Ee 113 Book 2008

DISCRETE-TIME
PROCESSING AND FILTERING
Ali H. Sayed
Electrical Engineering Department

University of California at Los Angeles
c
2008
All rights reserved.
These notes are only distributed to the students enrolled in the EE113
course in the Electrical Engineering Department at UCLA.
The notes cannot be reproduced or distributed without the explicit written consent from the author:
A. H. Sayed, Electrical Engineering Department, UCLA, CA 90095, sayed@ee.ucla.edu
The notes are complemented by an interactive website at http://www.ee.ucla.edu/dsplab
Please email typos and suggestions to sayed@ee.ucla.edu.
Contents
Motivation
1.1
1.2
1.3
1.4
1.5
1.6
1.7
1.8
1.9
2
Signals
Classification of Signals
Quantization
Sampling
Signal Processing
Systems
DSP Technology
Applications
1.8.1
Voiced and Unvoiced Speech
1.8.2
Estimation of DC Levels
Problems
1
1
3
4
6
7
9
10
10
10
12
16
Fundamental Sequences
21
2.1
2.2
2.3
2.4
2.5
2.6
2.7
2.8
21
24
29
31
35
37
40
43
Complex Numbers
Basic Sequences
Polar Plots
Symmetry Relations
Energy and Power Sequences
Signal Transformations
Application: Savings Account
Problems
Periodic Sequences
49
3.1
3.2
3.3
3.4
3.5
3.6
3.7
49
53
55
57
58
61
65
Periodic Signals
Complex Exponential Sequences
Angular Frequency
Eulers Relation
Relating Angular Frequencies and Periods
Application: Harmonics and Music Synthesis
Problems
Discrete-Time Systems
71
vii
viii
4.1
4.2
4.3
4.4
4.5
4.6
4.7
4.8
4.9
4.10
4.11
CONTENTS
4.12
5
105
5.1
5.2
5.3
5.4
5.5
5.6
5.7
5.8
105
110
111
113
114
115
117
119
119
121
123
Convolution Sum for LTI Systems

Causality of LTI Systems
BIBO Stability
Series Cascade of LTI Systems
Parallel Cascade of LTI Systems
FIR and IIR Systems
Inverse Problem
Applications
5.8.1
Multipath Channels
5.8.2
Financial Growth Model
Problems
Linear Convolution
131
6.1
131
131
132
133
135
135
136
136
140
140
144
148
6.2
6.3
6.4
7
71
73
76
77
78
81
82
83
86
87
90
90
92
95
100
Impulse Response Sequence
5.9
6
Systems
Classes of Systems
Relaxed Systems
Dynamic Systems
Time-Invariant Systems
Causal Systems
Stable Systems
Linear Systems
Constant-Coefficient Difference Equations
System Representations
Applications
4.11.1 Multipath Communications
4.11.2 Financial Growth Model
4.11.3 Population Growth Models
Problems
Properties of the Convolution Sum

6.1.1
Commutativity
6.1.2
Distributivity
6.1.3
Associativity
6.1.4
Convolution with the Unit-Sample Sequence
Evaluation of the Convolution Sum
6.2.1
Analytical Method
6.2.2
Graphical Method
Applications
6.3.1
Echo Cancellation
6.3.2
Population Growth Management
Problems
Homogeneous Difference Equations
155
7.1
7.2
7.3
7.4
7.5
7.6
7.7
7.8
8
155
157
157
158
159
160
163
167
169
170
171
171
173
176
Solving Difference Equations
181
8.1
8.2
8.3
8.4
8.5
8.6
8.7
8.8
8.9
182
186
188
189
191
193
194
195
197
197
201
205
8.10
9
Homogenous Equations
7.1.1
Distinct Roots
7.1.2
Repeated Roots
7.1.3
Complex Roots
7.1.4
Solution Method
Homogeneous Equations with Initial Conditions
Impulse Response of LTI Systems
Stability of Causal LTI Systems
Impulse Responses of non-LTI Systems
Complete Response of LTI Systems
Applications
7.7.1
Carbon Dating
7.7.2
Rabbit Population and Fibonacci Numbers
Problems
Particular Solution
Characterizing All Solutions
First Method for Finding Complete Solutions
Zero-State Response
Zero-Input Response
Second Method for Finding Complete Solutions
Transient and Steady-State Response
Third Method for Finding Complete Solutions
Applications
8.9.1
Macroeconomics Model
8.9.2
Cell Division in Biology
Problems
z-Transform
213
9.1
9.2
213
214
215
216
219
223
223
226
227
229
230
231
233
236
9.3
9.4
9.5
Bilateral z-Transform
Region of Convergence
9.2.1
Finite-Duration Sequences
9.2.2
Infinite-Duration Sequences
Exponential Sequences
Properties of the z-Transform
9.4.1
Linearity
9.4.2
Time Shifts
9.4.3
Exponential Modulation
9.4.4
Time Reversal
9.4.5
Linear Modulation
9.4.6
Complex Conjugation
9.4.7
Linear Convolution
Evaluating Series
ix
CONTENTS
9.6
9.7
CONTENTS
9.8
9.9
9.A
10
11
12
237
238
238
240
244
244
251
Partial Fractions
255
10.1
10.2
10.3
10.4
10.5
10.6
255
256
259
264
268
268
Rational Transforms
Elementary Rational Fractions
Partial Fractions Expansion
Integral Inversion Formula
Applications
Problems
Transfer Functions
273
11.1
11.2
11.3
11.4
11.5
11.6
11.7
11.8
11.9
11.10
273
273
275
278
278
282
283
284
286
286
Transfer Functions of LTI Systems

Eigenfunctions of LTI Systems
Evaluation from Difference Equations
Finding Output Sequences
Finding Difference Equations
Poles, Zeros, and Modes
Realizable LTI Systems
System Inversion
Applications
Problems
Unilateral z-Transform
295
12.1
12.2
12.3
295
296
299
300
301
302
304
305
306
308
309
309
12.4
12.5
12.6
12.7
13
Initial Value Theorem

Upsampling and Downsampling
9.7.1
Upsampling
9.7.2
Downsampling
Applications
Problems
Convergence of Power Series
z-Transform and Difference Equations

Properties of the Unilateral z-Transform
12.3.1 Linearity
12.3.2 Time Shifts
12.3.3 Exponential Modulation
12.3.4 Linear Modulation
12.3.5 Linear Convolution
Initial and Final Value Theorems
Applications
Problems
Discrete-Time Fourier Transform
313
13.1
13.2
13.3
13.4
13.5
13.6
13.7
14
343
14.1
14.2
343
344
345
349
351
352
354
357
358
360
365
367
368
372
372
374
377
378
14.4
14.5
16
313
320
325
328
333
335
335
Properties of the DTFT
14.3
15
Definition of the DTFT

Uniform Convergence
Inverse DTFT
Mean-Square Convergence
Inverse DTFT by Partial Fractions
Applications
Problems
Periodicity of the DTFT

Useful Properties
14.2.1 Linearity
14.2.2 Time Shifts
14.2.3 Frequency Shifts
14.2.4 Modulation
14.2.5 Time Reversal
14.2.6 inear Modulation
14.2.8 Multiplication in the Time Domain
14.2.9 Conjugation
14.2.10 Real Sequences
14.2.11 Parsevals Relation
Upsampling and Downsampling
14.3.1 Upsampling
14.3.2 Downsampling
Applications
Problems
Frequency Response
385
15.1
15.2
15.3
15.4
15.5
15.6
15.7
385
388
398
401
405
406
406
Frequency Content of a Sequence

Frequency Response of an LTI System
Linear Time-Invariant Systems
Ideal Filters
Realizable Filters
Applications
Problems
Minimum and Linear Phase Systems
415
16.1
16.2
16.3
415
418
421
422
424
426
Group Delay
Linear Phase Characteristics
Linear Phase FIR Filters
16.3.1 Type-I FIR Filters
16.3.2 Type-II FIR Filters
16.3.3 Type-III FIR Filters
xi
CONTENTS
xii
CONTENTS
16.4
16.5
16.6
16.7
16.8
17
18
427
431
433
435
438
439
440
442
444
445
449
449
Discrete Fourier Transform
457
17.1
17.2
17.3
17.4
17.5
17.6
17.7
457
462
471
475
478
480
481
Motivation
Relation to Original Sequence
Inverse DFT
Vector Representation
Applications
Problems
Properties of the DFT
487
18.1
18.2
488
489
489
493
500
503
505
508
512
520
523
525
525
18.3
18.4
19
16.3.4 Type-IV FIR Filters

16.3.5 Location of Zeros
All-Pass Systems
16.4.1 First-Order All-Pass Sections
16.4.2 Second-Order All-Pass Sections
16.4.3 Higher-Order All-Pass Sections
Minimum Phase Systems
Fundamental Decomposition
16.6.1 Minimum Group Delay Property
16.6.2 Minimum Energy Delay Property
Applications
Problems
Periodicity of the DFT

Useful Properties
18.2.1 Linearity
18.2.2 Circular Time Shifts
18.2.3 Circular Frequency Shifts
18.2.4 Modulation
18.2.5 Circular Time Reversal
18.2.6 Complex Conjugation in Time and Frequency
18.2.7 Circular Convolution
Applications
Problems
Computing Linear Convolutions
531
19.1
19.2
19.3
531
532
535
536
539
545
19.4
Relating Linear and Circular Convolutions

Computing Linear Convolutions via the DFT
Block Convolution Methods
19.3.1 Overlap-Add Convolution Method
19.3.2 Overlap-Save Convolution Method
Applications
19.5
Problems
545
xiii
CONTENTS
20
21
22
23
Fast Fourier Transform
549
20.1
20.2
20.3
20.4
20.5
549
551
560
568
568
Computational Complexity
Decimation-in-Time FFT
Decimation-in-Frequency FFT
Applications
Problems
Spectral Resolution
571
21.1
21.2
21.3
21.4
21.5
571
578
584
590
590
Windowing
Spectral Resolution of the DTFT
Spectral Resolution of the DFT
Applications
Problems
Sampling
591
22.1
22.2
22.3
22.4
22.5
22.6
Sampling Process
Fourier Transform
Linear Convolution
Linear Time-Invariant Systems
Nyquist Rate for Baseband Signals
Sampling of Bandpass Signals
22.6.1 Complex-valued Bandpass Signals
22.6.2 Real-valued Bandpass Signals
22.7 Relation of Fourier Transform to the DTFT
22.8 Relation of Fourier Transform to the DFT
22.9 Spectral Resolution
22.10 Applications
22.11 Problems
591
593
602
608
611
623
624
626
631
635
638
640
640
Discrete-Time Realizations
649
23.1
650
650
652
655
658
659
660
661
663
667
669
23.2
23.3
Realizations of FIR Filters

23.1.1 Direct or Tapped-Delay-Line Realizations
23.1.2 Cascade Realizations
23.1.3 Exploiting Symmetry
Realizations of IIR or ARMA Filters
23.2.1 Direct Realizations of AR Filters
23.2.2 Type-I Direct Realizations of ARMA Filters
23.2.3 Type-II Direct Realizations of ARMA Filters
23.2.4 Cascade Realizations
23.2.5 Parallel Realizations
Transposed Realizations
xiv
23.4
23.5
23.6
23.7
CONTENTS
24
672
675
687
687
Lattice Realizations
693
24.1
24.2
693
695
696
700
702
707
713
714
714
718
720
723
727
731
733
737
740
740
743
24.3
24.4
24.5
24.6
24.7
24.8
24.9
24.A
25
Masons Rule
State-Space Realizations
Applications
Problems
Motivation
Composite Cascades
24.2.1 Linear Fractional Transformations
24.2.2 Mapping Properties of All-Pass Sections
24.2.3 Stability Properties
24.2.4 Inverse Transformation
24.2.5 Listing of Mapping Properties
Lattice Realization of All-Pass Filters
24.3.1 Lattice Sections
24.3.2 Schur-Cohn Stability Test
24.3.3 Schur Algorithm
24.3.4 Levinson Algorithm
Lattice Realization of AR Filters
Lattice Realization of MA Filters
Lattice Realization of ARMA Filters
Summary of Lattice Realizations
Applications
Problems
Appendix: Maximum Modulus Principle
Quantization Effects
745
25.1
745
745
751
753
753
756
757
757
759
763
767
769
775
779
782
786
789
25.2
25.3
25.4
Binary Representations
25.1.1 Fixed-Point Representations
25.1.2 Floating-Point Representations
Quantization Errors
25.2.1 Rounding and Truncation
25.2.2 Overflow and Underflow
Finite Word-Length Effects
25.3.1 Effect on Processing and Filtering
25.3.2 Limit Cycles
25.3.3 Effect on Frequency Response
25.3.4 Effect on FIR Implementations
25.3.5 Propagation of Quantization Noise
25.3.6 Roundoff Noise Analysis
25.3.7 Pairing and Ordering of Sections
25.3.8 Scaling of Signals
25.3.9 Orthogonal Filters
Applications
25.5
26
Problems
790
Design of FIR Filters
799
26.1
26.2
799
803
805
807
815
824
825
826
833
839
840
840
26.3
26.4
26.5
27
Design of Analog Filters
843
27.1
27.2
27.3
27.4
27.5
843
846
849
851
857
858
868
872
876
876
878
881
883
884
886
27.6
27.7
27.8
27.9
27.A
28
29
Practical Filter Specifications

Window Method
26.2.1 Design Procedure
26.2.2 Mainlobes and Sidelobes
26.2.3 Kaiser Windows
Equiripple Design Method
26.3.1 Optimization Problem Formulation
26.3.2 Relation to Polynomial Approximation Theory
26.3.4 FIR Filters of Types II, III, and IV
Applications
Problems
Laplace Transform
Transfer Functions
Filter Specifications
Low-Pass Butterworth Filters
Low-Pass Chebyshev Filters
27.5.1 Type-I Chebyshev Filters
27.5.2 Type-II Chebyshev Filters
Low-Pass Elliptic Filters
Frequency Transformations
27.7.1 Design of High-Pass Filters
27.7.2 Design of Band-Pass Filters
27.7.3 Design of Band-Stop Filters
Applications
Problems
Convergence of Laplace Transform
Design of IIR Filters
889
28.1
28.2
28.3
28.4
28.5
28.6
28.7
28.8
889
892
899
903
907
913
922
923
Relating Laplace and z-Transforms

Impulse Invariance Method
Step Invariance Method
Matched z-Transformation
Bilinear Transformation Method
Frequency Transformations
Applications
Problems
Multirate Processing
929
xv
CONTENTS
xvi
29.1
CONTENTS
29.2
29.3
29.4
29.5
30
965
30.1
965
966
977
984
984
991
994
995
1000
1000
1002
1002
1003
1009
1016
1017
30.3
30.4
30.5
30.6
30.7
Analysis Filter Bank

30.1.1 Uniform Filter Banks
30.1.2 DFT Analysis Filter Bank
Synthesis Filter Bank
30.2.1 Uniform Filter Bank
30.2.2 DFT Synthesis Filter Bank
Subband Processing
Oversampled Filter Banks
Perfect Reconstruction Filter Banks
30.5.1 Alias-Free Reconstruction Condition
30.5.2 Perfect Reconstruction Condition
30.5.3 Perfect Reconstruction with FIR Filters
30.5.4 Quadrature Mirror Filter Banks
30.5.5 Orthogonal Perfect Reconstruction Filter Banks
Applications
Problems
Block Filtering
31.1
31.2
31.3
31.4
31.5
31.6
31.7
32
929
929
934
940
943
947
951
956
956
957
960
962
963
963
Filter Banks
30.2
31
Sampling Rate Conversion

29.1.1 Increasing the Sampling Rate by an Integer Factor
29.1.2 Decreasing the Sampling Rate by an Integer Factor
29.1.3 Modifying the Sampling Rate by a Rational Factor
Polyphase Realizations
29.2.1 Polyphase Decomposition
29.2.2 Polyphase Structures for Decimation and Interpolation
Nyquist Filters
29.3.1 Sample Preservation Property
29.3.2 Polyphase Characterization
29.3.4 Half-Band Filters
Applications
Problems
Block Processing
Overlap-Save DFT-Based Block Filtering
Overlap-Add DFT-Based Block Filtering
DCT-Based Block Filtering
DHT-Based Block Filtering
Applications
Problems
Random Signals
32.1
Probability Density Functions
1021
1021
1027
1037
1039
1042
1047
1047
1051
1051
32.2
32.3
32.4
32.5
32.6
Mean and Variance

Dependent Random Variables
Complex-Valued Random Variables
Random Vectors
Properties of Covariance Matrices
32.6.1 Covariance Matrices are Hermitian
32.6.2 Covariance Matrices are Non-Negative Definite
32.7 Random Processes
32.8 Power Spectral Densities
32.8.1 iltering of Stationary Processes
32.8.2 pectral Factorization
32.9 Applications
32.10 Problems
33
Linear Estimation
33.1
33.2
33.3
33.4
33.5
33.6
33.7
33.A
34
Estimation Without Observations

Using Correlated Observations
Using Multiple Observations
Design Examples
Vector Estimation
Applications
Problems
Appendix: Complex Gradients
33.A.1 auchy-Riemann Conditions
33.A.2 ector Arguments
Linear Prediction
34.1
34.2
34.3
34.4
34.5
34.6
34.7
34.8
34.9
Order-Update Estimation
34.1.1 Useful Property: Linear Transformations
34.1.2 Useful Property: Uncorrelated Components
34.1.3 Useful Property: Uncorrelated Observations
34.1.4 Main Decomposition Result
34.1.5 Interpretation
Forward Prediction Problem
Backward Prediction Problem
Relating the Prediction Problems
Residual Recursions
Levinson Algorithm
34.6.1 Polynomial Form
34.6.2 Lattice Implementation
Triangular Factorization
Applications
Problems
1054
1058
1061
1063
1067
1067
1073
1077
1078
1080
1083
1086
1087
1091
1091
1093
1099
1103
1107
1109
1110
1114
1114
1116
1117
1117
1117
1118
1119
1120
1120
1121
1127
1131
1134
1140
1141
1142
1143
1145
1146
xvii
CONTENTS
xviii
35
Wiener Filtering
1149
CONTENTS
35.1
35.2
35.3
35.4
35.5
35.6
36
Kalman Filtering
36.1
36.2
36.3
36.4
36.5
36.6
36.7
36.8
36.9
36.10
36.11
36.12
37
Wiener Smoothing Problem

Wiener Filtering Problem
Levinson and Spectral Factorization
Schur and Spectral Factorization
Applications
Problems
Uncorrelated Observations
Innovations Process
State-Space Model
Recursion for the State Estimator
Computing the Gain Matrix
Riccati Recursion
Covariance Form
Measurement and Time-Update Forms
Modeling and Whitening Filters
Relation to Wiener Filtering
Applications
Problems
Adaptive Filtering
37.1
37.2
37.3
Problem Formulation
Steepest-Descent Method
Stochastic Approximation
37.3.1 LMS Filter
37.3.2 NLMS Filter
37.3.3 Other LMS-Type Filters
37.3.4 RLS Filter
37.4 Application: Adaptive Channel Estimation
37.5 Application: Adaptive Channel Equalization
37.6 Application: Decision-Feedback Equalization
37.7 Ensemble-Average Learning Curves
37.8 Mean-Square Performance
37.8.1 Data Model
37.8.2 Energy Conservation Relation
37.8.3 Performance of LMS
37.8.4 Performance of NLMS
37.9 Applications
37.10 Problems
1149
1153
1164
1165
1168
1168
1171
1171
1174
1176
1177
1178
1179
1180
1181
1182
1183
1186
1186
1187
1187
1188
1193
1194
1195
1196
1198
1200
1202
1204
1206
1209
1210
1213
1215
1217
1218
1219
CHAPTER
Motivation
IFiltering.
n this initial chapter we explain what is meant by the title Discrete-Time Processing and
We first explain what is meant by the term signal, and then move on to explain
what discrete-time signals are. We also explain what signal processing entails and what
filtering means.
1.1 SIGNALS
For our purposes, the term signal will refer to a function of one or more independent
variables. The independent variable can be time, frequency, space coordinates, distance,
or some other variable of interest. In this book, we focus almost exclusively on functions
of a single variable and the independent variable will generally be either the time variable
or the frequency variable.
Example 1.1 (Moving cart)

Let x(t) denote the horizontal distance of a moving cart at time t relative to a point of reference.
Here the symbol t represents the independent variable and the symbol x represents the signal. In this
example, assuming that the cart starts moving at time 0, both t and x assume real values with t 0.
reference
point
x(t)
FIGURE 1.1
by x(t).
The horizontal distance of the cart at time t relative to a reference point is denoted
1
Discrete-Time Processing and Filtering, by Ali H. Sayed
c 2010 John Wiley & Sons, Inc.
Copyright
2
CHAPTER 1
MOTIVATION
Example 1.2 (Two-dimensional image)

Let x(m, ) denote a measure of the intensity or brightness of the (m, )-th pixel in an 8 8 twodimensional image with a total of 64 pixels. Here the indices {m, } represent the independent
variables and the symbol x represents the signal. The index m refers to the row location of the pixel
and the index denotes its column location. In this example, the variables {m, } assume integer
values in the range [0, 7], and it is assumed that the intensity variable x also assumes integer values
as well, say in the range [0, 255].
0 1 2 3 4 5 6 7
0
1
2
3
4
5
6
7
FIGURE 1.2 The intensity of the pixel at location (m, ) is denoted by x(m, ).
Example 1.3 (Students in a course)

Let x(n) denote the number of students attending a particular course at successive years, n. Here
the symbol n represents the independent variable and it assumes integer values, whereas the symbol
x denotes the signal and it also assumes integer values.
x(n) (students)
50
40
30
20
10
2007
FIGURE 1.3
2008
2009
2010
n (year)
The number of students attending a course during the years 2007 through 2010.
3
SECTION 1.2
Example 1.4 (Satellite orbiting the Earth)
CLASSIFICATION
OF SIGNALS
Let w(r) denote the angular speed of a satellite in uniform circular motion around the Earth at a
radial distance r from the center of the Earth. Here the symbol r represents the independent variable
and it assumes positive real values, whereas the symbol w represents the signal and it also assumes
positive real values.
w(r)
satellite
r
Earth
FIGURE 1.4
The angular speed of a satellite orbiting the Earth in uniform circular motion.
1.2 CLASSIFICATION OF SIGNALS

Signals can be classified in many ways. For the purposes of the treatment in this book, we
classify signals into three broad categories.
Continuous-Time Signals
In this case the independent variable assumes continuous real values as in Examples 1.1
and 1.4. We shall generally denote a continuous-time signal by x(t), with t denoting the
independent variable. An example of a continuous-time signal is the temperature variation
in a room over a continuous period of time.
Discrete-Time Signals
In this case the independent variable assumes discrete (i.e., integer) values as in Examples
1.2 and 1.3. We shall denote a discrete-time signal by x(n), with n denoting the independent variable. An example of a discrete-time signal is the average daily temperature in a
city, where the independent variable specifies the day of interest (say, day 1, day 2, and so
forth) and the signal corresponds to the average temperature on that day. We further note
that:
1. A discrete-time signal is also called a sequence. In this context, the notation x(n)
can be interpreted to refer to the nth term of the sequence. We shall employ both
terminologies, discrete-time signal and sequence, whenever convenient.
4
CHAPTER 1
MOTIVATION
2. The value (also called amplitude) of a discrete-time signal, x(n), at any particular
n, is not restricted; the amplitude may assume integer values, real values, or even
complex values.
Digital Signals
A digital signal is a sequence where the amplitude of its terms can belong to only a finite
number of possibilities. In Example 1.2, we saw that the intensity of every pixel (m, ) in
the image was limited to integer values in the range [0, 255]. Therefore, the signal x(m, )
in that example is a digital signal.
1.3 QUANTIZATION
Digital signals usually arise as the result of quantization. In quantization, a finite number
of bits is used to represent the amplitude of a signal. Assume, for instance, that 3 bits are
available to describe the amplitude of a signal, x(n). This means that we have 8 amplitude
possibilities described by the choices
000, 001, 010, 011
(used for nonnegative amplitudes)
100, 101, 110, 111
(used for negative amplitudes)
These eight choices are assigned as follows. Refer to Fig. 1.5 and assume the amplitudes
of the signal x(n) occur within the continuous range [4, 4]; the horizontal axis in the
figure represents the range of values that can be assumed by x(n). We divide the horizontal
axis into sub-intervals of width each. Then, whenever the amplitude of x(n) falls within
3
the interval [
2 , 2 ) we represent it by the value , which is assigned the bits 001. In other
words, we round the value of x to the nearest amplitude in the quantized domain. In this
3
way, all amplitudes in the range [
2 , 2 ) in the signal domain, x(n), are mapped into the
single amplitude in the quantized domain, xq (n). Likewise, whenever the amplitude of
x(n) falls within the interval [

2 , 2 ) we represent it by the value 0, which is assigned
the bits 000, and so forth. This construction starts with a discrete-time signal x(n) and
produces an amplitude-discretized version of it, which we are denoting by xq (n). The
amplitudes of xq (n) in this example are limited to the values:
{4, 3, 2, , 0, , 2, 3}
and we say that xq (n) is a digital signal.
5
SECTION 1.3
QUANTIZATION
xq
011
010
001
4 3 2
111
110
000
101
100
FIGURE 1.5 An example of 3-bit uniform quantization with rounding; the amplitude of a discretetime signal x(n) is quantized resulting in a digital signal xq (n).
Example 1.5 (3-bit quantization)

Consider the following three samples of a sequence x(n) at time instants n = 0, 1, 2:
x(0) = 0.2,
x(1) = 0.4,
x(2) = 0.7
The samples of x(n) assume values within the interval [1, 1]. We want to quantize the samples of
x(n) using 3 bits, as described in Fig. 1.5. The quantization step is set to = 0.25. We then find
that

x(0)
lies within the interval
x(1)
3
,
2 2
x(2)
3
5
,
2
2
5 7
,
2
2
Therefore, the corresponding quantized values and their bit representations would be
xq (0)
0.25
(001)
xq (1)
=
=
0.50
(110)
xq (2)
0.75
(011)
It is worth noting that computers and digital signal processors operate on digital signals
very effectively; digital signals are stored in computers and digital signal processors in the
form of bits or bytes. In most of this book, we shall deal with discrete-time signals as
opposed to digital signals. That is, the amplitude of each term in the sequence x(n) will
6
CHAPTER 1
MOTIVATION
not be restricted; it will generally be bounded but not quantized. There are at least two
reasons for proceeding in this manner:
1. First, discrete-time signals are more tractable to mathematical analysis than digital
signals.
2. Second, if we assume long enough word-lengths, i.e., if we assume digital signal representations that employ a sufficient number of bits to quantize the signal amplitudes,
then the loss in performance and accuracy that results from the use of quantization
may be assumed negligible.
Nevertheless, in Chapter 25 we shall examine in some detail the effect of quantization
errors on computations involving digital signals.
1.4 SAMPLING
How do sequences arise? In many cases, the data may already be available in discrete-time
form. For example, we may have available a table with entries that represent the yearly
levels of rainfall for the last 20 years in a city. In this case, we have a sequence with 20
entries and each entry in the table corresponds to a term in the sequence. Likewise, we may
have available a table indicating the number of students attending a course over a certain
number of years see Table 1.1
TABLE 1.1 Number of students attending a particular course over a 16 year period.
Year
Students
Year
Students
2008
2007
2006
2005
2004
2003
2002
2001
26
21
15
19
31
27
19
18
2000
1999
1998
1997
1996
1995
1994
1993
26
20
23
17
22
25
18
20
More often, however, sequences arise from sampling continuous-time signals such as
speech signals, radar signals, and biological signals. If x(t) is a continuous-time signal,
sampling it every T units of time (say, every T seconds or milliseconds) results in a sequence, x(n), whose terms are given by the values of x(t) evaluated at the time instants
t = nT , i.e.,
x(n) = x(t)|t=nT = x(nT )
(1.1)
In other words, only values of x(t) at time instants that are multiples of T are retained in
the sampling process and the other values of x(t) are ignored see Fig. 1.6. Usually, the
compact notation x(n) is used instead of x(nT ) to refer to the resulting sequence with the
letter T dropped. Besides begin a compact representation of the sequence, the notation
x(n) will also allow us to study properties of sequences independently of the sampling
period T .
7
SECTION 1.5
x(t)
SIGNAL
PROCESSING
x(n)
x(t)
FIGURE 1.6 Sampling of a continuous-time signal x(t) at multiples of the sampling period T to
generate a sequence x(n).
Remark. In general, the independent variable n in the notation x(n) does not necessarily refer to
time (for example, it may also refer to space or distance). Motivated by the sampling interpretation,
we shall nevertheless often use the time connotation to describe x(n). For example, when referring
to the sample x(n) we shall usually say the value of the sequence at time instant n.
1.5 SIGNAL PROCESSING

The term processing refers to the act of extracting information from a signal. For example, given a continuous-time signal x(t) that represents the temperature variation over an
interval of time T , we can extract information about the average temperature over this
interval of time via integration as follows:
x
=
1
T
x(t)dt
(1.2)
where the symbol x denotes the average temperature. This calculation amounts to evaluating the area under the temperature curve over the interval [0, T ] and dividing the result
by T see Fig. 1.7.
Likewise, given a sequence x(n) that represents the yearly rainfall over the last 20 years
in a city, we may extract information about the average annual rainfall as follows:
x
=
19
1 X
x(n)
20 n=0
(1.3)
Observe that we are numbering the terms of the sequence x(n) in this example from 0 to
19 and not from 1 to 20. It is common to use n = 0 as the origin of time when describing
sequences and we shall adopt this convention in the book.
Of course, we can perform more sophisticated processing on signals than simply evaluate their averages. For example, we can attempt to use the available data in order to predict
8
CHAPTER 1
x(t)
MOTIVATION
x
=
1
T
area
FIGURE 1.7
Extracting the average temperature by evaluating the area under the curve.
the level of rainfall two years ahead of time, or to estimate the number of students that will
be attending a particular course this year based on the attendance history over the previous
5 years. In another example, if x(n) denotes a speech sequence that is corrupted by background noise, we may want to process x(n) in order to remove the interfering noise and
generate a clear speech signal, y(n). This is one example of filtering whereby a signal is
processed to filter out undesired components or to transform the signal into another more
desirable form.
Figure 1.8 illustrates another filtering example. The top plot (a) shows a sinusoidal
signal, which is subjected to additive interference by the random fluctuations in part (b).
The result is the disturbed signal in part (c); which is the sum of the signals in parts (a) and
(b). A filter would then process the noisy signal in part (c) and attempt to recover the clear
sinusoidal version (a) or a good approximation for it. Obviously, the processing or the
algorithms that are needed to perform prediction and filtering tasks are more involved than
the processing that is involved in computing the signal averages mentioned in the earlier
examples.
amplitude
SECTION 1.6
sinusoidal
signal
SYSTEMS
(a)
disturbance
(b)
noisy
signal
(c)
time
FIGURE 1.8 Processing a noisy signal to remove (or reduce the effect of) the noise component
and recover the original sinusoidal signal.
1.6 SYSTEMS
The task of processing a signal in order to extract information from it is performed by a
system or filter. Systems operate on signals and transform an input signal into an output
signal, as shown in Fig. 1.9.
output signal
input signal
system
FIGURE 1.9
A system processes an input signal and transforms it into an output signal.
In this book we shall deal almost exclusively with discrete-time systems, namely, systems whose input and output signals are sequences. Hence, we shall deal with the processing of discrete-time signals. The discipline that studies discrete-time signals and systems,
as well as digital signals, is known as Discrete-Time Signal Processing. Sometimes it is
also called Digital Signal Processing (or DSP for short).
10
1.7 DSP TECHNOLOGY
CHAPTER 1
MOTIVATION
The relevance of discrete-time signal processing has been heightened by the enormous
technological advances in, and the increasing reliance on, digital computers and digital
signal processors since the 1970s. These advances have made it possible to deal efficiently
with sequences and digital data for the following reasons:
1. Digital hardware is efficient in storing and processing digital information. In particular, stored information can be moved from one location to another almost by the
click of a button, and the information can be processed at different locations and at
different times as dictated by user needs.
2. Digital hardware is usually programmable and offers more flexibility than analog
implementations. By modifying program codes we can use the same hardware to
perform different processing tasks. Analog implementations for different tasks tend
to be different and require elaborate testing and tuning.
3. Analog hardware is sensitive to component accuracy, temperature variations, and
thermal noise. Digital hardware is more reliable and more robust in this respect.
Still, despite its advantages, DSP technology may not be ideal for all applications. There
are situations where the signals exhibit rapid variations and require high sampling rates in
order to transform them into sampled signals for discrete-time processing. The requirement
of fast sampling rates generally translates into the requirement of digital hardware that is
capable of operating at high frequencies (or speeds), which in turn translates into costlier
and more challenging implementations. Likewise, faster sampling rates tend to generate
large amounts of data that may require significant storage and processing time and power.
Some of these challenging situations may be better handled by resorting to specialized
analog hardware, or even hybrid solutions combining analog and digital techniques. These
alternative technologies are beyond the scope of this book.
1.8 APPLICATIONS
In this section, we illustrate one application of some of the concepts covered in the chapter
in the context of some practical problems.
1.8.1 Voiced and Unvoiced Speech

