Hz.

Envelope and phase: Since a passband signal up is equivalent to a pair of real-valued baseband
waveforms (uc , us ), passband modulation is often called two-dimensional modulation. The repre-
sentation (2.68) in terms of I and Q components corresponds to thinking of this two-dimensional
waveform in rectangular coordinates (the “cosine axis” and the “sine axis”). We can also rep-
resent the passband waveform using polar coordinates. Consider the rectangular-polar transfor-
mation
p us (t)
e(t) = u2c (t) + u2s (t) , θ(t) = tan−1
uc (t)

65
where e(t) ≥ 0 is termed the envelope and θ(t) is the phase. This corresponds to uc (t) =
e(t) cos θ(t) and us (t) = e(t) sin θ(t). Substituting in (2.68), we obtain

up (t) = e(t) cos θ(t) cos 2πfc t − e(t) sin θ(t) sin 2πfc t = e(t) cos (2πfc t + θ(t)) (2.70)

This provides an alternate representation of the passband signal in terms of baseband envelope
and phase signals.

Q
u s (t) u(t)
e(t)
θ(t )
I
u c (t)

Figure 2.29: Geometry of the complex envelope.

Complex envelope: To obtain a third representation of a passband signal, we note that a

two-dimensional point can also be mapped to a complex number; see Section 2.1. We define the
complex envelope u(t) of the passband signal up (t) in (2.68) and (2.70) as follows:

u(t) = uc (t) + jus (t) = e(t)ejθ(t) (2.71)

We can now express the passband signal in terms of its complex envelope. From (2.70), we see
that
up (t) = e(t)Re ej(2πfc t+θ(t)) = Re e(t)ej(2πfc t+θ(t)) = Re e(t)ejθ(t) ej2πfc t

This leads to our third representation of a passband signal:

up (t) = Re u(t)ej2πfc t


(2.72)

While we have obtained (2.72) using the polar representation (2.70), we should also check that
it is consistent with the rectangular representation (2.68), writing out the real and imaginary
parts of the complex waveforms above as follows:

u(t)ej2πfc t = (uc (t) + jus (t)) (cos 2πfc t + j sin 2πfc t)

(2.73)
= (uc (t) cos 2πfc t − us (t) sin 2πfc t) + j (us (t) cos 2πfc t + uc (t) sin 2πfc t)

Taking the real part, we obtain the expression (2.68) for up (t).
The relationship between the three time domain representations of a passband signal in terms
of its complex envelope is depicted in Figure 2.29. We now specify the corresponding frequency
domain relationship.
Information resides in complex baseband: The complex baseband representation corre-
sponds to subtracting out the rapid, but predictable, phase variation due to the fixed reference

66
frequency fc , and then considering the much slower amplitude and phase variations induced by
baseband modulation. Since the phase variation due to fc is predictable, it cannot convey any
information. Thus, all the information in a passband signal is contained in its complex envelope.
Choice of frequency/phase reference is arbitrary: We can define the complex baseband
representation of a passband signal using an arbitrary frequency reference fc (and can also vary
the phase reference), as long as we satisfy fc > W , where W is the bandwidth. We may often wish
to transform the complex baseband representations for two different references. For example, we
can write
up (t) = uc1(t) cos(2πf1 t+θ1 )−us1 (t) sin(2πf1 t+θ1 ) = uc2 (t) cos(2πf2 t+θ2 )−us2 (t) sin(2πf2 t+θ2 )
We can express this more compactly in terms of the complex envelopes u1 = uc1 + jus1 and
u2 = uc2 + jus2:
up (t) = Re u1 (t)ej(2πf1 t+θ1 ) = Re u2 (t)ej(2πf2 t+θ2 )



(2.74)
We can now find the relationship between these complex envelopes by transforming the expo-
nential term for one reference to the other:
up (t) = Re u1 (t)ej(2πf1 t+θ1 ) = Re [u1 (t)ej(2π(f1 −f2 )t+θ1 −θ2 ) ]ej(2πf2 t+θ2 )



(2.75)
Comparing with the extreme right-hand sides of (2.74) and (2.75), we can read off that
u2 (t) = u1 (t)ej(2π(f1 −f2 )t+θ1 −θ2 )
While we derived this result using algebraic manipulations, it has the following intuitive interpre-
tation: if the instantaneous phase 2πfi t + θi of the reference is ahead/behind, then the complex
envelope must be correspondingly retarded/advanced, so that the instantaneous phase of the
overall passband signal stays the same. We illustrate this via some examples below.

Example 2.8.2 (Change of reference frequency/phase) Consider the passband signal up (t) =
I[−1,1] (t) cos 400πt.
(a) Find the output when up (t) cos 401πt is passed through a lowpass filter.
(b) Find the output when up (t) sin(400πt − π4 ) is passed through a lowpass filter.
Solution: From Figure 2.28, we recognize that both (a) and (b) correspond to downconversion
operations with different frequency and phase references. Thus, by converting the complex en-
velope with respect to the appropriate reference, we can read off the answers.
(a) Letting u1 = uc1 + jus1 denote the complex envelope with respect to the reference ej401πt , we
recognize that the output of the LPF is uc1/2. The passband signal can be written as
up (t) = I[−1,1] (t) cos 400πt = Re I[−1,1] (t)ej400πt

We can now massage it to read off the complex envelope for the new reference:
up (t) = Re I[−1,1] (t)e−jπt ej401πt

from which we see that u1 (t) = I[−1,1] (t)e−jπt = I[−1,1] (t) (cos πt − j sin πt). Taking real and
imaginary parts, we obtain uc1(t) = I[−1,1] (t) cos πt and us1 (t) = −I[−1,1] (t) sin πt, respectively.
Thus, the LPF output is 12 I[−1,1] (t) cos πt.
π
(b) Letting u2 = uc2 + jus2 denote the complex envelope with respect to the reference ej(400πt− 4 ) ,
we recognize that the output of the LPF is −us2 /2. We can convert to the new reference as
before: π π

up (t) = Re I[−1,1] (t)ej 4 ej(400πt− 4 )
π
which gives the complex envelope u2 = I[−1,1] (t)ej 4 = I[−1,1] (t) cos π4 + j sin π4 . Taking real and


imaginary parts, we obtain uc2 (t) = I[−1,1] (t) cos π4 and us2(t) = I[−1,1] (t) sin π4 , respectively. Thus,
the LPF output is given by −us2 /2 = − 12 I[−1,1] (t) sin π4 = − 2√1 2 I[−1,1] (t).

67
From a practical point of view, keeping track of frequency/phase references becomes important
for the task of synchronization. For example, the carrier frequency used by the transmitter for
upconversion may not be exactly equal to that used by the receiver for downconversion. Thus,
the receiver must compensate for the phase rotation incurred by the complex envelope at the
output of the downconverter, as illustrated by the following example.

Example 2.8.3 (Modeling and compensating for frequency/phase offsets in complex

baseband): Consider the passband signal up (2.68), with complex baseband representation
u = uc + jus . Now, consider a phase-shifted version of the passband signal
ũp (t) = uc (t) cos(2πfc t + θ(t)) − us (t) sin(2πfc t + θ(t))
where θ(t) may vary slowly with time. For example, a carrier frequency offset ∆f and a phase
offset γ corresponds to θ(t) = 2π∆f t + γ. Suppose, now, that the signal is downconverted as
in Figure 2.28, where we take the phase reference as that of the receiver’s local oscillator (LO).
How do the I and Q components depend on the phase offset of the received signal relative to the
LO? The easiest way to answer this is to find the complex envelope of ũp with respect to fc . To
do this, we write ũp in the standard form (2.70) as follows:

ũp (t) = Re u(t)ej(2πfc t+θ(t))

Comparing with the desired form

ũp (t) = Re(ũ(t)ej2πfc t )
we can read off
ũ(t) = u(t)ejθ(t) (2.76)
Equation (2.76) relates the complex envelopes before and after a phase offset. We can expand
out this “polar form” representation to obtain the corresponding relationship between the I and
Q components. Suppressing time dependence from the notation, we can rewrite (2.76) as
ũc + j ũs = (uc + jus )(cos θ + j sin θ)
using Euler’s formula. Equating real and imaginary parts on both sides, we obtain
ũc = uc cos θ − us sin θ
(2.77)
ũs = uc sin θ + us cos θ
The phase offset therefore results in the I and Q components being mixed together at the output
of the downconverter. Thus, for a coherent receiver recovers the original I and Q components uc ,
us , we must account for the (possibly time varying) phase offset θ(t). In particular, if we have
an estimate of the phase offset, then we can undo it by inverting the relationship in (2.76):

u(t) = ũ(t)e−jθ(t) (2.78)

which can be written out in terms of real-valued operations as follows:
uc = ũc cos θ + ũs sin θ
(2.79)
us = −ũc sin θ + ũs cos θ
The preceding computations provide a typical example of the advantage of working in complex
baseband. Relationships between passband signals can be compactly represented in complex
baseband, as in (2.76) and (2.78). For signal processing using real-valued arithmetic, these
complex baseband relationships can be expanded out to obtain relationships involving real-valued
quantities, as in (2.77) and (2.79). See Software Lab 2.1 for an example of such computations.

68
 Re(Up (f)) Im(Up (f))
B
A
−fc
f f
−fc fc fc

Re(C(f)) Im(C(f))
2B
2A

f f
fc fc

Re(U(f)) Im(U(f))
2B
2A

f f

Figure 2.30: Frequency domain relationship between a real-valued passband signal and its com-
plex envelope. The figure shows the spectrum Up (f ) of the passband signal, its scaled restriction
to positive frequencies C(f ), and the spectrum U(f ) of the complex envelope.

69
2.8.2 Frequency Domain Relationships
Consider an arbitrary complex-valued baseband waveform u(t) whose frequency content is con-
tained in [−W, W ], and suppose that fc > W . We want to show that

up (t) = Re u(t)ej2πfc t = Re (c(t))

(2.80)
is a real-valued passband signal whose frequency is concentrated around ±fc , away from DC. Let
c(t) = u(t)ej2πfc t ↔ C(f ) = U(f − fc ) (2.81)
That is, C(f ) is the complex envelope U(f ), shifted to the right by fc . Since U(f ) has frequency
content in [−W, W ], C(f ) has frequency content around [fc − W, fc + W ]. Since fc − W > 0, this
band does not include DC. Now,
1 1
up (t) = Re (c(t)) = (c(t) + c∗ (t)) ↔ Up (f ) = (C(f ) + C ∗ (−f ))
2 2
Since C ∗ (−f ) is the complex conjugated version of C(f ), flipped around the origin, it has fre-
quency content in the band of negative frequencies [−fc − W, −fc + W ] around −fc , which does
not include DC because −fc + W < 0. Thus, we have shown that up (t) is a passband signal. It
is real-valued by virtue of its construction using the time domain equation (2.80), which involves
taking the real part. But we can also doublecheck for consistency in the frequency domain:
Up (f ) is conjugate symmetric, since its positive frequency component is C(f ), and its nega-
tive frequency component is C ∗ (−f ). Substituting C(f ) by U(f − fc ), we obtain the passband
spectrum in terms of the complex baseband spectrum:
1
Up (f ) = (U(f − fc ) + U ∗ (−f − fc )) (2.82)
2

So far, we have seen how to construct a real-valued passband signal given a complex-valued
baseband signal. To go in reverse, we must answer the following: do the equivalent representa-
tions (2.68), (2.70), (2.72) and (2.82) hold for any passband signal, and if so, how do we find
the spectrum of the complex envelope given the spectrum of the passband signal? To answer
these questions, we simply trace back the steps we used to arrive at (2.82). Given the spec-
trum Up (f ) for a real-valued passband signal up (t), we construct C(f ) as a scaled version of
Up+ (f ) = Up (f )I[0,∞) (f ), the positive frequency part of Up (f ), as follows:

+ 2Up (f ) , f > 0
C(f ) = 2Up (f ) =
0, f <0

This means that Up (f ) = 21 C(f ) for positive frequencies. By the conjugate symmetry of Up (f ),
the negative frequency component must be 21 C ∗ (−f ), so that Up (f ) = 12 C(f ) + 12 C ∗ (−f ). In the
time domain, this corresponds to
1 1
up (t) = c(t) + c∗ (t) = Re (c(t)) (2.83)
2 2
Now, let us define the complex envelope as follows:
u(t) = c(t)e−j2πfc t ↔ U(f ) = C(f + fc )
Since c(t) = u(t)ej2πfc t , we obtain the desired relationship (2.68) on substituting into (2.83).
Since C(f ) has frequency content in a band around fc , U(f ), which is obtained by shifting C(f )
to the left by fc , is indeed a baseband signal with frequency content in a band around DC.

70
Frequency domain expressions for I and Q components: If we are given the time domain
complex envelope, we can read off the I and Q components as the real and imaginary parts:

uc (t) = Re(u(t)) = 12 (u(t) + u∗ (t))

us (t) = Im(u(t)) = 2j1 (u(t) − u∗ (t))

Taking Fourier transforms, we obtain

Uc (f ) = 12 (U(f ) + U ∗ (−f ))
Us (f ) = 2j1 (U(f ) − U ∗ (−f ))

Figure 2.30 shows the relation between the passband signal Up (f ), its scaled version C(f ) re-
stricted to positive frequencies, and the complex baseband signal U(f ). As this example em-
phasizes, all of these spectra can, in general, be complex-valued. Equation (2.80) corresponds to
starting with an arbitrary baseband signal U(f ) as in the bottom of the figure, and constructing
C(f ) as depicted in the middle of the figure. We then use C(f ) to construct a conjugate sym-
metric passband signal Up (f ), proceeding from the middle of the figure to the top. This example
also shows that U(f ) does not, in general, obey conjugate symmetry, so that the baseband signal
u(t) is, in general, complex-valued. However, by construction, Up (f ) is conjugate symmetric, and
hence the passband signal up (t) is real-valued.

Example 2.8.4 Let vp (t) denote a real-valued passband signal, with Fourier transform Vp (f )
specified as follows for negative frequencies:

−(f + 99) −101 ≤ f ≤ −99
Vp (f ) =
0 f < −101 or − 99 < f ≤ 0
(a) Sketch Vp (f ) for both positive and negative frequencies.
(b) Without explicitly taking the inverse Fourier transform, can you say whether vp (t) = vp (−t)
or not?
(c) Find and sketch Vc (f ) and Vs (f ), the Fourier transforms of the I and Q components with
respect to a reference frequency fc = 99. Do this without going to the time domain.
(d) Find an explicit time domain expression for the output when vp (t) cos 200πt is passed through
an ideal lowpass filter of bandwidth 4.
(e) Find an explicit time domain expression for the output when vp (t) sin 202πt is passed through
an ideal lowpass filter of bandwidth 4.
Solution:
Vp (f)

f
~
~

~
~

−101 −99 99 101

Figure 2.31: Sketch of passband spectrum for Example 2.8.4.

(a) Since vp (t) is real-valued, we have Vp (f ) = Vp∗ (−f ). Since the spectrum is also given to be
real-valued for f ≤ 0, we have Vp∗ (−f ) = Vp (−f ). The spectrum is sketched in Figure 2.31.
(b) Yes, vp (t) = vp (−t). Since vp (t) is real-valued, we have vp (−t) = vp∗ (−t) ↔ Vp∗ (f ). But
Vp∗ (f ) = Vp (f ), since the spectrum is real-valued.

71
 V(f)
4

f
0 2

Vc (f)

f
−2 2

j Vs (f)
2
−2
f
2
−2

Figure 2.32: Sketch of I and Q spectra in Example 2.8.4(c), taking reference frequency fc = 99.

(c) The spectrum of the complex envelope and the I and Q components are shown in Figure 2.32.
The complex envelope is obtained as V (f ) = 2Vp+ (f + fc ), while the I and Q components satisfy

V (f ) + V ∗ (−f ) V (f ) − V ∗ (−f )
Vc (f ) = , Vs (f ) =
2 2j
In our case, Vc (f ) = |f |I[−2,2] (f ) and jVs (f ) = f I[−2,2] (f ) are real-valued, and are plotted in the
figure.
V(f)
4

f
−1 0 1

Vc (f)
2

f
−1 1

Figure 2.33: Finding the I component in Example 2.8.4(d), taking reference frequency as fc =
100.

(d) The output of the LPF is vc (t)/2, where vc is the I component with respect to fc = 100. In
Figure 2.33, we construct the complex envelope and the I component as in (c), except that the
reference frequency is different. Clearly, the boxcar spectrum corresponds to vc (t) = 4sinc(2t),
so that the output is 2sinc(2t).

72
 V(f)
4

f
0 2

j Vs (f)
2

f
−2 2

−2
Figure 2.34: Finding the Q component in Example 2.8.4(e), taking reference frequency as fc =
101.

(e) The output of the LPF is −vs (t)/2, where vs is the I component with respect to fc = 101. In
Figure 2.34, we construct the complex envelope and the Q component as in (c), except that the
reference frequency is different. We now have to take the inverse Fourier transform, which is a
little painful if we do it from scratch. Instead, let us differentiate to see that

dVs (f )
j = I[−2,2] (f ) − 4δ(f ) ↔ 4sinc(4t) − 4
df
dVs (f )
But df
↔ −j2πtvs (t), so that j dVdf
s (f )
↔ 2πtvs (t). We therefore obtain that 2πtvs (t) =
2(sinc(4t)−1) (1−sinc(4t))
4sinc(4t) − 4, or vs (t) = πt
. Thus, the output of the LPF is −vs (t)/2, or πt
.

Passband Complex Baseband

Hp (f) H(f)
A 2A Filter

fc f f
Up (f) U(f)
B 2B Input

fc f f

Yp (f) Y(f)
AB 2AB Output

fc f f

Figure 2.35: The relationship between passband filtering and its complex baseband analogue.

73
 hc y (t)
c
−
hs
Passband uc (t)
Downconverter
Signal us (t)
u p (t) hc ys (t)

hs

Real baseband operations

1
Figure 2.36: Complex baseband realization of passband filter. The constant scale factors of 2
have been omitted.

2.8.3 Complex baseband equivalent of passband filtering

We now state another result that is extremely relevant to transceiver operations; namely, any
passband filter can be implemented in complex baseband. This result applies to filtering op-
erations that we desire to perform at the transmitter (e.g., to conform to spectral masks), at
the receiver (e.g., to filter out noise), and to a broad class of channels modeled as linear fil-
ters. Suppose that a passband signal up (t) = uc (t) cos 2πfc t − us (t) sin 2πfc t is passed through
a passband filter with impulse response hp (t) = hc (t) cos 2πfc t − hs (t) sin 2πfc t to get an output
yp (t) = (up ∗ hp )(t). In the frequency domain, Yp (f ) = Hp (f )Up (f ), so that the output yp (t) is
also passband, and can be written as yp (t) = yc (t) cos 2πfc t−ys (t) sin 2πfc t. How are the I and Q
components of the output related to those of the input and the filter impulse response? We now
show that a compact answer is given in terms of complex envelopes: the complex envelope y is the
convolution of the complex envelopes of the input and the impulse response, up to a scale factor.
Let y, u and h denote the complex envelopes for yp , up and hp , respectively, with respect to a
common frequency reference fc . Since real-valued passband signals are completely characterized
by their spectra for positive frequencies, the passband filtering equation Yp (f ) = Up (f )Hp (f )
can be separately (and redundantly) written out for positive and negative frequencies, because
the waveforms are conjugate symmetric around the origin, and there is no energy around f = 0.
Thus, focusing on the positive frequency segments Y + (f ) = Yp (f )I{f >0} , U + (f ) = Up (f )I{f >0} ,
H + (f ) = Hp (f )I{f >0} , we have Y + (f ) = U + (f )H + (f ), from which we conclude that the complex
envelope of y is given by
1
Y (f ) = 2Y + (f + fc ) = 2U + (f + fc )H + (f + fc ) = U(f )H(f )
2
Figure 2.35 depicts the relationship between the passband and complex baseband waveforms
in the frequency domain, and supplies a pictorial proof of the preceding relationship. We now
restate this important result in the time domain:
1
y(t) = (u ∗ h)(t) (2.84)
2
A practical consequence of this is that any desired passband filtering function can be realized in
complex baseband. As shown in Figure 2.36, this requires four real baseband filters: writing out
the real and imaginary parts of (2.84), we obtain
1 1
yc = (uc ∗ hc − us ∗ hs ), ys = (us ∗ hc + uc ∗ hs ) (2.85)
2 2

74
 s(t)
1
1 1
1/2
* =
−1 1 0 3 −1 1 2 4 t

Figure 2.37: Convolution of two boxes for Example 2.8.5.

Example 2.8.5 The passband signal u(t) = I[−1,1] (t) cos 100πt is passed through the passband
filter h(t) = I[0,3] (t) sin 100πt. Find an explicit time domain expression for the filter output.
Solution: We need to find the convolution yp (t) of the signal up (t) = I[−1,1] (t) cos 100πt with the
impulse response hp (t) = I[0,3] (t) sin 100πt, where we have inserted the subscript to explicitly
denote that the signals are passband. The corresponding relationship in complex baseband is
y = (1/2)u ∗ h. Taking a reference frequency fc = 50, we can read off the complex envelopes
u(t) = I[−1,1] (t) and h(t) = −jI[0,3] (t), so that

y = (−j/2)I[−1,1] (t) ∗ I[0,3] (t)

Let s(t) = (1/2)I[−1,1] (t) ∗ I[0,3] (t) denote the trapezoid obtained by convolving the two boxes, as
shown in Figure 2.37. Then
y(t) = −js(t)
That is, yc = 0 and ys = −s(t), so that yp (t) = s(t) sin 100πt.

2.8.4 General Comments on Complex Baseband

Remark 2.8.1 (Complex Baseband in Transceiver Implementations) Given the equiva-

lence of passband and complex baseband, and the fact that key operations such as linear filtering
can be performed in complex baseband, it is understandable why, in typical modern passband
transceivers, most of the intelligence is moved to baseband processing. For moderate bandwidths
at which analog-to-digital and digital-to-analog conversion can be accomplished inexpensively,
baseband operations can be efficiently performed in DSP. These digital algorithms are indepen-
dent of the passband over which communication eventually occurs, and are amenable to a variety
of low-cost implementations, including Very Large Scale Integrated Circuits (VLSI), Field Pro-
grammable Gate Arrays (FPGA), and general purpose DSP engines. On the other hand, analog
components such as local oscillators, power amplifiers and low noise amplifiers must be opti-
mized for the bands of interest, and are often bulky. Thus, the trend in modern transceivers is
to accomplish as much as possible using baseband DSP algorithms. For example, complicated
filters shaping the transmitted waveform to a spectral mask dictated by the FCC can be achieved
with baseband DSP algorithms, allowing the use of relatively sloppy analog filters at passband.
Another example is the elimination of analog phase locked loops for carrier synchronization in
many modern receivers; the receiver instead employs a fixed analog local oscillator for downcon-
version, followed by a digital phase locked loop, or a one-shot carrier frequency/phase estimate,
implemented in complex baseband.

Energy and power: The energy of a passband signal equals that of its complex envelope, up
to a scale factor which depends on the particular convention we adopt. In particular, for the
convention in (2.68), we have

1
1
||up ||2 = ||uc||2 + ||us ||2 = ||u||2 (2.86)
2 2

75
That is, the energy equals the sum of the energies of the I and Q components, up to a scalar
constant. The same relationship holds for the powers of finite-power passband signals and their
complex envelopes, since power is computed as a time average of energy. To show (2.86), consider

||up||2 = R (uc (t) cos 2πfc t − us (t) sin 2πfc t)2 dt

R

= u2c (t) cos2 (2πfc t)dt + u2s (t) sin2 (2πfc t)dt − 2 uc (t) cos 2πfc t us (t) sin 2πfc tdt
R R

The I-Q cross term drops out due to I-Q orthogonality, so that we are left with the I-I and Q-Q
terms, as follows:
Z Z
2
||up|| = u2c (t) cos2 (2πfc t)dt + u2s (t) sin2 (2πfc t)dt

Now, cos2 2πfc t = 1

2
+ 21 cos 4πfc t and sin2 2πfc t = 1
2
− 12 cos 4πfc t. We therefore obtain

1 1 1 1
Z Z Z Z
2
||up|| h= u2c (t)dt + u2s (t)dt + u2c (t) cos 4πfc tdt − u2s (t) cos 4πfc tdt
2 2 2 2

The last two terms are zero, since they are equal to the DC components of passband waveforms
centered around 2fc , arguing in exactly the same fashion as in our derivation of I-Q orthogonality.
This gives the desired result (2.86).
Correlation between two signals: The correlation, or inner product, of two real-valued
passband signals up and vp is defined as
Z ∞
hup , vp i = up (t)vp (t)dt
−∞

Using exactly the same reasoning as above, we can show that

1
hup , vp i = (huc , vc i + hus , vs i) (2.87)
2

That is, we can implement a passband correlation by first downconverting, and then employing
baseband operations: correlating I against I, and Q against Q, and then summing the results. It
is also worth noting how this is related to the complex baseband inner product, which is defined
as R∞ R∞
hu, vi = −∞ u(t)v ∗ (t)dt = −∞ (uc (t) + jus (t)) (vc (t) − jvs (t))
(2.88)
= (huc , vc i + hus , vs i) + j (hus , vc i − huc , vs i)

Comparing with (2.87), we obtain that

1
hup , vp i = Re (hu, vi)
2

That is, the passband inner product is the real part of the complex baseband inner product (up to
scale factor). Does the imaginary part of the complex baseband inner product have any meaning?
Indeed it does: it becomes important when there is phase uncertainty in the downconversion
operation, which causes the I and Q components to leak into each other. However, we postpone
discussion of such issues to later chapters.

76
2.9 Wireless Channel Modeling in Complex Baseband
We now provide a glimpse of wireless channel modeling using complex baseband. There are two
key differences between wireless and wireline communication. The first, which is what we focus
on now, is multipath propagation due to reflections off of scatterers adding up at the receiver.
This addition can be constructive or destructive (as we saw in Example 2.5.4), and is sensitive to
small changes in the relative location of the transmitter and receiver which produce changes in
the relative delays of the various paths. The resulting fluctuations in signal strength are termed
fading. The second key feature of wireless, which we explore in a different wireless module,
is interference: wireless is a broadcast medium, hence the receiver can also hear transmissions
other than the one it is interested in. We now explore the effects of multipath fading for some
simple scenarios. While we just made up the example impulse response in Example 2.5.4, we
now consider more detailed, but still simplified, models of the propagation environment and the
associated channel models.
Consider a passband transmitted signal at carrier frequency, of the form
up (t) = uc (t) cos 2πfc t − us (t) sin 2πfc t = e(t) cos(2πfc t + θ(t))
where
u(t) = uc (t) + jus (t) = e(t)ejθ(t)
is the complex baseband representation, or complex envelope. In order to model the propagation
of this signal through a multipath environment, let us consider its propagation through a path
of length r. The propagation attenuates the field by a factor of 1/r, and introduces a delay of
τ (r) = rc , where c denotes the speed of light. Suppressing the dependence of τ on r, the received
signal is given by
A
vp (t) = e(t − τ ) cos(2πfc (t − τ ) + θ(t − τ ) + φ)
r
where we consider relative values (across paths) for the constants A and φ. The complex envelope
of vp (t) with respect to the reference ej2ıfc t is given by
A
v(t) = u(t − τ )e−j(2πfc τ +φ) (2.89)
r
For example, we may take A = 1, φ = 0 for a direct, or line of sight (LOS), path from transmitter
to receiver, which we may take as a reference. Figure 2.38 shows the geometry of for a reflected
path corresponding to a single bounce, relative to the LOS path. Follow standard terminology, θi
denotes the angle of incidence, and θg = π2 −θi the grazing angle. The change in relative amplitude
and phase due to the reflection depends on the carrier frequency, the reflector material, the angle
of incidence, and the polarization with respect to the orientation of the reflector surface. Since
we do not wish to get into the underlying electromagnetics, we consider simplified models of
relative amplitude and phase. In particular, we note that for grazing incidence (θg ≈ 0), we have
A ≈ 1, φ ≈ π.
Generalizing (2.89) to multiple paths of length r1 , r2 , ..., the complex envelope of the received
signal is given by
X Ai
v(t) = u(t − τi )e−j(2πfc τi +φi ) (2.90)
i
r i

where τi = rci , and Ai , φi depend on the reflector characteristic and incidence angle for the ith
ray. This corresponds to the complex baseband channel impulse response
X Ai
h(t) = e−j(2πfc τi +φi ) δ(t − τi ) (2.91)
i
ri

77
 Receiver
Transmitter LOS path

θi Reflected path hr
ht
θg
Reflector

r=length of reflected path

Virtual source

Range R

Figure 2.38: Ray tracing for a single bounce path. We can reflect the transmitter around the
reflector to create a virtual source. The line between the virtual source and the receiver tells us
where the ray will hit the reflector, following the law of reflection that the angles of incidence
and reflection must be equal. The length of the line equals the length of the reflected ray to be
plugged into (2.92).