We show first how to use signal processing and the concept of the energy of a sequence in
order to classify speech into voiced and unvoiced frames. The processing we perform in
this example is relatively simple since it only involves segmenting a sequence into smaller
sequences and averaging the samples of each segment. Nevertheless, even simple processing tasks of this kind can reveal useful information about a signal and, in the application at
hand, help us classify speech into voiced and unvoiced segments.
Voiced and unvoiced speech are defined as follows. Speech is composed of phonemes,
which are produced by the vocal cords and the vocal tract (which includes the mouth and
the lips) see Fig. 1.10.1 Voiced signals are produced when the vocal cords vibrate during
the pronunciation of a phoneme. Unvoiced signals, by contrast, do not entail the use of
the vocal cords. For example, the only difference between the phonemes /s/ and /z/ or /f/
1 The
source for this public domain image of the vocal apparatus is Wikimedia Commons.
and /v/ is the vibration of the vocal cords. Also, voiced signals tend to be louder like the
vowels /a/, /e/, /i/, /u/, /o/. Unvoiced signals, on the other hand, tend to be more abrupt like
the stop consonants /p/, /t/, /k/.
FIGURE 1.10 A representation of the vocal tract.
The classification of speech into voiced and unvoiced segments is accomplished by

dividing a speech sequence into short frames and by computing the average power of each
frame. The speech in a particular frame is then declared to be voiced if its average power
exceeds a threshold level; otherwise it is declared to be unvoiced speech.
We define the power of a frame as follows. Assume each frame has N samples. For
example, the first frame is assumed to be the frame of index k = 0 and it consists of the
samples:
{x(0), x(1), x(2), . . . , x(N 1)}
(frame k = 0)
Likewise, the frame of index k = 1 contains the samples

{x(N ), x(N + 1), x(N + 2), . . . , x(2N 1)}
(frame k = 1)
and, more generally, the frame of index k contains the samples

{x(kN ), x(kN + 1), x(kN + 2), . . . , x((k + 1)N 1)}
(frame k)
(1.4)
The energy level of a frame is computed by evaluating the squared values of the samples
in the frame and adding these values together to yield:
(k+1)N 1
Energy of frame k = Ek =
n=kN
x2 (n)
(1.5)
11
SECTION 1.8
APPLICATIONS
12
CHAPTER 1
MOTIVATION
If the energy is divided by the number of samples, N , then the result is taken as the average
power of the frame:
Average power of frame k = Pk
1
=
N
(k+1)N 1
x2 (n)
(1.6)
n=kN
It is this power level that we shall compare against the threshold level:
Pk
Pk
threshold = voiced frame
< threshold = unvoiced frame
(1.7)
(1.8)
We divide the waveform of a speech signal into frames of duration 20ms each and compute the average power of each frame. This average power serves as an indication of the
loudness of the frame. We therefore expect higher average power for voiced signals than
for unvoiced signals. The top plot in Fig. 1.11 shows the amplitude values of 14000 samples of a speech signal. In the figure, the amplitude of all samples lie within the interval
[1, 1]. We set the threshold at 0.015 and show two other plots in the figure. The The
middle plot shows only those frames of the original speech signal whose average power
exceeds the threshold level (29% of the frames had their power level above the threshold);
the other frames are set to zero. The bottom plot shows the remaining frames of the original signal whose average power is lower than the threshold level (71% of the frames); the
other frames are set to zero.
Practice Questions:
1. Each frame is 20ms long and consists of N = 175 samples. The speech signal used is 14000
samples long. Into how many frames can the signal be divided? What is the duration of the
entire speech signal in seconds? At what rate in samples/second was the original continuoustime speech signal sampled?
2. The amplitudes of the speech samples lie within the interval [1, 1]. What is the largest
average power we can expect for a frame? If the amplitudes of the samples in a particular
frame lie within the interval [0.2, 0.2], what is the largest energy value we can expect for
this frame?
1.8.2 Estimation of DC Levels

In a variety of situations, users have access to noisy measurements around some constant
level. This situation can arise when different sources observe the same variable and report
their measurements. For example, in a laboratory experiment students may be measuring
the value of the same resistive component in a circuit. Due to measurement and numerical
errors, each student is likely to report a slightly different value for the resistance of the
component. We model this situation as follows. We denote the unknown resistance value
by R and the measurements reported by each student n by r(n). Then
r(n) = R + (n)
where (n) represents the perturbation that is present in the measurement of student n.
Assuming a group of N students, we would end up with N such measurements, say, one
for each student:
r(n) = R + (n), n = 0, . . . , N 1
(1.9)
13
Original speech signal
SECTION 1.8
amplitude
APPLICATIONS
amplitude
Voiced speech
amplitude
Unvoiced speech
n (sample index)
FIGURE 1.11 The top plot shows the original speech signal. The middle plot shows the voiced
frames whose average power exceeds the threshold. The bottom plot shows the unvoiced frames.
The horizontal axis represents the sample index, n.
where we are labeling the students from n = 0 up to n = N 1. The measurements r(n)

can be interpreted as some random variations around the constant value, R. We assume
that the noise components {(n)} are likely to assume both positive and negative values
across all students so that the average value of the {(n)} across these measurements can
be assumed to be zero:
N 1
1 X
(n) = 0
(1.10)
N n=0
Under this condition, if we now average the measurements {r(n)} in (1.9) we find that
R=
1
1 NP
r(n)
N n=0
(1.11)
In other words, the value of R is the average value of the sequence r(n). We refer to R
as the DC level of the sequence r(n). That is, we define the DC level of a sequence as
the average value of its samples. In this way, condition (1.9) amounts to assuming that the
noise components have zero DC level. Of course, in practice, assumption (1.10) may not
hold precisely and expression (1.11) would then provide an approximate estimate for R
rather than its exact value.
Expression (1.11) provides one way to evaluate the DC level of a sequence, r(n). This
solution method requires that we first collect all measurements, {r(n)}, and then average
them. If the time interval between consecutive measurements is long, the waiting time to
14
CHAPTER 1
MOTIVATION
collect the data can be undesirable. To illustrate this fact, let us return to the same circuits
laboratory class and let us assume that the class holds daily sessions. Let us further assume
that we now would like to use all measurements collected by all student groups during the
entire week in order to estimate R. To do so, we would need to wait until all laboratory
sessions have concluded by the end of the week and only then estimate R by averaging
across all measurements. We wonder whether it is possible to develop a procedure that
would allow us to estimate R on the fly as data become available, and then improve upon
this estimate as more data are collected. The discussion that follows derives one such
procedure. The purpose of the presentation is to show that it will often make sense to
look for alternative algorithms to achieve the same objective (that of computing R in this
example). This is because different algorithms will usually exhibit different properties that
are more suitable to one scenario than another. In the current example, expression (1.11)
provides a good solution if all data are available but is not convenient if we need to wait
until all the data becomes available. Let us examine an alternative that will lead to the same
result but that can handle data as they become available.
Recursive Computation
Assume the first group of students who meets on Monday reports its data by the end of the
day. Their measurements can be readily averaged to provide an initial estimate for R, say,
as
N1 1
1 X
r(n)
R1 =
N1 n=0
where we are assuming there are N1 students in the first group. We are also denoting the
estimate from this first set of data by R1 . In the absence of additional measurements, the
above calculation provides an initial estimate for R1 and it will serve as our estimate until
more data become available.
On Tuesday, the second group of students reports its measurements. We would now like
to determine the estimate of R that is based on both sets of data. Specifically, assuming the
second group has N2 students, we are now interested in evaluating the following average:
R1:2 =
1
N1 + N2
N1 +N
X2 1
r(n)
(1.12)
n=0
where we are averaging over the measurements from both groups. We are denoting the
estimate from the first two sets of data by R1:2 ; we are also labeling the students in the
second group from n = N1 up to n = N1 + N2 1. Evaluating R1:2 as above requires
that we re-use the data from the first group of students. One wonders whether the initial
computation of R1 can be useful here. To address this equation, let R2 denote the average
value that is based solely on the measurements from the second group of students:
R2 =
1
N2
N1 +N
X2 1
r(n)
n=N1
Using the definition (1.12) for R1:2 we note that

(N1 + N2 ) R1:2 =
NX
1 1
n=0
r(n) +
N1 +N
X2 1
n=N1
r(n)
15
Dividing both sides by the product N1 N2 we get
SECTION 1.8
N1 + N2
1
R1:2 =
N1 N2
N2
!
N1 1
1
1 X
1
r(n) +
N1 n=0
N1 N2
{z
}
|
|
n=N1
{z
N1 +N
X2 1
R1
R2
and we end up with a relation between R1,2 , R1 and R2 :

R1:2 =
N1
N1 + N2
R1 +
N2
N1 + N2
R2
APPLICATIONS
r(n)
(1.13)
This is a very interesting relation. It tells us that we can evaluate R1:2 , which is the average
based on both sets of data, by combining the individual averages. The measurements
themselves are not needed! This means that we could simply request that each group of
students report only their average value and then we would combine these values as in
(1.13) and obtain R1:2 . A similar argument will show that this result extends to include
the other student groups. For example, if R1:3 is the average value that is based on the
measurements from student groups 1, 2, and 3, then

N1 + N2
N3
R1:3 =
R1:2 +
R3
N1 + N2 + N3
N1 + N2 + N3
In this way, we only need to use the new average R3 (from group 3) and combine it with the
previous aggregate average R1:2 . This construction is an example of a recursive processing
algorithm, where new data (in this case, R3 ) is combined with a previous output of the
algorithm (in this case, R1:2 ) to update the algorithm to a new output value, R1:3 . More
generally, for data from student groups 1 through k, we have
R1:k =
Pk1
=1
Pk
=1
N
N
R1:k1 +
Nk
Pk
=1
Rk
(1.14)
Figures 1.12 and 1.13 illustrate an application of this procedure for the case of 5 students
groups with {12, 10, 11, 10, 9} students in each group. The value of the unknown quantity
R was set at R = 20 and all measurements were subjected to additive Gaussian noise with
zero-mean and unit variance. Figure 1.12 shows the variations of the measurements across
all 52 students; the measurements are represented by the filled circles in the plot, which are
connected by line segments for convenience of visualization. Figure 1.13 shows two plots;
one plot corresponds to the estimates of R that are reported by the individual groups (filled
squares), and the other plot corresponds to the estimates of R that are updated according
to the recursive procedure (1.14) and are indicated by the filled circles in the plot.
Practice Questions:
1. Assume the pairs (Rk , Nk ) assume the values (10.8, 5), (10.7, 8), and (11.1, 7). Evaluate
R1:2 and R1:3 . What would be the estimate of the mean value of the entire set of data?
2. For the same data in the previous part, evaluate R2:3 , which is the estimate of R that is based
on the measurements from groups 2 and 3.
3. According to (1.14), which group of students is weighted more heavily in the determination
of R?
16
22.5
CHAPTER 1
MOTIVATION
nominal
value of R
noisy
measurements
22
21.5
amplitude
21
20.5
20
19.5
19
18.5
18
17.5
10
15
20
25
30
35
n (measurements)
40
45
50
FIGURE 1.12 Noisy measurements by all 52 students. The measurements fluctuate around the
nominal value of R = 20.
20.5
20.4
20.3
cumulative
recursive estimate
R (estimate)
20.2
20.1
20
19.9
estimates by
individual groups
19.8
19.7
19.6
3
k (student group)
FIGURE 1.13 One line corresponds to the estimates of R that are reported by the 5 individual
groups, and the other line corresponds to the estimate of R that is updated according to the recursive
procedure (1.14).
1.9 PROBLEMS
Problem 1.1 Consider the continuous-time signal x(t) = 0.2t + 1, where t is in seconds. Plot the
samples of the sequence x(n) that are obtained by sampling x(t) at multiples of T = 1 second over
the interval t [0, 10].
Problem 1.2 Consider the continuous-time signal x(t) = 0.5 sin 2t + 3 , where t is in seconds. Plot the samples of the sequence x(n) that are obtained by sampling x(t) at multiples of
T = 0.25 second over the interval t [0, 3].
Problem 1.3 Figure 1.14 shows the samples of a sequence, x(n), over the interval 0 n 7.
Plot the quantized version, xq (n), according to the mapping of Fig. 1.5 and assuming = 1/4.
Write down the resulting bit sequence.
x(n)
5/4
3/4
1/2
1/4
2
4
3
1/4
1/2
3/4
FIGURE 1.14 Samples of a sequence x(n) over 0 n 7 for Prob. 1.3.
Problem 1.4 Repeat Prob. 1.3 assuming = 1/2. Determine the quantized version, xq (n), and
write down the corresponding bit sequence.
Problem 1.5 Refer to the quantization procedure described in Fig. 1.5. Assume = 1/4. Determine the samples xq (n) that correspond to the following sequence of bits (assume the first sample
occurs at n = 0):
001110011010010101110000100111
How many samples are represented in this sequence of bits? How many bits would you need to
represent 1024 samples of x(n)?
Problem 1.6 Refer again to the quantization procedure described in Fig. 1.5 and assume = 1/8.
Determine the samples xq (n) that correspond to the following sequence of bits (assume the first
sample occurs at n = 0):
001100110101101111100011110101001000001
How many samples are represented in this sequence of bits? How many bits would you need to
represent 2048 samples of x(n)?
Problem 1.7 Consider the scenario of Prob. 1.5. If it takes 1 microsecond to transmit one byte of
data from location A to location B, how long will it take to transfer the bits representing 1048576
samples of x(n)? Recall that one byte of data consists of 8 bits.
Problem 1.8 Consider the scenario of Prob. 1.7. Assume it took approximately 2.1 seconds to
transmit an amount of data from location A to location B. Approximately, how many samples of
x(n) were transmitted during this operation?
17
SECTION 1.9
PROBLEMS
18
CHAPTER 1
MOTIVATION
Problem 1.9 A speech signal x(t) is recorded for 15 seconds. In one implementation, the signal
is sampled at the rate of 8000 samples per second. If each sample is digitized to 8 bits, what is the
size of the recorded data in bytes? How would your answer change if the sampling rate is raised to
20000 samples per second?
Problem 1.10 A speech signal x(t) is sampled at the rate of 8000 samples per second and digitized
to 8 bits per sample. If a record of size 10MB (mega-bytes) is generated, what is roughly the time
duration of the recorded signal?
Problem 1.11 Consider the sequences
(
x(n) =
0n3
otherwise
0,
(
and

1 n
2
y(n) =

1 n1
4
0,
0n5
otherwise
(a) Plot the samples of x(n).

(b) Plot the samples of y(n).
(c) Plot the samples of the sequence z(n) = x(n)y(n), which are obtained from the point-wise
product of the samples of x(n) and y(n).
(d) Refer to Fig. 1.5 and assume = 1/4. Write the bit sequence for the samples of z(n) over
0 n 7.
Problem 1.12 Consider the sequences
(
x(n) =
y(n) =
0n4
otherwise
0n3
otherwise
0,
(
and

1 n2
2

1 n+1
4
0,
(a) Plot the samples of x(n).

(b) Plot the samples of y(n).
(c) Plot the samples of the sequence z(n) = x(n) + 2y(n).
(d) Refer to Fig. 1.5 and assume = 1/4. Write the bit sequence for the samples of z(n) over
0 n 6.
Problem 1.13 The energy of a real-valued sequence is defined as the sum of the squares of its
samples:
Ex =
x2 (n)
n=
What is the energy of the sequences x(n), y(n), and z(n) defined in Prob. 1.11.
Problem 1.14 What is the energy of the sequences x(n), y(n), and z(n) defined in Prob. 1.12.
Problem 1.15 Refer to the 88 image represented by Fig. 1.2. The intensity of its pixels are listed
in table below: The image is processed as follows. A 2 2 mask or filter moves across the image
from left to right one column at a time, and from top to bottom one row at a time. The mask replaces
the value of the image pixel located at the left-most top corner of the mask by the weighted average
of the pixels covered by the mask; the weights are the values included in the 2 2 mask: The average
is rounded to the closest integer in the range [0, 255]. Find the pixel values of the image that results
from this processing.
12
112
212
92
62
73
121
93
111
21
191
14
234
93
15
0
200
67
134
93
59
39
89
123
51
151
67
178
255
0
12
185
78
178
84
21
137
93
56
87
1
1/4
1/2
-1
-1
1/2
1/4
1
159
69
172
87
93
25
16
112
89
90
23
61
112
97
116
216
145
123
45
59
12
137
181
231
Problem 1.16 Repeat Prob. 1.15 for the alternative mask

Problem 1.17 Consider a discrete-time signal, x(n), defined over the interval n 0. At each time
n, let x
(n) denote the average value of the samples x(k) from time k = 0 up to time k = n, i.e.,
x
(n) =
n
1 X
x(k)
n + 1 k=0
This is an example of one processing algorithm; it acts on the data and generates x
(n). We would
like to motivate an alternative processing algorithm that operates on the data in a recursive manner
to generate the same x
(n). Show that x
(n) satisfies the recursion:
x
(n) =
n
1
x
(n 1) +
x(n)
n+1
n+1
with initial condition x

(0) = x(0). Note that the above algorithm is in terms of the previous value
x
(n 1) and the most recent term in the sequence, x(n). In this way, the second procedure for
evaluating the mean of the sequence does not need to save all prior data; the history of the prior data
is incorporated into x
(n 1) and only the most recent sample, x(n), is needed along with x
(n 1)
to evaluate x
(n).
Problem 1.18 Consider the following samples of x(n):
x(0) = 1, x(1) = 0.5, x(2) = 0.4, x(3) = 1.0, x(4) = 0.8, x(5) = 0.75
Compute the corresponding samples of x
(n) using the two procedures described in Prob. 1.17.
Problem 1.19 Consider a discrete-time signal whose samples at the time instants n = 0, 1, 2, 3, 4, 5, 6
are given by
x(0) = a, x(1) = b, x(2) = c, x(3) =?, x(4) = e, x(5) = f, x(6) = g
where the sample at time n = 3 is missing. Suggest one by which you would process the available
samples of x(n) to come up with an estimate for the missing sample.
Problem 1.20 Consider Prob. 1.19. Suggest at least two other ways by which you would process
the available samples of x(n) to come up with an estimate for the missing sample.
19
SECTION 1.9
PROBLEMS
CHAPTER
Fundamental Sequences
Complex numbers play a critical role in characterizing discrete-time signals and systems. For example, it will be seen in later chapters that complex numbers are needed
to describe the frequency content of a sequence and the frequency response of a system.
Complex numbers are also needed to describe some basic sequences in the time domain.
Accordingly, this chapter provides a brief review of complex numbers and explains how
they are useful in describing some important sequences, such as the complex exponential
sequence.
2.1 COMPLEX NUMBERS

Every complex number has the form
z = a + jb
(2.1)
(2.2)
where a and b are real numbers and

j=
In other words, j 2 = 1. The number a is called the real part of z and the number b is
called the imaginary part of z. We sometimes write
a = Re(z),
b = Im(z)
(2.3)
where the notation Re() and Im() refers to the real and imaginary components of their
argument.
Every complex number z can be expressed in an alternative form known as the polar form in terms of the magnitude of the number and its phase. Specifically, z can be
expressed as
z = ej ,
(2.4)
where denotes the magnitude of z; it is a nonnegative real number that is computed as

follows:
= a2 + b 2
(2.5)
and denotes the phase of z and is measured in radians. The phase can always be
chosen to lie within the interval [, ]; its value should be determined from the scalars
{a, b} with care as follows. First, we determine the angle
h i
,
2 2
21
Copyright
22
whose arctan is given by
CHAPTER 2
= arctan
FUNDAMENTAL
SEQUENCES

b
a
(2.6)
Then we may need to adjust by adding to it depending on which quadrant the number
z belongs to in the complex plane see Fig. 2.1. The phase is obtained from according
to the following rule:
, if z in quadrants I or IV, i.e., if Re(z) > 0

, if z in quadrant III, i.e., if Re(z) < 0 and Im(z) < 0
+ , if z in quadrant II, i.e., if Re(z) < 0 and Im(z) > 0
(2.7)
The correction to (i.e., the addition of ) is chosen in such a way that the resulting
phase will always lie within the interval [, ]. It is customary to express the phase of
a complex number using the arctan notation (2.6); it is to be understood, however, from the
above discussion, that the actual phase should be selected according to (2.7).
Im
II
=+
Re
III
IV
FIGURE 2.1 Division of the complex plane into four quadrants I, II, III, and IV.
Sometimes, the polar representation (2.4) of z is alternatively expressed as

z = |z| ejz
(2.8)
where |z| is used to denote the amplitude of z and z is used to denote its phase see
Fig. 2.2. Moreover, the letter e denotes the number whose natural logarithm is equal to 1:
ln e = 1
Example 2.1 (Polar form)

Consider the two complex numbers
z1 = 1 + j
and
z2 = 1 j
(2.9)
23
SECTION 2.1
Im
COMPLEX
NUMBERS
z
b
|z|
z
Re
FIGURE 2.2 Standard and polar form representations of a complex number z.
Both complex numbers have the same magnitude, =
2, and lead to the same phase angle
= arctan(1) = /4
However, z1 and z2 are distinct numbers: z1 lies in quadrant I while z2 lies in quadrant III see
Fig. 2.3. The correct phase angle for z2 is therefore
=
so that
z1 =
2 ej 4
3
=
4
4
and
z2 =
2 ej
3
4
Im
z1 = 1 + j
1
z2 = 1 j
Re
FIGURE 2.3 Location of z1 = 1 + j and z2 = 1 j in the complex plane.
Complex Conjugation
The complex conjugate of a complex number z, as in (2.1), is denoted by z and is defined
as
z = a jb
(2.10)
24
CHAPTER 2
FUNDAMENTAL
SEQUENCES
In other words, the imaginary part of z is negated. It is clear that both z and z have the
same magnitude. However, the phase of z is the opposite of the phase of z. Specifically,
if the polar form of z is given by (2.4) or (2.8), then
z = ej ,
(2.11)
z = |z| ejz
(2.12)
or
2.2 BASIC SEQUENCES

There are a handful of sequences that arise frequently in the study of discrete-time signals
and systems. We collect these sequences in the current section and comment on some of
their properties. Several of the sequences are so common (like the unit-sample sequence
and the unit-step sequence) that they have their own standard notation.
Unit-sample sequence. The unit-sample sequence is denoted by (n) and is defined

as follows:

1
n=0
(n) =
(2.13)
0
n 6= 0
In other words, the sequence is zero everywhere except at time n = 0 when it is equal to
one. The unit-sample sequence is often referred to more simply as the impulse sequence.
Unit-step sequence. The unit-step sequence is denoted by u(n) and is defined as follows:

1
n0
u(n) =
(2.14)
0
n<0
In other words, the sequence is equal to one for all nonnegative time instants. Note that it
is related to the unit-sample sequence in the following manner:
(n) = u(n) u(n 1)
and
u(n) =
n
X
(k)
(2.15)
k=0
The unit-step sequence is often referred to more simply as the step sequence. Fig. 2.4
plots the terms of the unit-sample and unit-step sequences over the interval 3 n 6.
Real exponential sequence. The real exponential sequence is defined as follows:
x(n) = An
where both A and are real.
(2.16)
25
(n)
SECTION 2.2
BASIC
SEQUENCES
u(n)
1
FIGURE 2.4 The top plot shows the terms of the unit-sample sequence, (n), over the interval
3 n 6, while the bottom plot shows the terms of the unit-step sequence, u(n), over the same
interval of time.
Example 2.2 (Decaying exponential sequence)

Figure 2.5 plots the terms of two real exponential sequences over the interval 5 n 10 for two
choices of that are less than one in magnitude. In one case, is positive and equal to 0.8 so that
x(n) = (0.8)n
while in the other case is negative and equal to 0.8 and, hence,
x(n) = (0.8)n
It is immediate to realize that when is less than one in magnitude, the samples of the exponential
sequence grow in magnitude for n < 0 (negative time) and decay in magnitude for n 0 (positive
time). It is also seen that when < 0, the samples of the exponential sequence alternate between
positive and negative values.
26
x(n)=(0.8)n
CHAPTER 2
FUNDAMENTAL
SEQUENCES
2
1
0
5
10
10
n
n
x(n)=(0.8)
2
1
0
1
2
5
3
n
FIGURE 2.5 The top plot shows the terms of the real exponential sequence (0.8)n over the
interval 5 n 10, while the bottom plot shows the terms of the real exponential sequence
(0.8)n over the same interval of time.
Example 2.3 (Two-sided exponential sequence)

Consider the case A = 1 and = 1/2. Then
n
x(n) =
1
2
Tables 2.1 and 2.2 list the samples of this sequence for both cases of positive and negative values
of n. It is seen in Table 2.1 that the samples of x(n) decay for positive n and in Table 2.2 that the
samples of x(n) grow for negative n.
TABLE 2.1 Samples x(n) = (0.5)n over the interval 0 n 10.

n
x(n)
10
1/2
1/4
1/8
1/16
1/32
1/64
1/128
1/256
1/512
1/1024
TABLE 2.2 Samples of x(n) = (0.5)n over the interval 10 n 1.

n
x(n)
-1
-2
-3
-4
-5
-6
-7
-8
-9
-10
16
32
64
128
256
512
1024
27
Sinusoidal sequences. The sinusoidal sequences are defined by the expressions:
SECTION 2.2
x(n) = A sin(o n + o )
(2.17)
x(n) = A cos(o n + o )
(2.18)
or
for some real-valued quantities {A, o , o }. The variable A is called the amplitude of the
sinusoidal signal and the variable o is the phase of the signal. As we are going to see in
Sec. 3.3, the variable o denotes the angular frequency of the sinusoidal signal.
Example 2.4 (Two sinusoidal plots)

Figure 2.6 plots the terms of two sinusoidal sequences over the interval 10 n 10 using A = 1.
In one case, o = /5 and o = 0 so that
x(n) = sin
n
while in the other case o = /5 and o = /3 so that

x(n) = sin
n

3
x(n)=sin( n/5)
1
0.5
0
0.5
1
10 9 8 7 6 5 4 3 2 1
0 1 2 3
n
x(n)=sin( n/5 +/3)
9 10
9 10
1
0.5
0
0.5
1
10 9 8 7 6 5 4 3 2 1
0
n
FIGURE 2.6 The top plot shows the terms of the sinusoidal sequence x(n) = sin(n/5) over
the interval 10 n 10, while the bottom plot shows the terms of the sinusoidal sequence
+ 3 ) over the same interval of time.
x(n) = sin( n
5
Complex exponential sequence. This sequence plays a fundamental role in the study
of discrete-time signals and systems. The general form of a complex exponential sequence
BASIC
SEQUENCES
28
is
x(n) = An
CHAPTER 2
FUNDAMENTAL
SEQUENCES
(2.19)
where is now a complex number and A is either real- or complex-valued. Assume, for
generality, that both A and are complex-valued and introduce their polar representations:
A = |A|ejo ,
= ||ejo , o , o [, ]
Then the expression for x(n) can be re-written in the equivalent form
x(n) = |A| ||n ej(o n+o )
(2.20)
which is the general form for a complex exponential sequence. An important special case
is the sequence
x(n) = ejo n
(2.21)
which corresponds to the choices A = 1 and = ejo .
Example 2.5 (Magnitude and phase plots)

Figure 2.7 shows the magnitude and phase plots of the complex exponential sequence that corresponds to the choices A = ej/2 and = 0.5ej/3 , i.e.,
n
x(n) =
1
2
ej(
n + )
3
2
over the interval 5 n 5. Observe that we are now using two plots to illustrate the sequence:
one for the magnitude of the samples and the other for their phases. This is because the samples
in this example assume complex values as opposed to real values (as was the case with the earlier
examples).
One-sided sequences. The unit-step sequence u(n) introduced in (2.14) plays a useful
role in defining one-sided sequences. Consider, for example, the exponential sequence
x(n) = (0.5)n
This sequence is defined for all values of n since the samples of x(n) exist for nonnegative
values of n as well as for negative values of n, as was illustrated earlier in Tables 2.1
and 2.2. On the other hand, the new sequence
n
x(n) = (0.5) u(n)

with the unit-step sequence multiplying from the right-hand side, is again a real exponential sequence but one that has nonzero values only for n 0; the samples of the sequence
are now zero for all n < 0 because u(n) is zero for n < 0. This example illustrates a useful
property of the unit-step sequence u(n), namely, it allows us to define one-sided sequences.
29
Magntiude plot
SECTION 2.3
30
POLAR
PLOTS
20
10
0
5
0
1
n
Phase plot
radians
6
4
2
0
2
4
5
0
n
FIGURE 2.7 The top plot shows the magnitude and the bottom plot shows the phase of the terms
n
of the exponential sequence x(n) = (0.5)n ej( 3 + 2 ) over the interval 5 n 5.
Example 2.6 (One-sided exponential sequence)

We reconsider the two-sided exponential sequences from Example. 2.2, and employ the unit-step
sequence to transform them into one-sided sequences, say, as
y(n) = (0.8)n u(n)
y(n) = (0.8)n u(n)
and
Figure 2.8 plots these sequences over the same interval of time, 5 n 10, as in Fig. 2.5.
Observe how the samples over negative time are all zero now.
2.3 POLAR PLOTS

It is clear that providing graphical illustrations of sequences that assume real-values is
generally straightforward. Nevertheless, illustrating complex-valued sequences is more
demanding. For example, in Fig. 2.7 we had to resort to two separate plots: one for the
magnitude of the samples and the other for the phase of the same samples.
Another convenient way to plot complex exponential sequences is by means of a polar
plot, which is a plot of the location of the samples over the complex plane. The polar plot
is best illustrated through an example. Consider the one-sided sequence
x(n) = ej 4 n u(n)
The first 8 terms of the sequence, corresponding to the time instants n = 0, 1, . . . , 7, are
given by
o
n
3
5
3
7
1, ej 4 , ej 2 , ej 4 , 1, ej 4 , ej 2 , ej 4
All 8 terms are complex numbers with unit magnitude and, hence, they all lie on a circle
of unit radius in the complex plane. Moreover, the terms are /4 radians apart on the unit
30
CHAPTER 2
FUNDAMENTAL
SEQUENCES
y(n)=(0.8) u(n)
3
0
5
10
10
n
n
y(n)=(0.8) u(n)
2
2
5
FIGURE 2.8 The top plot shows the terms of the one-sided exponential sequence y(n) =
(0.8)n u(n) over the interval 5 n 10, while the bottom plot shows the terms of the onesided exponential sequence y(n) = (0.8)n u(n) over the same interval of time. These plots should
be compared with the ones shown in Fig. 2.5 for the two-sided versions of these sequences.
circle; their phases increase from 0 to /4 to /2 and so forth. Thus, as illustrated in

Fig. 2.9, if we draw the samples in the complex plane, they will all lie on the unit circle
and they will be /4 radians apart from each other. The curved arrow in Fig. 2.9 is used to
indicate that the terms of the sequence cover the circle in a counter-clockwise direction.
Im
n=2
n=1
n=3
n=4
n=5
n=0
Re
n=7
n=6
FIGURE 2.9 A polar plot of the first 8 samples of the sequence x(n) = ej 4 n u(n).
31
SECTION 2.4
Example 2.7 (Decaying complex exponential)
SYMMETRY
RELATIONS
Figure 2.10 shows the first four terms of a second sequence, which is defined by
n
x(n) =
1
2
ej 4 n u(n)
This is an interesting sequence and its four terms are given by

1,
1 j 4 1 j 2 1 j 3
e , e , e 4
2
4
8
Observe now that the amplitudes of the successive samples decay by a factor of 1/2 each time,
moving from 1 to 1/2 to 1/4 and so forth. Thus the samples lie on circles of radii 1, 1/2, 1/4, and
so on, although they continue to be /4 radians apart from each other. The numbers {0, 1, 2, 3} in
the figure refer to the sample times n = 0, 1, 2, 3.
Im
n=1
n=2
n=0
n=3
1
4
FIGURE 2.10
1
2
Re
A polar plot of the first 4 samples of the sequence x(n) =

1 n
2
ej 4 n u(n).
2.4 SYMMETRY RELATIONS

We shall often encounter sequences that exhibit certain symmetry properties such as even
and odd symmetries.
Even sequences. A real-valued sequence, x(n), is said to be even if its samples satisfy
the following relation
x(n) = x(n)
for all n
(even sequence)
(2.22)
32
That is, the sequence x(n) is symmetric about the vertical axis.
CHAPTER 2
FUNDAMENTAL
SEQUENCES
Odd sequences. A real-valued sequence, x(n), is said to be odd if its samples satisfy
the following relation
x(n) = x(n)
for all n
(odd sequence)
(2.23)
That is, the sequence x(n) is symmetric about the origin of the cartesian plane.
Decomposition. Every real-valued sequence, x(n), can be decomposed into the sum of
an even component and an odd component, say as
x(n) = xe (n) + xo (n)
(2.24)
where xe (n) is an even sequence that denotes the even part of x(n), and xo (n) is an odd
sequence that denotes the odd part of x(n). We can determine expressions for xe (n) and
xo (n) in terms of x(n) by first using (2.24) to write
x(n) =
=
xe (n) + xo (n)
xe (n) xo (n)
so that
xe (n) =
1
[x(n) + x(n)]
2
(2.25)
xo (n) =
1
[x(n) x(n)]
2
(2.26)
and
Example 2.8 (Even and odd sequences)

The sequence
x(n) = sin
is odd while the sequence
x(n) = cos
n
3
n
3
is even in view of the properties sin() = sin() and cos() = cos() for any . Figure 2.11
plots the two sequences over the interval 5 n 5.
Now consider the sequence
x(n) = (0.5)n u(n)
This sequence is neither odd nor even. However, we can decompose it as the sum of even and odd
sequences as follows:
x(n) = xe (n) + xo (n)
33
x(n)=sin( n/3)
SECTION 2.4
SYMMETRY
RELATIONS
0.5
0
0.5
5
0
n
x(n)=cos( n/3)
0
n
1
0.5
0
0.5
1
5
FIGURE 2.11 The top plot shows the odd sequence x(n) = sin(n/3) while the bottom plot
shows the even sequence x(n) = cos(n/3). The dotted line in the top figure going through the
origin is meant to illustrate the symmetry of the odd sequence around the origin n = 0. Both
sequences are shown over the interval 5 n 5.
where
xe (n)
=
=
1
[x(n) + x(n)]
2

1
(0.5)n u(n) + (0.5)n u(n)
2
(
(0.5)|n|+1 ,
1,
=
xo (n)
=
=
1
[x(n) x(n)]
2

1
(0.5)n u(n) (0.5)n u(n)
2
8
>
<
n 6= 0
n=0
(0.5)n+1 ,
0,
>
: (0.5)n+1 ,
n>0
n=0
n<0
Figure 2.12 plots the resulting even and odd sequences of x(n) over 10 n 10.
Conjugate symmetric sequences. A complex-valued sequence, x(n), is said to be

conjugate symmetric if its samples satisfy the following relation
x(n) = x (n)
for all n
(conjugate symmetric sequence)
(2.27)
That is, if we conjugate the entries of the sequence and reflect the conjugated sequence
around the vertical axis, we arrive back at x(n). If we express x(n) in terms of its real and
34
x (n)
e
CHAPTER 2
FUNDAMENTAL
SEQUENCES
0.75
0.5
0.25
0
10
0
n
xo(n)
10
0
n
10
0.25
0.15
0.05
0.05
0.15
0.25
10
FIGURE 2.12 The top plot shows the even component of the exponential sequence x(n) =
(0.5)n u(n), while the bottom part shows the odd component of the same sequence. Note that while
x(n) is one-sided and its samples are nonzero over n 0, the odd and even sequences, xe (n) and
xo (n), have nonzero samples over all n. The plots show the samples over 10 n 10 only.
imaginary parts, say

x(n) = xR (n) + jxI (n)
(2.28)
where xR (n) and xI (n) are both real-valued, then

x (n) = xR (n) jxI (n)
so that the conjugate symmetry property (2.27) translates into
xR (n) = xR (n)
and
xI (n) = xI (n)
(2.29)
That is, the real part of x(n) should be an even sequence and the imaginary part of x(n)
should be an odd sequence. In addition, if we introduce the polar representation of x(n),
say
p
x(n) = (n) ej(n) , (n) = |xR (n)|2 + |xI (n)|2
then
x (n) = (n) ej(n)
and it again follows from the conjugate symmetry property (2.27) that we must have
(n) = (n)
and
(n) = (n)
(2.30)
That is, the magnitude sequence should be even and the phase sequence should be odd.
Example 2.9 (Conjugate symmetric sequence)
The sequence
x(n) = cos
n
+ j sin
n
35
is conjugate symmetric since

x (n)
=
=
=
=
(n)
(n)
j sin
3
3
n
n
cos
j sin
3
3

n
n
cos
+ j sin
3
3
x(n)
SECTION 2.5
ENERGY AND
POWER
SEQUENCES
cos
2.5 ENERGY AND POWER SEQUENCES

Energy sequences. The energy of a sequence x(n) is defined by
Ex =
n=
|x(n)|2
(2.31)
where the notation | | denotes the magnitude of its argument. When x(n) is real-valued,
the notation |x(n)| refers to the absolute value of the sample x(n). On the other hand,
when x(n) is complex-valued, the notation |x(n)| refers to the magnitude of x(n). Thus,
the energy of a sequence is given by the sum of the squared magnitude of all its samples.
When Ex < , we say that the sequence is an energy sequence. In other words, energy
sequences have finite energy.
Example 2.10 (Step and exponential sequences)
The step sequence, x(n) = u(n), is not an energy sequence since
X
n=
|u(n)|2 =
X
n=0
On the other hand, the exponential sequence

n
x(n) =
1
2
|x(n)|2
u(n)
is an energy sequence since
X
n=
n
X
1
n=0
=
=
1
1 1/4
4
<
3
In the second calculation, we used the fact that the values {(1/4)n , n 0} are the terms of a
geometric series with ratio 1/4 see Example 2.11 below.
36
Example 2.11 (Geometric series)
CHAPTER 2
FUNDAMENTAL
SEQUENCES
Consider the sequence

x(n) = ar n u(n)
where a and r are real. The successive samples of the sequence are given by
{a, ar, ar 2 , ar 3 , ar 4 , ar 5 , . . .}
Note that the first term is a and all successive terms are obtained by multiplying the preceding term
by the factor r. In this way, there is a constant ratio r between the successive terms of the sequence.
We say that x(n) represents a geometric sequence with ratio r and initial term a. For such sequences,
there is a closed-form expression for the sum of its terms, namely,
N
X
n=0
ar n = a(1 + r + r 2 + r 3 + . . . + r N ) = a
1 r N+1
1r
To establish the result, we let SN denote the sum of the first N + 1 terms:
SN = a(1 + r + r 2 + r 3 + . . . + r N )
Then, multiplying SN by r gives
rSN = a(r + r 2 + r 3 + . . . + r N+1 )
so that
SN rSN = a(1 r N+1 )
and, consequently,

SN = a
1 r N+1
1r
(sum of first N + 1 terms)
(2.32)
The above result holds for all finite values of N regardless of the value of r; in particular, if r = 1,
then SN = (N + 1)a. Now assume |r| < 1 and let S denote the geometric series
S = a(1 + r + r 2 + r 3 + . . .) =
lim SN
Then, since |r| < 1, it follows that r N+1 converges to zero in the expression for SN as N .
Consequently, the series converges to
S=
a
1r
(series)
(2.33)
Power sequences. The average power of a sequence x(n) is defined by
Px = lim
N
P
1
|x(n)|2
2N + 1 n=N
(2.34)
That is, we compute the energy of the sequence x(n) over the symmetric interval N
n N , normalize the result by the number of terms in this interval, which is 2N + 1, and
then evaluate the limit as the size of the interval increases indefinitely. When Px < ,
we say that the sequence is a power sequence. In other words, power sequences have finite
power.
37
Example 2.12 (Step sequence)
SECTION 2.6
Although the step sequence, x(n) = u(n), is not an energy sequence it is nevertheless a power
sequence since, for any N ,
N
N
X
X
1
N +1
1
1 =
|u(n)|2 =
2N + 1 n=N
2N + 1 n=0
2N + 1
so that the limit, as N , is Px = 1/2.
2.6 SIGNAL TRANSFORMATIONS

Sequences often appear in transformed versions and it is useful to become acquainted
with the following common signal transformations, which are illustrated in Fig. 2.13 for a
sequence x(n) whose samples are zero outside the indicated interval 5 n 5. If x(n)
is a given sequence, then:
1. x(n) corresponds to a sequence that is obtained from x(n) by reflecting it about
the vertical axis; this operation is also known as time-reversal.
2. x(n 1) corresponds to a sequence that is obtained from x(n) by shifting it by one
sample to the right.
3. x(n + 1) corresponds to a sequence that is obtained from x(n) by shifting it by one
sample to the left.
4. x(n + 1) corresponds to a sequence that is obtained from x(n) by first reflecting
x(n) about the vertical axis and then shifting it to the right by one sample; this is
not the same as shifting first to the right and then reflecting about the vertical axis.
5. x(n 1) corresponds to a sequence that is obtained from x(n) by first reflecting
it about the vertical axis and then shifting it to the left by sample. Again, this is not
the same as shifting first to the left and then reflecting about the vertical axis.
More generally, consider a sequence x(n) and define a new sequence y(n) that is obtained
from x(n) as follows:
y(n) = x(an + b)
(2.35)
for some integer values a and b. The five signal transformations listed above correspond to
particular choices of a and b. For arbitrary integers a and b, this is how we can obtain the
plot of y(n) from the plot of x(n). Introduce the variable
m = an + b
(2.36)
y(n) = x(m)
(2.37)
Then
Specifically, the following facts hold:
1. The value of y(n) at n = 0 is the value of the sequence x(m) at time m = b:
y(0) = x(b)
That is, we set n = 0 in an + b and arrive at the argument m = b for x(m).
SIGNAL
TRANSFORMATIONS
38
CHAPTER 2
FUNDAMENTAL
SEQUENCES
x(n)
0.5
0
-5
-4
-3
-2
-1
1
x(-n)
0.5
0
-5
-4
-3
-2
-1
1
x(n-1)
0.5
0
-5
-4
-3
-2
-1
1
x(n+1)
0.5
0
-5
-4
-3
-2
-1
1
x(-n+1)
0.5
0
-5
-4
-3
-2
-1
1
x(-n-1)
0.5
0
-5
-4
-3
-2
-1
0
n
FIGURE 2.13 Plots of various signal transformations of the sequence x(n) = (3/4)n u(n) over
the interval 5 n 5.
2. The value of y(n) at n = 1 is the value of the sequence x(m) at time m = a + b:

y(1) = x(a + b)
That is, we set n = 1 in an + b and arrive at the argument m = a + b for x(m).
3. The value of y(n) at n = 1 is the value of the sequence x(m) at time m = a + b:
y(1) = x(a + b)
That is, we set n = 1 in an + b and arrive at the argument m = a + b for x(m).
4. And so forth.
In more formal terms, we can describe the procedure for generating the plot of y(n) from
the plot of x(n) as follows:
1. We draw the n axis.
2. We define a new variable m = an + b and draw a new axis mapping the values of n
to the values of m. We also determine the orientation of this new m axis.
3. We plot x(m) versus m. This is the same plot as x(n) versus n except that it is done
on the m axis.
4. We replace the horizontal and vertical axes in the m domain by the horizontal and
vertical axes in the n domain. We only keep the samples that are defined for valid n
time instants.
We illustrate the construction by means of an example.
Example 2.13 (Plotting a transformed sequence)

Consider the sequence x(n) shown in the top plot of Fig. 2.14. It consists of 6 nonzero samples
between n = 1 and n = 4. We wish to plot the sequence that results from transforming x(n) as
follows:
y(n) = x(2n + 3)
The first step is to draw the horizontal m axis that relates to the original n axis via
m = 2n + 3
The m axis is shown in the middle plot. Note that its orientation is reversed relative to the direction
of n. The top and middle plots also show how the values of n correspond to the values of m. For
example, n = 0 is mapped to m = 3, n = 1 is mapped to m = 1, n = 1 is mapped to m = 5,
and so forth.
The middle plot shows the same sequence x(m) against the m axis; it is exactly the same sequence as the original x(n) except that now the samples are plotted against the m axis. Finally, the
bottom plot shows y(n) versus n. All we are doing here is replace the m axis from the middle plot
by the n axis and keep the samples from the middle plot that correspond to valid values of n. Thus
note, for example, that the samples of x(m) that occur at m = 4, 2, 0 do not map to samples of
y(n) since these values of m do not correspond to valid integer values of n in the transformation
m = 2n + 3. Therefore, only 3 of the original samples of x(n) are kept in y(n); these are the
samples marked with circles around their endpoints. The other samples are removed.
39
SECTION 2.6
SIGNAL
TRANSFORMATIONS
40
CHAPTER 2
FUNDAMENTAL
SEQUENCES
x(n) n
4
1
0
x(m) m
m = 2n + 3
3 2 1
2 3 4 5
y(n) n
2
1
FIGURE 2.14 The top plot shows a sequence x(n). The middle plot shows a new horizontal axis
m that is related to the axis n via the transformation m = 2n + 3. The same sequence x(m) is
plotted against the new m axis; observe how the orientation of the axis m is reversed relative to that
of the n axis. The bottom plot shows the sequence y(n) = x(2n + 3). Observe that y(n) retains
only 3 of the original samples of x(n); these are marked with circles around their endpoints.
2.7 APPLICATION: SAVINGS ACCOUNT

in the context of a practical problem. Specifically, we show how a one-sided exponential
sequence, and transformations thereof, arise in the context of a bank savings account.
Thus, assume a client opens a savings account with an initial deposit of US$1000. The
time at which the account was created is selected to be the origin of time, say, as n = 0.
Let y(n) denote the amount of funds in the account at a the beginning of a generic year n.
Then, obviously,
y(0) = US$1000
and
y(n) = 0 for all n < 0
where n < 0 covers the period prior to the creation of the account. Assume further that the
return rate on the account is 5% per year. Then the amount of funds that will be present at
41
the start of year n = 1 will be:
SECTION 2.7
APPLICATION
y(1) = y(0) (1 + 0.05) = 1.05 1000 = US$1050

Likewise, the amount of funds that will be present at the start of year n = 2 will be
y(2) = y(1) (1 + 0.05) = x(0) (1 + 0.05)2 = US$1102.50
More generally, the amount of funds that will be present at the start of a generic year n will
be
y(n) = y(0) (1.05)n = 1000 (1.05)n , n 0
If we incorporate the fact that y(n) is zero for negative time, we arrive at the expression
y(n) = 1000 (1.05)n u(n)
(2.38)
which describes a one-sided exponential sequence. In general, for an initial deposit value
of D at time n = 0, and assuming an annual return rate of %, the amount of funds that
will be available at the start of year n would be

n
u(n)
(2.39)
y(n) = D 1 +
100
Figure 2.15 depicts graphically the evolution of funds in a savings account.2
FIGURE 2.15 A depiction to illustrate the growth of funds in a savings account.
Continuing with our example (2.38), let us now consider the sequence z(n) = y(n 1).
It is obtained from shifting the samples of y(n) by one unit of time to the right. What would
the interpretation of the samples of z(n) be in the context of the savings account? To see
the relation, we write down the first few samples of the sequences y(n) and z(n):
It is clear from the data in the table that the value of z(n), at the start of year n, can be
interpreted as corresponding to the amount of funds that would be present in the account
2 Source
of this placeholder image is istockphoto.com.
42
CHAPTER 2
FUNDAMENTAL
SEQUENCES
n
y(n)
z(n)
0
US$1000
0
1
y(1)
US$1000
2
y(2)
y(1)
3
y(3)
y(2)
4
y(4)
y(3)
5
y(5)
y(4)
...
...
...
had the initial deposit of US$1000 been made at year n = 1 (and not at year n = 0, as was
the case with y(n)).
In a similar manner, let us consider the alternative sequence w(n) = y(2n). It is obtained from scaling the n axis by a factor of 2. Again, we are interested in the interpretation
of the samples of w(n) in the context of the savings account. To see the relation, we write
down the first few samples of the sequences y(n) and w(n):
n
y(n)
w(n)
0
US$1000
US$1000
1
y(1)
y(2)
2
y(2)
y(4)
3
y(3)
y(6)
4
y(4)
y(8)
5
y(5)
y(10)
...
...
...
Thus, observe that

w(1)
w(2)
w(3)
w(4)
..
.
= y(2) = 1000 (1.05)2
2

= y(4) = 1000 (1.05)4 = 1000 (1.05)2
3

= y(6) = 1000 (1.05)6 = 1000 (1.05)2
4

= y(8) = 1000 (1.05)8 = 1000 (1.05)2
.
= ..
so that, more generally,

w(n) = 1000 (1.1025)n u(n)
where we used (1.05)2 = 1.1025. It is clear that the value of w(n), at the start of year
n, can be interpreted as corresponding to the amount of funds that would be present in the
account when the initial deposit of US$1000 is still made at year n = 0 while the return
rate is increased from 5% to 10.25% per year. Observe that the return rate increased by a
factor larger than 2.
Practice Questions:
1. Starting with an initial deposit of US$500 at year n = 0, how much funds will be available at
the start of year n = 30 assuming an annual return rate of 2%.
2. Starting with an initial deposit of US$500 at year n = 3, how much funds will be available at
the start of year n = 30 assuming the same annual return rate of 2%.
3. Starting with an initial deposit of US$500 at year n = 0, what should the annual return rate
be such that the amount of funds that are present at the start of year n = 10 would be equal
to the amount of funds that are present at the start of year 30 at 2% annual return?
43
2.8 PROBLEMS
SECTION 2.8
PROBLEMS
Problem 2.1 Consider the complex numbers
1
3
z1 = j
,
2
2
z2 =
1
3
+j
2
2
Find the polar representations of the numbers: z1 , z2 , z1 z2 , z1 /z2 , z12 z2 , and |z1 |3 z2 .
Problem 2.2 Consider the complex numbers
3
1
z1 = + j
,
4
4
z2 =
3
1
j
4
4
Find the polar representations of the numbers: z1 , z2 , z1 z2 , z12 /z2 , z1 z23 , and |z1 |z2 .
Problem 2.3 Which of the following identities is incorrect?
(a) (3n 6) = (3n + 6).
(b) (n) = (5n).
(c) (5n 1) = (4n + 3).
(d) (5n 1) = (5(n 1)).
Problem 2.4 Which of the following identities is correct?
(a) 2(3n) = 2(n).
(b) (2n 2) = (n + 1).
(c) (5n + 10) = (2n 4).
(d) (5n) = (1)n (n).
Problem 2.5 Plot in polar coordinates the terms of the sequence x(n) =
What is the energy and average power of this sequence?
Problem 2.6 Plot in polar coordinates the terms of the sequence x(n) =
What is the energy and average power of this sequence?
Problem 2.7 Let
n
1
2
x(n) =
2
4
2
2
+j
2
4
n
2n
2
2
u(n).
u(n).
!

n+
3
4 1 +j 3
2
2
and denote its polar representation by x(n) = (n)ej(n) , where both and are functions of n.
(a) Determine (n) and (n).
(b) Determine the even and odd parts of (n).
(c) Determine the even and odd parts of (n).
Problem 2.8 Let
n2
x(n) =
1
4
!

n
1
3
6
3
j
2
2
and denote its polar representation by x(n) = (n)ej(n) .

(a) Determine (n) and (n).
(b) Determine the even and odd parts of (n).
(c) Determine the even and odd parts of (n).
44
CHAPTER 2
FUNDAMENTAL
SEQUENCES
Problem 2.9 Express the complex exponential sequence

n
1
3
+j
x(n) = (1 j)
2
2
in polar form and plot its terms at the time instants n = 1, 0, 1.
Problem 2.10 Express the complex exponential sequence

1
1
+j
2
2
x(n) =
2
1
3
j
2
2
2n
in polar form and plot its terms at the time instants n = 1, 0, 1.

Problem 2.11 Find the odd and even components of the sequence x(n) = (0.5)n u(n 1).
Plot x(n) and find its energy as well.
Problem 2.12 Find the odd and even components of the sequence x(n) = (1/4)n1 u(n + 2).
Plot x(n) and find its energy as well.
Problem 2.13 Let x(n) =

1 |n|
3
Problem 2.14 Let x(n) = 13
n
7
sin
n2
. Is x2 (n) even?
sin 5 n . Is x3 (n) odd?
Problem 2.15 Consider the sequence

n
n1
1
3
x(n) =
u(n 1) +
1
2
u(n 2)
(a) Find its energy and average power.

(b) Let y(n) = x(2n 3). For what values of n is y(n) zero?
"
n1 #
x(n) = 1 +
1
2
u(n 2)
(a) Find its energy and average power.

(b) Let y(n) = x(2n + 3). For what values of n is y(n) zero?
(c) Find the energy of x(2n).
Problem 2.17 Give, if possible, examples of nonzero sequences x(n) such that:
(a) x(n) and x(n 2) are identical.
(b) x(n)x(n + 1) = 0 for n = 0, 1, 2.

(c) x(n) + x (n) = cos

3
(n 1) .
Problem 2.18 Give, if possible, examples of nonzero sequences x(n) such that:
(a) x(n) and x(n + 2) are identical.
(b) x(n)x(n 3) = 0 for n = 0, 1, 2.
(c) x(n) + x (n) = sin

4
(n + 1) .
Problem 2.19 The DC level of a sequence x(n) is defined as the average value
X
1
x(n)
2N + 1 n=N
N
x
=
lim
45
Let x(n) be an odd sequence. Show that its DC level is zero.
SECTION 2.8
Problem 2.20 Consider an arbitrary sequence x(n) with even and odd parts denoted by xe (n) and
xo (n), respectively. Show that the DC level of x(n) coincides with the DC level of its even part.
Problem 2.21 Determine the DC level of the following sequences:
(a) x(n) =
(b) x(n) =

1 n2
2

1 |n|
3
u(n 3).
sin
n
7
Problem 2.22 Determine the DC level of the following sequences:

(a) x(n) =
(b) x(n) =

1 n
4
u(n + 1).
2
1 n
8
sin3
n
3
.

1 n
2
Problem 2.23 Plot the sequence x(n) = (n + 1) +

components.
Problem 2.24 Plot the sequence x(n) = 2(n 1) +
odd components.
u(n 3). Plot also its even and odd

1 n
4
u(n + 1). Plot also its even and
Problem 2.25 Let x(n) = (n 2) + 21 u(n 1) 14 u(n 3).

(a) Plot x(n).
(b) Plot x(2n).

(c) Plot x(n 1).
(d) Plot x(n + 1).

(e) Plot x(n 1).
(f) Plot x(n + 1).
(g) Plot x(2n + 1).

(h) Plot x(2n 1).
Problem 2.26 Let x(n) = 21 (n + 1) u(n + 2) 13 u(n + 2).
(a) Plot x(n).
(b) Plot x(2n).

(c) Plot x(n 1).
(d) Plot x(n + 1).

(e) Plot x(n 1).
(f) Plot x(n + 1).
(g) Plot x(2n + 1).

(h) Plot x(2n 1).
Problem 2.27 Let x(n) = xe (n) + xo (n) denote the even-odd decomposition of a sequence x(n).
Find the even-odd decomposition of the sequence x2 (n) in terms of xe (n) and xo (n). Apply the
result to

1
1
x(n) = cos
n + sin
n
2
4
4
3
Problem 2.28 Repeat Prob. 2.27 for the sequence
x(n) =
Problem 2.29 Is x(n) =

1 |n|
2
cos

1
1
sin
n + cos
n
4
4
2
6
n
4

1 |n|
3
sin
n
6
conjugate symmetric?
PROBLEMS
46
Problem 2.30 Is x(n) =
2
1 n
4
sin
n
6

1 |n|
2
+j
n
4
cos
conjugate symmetric?
CHAPTER 2
FUNDAMENTAL
SEQUENCES
Problem 2.31 Let x(n) be an odd sequence. Show that its average power is zero.
Problem 2.32 Let x(n) be a conjugate-symmetric sequence. Show that
x(n) =
n=
xR (n)
n=
Problem 2.33 Let xo (n) denote the odd component of an arbitrary sequence x(n). Show that
xo (0) = 0.
Problem 2.34 Let x(n) be an arbitrary odd sequence. Show that
x(n) = 0
n=

1 |n|
2
sin
n
5
. Find S =
Problem 2.36 Repeat Prob. 2.35 for x(n) = 13
n2
n=
sin
3
n
8
x(n).
Problem 2.37 Given the sequence x(n) = (0.5)n u(n), plot the sequences x(2n) and x(n/2).
Find the energies of the latter sequences as well. How do the even and odd components of x(2n) and
x(n/2) relate to those of x(n)?
Problem 2.38 Given the sequence x(n) = (1/4)n u(n), plot the sequences x(3n) and x(n/3).
Find the energies of the latter sequences as well. How do the even and odd components of x(3n) and
x(n/3) relate to those of x(n)?
Problem 2.39 Assume x(n) = 0 for n < 0. Show that x(n) can be expressed in terms of its even
part alone as follows
x(n) = 2xe (n)u(n) xe (0)(n)
Given xe (n) = (0.5)|n| , plot x(n).
Problem 2.40 Let xe (n) and xo (n) denote the even and odd components of an arbitrary sequence,
x(n). Let E, Ee , and Eo denote the energies of the sequences x(n), xe (n), and xo (n). Establish
the equality
E = Ee + Eo
Problem 2.41 Assume x(n) = 0 for n < 0. The even part of x(n) is given by xe (n) = |n| ,
where is real and satisfies || < 1. Find the energy of x(n). Find also the odd component of x(n)
and its energy.
Problem 2.42 Assume x(n) = 0 for n < 0 and x(0) = 1/2. The energy of x(n) is equal to one.
Find the energy of its even part.
Problem 2.43 Given a sequence x(n), we perform the following three operations:
(a) We plot the sequence y(n) = x(n 1) and then scale the time axis and plot y(2n).
(b) We plot the sequence z(n) = x(n 2) and then scale the time axis and plot z(2n).
(c) We plot the sequence w(n) = x(2n) and then shift it and plot w(n 2).
Which procedure results in the right plot for x(2n 2)? How would you modify the wrong procedure(s)?
Problem 2.44 How can you construct x(n + 3) from x(n)?
(a) First reflect x(n) about the vertical axis and then shift its samples to the left.
(b) First shift x(n) three samples to the left and then reflect the resulting signal about the vertical
axis.
(c) First shift x(n) three samples to the right and then reflect the resulting signal about the vertical
axis.
(d) Both parts (a) and (b) are correct.
Problem 2.45 The samples of the sequence ejn x(1 n) are {1, 2, 1 , 0, 2, 3, 1}, where the
box denotes the origin of time (i.e., n = 0). Samples to the right of the box occur at positive time
instants while samples to the left of the box occur at negative time instants. Samples outside the
specified interval are all zero. Which terms of x(n) can you determine from this information?
Problem 2.46 The samples of a sequence x(n) are zero except at the time instants shown in
Fig. 2.16. The amplitudes of the non-zero samples are either 1, 2, or 3. Plot the sequence h(n)
that is defined by
1
3
h(n) = x(n + 2) (n) + u(n 3)
2
2
Plot also the sequences x(3n 2), x(2n + 3), and x(2n 1).
x(n)
3
2
FIGURE 2.16 Sequence x(n) defined in Prob. 2.46.
Problem 2.47 Answer True or False. In each case, either prove your answer or give a counterexample.
(a) A power sequence is necessarily an energy sequence.
(b) Every energy sequence has zero average power.
(c) The sequence x(n) = 1/(n + 1), n 0, is an energy sequence.
(d) If x(n) is an energy sequence then x(n) 0 as n .
(e) There does not exist a sequence with infinite average power.
(f) The sum of two energy sequences, {x(n) = x1 (n) + x2 (n)}, is an energy sequence.
Problem 2.48 Which of the following statements is false?
(a) All energy signals are power signals.
(b) Some energy signals are power signals.
(c) All power signals are energy signals.
(d) Some power signals are energy signals.
Problem 2.49 In order to reconstruct x(n), it is enough to know which of the following signals?
(a) x(2n) and x(n2 ).
(b) x(2n + 3) and x(2n + 4).
(c) x(2n + 3) and x(2n 1).
47
SECTION 2.8
PROBLEMS
48
(d) Both parts (a) and (b) are correct.
CHAPTER 2
FUNDAMENTAL
SEQUENCES