Ai −j(2πfc τi +φi )
This is in exact correspondence with our original multipath model (2.36), with αi = ri
e .
The corresponding frequency domain response is given by

X Ai
H(f ) = e−j(2πfc τi +φi ) e−j2πf τi (2.92)
i
ri

Since we are modeling in complex baseband, f takes values around DC, with f = 0 corresponding
to the passband reference frequency fc .
Channel delay spread and coherence bandwidth: We have already introduced these con-
cepts in Example 2.5.4, but reiterate them here. Let τmin and τmax denote the minimum and
maximum of the delays {τi }. The difference τd = τmax − τmin is called the channel delay spread.
The reciprocal of the delay spread is termed the channel coherence bandwidth, Bc = τ1d . A base-
band signal of bandwidth W is said to be narrowband if W τd = W/Bc ≪ 1, or equivalently, if
its bandwidth is significantly smaller than the channel coherence bandwidth.
We can now infer that, for a narrowband signal around the reference frequency, the received
complex baseband signal equals a delayed version of the transmitted signal, scaled by the complex
channel gain
X Ai
h = H(0) = e−j(2πfc τi +φi ) (2.93)
i
r i

Example 2.9.1 (Two ray model) Suppose our propagation environment consists of the LOS
ray and the single reflectedpray shown in Figure 2.38. Then we have two rays, with r1 =
p
R2 + (hr − ht )2 and r2 = R2 + (hr + ht )2 . The corresponding delays are τi = ri /c, i = 1, 2,
where c denotes the speed of propagation. The grazing angle is given by θg = tan−1 ht +h
R
r
. Setting
A1 = 1 and φ1 = 0, once we specify A2 and φ2 for the reflected path, we can specify the complex
baseband channel. Numerical examples are explored in Problem 2.21, and in Software Lab 2.2.

78
2.10 Concept Inventory

In addition to a review of basic signals and systems concepts such as convolution and Fourier
transforms, the main focus of this chapter is to develop the complex baseband representation of
passband signals, and to emphasize its crucial role in modeling and implementation of commu-
nication systems.
Review
• Euler’s formula: ejθ = cos θ + j sin θ
• Important signals: delta function (sifting property), indicator function, complex exponential,
sinusoid, sinc
• Signals analogous to vectors: Inner product, energy and norm
• LTI systems: impulse response, convolution, complex exponentials as eigenfunctions, multipath
channel modeling
• Fourier series: complex exponentials or sinusoids as basis for periodic signals, conjugate sym-
metry for real-valued signals, Parseval’s identity, use of differentiation to simplify computation
• Fourier transform: standard pairs (sinc and boxcar, impulse and constant), effect of time de-
lay and frequency shift, conjugate symmetry for real-valued signals, Parseval’s identity, use of
differentiation to simplify computation, numerical computation using DFT
• Bandwidth: for physical signals, given by occupancy of positive frequencies; energy spectral
density equals magnitude squared of Fourier transform; computation of fractional energy con-
tainment bandwidth from energy spectral density

Complex baseband representation

• Complex envelope of passband signal: rectangular form (I and Q components), polar form
(envelope and phase), upconversion and downconversion, orthogonality of I and Q components
(under ideal synchronization), frequency domain relationship between passband signal and its
complex envelope
• Passband filtering can be accomplished in complex baseband
• Passband inner product and energy in terms of complex baseband quantities
Modeling in complex baseband
• Frequency and phase offsets: rotating phasor multiplying complex envelope, derotation to undo
offsets
• Wireless multipath channel: impulse response modeled as sum of impulses with complex-valued
coefficients, ray tracing, delay spread and coherence bandwidth

2.11 Endnotes

A detailed treatment of the material reviewed in Sections 2.1-2.5 can be found in basic textbooks
on signals and systems such as Oppenheim, Willsky and Nawab [17] or Lathi [18].
The Matlab code fragments and software labs interspersed in this textbook provide a glimpse of
the use of DSP in communication. However, for a background in core DSP algorithms, we refer
the reader to textbooks such as Oppenheim and Schafer [19] and Mitra [20].

79
Problems
LTI systems and Convolution

Problem 2.1 A system with input x(t) has output given by

Z t
y(t) = eu−t x(u)du
−∞

(a) Show that the system is LTI and find its impulse response.
(b) Find the transfer function H(f ) and plot |H(f )|.
(c) If the input x(t) = 2sinc(2t), find the energy of the output.

Problem 2.2 Find and sketch y = x1 ∗ x2 for the following:

(a) x1 (t) = e−t I[0,∞) (t), x2 (t) = x1 (−t).
(b) x1 (t) = I[0,2] (t) − 3I[1,4] (t), x2 (t) = I[0,1] (t).
Hint: In (b), you can use the LTI property and the known result in Figure 2.12 on the convolution
of two boxes.

Fourier Series

Problem 2.3 A digital circuit generates the following periodic waveform with period 0.5:

1, 0 ≤ t < 0.1
u(t) =
0, 1 ≤ t < 0.5
where the unit of time is microseconds throughout this problem.
(a) Find the complex exponential Fourier series for du/dt.
(b) Find the complex exponential Fourier series for u(t), using the results of (a).
(c) Find an explicit time domain expression for the output when u(t) is passed through an ideal
lowpass filter of bandwidth 100 KHz.
(d) Repeat (c) when the filter bandwidth is increased to 300 KHz.
(e) Find an explicit time domain expression for the output when u(t) is passed through a filter
with impulse response h2 (t) = sinc(t) cos(8πt).
(f) Can you generate a sinusoidal waveform of frequency 1 MHz by appropriately filtering u(t)?
If so, specify in detail how you would do it.

Fourier Transform and Bandwidth

Problem 2.4 Find and sketch the Fourier transforms for the following signals:
(a) u(t) = (1 − |t|)I[−1,1] (t).
(b) v(t) = sinc(2t)sinc(4t).
(c) s(t) = v(t) cos 200πt.
(d) Classify each of the signals in (a)-(c) as baseband or passband.

Problem
R∞ 2.5 Use Parseval’s identity to compute the following integrals:
(a) −∞ sinc2 (2t)dt
R∞
(b) 0 sinc(t)sinc(2t)dt

80
Problem 2.6 (a) For u(t) = sinc(t) sinc(2t), where t is in microseconds, find and plot the
magnitude spectrum |U(f )|, carefully labeling the units of frequency on the x axis.
(b) Now, consider s(t) = u(t) cos 200πt. Plot the magnitude spectrum |S(f )|, again labeling
the units of frequency and carefully showing the frequency intervals over which the spectrum is
nonzero.

Problem 2.7 The signal s(t) = sinc4t is passed through a filter with impulse response h(t) =
sinc2 t cos 4πt to obtain output y(t). Find and sketch the Fourier transform Y (f ) of the output
(sketch the real and imaginary parts separately if the spectrum is complex-valued).

Problem 2.8 Consider the tent signal s(t) = (1 − |t|)I[−1,1] (t).

(a) Find and sketch the Fourier transform S(f ).
(b) Compute the 99% energy containment bandwidth in KHz, assuming that the unit of time is
milliseconds.

Problem 2.9 Consider the cosine pulse

p(t) = cos πt I[−1/2,1/2] (t)

(a) Show that the Fourier transform of this pulse is given by

2 cos πf
P (f ) =
π(1 − 4f 2 )

(b) Use this result to derive the formula (2.63) for the sine pulse in Example 2.5.5.

Problem 2.10 (Numerical computation of the Fourier transform) Modify Code Frag-
ment 2.5.1 for Example 2.5.5 to numerically compute the Fourier transform of the tent function
in Problem 2.8. Display the magnitude spectra of the DFT-based numerically computed Fourier
transform and the analytically computed Fourier transform (from Problem 2.8) in the same plot,
over the frequency interval [−10, 10]. Comment on the accuracy of the DFT-based computation.

Introducing the matched filter

Problem 2.11 For a signal s(t), the matched filter is defined as a filter with impulse response
h(t) = smf (t) = s∗ (−t) (we allow signals to be complex valued, since we want to handle complex
baseband signals as well as physical real-valued signals).
(a) Sketch the matched filter impulse response for s(t) = I[1,3] (t).
(b) Find and sketch the convolution y(t) = (s ∗ smf )(t). This is the output when the signal is
passed through its matched filter. Where does the peak of the output occur?
(c) (True or False) Y (f ) ≥ 0 for all f .

Problem 2.12 Repeat Problem 2.11 for s(t) = I[1,3] (t) − 2I[2,5] (t).

Introducing delay spread and coherence bandwidth

Problem 2.13 A wireless channel has impulse response given by h(t) = 2δ(t − 0.1) + jδ(t −
0.64) − 0.8δ(t − 2.2), where the unit of time is in microseconds.
(a) What is the delay spread and coherence bandwidth?

81
(b) Plot the magnitude and phase of the channel transfer function H(f ) over the interval
[−2Bc , 2Bc ], where Bc denotes the coherence bandwidth computed in (a). Comment on how
the phase behaves when |H(f )| is small.
(c) Express |H(f )| in dB, taking 0 dB as the gain of a nominal channel hnom (t) = 2δ(t − 0.1)
corresponding to the first ray alone. What are the fading depths that you see with respect to
this nominal?
Define the average channel power gain over a band [−W/2, W/2] as
W/2
1
Z
Ḡ(W ) = |H(f )|2 df
W −W/2

This is a simplified measure of how increasing signal bandwidth W can help compensate for
frequency-selective fading: we hope that, as W gets large, we can average out fluctuations in
|H(f )|.
(d) Plot Ḡ(W ) as a function of W/Bc , and comment on how large the bandwidth needs to be
(as a multiple of Bc ) to provide “enough averaging.”

Complex envelope of passband signals

Problem 2.14 Consider a passband signal of the form

up (t) = a(t) cos 200πt

where a(t) = sinc(2t), and where the unit of time is in microseconds.

(a) What is the frequency band occupied by up (t)?
(b) The signal up (t) cos 199πt is passed through a lowpass filter to obtain an output b(t). Give
an explicit expression for b(t), and sketch B(f ) (if B(f ) is complex-valued, sketch its real and
imaginary parts separately).
(c) The signal up (t) sin 199πt is passed through a lowpass filter to obtain an output c(t). Give
an explicit expression for c(t), and sketch C(f ) (if C(f ) is complex-valued, sketch its real and
imaginary parts separately).
(d) Can you reconstruct a(t) from simple real-valued operations performed on b(t) and c(t)? If
so, sketch a block diagram for the operations required. If not, say why not.

s(t) LPF u (t)

2 cos(401π t)

Bandpass filter y(t)

LPF v (t)
h(t)

2 sin(400 π t+ π /4)

Figure 2.39: Operations involved in Problem 2.15.

82
Problem 2.15 Consider the signal s(t) = I[−1,1] (t) cos 400πt.
(a) Find and sketch the baseband signal u(t) that results when s(t) is downconverted as shown
in the upper branch of Figure 2.39.
(b) The signal s(t) is passed through the bandpass filter with impulse response h(t) = I[0,1] (t) sin(400πt+
π
4
). Find and sketch the baseband signal v(t) that results when the filter output y(t) = (s ∗ h)(t)
is downconverted as shown in the lower branch of Figure 2.39.

Problem 2.16 Consider the signals u1 (t) = I[0,1] (t)

R∞ cos 100πt and u2(t) = I[0,1] (t) sin 100πt.
(a) Find the numerical value of the inner product −∞ u1 (t)u2 (t)dt.
(b) Find an explicit time domain expression for the convolution y(t) = (u1 ∗ u2 )(t).
(c) Sketch the magnitude spectrum |Y (f )| for the convolution in (b).

Problem 2.17 Consider a real-valued passband signal vp (t) whose Fourier transform for positive
frequencies is given by 
 2, 30 ≤ f ≤ 32
Re(Vp (f )) = 0, 0 ≤ f < 30
0, 32 < f < ∞


 1 − |f − 32|, 31 ≤ f ≤ 33
Im(Vp (f )) = 0, 0 ≤ f < 31
0, 33 < f < ∞


(a) Sketch the real and imaginary parts of Vp (f ) for both positive and negative frequencies.
(b) Specify, in both the time domain and the frequency domain, the waveform that you get when
you pass vp (t) cos(60πt) through a low pass filter.

Problem 2.18 The passband signal u(t) = I[−1,1] (t) cos 100πt is passed through the passband
filter h(t) = I[0,3] (t) sin 100πt. Find an explicit time domain expression for the filter output.

Problem 2.19 Consider the passband signal up (t) = sinc(t) cos 20πt, where the unit of time is
in microseconds.
(a) Use Matlab to plot the signal (plot over a large enough time interval so as to include “most”
of the signal energy). Label the units on the time axis.
Remark: Since you will be plotting a discretized version, the sampling rate you should choose
should be large enough that the carrier waveform looks reasonably smooth (e.g., a rate of at least
10 times the carrier frequency).
(b) Write a Matlab program to implement a simple downconverter as follows. Pass x(t) =
2up (t) cos 20πt through a lowpass filter which consists of computing a sliding window average
Rt
over a window of 1 microsecond. That is, the LPF output is given by y(t) = t−1 x(τ ) dτ . Plot
the output and comment on whether it is what you expect to see.

Problem 2.20 Consider the following two passband signals:

up (t) = sinc(2t) cos 100πt
and
π
vp (t) = sinc(t) sin(101πt + )
4
(a) Find the complex envelopes u(t) and v(t) for up and vp , respectively, with respect to the
frequency reference fc = 50.
(b) What is the bandwidth of up (t)? What is the bandwidth of vp (t)?
(c) Find the inner product hup, vp i, using the result in (a).
(d) Find the convolution yp (t) = (up ∗ vp )(t), using the result in (a).

83
Wireless channel modeling

Problem 2.21 Consider the two-ray wireless channel model in Example 2.9.1.
(a) Show that, as long as the range R ≫ ht , hr the delay spread is well approximated as

2ht hr
τd ≈
Rc
where c denotes the propagation speed. We assume free space propagation with c = 3 × 108 m/s.
(b) Compare the approximation in (a) with the actual value of the delay spread for R = 200m,
ht = 2m, hr = 10m. (e.g., modeling an outdoor link with LOS and single ground bounce).
(c) What is the coherence bandwidth for the numerical example in (b).
(d) Redo (b) and (c) for R = 10m, ht = hr = 2m (e.g., a model for an indoor link modeling LOS
plus a single wall bounce).

Problem 2.22 Consider R = 200m, ht = 2m, hr = 10m in the two-ray wireless channel model
in Example 2.9.1. Assume A1 = 1 and φ1 = 0, set A2 = 0.95 and φ2 = π, and assume that the
carrier frequency is 5 GHz.
(a) Specify the channel impulse response, normalizing the LOS path to unit gain and zero delay.
Make sure you specify the unit of time being used.
(b) Plot the magnitude and phase of the channel transfer function over [−3Bc , 3Bc ], where Bc
denotes the channel coherence bandwidth.
(c) Plot the frequency selective fading gain in dB over [−3Bc , 3Bc ], using a LOS channel as
nominal. Comment on the fading depth.
(d) As in Problem 2.13, compute the frequency-averaged power gain Ḡ(W ) and plot it as a
function of W/Bc . How much bandwidth is needed to average out the effects of frequency-
selective fading?

Software Lab 2.1: Modeling Carrier Phase Uncertainty

PN
Consider a pair of independently modulated signals, uc (t) =
PN n=1 bc [n]p(t − n) and us (t) =
n=1 bs [n]p(t − n), where the symbols bc [n], bs [n] are chosen with equal probability to be +1 and
-1, and p(t) = I[0,1] (t) is a rectangular pulse. Let N = 100.
(1.1) Use Matlab to plot a typical realization of uc (t) and us (t) over 10 symbols. Make sure you
sample fast enough for the plot to look reasonably “nice.”
(1.2) Upconvert the baseband waveform uc (t) to get

up,1(t) = uc (t) cos 40πt

This is a so-called binary phase shift keyed (BPSK) signal, since the changes in phase due to
the changes in the signs of the transmitted symbols. Plot the passband signal up,1(t) over four
symbols (you will need to sample at a multiple of the carrier frequency for the plot to look nice,
which means you might have to go back and increase the sampling rate beyond what was required
for the baseband plots to look nice).
(1.3) Now, add in the Q component to obtain the passband signal

up (t) = uc (t) cos 40πt − us (t) sin 40πt

Plot the resulting Quaternary Phase Shift Keyed (QPSK) signal up (t) over four symbols.
(1.4) Downconvert up (t) by passing 2up (t) cos(40πt + θ) and 2up (t) sin(40πt + θ) through crude
lowpass filters with impulse response h(t) = I[0,0.25] (t). Denote the resulting I and Q components
by vc (t) and vs (t), respectively. Plot vc and vs for θ = 0 over 10 symbols. How do they compare

84
to uc and us ? Can you read off the corresponding bits bc [n] and bs [n] from eyeballing the plots
for vc and vs ?
(1.5) Plot vc and vs for θ = π/4. How do they compare to uc and us ? Can you read off the
corresponding bits bc [n] and bs [n] from eyeballing the plots for vc and vs ?
(1.6) Figure out how to recover uc and us from vc and vs if a genie tells you the value of θ (we are
looking for an approximate reconstruction–the LPFs used in downconversion are non-ideal, and
the original waveforms are not exactly bandlimited). Check whether your method for undoing
the phase offset works for θ = π/4, the scenario in (1.5). Plot the resulting reconstructions ũc and
ũs , and compare them with the original I and Q components. Can you read off the corresponding
bits bc [n] and bs [n] from eyeballing the plots for ũc and ũs ?

Software Lab 2.2: Modeling a lamppost based broadband network

The background for this lab is provided in Section 2.9, which discusses wireless channel modeling.
This material should be reviewed prior to doing the lab.

Direct path (200 m)

10 m 10 m

0 200 m
Lamppost to Lamppost Link (direct path + ground reflection)

Lamppost 1 Lamppost 2

10 m Car antenna 10 m
(2 m height)

0 D 200 m

Figure 2.40: Links in a lamppost-based network.

Consider a lamppost-based network supplying broadband access using unlicensed spectrum at 5

GHz. Figure 2.40 shows two kinds of links: lamppost-to-lamppost for backhaul, and lamppost-
to-mobile for access, where we show nominal values of antenna heights and distances. We explore
simple channel models for each case, consisting only of the direct path and the ground reflection.
For simplicity, assume throughout that A1 = 1, φ1 = 0 for the direct path, and A2 = 0.98, φ2 = π
for the ground reflection (we assume a phase shift of π for the reflected ray even though it may
not be at grazing incidence, especially for the lamppost to mobile link).
(2.1) Find the delay spread and coherence bandwidth for the lamppost-to-lamppost link. If the
message signal has 20 MHz bandwidth, is it “narrowband” with respect to this channel?
(2.2) Repeat item (2.1) for the lamppost-to-car link when the car is 100 m away from each
lamppost.
Fading and diversity for the backhaul link
First, let us explore the sensitivity of the lamppost to lamppost link to variations in range and

85
height. Fix the height of the transmitter on lamppost 1 at 10 m. Vary the height of the receiver
on lamppost 2 from 9.5 to 10.5 m.
(2.3) Letting hnom denote the nominal channel gain between two lampposts if you only consider
the direct path and h the net complex gain including the reflected path, plot the normalized
power gain in dB, 20 log10 |h|h|nom
, as a function of the variation in the receiver height. Comment
on the sensitivity of channel quality to variations in the receiver height.
(2.4) Modeling the variations in receiver height as coming from a uniform distribution over
[9.5, 10.5], find the probability that the normalized power gain is smaller than -20 dB? (i.e., that
we have a fade in signal power of 20 dB or worse).
(2.5) Now, suppose that the transmitter has two antennas, vertically spaced by 25 cm, with
the lower one at a height of 10 m. Let h1 and h2 denote the channels from the two antennas
to the receiver. Let hnom be defined as in item (2.3). Plot the normalized power gains in dB,
20 log10 |h|h i|
nom |
, i = 1, 2. Comment on whether or not both gains dip or peak at the same time.
(2.6) Plot 20 log10 max(|h 1 |,|h2 |)
|hnom |
, which is the normalized power gain you would get if you switched
to the transmit antenna which has the better channel. This strategy is termed switched diversity.
(2.7) Find the probability that the normalized power gain of the switched diversity scheme is
smaller than -20 dB.
(2.8) Comment on whether, and to what extent, diversity helped in combating fading.
Fading on the access link
Consider the access channel from lamppost 1 to the car. Let hnom (D) denote the nominal channel
gain from the lamppost to the car, ignoring the ground reflection. Taking into account the ground
reflection, let the channel gain be denoted as h(D). Here D is the distance of the car from the
bottom of lamppost 1, as shown in Figure 2.40.
(2.9) Plot |hnom | and |h| as a function of D on a dB scale (an amplitude α is expressed on the dB
scale as 20 log10 α). Comment on the “long-term” variation due to range, and the “short-term”
variation due to multipath fading.

86
Chapter 3

Analog Communication Techniques

Modulation is the process of encoding information into a signal that can be transmitted (or
recorded) over a channel of interest. In analog modulation, a baseband message signal, such
as speech, audio or video, is directly transformed into a signal that can be transmitted over
a designated channel, typically a passband radio frequency (RF) channel. Digital modulation
differs from this only in the following additional step: bits are encoded into baseband message
signals, which are then transformed into passband signals to be transmitted. Thus, despite
the relentless transition from digital to analog modulation, many of the techniques developed for
analog communication systems remain important for the digital communication systems designer,
and our goal in this chapter is to study an important subset of these techniques, using legacy
analog communication systems as examples to reinforce concepts.
From Chapter 2, we know that passband signals carry information in their complex envelope,
and that the complex envelope can be represented either in terms of I and Q components, or in
terms of envelope and phase. We study two broad classes of techniques: amplitude modula-
tion, in which the analog message signal appears directly in the I and/or Q components; and
angle modulation, in which the analog message signal appears directly in the phase or in the
instantaneous frequency (i.e., in the derivative of the phase), of the transmitted signal. Examples
of analog communication in space include AM radio, FM radio, and broadcast television, as well
as a variety of specialized radios. Examples of analog communication in time (i.e., for storage)
include audiocassettes and VHS videotapes.
The analog-centric techniques covered in this chapter include envelope detection, superhetero-
dyne reception, limiter discriminators, and phase locked loops. At a high level, these techniques
tell us how to go from baseband message signals to passband transmitted signals, and back
from passband received signals to baseband message signals. For analog communication, this
is enough, since we consider continuous time message signals which are directly transformed to
passband through amplitude or angle modulation. For digital communication, we need to also
figure out how to decode the encoded bits from the received passband signal, typically after down-
conversion to baseband; this is a subject discussed in later chapters. However, between encoding
at the transmitter and decoding at the receiver, a number of analog communication techniques
are relevant: for example, we need to decide between direct and superheterodyne architectures
for upconversion and downconversion, and tailor our frequency planning appropriately; we may
use a PLL to synthesize the local oscillator frequencies at the transmitter and receiver; and
the basic techniques for mapping baseband signals to passband remain the same (amplitude
and/or angle modulation). In addition, while many classical analog processing functionalities
are replaced by digital signal processing in modern digital communication transceivers, when we
push the limits of digital communication systems, in terms of lowering power consumption or
increasing data rates, it is often necessary to fall back on analog-centric, or hybrid digital-analog,
techniques. This is because the analog-to-digital conversion required for digital transceiver im-

87
plementations may often be too costly or power-hungry for ultra high-speed, or ultra low-power,
implementations.
Chapter Plan: After a quick discussion of terminology and notation in Section 3.1, we discuss
various forms of amplitude modulation in Section 3.2, including bandwidth requirements and the
tradeoffs between power efficiency and simplicity of demodulation. We discuss angle modulation
in Section 3.3, including the relation between phase and frequency modulation, the bandwidth
of angle modulated signals, and simple suboptimal demodulation strategies. The superhetero-
dyne up/downconversion architecture is discussed in Section 3.4, and the design considerations
illustrated via the example of analog AM radio. The phase locked loop (PLL) is discussed in
Section 3.5, including discussion of applications such as frequency synthesis and FM demodu-
lation, linearized modeling and analysis, and a glimpse of the insights provided by nonlinear
models. Finally, we discuss some legacy analog communication systems in Section 3.6, mainly to
highlight some of the creative design choices that were made in times when sophisticated digital
signal processing techniques were not available. This last section can be skipped if the reader’s
interest is limited to learning analog-centric techniques for digital communication system design.

3.1 Terminology and notation

Message Signal: In the remainder of this chapter, the analog baseband message signal is
denoted by m(t). Depending on convenience of exposition, we shall think of this message as
either finite power or finite energy. In practice, any message we would encounter in practice
would have finite energy when we consider a finite time interval. However, when modeling
transmissions over long time intervals, it is useful to think of messages as finite power signals
spanning an infinite time interval. On the other hand, when discussing the effect of the message
spectrum on the spectrum of the transmitted signal, it may be convenient to consider a finite
energy message signal. Since we consider physical message signals, the time domain signal is real-
valued, so that its Fourier transform (defined for a finite energy signal) is conjugate symmetric:
M(f ) = M ∗ (−f ). For a finite power (infinite energy) message, recall from Chapter 2 that the
power is defined as a time average in the limit of an infinite observation interval, as follows:
To
1
Z
m2 = lim m2 (t)dt
To →∞ To 0

Similarly, the DC value is defined as

To
1
Z
m = lim m(t)dt
To →∞ To 0

We typically assume that the DC value of the message is zero: m = 0.

A simple example, shown in Figure 3.1, that we shall use often is a finite-power sinusoidal
message signal, m(t) = Am cos 2πfm t, whose spectrum consists of impulses at ±fm : M(f ) =
Am
2
(δ(f − fm ) + δ(f + fm )). For this message, m = 0 and m2 = A2m /2.
Transmitted Signal: When the signal transmitted over the channel is a passband signal, it
can be written as (see Chapter 2)

up (t) = uc (t) cos(2πfc t) − us (t) sin(2πfc t) = e(t) cos(2πfc t + θ(t))

where fc is a carrier frequency, uc (t) is the I component, us (t) is the Q component, e(t) ≥ 0 is the
envelope, and θ(t) is the phase. Modulation consist of encoding the message in uc (t) and us (t), or
equivalently, in e(t) and θ(t). In most of the analog amplitude modulation schemes considered,

88
 1

0.8

0.6

0.4
M(f)
0.2
A m /2

m(t)/Am
0

−0.2

−0.4

−0.6

−0.8

−1 f
0 0.2 0.4 0.6 0.8 1
fm t
1.2 1.4 1.6 1.8 2

−fm fm

(a) Sinusoidal message waveform (b) Sinusoidal message spectrum

Figure 3.1: Sinusoidal message and its spectrum

the message modulates the I component (with the Q component occasionally playing a “sup-
porting role”) as discussed in Section 3.2. The exception is quadrature amplitude modulation, in
which both I and Q components carry separate messages. In phase and frequency modulation,
or angle modulation, the message directly modulates the phase θ(t) or its derivative, keeping the
envelope e(t) unchanged.