(
x(n) =
5
1 7 15 31 63 127
,
,
,
,...
, , ,
2
4 8 16 32 64 128
where the box denotes the origin of time (i.e., n = 0). Express x(n) in terms of the sequences (n),
n
u(n), and 12 .
Problem 2.51 Consider the expression for the sum of the geometric series:
S =
rn =
n=0
1
,
1r
|r| < 1
(a) Differentiate S with respect to r and conclude that
nr n =
n=0
r
(1 r)2
(b) Differentiate S again with respect to r and conclude that
n2 r n =
n=0
Problem 2.52 Evaluate the following series:

(a) S =
(b) S =
(c) S =
n=0
n(0.5)n .
n=0
n(0.5)2n .
n=0
n2 (0.5)n .
P
P

(a) S =
(b) S =
(c) S =
n=2
n=3
P0
n(0.25)n .
n(0.25)2n .
n= (4)

(a) S =
(b) S =
(c) S =
n=3 (n
n=3
P999
+ 2n2 )(1/3)n .
n2 (1/3)3n .
n=0 (1/4)
r(1 + r)
(1 r)3
CHAPTER
Periodic Sequences
eriodic sequences play an important role in the study of discrete-time signals and systems. A prominent role is played by the complex exponential sequence, x(n) = ejo n . In
this chapter, we define periodic sequences and highlight some of the properties of complex
exponential sequences.
3.1 PERIODIC SIGNALS

A sequence x(n) is said to be periodic with period N if N is the smallest positive integer
such that
x(n) = x(n + N ) for all integers n
(3.1)
In other words, N is the smallest positive integer for which the sequence repeats itself indefinitely. In particular, if we shift a periodic sequence by any multiple of N samples to the
left or to the right, then the resulting sequence would coincide with the original sequence.
The value of N is also called the fundamental period of the sequence.
Example 3.1 (A periodic sequence)

Figure 3.1 shows a periodic sequence x(n) that repeats itself every 4 samples. Therefore, its period
is N = 4.
x(n)
1
4 3 2 1
0 1
4 5
FIGURE 3.1 The plot shows a periodic sequence x(n) with period N = 4.
49
Copyright
50
CHAPTER 3
Example 3.2 (Continuous-time sinusoidal signals)

In continuous-time, sinusoidal signals of the form
x(t) = sin(o t + o )
are always periodic, namely, there always exists a positive real number To such that
x(t) = x(t + To ) for all real numbers t
In particular, for the sinusoidal signal in this example, the value of the period To is given by
To = 2/o
(3.2)
since then
sin (o (t + To ) + o )
sin (o (t + 2/o + o )
sin (o t + o + 2)
sin(o t + o )
The period To in this case is measured in seconds and o is measured in radians/second, and we we
say that the signal repeats itself every T seconds. We refer to o as the angular frequency of the
continuous-time sinusoidal signal. We also define the frequency of the signal as
Fo = 1/To
Fo = o /2
or
(3.3)
where Fo is measured in Hertz (or 1/sec). While the continuous-time sinusoidal signal is always
periodic and repeats itself every To seconds, the same conclusion does not hold for sinusoidal sequences; they may or may not be periodic. This is because the period N is now required to be an
integer as we verify in the next example.
For illustration purposes, Figure 3.2 plots the periodic continuous-time sinusoidal signal

x(t) = sin
5t
+
3
3
over 5 t 5. Its period is To = 6/5 = 1.2 seconds.
1
0.5
x(t)
PERIODIC
SEQUENCES
0
0.5
1
5
0
t
5
T=1.2
FIGURE 3.2 The plot shows a continuous-time periodic sinusoidal signal x(t) = sin(5t/3 +
/3) with period To = 1.2 seconds; the signal is shown over the interval 5 t 5.
51
SECTION 3.1
Example 3.3 (Discrete-time sinusoidal signals)
PERIODIC
SIGNALS
Consider now the sinusoidal sequence

x(n) = sin(o n + o )
for some parameters {o , o }. In order to verify whether this sequence is periodic or not, we need
to find the smallest positive integer, N , that satisfies
sin(o n + o ) = sin(o (n + N ) + o )
for all n
sin(o n + o ) = sin(o n + o + o N )
for all n
or, equivalently,
We know from the properties of the sine function that this equality holds if, and only if,
o N = 2k,
for some integer k
(3.4)
That is, if and only if, o N is an integer multiple of 2. The smallest possible integer N that satisfies
this relation would qualify as the period of the sinusoidal sequence.
The difficulty lies in the fact that there need not always exist an integer value of k that results in an
integer value for N satisfying (3.4); in other words, not every sinusoidal sequence is periodic! This
is just one of the subtle differences that exist between discrete-time and continuous-time signals. In
continuous-time, all sinusoidal signals are periodic, but not in discrete-time.
Example 3.4 (Periodic sinusoidal sequence)

Let us continue with the prior example of a sinusoidal sequence and assume o = 5/3, i.e.,

x(n) = sin
5
n + o
3
Then, according to (3.4), N and k must be related via

N=
6
k
5
In this case, the smallest integer k that results in an integer N is k = 5. It then follows that N = 6
and we conclude that the sequence x(n) repeats itself every 6 samples. We therefore say that it is
periodic with period N = 6 samples.
Figure 3.3 plots the periodic sinusoidal sequence

x(n) = sin
5n
+
3
3
It is seen that the sequence repeats itself every 6 samples so that the period is N = 6 samples.
Example 3.5 (Non-periodic sinusoidal sequence)

Consider now the sinusoidal sequence
x(n) = sin
2 n + o
52
1
CHAPTER 3
PERIODIC
SEQUENCES
0.5
0
0.5
1
15 13 11 9
1
n
11
13
15
N=6
FIGURE 3.3 The plot shows the periodic sinusoidal sequence x(n) = sin(5n/3 + /3) with
period N = 6 samples.; the signal is shown over the interval 15 n 15.
where o =
2. In this case, and according to (3.4), N and k should be related via
N = 2k
It is clear now that there does not exist any integer k that results in an integer N since 2 is an
irrational number. For this reason,
the sequence x(n) in thisexample is not periodic! In contrast, the
t + o are both periodic, since in continuouscontinuous-time signals sin( 2 t + o ) and sin 5

3
time the period of a signal is allowed to be any positive real number.
Figure 3.4 plots the sinusoidal sequence
x(n) = sin
2n
It is seen that the sequence is not periodic.
1
0.5
0
0.5
1
15 13 11 9
11
13
15
FIGURE 3.4 The plot shows the non-periodic sinusoidal sequence x(n) = sin( 2 n) over the
interval 15 n 15.
53
3.2 COMPLEX EXPONENTIAL SEQUENCES
SECTION 3.2
The observations that were made in the above examples for sinusoidal sequences hold also
for complex exponential sequences. Since the latter sequences play a prominent role in
characterizing discrete-time signals and systems, as is going to be seen in future chapters,
we proceed to examine the sequences and their properties more closely.
Thus, consider a complex exponential sequence of the form
x(n) = ejo n
(3.5)
This sequence will be periodic if we can find the smallest positive integer N such that
ejo n = ejo (n+N )
or, equivalently, if we can find the smallest positive integer N such that
ejo N = 1
(3.6)
On the left hand side of the above equality we have a complex number, ejo N , which lies
on the unit circle: its magnitude is one and its phase is o N , as illustrated in Fig. 3.5.
Im
ej0 N
1
0 N
Re
FIGURE 3.5 The plot shows the complex number ejo N , which lies on the circle of unit radius
in the complex plane.
In order for equality (3.6) to hold, the complex number ejo N must coincide with the
real number 1. This can happen if, and only if, the phase o N is a multiple of 2,
o N = 2k,
for some integer k
(3.7)
We again see that, as in the case of sinusoidal sequences, complex exponential sequences
may or may not be periodic. This is because we need to be able to select an integer value
for k that would result in an integer value for N satisfying (3.7).
COMPLEX
EXPONENTIAL
SEQUENCES
54
CHAPTER 3
PERIODIC
SEQUENCES
Example 3.6 (Periodic complex exponential sequence)

Consider the complex exponential sequence
x(n) = ej 6 n
for which o = /6. Then, according to (3.7), N and k must be related via
N = 12 k
and we conclude that the sequence x(n) = ejn/6 repeats itself every 12 samples. We therefore
say that the sequence is periodic with period N = 12 samples. This behavior can be illustrated on a
polar plot as follows.
Starting from n = 0, the terms of the sequence will be points on the unit circle at the following
successive angles (in degrees and relative to the positive horizontal axis):
0, 30, 60, 90, 120, 150, 180, 210, 240, 270, 300, 330, 0.
These points cover the unit circle in a counter clockwise direction see Fig. 3.6, and it is seen that it
takes 12 samples before the sequence repeats itself.
n=3
o
90
o
120
n=4
60
n=2
150
n=5
30
n=1
180
n=6
0
n=0
330o
n=11
210
n=7
240o
n=8
270
n=9
300o
n=10
FIGURE 3.6 A polar plot showing the samples of the sequence x(n) = ej 6 n over one period on
the circle of unit radius; the values of the angles are indicated in degrees and in steps of 30o .
55
SECTION 3.3
Example 3.7 (Non-periodic complex exponential sequence)
ANGULAR
FREQUENCY
Consider now the complex exponential sequence

x(n) = ej
where o =
2n
2. In this case, and according to (3.7), N and k should be related via
N = 2k
It is clear that
there does not exist any integer k that results in an integer N . For this reason, the
sequence ej 2 n is not periodic.
Example 3.8 (One-sided sequences)

Let us start from a periodic complex exponential sequence and transform it into a one-sided sequence, say by considering an example of the form
x(n) = ej 6 n u(n)
Then, the resulting sequence is not periodic anymore. For instance, the sequence assumes zero values
for negative time, n < 0, and nonzero values for nonnegative time, n 0. Although the samples of
the sequence repeat themselves every 12 samples over n 0, if we shift the above sequence by 12
samples to the right, then the new sequence will not coincide with the original sequence, namely,
x(n 12) 6= x(n)
This is because the first 12 samples of x(n 12) will all be zero, while the first 12 samples of x(n)
will consist of values on the unit circle.
3.3 ANGULAR FREQUENCY

Consider again the periodic complex exponential sequence
x(n) = ej 6 n
We already know that the sequence is periodic with period N = 12 samples. Moreover, the
samples of this sequence cover a phase change of 2 radians during every period, as was
illustrated in Fig. 3.6. In other words, when the sequence starts repeating itself, its samples
would have covered the circle once. Since the period is N = 12 samples, we therefore say
that the sequence covers
2
=
radians per sample
12
6
The value o = /6 therefore serves as a measure of how many radians are covered per
sample by the periodic sequence. For this reason, we shall refer to o as the angular
frequency and it is measured in radians/sample.
Consider now the alternative sequence
x(n) = ej 6 n
with a negative value for o . One might wonder about the meaning of a negative o . First
note that the sequence ej 6 n is still periodic with the same period N = 12. Now, however,
56
CHAPTER 3
PERIODIC
SEQUENCES
the terms of the sequence will be points on the unit circle at the following successive angles
(measured in degrees and relative to the positive horizontal axis):
0, 30, 60, 90, 120, 150, 180, 210, 240, 270, 300, 330, 0
These terms cover the unit circle in a clockwise direction. This is in contrast to the earlier
sequence ej 6 n , with positive o , whose terms cover the circle in a counter-clockwise

direction. Therefore, the sign of o indicates the direction by which the unit circle is
covered by the samples of the sequence. Figure 3.7 shows the polar plots of both sequences
ejn/6 and ejn/6 over one period; the arrows indicate the direction by which the unit
circles are covered by the samples.
n=9
n=3
o
120
n=5
90
60
n=2
n=4
o
150
n=8
o
150
30o
1
180
n=6
o
o
270
n=9
300
1 0o
180
330
o
n=10
30
n=11
n=6
n=0
o
240
n=8
60
n=0
210
n=7
90
n=7
n=1
o
120
210
n=5
n=11
330
n=1
240
n=4
n=10
270
n=3
300
n=2
j n/6
x(n)=e j n/6
x(n)=e
FIGURE 3.7 The polar plot on the left corresponds to one period of the sequence ejn/6 whose
samples cover the unit circle in a counter-clockwise direction. The polar plot on the right corresponds
to one period of the sequence ejn/6 whose samples cover the unit circle in a clockwise direction.
Accordingly, we say that the sequence ejn/6 has a negative angular frequency of
/6 radians/sample, while the sequence ejn/6 has a positive angular frequency of /6
radians/sample. Sometimes, both sequences are said to have an angular frequency of /6
radians/sample without being specific about whether the sequence is covering the unit circle in one direction or the other.
Example 3.9 (Angular frequency)

Consider the complex exponential sequence
x(n) = ej
3 n
5
for which o = 3/5. Then, according to (3.7), N and k must be related via
N=
10
k
3
and we conclude that the sequence ej3n/5 repeats itself every 10 samples. However, note that over
these 10 samples and, hence, over one period, the samples of the sequence cover the unit circle 3
times (as revealed by the value of k = 3). Indeed, starting from n = 0, the samples of the sequence
evolve according to the following angles (in radians and relative to the positive horizontal axis):
0,
3 6 9 12
18 21 24 27
,
,
,
, ,
,
,
,
, 2
5
5
5
5
5
5
5
5
Therefore, the angular frequency of the sequence is

3
2 3
=
10
5
radians/sample
3.4 EULERS RELATION

We might wonder about how negative values for o arise. One motivation stems from
the so-called Eulers relation, which allows us to express sinusoidal sequences as combinations of complex exponential sequences with positive and negative angular frequencies.
Specifically, Eulers relation states that, for any real scalar ,
ej = cos() + j sin()
(3.8)
The right-hand side of this expression can be interpreted as a complex number with real
part cos() and imaginary part sin(). The left-hand side of the equality can be interpreted
as the polar representation of the complex number: it has unit magnitude and phase .
Applying Eulers relation to the choice = o n, we get the result
ejo n = cos(o n) + j sin(o n)
(3.9)
which expresses a complex exponential in terms of cosine and sine sequences. By equating
the complex conjugates of both sides of the above relation we get
ejo n = cos(o n) j sin(o n)
(3.10)
We can combine (3.9)(3.10) to express the cosine and sine sequences in terms of the
complex exponential sequence. Indeed, by adding (3.9)(3.10) we arrive at
cos(o n) =

1 jo n
e
+ ejo n
2
(3.11)
Likewise, by subtracting (3.9)(3.10) we obtain

sin(o n) =

1 jo n
e
ejo n
2j
(3.12)
We therefore find that a cosine sequence can be obtained by combining the terms of two
exponential sequences: one with a negative angular frequency and the other with a positive
angular frequency. Later in the book, we shall arrive at a similar conclusion for generic
sequences x(n), namely, that under some conditions, a generic sequence x(n) can also be
expressed as a linear combination of (often more than two) exponential sequences with
57
SECTION 3.5
EULERS
RELATION
58
positive and negative angular frequencies see (15.3) and the discussion in Sec. 15.1.
CHAPTER 3
PERIODIC
SEQUENCES
3.5 RELATING ANGULAR FREQUENCIES AND PERIODS

Now that we have introduced the concept of angular frequencies, let us highlight some additional subtleties that arise when studying periodic sequences. Earlier we encountered one
such subtlety, namely, that sinusoidal sequences need not be periodic; likewise, complex
exponential sequences of the form (3.5) need not be periodic.
First, in continuous-time, there is a direct relationship between the period of a signal
and its angular frequency. In the sinusoidal signal of Example 3.2, namely,
x(t) = sin(o t + o )
(3.13)
the quantity o is called the angular frequency (measured in radians per second) and the
quantity
T = 2/o
(3.14)
is the period (measured in seconds). Observe that the larger the angular frequency, o , the
smaller the period, T , and vice-versa. The same conclusion does not hold in discrete-time.
Example 3.10 (Two periodic complex exponential sequences)

Let o = /3 and consider the exponential sequence
x(n) = ej 3 n
This is a periodic sequence with period N = 6 and k = 1. The value of k signifies that the sequence
covers a 2 phase change (i.e., a single rotation around the circle) every six samples. The resulting
angular frequency is therefore
2
=
6
3
radians per sample
which is equal to o . Note that the angular frequency (/3) in this example is larger than the
angular frequency in the earlier Example 3.2, which used o = /6. Moreover, the sequence in the
current example has period N = 6, which is correspondingly smaller than the period N = 12 from
Example 3.2. Therefore, in this case, we observe a similar behavior to continuous-time: a larger
angular frequency results in a proportionally smaller period.
The question is whether it always holds that higher angular frequencies correspond to lower
periods. The answer, for discrete-time signals, is negative!
Example 3.11 (Angular frequencies and periods)

Let o = 3/4 and consider now the exponential sequence
x(n) = ej
3 n
4
The value of o is higher than in the previous two cases, {/3, /6} from Examples 3.2 and 3.5.
The sequence is still periodic with N = 8 and k = 3. The value of k means that the sequence
goes around the circle 3 times before repeating itself. It therefore covers 3 2 = 6 radians every
period or every 8 samples. This corresponds to an angular frequency of
3
6
=
8
4
radians/sample
It follows that the sequence in this example has higher angular frequency than in the two earlier
examples (with o = /6 and o = /3). Its period, however, is smaller than the period of one of
them and larger than the period of the other; recall that the periods were found earlier as N = 6 and
N = 12!
We therefore conclude that, in discrete-time, we must always compare the (absolute)

angular frequencies (radians/sample), and not the periods, in order to determine which
sequence has higher angular speed (i.e., which sequence covers more radians per sample).
Moreover, it is not difficult to see that for values of o in the range [, ], the higher the
absolute value of o the higher the (absolute) angular frequency of the sequence. In fact,
o itself is the angular frequency. This has been the case in all examples considered so far.
However, for values of o outside the interval [, ], it does not hold that the higher
the (absolute value of) o the higher the angular frequency of the sequence. This is yet
another distinction from continuous-time, where the higher the value of the angular frequency (in radians per second) the faster the oscillations of the signal. We clarify this point
by explaining the notion of indistinguishable sequences.
Indistinguishable Sequences
To begin with, observe that it always holds that
ejo n = ej(o +2m)n
for all integers n and m
and for any o . That is, two complex exponential sequences whose angular frequencies
differ by multiples of 2 are indistinguishable. This is distinct from the continuous-time
case, where it does not hold that
ejo t = ej(o +2m)t
for all t
since 2mt is not necessarily a multiple of 2 for all t. Therefore, if a complex exponential
sequence has a value for o in the range [, 2], then by subtracting 2 from it we obtain
a new value o that lies between [, 0],
o = o 2
(3.15)
and both sequences, ejo n and ejo n , will be indistinguishable. In such cases, we would
choose the smaller number (in absolute value), o , to be the angular frequency for ejo n .
In a similar fashion, if an exponential sequence has a value for o in the range [2, ],
then by adding 2 to it we obtain a new value o that lies between [0, ],
o = o + 2
(3.16)
and both sequences, ejo n and ejo n , will again be indistinguishable. In this case, we
would also choose the smaller number (in absolute value), o , to be the angular frequency
for ejo n .
59
SECTION 3.5
RELATING
ANGULAR
FREQUENCIES
AND PERIODS
60
CHAPTER 3
PERIODIC
SEQUENCES
In summary, given an exponential sequence x(n) = ejo n , we can always reduce the
value of o to lie within the interval [, ], by adding or subtracting multiples of 2 as
needed. The new equivalent sequence, x(n) = ejo n , becomes our starting point for any
subsequent analysis.
Example 3.12 (Two indistinguishable sequences)

x(n) = ej
5 n
4
with a value for o = 5/4, which lies outside the interval [, ]. First, we note that the sequence
is periodic with N = 8 and k = 5. That is, the sequence repeats itself every 8 samples and during
one period it covers the circle 5 times (in the counter-clockwise direction). Hence, the sequence
covers 10 radians per 8 samples, which amounts to 5/4 radians per sample. However, note that
3
5
2 =
4
4
and, hence,
ej
5 n
4
= ej
3 n
4
The new sequence

y(n) = ej
3 n
4
is again periodic with the same period N = 8 but with k = 3. That is, the new sequence repeats
itself every 8 samples and during one period it covers the circle 3 times (in the clockwise direction).
Hence, the sequence y(n) covers 6 radians per 8 samples, which amounts to 3/4 radians per
sample.
The samples of y(n) coincide with those of x(n). That is, x(n) and y(n) are identical sequences.
The question then is which angular frequency should we adopt for x(n)? As mentioned above,
our convention throughout this book will be to adjust the angular frequencies so that they always
lie within the interval [, ]. Therefore, for the example at hand, we shall say that the angular
frequency of the sequence ej5n/4 is 3/4 radians/sample.
From the above discussion, we conclude that in discrete-time signal processing, the
range of values for the angular frequency o are always limited to a 2 interval, say
o
(3.17)
This means that we need not consider complex exponential sequences with angular frequencies outside this range. This is because we can always reduce an angular frequency
by integer multiples of 2 and get an identical sequence with angular frequency within the
[, ] range.
Aliases. Angular frequencies o and 1 that differ by integer multiples of 2 are called
aliases of each other,
o = 1 2k,
for any integer k
This is because they generate identical complex exponential sequences.
(3.18)
High and low frequencies. We shall say that angular frequencies close to the endpoints
of the interval [, ] correspond to high frequencies, while angular frequencies close
to 0 correspond to low frequencies. This is because the higher the absolute value of the
angular frequency, the more radians per sample are covered around the unit circle by the
samples of a periodic sequence see Fig. 3.8.
range of
low angular
frequencies
around 0
(radians/sample)
range of
high angular
frequencies
close to
FIGURE 3.8 The range of angular frequencies for discrete-time signals is [, ]. Higher angular
frequencies occur close to and lower angular frequencies occur around 0.
3.6 APPLICATION: HARMONICS AND MUSIC SYNTHESIS

in the context of a practical problem. Specifically, we comment on how periodic signals in
the form of harmonics arise during the generation of music notes.
Consider a periodic continuous-time signal x(t) with period To (measured in seconds)
and frequency Fo = 1/To (measured in Hertz). It is known that, under some mild technical conditions, such periodic signals admit a so-called Fourier series representation in the
following form:
x(t) = Ao +
Ak cos (2kFo t) +
k=1
k=1
Bk sin (2kFo t)
(3.19)
where k assumes integer values, and the Fourier coefficients {Ak , Bk } are evaluated as
follows:
Ao
Ak
Bk
=
=
=
1
To
2
To
2
To
To
x(t)dt
(3.20)
x(t) cos (2kFo t) dt
(3.21)
x(t) sin (2kFo t) dt
(3.22)
To
To
61
SECTION 3.6
APPLICATION
62
CHAPTER 3
PERIODIC
SEQUENCES
We refer to To as the fundamental period of x(t) and to Fo as its fundamental frequency.

Using the trigonometric identity
cos(a + b) = cos(a) cos(b) sin(a) sin(b)
(3.23)
we can easily verify the validity of the following equality

Ak cos (2kFo t) + Bk sin (2kFo t) = Ck cos(2kFo t k )
where the parameters {Ck , k } and {Ak , Bk } are defined in terms of each other as follows:
q
Ck =
A2k + Bk2
(3.24)
k
arctan (Bk /Ak )
(3.25)
Ak
Bk
=
=
Ck cos(k )
Ck sin(k )
(3.26)
(3.27)
It follows that we can rewrite the Fourier series (3.19) in a more compact and equivalent
form as:
P
Ck cos (2kFo t k )
x(t) = Ao +
(3.28)
k=1
Harmonics
Expression (3.28) shows that the periodic signal x(t) can always be expressed as the sum
of sinusoidal components that occur at multiples of the fundamental frequency Fo , namely,
at kFo . The DC level of the signal x(t) is equal to the first coefficient, Ao , and it simply
corresponds to the average value of the signal over one period. The sinusoidal components of the signal x(t) that occur at multiples of the fundamental frequency are called
harmonics. In general, for well-behaved periodic signals x(t), a sufficiently large number
of harmonics can be used to approximate (or synthesize) the signal reasonably well.
Now, when listening to a periodic signals x(t), both the frequency of the signal and its
shape affect the resulting sound. Two periodic signals with identical fundamental frequencies, Fo , but different waveform shapes (say, one is a square periodic waveform and the
other is a triangular periodic waveform) will sound differently. This effect is not surprising
since such different periodic signals will have different Fourier series representations and,
consequently, different content in terms of their harmonic frequencies; the coefficients
{Ak , Bk } or {Ck } will be different. The human ear is insensitive to phase offsets and,
hence, the phase parameters {k } do not affect the way a signal sounds.
Timbers
By combining several harmonic components with different amplitudes {Ck }, we can generate different timbres, which explains why the same notes on different musical instruments
can sound differently. For example, when we play a note on an instrument, we not only
excite the fundamental frequency Fo of the note (say, 440Hz for musical note A see
Fig. 3.9)3 but also the harmonics kFo of the fundamental frequency (880Hz, 1320Hz, etc.
for the same note A).
The presence of the harmonics is the reason why a piano key sounds more natural and
richer than a pure sinusoidal signal. The sound generated by the piano simultaneously
3 The
source for this public domain image of music notes is Wikipedia.
63
SECTION 3.6
APPLICATION
FIGURE 3.9 Several music notes and their respective frequencies in Hz.
contains a lot of harmonics with different amplitudes and the sound is then said to be
polytonic. Now even for the same note on a trumpet, the amplitudes (or energy) of the
harmonics might be different than those on a piano. Therefore, by varying the amplitudes
of the different harmonics that compose a polytonic note, we can give different timbres
to the note. Figure 3.10 shows the waveforms of three different sound signals composed
of the same fundamental frequency (440 Hz) and the same harmonic frequencies (880Hz,
1320Hz, 1760Hz, 2200Hz) but with different harmonic amplitudes {Ck } (with the DC
level, Ao , and the phases {k } set to zero).
Additive Synthesis
A discretized version of the periodic signal representation (3.19) can be obtained by sampling x(t) at some rate of Fs samples per second. Thus, substituting t by nTs in (3.19),
where Ts denotes the sampling period and is given by Ts = 1/Fs , we obtain the discretetime version:
x(n) = Ao +
k=1

P
Bk sin 2 FFos kn
Ak cos 2 FFos kn +
(3.29)
k=1
The cosine and sine sequences in (3.29) may or may not be periodic depending on the
relation between Fo and Fs . When the sampling frequency Fs is selected as a multiple of
the fundamental frequency, Fo , say as
Fs = N Fo ,
for some integer N
(3.30)
then the sequence x(n) becomes periodic and takes the form:
x(n) = Ao +
k=1
Ak cos
2
kn
N
k=1
Bk sin
2
kn
N
(3.31)
64
C1=1, C2=0.5, C3=0.4, C4=0.3, C5=0.2

5
0
5
C =1, C =0.2, C =0.3, C =0.6, C =0.1
1
amplitude
5
0
5
C1=1, C2=0.8, C3=0.1, C4=0.5, C5=0.4
5
amplitude
PERIODIC
SEQUENCES
amplitude
CHAPTER 3
0
5
0.005
0.01
time (seconds)
0.015
0.02
FIGURE 3.10 Waveforms of three signals with the same fundamental frequency (440 Hz) and
the same harmonic frequencies (880Hz, 1320Hz, 1760Hz, 2200Hz) but with different harmonic
amplitudes Ck .
We may express (3.29) in an alternative equivalent form in terms of exponential sequences

by using Eulers relations (3.11)(3.12). Replacing the cosine and sine sequences in terms
of exponential sequences and grouping terms gives

X
X
Fo
Fo
Ak jBk
Ak + jBk
x(n) = Ao +
ej2 Fs kn +
ej2 Fs kn
2
2
k=1
k=1
If we introduce the complex coefficient
Xk = Ak jBk
then we can rewrite the above expression in the form:
!
1 X
o kn
1
j2 F
F
s
x(n) = Ao +
Xk e
+
2
2
k=1
(3.32)
Xk
k=1
o
j2 F
Fs kn
which involves exponentials with both positive and negative angular frequencies. More
compactly, we can group the two summations and write
x(n) = Ao +
k=1

Fo
Re Xk ej2 Fs kn
(3.33)
in terms of the real part of the exponential sequence inside the summation symbol. In
the case when the sampling frequency Fs is selected as a multiple of the fundamental
frequency, Fo , say as Fs = N Fo , the above expression becomes
x(n) = Ao +
k=1

2
Re Xk ej N kn
(3.34)
Expression (3.29), or its equivalent form (3.33), can be used to synthesize audio and musical sounds digitally by combining together a sufficient number of sinusoidal sequences
through the selection of the coefficients {Ao , Ak , Bk } over some interval 1 k K and
setting all other coefficients to zero. The DC term Ao is undesirable in audio and music
synthesis applications and is therefore set to zero. This digital technique is a special case
of a more general method known as additive synthesis in the field of audio and music signal processing. Additive synthesis allows for the digital emulation of sounds. In one of
its forms, the restriction of constant coefficients {Ak , Bk } is lifted and the coefficients are
allowed to vary with time as well:
x(n) =
K
P
k=1

K
P
Bk (n) sin 2 FFos kn
Ak (n) cos 2 FFos kn +
(3.35)
k=1
Through the selection and control of the sequences {Ak (n), Bk (n)}, different sounds can
be generated.
Practice Questions:
1. By combining sinusoids at close enough frequencies we can generate beat signals. For example, we can produce beat frequencies by playing two neighboring notes of a piano simultaneously, say,
x(t) = cos(2F1 t) + cos(2F2 t)
Introduce the center and deviation frequencies:
Fc =
F2 + F1
,
2
Fd =
F2 F1
2
Verify that x(t) can be expressed in the equivalent representation:

x(t) = 2 cos(2Fd t) cos(2Fc t)
2. Can you explain why the above expression is said to have the form of an Amplitude Modulated
(AM) signal?
3. Is a continuous-time signal that is composed of a finite-number of non-harmonically related
sinusoids a periodic signal?
4. Write the general form of the sequence x(n) in (3.35) for digitally synthesizing the musical
note A assuming a sampling frequency of 20KHz and using 3 harmonics.
3.7 PROBLEMS
Problem 3.1 What is the angular frequency of the sequence x(n) = ej5n/4 ?
Problem 3.2 What is the angular frequency of the sequence x(n) = ej8n/7 ?
Problem 3.3 Order the following sequences according to (a) increasing angular frequencies and
(b) decreasing periods:
x1 (n) = ej 4 n ,
x2 (n) = ej 8 n ,
x3 (n) = ej
7 n
8
65
SECTION 3.7
PROBLEMS
66
CHAPTER 3
Problem 3.4 Order the following sequences according to (a) increasing angular frequencies and
(b) decreasing periods:
PERIODIC
SEQUENCES
x1 (n) = ej 3 n ,
x2 (n) = ej 6 n ,
x3 (n) = ej
Problem 3.5 What is the period of the sequence x(n) = sin
n
3
Problem 3.6 What is the period of the sequence x(n) = cos
n
6
5 n
6
Problem 3.7 What is the sequence that results from sampling x(t) = sin(120t) at the rate of 180
samples per second? What is the angular frequency and period of the sequence?
Problem 3.8 What is the sequence that results from sampling x(t) = cos(50t) at the rate of 150
samples per second? What is the angular frequency and period of the sequence?
Problem 3.9 Sample the sequence x2 (t) at the rate of 480 samples per second, where x(t) =
cos(120t). Is the resulting sequence periodic? If so, find its period.
Problem 3.10 Sample the sequence x2 (t) at the rate of 500 samples per second, where x(t) =
sin(50t). Is the resulting sequence periodic? If so, find its period.
Problem 3.11 Answer either True or False to the following statements and give brief justifications.
Statement
The sequence u(n) u(n 3) has only 3 nonzero samples.
The sequence ej 2.5 n has period N = 5 samples.