3.2 Amplitude Modulation

We now discuss a number of variants of amplitude modulation, in which the baseband message
signal modulates the amplitude of a sinusoidal carrier whose frequency falls in the passband over
which we wish to communicate.

3.2.1 Double Sideband (DSB) Suppressed Carrier (SC)

Here, the message m modulates the I component of the passband transmitted signal u as follows:
uDSB (t) = Am(t) cos(2πfc t) (3.1)
Taking Fourier transforms, we have
A
UDSB (f ) = (M(f − fc ) + M(f + fc )) (3.2)
2

The time domain and frequency domain DSB signals for a sinusoidal message are shown in Figure
3.2.
As another example, consider the finite-energy message whose spectrum is shown in Figure 3.3.
Since the time domain message m(t) is real-valued, its spectrum exhibits conjugate symmetry
(we have chosen a complex-valued message spectrum to emphasize the latter property). The
message bandwidth is denoted by B. The bandwidth of the DSB-SC signal is 2B, which is twice
the message bandwidth. This indicates that we are being redundant in our use of spectrum. To
see this, consider the upper sideband (USB) and lower sideband (LSB) depicted in Figure 3.4.
The shape of the signal in the USB (i.e., Up (f ) for fc < f ≤ fc + B) is the same as that of the
message for positive frequencies (i.e., M(f ), f > 0). The shape of the signal in the LSB (i.e.,
Up (f ) for fc − B ≤ f < fc ) is the same as that of the message for negative frequencies (i.e.,
M(f ), f < 0). Since m(t) is real-valued, we have M(−f ) = M ∗ (f ), so that we can reconstruct
the message if we know its content at either positive or negative frequencies. Thus, the USB and

89
 1

0.8

0.6

0.4

0.2
UDSB (f)
0

−0.2 A A m/4
−0.4

−0.6

−0.8
message waveform
DSB waveform
−1 −fc −fm −fc + fm fc − fm fc + fm f
0 2 4 6 8 10 12 14 16 18 20

(a) DSB time domain waveform (b) DSB spectrum

Figure 3.2: DSB-SC signal in the time and frequency domains for the sinusoidal message m(t) =
Am cos 2πfm t of Figure 3.1.

Re(M(f))
a

f
0

Im(M(f))

0
f

Figure 3.3: Example message spectrum.

90
 Re(UDSB (f)) 2B

A a/2

f
−fc fc

Upper Lower Lower Upper

Sideband Sideband Sideband Sideband
Im(UDSB (f))
A b/2

f
−fc fc

Figure 3.4: The spectrum of the passband DSB-SC signal for the example message in Figure 3.3.

LSB of u(t) each contain enough information to reconstruct the message. The term DSB refers to
the fact that we are sending both sidebands. Doing this, of course, is wasteful of spectrum. This
motivates single sideband (SSB) and vestigial sideband (VSB) modulation, which are discussed
a little later.
The term suppressed carrier is employed because, for a message with no DC component, we
see from (3.2) that the transmitted signal does not have a discrete component at the carrier
frequency (i.e., Up (f ) does not have impulses at ±fc ).

Passband received Lowpass Estimated

signal Filter message

2cos 2 π fc t

Figure 3.5: Coherent demodulation for AM.

Demodulation of DSB-SC: Since the message is contained in the I component, demodulation

consists of extracting the I component of the received signal, which we know how to do from
Chapter 2: multiply the received signal with the cosine of the carrier, and pass it through a low
pass filter. Ignoring noise, the received signal is given by
yp (t) = Am(t) cos(2πfc t + θr ) (3.3)

where θr is the phase of the received carrier relative to the local copy of the carrier produced
by the receiver’s local oscillator (LO), and A is the received amplitude, taking into account the
propagation channel from the transmitter to the receiver. The demodulator is shown in Figure
3.5. In order for this demodulator to work well, we must have θr as close to zero as possible;
that is, the carrier produced by the LO must be coherent with the received carrier. To see the
effect of phase mismatch, let us compute the demodulator output for arbitrary θr . Using the
trigonometric identity 2 cos θ1 cos θ2 = cos(θ1 − θ2 ) + cos(θ1 + θ2 ), we have

2yp (t) cos(2πfc t) = Am(t) cos(2πfc t + θr ) cos(2πfc t) = Am(t) cos θr + Am(t) cos(4πfc t + θr )

91
We recognize the second term on the extreme right-hand side as being a passband signal at 2fc
(since it is a baseband message multiplied by a carrier whose frequency exceeds the message
bandwidth). It is therefore rejected by the lowpass filter. The first term is a baseband signal
proportional to the message, which appears unchanged at the output of the LPF (except possibly
for scaling), as long as the LPF response has been designed to be flat over the message bandwidth.
The output of the demodulator is therefore given by

m̂(t) = Am(t) cos θr (3.4)

We can also infer this using the complex baseband representation, which is what we prefer to
employ instead of unwieldy trigonometric identities. The coherent demodulator in Figure 3.5
extracts the I component relative to the receiver’s LO. The received signal can be written as

yp (t) = Am(t) cos(2πfc t + θr ) = Re Am(t)ej(2πfc t+θr ) = Re Am(t)ejθr ej2πfc t

from which we can read off the complex envelope y(t) = Am(t)ejθr . The real part yc (t) =
Am(t) cos θr is the I component extracted by the demodulator.
The demodulator output (3.4) is proportional to the message, which is what we want, but
the proportionality constant varies with the phase of the received carrier relative to the LO.
In particular, the signal gets significantly attenuated as the phase mismatch increases, and gets
completely wiped out for θr = π2 . Note that, if the carrier frequency of the LO is not synchronized
with that of the received carrier (say with frequency offset ∆f ), then θr (t) = 2π∆f t+φ is a time-
varying phase that takes all values in [0, 2π), which leads to time-varying signal degradation in
amplitude, as well as unwanted sign changes. Thus, for coherent demodulation to be successful,
we must drive ∆f to zero, and make φ as small as possible; that is, we must synchronize to
the received carrier. One possible approach to use feedback-based techniques such as the phase
locked loop, discussed later in this chapter.

3.2.2 Conventional AM
In conventional AM, we add a large carrier component to a DSB-SC signal, so that the passband
transmitted signal is of the form:

uAM (t) = Am(t) cos(2πfc t) + Ac cos(2πfc t) (3.5)

Taking the Fourier transform, we have

A Ac
UAM (f ) = (M(f − fc ) + M(f + fc )) + (δ(f − fc ) + δ(f + fc ))
2 2
which means that, in addition to the USB and LSB due to the message modulation, we also have
impulses at ±fc due to the unmodulated carrier. Figure 3.6 shows the resulting spectrum.
The key concept behind conventional AM is that, by making Ac large enough, the message can be
demodulated using a simple envelope detector. Large Ac corresponds to expending transmitter
power on sending an unmodulated carrier which carries no message information, in order to
simplify the receiver. This tradeoff makes sense in a broadcast context, where one powerful
transmitter may be sending information to a large number of low-cost receivers, and is the
design approach that has been adopted for broadcast AM radio. A more detailed discussion
follows.
The envelope of the AM signal in (3.5) is given by

e(t) = |Am(t) + Ac |

92
 Ac /2 Re(UAM (f)) Ac /2

A a/2

f
−fc fc

2B

Im(UAM (f))
A b/2

f
−fc fc

Figure 3.6: The spectrum of a conventional AM signal for the example message in Figure 3.3.

If the term inside the magnitude operation is always nonnegative, we have e(t) = Am(t) + Ac .
In this case, we can read off the message signal directly from the envelope, using AC coupling to
get rid of the DC offset due to the second term. For this to happen, we must have
A m(t) + Ac ≥ 0 for all t ⇐⇒ A mint m(t) + Ac > 0 (3.6)
Let mint m(t) = −M0 , where M0 = |mint m(t)|. (Note that the minimum value of the message
must be negative if the message has zero DC value.) Equation (3.6) reduces to −AM0 + Ac ≥ 0,
or Ac ≥ AM0 . Let us define the modulation index amod as the ratio of the size of the biggest
negative incursion due to the message term to the size of the unmodulated carrier term:
AM0 A|mint m(t)|
amod = =
Ac Ac
The condition (3.6) for accurately recovering the message using envelope detection can now be
rewritten as
amod ≤ 1 (3.7)
It is also convenient to define a normalized version of the message as follows:
m(t) m(t)
mn (t) = = (3.8)
M0 |mint m(t)|
which satisfies
mint m(t)
mint mn (t) = = −1
M0
It is easy to see that the AM signal (3.5) can be rewritten as
uAM (t) = Ac (1 + amod mn (t)) cos(2πfc t) (3.9)
which clearly brings out the role of modulation index in ensuring that envelope detection works.
Figure 3.7 illustrates the impact of modulation index on the viability of envelope detection, where
the message signal is the sinusoidal message in Figure 3.1. For amod = 0.5 and amod = 1, we see
that envelope equals a scaled and DC-shifted version of the message. For amod = 1.5, we see that
the envelope no longer follows the shape of the message.
Demodulation of Conventional AM: Ignoring noise, the received signal is given by
yp (t) = B (1 + amod mn (t)) cos(2πfc t + θr ) (3.10)

93
 1.5

0.5

−0.5

−1

envelope
AM waveform
−1.5
0 2 4 6 8 10 12 14 16 18 20

(a) Modulation Index amod = 0.5

2

1.5

0.5

−0.5

−1

−1.5
envelope
AM waveform
−2
0 2 4 6 8 10 12 14 16 18 20

(b) Modulation Index amod = 1.0

2.5

1.5

0.5

−0.5

−1

−1.5

−2
envelope
AM waveform
−2.5
0 2 4 6 8 10 12 14 16 18 20

(c) Modulation Index amod = 1.5

Figure 3.7: Time domain AM waveforms for a sinusoidal message. The envelope no longer follows
the message for modulation index larger than one.

94
 + +
Passband R C Envelope detector
vin (t) vout(t)
AM signal output
− −

Figure 3.8: Envelope detector demodulation of AM. The envelope detector output is typically
passed through a DC blocking capacitance (not shown) to eliminate the DC offset due to the
carrier component of the AM signal.

v1 exp(−(t− t1)/RC)

v1 v2 exp(−(t− t2)/RC)
v2 Envelope detector output vout (t)

Envelope

t
t1 t2

Envelope detector input vin (t)

Figure 3.9: The relation between the envelope detector output vout (t) (shown in bold) and input
vin (t) (shown as dashed line). The output closely follows the envelope (shown as dotted line).

95
where θr is a phase offset which is unknown a priori, if we do not perform carrier synchronization.
However, as long as amod ≤ 1, we can recover the message without knowing θr using envelope
detection, since the envelope is still just a scaled and DC-shifted version of the message. Of
course, the message can also be recovered by coherent detection, since the I component of the
received carrier equals a scaled and DC-shifted version of the message. However, by doing enve-
lope detection instead, we can avoid carrier synchronization, thus reducing receiver complexity
drastically. An envelope detector is shown in Figure 3.8, and an example (where the envelope
is a straight line) showing how it works is depicted in Figure 3.9. The diode (we assume that it
is ideal) conducts in only the forward direction, when the input voltage vin (t) of the passband
signal is larger than the output voltage vout (t) across the RC filter. When this happens, the
output voltage becomes equal to the input voltage instantaneously (under the idealization that
the diode has zero resistance). In this regime, we have vout (t) = vin (t). When the input voltage is
smaller than the output voltage, the diode does not conduct, and the capacitor starts discharging
through the resistor with time constant RC. As shown in Figure 3.9, in this regime, starting at
time t1 , we have v(t) = v1 e−(t−t1 )/RC , where v1 = v(t1 ), as shown in Figure 3.9.
Roughly speaking, the capacitor gets charged at each carrier peak, and discharges between peaks.
The time interval between successive charging episodes is therefore approximately equal to f1c ,
the time between successive carrier peaks. The factor by which the output voltage is reduced
during this period due to capacitor discharge is exp (−1/(fc RC)). This must be close to one in
order for the voltage to follow the envelope, rather than the variations in the sinusoidal carrier.
That is, we must have fc RC ≫ 1. On the other hand, the decay in the envelope detector output
must be fast enough (i.e., the RC time constant must be small enough) so that it can follow
changes in the envelope. Since the time constant for envelope variations is inversely proportional
to the message bandwidth B, we must have RC ≪ 1/B. Combining these two conditions for
envelope detection to work well, we have
1 1
≪ RC ≪ (3.11)
fc B

This of course requires that fc ≫ B (carrier frequency much larger than message bandwidth),
which is typically satisfied in practice. For example, the carrier frequencies in broadcast AM
radio are over 500 KHz, whereas the message bandwidth is limited to 5 KHz. Applying (3.11),
the RC time constant for an envelope detector should be chosen so that

2 µs ≪ RC ≪ 200 µs

In this case, a good choice of parameters would be RC = 20µs, for example, with R = 50 ohms,
and C = 400 nanofarads.
Power efficiency of conventional AM: The price we pay for the receiver simplicity of conven-
tional AM is power inefficiency: in (3.5) the unmodulated carrier Ac cos(2πfc t) is not carrying
any information regarding the message. We now compute the power efficiency ηAM , which is
defined as the ratio of the transmitted power due to the message-bearing term Am(t) cos(2πfc t)
to the total power of uAM (t). In order to express the result in terms of the modulation index,
let us use the expression (3.9).

A2c A2
u2AM (t) = A2c (1 + amod mn (t))2 cos2 (2πfc t) = (1 + amod mn (t))2 + c (1 + amod mn (t))2 cos(4πfc t)
2 2
The second term on the right-hand side is the DC value of a passband signal at 2fc , which is
zero. Expanding out the first term, we have

A2c   A2  
u2AM (t) = 1 + a2mod m2n + 2amod mn = c 1 + a2mod m2n (3.12)
2 2

96
assuming that the message has zero DC value. The power of the message-bearing term can be
similarly computed as
A2
(Ac amod mn (t))2 cos2 (2πfc t) = c a2mod m2n
2
so that the power efficiency is given by

a2mod m2n
ηAM = (3.13)
1 + a2mod m2n

Noting that mn is normalized so that its most negative value is −1, for messages which have
comparable positive and negative excursions around zero, we expect |mn (t)| ≤ 1, and hence
average power m2n ≤ 1 (typical values are much smaller than one). Since amod ≤ 1 for envelope
detection to work, the power efficiency of conventional AM is at best 50%. For a sinusoidal
message, for example, it is easy to see that m2n = 1/2, so that the power efficiency is at most 33%.
For speech signals, which have significantly higher peak-to-average ratio, the power efficiency is
even smaller.

Example 3.2.1 (AM power efficiency computation): The message m(t) = 2 sin 2000πt −
3 cos 4000πt is used in an AM system with a modulation index of 70% and carrier frequency
of 580 KHz. What is the power efficiency? If the net transmitted power is 10 watts, find the
magnitude spectrum of the transmitted signal.
We need to find M0 = |mint m(t)| in order to determine the normalized form mn (t) = m(t)/M0 .
To simplify notation, let x = 2000πt, and minimize g(x) = 2 sin x − 3 cos 2x. Since g is periodic
with period 2π, we can minimize it numerically over a period. However, we can perform the
minimization analytically in this case. Differentiating g, we obtain

g ′(x) = 2 cos x + 6 sin 2x = 0

This gives
2 cos x + 12 sin x cos x = 2 cos x(1 + 6 sin x) = 0
There are two solutions cos x = 0 and sin x = − 61 . The first solution gives cos 2x = 2 cos2 x − 1 =
−1 and sin x = ±1, which gives g(x) = 1, 5. The second solution gives cos 2x = 1 − 2 sin2 x =
1 − 2/36 = 17/18, which gives g(x) = 2(−1/6) − 3(17/18) = −19/6. We therefore obtain

M0 = |mint m(t)| = 19/6

This gives
m(t) 12 18
mn (t) = = sin 10πt − cos 20πt
M0 19 19
This gives
m2n = (12/19)2(1/2) + (18/19)2(1/2) = 0.65
Substituting in (3.13), setting amod = 0.7, we obtain a power efficiency ηAM = 0.24, or 24%.
To figure out the spectrum of the transmitted signal, we must find Ac in the formula (3.9). The
power of the transmitted signal is given by (3.12) to be

A2c   A2
1 + a2mod m2n = c 1 + (0.72 )(0.65)


10 =
2 2
which yields Ac ≈ 3.9. The overall AM signal is given by

uAM (t) = Ac (1 + amod mn (t)) cos 2πfc t = Ac (1 + a1 sin 2πf1 t + a2 cos 4πf1 t) cos 2πfc t

97
where a1 = 0.7(12/19) = 0.44, a2 = 0.7(−18/19) = −0.66, f1 = 1 KHz and fc = 580KHz. The
magnitude spectrum is given by

|UAM (f )| = Ac /2 (δ(f − fc ) + δ(f + fc ))

+ Ac |a1 |/4 (δ(f − fc − f1 ) + δ(f − fc + f1 ) + δ(f + fc + f1 ) + δ(f + fc − f1 ))
+ Ac |a2 |/4 (δ(f − fc − 2f1 ) + δ(f − fc + 2f1 ) + δ(f + fc + 2f1 ) + δ(f + fc − 2f1 ))

with numerical values shown in Figure 3.10.

|UAM (f)|
1.95 1.95

0.644 0.644 0.644 0.644

0.429 0.429 0.429 0.429

~
~

~
~
−578 −579 −580 −581 −582 578 579 580 581 582 f (KHz)

Figure 3.10: Magnitude spectrum for the AM waveform in Example 3.2.1.

3.2.3 Single Sideband Modulation (SSB)

In SSB modulation, we send either the upper sideband or the lower sideband of a DSB-SC signal.
For the running example, the spectra of the passband USB and LSB signals are shown in Figure
3.11.
From our discussion of DSB-SC, we know that each sideband provides enough information to
reconstruct the message. But how do we physically reconstruct the message from an SSB signal?
To see this, consider the USB signal depicted in Figure 3.11(a). We can reconstruct the baseband
message if we can move the component near +fc to the left by fc , and the component near −fc
to the right by fc ; that is, if we move in the passband components towards the origin. These
two frequency translations can be accomplished by multiplying the USB signal by 2 cos 2πfc t =
ej2πfc t + e−j2πfc t , as shown in Figure 3.5, which creates the desired message signal at baseband,
as well as undesired frequency components at ±2fc which can be rejected by a lowpass filter. It
can be checked that the same argument applies to LSB signals as well.
It follows from the preceding discussion that SSB signals can be demodulated in exactly the
same fashion as DSB-SC, using the coherent demodulator depicted in Figure 3.5. Since this
demodulator simply extracts the I component of the passband signal, the I component of the
SSB signal must be the message. In order to understand the structure of an SSB signal, it remains
to identify the Q component. This is most easily done by considering the complex envelope of
the passband transmitted signal. Consider again the example USB signal in Figure 3.11(a). The
spectrum U(f ) of its complex envelope relative to fc is shown in Figure 3.12. Now, the spectra
of I and Q components can be inferred as follows:

U(f ) + U ∗ (−f ) U(f ) − U ∗ (−f )

Uc (f ) = , Us (f ) =
2 2j

Applying these equations, we get I and Q components as shown in Figure 3.13.

98
 Re(UUSB (f))

A a/2

f
−fc fc

B
Im(UUSB (f))
A b/2

f
−fc fc

(a) Upper Sideband Signaling

Re(ULSB (f))

A a/2

f
−fc fc

Im(ULSB (f))
A b/2

f
−fc fc

(b) Lower Sideband Signaling

Figure 3.11: Spectra for SSB signaling for the example message in Figure 3.3.

Re(U(f)) Im(U(f))

Aa

f f

−Ab

Figure 3.12: Complex envelope for the USB signal in Figure 3.11(a).

99
 I component Q component

Re(Uc (f))
Re(Us (f))
Aa/2

f f

−Ab/2

Im(Uc (f))

Ab/2 Im(Us (f))

Aa/2
f

Figure 3.13: I and Q components for the USB signal in Figure 3.11(a).

Thus, up to scaling, the I component Uc (f ) = M(f ), and the Q component is a transformation

of the message given by

−jM(f ), f > 0
Us (f ) = = M(f )(−jsgn(f )) (3.14)
jM(f ), f <0

That is, the Q component is a filtered version of the message, where the filter transfer function
is H(f ) = −jsgn(f ). This transformation is given a special name, the Hilbert transform.
Hilbert transform: The Hilbert transform of a signal x(t) is denoted by x̌(t), and is specified
in the frequency domain as
X̌(f ) = (−jsgn(f )) X(f )
This corresponds to passing u through a filter with transfer function
1
H(f ) = −jsgn(f ) ↔ h(t) =
πt
where the derivation of the impulse response is left as an exercise.
Figure 3.14 shows the spectrum of the Hilbert transform of the example message in Figure 3.3.
We see that it is the same (upto scaling) as the Q component of the USB signal, shown in Figure
3.13.
Physical interpretation of the Hilbert transform: If x(t) is real-valued, then so is its
Hilbert transform x̌(t). Thus, the Fourier transforms X(f ) and X̌(f ) must both satisfy conjugate
symmetry, and we only need to discuss what happens at positive frequencies. For f > 0, we have
X̌(f ) = −jsgn(f )X(f ) = −jX(f ) = e−jπ/2 X(f ). That is, the Hilbert transform simply imposes
a π/2 phase lag at all (positive) frequencies, leaving the magnitude of the Fourier transform
unchanged.

Example 3.2.2 (Hilbert transform of a sinusoid): Based on the preceding argument, a

sinusoid s(t) = cos(2πf0 t + φ) has Hilbert transform š(t) = cos(2πf0 t + φ − π2 ) = sin(2πf0 t + φ).

100
 Re(M(f))

f
0

−b

Im(M(f))
a

0
f

Figure 3.14: Spectrum of the Hilbert transform of the example message in Figure 3.3.

We can also do this the hard way, as follows:

s(t) = cos(2πf0 t + φ) = 12 ej(2πf0 t+φ) + e−j(2πf0 t+φ)

↔ S(f ) = 21 ejφ δ(f − f0 ) + e−jφ δ(f + f0 )

Thus,
Š(f ) = −jsgn(f )S(f ) = 21 ejφ (−j)δ(f − f0 ) +
e−jφ (j)δ(f + f0 )

↔ š(t) = 12 ejφ (−j)ej2πf0 t + e−jφ (j)ej2πf0 t

which simplifies to
1 j(2πf0 t+φ)
− e−j(2πf0 t+φ) = sin(2πf0 t + φ)


š(t) = e
2j

Equation (3.14) shows that the Q component of the USB signal is m̌(t), the Hilbert transform
of the message. Thus, the passband USB signal can be written as
uU SB (t) = m(t) cos(2πfc t) − m̌(t) sin(2πfc t) (3.15)
Similarly, we can show that the Q component of an LSB signal is −m̌(t), so that the passband
LSB signal is given by
uLSB (t) = m(t) cos(2πfc t) + m̌(t) sin(2πfc t) (3.16)

SSB modulation: Conceptually, an SSB signal can be generated by filtering out one of the
sidebands of a DSB-SC signal. However, it is difficult to implement the required sharp cutoff
at fc , especially if we wish to preserve the information contained at the boundary of the two
sidebands, which corresponds to the message information near DC. Thus, an implementation of
SSB based on sharp bandpass filters runs into trouble when the message has significant frequency
content near DC. The representations in (3.15) and (3.16) provide an alternative approach to
generating SSB signals, as shown in Figure 3.15. We have emphasized the role of 90◦ phase lags
in generating the I and Q components, as well as the LO signals used for upconversion.

Example 3.2.3 (SSB waveforms for a sinusoidal message): For a sinusoidal message
m(t) = cos 2πfm t, we have m̌(t) = sin 2πfm t from Example 3.2.2. Consider the DSB signal
uDSB (t) = 2 cos 2πfm t cos 2πfc t

101
 Message signal
m(t)

Hilbert transform 2cos 2 π fc t

(90 deg phase lag Local 90 deg USB/LSB signal
over message band) Oscillator phase lag
2sin 2 π fc t
−/+

m(t)

Figure 3.15: SSB modulation using the Hilbert transform of the message.

UDSB (f)

1/2

f
~
~

~
~

−fc −fm −fc + fm fc − fm fc + fm

UUSB (f)

1/2

f
~
~

~
~

−fc −fm fc + fm

ULSB (f)

1/2

f
~
~

~
~

−fc + fm fc − fm

Figure 3.16: DSB and SSB spectra for a sinusoidal message.

102
where we have normalized the signal power to one: u2DSB = 1. The DSB, USB and SSB spectrum
are shown in Figure 3.16. From the SSB spectra shown, we can immediately write down the
following time domain expressions:

uU SB (t) = cos 2π(fc + fm )t = cos 2πfm t cos 2πfc t − sin 2πfm t sin 2πfc t

uLSB (t) = cos 2π(fc − fm )t = cos 2πfm t cos 2πfc t + sin 2πfm t sin 2πfc t
The preceding equations are consistent with (3.15) and (3.16). For both the USB and LSB
signals, the I component equals the message: uc (t) = m(t) = cos 2πfm t. The Q component
for the USB signal is us (t) = m̌(t) = sin 2πfm t, and the Q component for the LSB signal is
us (t) = −m̌(t) = − sin 2πfm t.

SSB demodulation: We know now that the message can be recovered from an SSB signal
by extracting its I component using a coherent demodulator as in Figure 3.5. The difficulty of
coherent demodulation lies in the requirement for carrier synchronization, and we have discussed
the adverse impact of imperfect synchronization for DSB-SC signals. We now show that the
performance degradation is even more significant for SSB signals. Consider a USB received
signal of the form (ignoring scale factors):

yp (t) = m(t) cos(2πfc t + θr ) − m̌(t) sin(2πfc t + θr ) (3.17)

where θr is the phase offset with respect to the receiver LO. The complex envelope with respect
to the receiver LO is given by

y(t) = (m(t) + j m̌(t)) ejθr = (m(t) + j m̌(t)) (cos θr + j sin θr )

Taking the real part, we obtain that the I component extracted by the coherent demodulator is

yc (t) = m(t) cos θr − m̌(t) sin θr

Thus, as the phase error θr increases, not only do we get an attenuation in the first term corre-
sponding to the desired message (as in DSB), but we also get interference due to the second term
from the Hilbert transform of the message. Thus, for coherent demodulation, accurate carrier
synchronization is even more crucial for SSB than for DSB.
Noncoherent demodulation is also possible for SSB if we add a strong carrier term, as in conven-
tional AM. Specifically, for a received signal given by

yp (t) = (A + m(t)) cos(2πfc t + θr ) ± m̌(t) sin(2πfc t + θr )

the envelope is given by

p
e(t) = (A + m(t))2 + m̌2 (t) ≈ A + m(t) (3.18)

if |A + m(t)| ≫ |m̌(t)|. Subject to the approximation in (3.18), an envelope detector works just
as in conventional AM.

3.2.4 Vestigial Sideband (VSB) Modulation

VSB is similar to SSB, in that it also tries to reduce the transmitted bandwidth relative to DSB,
and the transmitted signal is a filtered version of the DSB signal. The idea is to mainly transmit
one of the two sidebands, but to leave a vestige of the other sideband in order to ease the filtering
requirements. The passband filter used to shape the DSB signal in this fashion is chosen so that

103
 Hp (f)

M(f+fc ) M(f−fc )

−fc fc f

H p (f−fc )+ H p (f+ fc ) constant over message band

M(f)

Figure 3.17: Relevant passband and baseband spectra for VSB.

the I component of the transmitted signal equals the message. To see this, consider the DSB-SC
signal
2m(t) cos 2πfc t ↔ M(f − fc ) + M(f + fc )
This is filtered by a passband VSB filter with transfer function Hp (f ), as shown in Figure 3.17,
to obtain the transmitted signal with spectrum
UV SB (f ) = Hp (f ) (M(f − fc ) + M(f + fc )) (3.19)
A coherent demodulator extracting the I component passes 2uV SB (t) cos 2πfc t through a lowpass
filter. But
2uV SB (t) cos 2πfc t ↔ UV SB (f − f c) + UV SB (f + fc )
which equals (substituting from (3.19),
Hp (f − fc ) (M(f − 2fc ) + M(f )) + Hp (f + fc ) (M(f ) + M(f + 2fc )) (3.20)

The 2fc term, Hp (f − fc )M(f − 2fc ) + Hp (f + fc )M(f + 2fc ), is filtered out by the lowpass filter.
The output of the LPF are the lowpass terms in (3.20), which equal the I component, and are
given by
M(f ) (Hp (f − fc ) + Hp (f + fc ))

In order for this to equal (a scaled version of) the desired message, we must have
Hp (f + fc ) + Hp (f − fc ) = constant , |f | < W (3.21)
as shown in the example in Figure 3.17. To understand what this implies about the structure
of the passband VSB filter, note that the filter impulse response can be written as hp (t) =

104
hc (t) cos 2πfc t − hs (t) sin 2πfc t, where hc (t) is obtained by passing 2hp (t) cos(2πfc t) through a
lowpass filter. But 2hp (t) cos(2πfc t) ↔ Hp (f − fc ) + Hp (f + fc ). Thus, the Fourier transform
involved in (3.21) is precisely the lowpass restriction of 2hp (t) cos(2πfc t), i.e., it is Hc (f ). Thus,
the correct demodulation condition for VSB in (3.21) is equivalent to requiring that Hc (f ) be
constant over the message band. Further discussion of the structure of VSB signals is provided
via problems.
As with SSB, if we add a strong carrier component to the VSB signal, we can demodulate it
noncoherently using an envelope detector, again at the cost of some distortion from the presence
of the Q component.

3.2.5 Quadrature Amplitude Modulation

The transmitted signal in quadrature amplitude modulation (QAM) is of the form

uQAM (t) = mc (t) cos 2πfc t − ms (t) sin 2πfc t

where mc (t) and ms (t) are separate messages (unlike SSB and VSB, where the Q component is
a transformation of the message carried by the I component). In other words, a complex-valued
message m = mc (t) + jms (t) is encoded in the complex envelope of the passband transmitted
signal. QAM is extensively employed in digital communication, as we shall see in later chapters.
It is also used to carry color information in analog TV.

Lowpass ^ (t)
mc
Filter

2 cos 2 πf ct
Passband
QAM
signal −2 sin 2 πf c t

Lowpass ^ (t)
ms
Filter

Figure 3.18: Demodulation for quadrature amplitude modulation.

Demodulation is achieved using a coherent receiver which extracts both the I and Q components,
as shown in Figure 3.18. If the received signal has a phase offset θ relative to the receiver’s LO,
then we get both attenuation in the desired message and interference from the undesired message,
as follows. Ignoring noise and scale factors, the reconstructed complex baseband message is given
by
m̂(t) = m̂c (t) + j m̂s (t) = (mc (t) + jms (t))ejθ(t) = m(t)ejθ(t)
from which we conclude that

m̂c (t) = mc (t) cos θ(t) − ms (t) sin θ(t)

m̂s (t) = ms (t) cos θ(t) + mc (t) sin θ(t)

Thus, accurate carrier synchronization (θ(t) as close to zero as possible) is important for QAM
demodulation to function properly.

105
 M(f) X(f)
5/2
1 1

(spectrum for
−20 20 negative frequencies 1/2 1/2
f (KHz)
−10 10 not shown)
580 620
−1/2 −1/2 f (KHz)
590 600 610
−1/4 −1/4

(a) Message spectrum (b) AM spectrum

Figure 3.19: Spectrum of message and the corresponding AM signal in Example 3.2.4. Axes are
not to scale.

~
Y(f)

Y(f) 5
5/2

(spectrum for 1
negative frequencies 1/2
not shown) 20
620
f (KHz) 0
f (KHz)
600 610 10
−1/4 −1/2

(a) Passband output of BPF (b) Complex envelope

Figure 3.20: Passband output of bandpass filter and its complex envelope with respect to 600
KHz reference, for Example 3.2.4. Axes are not to scale.

3.2.6 Concept synthesis for AM

Here is a worked problem that synthesizes a few of the concepts we have discussed for AM.

Example 3.2.4 The signal m(t) = 2 cos 20πt − cos 40πt, where the unit of time is millisec-
onds, is amplitude modulated using a carrier frequency fc of 600 KHz. The AM signal is given
by
x(t) = 5 cos 2πfc t + m(t) cos 2πfc t
(a) Sketch the magnitude spectrum of x. What is its bandwidth?
(b) What is the modulation index?
(c) The AM signal is passed through an ideal highpass filter with cutoff frequency 595 KHz (i.e.,
the filter passes all frequencies above 595 KHz, and cuts off all frequencies below 595 KHz). Find
an explicit time domain expression for the Q component of the filter output with respect to a
600 KHz frequency reference.
Solution: (a) The message spectrum M(f ) = δ(f − 10) + δ(f + 10) − 21 δ(f − 20) − 12 δ(f + 20).
The spectrum of the AM signal is given by
5 5 1 1
X(f ) = δ(f − fc ) + δ(f + fc ) + M(f − fc ) + M(f + fc )
2 2 2 2
These spectra are sketched in Figure 3.19.
(b) From Figure 3.19, it is clear that a highpass filter with cutoff at 595 KHz selects the USB
signal plus the carrier. The passband output has spectrum as shown in Figure 3.20(a), and the
complex envelope with respect to 600 KHz is shown in Figure 3.20(b). Taking the inverse Fourier
transform, the time domain complex envelope is given by
1
ỹ(t) = 5 + ej20πt − ej40πt
2

106
We can now find the Q component to be
1
ys (t) = Im (ỹ(t)) = sin 20πt − sin 40πt
2
where t is in milliseconds. Another approach is to recognize that the Q component is the Q
component of the USB signal, which is known to be the Hilbert transform of the message.
Yet another approach
  is to find the Q component in the frequency domain using jYs (f ) =
∗
Ỹ (f ) − Ỹ (f ) /2 and then take inverse Fourier transform. In this particular example, the first
approach is probably the simplest.

3.3 Angle Modulation

We know that a passband signal can be represented as e(t) cos(2πfc t + θ(t)), where e(t) is the
envelope, and θ(t) is the phase. Let us define the instantaneous frequency offset relative to the
carrier as
1 dθ(t)
f (t) =
2π dt
In frequency modulation (FM) and phase modulation (PM), we encode information into the
phase θ(t), with the envelope remaining constant. The transmitted signal is given by

u(t) = Ac cos(2πfc t + θ(t)), Angle Modulation (information carried in θ)

For a message m(t), we have

θ(t) = kp m(t) , Phase Modulation, (3.22)

and
1 dθ(t)
= f (t) = kf m(t) , Frequency Modulation, (3.23)
2π dt
where kp , kf are constants. Integrating (3.23), the phase of the FM waveform is given by:
Z t
θ(t) = θ(0) + 2πkf m(τ )dτ (3.24)
0

Comparing (3.24) with (3.22), we see that FM is equivalent to PM with the integral of the
message. Similarly, for differentiable messages, PM can be interpreted as FM, with the input
to the FM modulator being the derivative of the message. Figure 3.21 provides an example
illustrating this relationship; this is actually a digital modulation scheme called continuous phase
modulation, as we shall see when we study digital communication. In this example, the digital
message +1, −1, −1, +1 is the input to an FM modulator: the instantaneous frequency switches
from fc + kf (for one time unit) to fc − kf (for two time units) and then back to fc + kf again.
The same waveform is produced when we feed the integral of the message into a PM modulator,
as shown in the figure.
When the digital message of Figure 3.21 is input to a phase modulator, then we get a modulated
waveform with phase discontinuities when the message changes sign. This is in contrast to the
output in Figure 3.21, where the phase is continuous. That is, if we compare FM and PM
for the same message, we infer that FM waveforms should have less abrupt phase transitions
due to the smoothing resulting from integration: compare the expressions for the phases of the

107
 +1 +1

Frequency Modulator 0.8

0.6

−1 −1 0.4

0.2

u(t)
Integrate 0

−0.2

−0.4

−0.6

Phase Modulator −0.8

−1
0 0.5 1 1.5 2 2.5 3 3.5 4
t

(a) Messages used for angle modulation (b) Angle modulated signal

Figure 3.21: The equivalence of FM and PM

0.8

0.6

0.4

+1 +1 0.2
u(t)

−0.2

−0.4

−0.6

−0.8

−1
0 0.5 1 1.5 2 2.5 3 3.5 4

−1 −1 t

(a) Digital input to phase modu- (b) Phase shift keyed signal
lator

Figure 3.22: Phase discontinuities in PM signal due to sharp message transitions.

108
modulated signals in (3.22) and (3.24) for the same message m(t). Thus, for a given level of
message variations, we expect FM to have smaller bandwidth. FM is therefore preferred to
PM for analog modulation, where the communication system designer does not have control
over the properties of the message signal (e.g., the system designer cannot require the message
to be smooth). For this reason, and also given the basic equivalence of the two formats, we
restrict the discussion in the remainder of this section to FM for the most part. PM, however,
is extensively employed in digital communication, where the system designer has significant
flexibility in shaping the message signal. In this context, we use the term Phase Shift Keying
(PSK) to denote the discrete nature of the information encoded in the message. Figure 3.22 is
actually a simple example of PSK, although in practice, the phase of the modulated signal is
shaped to be smoother in order to improve bandwidth efficiency.
Frequency Deviation and Modulation Index: The maximum deviation in instantaneous
frequency due to a message m(t) is given by
∆fmax = kf maxt |m(t)|
If the bandwidth of the message is B, the modulation index is defined as
∆fmax kf maxt |m(t)|
β= =
B B
We use the term narrowband FM if β < 1 (typically much smaller than one), and the term
wideband FM if β > 1. We discuss the bandwidth occupancy of FM signals in more detail a little
later, but note for now that the bandwidth of narrowband FM signals is dominated by that of the
message, while the bandwidth of wideband FM signals is dominated by the frequency deviation.
Consider the FM signal corresponding to a sinusoidal message m(t) = Am cos 2πfm t. The phase
deviation due to this message is given by
Z t
Am kf
θ(t) = 2πkf Am cos(2πfm τ ) dτ = sin(2πfm t)
0 fm
Since the maximum frequency deviation ∆fmax = Am kf and the message bandwidth B = fm ,
A k
the modulation index is given by β = fmm f , so that the phase deviation can be written as
θ(t) = β sin 2πfm t (3.25)

Modulation: An FM modulator, by definition, is a Voltage Controlled Oscillator (VCO), whose

output is a sinusoidal wave whose instantaneous frequency offset from a reference frequency is
proportional to the input signal. VCO implementations are often based on the use of varactor
diodes, which provide voltage-controlled capacitance, in LC tuned circuits. This is termed direct
FM modulation, in that the output of the VCO produces a passband signal with the desired
frequency deviation as a function of the message. The VCO output may be at the desired carrier
frequency, or at an intermediate frequency. In the latter scenario, it must be upconverted further
to the carrier frequency, but this operation does not change the frequency modulation. Direct
FM modulation may be employed for both narrowband and wideband modulation.
An alternative approach to wideband modulation is to first generate a narrowband FM signal
(typically using a phase modulator), and to then multiply the frequency (often over multiple
stages) using nonlinearities, thus increasing the frequency deviation as well as the carrier fre-
quency. This method, which is termed indirect FM modulation, is of historical importance, but is
not used in present-day FM systems because direct modulation for wideband FM is now feasible
and cost-effective.
Demodulation: Many different approaches to FM demodulation have evolved over the past cen-
tury. Here we discuss two important classes of demodulators: limiter-discriminator demodulator
in Section 3.3.1, and the phase locked loop in Section 3.5.

109
3.3.1 Limiter-Discriminator Demodulation
Limiter

Passband Bandpass Ideal dθ (t)/dt

received filter
Discriminator
signal (fc )

A cos(2 π fc t + θ (t))

Figure 3.23: Limiter-Discriminator Demodulation of FM.

The task of an FM demodulator is to convert frequency variations in the passband received

signal into amplitude variations, thus recovering an estimate of the message. Ideally, therefore,
an FM demodulator would produce the derivative of the phase of the received signal; this is
termed a discriminator, as shown in Figure 3.23. While an ideal FM signal as in (3.26) does not
have amplitude fluctuations, noise and channel distortions might create such fluctuations, which
leads to unwanted contributions to the discriminator output. In practice, therefore, as shown
in the figure, the discriminator is typically preceded by a limiter, which removes amplitude
fluctuations due to noise and channel distortions which might lead to unwanted contributions
to the discriminator output. This is achieved by passing the modulated sinusoidal waveform
through a hardlimiter, which generates a square wave, and then selecting the right harmonic
using a bandpass filter tuned to the carrier frequency. The overall structure is termed a limiter-
discriminator.
Ideal limiter-discriminator: Following the limiter, we have an FM signal of the form:
yp (t) = A cos(2πfc t + θ(t))
where θ(t) may include contributions due to channel and noise impairments (to be discussed
later), as well as the angle modulation due to the message. An ideal discriminator now produces
the output dθ(t)
dt
(where we ignore scaling factors).

A cos(2 π fc t + θ (t)) d/dt Envelope DC dθ (t)/dt

(from limiter) Detector Block

2 π fc + dθ (t)/dt

Figure 3.24: A crude discriminator based on differentiation and envelope detection.

A crude realization of a discriminator, which converts fluctuations in frequency to fluctuations

in envelope, is shown in Figure 3.24. Taking the derivative of the FM signal
 Z t 
uF M (t) = Ac cos 2πfc t + 2πkf m(τ )dτ + θ0 (3.26)
0

we have
 Z t 
duF M (t)
v(t) = = −Ac (2πfc + 2πkf m(t)) sin 2πfc t + 2πkf m(τ )dτ + θ0
dt 0

The envelope of v(t) is 2πAc |fc + kf m(t)|. Noting that kf m(t) is the instantaneous frequency
deviation from the carrier, whose magnitude is much smaller than fc for a properly designed

110
system, we realize that fc + kf m(t) > 0 for all t. Thus, the envelope equals 2πAc (fc + kf m(t)),
so that passing the discriminator output through an envelope detector yields a scaled and DC-
shifted version of the message. Using AC coupling to reject the DC term, we obtain a scaled
version of the message m(t), just as in conventional AM.

Approximately
linear response

fc f
f0
Band
occupied by
FM Signal

Figure 3.25: Slope detector using a tuned circuit offset from resonance.

The discriminator as described above corresponds to the frequency domain transfer function
H(f ) = j2πf , and can therefore be approximated (up to DC offsets) by transfer functions that
are approximately linear over the FM band of interest. An example of such a slope detector is
given in Figure 3.25, where the carrier frequency fc is chosen at an offset from the resonance
frequency f0 of a tuned circuit.
One problem with the simple discriminator and its approximations is that the envelope detector
output has a significant DC component: when we get rid of this using AC coupling, we also
attenuate low frequency components near DC. This limitation can be overcome by employing
circuits that rely on the approximately linear variations in amplitude and phase of tuned circuits
around resonance to synthesize approximations to an ideal discriminator whose output is the
derivative of the phase. These include the Foster-Seely detector and the ratio detector. Circuit
level details of such implementations are beyond our scope.

3.3.2 FM Spectrum
We first consider a naive but useful estimate of FM bandwidth termed Carson’s rule. We
then show that the spectral properties of FM are actually quite complicated, even for a simple
sinusoidal message, and outline methods of obtaining more detailed bandwidth estimates.
Consider an angle modulated signal, up (t) = Ac cos (2πfc t + θ(t)), where θ(t) contains the mes-
sage information. For a baseband message m(t) of bandwidth B, the phase θ(t) for PM is also
a baseband signal with the same bandwidth. The phase θ(t) for FM is the integral of the mes-
sage. Since integration smooths out the time domain signal, or equivalently, attenuates higher
frequencies, θ(t) is a baseband signal with bandwidth at most B. We therefore loosely think of
θ(t) as having a bandwidth equal to B, the message bandwidth, for the remainder of this section.
The complex envelope of up with respect to fc is given by

u(t) = Ac ejθ(t) = Ac cos θ(t) + jAc sin θ(t)

Now, if |θ(t)| is small, as is the case for narrowband angle modulation, then cos θ(t) ≈ 1 and
sin θ(t) ≈ θ(t), so that the complex envelope is approximately given by
u(t) ≈ Ac + jAc θ(t)

111
Thus, the passband signal is approximately given by

up (t) ≈ Ac cos 2πfc t − θ(t)Ac sin 2πfc t

Thus, the I component has a large unmodulated carrier contribution as in conventional AM, but
the message information is now in the Q component instead of in the I component, as in AM.
The Fourier transform is given by
Ac Ac
Up (f ) = (δ(f − fc ) + δ(f + fc )) − (Θ(f − fc ) − Θ(f + fc ))
2 2j
where Θ(f ) denotes the Fourier transform of θ(t). The magnitude spectrum is therefore given
by
Ac Ac
|Up (f )| = (δ(f − fc ) + δ(f + fc )) + (|Θ(f − fc )| + |Θ(f + fc )|) (3.27)
2 2
Thus, the bandwidth of a narrowband FM signal is 2B, or twice the message bandwidth, just as
in AM. For example, narrowband angle modulation with a sinusoidal message m(t) = cos 2πfm t
k
occupies a bandwidth of 2fm : θ(t) = fmf sin 2πfm t for FM, and θ(t) = kp cos 2πfm t) for PM.
For wideband FM, we would expect the bandwidth to be dominated by the frequency deviation
kf m(t). For messages that have positive and negative peaks of similar size, the frequency devia-
tion ranges between −∆fmax and ∆fmax , where ∆fmax = kf maxt |m(t)|. In this case, we expect
the bandwidth to be dominated by the instantaneous deviations around the carrier frequency,
which spans an interval of length 2∆fmax .
Carson’s rule: This is an estimate for the bandwidth of a general FM signal, based on simply
adding up the estimates from our separate discussion of narrowband and wideband modulation:

BF M ≈ 2B + 2∆fmax = 2B(β + 1) , Carson′ s rule (3.28)

where β = ∆fmax /B is the modulation index, also called the FM deviation ratio, defined earlier.
FM Spectrum for a Sinusoidal Message: In order to get more detailed insight into what
the spectrum of an FM signal looks like, let us now consider the example of a sinusoidal message,
for which the phase deviation is given by θ(t) = β sin 2πfm t, from (3.25). The complex envelope
of the FM signal with respect to fc is given by

u(t) = ejθ(t) = ejβ sin 2πfm t

1
Since the sinusoid in the exponent is periodic with period fm
, so is u(t). It can therefore be
expanded into a Fourier series of the form
∞
X
u(t) = u[n]ej2πnfm t
n=−∞

where the Fourier coefficients {u[n]} are given by

Z 1 Z 1
2fm 2fm
−j2πnfm t
u[n] = fm u(t)e dt = fm ejβ sin 2πfm t e−j2πnfm t dt
− 2f1 − 2f1
m m

Using the change of variables 2πfm t = x, we have

Z π
1
u[n] = ej(β sin x−nx) dx = Jn (β)
2π −π

112
where Jn (·) is the Bessel function of the first kind of order n. While the integrand above is
complex-valued, the integral is real-valued. To see this, use Euler’s formula:

ej(β sin x−nx) = cos(β sin x − nx) + j sin(β sin x − nx)

Since β sin x − nx and the sine function are both odd, the imaginary term sin(β sin x − nx) above
is an odd function, and integrates out to zero over [−π, π]. The real part is even, hence the
integral over [−π, π] is simply twice that over [0, π]. We summarize as follows:
Z π
1 1 π
Z
j(β sin x−nx)
u[n] = Jn (β) = e dx = cos(β sin x − nx)dx (3.29)
2π −π π 0

1
J0(β)
J1(β)
J2(β)
J3(β)

0.5

−0.5
0 1 2 3 4 5 6 7 8 9 10
β

Figure 3.26: Bessel functions of the first kind, Jn (β) versus β, for n = 0, 1, 2, 3.

Bessel functions are available in mathematical software packages such as Matlab and Mathemat-
ica. Figure 3.26 shows some Bessel function plots. Some properties of Bessel functions worth
noting are as follows:
• For n integer, Jn (β) == (−1)n J−n (β) = (−1)n Jn (−β).
• For fixed β, Jn (β) tends to zero fast as n gets large, so that the complex envelope is well ap-
proximated by a finite number of Fourier series components. In particular, a good approximation
is that Jn (β) is small for n > β + 1. This leads to an approximation for the bandwidth of the
FM signal given by 2(β + 1)fm , which is consistent with Carson’s rule.
• For fixed n, Jn (β) vanishes for specific values of β, a fact that can be used for spectral shaping.

To summarize, the complex envelope of an FM signal modulated by a sinusoidal message can be

written as ∞
X
u(t) = ejβ sin 2πfm t = Jn (β)ej2πnfmt (3.30)
n=−∞

The corresponding spectrum is given by

∞
X
U(f ) = Jn (β)δ(f − nfm ) (3.31)
n=−∞

113
Noting that |J−n (β)| = |Jn (β)|, the complex envelope has discrete frequency components at ±nfm
of strength |Jn (β)|: these correspond to frequency components at fc ± nfm in the passband FM
signal.
Fractional power containment bandwidth: By Parseval’s identity for Fourier series, the
power of the complex envelope is given by
∞
X ∞
X
2
1 = |u(t)| = |u(t)|2 = Jn2 (β) = J02 (β) +2 Jn2 (β)
n=−∞ n=1

we can compute the fractional power containment bandwidth as 2Kfm , where K ≥ 1 is the
smaller integer such that
K
X
2
J0 (β) + 2 Jn2 (β) ≥ α
n=1

where α is the desired fraction of power within the band. (e.g., α = 0.99 for the 99% power
containment bandwidth). For integer values of β = 1, ..., 10, we find that K = β + 1 provides a
good approximation to the 99% power containment bandwidth, which is again consistent with
Carson’s formula.

3.3.3 Concept synthesis for FM

a(t)

... 2 mV
100 ...
t (microsec)
200
−2 mV

Figure 3.27: Input to the VCO in Example 3.3.1.

The following worked problem brings together some of the concepts we have discussed regarding
FM.

Example 3.3.1 The signal a(t) shown in Figure 3.27 is fed to a VCO with quiescent frequency
of 5 MHz and frequency deviation of 25 KHz/mV. Denote the output of the VCO by y(t).
(a) Provide an estimate of the bandwidth of y. Clearly state the assumptions that you make.
(b) The signal y(t) is passed through an ideal bandpass filter of bandwidth 5 KHz, centered at
5.005 MHz. Provide the simplest possible expression for the power at the filter output (if you
can give a numerical answer, do so).
Solution: (a) The VCO output is an FM signal with
∆fmax = kf maxt m(t) = 25 KHz/mV × 2 mV = 50 KHz
The message is periodic with period 100 microseconds, hence its fundamental frequency is 10
KHz. Approximating its bandwidth by its first harmonic, we have B ≈ 10 KHz. Using Carson’s
formula, we can approximate the bandwidth of the FM signal at the VCO output as
BF M ≈ 2∆fmax + 2B ≈ 120 KHz
(b) The complex envelope of the VCO output is given by ejθ(t) , where
Z
θ(t) = 2πkf m(τ )dτ

114
For periodic messages with zero DC value (as is the case for m(t) here), θ(t), and hence, ejθ(t)
has the same period as the message. We can therefore express the complex envelope as a Fourier
series with complex exponentials at frequencies nfm , where fm = 10 KHz is the fundamental
frequency for the message, and where n takes integer values. Thus, the FM signal has discrete
components at fc + nfm , where fc = 5 MHz in this example. A bandpass filter at 5.005 MHz
with bandwidth 5 KHz does not capture any of these components, since it spans the interval
[5.0025, 5.0075] MHz, whereas the nearest Fourier components are at 5 MHz and 5.01 MHz.
Thus, the power at the output of the bandpass filter is zero.

3.4 The Superheterodyne Receiver

The receiver in a radio communication system must downconvert the passband received signal
down to baseband in order to recover the message. At the turn of the twentieth century, it
was difficult to produce amplification at frequencies beyond a few MHz) using the vacuum tube
technology of that time. However, higher carrier frequencies are desirable because of the larger
available bandwidths and the smaller antennas required. The invention of the superheterodyne,
or superhet, receiver was motivated by these considerations. Basically, the idea is to use sloppy
design for front end filtering of the received radio frequency (RF) signal, and for translating it to a
lower intermediate frequency (IF). The IF signal is then processed using carefully designed filters
and amplifiers. Subsequently, the IF signal can be converted to baseband in a number of different
ways: for example, an envelope detector for AM radio, a phase locked loop or discriminator
for FM radio, and a coherent quadrature demodulator for digital cellular telephone receivers.
While the original motivation for the superheterodyne receiver is no longer strictly applicable
(modern analog electronics are capable of providing amplification at the carrier frequencies in
commercial use), it is still true that gain is easier to provide at lower frequencies than at higher
frequencies. Furthermore, it becomes possible to closely optimize the processing at a fixed IF
(in terms of amplifier and filter design), while permitting a tunable RF front end with more
relaxed specifications, which is important for the design of radios that operate over a wide
range of carrier frequencies. For example, the superhet architecture is commonly employed for
AM and FM broadcast radio receivers, where the RF front end tunes to the desired station,
translating the received signal to a fixed IF. Radio receivers built with discrete components often
take advantage of the widespread availability of inexpensive filters at certain commonly used
IF frequencies, such as 455 KHz (used for AM radio) and 10.7 MHz (used for FM radio). As
carrier frequencies scale up to the GHz range (as is the case for modern digital cellular and
wireless local area network transceivers), circuit components shrink with the carrier wavelength,
and it becomes possible to implement RF amplifiers and filters using integrated circuits. In such
settings, a direct conversion architecture, in which the passband signal is directly translated to
baseband, becomes increasingly attractive, as discussed later in this section.
The key element in frequency translation is a mixer, which multiplies two input signals. For our
present purpose, one of these inputs is a passband received signal A cos(2πfRF t + θ), where the
envelope A(t) and phase θ(t) are baseband signals that contain message information. The second
input is a local oscillator (LO) signal, which is a locally generated sinusoid cos(2πfLO t) (we set
the LO phase to zero without loss of generality, effectively adopting it as our phase reference).
The output of the mixer is therefore given by

A A
A cos(2πfRF t + θ) cos(2πfLO t) = cos (2π(fRF − fLO )t + θ) + cos (2π(fRF + fLO )t + θ)
2 2
Thus, there are two frequency components at the output of the mixer, fRF + fLO and |fRF −
fLO | (remember that we only need to talk about positive frequencies when discussing physically

115
realizable signals, due to the conjugate symmetry of the Fourier transform of real-valued time
signals). In the superhet receiver, we set one of these as our IF, typically the difference frequency:
fIF = |fRF − fLO |.
RF signal
into antenna
Image reject Mixer Channel select

LNA BPF (IF) IF to baseband

BPF (RF)
conversion

Local Oscillator

Figure 3.28: Generic block diagram for a superhet receiver.

Antenna
receiving Automatic gain control
entire AM band

Mixer
Tunable IF amplifier Envelope Audio To
RF Amplifier (455 KHz) detector amplifier speaker
Center frequency f RF

Station Tunable
selection Local Oscillator

fLO = f RF + 455 KHz

Figure 3.29: A superhet AM receiver.