The samples of the sequence e0.5n have unit magnitude for all n.
The frequency of sin(t) is 0.5 Hz or rad/sec.
This sinusoid covers a 2 phase change per period.
Sampling it every 0.3 sec leads to the sequence sin(0.3n)
of period N = 20 samples.
This sequence also covers 2 phase change per period.
cos(0.1n) and cos(1.9n) have the same angular frequency.
Problem 3.12 Answer either True or False to the following statements and give brief justifications.
Statement
The sequence u(n) u(n + 3) has only 3 nonzero samples.
The sequence ej 4.5 n has period N = 18 samples.

The samples of the sequence ejn have unit magnitude for all n.
The frequency of sin(20t) is 10 Hz
This sinusoid covers a phase change per period.
Sampling it every 0.05 sec leads to the sequence sin(n)
of period N = 2 samples.
This sequence also covers phase change per period.
sin(0.3n) and sin(3.7n) have the same angular frequency.
Problem 3.13 Which of the following sequences are periodic?
(a) ej 9 n .
(b) e
j
n
9
(c) e
n2
j
9
(d) e
j
(n2)
9
u(n).
.
.
67

(a) e
n
j 3
7
SECTION 3.7
PROBLEMS
(b) ej
3 n
7
(c) ej
3 n3
7
(d) ej
3 (n+3)
7
u(n).
.
.

(b)
(c)
(d)
(n 4) .
3

4
sin 3 n .

sin 3 n + 4 .

sin2 3 n .
(a) sin

(b)
(c)
(d)
(n + 2) .
4

cos 4 n3 .

cos 4 n 7 .

cos2 4 n .
(a) cos
Problem 3.17 Find the period of the sequence

x(n) = cos

sin
n +
6
6
8
n +

x(n) = cos

cos
n +
4
3
6
7 n
8
x(n) = ej 4 n + ej 8 n ej
x(n) = ej
5 n
7
+ ej
2 n
5
+ ej
4 n
9
Problem 3.21 Are the following sequences aliases of each other

x(n) = ej
3 n
4
and x(n) = ej
13 n
4
What are their angular frequencies and periods?

Problem 3.22 Are the following sequences aliases of each other
x(n) = ej
5 n
6
and x(n) = ej
7 n
6
What are their angular frequencies and periods?

Problem 3.23 What are the low and high frequency components of the sequence
x(n) = ej 12 n + ej
7 n
8
+ ej
8 n
9
+ ej 24 n ?
Problem 3.24 What are the low and high frequency components of the sequence
x(n) = ej
12 n
13
+ ej 9 n ej
6 n
7
+ ej 25 n ?
68
Problem 3.25 Is the sequence

CHAPTER 3
PERIODIC
SEQUENCES
x(n) = cos
2
n +
3
6
+ 2 sin

periodic? If so, what is its period? Determine also its energy and average-power.
Problem 3.26 Is the sequence
x(n) = sin

2 sin
n
3
3
periodic? If so, what is its period? Determine also its energy and average-power.
Problem 3.27 Assume x(n) has period N. Are the following sequences periodic? If so, determine
their periods in terms of N :
(a) x(1 2n)?
(b) x(n) + (1)n x(0)?
Problem 3.28 Assume x(n) has period N. Are the following sequences periodic? If so, determine
their periods in terms of N :
(a) x(3n + 3)?
(b) x(2n)?
Problem 3.29 If x(n) is periodic, prove that ej 4 n x(n) is also periodic no matter what the period
of x(n) is.
Problem 3.30 If x(n) is periodic, prove that cos
period of x(n) is.
n
4
x(n) is also periodic no matter what the
Problem 3.31 Prove that sin(n) is periodic if, and only if, 2/ is a rational number.
Problem 3.32 Assume x(n) is real-valued and periodic with period N . Are its even and odd
components periodic?
Problem 3.33 True or false? Except for the zero sequence, every periodic sequence has infinite
energy.
Problem 3.34 True or false? The period of the sum of two periodic sequences is always the leastcommon multiple of their periods.
5
Problem 3.35 Consider the sequence x(n) = ej 12 n + ej 12 n . Show that it can be written in the
form
x(n) = A ejo n cos(1 n)
for some positive real number A, and for some o > 1 . Is x(n) periodic?
3
Problem 3.36 Consider the sequence x(n) = ej 7 n ej 7 n . Show that it can be written in
the form
x(n) = A ej (o n 2 ) sin(1 n)
for some positive real number A, and for some o < 1 . Is x(n) periodic?
Problem 3.37 Let x(n) be a periodic sequence of period N and assume its energy over a period is
equal to E. Show that its average power is equal to E/N .
Problem 3.38 Let x(n) be a periodic sequence of period N and assume its DC level (i.e., the
average of its samples) over one period is C. Show that the DC level of the entire sequence is C as
well (refer to Prob. 2.19 for the definition of the DC level of a sequence).
Problem 3.39 Consider an arbitrary sequence x(n) and an arbitrary finite positive integer N . Define the sequence
xp (n) =
69
SECTION 3.7
PROBLEMS
x(n + N )
Show that xp (n) is periodic and find its period.

Problem 3.40 Refer to the periodic sequence defined in Prob. 3.39. Let N = 3.
(a) Assume x(n) = (n) + 0.5(n 1) 2(n 3). Determine the samples of xp (n) over the
period 0 n N 1.
(b) Repeat when x(n) = (n) + 0.5(n 1) 2(n 2). How do the samples of xp (n) and
x(n) compare with each other over 0 n N 1.
Problem 3.41 Consider the sequence x(n) = 0.5(n + 1) + (n) + 0.5(n 1) and let N = 4.
Find the average power of the sequence xp (n) defined in Prob. 3.39.
Problem 3.42 Consider the sequence x(n) = (n 2) + (n + 1) and let N = 4. Find the DC
level of the sequence xp (n) defined in Prob. 3.39.
Problem 3.43 Consider two periodic sequences x(n) and h(n) with periods Nx and Nh , respectively. Let
Nx Nh
N=
gcd(Nx , Nh )
in terms of the greatest common divisor of Nx and Nh . Argue that the sum sequence, y(n) =
x(n) + h(n), is periodic. How does its period relate to N ? Can the period be less than N ?
Problem 3.44 Consider the same setting as Prob. 3.43. Argue that the product sequence, z(n) =
x(n)h(n), is also periodic. How is the period of z(n) related to N ? Provide examples of sequences
x(n) and h(n) for which the product sequences have periods either equal to N or less than N .
CHAPTER
Discrete-Time Systems
Now that we have developed a basic understanding of what discrete-time signals (or
sequences) are, we move on to study discrete-time systems and some of their properties.
As the reader will soon realize, this chapter includes several definitions about systems and
their characterizations. While the multitude of definitions might be overwhelming at first
sight, the interpretations of the definitions are in most cases straightforward to understand.
We start with the definition of a system.
4.1 SYSTEMS
A system is defined as a mapping between an input sequence and an output sequence; it
operates on an input sequence and generates an output sequence through some transformation see Fig. 4.1. What makes a system special is that the input sequence, x(n), must
uniquely define the output sequence, y(n). In other words, there should be no ambiguity
about what the output sequence will be for any given input sequence. The output sequence
of a system is sometimes called the response sequence.
input sequence
x(n)
system
S
output sequence
y(n)
FIGURE 4.1 A discrete-time system S maps an input sequence, x(n), into an output or response
sequence, y(n).
Schematically, we may denote the input-output mapping of a generic system by writing

y(n) = S[x(n)]
(4.1)
where the letter S[] refers to the transformation that is carried out by the system. The
above notation signifies that system S is being applied to the input sequence, x(n), in
order to generate the output sequence, y(n). In general, each term of the output sequence
y(n) can be a function of present, past, or future terms of the input sequence x(n). For
the purposes of the treatment in this book, all input-output transformations (or relations) S
corresponding to systems will be described by mathematical equations as the discussions
will illustrate.
71
Copyright
72
CHAPTER 4
DISCRETETIME
SYSTEMS
Example 4.1 (Examples of systems)

The following transformations are examples of systems:
1. S is described by y(n) = x(n). This is a very simple system. It maps the input sequence
to itself. Such a system can be viewed as a model for an ideal (i.e., lossless) wire connection
transmitting a signal from one point to another in a communication system.
2. S is described by y(n) = x(n 1). This is a unit delay system. It delays the input sequence
by one unit of time.
3. S is described by y(n) = 21 [x(n) + x(n 1)]. This system averages two successive samples
of the input sequence.
4. S is described by y(n) = y(n 1) + x(n) for n 0 and with initial condition y(1) = 0.
This system generates output values for n 0 by adding to its previous output sample the
value of the present input sample.
In all these examples, given the input sequence x(n) we can readily and uniquely evaluate the corresponding output sequence y(n).
Example 4.2 (Evaluating the response of a system)

Consider the averaging system
y(n) =
1
[x(n) + x(n 1)]
2
and assume the input sequence is chosen as

x(n) = (1)n
Then one way to evaluate the output sequence is to employ the system description to evaluate the
value of y(n) for every choice of n. Doing so leads to
..
.
y(1)
y(0)
y(1)
..
.
=
=
=
=
=
..
.
1
[x(1) + x(2)]
2
1
[x(0) + x(1)]
2
1
[x(1) + x(0)]
2
=
=
=
..
.
1
[(1)1 + (1)2 ]
2
1
[(1)0 + (1)1 ]
2
1
[(1)1 + (1)0 ]
2
=
=
=
0
0
0
It is easy to see that y(n) = 0 for all n.

Alternatively, we can proceed more generally and evaluate the value of y(n) for a generic n as
follows. We start from the system description and replace the input sequence by its expressions so
that
y(n)
=
=
1
[x(n) + x(n 1)]
2

1
(1)n + (1)n1
2
Now, we note that if n is even we have

(1)n = 1
and if n is odd we have
(1)n = 1
and
and
(1)n1 = 1
(1)n1 = 1
Therefore, in both cases of n even or odd, we conclude again that the expression for y(n) evaluates
to zero. In summary,
x(n) = (1)n = y(n) = 0
Let us now evaluate the output of the same system in response to the one-sided input sequence
x(n) = (1)n u(n)
Again, we can do so either by evaluating the output samples y(n) for different values of n or by
using the system description to write
y(n)
1
[x(n) + x(n 1)]
2

1
(1)n u(n) + (1)n1 u(n 1)
2
=
=
Since the step sequences u(n) and u(n 1) are both equal to zero for n < 0, we conclude that
y(n) = 0 for n < 0. Let us determine the values of y(n) for n 0.
To begin with, observe further that the step sequence u(n) is such that
u(n) = u(n 1) = 1
for all n 1
This is because u(n 1) is simply a one-unit delayed version of u(n). It follows that the expression
for y(n) reduces to

1
y(n) =
(1)n + (1)n1 , for all n 1
2
Here again, depending on whether n is odd or even, the value of y(n) reduces to 0 for n 1. What
about the value of y(n) at time 0? We can use the relation
y(n) =

1
(1)n u(n) + (1)n1 u(n 1)
2
and replace n by 0 to get

y(0) =

1
(1)0 u(0) + (1)1 u(1)
2
But since u(1) = 0 and u(0) = 1, we arrive at y(0) = 1/2. In summary,

x(n) = (1)n u(n) = y(n) = 12 (n)
The response of the system in this case is an impulse sequence with amplitude 1/2.
4.2 CLASSES OF SYSTEMS

Sometimes knowledge of the input sequence alone is not sufficient to evaluate the response
of a system. This is because there are situations in which an input-output transformation
defines a class of systems rather than a unique system.
Consider the input-output relation
y(n) = y(n 1) + x(n)
(class of systems)
(4.2)
with input sequence x(n) and output sequence y(n). Knowledge of x(n) alone is not
sufficient to determine the response of the system y(n) in this case. Additional information
73
SECTION 4.2
CLASSES OF
SYSTEMS
74
CHAPTER 4
DISCRETETIME
SYSTEMS
is needed. For instance, in order to evaluate y(0) we need to know y(1) in addition to
x(0).
So assume, for this example, that we are given the additional piece of information that
y(1) = 1. We say that y(1) = 1 specifies the initial condition of the system at time instant 1. Then, given x(n) we can proceed to compute the output sequence y(n). Indeed,
assume x(n) = u(n). Then we can compute all values of y(n) recursively as follows:
y(0)
y(1)
y(2)
y(3)
y(4)
..
.
=
=
=
=
=
=
y(1) + x(0)
y(0) + x(1)
y(1) + x(2)
y(2) + x(3)
y(3) + x(4)
..
.
=
=
=
=
=
1+1
2+1
3+1
4+1
5+1
=
=
=
=
=
2
3
4
5
6
We recognize a pattern in these calculations and deduce that

y(n) = n + 2 for n 0
The values of y(n) for n < 1 can be similarly determined albeit by running the recursion
backwards in time:
y(n 1) = y(n) x(n)
Doing so leads to the values
y(2) =
y(3) =
..
.
=
y(1) x(1) = 1 0
y(2) x(2) = 1 0
..
.
= 1
= 1
That is, y(n) = 1 for all n 1. We thus find that for the present example and using
x(n) = u(n) and y(1) = 1, we get
x(n) = u(n)
and
y(1) = 1
= y(n) =
(n + 2) for n 0
1
for n 1
For any other input sequence x(n), and for any other initial condition (whether specified
at time n = 1 or at some other time), we can proceed in a similar manner and determine
the corresponding output sequence y(n). We therefore say that the transformation
{y(n) = y(n 1) + x(n) , y(1) = 1}
(a system)
with a particular initial condition specified, describes a system.

Example 4.3 (The same input-output relation with a different initial condition)
Consider again the input-output relation
y(n) = y(n 1) + x(n)
with the same input sequence x(n) = u(n). Assume now, however, that the additional information
that we have available is y(2) = 1, i.e., we are given the initial condition at time 2 as opposed to
75
time 1. Repeating the previous arguments, we can determine y(n) for n 3 by recursion:
y(3)
y(4)
y(5)
y(6)
..
.
=
=
=
=
=
y(2) + x(3)
y(3) + x(4)
y(4) + x(5)
y(5) + x(6)
..
.
=
=
=
=
1 + 1
0+1
1+1
2+1
=
=
=
=
SECTION 4.3
CLASS OF
SYSTEMS
0
1
2
3
so that
y(n) = n 3 for n 2
Likewise, for n < 2, we run the recursion backwards:
y(n 1) = y(n) x(n)

and find
y(1)
y(0)
y(1)
y(2)
..
.
=
=
=
=
=
y(2) x(2)
y(1) x(1)
y(0) x(0)
y(1) x(1)
..
.
=
=
=
=
1 1
2 1
3 1
4 0
=
=
=
=
2
3
4
4
so that y(n) = 4 for n 1. In other words, we now obtain
x(n) = u(n) and y(2) = 1
= y(n) =
(n 3)
4
n0
n 1
Similarly, for any other input sequence x(n), we can determine the corresponding output sequence
y(n) resulting from y(n) = y(n 1) + x(n) and y(2) = 1. We thus say that the transformation
{y(n) = y(n 1) + x(n) , y(2) = 1}
(a system)
describes a system.
What we have seen so far are examples of an input-output relation that leads to two
different output sequences {y(n)} in response to the same input sequence {x(n) = u(n)}.
For this reason, we say that
{y(n) = y(n 1) + x(n) , y(1) = 1}

defines one system, while
{y(n) = y(n 1) + x(n) , y(2) = 1}

defines another system. In these two examples, the systems differ not by their input-output
transformation but by their assumed initial conditions. We also say that the transformation
{y(n) = y(n 1) + x(n)}
76
CHAPTER 4
DISCRETETIME
SYSTEMS
describes a class of systems; it would define a system once an initial condition is specified.
We now proceed to characterize several important properties of systems.
4.3 RELAXED SYSTEMS

A relaxed system or, equivalently, a system that is initially at rest, is defined as one whose
output sequence is zero as long as the input sequence is zero. That is, if
y(n) = S[x(n)]
describes a relaxed system, then
y(m) = 0 for all m n as long as x(m) = 0 over m n
(4.3)
In other words, the output sequence stays at zero while the input sequence is at zero. When
the input sequence moves away from zero, the output sequence may (or may not) move
away from zero. We can alternatively describe a relaxed system as follows. Let no denote
the time instant at which the input sequence x(n) becomes nonzero, i.e., x(n) = 0 for all
n < n0 (if x(n) is always nonzero, then we select no = ). For a relaxed system we
must have y(n) = 0 for all n < no .
Example 4.4 (Non-relaxed systems)
The system
y(n) = x(n + 1)
is not relaxed. This is because we can find a counter-example that violates the definition of a relaxed
system. Consider the input sequence x(n) = (n). This input sequence is zero for all n < 0
and assumes the value 1 at n = 0. Now the corresponding output sequence is y(n) = (n + 1),
which is obtained by replacing x(n) in the input-output transformation by its expression. This output
sequence is zero for all n < 1 but it assumes the value 1 at n = 1 see Fig. 4.2. Therefore,
we have an example of an output sequence that does not remain at zero while the input sequence is
at zero. Instead, the output sequence assumes the value 1 at time n = 1 even though the input
sequence is still zero at that time.
x(n) = (n)
y(n) = (n + 1)
1
3 2 1
1
1 2 3
3 2 1
1 2 3
FIGURE 4.2 An input-output pair corresponding to the system y(n) = x(n + 1); the plot on the
left illustrates a particular input sequence, namely, x(n) = (n), while the plot on the right illustrates
the response sequence. Only samples in the range 3 n 3 are shown. All other samples are
zero.
77
SECTION 4.4
Example 4.5 (Systems with initial conditions)
DYNAMIC
SYSTEMS
Likewise, the systems

{y(n) = y(n 1) + x(n) , y(1) = 1},
{y(n) = y(n 1) + x(n), y(2) = 1}
are not relaxed. For instance, if we choose x(n) = 0 for all n, then the corresponding output
sequences y(n) for both systems will not be identically zero. Even the system
{y(n) = y(n 1) + x(n) , y(1) = 0}
with a zero initial condition is not relaxed! For example, if we evaluate its response to x(n) =
u(n + 2), we find that y(3) = 2, which is nonzero even though the input is zero up to and
including n = 3.
Example 4.6 (Relaxed systems)

The following are examples of relaxed systems:
1. S is described by y(n) = x(n 1). The output sequence will stay at zero as long as the input
sequence stays at zero.
2. S is described by y(n) = y(n 1) + x(n) with the requirement that the output sample
right before the first nonzero sample of the input sequence is zero. For example, assume we
limit the above input-output relation to the interval n 0 and assume further that all input
sequences are one-sided and exist over n 0. If it holds that y(1) = 0, then the system
will be relaxed. In other words, the following system is relaxed
y(n) = y(n 1) + x(n), y(1) = 0, n 0 (relaxed)
Note that we are explicitly adding the requirement n 0 to stress the fact that we are only
interested in the operation of the system over n 0. To verify that the system so defined is
indeed relaxed, we iterate the recursion starting from n = 0 and find that, at any particular
time n,
y(n) =
n
X
k=0
x(k),
n0
It follows that y(n) will stay at zero as long as the input sequence stays at zero. The output
sequence y(n) will move away from zero only after the input sequence moves away from
zero.
3. S is described by y(n) = 2y(n 2) + y(n 1) + x(n) with the requirement that the output
samples at the two time instants right before the first nonzero sample of the input sequence is
zero. For instance, the following system is relaxed
y(n) = 2y(n 2) + y(n 1) + x(n), y(1) = 0, y(2) = 0, n 0 (relaxed)
4.4 DYNAMIC SYSTEMS

A system is static or memoryless if its output at time n depends only on its input at the
same time instant n. Otherwise, the system is said to be dynamic or with memory.
78
Example 4.7 (Systems with memory)
CHAPTER 4
DISCRETETIME
SYSTEMS
(1) The systems y(n) = ax(n) and y(n) = cos[x(n)] are memoryless.
(2) The system {y(n) = y(n1)+x(n), y(1) = 0, n 0} is dynamic since at any particular
time instant n, the value of y(n) does not only depend on x(n) but also on prior values of
x(n) through the dependence on y(n1). Indeed, assume we iterate the input-output relation.
Then we find
y(0) = x(0)
y(1) = y(0) + x(1) = x(0) + x(1)
y(2) = y(1) + x(2) = x(0) + x(1) + x(2)
..
.
.
= ..
and it is seen, for example, that the values of y(1) and y(2) depend on the value of x(0) as
well.
(3) The system y(n) = x(n2 ) is dynamic. Note for instance that y(2) = x(4) so that the value of
the output sequence at time n = 2 depends on the value of the input sequence at time n = 4.
(4) The system y(n) = x(n 1) is also dynamic. Note that y(2) = x(1) so that the output value
at time n = 2 depends on the input value at time n = 1.
(5) The system
1
[x(n) + x(n 1)]
2
is dynamic since the value of y(n) depends on x(n 1) as well.
y(n) =
(6) The system y(n) = x2 (n) is static.
4.5 TIME-INVARIANT SYSTEMS

A system is time-invariant if a time delay (or advance) in the input sequence yields an
identical time delay (or time advance) in the output sequence. In other words, if
y(n) = S[x(n)]
describes a system, then it should hold that
y(n K) = S[x(n K)]
for any integer K
(4.4)
Otherwise, the system is said to be time-variant. The integer k can be positive (in which
case the input and output sequences are delayed) or negative (in which case the input
and output sequences are advanced). What the definition of time-invariance implies is
the following. Assume the system is excited with an input sequence x(n) to generate
y(n). Then, if the same experiment is repeated some time later, the system will respond
in the same manner except that the new output sequence will be delayed in relation to the
previous output sequence. And the amount of delay will be the same as the delay in the
input sequence. Time-invariant systems are also referred to as shift-invariant systems to
accommodate situations where the variable n does not necessarily denote time.
To prove that a system is time-variant, it is enough to find a counter-example. That is,
it is enough to find a sequence x(n) for which the definition of time-invariance does not
hold.
On the other hand, to establish that a system is time-invariant we proceed as follows.
We denote by y(n) the output sequence that corresponds to a generic input sequence x(n).
We denote by yK (n) the output sequence that corresponds to the shifted input sequence
x(n K), for an arbitrary integer K (positive or negative). We then use the input-output
relation of the system to establish that for any such K, it holds that yK (n) = y(n K).
Example 4.8 (Time-invariant systems)
Let us prove that the system
y(n) = 2x(n 1)
is time-invariant. Let yK (n) denote the output sequence that corresponds to the input sequence
x(n K). It follows from the equation defining the system that
yK (n) = 2x(n K 1)
Now using the system description y(n) = 2x(n 1) we see that
y(n K) = 2x(n K 1)
Comparing the expressions for yK (n) and y(n K) we conclude that yK (n) = y(n K) for any
K and the system is therefore time-invariant.
Likewise, the system
y(n) = x(n 1) + x(n)
is time-invariant. Thus, let yK (n) denote the output sequence that corresponds to the input sequence
x(n K). It follows from the equation defining the system that
yK (n) = x(n K 1) + x(n K)
Using the system description y(n) = x(n 1) + x(n) we conclude that
y(n K) = x(n K 1) + x(n K)
Therefore, yK (n) = y(n K) for any K and the system is time-invariant.
Example 4.9 (Time-variant systems)

The system
{y(n) = y(n 1) + x(n), y(1) = 0}
is time-variant. To see this, we only need to compare the output sequences that result from the
choices x(n) = u(n) and x(n) = u(n + 5). The responses to these input sequences can be found
by iterating the input-output recursion for all values of n (positive and negative) as was done in
Example 4.2. Doing so leads to the following conclusion
(
n+1
0
x(n) = u(n) = y(n) =
for n 0
otherwise
and
x(n) = u(n + 5) = y(n) =
n+1
5
for n 5
otherwise
Both responses are depicted in Fig. 4.3. It is seen that the response to u(n) is not a delayed version
of the response to u(n + 5).
Likewise, the system, for n 0,
(
y(n) =
x(n)
0
n even
n odd
79
SECTION 4.5
TIME-INVARIANT
SYSTEMS
80
response to x(n) = u(n)
CHAPTER 4
DISCRETETIME
SYSTEMS
4
3
2
1
4 3 2 1
1 2 3
response to x(n) = u(n + 5)
4
3
2
1
8 7 6 5 4 3 2
1
1 2 3
2
3
4
5
FIGURE 4.3 Response sequences of the system {y(n) = y(n 1) + x(n), y(1) = 0} to the
input sequences x(n) = u(n) (top) and x(n) = u(n + 5) (bottom).
is time-variant. To verify this fact it is enough to consider x(n) = (n) and to check the forms of
the resulting output sequences for x(n) and x(n 1).
Also, the system y(n) = x(n2 ) is time-variant. This is because
y(n K) = x((n K)2 )
while the response to x(n K) is actually
yK (n) = x(n2 K)
To convince yourself that this is the case, choose x(n) = (n). Then plot the response y(n) =
x(n2 ). Now choose x(n) = (n 1) and plot the sequences {x((n 1)2 ), x(n2 1)}.
Which one of these sequences is the response to (n 1)?
81
Example 4.10 (Time-invariant relaxed systems)
SECTION 4.6
CAUSAL
SYSTEMS
The relaxed system

{y(n) = y(n 1) + x(n)
(relaxed)}
is time-invariant. Indeed, for an arbitrary sequence x(n), let no denote the time instant at which x(n)
becomes nonzero, i.e., x(n) = 0 for all n < no (if x(n) is always nonzero, then we set no = ).
As defined before, since the system is relaxed, we must have y(n) = 0 for all n < no . Iterating the
above input-output relation we get
y(n) =
n
X
x()
=n0
which shows that y(n) is the sum of all input samples up to and including time n. If we now delay
x(n) to x(n K), for an arbitrary positive integer K, then no is replaced by no + K (since the new
sequence is now zero for n < no + K. Let yK (n) denote the new output sequence and let xK (n)
denote the input sequence x(n K). Then yK (n) is given by
yK (n) =
n
X
n
X
xK () =
=no +K
=no +K
x( K)
Introduce the change of variables

= K
Then the above equality gives
yK (n) =
nK
X
x( )
=n0
In other words, it follows that

yK (n) = y(n K)
The same conclusion will hold when K is negative and we conclude that the relaxed system is timeinvariant.
4.6 CAUSAL SYSTEMS

A system is causal if its output at time n depends only on present and past values of the
input. In other words, y(n) depends only on x(m) for m n. Otherwise, the system is
noncausal. Likewise, a system is strictly causal if its output at time n depends only on
past values of the input sequence x(n) (i.e., for strict causality, the present value of x(n)
is excluded).
Example 4.11 (Causal and noncausal systems)
(1) The system y(n) = nx(n 1) is strictly causal.
(2) The systems y(n) =
1
2
[x(n) + x(n 1)] and y(n) = x2 (n) are causal.
(3) The system y(n) = x(n2 ) is not causal. For example, y(2) = x(4) and it is seen that the
output value at time n = 2 depends on a future value of the input sequence.
An important equivalent characterization of causality is the following. For any two input
sequences {x1 (n), x2 (n)}, it should hold that:
if x1 (n) = x2 (n) for n < N , then y1 (n) = y2 (n) for n < N
(4.5)
82
CHAPTER 4
DISCRETETIME
SYSTEMS
In other words, as long as the input sequences agree up to a certain time instant, the corresponding output sequences must also be identical up to that same time instant. Indeed,
since the output of a causal system at any particular time instant can only depend on present
and past input samples, we conclude that the values of the output sequences y1 (n) and
y2 (n) over n < N must agree because these samples are being evaluated from identical
input values.
4.7 STABLE SYSTEMS

A system is said to be bounded-input bounded-output (BIBO) stable if every bounded input
sequence x(n) yields a bounded output sequence y(n). By definition, a bounded sequence
x(n) is one for which all samples are bounded by some finite positive number Bx , namely,
|x(n)| Bx <
for all n
(4.6)
To prove that a system is not BIBO stable, it is sufficient to find a counter-example.

That is, it is enough to find a bounded sequence x(n) for which the output sequence y(n)
is unbounded.
On the other hand, in order to establish that a system is BIBO stable we proceed as
follows. We let y(n) denote the output sequence that corresponds to an arbitrary bounded
input sequence x(n). Then we use the input-output relation of the system to show that
there exists a finite positive number By < such that |y(n)| < By for all n.
Example 4.12 (Stable systems)
(1) The systems y(n) = 12 [x(n) + x(n 1)] and y(n) = x(n2 ) are BIBO stable. Indeed, let
x(n) denote any bounded sequence, say
|x(n)| Bx < for all n
Then for the first system we have
|y(n)|
1
|x(n) + x(n 1)|
2
1
1
|x(n)| + |x(n 1)|
2
2
Bx
Bx
+
2
2
Bx
and we conclude that the output sequence is also bounded. With regards to the second system
we have
|y(n)|
|x(n2 )| Bx
so that the output sequence is again bounded.