For a given RF and a fixed IF, we therefore have two choices of LO frequency when fIF =
|fRF − fLO |: fLO = fRF − fIF and fLO = fRF + fIF To continue the discussion, let us consider
the example of AM broadcast radio, which operates over the band from 540 to 1600 KHz, with
10 KHz spacing between the carrier frequencies for different stations. The audio message signal
is limited to 5 KHz bandwidth, modulated using conventional AM to obtain an RF signal of
bandwidth 10 KHz. Figure 3.29 shows a block diagram for the superhet architecture commonly
used in AM receivers. The RF bandpass filter must be tuned to the carrier frequency for the
desired station, and at the same time, the LO frequency into the mixer must be chosen so that
the difference frequency equals the IF frequency of 455 KHz. If fLO = fRF + fIF , then the
LO frequency ranges from 995 to 2055 KHz, corresponding to an approximately 2-fold variation
in tuning range. If fLO = fRF − fIF , then the LO frequency ranges from 85 to 1145 KHz,
corresponding to more than 13-fold variation in tuning range. The first choice is therefore
preferred, because it is easier to implement a tunable oscillator over a smaller tuning range.
Having fixed the LO frequency, we have a desired signal at fRF = fLO − fIF that leads to a
component at IF, and potentially an undesired image frequency at fIM = fLO + fIF = fRF + 2fIF
that also leads to a component at IF. The job of the RF bandpass filter is to block this image
frequency. Thus, the filter must let in the desired signal at fRF (so that its bandwidth must be
larger than 10 KHz), but severely attenuate the image frequency which is 910 KHz away from
the center frequency. It is therefore termed an image reject filter. We see that, for the AM
broadcast radio application, a superhet architecture allows us to design the tunable image reject
filter to somewhat relaxed specifications. However, the image reject filter does let in not only
the signal from the desired station, but also those from adjacent stations. It is the job of the
IF filter, which is tuned to the fixed frequency of 455 KHz, to filter out these adjacent stations.

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 BEFORE TRANSLATION TO IF
2fIF
HRF (f)
B channel Image frequency
gets blocked

finterf fRF fLO f IM f

Image reject = fLO − IF
f
filter has = fLO + fIF
relaxed specs

AFTER TRANSLATION TO IF

HIF (f) IF filter must have

sharp cutoff to block
interfering signal

f IF

Figure 3.30: The role of image rejection and channel selection in superhet receivers.

For this purpose, we use a highly selective filter at IF with a bandwidth of 10 KHz. Figure 3.30
illustrates these design considerations more generally.
Receivers for FM broadcast radio also commonly use a superhet architecture. The FM broadcast
band ranges from 88-108 MHz, with carrier frequency separation of 200 KHz between adjacent
stations. The IF is chosen at 10.7 MHz, so that the LO is tuned from 98.7 to 118.7 MHz for the
choice fLO = fRF + fIF . The RF filter specifications remain relaxed: it has to let in the desired
signal of bandwidth 200 KHz, while rejecting an image frequency which is 2fIF = 21.4 MHz away
from its center frequency. We discuss the structure of the FM broadcast signal, particularly the
way in which stereo FM is transmitted, in more detail in Section 3.6.
Roughly indexing the difficulty of implementing a filter by the ratio of its center frequency to its
bandwidth, or its Q factor, with high Q being more difficult to implement, we have the following
fundamental tradeoff for superhet receivers. If we use a large IF, then the Q needed for the
image reject filter is smaller. On the other hand, the Q needed for the IF filter to reject an
interfering signal whose frequency is near that of the desired signal becomes higher. In modern
digital communication applications, superheterodyne reception with multiple IF stages may be
used in order to work around this tradeoff, in order to achieve the desired gain for the signal of
interest and to attenuate sufficiently interference from other signals, while achieving an adequate
degree of image rejection. Image rejection can be enhanced by employing appropriately designed
image-reject mixer architectures.
Direct conversion receivers: With the trend towards increasing monolithic integration of
digital communication transceivers for applications such as cellular telephony and wireless local
area networks, the superhet architecture is often being supplanted by direct conversion (or zero
IF) receivers, in which the passband received signal is directly converted down to baseband
using a quadrature mixer at the RF carrier frequency. In this case, the desired signal is its
own image, which removes the necessity for image rejection. Moreover, interfering signals can be
filtered out at baseband, often using sophisticated digital signal processing after analog-to-digital
conversion (ADC), provided that there is enough dynamic range in the circuitry to prevent a
strong interferer from swamping the desired signal prior to the ADC. In contrast, the high Q
bandpass filters required for image rejection and interference suppression in the superhet design

117
must often be implemented off-chip using, for example, surface acoustic wave (SAW) devices,
which is bulky and costly. Thus, direct conversion is in some sense the “obvious” thing to do,
except that historically, people were unable to make it work, leading to the superhet architecture
serving as the default design through most of the twentieth century. A key problem with direct
conversion is that LO leakage into the RF input of the mixer causes self-mixing, leading to a DC
offset. While a DC offset can be calibrated out, the main problem is that it can saturate the
amplifiers following the mixer, thus swamping out the contribution of the weaker received signal.
Note that the DC offset due to LO leakage is not a problem with a superhet architecture, since
the DC term gets filtered out by the passband IF filter. Other problems with direct conversion
include 1/f noise and susceptibility to second order nonlinearities, but discussion of these issues
is beyond our current scope. However, since the 1990s, integrated circuit designers have managed
to overcome these and other obstacles, and direct conversion receivers have become the norm
for monolithic implementations of modern digital communication transceivers. These include
cellular systems in various licensed bands ranging from 900 MHz to 2 GHz, and WLANs in the
2.4 GHz and 5 GHz unlicensed bands.
The insatiable demand for communication bandwidth virtually assures us that we will seek to
exploit frequency bands well beyond 5 GHz, and circuit designers will be making informed choices
between the superhet and direct conversion architectures for radios at these higher frequencies.
For example, the 60 GHz band in the United States has 7 GHz of unlicensed spectrum; this
band is susceptible to oxygen absorption, and is ideally suited for short range (e.g. 10-500
meters range) communication both indoors and outdoors. Similarly, the 71-76 GHz and 81-
86 GHz bands, which avoid oxygen absorption loss, are available for semi-unlicensed point-to-
point “last mile” links. Just as with cellular and WLAN applications in lower frequency bands,
we expect that proliferation of applications using these “millimeter (mm) wave” bands would
require low-cost integrated circuit transceiver implementations. Based on the trends at lower
frequencies, one is tempted to conjecture that initial circuit designs might be based on superhet
architectures, with direct conversion receivers becoming subsequently more popular as designers
become more comfortable with working at these higher frequencies. It is interesting to note
that the design experience at lower carrier frequencies does not go to waste; for example, direct
conversion receivers at, say, 5 GHz, can serve as the IF stage for superhet receivers for mm wave
communication.

3.5 The Phase Locked Loop

The phase locked loop (PLL) is an effective FM demodulator, but also has a far broader range
of applications, including frequency synthesis and synchronization. We therefore treat it sepa-
rately from our coverage of FM. The PLL provides a canonical example of the use of feedback
for estimation and synchronization in communication systems, a principle that is employed in
variants such as the Costas’ loop and the delay locked loop.
The key idea behind the PLL, depicted in Figure 3.31, is as follows: we would like to lock on to
the phase of the input to the PLL. We compare the phase of the input with that of the output of
a voltage controlled oscillator (VCO) using a phase detector. The difference between the phases
drives the input of the VCO. If the VCO output is ahead of the PLL input in phase, then we
would like to retard the VCO output phase. If the VCO output is behind the PLL input in phase,
we would like to advance the VCO output phase. This is done by using the phase difference to
control the VCO input. Typically, rather than using the output of the phase detector directly
for this purpose, we smooth it out using a loop filter in order to reduce the effect of noise.
Mixer as phase detector: The classical analog realization of the PLL is based on using a
mixer (i.e., a multiplier) as a phase detector. To see how this works, consider the product of two

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 Function of phase difference θ i − θ o
PLL Input
Phase Loop
Phase θ i Detector Filter

Phase θ o

VCO
VCO Output

Figure 3.31: PLL block diagram.

1/2 A c A v sin ( θ i −θo )

+ passband (2 fc ) term
PLL Input
Ac cos (2 π fc t+ θ i ) Loop
Filter

−Av sin (2 π fc t+ θo )
VCO Output VCO
x(t)

Figure 3.32: PLL realization using a mixer as phase detector.

sinusoids whose phases we are trying to align:

1 1
cos(2πfc t + θ1 ) cos(2πfc t + θ2 ) = cos(θ1 − θ2 ) + cos(4πfc t + θ1 + θ2 ))
2 2
The second term on the right-hand side is a passband signal at 2fc which can be filtered out
by a lowpass filter. The first term contains the phase difference θ1 − θ2 , and is to be used to
drive the VCO so that we eventually match the phases. Thus, the first term should be small
when we are near a phase match. Since the driving term is the cosine of the phase difference,
the phase match condition is θ1 − θ2 = π/2. That is, using a mixer as our phase detector means
that, when the PLL is locked, the phase at the VCO output is 90◦ offset from the phase of the
PLL input. Now that we know this, we adopt a more convenient notation, changing variables
to define a phase difference whose value at the desired matched state is zero rather than π/2.
Let the PLL input be denoted by Ac cos(2πfc + θi (t)), and let the VCO output be denoted by
Av cos(2πfc + θo (t) + π2 ) = −Av sin(2πfc + θo (t)). The output of the mixer is now given by

−Ac Av cos (2πfc + θi (t)) sin (2πfc + θo (t))

Ac Av Ac Av
= 2
sin (θi (t) − θo (t)) − 2
sin (4πfc t + θi (t) + θo (t))
The second term on the right-hand side is a passband signal at 2fc which can be filtered out
as before. The first term is the desired driving term, and with the change of notation, we note
that the desired state, when the driving term is zero, corresponds to θi = θo . The mixer based
realization of the PLL is shown in Figure 3.32.
The instantaneous frequency of the VCO is proportional to its input. Thus the phase of the
VCO output −sin(2πfc t + θo (t)) is given by
Z t
θo (t) = Kv x(τ )dτ
0

119
ignoring integration constants. Taking Laplace transforms, we have Θo (s) = Kv X(s)/s. The
reference frequency fc is chosen as the quiescent frequency of the VCO, which is the frequency it
would produce when its input voltage is zero.
PLL Input
XOR gate
(square wave) Loop
Filter

VCO Output VCO

(square wave)

VCO Output

PLL Input

γ
Output of XOR gate VHI

VLO

Figure 3.33: PLL realization using XOR gate as phase detector.

V’
V = V’ − V
( LO +VHI )/2
VHI

−2π −π 0 π 2π γ −π −π/2 0 π/2 π θ=γ − π/2

VLO

(a) DC value of output of XOR gate. (b) XOR phase detector output after axes translation.

Figure 3.34: Response for the XOR phase detector.

Mixed signal phase detectors: Modern hardware realizations of the PLL, particularly for
applications involving digital waveforms (e.g., a clock signal), often realize the phase detector
using digital logic. The most rudimentary of these is an exclusive or (XOR) gate, as shown in
Figure 3.33. For the scenario depicted in the figure, we see that the average value of the output
of the XOR gate is linearly related to the phase offset γ. Normalizing a period of the square
wave to length 2π, this DC value V ′ is related to γ as shown in Figure 3.34(a). Note that, for
zero phase offset, we have V ′ = VHI , and that the response is symmetric around γ = 0. In order
to get a linear phase detector response going through the origin, we translate this curve along
both axes: we define V = V ′ − (VLO + VHI ) /2 as a centered response, and we define the phase
offset θ = γ − π2 . Thus, the lock condition (θ = 0) corresponds to the square waves being 90◦ out
of phase. This translation gives us the phase response shown in Figure 3.34(b), which looks like
a triangular version of the sinusoidal response for the mixer-based phase detector.
The simple XOR-based phase detector has the disadvantage of requiring that the waveforms
have 50% duty cycle. In practice, more sophisticated phase detectors, often based on edge
detection, are used. These include “phase-frequency detectors” that directly provide information

120
on frequency differences, which is useful for rapid locking. While discussion of the many phase
detector variants employed in hardware design is beyond our scope, references for further study
are provided at the end of this chapter.

3.5.1 PLL Applications

Before trying to understand how a PLL works in more detail, let us discuss how we would use
it, assuming that it has been properly designed. That is, suppose we can design a system such
as that depicted in Figure 3.32, such that θo (t) ≈ θi (t). What would we do with such a system?
Function of phase difference θ i − θ o

FM received Phase Loop

signal Detector Filter

VCO

FM Demodulator
Output

Figure 3.35: The PLL is an FM demodulator.

PLL as FM demodulator: If the PLL input is an FM signal, its phase is given by

Z t
θi (t) = 2πkf m(τ )dτ
0

The VCO output phase is given by

Z t
θo (t) = Kv x(τ )dτ
0
dθo dθi
If θo ≈ θi , then dt
≈ dt
, so that
Kv x(t) ≈ 2πkf m(t)
That is, the VCO input is approximately equal to a scaled version of the message. Thus, the
PLL is an FM demodulator, where the FM signal is the input to the PLL, and the demodulator
output is the VCO input, as shown in Figure 3.35.
PLL as frequency synthesizer: The PLL is often used to synthesize the local oscillators used
in communication transmitters and receivers. In a typical scenario, we might have a crystal
oscillator which provides an accurate frequency reference at a relatively low frequency, say 40
MHz. We wish to use this to derive an accurate frequency reference at a higher frequency, say 1
GHz, which might be the local oscillator used at an IF or RF stage in the transceiver. We have
a VCO that can produce frequencies around 1 GHz (but is not calibrated to produce the exact
value of the desired frequency), and we wish to use it to obtain a frequency f0 that is exactly
K times the crystal frequency fcrystal . This can be achieved by adding a frequency divider into
the PLL loop, as shown in Figure 3.36. Such frequency dividers can be implemented digitally
by appropriately skipping pulses. Many variants of this basic concept are possible, such as using
multiple frequency dividers, frequency multipliers, or multiple interacting loops.
All of these applications rely on the basic property that the VCO output phase successfully tracks
some reference phase using the feedback in the loop. Let us now try to get some insight into how
this happens, and into the impact of various parameters on the PLL’s performance.

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 fcrystal
Crystal Phase Loop
Oscillator Frequency Detector Filter
reference

VCO

Kfcrystal Frequency
synthesizer
output
Frequency Divider
(divide by K)

Figure 3.36: Frequency synthesis using a PLL by inserting a frequency divider into the loop.

3.5.2 Mathematical Model for the PLL

Loop gain
and filter

sin( ) K G(s)
θi
−
θo
1/s
VCO functionality
(normalized)

Figure 3.37: Nonlinear model for mixer-based PLL.

The mixer-based PLL in Figure 3.32 can be modeled as shown in Figure 3.37, where θi (t) is
the input phase, and θo (t) is the output phase. It is also useful to define the corresponding
instantaneous frequencies (or rather, frequency deviations from the VCO quiescent frequency
fc ):
1 dθi (t) 1 dθo (t)
fi (t) = , fo (t) =
2π dt 2π dt
The phase and frequency errors are defined as

θe (t) = θi (t) − θo (t) , fe (t) = fi (t) − fo (t)

In deriving this model, we can ignore the passband term at 2fc , which will get rejected by the
integration operation due to the VCO, as well as by the loop filter (if a nontrivial lowpass loop
filter is employed). From Figure 3.32, the sine of the phase difference is amplified by 12 Ac Av due
to the amplitudes of the PLL input and VCO output. This is passed through the loop filter,
which has transfer function G(s), and then through the VCO, which has a transfer function Kv /s.
The loop gain K shown in Figure 3.37 is set to be the product K = 21 Ac Av Kv (in addition, the
loop gain also includes additional amplification or attenuation in the loop that is not accounted
for in the transfer function G(s)).
The model in Figure 3.37 is difficult to analyze because of the sin(·) nonlinearity after the phase
difference operation. One way to avoid this difficulty is to linearize the model by simply dropping

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 Loop gain
and filter

θi K G(s)
−
θo
1/s
VCO functionality
(normalized)

Figure 3.38: Linearized PLL model.

the nonlinearity. The motivation is that, when the input and output phases are close, as is the
case when the PLL is in tracking mode, then

sin(θi − θo ) ≈ θi − θo

Applying this approximation, we obtain the linearized model of Figure 3.38. Note that, for the
XOR-based response shown in Figure 3.34(b), the response is exactly linear for |θ| ≤ π2 .

3.5.3 PLL Analysis

Under the linearized model, the PLL becomes an LTI system whose analysis is conveniently
performed using the Laplace transform. From Figure 3.38, we see that

(Θi (s) − Θo (s)) KG(s)/s = Θo (s)

from which we infer the input-output relationship

Θo (s) KG(s)
H(s) = = (3.32)
Θi (s) s + KG(s)

It is also useful to express the phase error θe in terms of the input θi , as follows:

Θe (s) Θi (s) − Θo (s) s

He (s) = = = 1 − H(s) = (3.33)
Θi (s) Θi (s) s + KG(s)

For this LTI model, the same transfer functions also govern the relationships between the input
s s
and output instantaneous frequencies: since Fi (s) = 2π Θi (s) and Fo (s) = 2π Θo (s), we obtain
Fo (s)/Fi (s) = Θo (s)/Θi (s). Thus, we have

Fo (s) KG(s)
= H(s) = (3.34)
Fi (s) s + KG(s)

Fi (s) − Fo (s) s
= He (s) = (3.35)
Fi (s) s + KG(s)

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First order PLL: When we have a trivial loop filter, G(s) = 1, we obtain the first order response

K s
H(s) = , He (s) =
s+K s+K
which is a stable response for loop gain K > 0, with a single pole at s = −K. It is interesting to
see what happens when the input phase is a step function, θi (t) = ∆θI[0,∞) (t), or Θi (s) = ∆θ/s.
We obtain
K∆θ ∆θ ∆θ
Θo (s) = H(s)Θi (s) = = −
s(s + K) s s+K
Taking the inverse Laplace transform, we obtain

θo (t) = ∆θ(1 − e−Kt )I[0,∞)(t)

so that θo (t) → ∆θ as t → ∞. Thus, the first order PLL can track a sudden change in phase,
with the output phase converging to the input phase exponentially fast. The residual phase error
is zero. Note that we could also have inferred this quickly from the final value theorem, without
taking the inverse Laplace transform:

lim θe (t) = lim sΘe (s) = lim sHe (s)Θi (s) (3.36)
t→∞ s→0 s→0

Specializing to the setting of interest, we obtain

s ∆θ0
lim θe (t) = lim s =0
t→∞ s→0 s+K s

We now examine the response of the first order PLL to a frequency step ∆f , so that the instanta-
neous input frequency is fi (t) = ∆f I[0,∞) (t). The corresponding Laplace transform is Fi (s) = ∆f
s
.
The input phase is the integral of the instantaneous frequency:
Z t
θi (t) = 2π fi (τ )dτ
0

The Laplace transform of the input phase is therefore given by

2π∆f
Θi (s) = 2πF (s)/s =
s2
Given that the input-output relationships are identical for frequency and phase, we can reuse
the computations we did for the phase step input, replacing phase by frequency, to conclude that
fo (t) = ∆f (1 − e−Kt )I[0,∞) (t) → ∆f as t → ∞, so that the steady-state frequency error is zero.
The corresponding output phase trajectory is left as an exercise, but we can use the final value
theorem to compute the limiting value of the phase error:

s 2π∆f 2π∆f
lim θe (t) = lim s 2
=
t→∞ s→0 s+K s K
Thus, the first order PLL can adapt its frequency to track a step frequency change, but there is
a nonzero steady-state phase error. This can be fixed by increasing the order of the PLL, as we
now show below.
Second order PLL: We now introduce a loop filter which feeds back both the phase error and
the integral of the phase error to the VCO input (in control theory terminology, we are using

124
”proportional plus integral” feedback). That is, G(s) = 1 + a/s, where a > 0. This yields the
second order response
KG(s) K(s + a)
H(s) = = 2
s + KG(s) s + Ks + Ka
s s2
He (s) = = 2
s + KG(s) s + Ks + Ka
√
2
The poles of the response are at s = −K± K2 −4Ka . It is easy to check that the response is stable
(i.e., the poles are in the left half plane) for K > 0. The poles are conjugate symmetric with an
imaginary component if K 2 − 4Ka < 0, or K < 4a, otherwise they are both real-valued. Note
that the phase error due to a step frequency response does go to zero. This is easily seen by
invoking the final value theorem (3.36):

s2 2π∆f
lim θe (t) = lim s =0
t→∞ s→0 s + Ks + Ka s2
2

Thus, the second order PLL has zero steady state frequency and phase errors when responding
to a constant frequency offset.
We have seen now that the first order PLL can handle step phase changes, and the second order
PLL can handle step frequency changes, while driving the steady-state phase error to zero. This
pattern continues as we keep increasing the order of the PLL: for example, a third order PLL
can handle a linear frequency ramp, which corresponds to Θi (s) being proportional to 1/s3.
Linearized analysis provides quick insight into the complexity of the phase/frequency variations
that the PLL can track, as a function of the choice of loop filter and loop gain. We now take
another look at the first order PLL, accounting for the sin(·) nonlinearity in Figure 3.37, in
order to provide a glimpse of the approach used for handling the nonlinear differential equations
involved, and to compare the results with the linearized analysis.
Nonlinear model for the first order PLL: Let us try to express the phase error θe in terms of
the input phase for a first order PLL, with G(s) = 1. The model of Figure 3.37 can be expressed
in the time domain as: Z t
K sin(θe (τ ))dτ = θo (t) = θi (t) − θe (t)
0
Differentiating with respect to t, we obtain
dθi dθe
K sin θe = − (3.37)
dt dt
(Both θe and θi are functions of t, but we suppress the dependence for notational simplicity.)
Let us now specialize to the specific example of a step frequency input, for which
dθi
= 2π∆f
dt
Plugging into (3.37) and rearranging, we get
dθe
= 2π∆f − K sin θe (3.38)
dt

We cannot solve the nonlinear differential equation (3.38) for θe analytically, but we can get
useful insight by a “phase plane plot” of dθdte against θe , as shown in Figure 3.39. Since sin θe ≤ 1,
we have dθdte ≥ 2π∆f − K, so that, if ∆f > 2π K
, then dθdte > 0 for all t. Thus, for large enough

125
 d θ e /dt PLL locks

2π ∆f

θe
2π ∆f − K

θ e (0) θ e (1)

PLL does not lock

d θ e /dt

2π ∆f

2π ∆f − K
θe

Figure 3.39: Phase plane plot for first order PLL.

K
frequency offset, the loop never locks. On the other hand, if ∆f < 2π , then the loop does lock.
In this case, starting from an initial error, say θe (0), the phase error follows the trajectory to the
right (if the derivative is positive) or left (if the derivative is negative) until it hits a point at
which dθdte = 0. From (3.38), this happens when

2π∆f
sin θe = (3.39)
K
Due to the periodicity of the sine function, if θ is a solution to the preceding equation, so is
θ + 2π. Thus, if the equation has a solution, there must be at least one solution in the basic
interval [−π, π]. Moreover, since sin θ = sin(π − θ), if θ is a solution, so is π
− θ, so that there
are actually two solutions in [−π, π]. Let us denote by θe (0) = sin−1 2π∆f K
the solution that
lies in the interval [−π/2, π/2]. This forms a stable equilibrium: from (3.38), we see that the
derivative is negative for phase error slightly above θe (0), and is positive as the phase error
slightly below θe (0), so that the phase error is driven back to θe (0) in each case. Using exactly
the same argument, we see that the points θe (0) + 2nπ are also stable equilibria, where n takes
integer values. However, another solution to (3.39) is θe (1) = π − θ(0), and translations of it
by . It is easy to see that this is an unstable equilibrium: when there is a slight perturbation,
the sign of the derivative is such that it drives the phase error away from θe (1). In general,
θe (1) + 2nπ are unstable equlibria, where n takes integer values. Thus, if the frequency offset is
K
within the “pull-in range” 2π of the first order PLL, then the steady state phase offset (modulo
−1 2π∆f
, which, for small values of 2π∆f


2π) is θe (0) = sin K K
, is approximately equal to the value
2π∆f
K
predicted by the linearized analysis.
Linear versus nonlinear model: Roughly speaking, the nonlinear model (which we simply
simulate when phase-plane plots get too complicated) tells us when the PLL locks, while the

126
linearized analysis provides accurate estimates when the PLL does lock. The linearized model
also tells us something about scenarios when the PLL does not lock: when the phase error blows
up for the linearized model, it indicates that the PLL will perform poorly. This is because the
linearized model holds under the assumption that the phase error is small; if the phase error
under this optimistic assumption turns out not to be small, then our initial assumption must
have been wrong, and the phase error must be large.

PLL Input Phase Detector

(0.5 V/radian)

VCO
(10 KHz/V)

Figure 3.40: PLL for Example 3.5.1.

The following worked problem illustrates application of linearized PLL analysis.

Example 3.5.1 Consider the PLL shown in Figure 3.40, assumed to be locked at time zero.
(a) Suppose that the input phase jumps by e = 2.72 radians at time zero (set the phase just
before the jump to zero, without loss of generality). How long does it take for the difference
between the PLL input phase and VCO output phase to shrink to 1 radian? (Make sure you
specify the unit of time that you use.)
(b) Find the limiting value of the phase error (in radians) if the frequency jumps by 1 KHz just
after time zero.
Solution: Let θe (t) = θi (t) − θo (t) denote the phase error. In the s domain, it is related to the
input phase as follows:
K
Θi (s) − Θe (s) = Θe (s)
s
so that
Θe (s) s
=
Θi (s) s+K
(a) For a phase jump of e radians at time zero, we have Θi (s) = es , which yields
s e
Θe (s) = Θi (s) =
s+K s+K
Going to the time domain, we have
θe (t) = ee−Kt = e1−Kt
so that θe (t) = 1 for 1 − Kt = 0, or t = K1 = 51 milliseconds.
(b) For a frequency jump of ∆f , the Laplace transform of the input phase is given by
2π∆f
Θi (s) =
s2
so that the phase error is given by
s 2π∆f
Θe (s) = Θi (s) =
s+K s(s + K)

127
Using the final value theorem, we have
2π∆f
lim θe (t) = lim sΘe (s) =
t→∞ s→0 K
For ∆f = 1 KHz and K = 5 KHz/radian, this yields a phase error of 2π/5 radians, or 72◦ .

3.6 Some Analog Communication Systems

Some of the analog communication systems that we encounter (or at least, used to encounter)
in our daily lives include broadcast radio and television. We have already discussed AM radio in
the context of the superhet receiver. We now briefly discuss FM radio and television. Our goal
is to highlight design concepts, and the role played in these systems by the various modulation
formats we have studied, rather than to provide a detailed technical description. Other commonly
encountered examples of analog communication that we do not discuss include analog storage
media (audiotapes and videotapes), analog wireline telephony, analog cellular telephony, amateur
ham radio, and wireless microphones.

3.6.1 FM radio
Pilot
L+R signal DSB−SC modulated
L−R signal

15 19 23 38 53 Frequency (KHz)

Figure 3.41: Spectrum of baseband input to FM modulator for FM stereo broadcast.

FM mono radio employs a peak frequency deviation of 75 KHz, with the baseband audio message
signal bandlimited to 15 KHz; this corresponds to a modulation index of 5. Using Carson’s
formula, the bandwidth of the FM radio signal can be estimated as 180 KHz. The separation
between adjacent radio stations is 200 KHz. FM stereo broadcast transmits two audio channels,
“left” and “right,” in a manner that is backwards compatible with mono broadcast, in that
a standard mono receiver can extract the sum of the left and right channels, while remaining
oblivious to whether the broadcast signal is mono or stereo. The structure of the baseband signal
into the FM modulator is shown in Figure 3.41. The sum of the left and right channels, or the
L + R signal, occupies a band from 30 Hz to 15 KHz. The difference, or the L − R signal (which
also has a bandwidth of 15 KHz), is modulated using DSB-SC, using a carrier frequency of 38
KHz, and hence occupies a band from 23 KHz to 53 KHz. A pilot tone at 19 KHz, at half the
carrier frequency for the DSB signal, is provided to enable coherent demodulation of the DSB-SC
signal. The spacing between adjacent FM stereo broadcast stations is still 200 KHz, which makes
it a somewhat tight fit (if we apply Carson’s formula with a maximum frequency deviation of 75
KHz, we obtain an RF bandwidth of 256 KHz).
The format of the baseband signal in Figure 3.41 (in particular, the DSB-SC modulation of the
difference signal) seems rather contrived, but the corresponding modulator can be implemented
quite simply, as sketched in Figure 3.42: we simply switch between the L and R channel audio
signals using a 38 KHz clock. As we shown in one of the problems, this directly yields the L + R

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 Frequency divide
by two
19 KHz pilot
38 KHz clock
L channel signal Transmitted
FM modulator
Composite signal
R channel signal
message
signal

Figure 3.42: Block diagram of a simple FM stereo transmitter.

signal, plus the DSB-SC modulated L − R signal. It remains to add in the 19 KHz pilot before
feeding the composite baseband signal to the FM modulator.
The receiver employs an FM demodulator to obtain an estimate of the baseband transmitted
signal. The L + R signal is obtained by bandlimiting the output of the FM demodulator to 15
KHz using a lowpass filter; this is what an oblivious mono receiver would do. A stereo receiver,
in addition, processes the output of the FM demodulator in the band from 15 KHz to 53 KHz.
It extracts the 19 KHz pilot tone, doubles its frequency to obtain a coherent carrier reference,
and uses that to demodulate the L − R signal sent using DSB-SC. It then obtains the L and R
channels by adding and subtracting the L + R and L − R signals from each other, respectively.

3.6.2 Analog broadcast TV

While analog broadcast TV is obsolete, and is being replaced by digital TV as we speak, we
discuss it briefly here to highlight a few features. First, it illustrates an application of several
modulation schemes: VSB (for intensity information), quadrature modulation (for the color in-
formation), and FM (for audio information). Second, it is an interesting example of how the
embedding of different kinds of information in analog form must account for the characteris-
tics of the information source (video) and destination (a cathode ray tube TV monitor). This
customized, and rather painful, design process is in contrast to the generality and conceptual
simplification provided by the source-channel separation principle in digital communication (men-
tioned in Chapter 1). Indeed, from Chapter 4 onwards, where we restrict attention to digital
communication, we do not need to discuss source characteristics.
We first need a quick discussion of CRT TV monitors. An electron beam impinging on a fluores-
cent screen is used to emit the light that we perceive as the image on the TV. The electron beam
is “raster scanned” in horizontal lines moving down the screen, with its horizontal and vertical
location controlled by two magnetic fields created by voltages, as shown in Figure 3.43. We rely
on the persistence of human vision to piece together these discrete scans into a continuous image
in space and time. Black and white TV monitors use a phosphor (or fluorescent material) that
emits white light when struck by electrons. Color TV monitors use three kinds of phosphors,
typically arranged as dots on the screen, which emit red, green and blue light, respectively, when
struck by electrons. Three electron beams are used, one for each color. The intensity of the
emitted light is controlled by the intensity of the electron beam. For historical reasons, the scan
rate is chosen to be equal to the frequency of the AC power (otherwise, for the power supplies
used at the time, rolling bars would appear on the TV screen). In the United States, this means
that the scan rate is set at 60 Hz (the frequency of the AC mains).
In order to enable the TV receiver to control the operation of the CRT monitor, the received signal
must contain not only intensity and color information, but also the timing information required
to correctly implement the raster scan. Figure 3.44 shows the format of the composite video signal

129
 Electron beam Fluorescent screen
Horizontal line scan

Horizontal retrace

Magnetic fields
controlling beam
trajectory
CRT Schematic Raster scan pattern

Horizontal position control

Vertical position control

Controls needed for raster scan

Figure 3.43: Implementing raster scan in a CRT monitor requires magnetic fields controlled by
sawtooth waveforms.

Video information Video information

(odd field) (even field)
Line 1 Line 3 Line 479 Line 2 Line 4 Line 480

Horizontal sync Horizontal sync

pulses pulses

Vertical sync
waveforms
(not shown)

Figure 3.44: The structure of a black and white composite video signal (numbers apply to the
NTSC standard).

130
containing this information. In order to reduce flicker (again a historical legacy, since older CRT
monitors could not maintain intensities long enough if the time between refreshes is too long), the
CRT screen is painted in two rounds for each image (or frame): first the odd lines (comprising the
odd field) are scanned, then the even lines (comprising the even field) are scanned. For the NTSC
standard, this is done at a rate of 60 fields per second, or 30 frames per second. A horizontal sync
pulse is inserted between each line. A more complex vertical synchronization waveform is inserted
between each field; this enables vertical synchronization (as well as other functionaliities that
we do not discuss here). The receiver can extract the horizontal and vertical timing information
from the composite video signal, and generate the sawtooth waveforms required for controlling
the electron beam (one of the first widespread commercial applications of the PLL was for this
purpose). For the NTSC standard, the composite video signal spans 525 lines, about 486 of
which are actually painted (counting both the even and odd fields). The remaining 39 lines
accommodate the vertical synchronization waveforms.
The bandwidth of the baseband video signal can be roughly estimated as follows. Assuming
about 480 lines, with about 640 pixels per line (for an aspect ratio of 4:3), we have about 300,000
pixels, refreshed at the rate of 30 times per second. Thus, our overall sampling rate is about
9 Msamples/second. This can accurately represent a signal of bandwidth 4.5 MHz. For a 6
MHz TV channel bandwidth, DSB and wideband FM are therefore out of the question, and
VSB was chosen to modulate the composite video signal. However, the careful shaping of the
spectrum required for VSB is not carried out at the transmitter, because this would require
the design of high-power electronics with tight specifications. Instead, the transmitter uses a
simple filter, while the receiver, which deals with a low-power signal, accomplishes the VSB
shaping requirement in (3.21). Audio modulation is done using FM in a band adjacent to the
one carrying the video signal.
While the signaling for black and white TV is essentially the same for all existing analog TV
standards, the insertion of color differs among standards such as NTSC, PAL and SECAM. We
do not go into details here, but, taking NTSC as an example, we note that the frequency domain
characteristics of the black and white composite video signal is exploited in rather a clever way
to insert color information. The black and white signal exhibits a clustering of power around the
Fourier series components corresponding to the horizontal scan rate, with the power decaying
around the higher order harmonics. The color modulated signal uses the same band as the black
and white signal, but is inserted between two such harmonics, so as to minimize the mutual
interference between the intensity information and the color information. The color information
is encoded in two baseband signals, which are modulated on to the I and Q components using
QAM. Synchronization information that permits coherent recovery of the color subcarrier for
quadrature demodulation is embedded in the vertical synchronization waveform.

3.7 Problems
Amplitude modulation

Problem 3.1 Figure 3.45 shows a signal obtained after amplitude modulation by a sinusoidal
message. The carrier frequency is difficult to determine from the figure, and is not needed for
answering the questions below.
(a) Find the modulation index.
(b) Find the signal power.
(c) Find the bandwidth of the AM signal.

Problem 3.2 Consider a message signal m(t) = 2 cos 2πt + π4 .

(a) Sketch the spectrum U(f ) of the DSB-SC signal up (t) = 8m(t) cos 400πt. What is the power

131
 30

20

10

−10

−20

−30
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
Time (milliseconds)

Figure 3.45: Amplitude modulated signal for Problem 3.1.

of u?
(b) Carefully sketch the output of an ideal envelope detector with input up . On the same plot,
sketch the message signal m(t).
(c) Let vp (t) denote the waveform obtained by high-pass filtering the signal u(t) so as to let
through only frequencies above 200 Hz. Find vc (t) and vs (t) such that we can write

vp (t) = vc (t) cos 400πt − vs (t) sin 400πt

and sketch the envelope of v.

Problem 3.3 A message to be transmitted using AM is given by

m(t) = 3 cos 2πt + 4 sin 6πt

where the unit of time is milliseconds. It is to be sent using a carrier frequency of 600 KHz.
(a) What is the message bandwidth? Sketch its magnitude spectrum, clearly specifying the units
used on the frequency axis.
(b) Find an expression for the normalized message mn (t).
(c) For a modulation index of 50%, write an explicit time domain expression for the AM signal.
(d) What is the power efficiency of the AM signal?
(e) Sketch the magnitude spectrum for the AM signal, again clearly specifying the units used on
the frequency axis.
(f) The AM signal is to be detected using an envelope detector (as shown in Figure 3.8), with
R = 50 ohms. What is a good range of choices for the capacitance C?

Problem 3.4 Consider a message signal m(t) = cos(2πfm t + φ), and a corresponding DSB-SC
signal up (t) = Am(t) cos 2πfc t, where fc > fm .
(a) Sketch the spectra of the corresponding LSB and USB signals (if the spectrum is complex-
valued, sketch the real and imaginary parts separately).
(b) Find explicit time domain expressions for the LSB and USB signals.

Problem 3.5 One way of avoiding the use of a mixer in generating AM is to pass x(t) =
m(t) + α cos 2πfc t through a memoryless nonlinearity and then a bandpass filter.
(a) Suppose that M(f ) = (1 − |f |/10)I[−10,10] (the unit of frequency is in KHz) and fc is 900

132
KHz. For a nonlinearity f (x) = βx2 + x, sketch the magnitude spectrum at the output of the
nonlinearity when the input is x(t), carefully labeling the frequency axis.
(b) For the specific settings in (a), characterize the bandpass filter that you should use at the
output of the nonlinearity so as to generate an AM signal carrying the message m(t)? That is,
describe the set of the frequencies that the BPF must reject, and those that it must pass.

Problem 3.6 Consider a DSB signal corresponding to the message m(t) = sinc(2t) and a carrier
frequency fc which is 100 times larger than the message bandwidth, where the unit of time is
milliseconds.
(a) Sketch the magnitude spectrum of the DSB signal 10m(t) cos 2πfc t, specifying the units on
the frequency axis.
(b) Specify a time domain expression for the corresponding LSB signal.
(c) Now, suppose that the DSB signal is passed through a bandpass filter whose transfer function
is given by
1
Hp (f ) = (f − fc + )I[fc − 1 ,fc + 1 ] + I[fc + 1 ,fc + 3 ] , f > 0
2 2 2 2 2

Sketch the magnitude spectrum of the corresponding VSB signal.

(d) Find a time domain expression for the VSB signal of the form

uc (t) cos 2πfc t − us (t) sin 2πfc t

carefully specifying uc and us . the I and Q components.

Lowpass
filter

Message cos 2 π f 1t cos 2 π f 2t

Output
waveform waveform
sin 2 π f 1t sin 2 π f 2t _
+

Lowpass
filter

Figure 3.46: Block diagram of Weaver’s SSB modulator for Problem 3.7.

Problem 3.7 Figure 3.46 shows a block diagram of Weaver’s SSB modulator, which works if we
choose f1 , f2 and the bandwidth of the lowpass filter appropriately. Let us work through these
choices for a waveform of the form m(t) = AL cos(2πfL t + φL ) + AH cos(2πfH t + φH ), where
fH > fL (the design choices we obtain will work for any message whose spectrum lies in the band
[fL , fH ].
(a) For f1 = (fL + fH )/2 (i.e., choosing the first LO frequency to be in the middle of the message
band), find the time domain waveforms at the outputs of the upper and lower branches after the
first mixer.
(b) Choose the bandwidth of the lowpass filter to be W = fH +2f 2
L
(assume the lowpass filter is
ideal). Find the time domain waveforms at the outputs of the upper and lower branches after
the LPF.
(c) Now, assuming that f2 ≫ fH , find a time domain expression for the output waveform, as-
suming that the upper and lower branches are added together. Is this an LSB or USB waveform?
What is the carrier frequency?

133
(d) Repeat (c) when the lower branch is subtracted from the upper branch.
Remark: Weaver’s modulator does not require bandpass filters with sharp cutoffs, unlike the
direct approach to generating SSB waveforms by filtering DSB-SC waveforms. It is also simpler
than the Hilbert transform method (the latter requires implementation of a π/2 phase shift over
the entire message band).

Hp (f)

880 900 920 f (MHz)

Figure 3.47: Bandpass filter for Problem 3.8.

Problem 3.8 Consider the AM signal up (t) = 2(10 + cos 2πfm t) cos 2πfc t, where the message
frequency fm is 1 MHz and the carrier frequency fc is 885 MHz.
(a) Suppose that we use superheterodyne reception with an IF of 10.7 MHz, and envelope detec-
tion after the IF filter. Envelope detection is accomplished as in Figure 3.8, using a diode and
an RC circuit. What would be a good choice of C if R = 100 ohms?
(b) The AM signal up (t) is passed through the bandpass filter with transfer function Hp (f ) de-
picted (for positive frequencies) in Figure 3.47. Find the I and Q components of the filter output
with respect to reference frequency fc of 885 MHz. Does the filter output represent a form of
modulation you are familiar with?

Problem 3.9 Consider a message signal m(t) with spectrum M(f ) = I[−2,2] (f ).
(a) Sketch the spectrum of the DSB-SC signal uDSB−SC = 10m(t) cos 300πt. What is the power
and bandwidth of u?
(b) The signal in (a) is passed through an envelope detector. Sketch the output, and comment
on how it is related to the message.
(c) What is the smallest value of A such that the message can be recovered without distortion
from the AM signal uAM = (A + m(t)) cos 300πt by envelope detection?
(d) Give a time-domain expression of the form

up (t) = uc (t) cos 300πt − us (t) sin 300πt

obtained by high-pass filtering the DSB signal in (a) so as to let through only frequencies above
150 Hz.
(e) Consider a VSB signal constructed by passing the signal in (a) through a passband filter with
transfer function for positive frequencies specified by:

f − 149 149 ≤ f ≤ 151
Hp (f ) =
2 f ≥ 151

(you should be able to sketch Hp (f ) for both positive and negative frequencies.) Find a time
domain expression for the VSB signal of the form

up (t) = uc (t) cos 300πt − us (t) sin 300πt

134
Problem 3.10 Consider Figure 3.17 depicting VSB spectra. Suppose that the passband VSB
filter Hp (f ) is specified (for positive frequencies) as follows:

 1, 101 ≤ f < 102
1
Hp (f ) = (f − 99) , 99 ≤ f ≤ 101
 2
0, else

(a) Sketch the passband transfer function Hp (f ) for both positive and negative frequencies.
(b) Sketch the spectrum of the complex envelope H(f ), taking fc = 100 as a reference.
(c) Sketch the spectra (show the real and imaginary parts separately) of the I and Q components
of the impulse response of the passband filter.
(d) Consider a message signal of the form m(t) = 4sinc4t − 2 cos 2πt. Sketch the spectrum of the
DSB signal that results when the message is modulated by a carrier at fc = 100.
(e) Now, suppose that the DSB signal in (d) is passed through the VSB filter in (a)-(c). Sketch the
spectra of the I and Q components of the resulting VSB signal, showing the real and imaginary
parts separately.
(f) Find a time domain expression for the Q component.

Problem 3.11 Consider the periodic signal m(t) = ∞

P
n=−∞ p(t − 2n), where p(t) = tI[−1,1] (t).
(a) Sketch the AM signal x(t) = (4 + m(t)) cos 100πt.
(b) What is the power efficiency?

Superheterodyne reception

Problem 3.12 A dual band radio operates at 900 MHz and 1.8 GHz. The channel spacing in
each band is 1 MHz. We wish to design a superheterodyne receiver with an IF of 250 MHz. The
LO is built using a frequency synthesizer that is tunable from 1.9 to 2.25 GHz, and frequency
divider circuits if needed (assume that you can only implement frequency division by an integer).
(a) How would you design a superhet receiver to receive a passband signal restricted to the band
1800-1801 MHz? Specify the characteristics of the RF and IF filters, and how you would choose
and synthesize the LO frequency.
(b) Repeat (a) when the signal to be received lies in the band 900-901 MHz.

Angle modulation

Problem 3.13 Figure 3.48 shows, as a function of time, the phase deviation of a bandpass FM
signal modulated by a sinusoidal message.
(a) Find the modulation index (assume that it is an integer multiple of π for your estimate).
(b) Find the message bandwidth.
(c) Estimate the bandwidth of the FM signal using Carson’s formula.

Problem 3.14 The input m(t) to an FM modulator with kf = 1 has Fourier transform

j2πf |f | < 1
M(f ) =
0 else

The output of the FM modulator is given by

u(t) = A cos(2πfc t + φ(t))

135
 600

400

Phase deviation (degrees)

200

−200

−400

−600
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Time (milliseconds)

Figure 3.48: Phase deviation of FM signal for Problem 3.13.

where fc is the carrier frequency.

(a) Find an explicit time domain expression for φ(t) and carefully sketch φ(t) as a function of
time.
(b) Find the magnitude of the instantaneous frequency deviation from the carrier at time t = 41 .
(c) Using the result from (b) as an approximation for the maximum frequency deviation, estimate
the bandwidth of u(t).

Problem 3.15 Let p(t) = I[− 1 , 1 ] (t) denote a rectangular pulse of unit duration. Construct the
2 2
signal
X∞
m(t) = (−1)n p(t − n)
n=−∞

The signal m(t) is input to an FM modulator, whose output is given by

u(t) = 20 cos(2πfc t + φ(t))

where Z t
φ(t) = 20π m(τ )dτ + a
−∞

and a is chosen such that φ(0) = 0.

(a) Carefully sketch both m(t) and φ(t) as a function of time.
(b) Approximating the bandwidth of m(t) as W ≈ 2, estimate the bandwidth of u(t) using
Carson’s formula.
(c) Suppose that a very narrow ideal BPF (with bandwidth less than 0.1) is placed at fc + α.
For which (if any) of the following choices of α will you get nonzero power at the output of the
BPF: (i) α = .5, (ii) α = .75, (iii) α = 1.

Problem 3.16 Let u(t) = 20 cos(2000πt + φ(t)) denote an angle modulated signal.
(a) For φ(t) = 0.1 cos 2πt, what is the approximate bandwidth of u?
(b) Let y(t) = u12 (t). Specify the frequency bands spanned by y(t). In particular, specify the
output when y is passed through:
(i) A BPF centered at 12KHz. Using Carson’s formula, determine the bandwidth of the BPF
required to recover most of the information in φ from the output.
(ii) An ideal LPF of bandwidth 200 Hz.

136
(iii) A BPF of bandwidth
P 100 Hz centered at 11 KHz.
(c) For φ(t) = 2 n s(t − 2n), where s(t) = (1 − |t|)I[−1,1] .
(i) Sketch the instantaneous frequency deviation from the carrier frequency of 1 KHz.
(ii) Show that we can write X
u(t) = cn cos(2000πt + nαt)
n

Specify α, and write down an explicit integral expression for cn .

Problem 3.17 Consider the set-up of Problem 3.15, taking the unit of time in milliseconds for
concreteness. You do not need the value of fc , but you can take it to be 1 MHz.
(a) Numerically (e.g., using Matlab) compute the Fourier series expansion for the complex enve-
lope of the FM waveform, in the same manner as was done for a sinusoidal message. Report the
magnitudes of the Fourier series coefficients for the first 5 harmonics.
(b) Find the 90%, 95% and 99% power containment bandwidths. Compare with the estimate
from Carson’s formula obtained in Problem 3.15(b).

Problem 3.18 A VCO with a quiescent frequency of 1 GHz, with a frequency sweep of 2
MHz/mV produces an angle modulated signal whose phase deviation θ(t) from a carrier frequency
fc of 1 GHz is shown in Figure 3.49.
θ (t)

VCO 10 π
m(t) cos(2π fc t + θ (t))
2 MHz/mV

−3 −1 1 3 t(microseconds)

Figure 3.49: Set-up for Problem 3.18.

(a) Sketch the input m(t) to the VCO, carefully labeling both the voltage and time axes.
(b) Estimate the bandwidth of the angle modulated signal at the VCO output. You may ap-
proximate the bandwidth of a periodic signal by that of its first harmonic.

Uncategorized problems

Problem 3.19 The signal m(t) = 2 cos 20πt − cos 40πt, where the unit of time is millisec-
onds, and the unit of amplitude is millivolts (mV), is fed to a VCO with quiescent frequency
of 5 MHz and frequency deviation of 100 KHz/mV. Denote the output of the VCO by y(t).
(a) Provide an estimate of the bandwidth of y.
(b) The signal y(t) is passed through an ideal bandpass filter of bandwidth 5 KHz, centered at
5.005 MHz. Describe in detail how you would compute the power at the filter output (if you can
compute the power in closed form, do so).

Problem 3.20 Consider the AM signal up (t) = (A + m(t)) cos 400πt (t in ms) with message
signal m(t) as in Figure 3.50, where A is 10 mV.
(a) If the AM signal is demodulated using an envelope detector with an RC filter, how should
you choose C if R = 500 ohms? Try to ensure that the first harmonic (i.e., the fundamental)
and the third harmonic of the message are reproduced with minimal distortion.
(b) Now, consider an attempt at synchronous demodulation, where the AM signal is downcon-
verted using a 201 KHz LO, as shown in Figure 3.51, find and sketch the I and Q components,

137
 m(t)

... 10 mV

t (ms)
0 1 2 3
...
−10 mV

Figure 3.50: Message signal for Problems 3.20 and 3.21.

Lowpass u c (t)
Filter

2cos 402 π t
u p (t)
−2sin 402π t

Lowpass u s (t)
Filter

Figure 3.51: Downconversion using 201 KHz LO (t in ms in the figure) for Problem 3.20(b)-(c).

uc (t) and us (t), for 0 ≤ t ≤ 2 (t in ms).

(c) Describe how you would recover the original message m(t) from the downconverter outputs
uc (t) and us (t), drawing block diagrams as needed.

Problem 3.21 The square wave message signal m(t) in Figure 3.50 is input to a VCO with
quiescent frequency 200 KHz and frequency deviation 1 KHz/mV. Denote the output of the
VCO by up (t).
(a) Sketch the I and Q components of the FM signal (with respect to a frequency reference of
200 KHz and a phase reference chosen such that the phase is zero at time zero) over the time
interval 0 ≤ t ≤ 2 (t in ms), clearly labeling the axes.
(b) In order to extract the I and Q components using a standard downconverter (mix with LO
and then lowpass filter), how would you choose the bandwidth of the LPFs used at the mixer
outputs?

φ (t)

8π

4π

1 2 3 4 5 6 7 8 t (msec)
−4π

Figure 3.52: Phase Evolution in Problem 3.22.

Problem 3.22 The output of an FM modulator is the bandpass signal y(t) = 10 cos(300πt +
φ(t)), where the unit of time is milliseconds, and the phase φ(t) is as sketched in Figure 3.52.

138
(a) Suppose that y(t) is the output of a VCO with frequency deviation 1 KHz/mV and quiescent
frequency 149 KHz, find and sketch the input to the VCO.
(b) Use Carson’s formula to estimate the bandwidth of y(t), clearly stating the approximations
that you make.

Phase locked loop

Set-up for PLL problems: For the next few problems on PLL modeling and analysis, consider
the linearized model in Figure 3.38, with the following notation: loop filter G(s), loop gain K,
and VCO modeled as 1/s. Recall from your background on signals and systems that a second
order system of the form s2 +2ζω1n s+ω2 is said to have natural frequency ωn (in radians/second)
n
and damping factor ζ.

Problem 3.23 Let H(s) denote the gain from the PLL input to the output of the VCO. Let
He (s) denote the gain from the PLL input to the input to the loop filter. Let Hm (s) denote the
gain from the PLL input to the VCO input.
(a) Write down the formulas for H(s), He (s), Hm (s), in terms of K and G(s).
(b) Which is the relevant transfer function if the PLL is being used for FM demodulation?
(c) Which is the relevant transfer function if the PLL is being used for carrier phase tracking?
(d) For G(s) = s+8
s
and K = 2, write down expressions for H(s), He (s) and Hm (s). What is the
natural frequency and the damping factor?

Problem 3.24 Suppose the PLL input exhibits a frequency jump of 1 KHz.
(a) How would you choose the loop gain K for a first order PLL (G(s) = 1) to ensure a steady
state error of at most 5 degrees?
(b) How would you choose the parameters a and K for a second order PLL (G(s) = s+a s
) to have
1
a natural frequency of 1.414 KHz and a damping factor of 2 . Specify the units for a and K.
√

(c) For the parameter choices in (b), find and roughly sketch the phase error as a function of
time for a frequency jump of 1 KHz.

Problem 3.25 Suppose that G(s) = s+16 s+4

and K = 4.
Θo (s)
(a) Find the transfer function Θi (s) .
(b) Suppose that the PLL is used for FM demodulation, with the input to the PLL is being an
FM signal with instantaneous frequency deviation of the FM signal 10 π
m(t), where the message
m(t) = 2 cos t + sin 2t. Using the linearized model for the PLL, find a time domain expression
for the estimated message provided by the PLL-based demodulator.
Hint: What happens to a sinusoid of frequency ω passing through a linear system with transfer
function H(s)?

Problem 3.26 Consider the PLL depicted in Figure 3.53, with input phase φ(t). The output
signal of interest to us here is v(t), the VCO input. The parameter for the loop filter G(s) is
given by a = 1000π radians/sec.
(a) Assume that the PLL is locked at time 0, and suppose that φ(t) = 1000πtI{t>0} . Find the
limiting value of v(t).
(b) Now, suppose that φ(t) = 4π sin 1000πt. Find an approximate expression for v(t). For full
credit, simplify as much as possible.
(c) For part (b), estimate the bandwidth of the passband signal at the PLL input.

139
 Phase Detector
G(s) = (s+a)/s
1 volt/radian

VCO
1KHz/volt
v(t)

Figure 3.53: System for Problem 3.26.

Quiz on analog communication systems

Problem 3.27 Answer the following questions regarding commercial analog communication sys-
tems (some of which may no longer exist in your neighborhood).
(a) (True or False) The modulation format for analog cellular telephony was conventional AM.
(b) (Multiple choice) FM was used in analog TV as follows:
(i) to modulate the video signal
(ii) to modulate the audio signal
(iii) FM was not used in analog TV systems.
(c) A superheterodyne receiver for AM radio employs an intermediate frequency (IF) of 455 KHz,
and has stations spaced at 10 KHz. Comment briefly on each of the following statements:
(i) The AM band is small enough that the problem of image frequencies does not occur.
(ii) A bandwidth of 20 KHz for the RF front end is a good choice.
(iii) A bandwidth of 20 KHz for the IF filter is a good choice.

140
Chapter 4

Digital Modulation

+1 +1 +1 +1

Bit−to−Symbol Map
0 +1
Pulse ... ...
...0110100... Modulation
1 −1

−1 −1 −1

Symbol interval
T

Figure 4.1: Running example: Binary antipodal signaling using a timelimited pulse.

Digital modulation is the process of translating bits to analog waveforms that can be sent over
a physical channel. Figure 4.1 shows an example of a baseband digitally modulated waveform,
where bits that take values in {0, 1} are mapped to symbols in {+1, −1}, which are then used
to modulate translates of a rectangular pulse, where the translation corresponding to successive
symbols is the symbol interval T . The modulated waveform can be represented as a sequence of
symbols (taking values ±1 in the example) multiplying translates of a pulse (rectangular in the
example). This is an example of a widely used form of digital modulation termed linear modula-
tion, where the transmitted signal depends linearly on the symbols to be sent. Our treatment of
linear modulation in this chapter generalizes this example in several ways. The modulated signal
in Figure 4.1 is a baseband signal, but what if we are constrained to use a passband channel
(e.g., a wireless cellular system operating at 900 MHz)? One way to handle this to simply trans-
late this baseband waveform to passband by upconversion; that is, send up (t) = u(t) cos 2πfc t,
where the carrier frequency fc lies in the desired frequency band. However, what if the frequency
occupancy of the passband signal is strictly constrained? (Such constraints are often the result
of guidelines from standards or regulatory bodies, and serve to limit interference between users
operating in adjacent channels.) Clearly, the timelimited modulation pulse used in Figure 4.1
spreads out significantly in frequency. We must therefore learn to work with modulation pulses
which are better constrained in frequency. We may also wish to send information on both the
I and Q components. Finally, we may wish to pack in more bits per symbol; for example, we
could send 2 bits per symbol by using 4 levels, say {±1, ±3}.
Chapter plan: In Section 4.1, we develop an understanding of the structure of linearly mod-
ulated signals, using the binary modulation in Figure 4.1 to lead into variants of this example,
corresponding to different signaling constellations which can be used for baseband and passband
channels. In Section 4.2, we discuss how to quantify the bandwidth of linearly modulated signals
by computing the power spectral density. With these basic insights in place, we turn in Section
4.3 to a discussion of modulation for bandlimited channels, treating signaling over baseband and
passband channels in a unified framework using the complex baseband representation. We note,

141
invoking Nyquist’s sampling theorem to determine the degrees of freedom offered by bandlimited
channels, that linear modulation with a bandlimited modulation pulse can be used to fill all of
these degrees of freedom. We discuss how to design bandlimited modulation pulses based on
the Nyquist criterion for intersymbol interference (ISI) avoidance. Finally, we discuss orthogonal
and biorthogonal modulation in Section 4.4.
Software: Over the course of this and later chapters, we develop a simulation framework for
simulating linear modulation over noisy dispersive channels. Software Lab 4.1 in this chapter is
a first step in this direction. Appendix 4.B provides guidance for developing the software for this
lab.