(2) The system y(n) = nx(n) is not BIBO stable. For instance, if we choose x(n) = 1 for all n,
then x(n) is clearly bounded while y(n) = n is unbounded.
(3) The system
y(n) =
1
y(n 1) + x(n), y(1) = 0, n 0
2
is BIBO stable. We can verify the stability of this system from first principles by applying the
definition. Iterating the input-output relation for n 0 we find that
83
SECTION 4.8
y(0)
y(1)
y(2)
..
.
In general, for any n 0, we get

y(n) =
x(0)
1
x(0) + x(1)
2
1
1
x(0) + x(1) + x(2)
4
2
..
.
n k
X
1
k=0
LINEAR
SYSTEMS
x(n k)
Now let x(n) denote any bounded input sequence satisfying

|x(n)| Bx <
for all n 0
Then
|y(n)|
n k
X
1
k=0
Bx
Bx
Bx
2Bx
|x(n k)|
n k
X
1
k=0

X
k=0
1
1
2
1
2
k

1
2
and the output sequence is bounded as well.
4.8 LINEAR SYSTEMS

A system is linear if it satisfies the superposition principle, which states that
ay1 (n) + by2 (n) = S[ax1 (n) + bx2 (n)]
(4.7)
for any constants {a, b}. In other words, the response of the system to any linear combination of input sequences is the same linear combination of the individual output sequences.
To be precise, we should also require the superposition property to hold for an infinite
combination of input signals but we forgo this technical detail here.
To prove that a system is not linear, it is sufficient to find a counter-example. That is, it
is enough to find a sequence x(n) for which the superposition property does not hold.
On the other hand, to establish that a system is linear we proceed as follows. We denote by y1 (n) the output sequence that corresponds to a generic input sequence x1 (n). We
denote by y2 (n) the output sequence that corresponds to another generic input sequence
x2 (n). We denote by y(n) the output sequence that corresponds to the linear combination ax1 (n) + bx2 (n) for any scalars {a, b}. We then proceed to show that y(n) satisfies
y(n) = ay1 (n) + by2 (n).
84
Example 4.13 (Two linear systems)
CHAPTER 4
DISCRETETIME
SYSTEMS
Let us verify that y(n) = 2x(n 1) is a linear system. Let y1 (n) denote the output sequence
that corresponds to an input sequence x1 (n). Likewise, let y2 (n) denote the output sequence that
corresponds to another input sequence x2 (n). Then
y1 (n) = 2x1 (n 1),
y2 (n) = 2x2 (n 1)
Now let y(n) denote the output sequence that corresponds to the linear combination ax1 (n) +
bx2 (n), for any scalars {a, b}. Then, from the system description,
y(n)
=
=
=
2[ax1 (n 1) + bx2 (n 1)]
a[2x1 (n 1)] + b[2x2 (n 1)]

ay1 (n) + by2 (n)
Therefore, the system is linear. Let us now prove that the following system
{y(n) = y(n 1) + x(n), y(1) = 0}
is also linear. Let y1 (n) denote the output sequence that corresponds to an input sequence x1 (n).
Likewise, let y2 (n) denote the output sequence that corresponds to another input sequence x2 (n).
Then it is easy to verify that for n 0:
y1 (n)
y2 (n)
x1 (0) + x1 (1) + x1 (2) + . . . + x1 (n) =

x2 (0) + x2 (1) + x2 (2) + . . . + x2 (n) =
n
X
k=0
n
X
x1 (k)
x2 (k)
k=0
while for n < 1:

y1 (n)
y2 (n)
x1 (n + 1) x1 (n + 2) x1 (n + 3) . . . x1 (1) =
x2 (n + 1) x2 (n + 2) x2 (n + 3) . . . x2 (1) =
1
X
x1 (k)
k=n+1
1
X
x2 (k)
k=n+1
and
y1 (1) = 0 = y2 (1) = 0
Now let y(n) denote the output sequence that corresponds to the linear combination ax1 (n) +
bx2 (n), for any scalars {a, b}. Then, we also get
8
>
>
>
>
<
y(n) =
n
P
[ax1 (k) + bx2 (k)]
k=0
>
1
>
P
>
>
[ax1 (k) + bx2 (k)]
:
k=n+1
n0
n = 1
n < 1
so that
y(n) = ay1 (n) + by2 (n),
and the system is linear.
Example 4.14 (Linear and nonlinear systems)
for all n
(1) The systems y(n) = nx(n 1) and y(n) = 12 [x(n) + x(n 1)] are linear, as can be
immediately verified by applying the definition of linearity.
(2) The system, for n 0,
85
y(n) =
x(n)
0
SECTION 4.8
n even
n odd
LINEAR
SYSTEMS
is also linear.
(3) The systems y(n) = x2 (n) and y(n) = ejx(n) are not linear.
(4) The system y(n) = x(2n + 3) is linear while y(n) = 2x(n) + 3 is not linear.
(5) The system {y(n) = y(n 1) + x(n), y(1) = 1} is not linear. As a general rule, nonzero
initial conditions for constant-coefficient difference equations (which are defined in the sequel
in Sec. 4.9) destroy linearity.
(7) The system y(n) = y(n 1) + x(n) when relaxed is linear. Recall again that being relaxed
does not mean that y(1) = 0 but rather that the output is zero while the input is zero. For
this example, this is equivalent to saying that the output is zero just prior to the first nonzero
sample of the input.
Additivity and Homogeneity

We further note that the following two conditions combined are equivalent to the property
of linearity. In other words, if a system satisfies both of these properties then the system is
necessarily linear and vice-versa:
(A) Additivity. If y1 (n) = S[x1 (n)], y2 (n) = S[x2 (n)], then
S[x1 (n) + x2 (n)] = y1 (n) + y2 (n)
This means that the response to the sum of input sequences is the sum of the individual output sequences.
(H) Homogeneity. If y(n) = S[x(n)] then
S[ax(n)] = ay(n) for any scalar a
This means that the response to a scaled input sequence is the corresponding scaled
output sequence.
Proof: To establish the equivalence between these two properties and linearity, let us assume first that
the properties of additivity and homogeneity hold and let us prove that the system y(n) = S[x(n)] is
linear. Thus consider any linear combination of input sequences of the form ax1 (n) + bx2 (n). Then
S[ax1 (n) + bx2 (n)]
=
=
S[ax1 (n)] + S[bx2 (n)]
aS[x1 (n)] + bS[x2 (n)]
(by additivity)
(by homogeneity)
which shows that the system is linear. Conversely, assume the system y(n) = S[x(n)] is linear so
that, by definition of linearity,
S[ax1 (n) + bx2 (n)] = aS[x1 (n)] + bS[x2 (n)]
for any scalars {a, b}. Choosing b = 0, we get
S[ax1 (n)] = aS[x1 (n)]
which shows that the system satisfies the homogeneity property. Choosing a = 1 and b = 1, we get
S[x1 (n) + x2 (n)] = S[x1 (n)] + S[x2 (n)]
86
which shows that the system satisfies the additivity property.
CHAPTER 4
DISCRETETIME
SYSTEMS
Absence of Excitation
A useful property of linear systems is the following: the output sequence of the system is
necessarily zero when the input sequence is the zero sequence. That is,
if x(n) = 0 for all n then y(n) = 0 for all n
(4.8)
We therefore conclude that a linear system cannot generate a nonzero output sequence
without excitation. Only nonlinear systems can generate nonzero outputs without excitation at the input.
Proof: Let y1 (n) denote the output sequence that corresponds to a sequence x1 (n). Let y2 (n) denote
the output sequence that corresponds to x2 (n) = x1 (n). We know from the homogeneity property
that y2 (n) = y1 (n). Now, by the additivity property, the output that corresponds to x1 (n) + x2 (n)
is y1 (n) + y2 (n). But both sequences are zero since x1 (n) + x2 (n) = 0 and y1 (n) + y2 (n) = 0.
Hence, the response to the zero input sequence is the zero output sequence.
The absence of excitation property of a linear systems does not mean that a linear system is a relaxed system. Consider, for example, the system y(n) = x(n + 1). This is a
linear system but, as mentioned earlier, it is not relaxed.
4.9 CONSTANT-COEFFICIENT DIFFERENCE EQUATIONS

In this book we shall primarily deal with systems that are described by difference equations
with constant coefficients, namely, by input-output relations of the form:
y(n) =
M
X
k=1
ak y(n k) +
N
X
k=0
bk x(n k)
(4.9)
for some constant coefficients {ak , bk }. In this description, for each n, the sample y(n) is
a linear combination of past output samples and of present and past input samples. More
explicitly, we write
y(n) = a1 y(n1)+a2 y(n2)+. . .+aM y(nM ) + b0 x(n)+b1 x(n1)+. . .+bN x(nN )
The following remarks are therefore in place:
(1) As explained before, the above difference equation describes a class of systems,
unless additional information (such as initial conditions) is available along with the
input sequence.
(2) The above difference equation describes a relaxed system if it is assumed that the
output sequence is zero while the input sequence is zero (see Example 4.10):
if x(n) = 0 for n < no , then y(n) = 0 for n < no
(3) A relaxed difference equation as above describes a linear time-invariant (LTI) system
(recall Examples 4.10 and 4.13).
In other words, whenever we say that we are given a relaxed system that is described by a
constant-coefficient difference equation, then we can immediately conclude that the system
should be linear and time-invariant, written as LTI. This a very important subclass of systems and it will be the focus of much of our studies in the remainder of this book. Specifically, we shall often deal with systems described by difference equations similar to (4.9) but
restricted to the interval n 0 and with initial conditions {y(1), y(2), . . . , y(M )}.
Such systems are linear (Example 4.13) but generally time-variant (Example 4.9).
It should be noted that a constant-coefficient difference equation can describe either a
causal system or a noncausal system. For example, the description
y(n) = y(n 1) + x(n)
when used to determine y(n) from y(n 1) and x(n) is a causal system. If we instead
rewrite it as
y(n 1) = y(n) x(n)
and use this recursion to compute y(n 1) from y(n) and x(n) then we have a non-causal
system. In the first case, time progresses forwards while in the second case time progresses
backwards. For this reason, we always need to specify the direction of evolution of time
in order to be able to conclude whether a given difference equation refers to a causal or
noncausal system.
4.10 SYSTEM REPRESENTATIONS

Besides mathematical equations, most systems that we shall encounter in this book can
be described in terms of block diagrams or signal flowgraph diagrams by employing three
basic components, as we explain below. We shall have much more to say about system
realizations in Chapter 23. Here we only introduce some initial ideas.
Block Diagrams
The three basic elements that we shall employ to describe systems in block diagrams are
unit-time delays, multipliers by constants, and adders.
Unit time-delays. These elements are denoted by the symbol z 1 . If the input to a unittime delay is x(n) then its output is x(n 1), as illustrated in Fig. 4.4. The reason for the
use of the symbol z 1 to denote the unit delay block will become clearer later when we
study the shift property of the ztransform in Sec. 9.4.
x(n)
z 1
x(n 1)
FIGURE 4.4 The unit delay operator is denoted by the symbol z 1 . If x(n) is the input to the
block then x(n 1) is its output. In other words, the input sequence is delayed by one unit of time.
Multipliers by constants. This operation is usually denoted by an arrow with the value
of the constant written next to it see Fig. 4.5. If the input to such a block is x(n), then
its output is x(n) with denoting the multiplication constant.
87
SECTION 4.10
SYSTEM
REPRESENTATIONS
88
CHAPTER 4
DISCRETETIME
SYSTEMS
x(n)
FIGURE 4.5
by .
x(n)
An arrow with a constant next to it signifies multiplication of the signal samples
Adders. An input to an adder usually consists of two sequences, x1 (n) and x2 (n), and
the output is the sequence x1 (n) + x2 (n). An adder is often denoted by a small circle with
a plus sign, two arrows arriving at the circle, and one arrow leaving it see Fig. 4.6.
x1 (n)
x1 (n) + x2 (n)
+
x2 (n)
FIGURE 4.6 An adder adds the samples of the incident signals, x1 (n) and x2 (n), and generates
x1 (n) + x2 (n).
Signal Flowgraph Diagrams

An alternative representation to block diagrams is in terms of signal flowgraph diagrams.
In these diagrams, unit-time delays are denoted by an arrow with the symbol z 1 next to it,
and adders are denoted by small circles without the plus sign inside them. Multiplications
by constants continue to be represented by arrows with the constants listed next to them.
The direction of the arrows indicate the direction of propagation of the signals. Figure 4.7
illustrates the flowchart representations of the three basic elements just mentioned.
Example 4.15 (Two block diagram representations)

Figure 4.8 shows two block diagram representations for the class of systems
{y(n) = y(n 1) + x(n 1) 2x(n)}
The diagrams are obtained as follows.
(1) Consider initially the block diagram representation shown in the top part of Fig. 4.8. We start
by writing down the input and output terminals, x(n) and y(n). We then employ a unit-time
delay to generate x(n 1) from x(n), and another unit-time delay to generate y(n 1) from
y(n). We further multiply x(n) by the constant 2 and add outputs of the multiplier and the
delay on the leftside to obtain x(n 1) 2x(n). We subsequently add this result to y(n 1)
to obtain y(n).
(2) Block diagram representations are not unique. While the block diagram representation in the
top part of Fig. 4.8 requires two unit-time delays, the representation in the bottom part of the
89
x(n)
z 1
x(n)
SECTION 4.11
APPLICATIONS
x(n 1)
x(n)
x1 (n)
x1 (n) + x2 (n)
x2 (n)
FIGURE 4.7 Flowgraph representations of the unit-time delay (top), multiplication by a constant
(middle), and adders (bottom).
same figure employs only one unit-time delay. This alternative implementation is obtained as
follows. We first add x(n) to y(n) and then delay the combination to obtain x(n 1) + y(n
1). The delayed combination is then added to 2x(n) to generate y(n).
(3) Figure 4.9 shows the corresponding flowchart representations for the same diagrams.
x(n)
z 1
y(n)
z 1
x(n)
z 1
y(n)
FIGURE 4.8 Two block diagram representations for the class of systems {y(n) = y(n 1) +
x(n1)2x(n)}. The representation in the top employs two unit delays, whereas the representation
in the bottom employs a single unit delay.
90
CHAPTER 4
x(n)
DISCRETETIME
SYSTEMS
y(n)
z 1
z 1
x(n)
FIGURE 4.9
z 1
y(n)
Signal flowgraph representations of the block diagrams shown in Fig. 4.8.
4.11 APPLICATIONS
In this section, we illustrate applications of some of the concepts covered in the chapter
in the context of several practical problems. Specifically, we comment on how constantcoefficient difference equations are useful modeling tools in communications, finance, and
population growth.
4.11.1 Multipath Communications

In wireless communications, a signal is transmitted from a user to a base station through
the air using electromagnetic waves. The base station subsequently routes the signal to
its intended recipient. Propagation in a wireless medium suffers from multipath effects.
During propagation, the waves may bounce off obstacles such as hills, trees, and buildings
before arriving at the base station through different paths. The bounced signals arrive at the
base station with different delays and different energy levels. In this way, the base station
receives several replicas of the transmitted signals: these replicas generally arrive at different time instants and with different attenuation levels relative to the original transmitted
signal.
Let us denote the signal that leaves the originating users cell phone by x(n). There
may exist a direct path between the user and the base station, one that does not entail any
reflections or bouncing. If the signal travels through this path, it will still arrive at the base
station with some delay and with some attenuation, say, after some delay of 2 units of time
and amplitude attenuation equal to 5/6. Therefore, the first component of the received
signal at the base station can be written as
y1 (n) =
5
x(n 2)
6
Assuming some hills and high-rise buildings are present in the surroundings, the signal
x(n) may get bounced off these obstacles and replicas will also arrive at the base station.
Let us consider the case of two replicas: one delayed by 3 units of time and scaled by 1/4
91
and the other delayed by 5 units of time and scaled by 1/8. Then
SECTION 4.11
y2 (n) =
1
x(n 3)
4
and
y3 (n) =
1
x(n 5)
8
APPLICATIONS
In this way, the overall received signal at the base station is the combination of all three
signals {y1 (n), y2 (n), y3 (n)} and is given by
y(n) =
5
1
1
x(n 2) + x(n 3) + x(n 5)
6
4
8
(4.10)
We therefore arrive at a constant-coefficient difference equation relating the input sequence

x(n) to the received signal y(n). We say that this equation models the multipath communications channel; here x(n) denotes the input sequence and y(n) denotes the output
sequence, as illustrated in Fig. 4.10. The system in this case can be easily verified to be a
causal LTI system.
Path 2:
scaling = 1/4
delay = 3
Path 1:
scaling = 5/6
delay = 2
y(n)
x(n)
caller
Path 3:
scaling = 1/8
delay = 5
base station
FIGURE 4.10 A multipath communications channel consisting of three paths: the delays and
scaling factors of the paths are indicated in the figure.
Some questions of interest in this wireless communications scenario relate to the following:
(a) Assume the base station collects N data samples {y(0), y(1), . . . , y(N 1)} and has
access to the transmitted data {x(0), x(1), . . . , x(N 1)}. How can these samples
be used to estimate the channel coefficients { 65 , 14 , 81 }? This question relates to a
so-called channel estimation problem, which we shall study later in Sec. 37.4.
(b) Assume again that the base station collects N samples {y(0), y(1), . . . , y(N 1)},
and that the channel coefficients are now known. How can this information be used
to recover the original input sequence x(n) over the same interval 0 n N 1?
This question relates to a so-called channel equalization problem, which we shall
study later in Sec. 37.5.
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SYSTEMS
Practice Questions:
1. Draw a signal flowgraph diagram for the multipath channel described by the difference equation (4.10).
2. The transmitted signal x(n) is usually zero for negative time since transmission starts at n =
0. What are the values of y(0) and y(1) in (4.10)?
3. Assume we know that y(2) = 1, y(3) = 3/4, y(4) = 1/3, and y(5) = 31/48. Can you
determine the values of {x(0), x(1), x(2), x(3)}?
4. Which measurements y(n) are needed to recover the values of {x(4), x(5), x(6), x(7)}?

We reconsider in greater detail the savings account example of Sec. 2.7. A client opens a
savings account with an initial deposit of D dollars. The time at which the account was
created is selected as the origin of time, say, as n = 0. We assume an annual return rate
of % and let y(n) denote the amount of funds that will be present at the start of year n.
Then, the variable y(n) evolves according to the recursion:

y(n) = 1 +
y(n 1), y(0) = D = initial funds
100
(4.11)
This is a constant-coefficient difference equation, which evolves over n > 0 starting from
the initial condition y(0) = D. It is easy to determine an expression for y(n) as a function of n by iterating the above recursion, which leads to the exponential growth model
(illustrated generically in Fig. 4.11):4
h
in
D
y(n) = 1 +
100
(4.12)
Before generalizing the model, it is worth noting that we can rewrite recursion (4.11)
in an alternative and useful form. This second form assumes a zero initial condition at
time n = 1 and incorporates an input sequence of the form x(n) = 1000 (n) into the
recursion as follows:

y(n) = 1 +
y(n 1) + 1000 (n), y(1) = 0
(4.13)
100
Recursion (4.13) now runs over n 0 (and not n > 0). The unit-sample sequence, (n),
assumes the value 1 at n = 0 and disappears for all other values of n. Therefore, if we use
(4.13) to evaluate y(0) we get

y(1) + 1000 (0)
y(0) =
1+
100

=
1+
0 + 1000 1
100
= 1000
which is the same value used in (4.11). Subsequently, for all values of n > 0, recursion
(4.13) becomes identical to recursion (4.11) since the (n) term will only contribute with
zero values over n > 0.
4 Source
of this placeholder image is istockphoto.com
93
SECTION 4.11
APPLICATIONS
FIGURE 4.11 An illustration of a growing financial application.
Recursion (4.13) describes a causal LTI system with input sequence equal to 1000 (n)
and output sequence y(n). More generally, the client may wish to influence the growth of
the funds in the account by making additional annual deposits (or even withdrawals; apart
from the initial deposit of US$1000 at time n = 0). Thus, let x(n) denote the amount of
funds that the client deposits (or withdraws) into the account at the start of year n. Then
recursion (4.13) can be replaced by the more general form:

y(n) = 1 +
y(n 1) + x(n),
100
y(1) = 0
(4.14)
where x(0) = D corresponds to the initial deposit. This is a constant-coefficient difference

equation and it represents a causal LTI system with input x(n) and output y(n); the system
is clearly relaxed since the funds in the account will be zero while the client does not make
any deposits.
We can evaluate the successive values of y(n) by iterating the recursion. However, it
is generally not possible to (guess or) arrive at a closed-form expression for y(n) in terms
of n by simply iterating (4.14) for arbitrary sequences x(n), as was the case in (4.12) for
recursion (4.11). Nevertheless, later in Chapters 8 and 12, we shall develop techniques that
will allow us to determine, in a systematic manner, a closed-form expression for y(n) as a
function of n for difference equations involving input sequences x(n), such as (4.14).
When the client makes regular annual deposits, say, of d dollars from year one onwards,
then the sequence x(n) can be expressed as
x(n) = D(n) + d u(n 1)
in terms of the step sequence. In this case, we can arrive at a closed-form expression for
y(n) by iterating (4.14). Indeed, let

= 1+
100
(4.15)
94
and note the following sequence of calculations that follow from (4.14):
CHAPTER 4
DISCRETETIME
SYSTEMS
y(0) =
y(1) =
D
D + d
y(2)
y(3)
..
.
y(n)
y(1) + d
= 2 D + (1 + )d
y(2) + d
= 3 D + (1 + + 2 )d
..
.
y(n 1) + d = n D + (1 + + 2 + . . . + n1 )d
=
=
=
=
Using the result of Example 2.11 for the sum of the first n terms of a geometric series, we
get
1 n
d
y(n) = n D +
1
Substituting by its value (4.15) in terms of , we arrive at
h
i
in
100 h
n
y(n) = 1 +
D +
1 d, n 0
1+
100
100
(4.16)
which can be compared with (4.12). Let us consider a numerical example. Assume =
2%, D =US$1000, and the client makes regular annual deposits of US$50 from year one
onwards. Then the amount of funds in the account will evolve according to the recursion
y(n) = 1.02 y(n 1) + 1000 (n) + 50 u(n 1),
y(1) = 0
(4.17)
Without the regular deposits, the amount of funds in the account would evolve instead
according to
y(n) = 1.02 y(n 1) + 1000 (n), y(1) = 0
(4.18)
Table 4.1 and Fig. 4.12 compare the funds in both kinds of accounts (with and without
regular annual deposits).
TABLE 4.1 Evolution of the funds in the savings account with and without regular annual deposits
of US$50, assuming an annual return rate of 2% and an initial deposit of US$1000.
Without regular
With regular
Year
deposits
deposits
n=0
n=1
n=2
n=3
n=4
n=5
1000.00
1020.00
1040.40
1061.20
1082.40
1104.00
1000.00
1070.00
1141.40
1214.20
1288.50
1364.30
Practice Questions:
1. Starting with an initial deposit of US$500 at year n = 0, and making regular annual deposits
of US$20 from year one onwards, how much funds will be available at the start of year n = 30
assuming an annual return rate of 2%.
of US$20 from year four onwards, how much funds will be available at the start of year
n = 30 assuming the same annual return rate of 2%.
95
1800
SECTION 4.11
APPLICATIONS
1700
only based on
initial deposit
1600
with regular
annual deposits
funds
1500
1400
1300
1200
1100
1000
5
6
n (years)
10
FIGURE 4.12 Evolution of the funds in the savings account with and without regular annual
deposits of US$50, assuming an annual return rate of 2% and an initial deposit of US$1000.
of US$20 from year one onwards, what should the annual return rate be such that the amount
of funds that are present at the start of year n = 10 would be equal to the amount of funds
that are present at the start of year 30 at 2% annual return?
4.11.3 Population Growth Models

Mathematical models can be used to model the dynamics of population growth in humans
and animal species. Let y(0) = Po denote the initial number of individuals in a population
at some reference of time, which we select to correspond to n = 0. We would like to determine an expression for the evolution of the population size, y(n), as a function of time.
We assume the time index, n, is measured in years so that y(0) is the population size at the
beginning of year n = 0, and y(1) is the population size at the beginning of year n = 1,
and so on. Figure 4.13 depicts a trend with a growing population size.5
Malthusian Model
Now the size of the population decreases or increases as a function of the number of births
and deaths that occur annually. Let us assume that the birth rate is b% per year and that
the death rate is d% per year, where both rates are percentages relative to the existing
population size. In this way, given y(n 1), the population size at year n will be given by
y(n) = y(n 1) +
5 Source
b
d
y(n 1)
y(n 1)
100
100
of this placeholder image is istockphoto.com
96
CHAPTER 4
DISCRETETIME
SYSTEMS
FIGURE 4.13 An illustration of a growing population trend. Mathematical models can be used
to model the dynamics of population growth in humans and animal groups.
Grouping terms we arrive at the following so-called Malthusian model for population
growth

b
d
y(n) = 1 +
y(n 1), y(0) = initial population size
100 100
(4.19)
This model involves a constant-coefficient difference equation, which evolves over n >
0 starting from an initial condition, y(0) = Po , and progresses forward in time. It is
straightforward to verify by iterating the Malthusian recursion that the population size
grows exponentially as follows:

n
b
d
y(n) = 1 +
Po
(4.20)
100 100
For this reason, the Malthusian model is sometimes referred to as an exponential growth
model. For example, given an initial population size of Po = 100 individuals and assuming
the birth and death rates are 10% and 2%, respectively, we find that the population size
evolves according to the numbers listed in Table 4.2, which follow from the relation
y(n) = 1.12 y(n 1), y(0) = 100
Before generalizing the Malthusian model (4.19), it is worth noting that we can rewrite
the recursion in an alternative and useful form. This second form assumes a zero initial
condition at time n = 1 and incorporates an input sequence of the form x(n) = Po (n)
into the recursion as follows:

b
d
y(n) = 1 +
y(n 1) + Po (n), y(1) = 0
(4.21)
100 100
97
TABLE 4.2 Evolution of population size using the Malthusian model (4.19) with y(0) = 100,
b = 10% and d = 2%.
Year
Size
n=0
n=1
n=2
n=3
n=4
n=5
..
.
100
108
116
125
135
145
..
.
Recursion (4.21) now runs over n 0 (and not n > 0). The unit-sample sequence, (n),
assumes the value 1 at n = 0 and disappears for all other values of n. Therefore, if we use
(4.21) to evaluate y(0) we get
y(0) =
=
=

d
b
y(1) + Po (0)
100 100

b
d
1+
0 + Po 1
100 100
Po
1+
which is the same value used in (4.19). Subsequently, for all values of n > 0, recursion
(4.21) becomes identical to recursion (4.19) since the (n) term will only contribute with
zero values over n > 0.
Recursion (4.21) describes a causal LTI system with input sequence equal to Po (n)
and output sequence y(n). More generally, we can study the situation where one wishes to
influence the dynamics of the population growth in a given society by adding (or even removing) individuals at various instants of time. This fact can be modeled by incorporating
a more general input sequence, x(n), into the Malthusian recursion. For example, assume
at year n = 5, a total of 50 individuals are added to the population. Then the Malthusian
model (4.21) can be adjusted to capture this addition as follows:

d
b
y(n 1) + Po (n) + 50 (n 5), y(1) = 0 (4.22)
y(n) = 1 +
100 100

This recursion incorporates a driving signal in the form of two unit-sample sequences located at n = 0 and n = 5. For more general input sequences, x(n), the model becomes

b
d
y(n) = 1 +
y(n 1) + x(n),
100 100
y(1) = 0
(4.23)
where x(0) = Po corresponds to the initial population size. In this case, the sequence
x(n) represents the input to the system at the various time instants. The above constantcoefficient difference equation can be easily seen to represent a causal LTI system with
input x(n) and output y(n).
SECTION 4.11
APPLICATIONS
98
CHAPTER 4
DISCRETETIME
SYSTEMS
Logistic Model
Continuing with the original Malthusian model (4.20), we see that according to this model,
population growth follows an exponential evolution. This is not a realistic model since
it predicts that the population size will grow or decline in an exponential manner. An
alternative model is often used in practice and is known as the discrete-time logistic model.
The main difference is that in the new model, the birth and death rates are not constants but
are now functions of the size of the population at any particular time instant. For example,
when the population size is very large, it is expected that the death rate in the population
should become higher due to competition on limited resources. On the other hand, when
the population size is relatively small, the birth rate should become higher to enable the
perpetuation of the species. We arrive at the logistic model as follows.
At any time n, we measure the rate of change in the population size relative to time
n 1 through the ratio
y(n) y(n 1)
(n) =
(4.24)
y(n 1)
We refer to this ratio as the population growth rate. We would like this ratio to assume
large values when the population size, y(n 1), is small (so that the population would
grow), and to assume small values (or even negative values) when the population size,
y(n 1), is large (so that the population would decline). One way to achieve this behavior
is to set

y(n 1)
(n) = 1
(4.25)
P
for some positive constants and P . If we drop the time indices for simplicity, the above
equation can be rewritten as

y
= 1
(4.26)
P
which can be recognized as the equation of a straight line with a negative slope relating
and y see Fig. 4.14. From the figure we see that when y is small, y is large (attaining
the value at y = 0), and when y is large, y is small or negative (attaining the value zero
at y = P ).
Reincorporating the time indices, and expanding the equation (4.25), we find that the logistic model is given by the following expression:

y(n 1)
y(n) = y(n 1) 1 + 1
, y(0) = initial condition
P
(4.27)
This model now involves a quadratic term in y(n 1) on the right-hand side. For this
reason, the logistic model does not correspond to a constant-coefficient difference equation
of the form we studied in this chapter. In such cases, it is generally difficult to obtain a
closed form expression that describes the evolution of y(n) as a function of n, as was
the case with (4.20). Nevertheless, we can iterate the logistic model and simulate the
evolution of the population growth. We may again consider the more general case in which
one attempts to influence the evolution of the population growth through the addition of
individuals. In this case, we can adjust the logistic model to the following form:

y(n 1)
+ x(n),
y(n) = y(n 1) 1 + 1
P
y(1) = 0
(4.28)
and iterate it over n 0. Here, the sequence x(n) represents the individuals that are added
at the various time instants n with x(0) = Po . Relation (4.28) represents a nonlinear
system with input sequence, x(n), and output sequence, y(n).
99
SECTION 4.11
APPLICATIONS
Population size at which the

rate of growth becomes negative.
Value of rate of
growth when
the population size
is small.
FIGURE 4.14 In the logistic model, the graph of the population growth rate () versus the
population size (y) is a straight line with a negative slope.
Let us reconsider the earlier example with an initial population size of 100 individuals
and use P = 400 and = 0.8, and apply the logistic model (4.27) to it. Iterating (4.27)
we find that the population size evolves according to the numbers listed in Table 4.3, which
follow from the recursion

y(n 1)
, y(0) = 100
y(n) = y(n 1) 1 + 0.8 1
400
In addition, Fig. 4.15 compares the evolution of an initial population of y(0) = 1000
individuals according to the Malthusian and logistic models; observe how the population
size in the latter model tends to a steady-state value.
TABLE 4.3 Evolution of population size using the logistic model (4.27) with y(0) = 100, =
0.8 and P = 400.
Year
Size
n=0
n=1
n=2
n=3
n=4
n=5
n=6
..
.
100
160
236
313
367
391
398
..
.
100
CHAPTER 4
500
DISCRETETIME
SYSTEMS
450
Malthusian
model
Logistic
model
population size
400
350
300
250
200
150
100
10
n (years)
FIGURE 4.15 Evolution of population size using both the Malthusian model (4.19) with y(0) =
100, b = 10% and d = 2%, and the logistic model (4.27) with y(0) = 100, = 0.8 and P = 400.
Practice Questions:
1. Refer to the Malthusian model (4.19). For what conditions on b and d, the model will correspond to a growing population, a declining population, a stable population?
2. Assume you are given the data in Table 4.2 showing the population growth as predicted by the
Malthusian model starting from an initial population of y(0) = 100. Can you work backwards
and find the birth rate and the death rate in the population?
3. Assume the population size y(n 1) is very small in the logistic model (4.27) so that the term
y(n 1)/P 0 and y(n) (1 + )y(n 1). Compare this relation with the Malthusian
model and provide an interpretation for in terms of b and d.
4. Verify that the system described by the logistic model (4.28) is nonlinear.
4.12 PROBLEMS
Problem 4.1 Determine whether each of the following systems is linear:
(a) y(n) = x(n3 ).
(b) y(n) = x(2n).
(c) y(n) = x(n/3) when n = 0, 3, 6, . . . and y(n) = 0 otherwise.
Problem 4.2 Determine whether each of the following systems is linear:
(a) y(n) = x(n2 ).
(b) y(n) = x(3n 2).
Problem 4.3 Determine whether each of the following systems is time-invariant:
(a) y(n) = x(n3 ).
101
(b) y(n) = x(2n).
SECTION 4.12
PROBLEMS

Problem 4.4 Determine whether each of the following systems is time-invariant:
(a) y(n) = x(n2 ).
(b) y(n) = x(3n 2).
Problem 4.5 Determine whether each of the following systems is causal:

(a) y(n) = x(n3 ).
(b) y(n) = x(2n).
Problem 4.6 Determine whether each of the following systems is causal:
(a) y(n) = x(n2 ).
(b) y(n) = x(3n 2).
Problem 4.7 Consider the moving average system with exponential weighting
y(n) =
M
1 X k
x(n k)
M +1
k=0
where || < 1. Is the system linear? causal? time-invariant? stable?

Problem 4.8 Is the system
y(n) =
linear? causal? time-invariant? stable?

1
x n2
|n| + 1
Problem 4.9 True or False:

(a) A relaxed system is linear.
(b) A linear system is relaxed.
Problem 4.10 True or False:
(a) An LTI system is relaxed.
(b) A linear and relaxed system is time-invariant.
Problem 4.11 The response of a linear time-invariant system to x(n) = u(n) is y(n) = (0.5)n u(n).
Find its response to (n).
Problem 4.12 The response of a linear time-invariant system to x(n) = 2(n 1) is y(n) =
(0.5)n u(n). Find its response to (n) + 0.5(n + 2).
Problem 4.13 System S1 is defined by y(n) = log (|(x(n 1)|) and system S2 is defined by
y(n) = exp(x(2n)). Which of the following statements is correct?
(a) Both systems are BIBO stable.
(b) Both systems are unstable.
(c) System S1 is unstable and system S2 is BIBO stable.
(d) Both systems are time invariant.
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CHAPTER 4
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SYSTEMS
Problem 4.14 Which of the following statements is true?

(a) System y(n) = sin(5n + 3)x(n) is linear.
(b) System y(n) = x(2n) is time invariant.
(c) System y(n) = x(n) is causal.
(d) System y(n) = x(n)x(n2 )x(n3 ) is unstable.
Problem 4.15 Determine the response of the system
y(n) = 0.5n x(n) + x(n 1)
to the unit sample x(n) = (n). Is this a linear system? What would its response be to x(n) =
(n 3)?
Problem 4.16 Determine the response of the system
2
y(n) = 0.5n x(n 1) + x(n)

to the unit sample x(n) = (n). Is this a linear system? What would its response be to x(n) =
(n + 1)?
Problem 4.17 Determine whether each of the following systems is linear or not, time-invariant or
not, causal or not, relaxed or not:
(a) y(n) = y(n 1) + x(n), y(1) = 0.
(b) y(n) = y(n 1) + x(n), y(1) = 1.
(c) y(n) = y(n 1) + x(n), and the output is zero as long as the input is zero.
(d) y(n) = y(n 1) + x(n), and the output is zero right before the first nonzero sample in the
input sequence.
(e) y(n 1) = y(n) x(n), and the output is zero as long as the input is zero.
Problem 4.18 Determine whether each of the following systems is linear or not, time-invariant or
not, causal or not, BIBO stable or not, relaxed or not:
(a) y(n) = ln[|x(n)| + 1].
(b) y(n) = y(n 1) + x(n), y(1) = 0.
(c) y(n) = y(n 1) + x(n), y(1) = 1.
(d) y(n) = 2 + x(n).
Problem 4.19 Show that the following system is BIBO stable for any || < 1:
{y(n) = y(n 1) + x(n), y(1) = 0, n 0}
Problem 4.20 Prove that the system below is BIBO stable:

y(n) =
1
y(n 2) + x(n), y(1) = 0, y(2) = 0, n 0
4
Problem 4.21 Show that y(n) = x(2n + 1) is a time-variant system.

Problem 4.22 Let y(n) = x(n2 1). Is the system linear? time-invariant? causal? stable?
Problem 4.23 Let y(n) = ejx(n) . Is the system linear? time-invariant? causal? stable?
Problem 4.24 Let y(n) = ejx(n) cos(2x(n)). Is the system linear? time-invariant? causal?
stable?
Problem 4.25 The response of a linear system to 0.5n u(n) is u(n) and to 0.5n1 u(n 1) is
u(n 1). Is the system time-invariant?
Problem 4.26 The response of an LTI system to 0.5n u(n) is u(n). Is the system causal?
103
SECTION 4.12
Problem 4.27 Given an example of a system that satisfies the homogeneity property but does not
satisfy the additivity property.
Problem 4.28 Given an example of a system that satisfies the additivity property but does not
satisfy the homogeneity property.
Problem 4.29 Consider the system with output sequence y(n) and input sequence x(n) related via
y(n) =
m=
x(m)x(n m)
Verify whether the system is linear, causal, time-invariant, or stable. Justify your statements or give
valid counter-examples.
Problem 4.30 Verify whether the system

y(n) =
1
y(n 2) + x(n 3), y(1) = 0, y(2) = 0, n 0
4
satisfies the homogeneity property, the additivity property, and the superposition property. Is the
system linear? time-invariant?
Problem 4.31 Establish the validity of the following equivalent characterization of causality: as
long as two input sequences to a causal system are identical, the corresponding output sequences
must also be identical. That is, for any sequences {x1 (n), x2 (n)}, if x1 (n) = x2 (n) for n < N
then y1 (n) = y2 (n) for n < N .
Problem 4.32 Give examples of systems that are described by a constant-coefficient difference
equation and are
(a) Linear, time-invariant, and causal.
(b) Linear, time-invariant, and non-causal.
(c) Linear, time-variant, and causal.
(d) Linear, time-variant, and non-causal.
(e) Nonlinear.
Problem 4.33 A relaxed system is described by the difference equation
y(n)
1
y(n 1) = x2 (n)
2
where x(n) denotes the input sequence and y(n) denotes the output sequence. Prove or give counterexamples:
(a) Is the system linear?
(b) Is the system time-invariant?
(c) Is the system causal?
(d) Is the system BIBO stable?
Problem 4.34 A relaxed system is described by the equation
y(n)
1
y(n 2) = x(n2 )
2
where x(n) denotes the input sequence and y(n) denotes the output sequence. Prove or give counterexamples:
(a) Is the system linear?
PROBLEMS
104
CHAPTER 4
DISCRETETIME
SYSTEMS
(b) Is the system time-invariant?

(c) Is the system causal?
(d) Is the system BIBO stable?
Problem 4.35 If the response of an LTI system to x(n) = ej 3 n is y(n) =

the response of the system to
1 j(
e 3 n+ 6 ) ,
2
what is

1

?
cos
(n 1) +
2
3
4
x(n) =
Problem 4.36 Refer to Fig. 4.16, which shows two LTI systems S1 and S2 . The response of S1 to
an input sequence x1 (n) is
n
o
y1 (n) =
1 , 3/2, 2, 2
where the boxed entry occurs at the origin of times, and the samples to its right occur at time instants
n = 1, 2,
n
o 3. All other entries are zero. The response of S2 to the input sequence x2 (n) =
1 , 1/2, 1
is y2 (n) =

1 n1
3
u(n 2). The systems are connected in cascade where the
output of S1 is the input to S2 . What would the response y(n) of the cascade be when x(n) =
x1 (n) 2x1 (n 1) is applied to S1 ?
x1 (n)
S1

1 , 1/2, 1
S2
x(n)
S1

1 , 3/2, 2, 2
1 n1
3
S2
u(n 2)
y(n)
FIGURE 4.16 The top row shows the response of system S1 to x1 (n). The middle row shows the
given sequences at the input and output of system S2 . The bottom row shows the series connection
of both systems for Prob. 4.36
Problem 4.37 Draw block diagram and signal flowgraph representations for the classes of systems
(a) y(n) = 12 y(n 2) + y(n 1) + 13 x(n 2).
(b) y(n 1) = 2y(n) + 51 x(n 3).
Problem 4.38 Draw block diagram and signal flowgraph representations for the classes of systems
(a) y(n) = 41 y(n 3) 12 y(n 1) + x(n) + 41 x(n 3).
(b) y(n 2) = 21 y(n) + y(n 1) 51 x(n 2).
Problem 4.39 The response of an LTI system to an even sequence x(n) is 0.5n u(n). Can you
determine the response of the system to x(n + 1)?
Problem 4.40 The response of an LTI system to an odd sequence x(n) is 0.5n u(n). Can you
determine the response of the system to x(n 2)?
CHAPTER
Impulse Response Sequence
T
he impulse response sequence plays a fundamental role in characterizing the behavior
of linear time-invariant (LTI) systems, so much so that knowledge of the impulse response
sequence alone is sufficient in order to fully describe the behavior of an LTI system. For
example, explicit knowledge of a mathematical model describing the input-output relation
of the LTI system is not even needed; such a model can be inferred from the impulse
response sequence itself. Moreover, for LTI systems, knowledge of the impulse response
sequence enables us to determine whether the system is BIBO stable or not and whether it
is causal or not. Knowledge of the impulse response sequence also enables us to determine
the response of the system to any input sequence, x(n). In later chapters, we shall see
that the impulse response sequence also conveys information about the properties of LTI
systems in the frequency domain.
This chapter expands on these remarks and emphasizes the significance of the impulse
response sequence in the context of LTI systems.
5.1 CONVOLUTION SUM FOR LTI SYSTEMS

Consider a system that is described generically by the input-output relation
y(n) = S[x(n)]
where x(n) denotes the input sequence and y(n) denotes the output sequence. The symbol
S denotes the transformation that the system performs on the input sequence in order to
generate the output sequence. Initially, we will not restrict the system to be linear or even
LTI.
The impulse response sequence of S is defined as the response of the system to the unitsample sequence, x(n) = (n). We denote the impulse response sequence by the special
symbol h(n) so that
h(n) = S[(n)]
(5.1)
Example 5.1 (Response to the unit-sample sequence)

Consider the system described by the input-output relation
y(n) = nx(n 1)
The impulse response sequence is obtained by setting x(n) = (n) so that
h(n) = n(n 1)
105
Copyright
106
CHAPTER 5
IMPULSE
RESPONSE
SEQUENCE
Observe that, in this case, since the unit-sample sequence, (n), assumes the value 1 at n = 0 and is
zero elsewhere, we have that (n 1) will assume the value 1 at n = 1 and will be zero for all other
values of n. It follows that we can simplify the above expression for h(n) and write instead
h(n) = (n 1)
Example 5.2 (Identical impulse response sequences)

Consider now the system described by the input-output relation
y(n) = x(n 1)
(system I)
Its impulse response sequence is easily seen to be

h(n) = (n 1)
which is the same result we obtained in Example 5.1 for the system
y(n) = nx(n 1)
(system II)
However, both systems are different. System I is linear and time-invariant (LTI) while system II is
linear but time-variant. Therefore, given knowledge of h(n) alone, it is not possible to conclude
which system gave rise to it. Such ambiguous situations do not arise when we focus exclusively on
LTI systems. As the discussion will show, no two LTI systems will share the same impulse response
sequence. In other words, for LTI systems, the impulse response sequence serves as a defining
property that uniquely identifies the system.
Example 5.3 (Impulse response sequence of an LTI system)

Consider a relaxed system that is described by the input-output relation
y(n) =
1
y(n 1) + x(n)
2
We already know from the discussion in Sec. 4.9 that relaxed systems that are described by such
constant-coefficient difference equations are LTI. We would like to determine the impulse response
sequence of the given system. Thus, let x(n) = (n) and let us denote the corresponding output
sequence by h(n). It follows from the input-output relation that h(n) satisfies
h(n) =
1
h(n 1) + (n)
2
Now since the input sequence (n) is zero for all n < 0 and the system is relaxed, we conclude from
the definition of a relaxed system that h(n) = 0 for n < 0 as well. To determine the values of h(n)
for n > 0, we iterate the recursion to get
h(0)
(0) = 1
h(1)
1/2
h(2)
1/4
h(3)
..
.
1/8
..
.
107
We identify a pattern in the successive values of {h(n)} and conclude that

n
h(n) =
1
2
SECTION 5.1
CONVOLUTION
SUM FOR
LTI SYSTEMS
u(n)
where we added the unit-step sequence u(n) to enforce the fact that h(n) = 0 for n < 0.
LTI Systems
From now on, we focus on the study of linear time-invariant (LTI) systems. In particular,
we establish several useful results concerning the impulse response sequence of such systems. To begin with, we argue that the response of an LTI system to any input sequence
can be determined solely from knowledge of the impulse response sequence of the system.
In other words, a mathematical input-output relation for the system is not needed.
Thus, consider an LTI system S and let h(n) denote its impulse response sequence. Due
to the assumed time-invariance of the system, its response to (n k), for any k, will be
h(n k). That is, if the input sequence is a time-shifted unit-sample sequence then the
result will be an equally time-shifted impulse response sequence:
h(n) = S[ (n) ] = h(n k) = S[ (n k) ]
(by time-invariance)
(5.2)
Now, any arbitrary input sequence x(n) can be expressed as a linear combination of shifted
unit-sample sequences, namely, for any x(n) we can write
x(n) = . . . + x(1)(n + 1) + x(0)(n) + x(1)(n 1) + . . .
(5.3)
Specifically, note that the unit-sample (or impulse) sequence (n), which is located at the
origin n = 0, has amplitude x(0). Likewise, the unit-sample sequence (n 1), which is
located at time n = 1, has amplitude x(1), and so forth see Fig. 5.1.
x(n)
x(3)(n 3)
x(1)(n 1)
x(3)(n + 3)
x(0)(n)
2
3
x(2)(n + 2)
x(2)(n 2)
FIGURE 5.1 An arbitrary sequence x(n) can be expressed as a linear combination of shifted
unit-sample sequences.
108
More compactly we write
CHAPTER 5
IMPULSE
RESPONSE
SEQUENCE
x(n) =
x(k)(n k)
(5.4)
k=
This is a representation for x(n) as a linear combination of the sequences {(n k)} for
all k. The coefficients of the linear combination are the samples of the sequence itself.
Proceeding with the argument, we invoke the superposition principle for linear systems
and note that the response of the LTI system S to an arbitrary input sequence x(n) is given
by
y(n) =
=
S[ x(n) ]
"
#
X
S
x(k)(n k)
(using (5.4))
k=
k=
x(k) S[ (n k) ]
(by superposition principle)
where in the second equality we replaced x(n) by its alternative representation (5.4), and in
the third equality we invoked the superposition principle to move the system operation S[]
inside the summation. Using (5.2)), and invoking the definition of the impulse response
sequence, we get
y(n) =
k=
x(k)h(n k)
(5.5)
The series
k=
x(k)h(n k)]
that appears in (5.5) is called the convolution sum of x(n) and h(n); it is usually denoted
more compactly as
x(n) h(n)
so that
x(n) h(n) =
k=
x(k)h(n k)
(5.6)
We therefore find that the response of the LTI system S to any input sequence x(n) can
be determined by evaluating the convolution sum of this input sequence with the impulse
response sequence of the system, i.e.,
y(n) = x(n) h(n)
(5.7)
As we are going to see in the next chapter, the convolution sum can be evaluated in many
ways, some are easier than others depending on the context. Here, we continue to describe
several useful properties of the impulse response sequence of LTI systems.
109
SECTION 5.1
Example 5.4 (Convolution of two sequences)

Consider the sequences h(n) = (0.5)n u(n) and x(n) = u(n 1). Then, by definition, their
convolution is given by
y(n)
k=
k=
x(k)h(n k)
u(k 1) [0.5nk u(n k)]
We evaluate the summation as follows. First, we note that the step-sequence u(k 1) is zero for
k < 1, while the step-sequence u(n k) is zero for k > n. This means that the product that appears
inside the summation will be zero for all values of k in the interval k < 1 and k > n. Therefore, the
limits of the summation should go from k = 1 to k = n and the expression for y(n) reduces to
y(n)
n
X
0.5nk
k=1
Clearly, the summation will include nontrivial elements only for values of n that are larger than or
equal to one (which is the lower limit for the summation index). For n < 1, we get y(n) = 0, while
for n 1 we have
y(n)
=
=
1 + 0.5 + 0.52 + . . . + 0.5n1

1 0.5n
1
= 2 n1
1 0.5
2
In summary, we find that

y(n) = 2
u(n 1),
2n1
for all n
where we added the step-sequence u(n 1) to enforce the condition y(n) = 0 for n < 1.
Example 5.5 (Response to the unit-step sequence)

Consider a relaxed system that is described by the input-output relation
y(n) =
1
y(n 1) + x(n)
2
and let us determine its response to the input sequence x(n) = u(n 1). We already know from
Example 5.1 that this system is LTI and its impulse response sequence is given by h(n) = 0.5n u(n).
Therefore, the response of the system to the input sequence x(n) = u(n 1) is given by the
convolution sum
y(n)
u(n 1) (0.5)n u(n)
which we already evaluated in Example 5.4, so that the desired output sequence is

y(n) = 2
1
2n1
u(n 1),
for all n
Alternatively, we may proceed from first principles as follows. Since the system is relaxed, and
since the input sequence u(n 1) is zero for n < 1, then y(n) = 0 for n < 1. For n 1, we iterate
CONVOLUTION
SUM FOR
LTI SYSTEMS
110
CHAPTER 5
the recursion using x(n) = u(n 1) to get

y(1)
IMPULSE
RESPONSE
SEQUENCE
y(2)
y(3)
y(4)
..
.
x(1) = 1
1/2 + 1 = 3/2 = 2 1/2
3/4 + 1 = 7/4 = 2 1/4
7/8 + 1 = 15/8 = 2 1/8

..
.
A pattern can be recognized in these sample values and we are led to the same expression

y(n) = 2
1
2n1
u(n 1),
for all n
5.2 CAUSALITY OF LTI SYSTEMS

The impulse response sequence can be used to determine whether an LTI system is causal
or not without the need to know a mathematical model describing the input-output relation.
Before explaining how this step can be achieved, we first define what we mean by a causal
sequence (as opposed to a causal system).
We say that a sequence x(n) is causal if its samples are zero for n < 0, i.e.,
x(n) = 0
for
n<0
(causal sequence)
(5.8)
Note that this is the definition of a causal sequence as opposed to a causal system, which
was defined earlier in Sec. 4.6. Recall that a causal system is one for which the output at
time n depends on present and past values of the input sequence (it does not depend on
future values of the input sequence).
With these definitions, it is straightforward to verify that the causality of an LTI system
is equivalent to the causality of its impulse response sequence, namely,
LTI system is causal h(n) = 0 for n < 0
(5.9)
Proof: Assume first that the LTI system is causal. By linearity, its response to the zero sequence
has to be the zero sequence (recall the absence of excitation property for linear systems as was given
by (4.8)). Now, by definition, the response of the system to (n) is h(n). Moreover, (n) and the
zero sequence are identical for n < 0. It follows from the assumed causality of the system (recall
property (4.5)), that the corresponding output sequences must agree for n < 0. This shows that we
must have h(n) = 0 for n < 0.
Conversely, assume h(n) = 0 for n < 0. Then, in view of the convolution sum (5.7), the output
y(n) in response to any input x(n) is given by
y(n)
X
k=
X
j=
X
j=0
x(k)h(n k)
h(j)x(n j)
h(j)x(n j)
(using j = n k)
where we employed in the second step a change of variables (j = n k), and used in the last step
the fact that h(j) = 0 for j < 0. The last expression shows that y(n) depends only on past and
present values of the input sequence. Hence, the system is causal.
Example 5.6 (Causal and noncausal systems)

n
If
1
u(n)
2
is the impulse response sequence of an LTI system, then this system must be causal. Accordingly,
the relaxed system that is described by the input-output relation
h(n) =
y(n) =
1
y(n 1) + x(n)
2
is causal because, as we already know, its impulse response is given by the above h(n).
On the other hand, the sequence h(n) = u(n + 2) cannot be the impulse response of a causal
LTI system. This is because u(n + 2) is not a causal sequence; it is nonzero for n < 0.
Limits of Summation
For a causal LTI system, the convolution sum can be written in either of two forms:
y(n) =
n
P
k=
x(k)h(n k) =
k=0
h(k)x(n k)
(5.10)
That is, the limits of the summation can be restricted to the intervals (, n] or [0, ).
This is because for causal LTI systems, h(n) = 0 for n < 0. If, in addition, the input
sequence itself is causal, then the above relations simplify to
y(n) =
n
P
k=0
h(k)x(n k) =
n
P
k=0
x(k)h(n k)
(5.11)
since, for a causal input sequence, x(n) = 0 for n < 0.
5.3 BIBO STABILITY

The BIBO stability of an LTI system can also be inferred from its impulse response. Recall
that a system is BIBO if, and only if, its output sequence remains bounded for any bounded
input sequence. It holds that for LTI systems, the following is an equivalent characterization of BIBO stability in terms of the impulse response sequence:
LTI system is BIBO stable
k=
|h(k)| <
(5.12)
In other words, an LTI system is BIBO stable if, and only if, its impulse response sequence
is absolutely summable.
Proof: Let x(n) be any bounded sequence, say
|x(n)| Bx < for all n
111
SECTION 5.3
BIBO STABILITY
112
CHAPTER 5
Let y(n) denote the corresponding output sequence and assume initially that the impulse response
sequence is absolutely summable, say
IMPULSE
RESPONSE
SEQUENCE
X
k
|h(k)| < K <
for some positive scalar K. Then, according to the convolution sum, the output sequence of the LTI
system is given by
y(n) =
k=
which shows that

|y(n)|
x(k)h(n k)
X

x(k)h(n k)

k=
X
k=
|x(k)h(n k)|
Bx
<
Bx K
k=
|h(n k)| ,
since x(n) is bounded
We therefore conclude that y(n) is also bounded so that the LTI system is BIBO stable.
Conversely, assume the LTI system is BIBO stable and let us prove that its impulse response
sequence must be absolutely summable. Assume not. That is, assume the system is BIBO stable but
X
k=
|h(k)| =
We now verify that this assumption leads to a contradiction. Indeed, define the obviously bounded
input sequence
(
h (n)
if h(n) 6= 0
|h(n)|
x(n) =
0
if h(n) = 0
where h (n) denotes the complex conjugate value of h(n). The corresponding output sample at
time 0 is given by
y(0) =
x(k)h(k) =
k=
k=
|h(k)|
which in view of the assumption on h(n) is unbounded. We therefore have an example of a bounded
input sequence that leads to an unbounded output sequence. This fact contradicts the assumed BIBO
stability of the system. We conclude that the assumption on h(n) is false and we must necessarily
have
k=
|h(k)| <
Example 5.7 (Stable and unstable systems)
n
h(n) =
1
2
u(n)
This sequence can be the impulse response sequence of a BIBO LTI system since it is absolutely
summable. Indeed, note that
X
n=
|h(n)| =
X
1
n=0
2n
1
=
1
1
2
SERIES
CASCADE OF
LTI SYSTEMS
= 2<
On the other hand, the sequence h(n) = u(n) cannot be the impulse response sequence of a BIBO
stable LTI system because it is not absolutely summable.
5.4 SERIES CASCADE OF LTI SYSTEMS

We now illustrate what happens when we combine LTI systems together. We consider the
cases of series and parallel cascades of such systems.
Thus, consider two LTI systems S1 and S2 , with impulse response sequences h1 (n) and
h2 (n), respectively. Both systems are connected in series, with the output of S1 connected
to the input of S2 , as shown in the top part of Fig. 5.2.
S
x(n)
(n)
S1
S1
S2
h1 (n)
S2
113
SECTION 5.4
y(n)
h1 (n) h2 (n)
FIGURE 5.2 Series cascade of two systems. The input to the cascade is x(n) and the output of
the cascade is y(n).
The combined system, say S, whose input is x(n) and output is y(n), is also LTI. Moreover, the impulse response sequence of the combination is denoted by h(n) and is given
by
h(n) = h1 (n) h2 (n)
(5.13)
Proof: Let us first determine the impulse response sequence of S. For this purpose, take x(n) =
(n). Then the output of S1 is, by definition, h1 (n), as illustrated in the bottom plot of Fig. 5.2. The
sequence h1 (n) is now an input to S2 and the output y(n) will therefore be y(n) = h1 (n) h2 (n).
This shows that the impulse response sequence of the combined system is h(n) = h1 (n) h2 (n).
To establish linearity of the combined system, let x1 (n) and w1 (n) be any two input sequences
to S1 and denote by y1 (n) and z1 (n) the corresponding outputs of S1 . Let also y2 (n) and z2 (n)
denote the outputs of S2 when excited by y1 (n) and z1 (n), respectively. By linearity of S1 , we
get that ax1 (n) + bw1 (n) outputs ay1 (n) + bz1 (n), which now excites S2 . By linearity of S2 , we
get that ay1 (n) + bz1 (n) outputs ay2 (n) + bz2 (n). It then follows that the cascade satisfies the
superposition principle and it is therefore a linear system.
114
CHAPTER 5
IMPULSE
RESPONSE
SEQUENCE
To establish time-invariance of the combined system, we first note from the time-invariance of
S1 that the time-shifted sequence x1 (n k) outputs y1 (n k). This sequence now excites S2 . By
time-invariance of S2 , we get that y1 (n k) outputs y2 (n k). It is then clear that the cascade is a
time-invariant system.
5.5 PARALLEL CASCADE OF LTI SYSTEMS

Assume now that the systems S1 and S2 are connected in parallel, having the same input
sequence x(n) while their output sequences are added together to yield y(n), as illustrated
in the top part of Fig. 5.3. The combined system, say S, whose input is x(n) and output is
y(n) is also LTI. Moreover, its impulse response sequence h(n) is given by
h(n) = h1 (n) + h2 (n)
(5.14)
Proof: The proof is similar to the case of a series cascade and is left as an exercise see the bottom
part of Fig. 5.3.
S
x(n)
S1
y(n)
+
S2
(n)
S1
h1 (n)
h1 (n) + h2 (n)
S2
h2 (n)
FIGURE 5.3 Parallel connection of two systems. The input to both systems is x(n) and the output
of the cascade is y(n).
115
5.6 FIR AND IIR SYSTEMS
SECTION 5.6
There is a useful characterization of systems in terms of the length of their impulse response sequences. Systems can be classified into finite-impulse response (FIR) systems or
infinite-impulse response (IIR) systems. In the FIR case, the impulse response sequence
has a finite duration (and, therefore, the number of nonzero samples is finite). In the IIR
case, the impulse response sequence has infinite duration.
Example 5.8 (FIR and IIR)
Consider, for example, the systems
{y(n) = x(n) + x(n 1)}
and
{y(n) = y(n 1) + x(n), relaxed}
Both systems are linear and time-invariant. The impulse response sequence of either system can be
determined by inspection. For the first system we have
h(n) = (n) + (n 1)
(first system)
which has finite duration. Therefore, the first system corresponds to an FIR system. On the other
hand, the impulse response sequence for the second system is the step sequence
h(n) = u(n)
(second system)
which has infinite duration. Therefore, the second system corresponds to an IIR system.
Moving-Average (MA) Systems

An FIR LTI system is also called a Moving-Average (MA) system for the following reason.
Let h(n) denote the impulse-response sequence of an FIR LTI system. Given knowledge
of h(n) alone, we can determine a difference equation that describes the input-output mapping of the system. Consider, for example, the case
h(n) = 2(n) (n 1) + 3(n 2)
(5.15)
This is a sequence of duration 3. The response y(n) of the corresponding LTI system for
any input sequence x(n) is given by
y(n) =
=
k=
k=
h(k)x(n k)
[2(k) (k 1) + 3(k 2)] x(n k)
= 2x(n) x(n 1) + 3x(n 2)
(5.16)
This result provides a mathematical description for the input-output relation of the FIR LTI
system whose impulse response sequence is given by (5.15). The relation shows that the
output at time n is obtained by linearly combining the present input sample, x(n), and two
previous input samples, {x(n 1), x(n 2)}. Such linear combinations of a finite number
of present and past input samples are called moving-average (MA) representations. We
thus say that an FIR LTI system admits a MA representation; the coefficients of the linear
combination are the samples of the impulse-response sequence. A moving-average or FIR
FIR AND
IIR SYSTEMS
116
CHAPTER 5
IMPULSE
RESPONSE
SEQUENCE
system is also called a tapped-delay-line system. This is because, as shown by the example (5.16), FIR systems can be implemented by using a finite number of delays and linear
combiners (a collection of multipliers by constants and adders) see Fig. 5.4.
x(n)
z 1
x(n 1)
z 1
x(n 2)
3
1
+
y(n)
FIGURE 5.4 A tapped-delay-line implementation of the FIR system (5.16).
Auto-Regressive Moving-Average (ARMA) Systems

The above example shows that for FIR LTI systems, knowledge of the impulse-response
sequence can be used to determine a mathematical input-output relation for the system.
The same argument, however, cannot be used for IIR LTI systems since y(n) will then
be expressed in terms of an infinite number of present and past input samples (because
h(n) has infinite duration in this case). Nevertheless, we shall see later in the book (see
Sec. 11.4) that by using transform techniques, it will be possible to determine compact
mathematical input-output relations for IIR LTI systems as well from knowledge of their
impulse-response sequences.
Since the output of an IIR LTI system at time n is a function of an infinite number of
present and past input samples, it appears then that the implementation of such systems
would require an infinite number of delays. It turns out that for the case of IIR LTI systems
that are described by constant-coefficient difference equations, an implementation with a
finite number of delays is possible by using the block diagram technique of Sec. 4.10.
Specifically, such systems are described by difference equations of the form
y(n) =
M
X
k=1
ak y(n k) +
N
X
k=0
bk x(n k)
(5.17)
for some coefficients {ak , bk }. It is seen that the current output sample is not only a
function of current and past input samples (as in the FIR case) but also of past output
samples (compare with (5.16)). Such representations are called auto-regressive movingaverage (ARMA) representations. If y(n) were only dependent on past output samples
and the present input sample, say,
y(n) =
M
X
k=1
ak y(n k) + b0 x(n)
(5.18)
then the representation is said to be of the auto-regressive (AR) type. Such recursive or
ARMA systems can be implemented with a finite number of delays by employing delayed
versions of the output sequence.
117
SECTION 5.7
Example 5.9 (Implementation of an IIR system)
INVERSE
PROBLEM
Consider the difference equation

y(n) = y(n 1) + x(n 1) 2x(n)
which was studied earlier in Example 4.15. In that example, we explained how to arrive at the two
different implementations shown in Fig. 5.5. One implementation employs two delays, while the
other implementation employs one delay. An implementation with the smallest number of delays is
called minimal.
x(n)
y(n)
z 1
z 1
x(n)
z 1
y(n)
FIGURE 5.5 Two block diagram representations for the class of ARMA systems {y(n) = y(n
1) + x(n 1) 2x(n)}.
5.7 INVERSE PROBLEM

We have seen that the impulse-response sequence of an LTI system is sufficient to fully
characterize the response of the system to any input sequence via the convolution sum.
What about the converse statement? Given any nonzero input-output pair {x(n), y(n)},
can we recover the impulse response sequence of the system? The answer is positive. We
shall describe several different ways to solve this problem in this book. Among them we
list the following:
1. Using transform techniques. This method will be described in subsequent
chapters (see, for example, Sec. 11.4).
2. Invoking linearity and time-invariance. This method is useful in some special
cases. For example, assume that we know that the step response, i.e., the response to
x(n) = u(n)
118
of an LTI system is
CHAPTER 5
IMPULSE
RESPONSE
SEQUENCE
y(n) = (0.5)n u(n 1)