4.1 Signal Constellations

1

0.8

0.6

0.4

0.2

−0.2

−0.4

−0.6

−0.8

−1
0 0.5 1 1.5 2 2.5 3
t/T

Figure 4.2: BPSK illustrated for fc = T4 and symbol sequence +1, −1, −1. The solid line corre-
sponds to the passband signal up (t), and the dashed line to the baseband signal u(t). Note that,
due to the change in sign between the first and second symbols, there is a phase discontinuity of
π at t = T .

The linearly modulated signal depicted in Figure 4.1 can be written in the following general
form: X
u(t) = b[n]p(t − nT ) (4.1)
n

where {b[n]} is a sequence of symbols, and p(t) is the modulating pulse. The symbols take values
in {−1, +1} in our example, and the modulating pulse is a rectangular timelimited pulse. As we
proceed along this chapter, we shall see that linear modulation as in (4.1) is far more generally
applicable, in terms of the set of possible values taken by the symbol sequence, as well as the
choice of modulating pulse.
The modulated waveform (4.1) is a baseband waveform. While it is timelimited in our example,
and hence cannot be strictly bandlimited, it is approximately bandlimited to a band around DC.
Now, if we are given a passband channel over which to send the information encoded in this
waveform, one easy approach is to send the passband signal
up (t) = u(t) cos 2πfc t (4.2)
where fc is the carrier frequency. That is, the modulated baseband signal is sent as the I
component of the passband signal. To see what happens to the passband signal as a consequence
of the modulation, we plot it in Figure 4.2. For the nth symbol interval nT ≤ t < (n + 1)T , we
have up (t) = cos 2πfc t if b[n] = +1, and up (t) = − cos 2πfc t = cos(2πfc t + π) if b[n] = −1. Thus,
binary antipodal modulation switches the phase of the carrier between two values 0 and π, which
is why it is termed Binary Phase Shift Keying (BPSK) when applied to a passband channel:

142
We know from Chapter 2 that any passband signal can be represented in terms of two real-valued
baseband waveforms, the I and Q components.
up (t) = uc (t) cos 2πfc t − us (t) sin 2πfc t
The complex envelope of up (t) is given by u(t) = uc (t) + jus (t). For BPSK, the I component is
modulated using binary antipodal signaling, while the Q component is not used, so that u(t) =
uc (t). However, noting that the two signals, uc (t) cos 2πfc t and us (t) sin 2πfc t are orthogonal
regardless of the choice of uc and us , we realize that we can modulate both I and Q components
independently, without affecting their orthogonality. In this case, we have
X X
uc (t) = bc [n]p(t − nT ), us (t) = bs [n]p(t − nT )
n n

The complex envelope is given by

X X
u(t) = uc (t) + jus (t) = (bc [n] + jbs [n]) p(t − nT ) = b[ n]p(t − nT ) (4.3)
n n

where {b[n] = bc [n] + jbs [n]} are complex-valued symbols.

1.5

0.5

−0.5

−1

−1.5
0 0.5 1 1.5 2 2.5 3
t/T

Figure 4.3: QPSK illustrated for fc = T4 , with symbol sequences {bc [n]} = {+1, −1, −1} and
{bs [n]} = {−1, +1, −1}. The phase of the passband signal is −π/4 in the first symbol interval,
switches to 3π/4 in the second, and to −3π/4 in the third.

Let us see what happens to the passband signal when bc [n], bs [n] each take values in {±1 ± j}.
For the nth symbol interval nTò t < (n + 1)T :
up (t) = cos 2πfc t − sin 2πfc t = √2 cos (2πfc t + π/4) if bc [n] = +1, bs [n] = +1;
up (t) = cos 2πfc t + sin 2πfc t = 2√cos (2πfc t − π/4) if bc [n] = +1, bs [n] = −1;
up (t) = − cos 2πfc t − sin 2πfc t = √2 cos (2πfc t + 3π/4) if bc [n] = −1, bs [n] = +1;
up (t) = − cos 2πfc t + sin 2πfc t = 2 cos (2πfc t − 3π/4) if bc [n] = −1, bs [n] = −1.
Thus, the modulation causes the passband signal to switch its phase among four possibilities,
{±π/4, ±3π/4}, as illustrated in Figure 4.3, which is why we call it Quadrature Phase Shift
Keying (QPSK).
Equivalently, we could have √ seen this from the complex envelope. Note that the QPSK symbols
jθ[n]
can be written as b[n] = 2e , where θ[n] ∈ {±π/4, ±3π/4}. Thus, over the nth symbol, we
have
√  √
j2πfc t jθ[n] j2πfc t


up (t) = Re b[n]e = Re 2e e = 2 cos (2πfc t + θ[n]) , nT ≤ t < (n + 1)T

This indicates that it is actually easier to figure out what is happening to the passband signal
by working with the complex envelope. We therefore work in the complex baseband domain for
the remainder of this chapter.

143
In general, the complex envelope for a linearly modulated signal is given by (4.1), where b[n] =
bc [n] + jbs [n] = r[n]ejθ[n] can be complex-valued. We can view this as bc [n] modulating the
I component and bs [n] modulating the Q component, or as scaling the envelope by r[n] and
switching the phase by θ[n]. The set of values that each symbol can take is called the signaling
alphabet, or constellation. We can plot the constellation in a two-dimensional plot, with the x-
axis denoting the real part bc [n] (corresponding to the I component) and the y-axis denoting the
imaginary part bs [n] (corresponding to the Q component). Indeed, this is why linear modulation
over passband channels is also termed two-dimensional modulation. Note that this provides a
unified description of constellations that can be used over both baseband and passband channels:
for physical baseband channels, we simply constrain b[n] = bc [n] to be real-valued, setting bs [n] =
0.

BPSK/2PAM
4PAM

16QAM

QPSK/4PSK/4QAM 8PSK

Figure 4.4: Some commonly used constellations. Note that 2PAM and 4PAM can be used over
both baseband and passband channels, while the two-dimensional constellations QPSK, 8PSK
and 16QAM are for use over passband channels.

Figure 4.4 shows some common constellations. Pulse Amplitude Modulation (PAM) corresponds
to using multiple amplitude levels along the I component (setting the Q component to zero).
This is often used for signaling over physical baseband channels. Using PAM along both I and Q
axes corresponds to Quadrature Amplitude Modulation (QAM). If the constellation points lie on
a circle, they only affect the phase of the carrier: such signaling schemes are termed Phase Shift
Keying (PSK). When naming a modulation scheme, we usually indicate the number of points
in the constellations. BPSK and QPSK are special: BPSK (or 2PSK) can also be classified as
2PAM, while QPSK (or 4PSK) can also be classified as 4QAM.
Each symbol in a constellation of size M can be uniquely mapped to log2 M bits. For a symbol
rate of 1/T symbols per unit time, the bit rate is therefore logT2 M bits per unit time. Since the
transmitted bits often contain redundancy due to a channel code employed for error correction or
detection, the information rate is typically smaller than the bit rate. The choice of constellation
for a particular application depends on considerations such as power-bandwidth tradeoffs and
implementation complexity. We shall discuss these issues once we develop more background.

144
4.2 Bandwidth Occupancy
Bandwidth is a precious commodity, hence it is important to quantify the frequency occupancy
of communication signals. To this end, consider the complex envelope of a linearly modulated
signal (the two-sided bandwidth of this complex envelope equals the physicalP bandwidth of the
corresponding passband signal), which has the form given in (4.1): u(t) = n b[n]p(t − nT ).
The complex-valued symbol sequence {b[n]} is modeled as random. Modeling the sequence as
random at the transmitter makes sense because the latter does not control the information being
sent (e.g., it depends on the specific computer file or digital audio signal being sent). Since this
information is mapped to the symbols in some fashion, it follows that the symbols themselves are
also random rather than deterministic. Modeling the symbols as random at the receiver makes
even more sense, since the receiver by definition does not know the symbol sequence (otherwise
there would be no need to transmit). However, for characterizing the bandwidth occupancy of the
digitally modulated signal u, we do not compute statistics across different possible realizations
of the symbol sequence {b[n]}. Rather, we define the quantities of interest in terms of averages
across time, treating u(t) as a finite power signal which can be modeled as deterministic once the
symbol sequence {b[n]} is fixed. (We discuss concepts of statistical averaging across realizations
later, when we discuss random processes in Chapter 5.)
We introduce the concept of PSD in Section 4.2.1. In Section 4.2.2, we state our main result on
the PSD of digitally modulated signals, and discuss how to compute bandwidth once we know
the PSD.

4.2.1 Power Spectral Density

H(f)
∆f
1
Power Sx ( f*) ∆ f
x(t)
Meter

f*

Figure 4.5: Operational definition of PSD.

We now introduce the important concept of power spectral density (PSD), which specifies how
the power in a signal is distributed in different frequency bands.
Power Spectral Density: The power spectral density (PSD), Sx (f ), for a finite-power signal
x(t) is defined through the conceptual measurement depicted in Figure 4.5. Pass x(t) through
an ideal narrowband filter with transfer function

1, f ∗ − ∆f < f < f ∗ + ∆f

Hf ∗ (f ) = 2 2
0, else

The PSD evaluated at f ∗ , Sx (f ∗ ), is defined as the measured power at the filter output, divided
by the filter width ∆f (in the limit as ∆f → 0).
Example (PSD of complex exponentials): Let us now find the PSD of x(t) = Aej(2πf0 t+θ) .
Since the frequency content of x is concentrated at f0 , the power meter in Figure 4.5 will have
zero output for f ∗ 6= f0 (as ∆f → 0, f0 falls outside the filter bandwidth for any such f0 ). Thus,

145
Sx (f ) = 0 for f 6= f0 . On the other hand, for f ∗ = f0 , the output of the power meter is the
entire power of x, which is
Z f0 + ∆f
2
2
Px = A = Sx (f )df
f0 − ∆f
2

We conclude that the PSD is Sx (f ) = A2 δ(f −f0 ). Extending this reasoning to a sum of complex
exponentials, we have
X X
PSD of Ai ej(2πfi t+θi ) = A2i δ(f − fi )
i i

where fi are distinct frequencies (positive or negative), and Ai , θi are the amplitude and phase,
respectively, of the ith complex exponential. Thus, for a real-valued sinusoid, we obtain
1 1 1 1
Sx (f ) = δ(f − f0 ) + δ(f + f0 ) , for x(t) = cos(2πf0 t + θ) = ej(2πf0 t+θ) + e−j(2πf0 t+θ) (4.4)
4 4 2 2

Periodogram-based PSD estimation: One way to carry out the conceptual measurement in
Figure 4.5 is to limit x(t) to a finite observation interval, compute its Fourier transform and hence
its energy spectral density (which is the magnitude square of the Fourier transform), and then
divide by the length of the observation interval. The PSD is obtained by letting the observation
interval get large. Specifically, define the time-windowed version of x as
xTo (t) = x(t)I[− To , To ] (t) (4.5)
2 2

where To is the length of the observation interval. Since To is finite and x(t) has finite power,
xTo (t) has finite energy, and we can compute its Fourier transform
XTo (f ) = F (xTo )

The energy spectral density of xTo is given by |XTo (f )|2 . Averaging this over the observation
interval, we obtain the estimated PSD
|XTo (f )|2
Ŝx (f ) = (4.6)
To
The estimate in (4.6), which is termed a periodogram, can typically be obtained by taking the
DFT of a sampled version of the time windowed signal; the time interval To must be large enough
to give the desired frequency resolution, while the sampling rate must be large enough to capture
the variations in x(t). The estimated PSDs obtained over multiple observation intervals can then
be averaged further to get smoother estimates.
Formally, we can define the PSD in the limit of large time windows as follows:
|XTo (f )|2
Sx (f ) = lim (4.7)
To →∞ To

Units for PSD: Power per unit frequency has the same units as power multiplied by time, or
energy. Thus, the PSD is expressed in units of Watts/Hertz, or Joules.
Power in terms of PSD: The power Px of a finite power signal x is given by integrating its
PSD: Z ∞
Px = Sx (f )df (4.8)
−∞

146
4.2.2 PSD of a linearly modulated signal
P
We are now ready to state our result on the PSD of a linearly modulated signal u(t) = n b[n]p(t−
nT ). While we derive a more general result in Appendix 4.A, our result here applies to the fol-
lowing important special case:
(a) the symbols have zero DC value: limN →∞ 2N1+1 N
P
n=−N b[n] = 0; and
1
PN
(b) the symbols are uncorrelated: limN →∞ 2N n=−N b[n]b∗ [n − k] = 0 for k 6= 0.

Theorem 4.2.1 (PSD of a linearly modulated signal) Consider a linearly modulated signal
X
u(t) = b[n]p(t − nT )
n

where the symbol sequence {b[n]} is zero mean and uncorrelated with average symbol energy

N
1 X
|b[n]|2 = lim |b[n]|2 = σb2
N →∞ 2N + 1
n=−N

Then the PSD is given by

|P (f )|2 2
Su (f ) = σb (4.9)
T
and the power of the modulated signal is

σb2 ||p||2
Pu = (4.10)
T
where ||p||2 denotes the energy of the modulating pulse.

See Appendix 4.A for a proof of (4.9), which follows from specializing a more general expression.
The expression for power follows from integrating the PSD:
∞ ∞ ∞
σ2 σ2 σb2 ||p||2
Z Z Z
Pu = Su (f )df = b |P (f )| df = b
2
|p(t)|2 dt =
−∞ T −∞ T −∞ T

where we have used Parseval’s identity.

An intuitive interpretation of this theorem is as follows. Every T time units, we send a pulse
of the form b[n]p(t − nT ) with average energy spectral density σb2 |P (f )|2, so that the PSD is
obtained by dividing this by T . The same reasoning applies to the expression for power: every
T time units, we send a pulse b[n]p(t − nT ) with average energy σb2 ||p||2, so that the power is
obtained by dividing by T . The preceding intuition does not apply when successive symbols are
correlated, in which case we get the more complicated expression (4.32) for the PSD in Appendix
4.A.
Once we know the PSD, we can define the bandwidth of u in a number of ways.
3 dB bandwidth: For symmetric Su (f ) with a maximum at f = 0, the 3 dB bandwidth B3dB
is defined by Su (B3dB /2) = Su (−B3dB /2) = 21 Su (0). That is, the 3 dB bandwidth is the size
of the interval between the points at which the PSD is 3 dB, or a factor of 12 , smaller than its
maximum value.
Fractional power containment bandwidth. This is the size of the smallest interval that

147
contains a given fraction of the power. For example, for symmetric Su (f ), the 99% fractional
power containment bandwidth B is defined by
Z B/2 Z ∞
Su (f )df = 0.99Pu = 0.99 Su (f )df
−B/2 −∞

(replace 0.99 in the preceding equation by any desired fraction γ to get the corresponding γ
power containment bandwidth).
Time/frequency normalization: Before we discuss examples in detail, let us simplify our
life by making a simple observation on time and frequency scaling. Suppose we have a linearly
modulated system operating at a symbol rate of 1/T , as in (4.1). We can think of it as a
normalized system operating at a symbol rate of one, where the unit of time is T . This implies
that the unit of frequency is 1/T . In terms of these new units, we can write the linearly modulated
signal as X
u1 (t) = b[n]p1 (t − n)
n

where p1 (t) is the modulation pulse for the normalized system. For example, for a rectangular
pulse timelimited to the symbol interval, we have p1 (t) = I[0,1] (t). Suppose now that the band-
width of the normalized system (computed using any definition that we please) is B1 . Since
the unit of frequency is 1/T , the bandwidth in the original system is B1 /T . Thus, in terms of
determining frequency occupancy, we can work, without loss of generality, with the normalized
system. In the original system, what we are really doing is working with the normalized time
t/T and the normalized frequency f T .
1
rect. pulse
sine pulse
0.9

0.8

0.7

0.6

0.5

0.4

0.3

0.2

0.1

0
−5 −4 −3 −2 −1 0 1 2 3 4 5
fT

Figure 4.6: PSD corresponding to rectangular and sine timelimited pulses. The main lobe of the
PSD is broader for the sine pulse, but its 99% power containment bandwidth is much smaller.

Rectangular pulse: Without loss of generality, consider a normalized system with p1 (t) =
I[0,1] (t), for which P1 (f ) = sinc(f )e−jπf . For {b[n]} i.i.d., taking values ±1 with equal probability,
we have σb2 = 1. Applying (4.9), we obtain

Su1 (f ) = σb2 sinc2 (f ) (4.11)

Integrating, or applying (4.10), we obtain Pu = σb2 . The scale factor of σb2 is not important, since
it drops out for any definition of bandwidth. We therefore set it to σb2 = 1. The PSD for the
rectangular pulse, along with that for a sine pulse introduced shortly, is plotted in Figure 4.6.

148
Note that the PSD for the rectangular pulse has much fatter tails, which does not bode well for
its bandwidth efficiency. For fractional power containment bandwidth with fraction γ, we have
the equation
Z B1 /2 Z ∞ Z 1
2 2
sinc f df = γ sinc f df = γ 12 dt = γ
−B1 /2 −∞ 0

using Parseval’s identity. We therefore obtain, using the symmetry of the PSD, that the band-
width is the numerical solution to the equation
Z B1 /2
sinc2 f df = γ/2 (4.12)
0

For example, for γ = 0.99, we obtain B1 = 10.2, while for γ = 0.9, we obtain B1 = 0.85.
Thus, if we wish to be strict about power containment (e.g., in order to limit adjacent channel
interference in wireless systems), the rectangular timelimited pulse is a very poor choice. On the
other hand, in systems where interference or regulation are not significant issues (e.g., low-cost
wired systems), this pulse may be a good choice because of its ease of implementation using
digital logic.

Example 4.2.1 (Bandwidth computation): A passband system operating at a carrier fre-

quency of 2.4 GHz at a bit rate of 20 Mbps. A rectangular modulation pulse timelimited to the
symbol interval is employed.
(a) Find the 99% and 90% power containment bandwidths if the constellation used is 16-QAM.
(b) Find the 99% and 90% power containment bandwidths if the constellation used is QPSK.
Solution:
(a) The 16-QAM system sends 4 bits/symbol, so that the symbol rate 1/T equals 420bits/symbol
Mbits/sec
=5
Msymbols/sec. Since the 99% power containment bandwidth for the normalized system is
B1 = 10.2, the required bandwidth is B1 /T = 51 MHz. Since the 90% power containment
for the normalized system is B1 = 0.85, the required bandwidth B1 /T equals 4.25 MHz.
(b) The QPSK system sends 2 bits/symbol, so that the symbol rate is 10 Msymbols/sec. The
bandwidths required are therefore double those in (a): the 99% power containment bandwidth
is 102 MHz, while the 90% power containment bandwidth is 8.5 MHz.
Clearly, when the criterion for defining bandwidth is the same, then 16-QAM consumes half the
bandwidth compared to QPSK for a fixed bit rate. However, it is interesting to note that, for
the rectangular timelimited pulse, a QPSK system where we are sloppy about power leakage
(90% power containment bandwidth of 8.5 MHz) can require far less bandwidth than a system
using a more bandwidth-efficient 16-QAM constellation where we are strict about power leakage
(99% power containment bandwidth of 51 MHz). This extreme variation of bandwidth when we
tweak definitions slightly is because of the poor frequency domain containment of the rectangular
timelimited pulse. Thus, if we are serious about limiting frequency occupancy, we need to think
about more sophisticated designs for the modulation pulse.

Smoothing out the rectangular pulse: A useful alternative to using the rectangular pulse,
while still keeping the modulating pulse timelimited to a symbol interval, is the sine pulse, which
for the normalized system equals
√
p1 (t) = 2 sin(πt) I[0,1] (t)

Since the sine pulse does not have the sharp edges of the rectangular pulse in the time domain,
we expect it to be more compact in the frequency domain. Note that we have normalized the
pulse to have unit energy, as we did for the normalized rectangular pulse. This implies that the
power of the modulated signal is the same in the two cases, so that we can compare PSDs under

149
the constraint that the area under the PSDs remains constant. Setting σb2 = 1 and using (4.9),
we obtain (see Problem 4.1):

8 cos2 πf
Su1 (f ) = |P1 (f )|2 = (4.13)
π 2 (1 − 4f 2 )2

Proceeding as we did for obtaining (4.12), the fractional power containment bandwidth for frac-
tion γ is given by the formula:
B1 /2
8 cos2 πf
Z
df = γ/2 (4.14)
0 π 2 (1 − 4f 2 )2

For γ = 0.99, we obtain B1 = 1.2, which is an order of magnitude improvement over the
corresponding value of B1 = 10.2 for the rectangular pulse.
While the sine pulse has better frequency domain containment than the rectangular pulse, it is
still not suitable for strictly bandlimited channels. We discuss pulse design for such channels
next.

4.3 Design for Bandlimited Channels

Suppose that you are told to design your digital communication system so that the transmitted
signal fits between 2.39 and 2.41 GHz; that is, you are given a passband channel of bandwidth 20
MHz at a carrier frequency of 2.4 GHz. Any signal that you transmit over this band has a complex
envelope with respect to 2.4 GHz that occupies a band from -10 MHz to 10 MHz. Similarly, the
passband channel (modeled as an LTI system) has an impulse response whose complex envelope is
bandlimited from -10 MHz to 10 MHz. In general, for a passband channel or signal of bandwidth
W , with an appropriate choice of reference frequency, we have a corresponding complex baseband
signal spanning the band [−W/2, W/2]. Thus, we restrict our design to the complex baseband
domain, with the understanding that the designs can be translated to passband channels by
upconversion of the I and Q components at the transmitter, and downconversion at the receiver.
Also, note that the designs specialize to physical baseband channels if we restrict the baseband
signals to be real-valued.

4.3.1 Nyquist’s Sampling Theorem and the Sinc Pulse

Our first step in understanding communication system design for such a bandlimited channel is
to understand the structure of bandlimited signals. To this end, suppose that the signal s(t) is
bandlimited to [−W/2, W/2]. We can now invoke Nyquist’s sampling theorem (proof postponed
to Section 4.5) to express the signal in terms of its samples at rate W .

Theorem 4.3.1 (Nyquist’s sampling theorem) Any signal s(t) bandlimited to [− W2 , W2 ] can
be described completely by its samples {s( Wn )} at rate W . The signal s(t) can be recovered from
its samples using the following interpolation formula:
∞ n 
X n
s(t) = s p t− (4.15)
n=−∞
W W

where p(t) = sinc(W t).

150
Degrees of freedom: What does the sampling theorem tell us about digital modulation? The
interpolation formula (4.15) tells us that we can interpret s(t) as a linearly modulated signal
with symbol sequence equal to the samples {s(n/W )}, symbol rate 1/T equal to the bandwidth
W , and modulation pulse given by p(t) = sinc(W t) ↔ P (f ) = W1 I[−W/2,W/2] (f ). Thus, linear
modulation with the sinc pulse is able to exploit all the “degrees of freedom” available in a
bandlimited channel.
Signal space: If we signal over an observation interval of length To using linear modulation
according to the interpolation formula (4.15), then we have approximately W To complex-valued
samples. Thus, while the signals we send are continuous-time signals, which in general, lie in an
infinite-dimensional space, the set of possible signals we can send in a finite observation interval
of length To live in a complex-valued vector space of finite dimension W To , or equivalently, a
real-valued vector space of dimension 2W To . Such geometric views of communication signals as
vectors, often termed signal space concepts, are particularly useful in design and analysis, as we
explore in more detail in Chapter 6.

0.8

0.6

0.4

0.2

−0.2

−0.4

−0.6

−0.8

−1
−10 −5 0 5 10 15
t/T

Figure 4.7: Three successive sinc pulses (each pulse is truncated to a length of 10 symbol intervals
on each side) modulated by +1,-1,+1. The actual transmitted signal is the sum of these pulses
(not shown). Note that, while the pulses overlap, the samples at t = 0, T, 2T are equal to the
transmitted bits because only one pulse is nonzero at these times.

The concept of Nyquist signaling: Since the sinc pulse is not timelimited to a symbol interval,
in principle, the symbols could interfere with each other. The time domain signal corresponding
to a bandlimited modulation pulse such as the sinc spans an interval significantly larger than the
symbol interval (in theory, the interval is infinitely large, but we always truncate the waveform
in implementations). This means that successive pulses corresponding to successive symbols
which are spaced by the symbol interval (i.e., b[n]p(t − nT ) as we increment n) overlap with,
and therefore can interfere with, each other. Figure 4.7 shows the sinc pulse modulated by three
bits, +1,-1,+1. While the pulses corresponding to the three symbols do overlap, notice that, by
sampling at t = 0, t = T and t = 2T , we can recover the three symbols because exactly one of the
pulses is nonzero at each of these times. That is, at sampling times spaced by integer multiples of
the symbol time T , there is no intersymbol interference. We call such a pulse Nyquist for signaling
at rate T1 , and we discuss other examples of such pulses soon. Designing pulses based on the
Nyquist criterion allows us the freedom to expand the modulation pulses in time beyond the
symbol interval (thus enabling better containment in the frequency domain), while ensuring that
there is no ISI at appropriately chosen sampling times despite the significant overlap between
successive pulses.

151
 1.5

0.5

−0.5

−1

−1.5
−10 −5 0 5 10 15 20
t/T

Figure 4.8: The baseband signal for 10 BPSK symbols of alternating signs, modulated using the
sinc pulse. The first symbol is +1, and the sample at time t = 0, marked with ’x’, equals +1, as
desired (no ISI). However, if the sampling time is off by 0.25T , the sample value, marked by ’+’,
becomes much smaller because of ISI. While it still has the right sign, the ISI causes it to have
significantly smaller noise immunity. See Problem 4.14 for an example in which the ISI due to
timing mismatch actually causes the sign to flip.

152
The problem with sinc: Are we done then? Should we just use linear modulation with a sinc
pulse when confronted with a bandlimited channel? Unfortunately, the answer is no: just as the
rectangular timelimited pulse decays too slowly in frequency, the rectangular bandlimited pulse,
corresponding to the sinc pulse in the time domain, decays too slowly in time. Let us see what
happens as a consequence. Figure 4.8 shows a plot of the modulated waveform for a bit sequence
of alternating sign. At the correct sampling times, there is no ISI. However, if we consider a small
timing error of 0.25T , the ISI causes the sample value to drop drastically, making the system
more vulnerable to noise. What is happening is that, when there is a small sampling offset,
we can make the ISI add up to a large value by choosing the interfering symbols so that their
contributions all have signs opposite to that of the desired symbol at the sampling time. Since
the sinc pulse decays as 1/t, the ISI created for a given symbol by an interfering symbol which
is n symbol intervals away decays as 1/n, soP that, in the worst-case, the contributions from the
interfering symbols roughly have the form n n1 , a series that is known to diverge. Thus, in
theory, if we do not truncate the sinc pulse, we can make the ISI arbitrarily large when there is
a small timing offset. In practice, we do truncate the modulation pulse, so that we only see ISI
from a finite number of symbols. However, even when we do truncate, as we see from Figure 4.8,
the slow decay of the sinc pulse means that the ISI adds up quickly, and significantly reduces
the margin of error when noise is introduced into the system.
While the sinc pulse may not be a good idea in practice, the idea of using bandwidth-efficient
Nyquist pulses is a good one, and we now develop it further.

4.3.2 Nyquist Criterion for ISI Avoidance

Nyquist signaling: Consider a linearly modulated signal
X
u(t) = b[n]p(t − nT )
n

We say that the pulse p(t) is Nyquist (or satisfies the Nyquist criterion) for signaling at rate T1
if the symbol-spaced samples of the modulated signal are equal to the symbols (or a fixed scalar
multiple of the symbols); that is, u(kT ) = b[k] for all k. That is, there is no ISI at appropriately
chosen sampling times spaced by the symbol interval.
In the time domain,
Pit is quite easy to see what is required to satisfy the Nyquist criterion. The
samples u(kT ) = n b[n]p(kT − nT ) = b[k] (or a scalar multiple of b[k]) for all k if and only
if p(0) = 1 (or some nonzero constant) and p(mT ) = 0 for all integers m 6= 0. However, for
design of bandwidth efficient pulses, it is important to characterize the Nyquist criterion in the
frequency domain. This is given by the following theorem.

Theorem 4.3.2 (Nyquist criterion for ISI avoidance): The pulse p(t) ↔ P (f ) is Nyquist
for signaling at rate T1 if 
1 m=0
p(mT ) = δm0 = (4.16)
0 m 6= 0
or equivalently,
∞
1 X k
P (f + ) = 1 for all f (4.17)
T T
k=−∞

The proof of this theorem is given in Section 4.5, where we show that both the Nyquist sampling
theorem, Theorem 4.3.1, and the preceding theorem are based on the same mathematical result,
that the samples of a time domain signal have a one-to-one mapping with the sum of translated
(or aliased) versions of its Fourier transform.

153
In this section, we explore the design implications of Theorem 4.3.2. In the frequency domain,
the translates of P (f ) by integer multiples of 1/T must add up to a constant. As illustrated by
Figure 4.9, the minimum bandwidth pulse for which this happens is the ideal bandlimited pulse
over an interval of length 1/T .
Not Nyquist Nyquist with minimum bandwidth

P(f + 1/T) P(f) P(f − 1/T) P(f + 1/T) P(f) P(f − 1/T)

... ... ... ...

f f
1/T 1/T

Figure 4.9: The minimum bandwidth Nyquist pulse is a sinc.

Minimum bandwidth Nyquist pulse: The minimum bandwidth Nyquist pulse is

1

T, |f | ≤ 2T
P (f ) =
0, else
corresponding to the time domain pulse
p(t) = sinc(t/T )

As we have already discussed, the sinc pulse is not a good choice in practice because of its slow
decay in time. To speed up the decay in time, we must expand in the frequency domain, while
conforming to the Nyquist criterion. The trapezoidal pulse depicted in Figure 4.9 is an example
of such a pulse.

P(f + 1/T) T P(f) P(f − 1/T)

f
−(1+a)/(2T) −(1−a)/(2T) (1−a)/(2T) (1+a)/(2T)

Figure 4.10: A trapezoidal pulse which is Nyquist at rate 1/T . The (fractional) excess bandwidth
is a.

The role of excess bandwidth: We have noted earlier that the problem P∞with the sinc pulse
1
arises because of its 1/t decay and the divergence of the harmonic series n=1 n , which implies
that the worst-case contribution from “distant” interfering symbols at a given sampling instant
can blow up. Using thePsame reasoning, however, a pulse p(t) decaying as 1/tb for b > 1 should
work, since the series ∞ 1
n=1 nb does converge for b > 1. A faster time decay requires a slower
decay in frequency. Thus, we need excess bandwidth, beyond the minimum bandwidth dictated
by the Nyquist criterion, to fix the problems associated with the sinc pulse. The (fractional)
excess bandwidth for a linear modulation scheme is defined to be the fraction of bandwidth
over the minimum required for ISI avoidance at a given symbol rate. In particular, Figure 4.10
shows that a trapezoidal pulse (in the frequency domain) can be Nyquist for suitably chosen
parameters, since the translates {P (f + k/T )} as shown in the figure add up to a constant. Since
trapezoidal P (f ) is the convolution of two boxes in the frequency domain, the time domain pulse
p(t) is the product of two sinc functions, as worked out in the example below. Since each sinc
decays as 1/t, the product decays as 1/t2 , which implies that the worst-case ISI with timing
mismatch is indeed bounded.

154
Example 4.3.1 Consider the trapezoidal pulse of excess bandwidth a shown in Figure 4.10.
(a) Find an explicit expression for the time domain pulse p(t).
(b) What is the bandwidth required for a passband system using this pulse operating at 120
Mbps using 64QAM, with an excess bandwidth of 25%?
Solution: (a) It is easy to check that the trapezoid is a convolution of two boxes as follows (we
assume 0 < a ≤ 1):
T2
P (f ) = I 1 1 (f ) ∗ I[− 2Ta , 2Ta ] (f )
a [− 2T , 2T ]
Taking inverse Fourier transforms, we obtain
T2 1
 
a 
p(t) = sinc(t/T ) sinc(at/T ) = sinc(t/T )sinc(at/T ) (4.18)
a T T
The presence of the first sinc provides the zeroes required by the time domain Nyquist criterion:
p(mT ) = 0 for nonzero integers m 6= 0. The presence of a second sinc yields a 1/t2 decay,
providing robustness against timing mismatch.
(b) Since 64 = 26 , the use of 64QAM corresponding to sending 6 bits/symbol, so that the symbol
rate is 120/6 = 20 Msymbols/sec. The minimum bandwidth required is therefore 20 MHz, so
that 25% excess bandwidth corresponds to a bandwidth of 20 × 1.25 = 25 MHz.

Raised cosine pulse: Replacing the straight line of the trapezoid with a smoother cosine-
shaped curve in the frequency domain gives us the raised cosine pulse shown in Figure 4.12,
which has a faster, 1/t3 , decay in the time domain.

|f | ≤ 1−a

 T, 2T
T
P (f ) = 2
[1 + cos((|f | − 1−a
2T
) πT
a
)], 1−a
2T
≤ |f | ≤ 1+a
2T
0, |f | > 1+a

2T

where a is the fractional excess bandwidth, typically chosen in the range where 0 ≤ a < 1. As
shown in Problem 4.11, the time domain pulse s(t) is given by

t cos πa Tt
p(t) = sinc( )
T 1 − 2at 2


T

This pulse inherits the Nyquist property of the sinc pulse, while having an additional multiplica-
tive factor that gives an overall 1/t3 decay with time. The faster time decay compared to the
sinc pulse is evident from a comparison of Figures 4.12(b) and 4.11(b).

4.3.3 Bandwidth efficiency

We define the bandwidth efficiency of linear modulation with an M-ary alphabet as
ηB = log2 M bits/symbol
The Nyquist criterion for ISI avoidance says that the minimum bandwidth required for ISI-free
transmission using linear modulation equals the symbol rate, using the sinc as the modulation
pulse. For such an idealized system, we can think of ηB as bits/second per Hertz, since the symbol
rate equals the bandwidth. Thus, knowing the bit rate Rb and the bandwidth efficiency ηB of
the modulation scheme, we can determine the symbol rate, and hence the minimum required
bandwidth Bmin . as follows:
Rb
Bmin =
ηB

155
 1

0.8

0.6

0.4

0.2
X(f)

T 0

−0.2

fT −0.4
−1/2 0 1/2 −5 −4 −3 −2 −1 0 1 2 3 4 5
t/T

(a) Frequency domain boxcar (b) Time domain sinc pulse

Figure 4.11: Sinc pulse for minimum bandwidth ISI-free signaling at rate 1/T . Both time and
frequency axes are normalized to be dimensionless.

0.8

0.6

0.4

X(f)
0.2
T

0
T/2

fT −0.2
−(1+a)/2 −1/2 −(1−a)/2 0 (1−a)/2 1/2 (1+a)/2 −5 −4 −3 −2 −1 0 1 2 3 4 5
t/T

(a) Frequency domain raised cosine (b) Time domain pulse (excess bandwidth a = 0.5)

Figure 4.12: Raised cosine pulse for minimum bandwidth ISI-free signaling at rate 1/T , with
excess bandwidth a. Both time and frequency axes are normalized to be dimensionless.

156
This bandwidth would then be expanded by the excess bandwidth used in the modulating pulse.
However, this is not included in our definition of bandwidth efficiency, because excess bandwidth
is a highly variable quantity dictated by a variety of implementation considerations. Once we
decide on the fractional excess bandwidth a, the actual bandwidth required is

Rb
B = (1 + a)Bmin = (1 + a)
ηB

4.3.4 Power-bandwidth tradeoffs: a sneak preview

Clearly, we can increase bandwidth efficiency simply by increasing M, the constellation size.
For example, the bandwidth efficiency of QPSK is 2 bits/symbol, while that of 16QAM is 4
bits/symbol. What stops us from increasing constellation size, and hence bandwidth efficiency,
indefinitely is noise, and the fact that we cannot use arbitrarily large transmit power (typically
limited by cost or physical and regulatory constraints) to overcome it. Noise in digital communi-
cation systems must be modeled statistically, hence rigorous discussion of a formal model and its
design consequences is postponed to Chapters 5 and 6. However, that does not prevent us from
giving a handwaving sneak preview of the bottomline here. Note that this subsection is meant
as a teaser: it can be safely skipped, since these issues are covered in detail in Chapter 6.

dmin
dmin 2
Scale up
by factor
1 Es
Es
of two
−1 1 −2 2
−1

−2

Figure 4.13: Scaling of minimum distance and energy per symbol.

Intuitively speaking, the effect of noise is to perturb constellation points from the nominal loca-
tions shown in Figure 4.4, which leads to the possibility of making an error in deciding which
point was transmitted. For a given noise “strength” (which determines how much movement the
noise can produce), the closer the constellation points, the more the possibility of such errors.
In particular, as we shall see in Chapter 6, the minimum distance between constellation points,
termed dmin , provide a good measure of how vulnerable we are to noise. For a given constellation
shape, we can increase dmin simply by scaling up the constellation, as shown in Figure 4.13, but
this comes with a corresponding increase in energy expenditure. To quantify this, define the
energy per symbol Es for a constellation as the average of the squared Euclidean distances of the
points from the origin. For an M-ary constellation, each symbol carries log2 M bits of informa-
tion, and we can define the average energy per bit Eb as Eb = logEsM . Specifically, dmin increases
2
from 2 to 4 by scaling as shown in Figure 4.13. Correspondingly, Es = 2 and Eb = 1 is increased
to Es = 8 and Eb = 4 in Figure 4.13(b). Thus, doubling the minimum distance in Figure 4.13
d2min
leads to a four-fold increase in Es and Eb . However, the quantity E b
does not change due to
scaling; it depends only on the relative geometry of the constellation points. We therefore adopt

157
this scale-invariant measure as our notion of power efficiency for a constellation:

d2min
ηP = (4.19)
Eb
Since this quantity is scale-invariant, we can choose any convenient scaling in computing it: for
QPSK, choosing the scaling on the left in Figure 4.13, we have dmin = 2, Es = 2, Eb = 1, which
gives ηP = 4.
It is important to understand how these quantities relate to physical link parameters. For a
given bit rate Rb and received power PRX , the energy per bit is given by Eb = PRRX b
. It is worth
verifying that the units make sense: the numerator has units of Watts, or Joules/sec, while the
denominator has units of bits/sec, so that Eb has units of joules/bit. We shall see in Chapter 6
that the reliability of communication is determined by the power efficiency ηP (a scale-invariant
quantity which is a function of the constellation shape) and the dimensionless signal-to-noise ratio
(SNR) measure Eb /N0 , where N0 is the noise power spectral density, which has units of watts/Hz,
Eb
or Joules. Specifically, the reliability can be approximately characterized by the product ηP N 0
, so
that, for a given desired reliability, the required energy per bit (and hence power) scales inversely
as power efficiency for a fixed bit rate. Communication link designers use such concepts as the
basis for forming a “link budget” that can be used to choose link parameters such as transmit
power, antenna gains and range.
Even based on these rather sketchy and oversimplified arguments, we can draw quick conclusions
on the power-bandwidth tradeoffs in using different constellations, as shown in the following
example.

Example 4.3.2 We wish to design a passband communication system operating at a bit rate of
40 Mbps.
(a) What is the bandwidth required if we employ QPSK, with an excess bandwidth of 25%.
(b) What if we now employ 16QAM, again with excess bandwidth 25%.
(c) Suppose that the QPSK system in (a) attains a desired reliability when the transmit power is
50 mW. Give an estimate of the transmit power needed for the 16QAM system in (b) to attain
a similar reliability.
(d) How does the bandwidth and transmit power required change for the QPSK system if we
increase the bit rate to 80 Mbps.
(e) How does the bandwidth and transmit power required change for the QPSK system if we
increase the bit rate to 80 Mbps.
Solution: (a) The bandwidth efficiency of QPSK is 2 bits/symbol, hence the minimum bandwidth
required is 20 MHz. For excess bandwidth of 25%, the bandwidth required is 25 MHz.
(b) The bandwidth efficiency of 16QAM is 4 bits/symbol, hence, reasoning as in (a), the band-
width required is 12.5 MHz.
(c) We wish to set ηP Eb /N0 to be equal for both systems in order to keep the reliability roughly
the same. Assuming that the noise PSD N0 is the same for both systems, the required Eb scales
as 1/ηP . Since the bit rates Rb for both systems are equal, the required received power P = Eb Rb
(and hence the required transmit power, assuming that received power scales linearly with trans-
mit power) also scales as 1/ηP . We already know that ηP = 4 for QPSK. It remains to find ηP for
16QAM, which is shown in Problem 4.15 to equal 8/5. We therefore conclude that the transmit
power for the 16QAM system can be estimated as
ηP (QP SK)
PT (16QAM) = PT (QP SK)
ηP (16QAM)
which evaluates for 125 mW.
(d) For fixed bandwidth efficiency, required bandwidth scales linearly with bit rate, hence the

158
new bandwidth required is 50 MHz. In order to maintain a given reliability, we must maintain
the same value of ηP Eb /N0 as in (c). The power efficiency ηP is unchanged, since we are using
the same constellation. Assuming that the noise PSD N0 is unchanged, the required energy per
bit Eb is unchanged, hence transmit power must scale up linearly with bit rate Rb . Thus, the
power required using QPSK is now 100 mW.
(e) Arguing as in (d), we require a bandwidth of 25 MHz and a power of 250 mW for 16QAM,
using the results in (b) and (c).

4.3.5 The Nyquist criterion at the link level

Symbols Transmit Filter Channel Filter Receive Filter z(t)

g (t) Sampler z(nT)
{b[n]} g (t) g (t) RX
TX C When is z(nT) = b[n]?
rate 1/T rate 1/T

Figure 4.14: Nyquist criterion at the link level.

Figure 4.14 shows a block diagram for a link using linear modulation, with the entire model
expressed in complexP baseband. The symbols {b[n]} are passed through the transmit filter to
obtain the waveform n b[n]gT X (t − nT ). This then goes through the channel filter gC (t), and
then the receive filter
P gRX (t). Thus, at the output of the receive filter, we have the linearly
modulated signal n b[n]p(t − nT ), where p(t) = (gT X ∗ gC ∗ gRX )(t) is the cascade of the
transmit, channel and receive filters. We would like the pulse p(t) to be Nyquist at rate 1/T , so
that, in the absence of noise, the symbol rate samples at the output of the receive filter equal
the transmitted symbols. Of course, in practice, we do not have control over the channel, hence
we often assume an ideal channel, and design such that the cascade of the transmit and receive
filter, given by (gT X ∗ gRX ) (t)GT X (f )GRX (f ) is Nyquist. One possible choice is to set GT X to
be a Nyquist pulse, and GRX to be a wideband filter whose response is flat over the band of
interest. Another choice that is even more popular is to set GT X (f ) and GRX (f ) to be square
roots of a Nyquist pulse. In particular, the square root raised cosine (SRRC) pulse is often used
in practice.
A framework for software simulations of linear modulated systems with raised cosine and SRRC
pulses, including Matlab code fragments, is provided in the appendix, and provides a foundation
for Software Lab 4.1.
Square root Nyquist pulses and their time domain interpretation: A pulse g(t) ↔ G(f )
is defined to be square root Nyquist at rate 1/T if |G(f )|2 is Nyquist at rate 1/T . Note that
P (f ) = |G(f )|2 ↔ p(t) = (g ∗ gM F )(t), where gM F (t) = g ∗ (−t). The time domain Nyquist
condition is given by
Z
p(mT ) = (g ∗ gM F )(mT ) = g(t)g ∗(t − mT )dt = δm0 (4.20)

That is, a square root Nyquist pulse has an autocorrelation function that vanishes at nonzero
integer multiples of T . In other words, the waveforms {g(t − kT, k = 0, ±1, ±2, ...} are orthonor-
mal, and can be used to provide a basis for constructing more complex waveforms, as we see in
Section 4.3.6.
Food for thought: True or False? Any pulse timelimited to [0, T ] is square root Nyquist at
rate 1/T .

159
4.3.6 Linear modulation as a building block
Linear modulation can be used as a building block for constructing more sophisticated waveforms,
using discrete-time sequences modulated by square root Nyquist pulses. Thus, one symbol would
be made up of multiple “chips,” linearly modulated by a square root Nyquist “chip waveform.”
Specifically, suppose that ψ(t) is square root Nyquist at a chip rate T1c . N chips make up
one symbol, so that the symbol rate is T1s = N1Tc , and a symbol waveform is given by linearly
modulating a code vector s = (s[0], ..., s[N − 1]) consisting of N chips, as follows:

N
X
s(t) = s[k]ψ(t − kTc )
k=0

Since {ψ(t − kTc )} are orthonormal (see (4.20)), we have simply expressed the code vector in a
continuous time basis. Thus, the continuous time inner product between two symbol waveforms
(which determines their geometric relationships and their performance in noise, as we see in
the next chapter) is equal to the discrete time inner product between the corresponding code
vectors. Specifically, suppose that s1 (t) and s2 (t) are two symbol waveforms corresponding to
code vectors s1 and s2 , respectively. Then their inner product satisfies
N
X −1 N
X −1 Z N
X −1
hs1 , s2 i = s1 [k]s∗2 [l] ∗
ψ(t − kTc )ψ (t − lTc )dt = s1 [k]s∗2 [k] = hs1 , s2 i
k=0 l=0 k=0

where we have use the orthonormality of the translates {ψ(t − kTc )}. This means that we can
design discrete time code vectors to have certain desired properties, and then linearly modulate
square root Nyquist chip waveforms to get symbol waveforms that have the same desired prop-
erties. For example, if s1 and s2 are orthogonal, then so are s1 (t) and s2 (t); we use this in the
next section when we discuss orthogonal modulation.
Examples of square root Nyquist chip waveforms include a rectangular pulse timelimited to an
interval of length Tc , as well as bandlimited pulses such as the square root raised cosine. From
Theorem 4.2.1, we see that the PSD of the modulated waveform is proportional to |Ψ(f )|2 (it is
typically a good approximation to assume that the chips {s[k]} are uncorrelated). That is, the
bandwidth occupancy is determined by that of the chip waveform ψ.

4.4 Orthogonal and Biorthogonal Modulation

While linear modulation with larger and larger constellations is a means of increasing bandwidth
efficiency, we shall see that orthogonal modulation with larger and larger constellations is a
means of increasing power efficiency (at the cost of making the bandwidth efficiency smaller).
Consider first M-ary frequency shift keying (FSK), a classical form of orthogonal modulation in
which one of M sinusoidal tones, successively spaced by ∆f , are transmitted every T units of
time, where T1 is the symbol rate. Thus, the bit rate is logT2 M , and for a typical symbol interval,
the transmitted passband signal is chosen from one of M possibilities:

up,k (t) = cos (2π(f0 + k∆f )t) , 0 ≤ t ≤ T, k = 0, 1, ..., M − 1

where we typically have f0 ≫ T1 . Taking f0 as reference, the corresponding complex baseband

waveforms are
uk (t) = exp (j2πk∆f t) , 0 ≤ t ≤ T, k = 0, 1, ..., M − 1

160
Let us now understand how the tones should be chosen in order to ensure orthogonality. Recall
that the passband and complex baseband inner products are related as follows:
1
hup,k , up,l i = Rehuk , ul i
2
so we can develop criteria for orthogonality working in complex baseband. Setting k = l, we see
that
||uk ||2 = T
For two adjacent tones, l = k + 1, we leave it as an exercise to show that

sin 2π∆f T
Rehuk , uk+1i =
2π∆f

We see that the minimum value of ∆f for which the preceding quantity is zero is given by
1
2π∆f T = π, or ∆f = 2T .
1
Thus, from the point of view of the receiver, a tone spacing of 2T ensures that when there is an
incoming wave at the kth tone, then correlating against the kth tone will give a large output, but
correlating against the (k + 1)th tone will give zero output (in the absence of noise). However,
this assumes a coherent system in which the tones we are correlating against are synchronized in
phase with the incoming wave. What happens if they are 90◦ out of phase? Then correlation of
the kth tone with itself yields
T
π
Z 
cos (2π(f0 + k∆f )t) cos 2π(f0 + k∆f )t + dt = 0
0 2

(by orthogonality of the cosine and sine), so that the output we desire to be large is actually
zero! Robustness to such variations can be obtained by employing noncoherent reception, which
we describe next.
Noncoherent reception: Let us develop the concept of noncoherent reception in generality,
because it is a concept that is useful in many settings, not just for orthogonal modulation. Sup-
pose that we transmit a passband waveform, and wish to detect it at the receiver by correlating
it against the receiver’s copy of the waveform. However, the receiver’s local oscillator may not
be synchronized in phase with the phase of the incoming wave. Let us denote the receiver’s copy
of the signal as
up (t) = uc (t) cos 2πfc t − us (t) sin 2πfc t
and the incoming passband signal as

yp (t) = yc (t) cos 2πfc t − ys (t) sin 2πfc t = uc (t) cos (2πfc t + θ) − us (t) sin (2πfc t + θ)

Using the receiver’s local oscillator as reference, the complex envelope of the receiver’s copy is
u(t) = uc + jus (t), while that of the incoming wave is y(t) = u(t)ejθ . Thus, the inner product

1 1 1
||u||2
hyp , up i = Rehy, ui = Rehuejθ , ui = Re ||u||2ejθ = cos θ
2 2 2 2
Thus, the output of the correlator is degraded by the factor cos θ, and can actually become zero,
as we have already observed, if the phase offset θ = π/2. In order to get around this problem,
let us look at the complex baseband inner product again:

hy, ui = huejθ , ui = ejθ ||u||2

161
We could ensure that this output remains large regardless of the value of θ if we took its magni-
tude, rather than the real part. Thus, noncoherent reception corresponds to computing |hy, ui|
or |hy, ui|2. Let us unwrap the complex inner product to see what this entails:
Z Z
∗
hy, ui = y(t)u (t)dt = (yc (t)+jys (t))(uc (t)−jus (t))dt = (hyc , uc i + hys , us i)+j (hys , uc i − hyc , us i)

Thus, the noncoherent receiver computes the quantity

|hy, ui|2 = (hyc , uc i + hys , us i)2 + (hys , uc i − hyc , us i)2

In contrast, the coherent receiver computes

Rehy, ui = hyc , uc i + hys , us i

That is, when the receiver LO is synchronized to the phase of the incoming wave, we can correlate
the I component of the received waveform with the I component of the receiver’s copy, and
similarly correlate the Q components, and sum them up. However, in the presence of phase
asynchrony, the I and Q components get mixed up, and we must compute the magnitude of the
complex inner product to recover all the energy of the incoming wave. Figure 4.15 shows the
receiver operations corresponding to coherent and noncoherent reception.

Coherent
) receiver output

Low Pass yc (t) uc (t) Zc

Filter
Passband Squarer
Received
2 cos 2 π fc t
Signal
us (t)
− Noncoherent
receiver
Zs Squarer
output
y s (t)
Low Pass uc (t)
Filter

− 2 sin 2π fc t
us (t)

Figure 4.15: Structure of coherent and noncoherent receivers.

Back to FSK: Going back to FSK, if we now use noncoherent reception, then in order to
ensure that we get a zero output (in the absence of noise) when receiving the kth tone with a
noncoherent receiver for the (k + 1)th tone, we must ensure that

|huk , uk+1i| = 0

We leave it as an exercise (Problem 4.18) to show that the minimum tone spacing for noncoherent
FSK is T1 , which is double that required for orthogonality in coherent FSK. The bandwidth for
M
coherent M-ary FSK is approximately 2T , which corresponds to a time-bandwidth product of
M
approximately 2 . This corresponds to a complex vector space of dimension M2 , or a real vector
space of dimension M, in which we can fit M orthogonal signals. On the other hand, M-ary
noncoherent signaling requires M complex dimensions, since the complex baseband signals must
remain orthogonal even under multiplication by complex-valued scalars.
Summarizing the concept of orthogonality: To summarize, when we say “orthogonal”
modulation, we must specify whether we mean coherent or noncoherent reception, because the

162
concept of orthogonality is different in the two cases. For a signal set {sk (t)}, orthogonality
requires that, for k 6= l, we have
Re(hsk , sl i) = 0 coherent orthogonality criterion
(4.21)
hsk , sl i = 0 noncoherent orthogonality criterion

Bandwidth efficiency: We conclude from the example of orthogonal FSK that the bandwidth
efficiency of orthogonal signaling is ηB = log2M(2M ) bits/complex dimension for coherent systems,
and ηB = logM 2 (M )
bits/complex dimension for noncoherent systems. This is a general observation
that holds for any realization of orthogonal signaling. In a signal space of complex dimension
D (and hence real dimension 2D), we can fit 2D signals satisfying the coherent orthogonality
criterion, but only D signals satisfying the noncoherent orthogonality criterion. As M gets large,
the bandwidth efficiency tends to zero. In compensation, as we see in Chapter 6, the power
efficiency of orthogonal signaling for large M is the “best possible.”
Orthogonal Walsh-Hadamard codes
Section 4.3.6 shows how to map vectors to waveforms while preserving inner products, by using
linear modulation with a square root Nyquist chip waveform. Applying this construction, the
problem of designing orthogonal waveforms {si } now reduces to designing orthogonal code vectors
{si }. Walsh-Hadamard codes are a standard construction employed for this purpose, and can
be constructed recursively as follows: at the nth stage, we generate 2n orthogonal vectors, using
the 2n−1 vectors constructed in the n − 1 stage. Let Hn denote a matrix whose rows are 2n
orthogonal codes obtained after the nth stage, with H0 = (1). Then
 
Hn−1 Hn−1
Hn =
Hn−1 −Hn−1
We therefore get
 
  1 1 1 1
1 1  1 −1 1 −1 
H1 = ,  1 1 −1 −1  ,
H2 =   etc.
1 −1
1 −1 −1 1
Figure 4.16 depicts the waveforms corresponding to the 4-ary signal set in H2 using a rectangular
timelimited chip waveform to go from sequences to signals, as described in Section 4.3.6.
The signals {si } obtained above can be used for noncoherent orthogonal signaling, since they
satisfy the orthogonality criterion hsi , sj i = 0 for i 6= j. However, just as for FSK, we can
fit twice as many signals into the same number of degrees of freedom if we used the weaker
notion of orthogonality required for coherent signaling, namely Re(hsi , sj i = 0 for i 6= j. It
is easy to check that for M-ary Walsh-Hadamard signals {si , i = 1, ..., M}, we can get 2M
orthogonal signals for coherent signaling: {si , jsi , i = 1, ..., M}. This construction corresponds
to independently modulating the I and Q components with a Walsh-Hadamard code; that is,
using passband waveforms si (t) cos 2πfc t and −si (t) sin 2πfc t (the negative sign is only to conform
to our convention for I and Q, and can be dropped, which corresponds to replacing jsi by −jsi
in complex baseband), i = 1, ..., M.
Biorthogonal modulation
Given an orthogonal signal set, a biorthogonal signal set of twice the size can be obtained by
including a negated copy of each signal. Since signals s and −s cannot be distinguished in a
noncoherent system, biorthogonal signaling is applicable to coherent systems. Thus, for an M-ary
Walsh-Hadamard signal set {si } with M signals obeying the noncoherent orthogonality criterion,
we can construct a coherent orthogonal signal set {si , jsi } of size 2M, and hence a bior