Can we determine its impulse response? To do so, we proceed as follows. Note that
the unit-sample sequence is related to the step sequence via the relation:
(n) = u(n) u(n 1)
This means that we can regard (n) as a linear combination of u(n) and a shifted
version of u(n). Now, by linearity and time-invariance, the response of the system to
(n) should be the same linear combination of the corresponding output sequences,
so that
h(n) = y(n) y(n 1)
and we arrive at
h(n) = (0.5)n u(n 1) (0.5)n1 u(n 2)
3. Solving a triangular linear system of equations. We start from the convolution sum
X
y(n) =
x(k)h(n k)
k=
and observe that it leads to a linear equation between the observations {y(n)} and
the input samples {x(n)}. To illustrate this fact, let us consider the case of a causal
LTI system for which h(n) = 0 for n < 0 and, hence,
y(n) =
n
X
k=
x(k)h(n k)
We also assume that the input is a causal sequence, i.e., x(n) = 0 for n < 0, so that
y(n) =
n
X
k=0
x(k)h(n k) =
n
X
k=0
h(k)x(n k)
It follows that the output is a causal sequence as well (y(n) = 0 for n < 0). Expanding the convolution sum for each value of n, we find that the samples of the
sequences {y(), x(), h()} are related via the triangular system of equations:
y(0)
y(1)
y(2)
y(3)
..
.
x(0)
x(1)
x(2)
x(3)
..
.
x(0)
x(1) x(0)
x(2) x(1)
x(0)
..
h(0)
h(1)
h(2)
h(3)
..
.
The entries {h(n)} can now be determined by solving this linear system of equations
by back-substitution. Assume, for instance, that x(0) is nonzero, then
1
y(0)
x(0)
1
h(1) =
[y(1) x(1)h(0)]
x(0)
1
h(2) =
[y(2) x(2)h(0) x(1)h(1)]
x(0)
..
.
. = ..
h(0) =
Note the following facts:

(a) If x(0) = 0 then we must have y(0) = 0. This is because we are dealing with
a causal linear system and, hence, the output remains zero as long as there is
no excitation.
(b) Accordingly, when x(0) = 0 we cannot recover h(0) from the first equation.
We would recover it from the other equations.
5.8 APPLICATIONS
In this section, we illustrate applications of some of the concepts covered in the chapter in
the context of some practical problems.
5.8.1 Multipath Channels

In the wireless communications application of Sec. 4.11 we provided one example of a
multipath channel, which was described by the difference equation
y(n) =
1
1
5
x(n 2) + x(n 3) + x(n 5)
6
4
8
(5.19)
The equation describes a causal LTI system. To determine its impulse response sequence,
we simply set x(n) = (n) and y(n) = h(n) and use the difference equation to conclude
that
1
1
5
(5.20)
h(n) = (n 2) + (n 3) + (n 5)
6
4
8
In other words, in this case, the impulse response sequence consists of three unit-sample
sequences: one is located at n = 2 and has amplitude 5/6, another is located at n = 3 and
has amplitude 1/4, and a third is located at n = 5 and has amplitude 1/8. We say that the
channel exhibits three multipath components.
Assume now that the signal that is transmitted over the multipath channel has the form
x(n) = (0.5)n u(n). We can use the above expression for h(n) to determine what the
corresponding received sequence will be. Using the fact that the channel is LTI we can
119
SECTION 5.8
APPLICATIONS
120
write
CHAPTER 5
y(n) =
=
=
x(n) h(n)
n

1
1
1
5
u(n) (n 2) + (n 3) + (n 5)
2
6
4
8
n2
n3
n5
1
1
1
1
1
5
u(n 2) +
u(n 3) +
u(n 5)
6
2
4
2
8
2
where we are using the following useful property about convolving sequences with the
unit-sample sequence:
z(n) (n no ) = z(n no )
(5.21)
for any sequence z(n). That is, convolving a sequence z(n) with a unit-sample sequence
that is located at n = no simply shifts z(n) by no . This property can be easily established
from the definition of the convolution sum:
z(n) (n no ) =
=
k=
z(k)(n no k)
z(n no )
Returning to the above expression for y(n) we can group terms as follows:
y(n) =
5
2
1
28
(n 2) + (n 3) + (n 4) +
6
3
3
3
n
1
u(n 5)
2
Impulse response sequence, h(n)

1
h(n)
0.8
0.6
0.4
0.2
0
10
n
Output sequence, y(n)
15
20
10
n
15
20
1
0.8
y(n)
IMPULSE
RESPONSE
SEQUENCE
0.6
0.4
0.2
0
FIGURE 5.6 Plot of the impulse response sequence (top) and the output sequence (bottom) in
response to x(n) = (0.5)n u(n) over the first 20 samples.
Figure 5.6 plots the first 20 samples of h(n) and the output sequence y(n) computed
above in response to the input sequence, x(n) = (0.5)n u(n). It is not uncommon to
illustrate the plots of discrete-time sequences by connecting the dots of the stem plot with
a continuous line, as we illustrate in Fig. 5.7. This second type of illustration is useful
for better visualization of the behavior of the sequence, especially when the number of
samples is large. We shall employ both types of plots in our presentation.
Impulse response sequence, h(n)
1
h(n)
0.8
0.6
0.4
0.2
0
10
15
n
Output response sequence, y(n)
20
y(n)
0.8
0.6
0.4
0.2
0
10
n
15
20
FIGURE 5.7 An alternative plot representation of discrete-time sequences. The figure plots the
same impulse response sequence (top) and output sequence (bottom) that are shown in Fig. 5.6,
except that now the stems are connected by a continuous line running through the ends of the sample
values.
Practice Questions:
1. Plot the impulse response sequence of the multipath channel.
2. What are the energy and average power of the impulse response sequence?
3. Determine the response of the channel to the input sequence x(n) = u(n) by convolving
x(n) and h(n). Plot the samples of y(n) over the interval 0 n 8.
4. Is the multipath channel BIBO stable?
5. Is the multipath channel MA, AR, or ARMA? Is the channel FIR or IIR?

In Sec. 4.11 we studied the problem of a financial account that grows at the annual return
rate of % starting from an initial deposit of D dollars. We considered a general scenario
where, in addition to the initial deposit, the client may consider making additional deposits
(or withdrawals) at other time instants. In this case, the evolution of the funds in the account
are governed by recursion (4.14), namely,

y(n 1) + x(n), y(1) = 0
(5.22)
y(n) = 1 +
100
where x(0) = D corresponds to the initial deposit. This is a constant-coefficient difference equation and it represents a causal LTI system with input sequence x(n) and output
sequence y(n).
121
SECTION 5.8
APPLICATIONS
122
CHAPTER 5
IMPULSE
RESPONSE
SEQUENCE
Let us determine the impulse response sequence of system (5.22). To do so, we set
x(n) = (n) and y(n) = h(n) and use the difference equation to write

h(n 1) + (n), h(1) = 0
(5.23)
h(n) = 1 +
100
By iterating the recursion, we are able in this case to find a closed-form expression for h(n).
First note from the recursion that h(n) = 0 for all n < 0; this result is consistent with the
fact that we are dealing with a causal LTI system, and the impulse response sequences of
such systems must be causal sequences. Moreover, for n 0, we obtain from recursion
(5.23) the following sequence of results:
h(0) =
h(1) =
h(2) =
h(3) =
..
.
so that
1

1+

1+

1+
..
.

h(0) = 1 +
100
100

2

h(1) = 1 +
100
100

3
h(2) = 1 +
100
100

n
h(n) = 1 +
u(n)
100
(5.24)
Let us use this result to examine the case when the client makes regular annual deposits
of d dollars from year one onwards. That is, we now wish to determine the response of the
LTI system (5.22) to the input sequence
x(n) = D(n) + d u(n 1)
We solved this problem earlier in Sec. 4.11 by iterating (5.22). Here, we address the
same problem by appealing instead to the convolution result (5.7), i.e., by determining the
convolution sum of x(n) with h(n). First, let

= 1+
(5.25)
100
Now since the system is causal and the input sequence, x(n), is a causal sequence, we can
use (5.11) to write
y(n)
=
=
n
X
k=0
n
X
h(k)x(n k)
k u(k) [D (n k) + d u(n k 1)]
k=0
n
X
k=0
n
D (n k)
n1
X
k=0

n1
= D + 1 + + 2 + . . . +
Using the result of Example 2.11 for the sum of the first n terms of a geometric series, we
get
1 n
d
y(n) = n D +
1
Substituting by its value (5.25) in terms of , we arrive at the same result (4.16), namely,
123
SECTION 5.9
h
i
in
100 h
n
y(n) = 1 +
D +
1 d, n 0
1+
100
100
PROBLEMS
(5.26)
Figure 5.8 illustrates the evolution of the funds in the account assuming an initial deposit
of US$100, an annual return rate of 5%, and regular annual deposits of US$8.
Return rate =5%, initial deposit = US$100, regular annual deposit = US$8
400
350
y(n)
300
250
200
150
100
10
n (year)
15
20
FIGURE 5.8 Evolution of the amount of funds in the account assuming an initial deposit of
US$100, an annual return rate of 5%, and regular annual deposits of US$8.
Practice Questions:
1. Start with an initial deposit of D dollars at year n = 0 and make regular annual deposits of d
dollars every other year, starting from year one. Assuming an annual return rate of %, use
the convolution sum method to determine an expression for the total funds, y(n), at year n.
of US$20 every other year starting from year one, how much funds will be available at the
start of year n = 30 assuming an annual return rate of 2%.
3. Start with an initial deposit of D dollars at year n = 0 and assume an annual return rate of
%. Assume further that starting from year one onwards, the client makes regular deposits of
d dollars during odd years and regular withdrawals of w dollars during even years. Use the
convolution sum method to determine an expression for the total funds, y(n), at year n.
4. Starting with an initial deposit of US$500 at year n = 0, and making regular deposits of
US$20 every odd year and regular withdrawals of US$10 every even year, how much funds
will be available at the start of year n = 30 assuming an annual return rate of 2%.
5.9 PROBLEMS
Problem 5.1 Find the impulse response sequences of the following systems:
(a) The relaxed system y(n) = 31 y(n 1) + x(n + 1).

(b) The system y(n) = 31 y(n 1) + x(n + 1), y(1) = 1 .
124
CHAPTER 5
IMPULSE
RESPONSE
SEQUENCE
(c) The relaxed system y(n) = 13 y(n 1) + x2 (n + 1).
Which system is LTI?
Problem 5.2 Find the impulse response sequences of the following systems:
(a) The relaxed system y(n) = 41 y(n 1) + x(n 1).

(b) The system y(n) = 41 y(n 1) + x(n 1), y(1) = 1 .

(c) The relaxed system y(n) = 41 y(n 1) + x2 (n 1).
Which system is LTI?

Problem 5.3 Answer True or False.
(a) The series cascade of two causal LTI systems is causal.
(b) The series cascade of two stable LTI systems is stable.
(c) The parallel cascade of two causal LTI systems is causal.
(d) The parallel cascade of two stable LTI systems is stable.
(a) The series cascade of two FIR LTI systems is FIR.
(b) The parallel cascade of two IIR LTI system is IIR.
(c) An FIR LTI system is always BIBO stable.
(d) An IIR LTI system is always causal.
(a) If the input sequence to an LTI system is an even sequence, then the corresponding output
sequence is also even.
(b) If the input sequence to an LTI system is an odd sequence, then the corresponding output
sequence is also odd.
(c) If the output sequence of an LTI system is an odd sequence, then the corresponding input
sequence is odd.
(a) If both the input sequence to an LTI system and its impulse response sequence are odd sequences, then the output sequence is also odd.
(b) If both the input sequence to an LTI system and its impulse response sequence are even sequences, then the output sequence is also even.
(c) If the input sequence to an LTI system is odd, and its impulse response sequence is even, then
then the output sequence is odd.
Problem 5.7 Classify each of the following LTI systems as MA, AR, or ARMA.
(a) y(n) = 12 x(n) + 13 x(n 1).
(b) The relaxed system y(n) = 21 y(n 2) + x(n 1).

(c) The relaxed system y(n) = 12 y(n 2) + x(n).
(d) h(n) =

1 n
2
u(n).
Problem 5.8 Classify each of the following LTI systems as MA, AR, or ARMA.
(a) The relaxed system y(n) = 12 y(n 1) + 31 y(n 3) + x(n) 12 x(n 1).
(b) The relaxed system y(n) = 21 y(n 3) + x(n 2).

(c) h(n) = (n) + 12 (n 1).
(d) h(n) =

1 n1
2
u(n 2).
Problem 5.9 Which of the following LTI systems are FIR or IIR?
(a) h(n) = u(n) u(n 10).
(b) h(n) =

1 n
2
PROBLEMS
u(n 3).
(c) The relaxed system y(n) = 21 y(n 1) + x(n).
(d) The relaxed system y(n) = x(n) + 12 [x(n 1) + x(n 2)].

(e) The series cascade of systems (a) and (b).
(f) The series cascade of systems (c) and (d).
(g) The parallel cascade of systems (b) and (c).
Problem 5.10 Which of the following LTI systems are FIR or IIR?
(a) h(n) = u(n) u(n + 5).
(b) h(n) =

1 n+3
3
u(n + 1).
(c) The relaxed system y(n) = 12 y(n 3) + x(n 1).
(d) The relaxed system y(n) = x(n) x(n 3).

(e) The series cascade of systems (a) and (b).
(f) The series cascade of systems (c) and (d).
(g) The parallel cascade of systems (b) and (c).
Problem 5.11 Which of the following LTI systems are causal?

(a) h(n) =
(b) h(n) =

1 n3
u(n
2

1 n
u(n +
3
1).
1) +

1 n
2
u(n + 1).
Problem 5.12 Which of the following LTI systems are causal?

(a) h(n) =
(b) h(n) =

1 n
u(n + 2).
2

1 n
u(n
+ 1).
3
Problem 5.13 Consider an LTI system with impulse response sequence:

h(n) = 2(n) (n 1) + (n 2)
(a) Implement it as the parallel cascade of two LTI systems.
(b) Implement it as the series cascade of two LTI systems.
Problem 5.14 Consider an LTI system with impulse response sequence:
h(n) =
125
SECTION 5.9
1
15
1
(n) +
(n 1) (n 2)
2
8
2
(a) Implement it as the parallel cascade of two LTI systems.

(b) Implement it as the series cascade of two LTI systems.
Problem 5.15 A system is described by the input-output relation
y(n) = x(3n 1) cos

n u(n + 5)
Is the system linear? causal? time-invariant? BIBO stable?

Problem 5.16 A relaxed system is described by the input-output relation:
y(n) = y 2 (n 1) + 2(1)n x(2n)
Is the system linear? time-invariant? causal? BIBO stable?
126
CHAPTER 5
IMPULSE
RESPONSE
SEQUENCE
Problem 5.17 If h(n) is the impulse response sequence of an LTI system, what would the response
of the system be to the input x(n) = (2n 4)?
Problem 5.18 If h(n) is the impulse response sequence of an LTI system, what would the response
of the system be to the input x(n) = (n + 1) + 3(3n + 6)?
Problem
5.19 The response of an LTI system to the input sequence x(n) = u(n) is y(n) =

1 n
u(n).
Is this a BIBO stable system? Is it a causal system? Is it an IIR system? Compute its
2
response to the sequence
8
>
1/3
n = 1
>
>
<
1
n=0
x(n) =
>
1/2
n=1
>
>
:
0
otherwise
without using convolution.
Problem 5.20 The response of an LTI system to the input sequence x(n) = u(n + 2) is y(n) =

1 n1
u(n). Is this a BIBO stable system? Is it a causal system? Is it an IIR system? Compute its
3
response to the sequence
8
>
1/4
n = 2
>
>
<
0
n=0
x(n) =
>
1/4
n=2
>
>
:
0
otherwise
without using convolution.
Problem 5.21 The step response of an LTI system S is y(n) = (1)n u(n). A cascade connection
consists of S followed by the system y(n) = sign[x(n)], where
(
sign[x(n)] =
+1
1
if x(n) 0
otherwise
Find the impulse response of the cascade.

Problem 5.22 The impulse response of an LTI system S is h(n) = (0.2)n1 u(n 1). A cascade
connection consists of the system
y(n) = sign[x(n)]
n
o followed by S. What is the response of the
cascade to the input x(n) =
0.5 , 2.2, 3.1, 4 ?
Problem 5.23 Let

x(n) = (n + 2)
(a) Plot the sequence
h(n) =
1
(n) + 2(n 2)
2
1
1
x(n 1) + (n 1) + u(n 2)
2
4
(b) Can h(n) be the impulse response of a BIBO stable LTI system?
(c) Can h(n 4) be the impulse response of a causal LTI system?
(d) Define the sequence
n1
h1 (n) =
1
3
h(n)u(n)
If h1 (n) were the impulse response sequence of an LTI system, will the system be BIBO
stable?
Problem 5.24 The samples of a sequence x(n) are zero except at the time instants shown Fig. 5.9.
The amplitudes of the non-zero samples are either 1, 2, or 3.
127
(a) Plot the sequence
3
1
h(n) = x(n + 2) (n) + u(n 3)
2
2
SECTION 5.9
PROBLEMS
(b) Can h(n) be the impulse response of a BIBO stable LTI system?
(c) Can h(n 3) be the impulse response of a causal LTI system?
(d) Define the sequence
n
h1 (n) =
1
2
h(n)u(n)
If h1 (n) were the impulse response sequence of an LTI system, will the system be BIBO
stable?
x(n)
3
2
Problem 5.25 The response of a causal LTI system to the input sequence x(n) =
n
is the sequence y(n) =
1 , 1, 2
1/2 , 2, 3 , where the boxed entries occur at the origin of time. Deter-
mine its impulse response sequence, h(n).
Problem 5.26 The response of an LTI system to x(n) = u(n 2) is y(n) =

Find its impulse response sequence. Is this a BIBO stable system? Is it causal?

1 n2
2
u(n 4).
Problem 5.27 The response of an LTI system to x(n) = u(n) is y(n) = (n + 1)u(n). Find h(n)
by solving a triangular system of equations. Is the system BIBO stable?
Problem 5.28 The response of an LTI system to x(n) = 2u(n 1) is y(n) =
Find h(n) by solving a triangular system of equations. Is the system BIBO stable?

1 n
2
u(n 2).
Problem 5.29 Find a difference equation to describe the input-output relation of the LTI system
whose impulse response sequence is given by
h(n) = (n 1) +
1
1
(n 2) (n 3)
3
4
(a) What is the response of the system to x(n) = ej 3 n u(n)?

(a) What is the response of the system to x(n) = 2 cos
n
6
u(n)?
Problem 5.30 Find a difference equation to describe the input-output relation of the LTI system
whose impulse response sequence is given by
h(n) = (n 2) +
1
(n 4)
2
128
(a) What is the response of the system to x(n) = ej 4 (n1) u(n)?
CHAPTER 5
(a) What is the response of the system to x(n) = sin
IMPULSE
RESPONSE
SEQUENCE
n
4
u(n)?
Problem 5.31 What is the impulse response sequence of the overall LTI system whose input sequence is x(n) and output sequence is y(n) in Fig. 5.10. The box in the top branch denotes an LTI
system with impulse response sequence equal to (0.5)n u(n).
x(n)
h(n) =
1 n
2
u(n)
y(n)
FIGURE 5.10 An LTI system involving a feedback connection for Prob. 5.31.
Problem 5.32 What is the impulse response sequence of the overall LTI system whose input sequence is x(n) and output sequence is y(n) in Fig. 5.11. The box in the top branch denotes an LTI
system with impulse response sequence equal to (0.5)n u(n).
x(n)
+
y(n)
h(n) =
1 n
2
u(n)
FIGURE 5.11 An LTI system involving a feedback connection for Prob. 5.32.
Problem 5.33 When the sequence x(n) = { 0 , 1, 1} is applied to a causal LTI system, the odd
part of the resulting output sequence is known to be:
yo (n) = (1/4)n1 u(n 2) + (1/3)n u(n 1),
n0
(a) Find the impulse response sequence of the system.

(b) Find the energy of the impulse response sequence.
(c) Find the power of the impulse response sequence.
Problem 5.34 The even part of the impulse response sequence of a causal LTI system is given by
n
he (n) =
1
2
n2
u(n 1)
1
4
u(n)
129
(a) What is the energy of the impulse response sequence, h(n)?
SECTION 5.9
(b) Find the unit-step response of the system.
PROBLEMS
(c) Find a constant-coefficient difference equation describing the system.

(d) Draw a block diagram representation for the system.
Problem 5.35 What is the impulse response sequence of the overall LTI system whose input sequence is x(n) and output sequence is y(n) in Fig. 5.12. The boxes denote LTI subsystems with
impulse response sequences
n
h1 (n) =
1
2
n
u(n 1),
x(n)
1
3
h2 (n) =
u(n),
h1 (n)
h3 (n) = h1 (n) = 4u(n 4)
y(n)
h2 (n)
h3 (n)
FIGURE 5.12 An LTI system involving parallel and series cascades for Prob. 5.35.
Problem 5.36 What is the impulse response sequence of the overall LTI system whose input sequence is x(n) and output sequence is y(n) in Fig. 5.13. The boxes denote LTI subsystems with
impulse response sequences
n
h1 (n) =
1
2
x(n)
n
u(n 1),
h1 (n)
h2 (n) =
1
3
u(n),
h3 (n) = h1 (n) = 4u(n 4)
h2 (n)
y(n)
h3 (n)
FIGURE 5.13 An LTI system involving parallel and series cascades for Prob. 5.36.
Problem 5.37 An LTI system is excited by a periodic sequence x(n) with period N . Show that
the output sequence is also periodic with the same period.
130
CHAPTER 5
IMPULSE
RESPONSE
SEQUENCE
Problem 5.38 An LTI system is excited by the exponential sequence x(n) = ejo n . Show that
the output sequence is also an exponential sequence with the same angular frequency.
Problem 5.39 Consider the relaxed system
y(n) = y(n 2) + bx(n) + x(n 1)
for some unknown finite positive number b. A student claims that, regardless of the value of b,
the impulse response sequence has to be a power sequence. Do you agree? Prove or disprove
the students statement. The student even claims that knowledge of the average power of h(n) is
sufficient to identify b. Do you agree? If so, what would the value of b be if the average power were
3? If the difference equation were instead
y(n) =
1
y(n 2) + bx(n) + x(n 1)
2
how would your answers change?

Problem 5.40 The input-output relations of two systems S1 and S2 are shown in Fig. 5.14. System
S1 is called an interpolator or upsampler, and system S2 is called a decimator or downsampler.
(a) Verify that S1 and S2 are time-variant systems.
(b) Show that their series cascade is a time-invariant system. Express y(n) in terms of x(n).
(c) What if the order of the systems is reversed in the series cascade? Will the cascade continue
to be a time-invariant system?
x(n)
S1
x(n)
n = 0, 3, 6, . . .
otherwise
x(3n)
S2
x(n)
x( n3 )
0
S1
S2
y(n)
FIGURE 5.14 An upsampler (top plot) is cascaded in series with a downsampler (middle plot) to
obtain the cascade shown in the bottom plot for Prob. 5.40.
CHAPTER
Linear Convolution
he discussion in the previous chapter established that the response of an LTI system
with impulse response sequence h(n) to any input sequence x(n) is given by the convolution sum
X
y(n) = x(n) h(n) =
x(k)h(n k)
(6.1)
k=
In this chapter we study more closely such convolution sums and establish several of their
properties. We also provide physical interpretations for the derived properties.
6.1 PROPERTIES OF THE CONVOLUTION SUM

The convolution sum exhibits several useful properties, which can be used to facilitate the
computation of the convolution, as well as to establish additional properties of LTI systems.
In the discussion that follows we let x(n) and h(n) denote two arbitrary sequences. When
more than two sequences are needed, we will use the notation x1 (n), x2 (n), h1 (n), h2 (n),
with subscripts to refer to multiple sequences.
6.1.1 Commutativity
It holds that
x(n) h(n) = h(n) x(n)
That is,
k=
x(k)h(n k) =
k=
h(k)x(n k)
(6.2)
(6.3)
In other words, the order by which the sequences are convolved does not matter.
Proof: Introduce the new variable = n k. Then
k=
x(k)h(n k) =
X
=
x(n )h()
Rename as k again to obtain
X
=
x(n )h() =
X
k=
h(k)x(n k)
131
Copyright
132
CHAPTER 6
LINEAR
CONVOLUTION
Physical interpretation. We can view x(n) and h(n) as the impulse-response sequences of two LTI systems. These two systems can be cascaded in series in one of
two ways, as shown in Fig. 6.1. In one case, the system represented by h(n) comes first
and in the second case, the system represented by x(n) comes first.
(n)
(n)
x(n)
h(n)
h(n)
x(n)
y1 (n)
y2 (n)
FIGURE 6.1 Two possibilities for the series cascade of the systems with impulse response
sequences h(n) and x(n).
If we feed (n) into the first cascade (top of Fig. 6.1), then the output of the system represented by x(n) would be the sequence x(n) itself. With x(n) feeding into the second
system represented by h(n), the output sequence will be given by the convolution sum
y1 (n) = x(n) h(n)
Likewise, if we feed (n) into the second cascade (bottom of Fig. 6.1), the response of this
second combination will be
y2 (n) = h(n) x(n)
The commutativity property therefore states that both output sequences must agree, namely,
y1 (n) = y2 (n)
It follows that we can always switch the order of LTI systems in a series cascade.
6.1.2 Distributivity
For arbitrary sequences {x(n), x1 (n), x2 (n), h1 (n), h2 (n)}, the following relations hold:
x(n) [h1 (n) + h2 (n)] = x(n) h1 (n) + x(n) h2 (n)
(6.4)
[x1 (n) + x2 (n)] h(n) = x1 (n) h(n) + x2 (n) h(n)
(6.5)
and
Proof: We prove the first relation. An identical argument applies to the second relation. Using the
definition of convolution sum we have
x(n) [h1 (n) + h2 (n)]
k=
k=
PROPERTIES
OF THE
CONVOLUTION
SUM
x(k)[h1 (n k) + h2 (n k)]
x(k)h1 (n k) +
X
k=
x(k)h2 (n k)
x(n) h1 (n) + x(n) h2 (n)
Physical interpretation. Consider two LTI systems with impulse responses h1 (n) and
h2 (n) and assume they are connected in parallel, as shown in Fig. 6.2.
x(n)
h1 (n)
y(n)
h2 (n)
FIGURE 6.2
convolution.
Parallel connection of two LTI systems to illustrate the distributivity property of
Let x(n) be the input sequence that is applied to the cascade connection. We already know
from Sec. 5.5 that the impulse response sequence of the parallel connection is h1 (n) +
h2 (n). Therefore, the output of the system will be x(n) [h1 (n) + h2 (n)]. On the other
hand, the output of h1 (n) is x(n) h1 (n) and the output of h2 (n) is x(n) h2 (n). Hence,
x(n) [h1 (n) + h2 (n)] = x(n) h1 (n) + x(n) h2 (n)
The distributivity property therefore states that the parallel connection of two LTI systems
can be replaced by a single LTI system whose impulse response sequence is the sum of the
individual impulse response sequences.
6.1.3 Associativity
For arbitrary sequences {h1 (n), h2 (n), h3 (n)} it holds that
h1 (n) [h2 (n) h3 (n)] = [h1 (n) h2 (n)] h3 (n) = [h1 (n) h3 (n)] h2 (n)
Proof: Let z(n) denote the result of the convolution h1 (n) h2 (n). That is,
z(n) =
X
k=
h1 (k)h2 (n k)
133
SECTION 6.1
(6.6)
134
Then, using the definition of convolution sum,
CHAPTER 6
[h1 (n) h2 (n)] h3 (n)
LINEAR
CONVOLUTION
z(n) h3 (n)
z()h3 (n )
"
"
h1 (k)h2 ( k) h3 (n )
k=
h1 (k)h2 ( k)h3 (n )
k=
"
k=
X
=
h1 (k)h2 ( k)h3 (n )
"
h1 (k)
k=
h2 ( k)h3 (n )
Introduce the variable m = k and let w(n) = h2 (n) h3 (n). That is,
w(n) =
k=
h2 (k)h3 (n k)
Then we can write
"
h1 (k)
k=
X
=
h2 ( k)h3 (n )
"
h1 (k)
i=
k=
k=
h2 (m)h3 (n k m)
h1 (k)w(n k)
h1 (n) w(n)
h1 (n) [h2 (n) h3 (n)]
We therefore established that

h1 (n) [h2 (n) h3 (n)] = [h1 (n) h2 (n)] h3 (n)
A similar argument can be used to establish equality with [h1 (n) h3 (n)] h2 (n).
Physical interpretation. Consider three LTI systems with impulse response sequences
{h1 (n), h2 (n), h3 (n)} and assume they are connected in series, as illustrated in Fig. 6.3.
h2 (n) h3 (n)
(n)
FIGURE 6.3
convolution.
h1 (n)
h2 (n)
h3 (n)
y(n)
Series connection of three LTI systems to illustrate the associativity property of
Let (n) be the input sequence to this cascade connection. We already know from Sec. 5.4
that the impulse response sequence of the series cascade of the last two systems is h2 (n)
h3 (n). Now the output of the system h1 (n), when excited by an impulse sequence, is
h1 (n). The output of the combined system will therefore be h1 (n) [h2 (n) h3 (n)]. In
a similar manner, if we combine the first two systems together we get an LTI system with
impulse response sequence h1 (n) h2 (n). When this sequence is applied to the system
h3 (n), the resulting output sequence will be [h1 (n) h2 (n)] h3 (n). Therefore,
h1 (n) [h2 (n) h3 (n)] = h1 (n) h2 (n)] h3 (n) .
Similar arguments lead to the other equality. The associativity property therefore states that
in a series cascade of LTI systems, we can combine any pair of systems together. For example, we can combine {h1 (n), h2 (n)} first, or {h2 (n), h3 (n)} first, or even {h1 (n), h3 (n)}
first, and then combine the result with the remaining system. If we group the commutativity and associativity properties together, then we conclude that in a series cascade of
LTI systems, we can reorder the systems at will and the impulse response sequence of the
combination will not change.
6.1.4 Convolution with the Unit-Sample Sequence

For any sequence h(n), it holds that
h(n) (n k) = h(n k)
(6.7)
That is, convolution with a unit-sample sequence always shifts the original sequence, h(n),
to the location of the unit sample.
Proof: Using the definition of the convolution sum we have
h(n) (n k)
X
=
h()(n k ) = h(n k)
since the impulse sequence (n k ) is nonzero only at = n k.
Physical interpretation. Consider an LTI system with impulse response sequence

h(n). Its response to (n) is, obviously, h(n). This result can also be expressed by saying
h(n) = (n) h(n)
where on the left-hand side we have the output sequence (which in this case is h(n) itself),
and on the right-hand side we have the convolution of the impulse response sequence and
the input sequence (which in this case is (n)). Now, by the time-invariance of the LTI
system, the response to (n k) is equal to h(n k). Therefore, it must hold that
h(n k) = (n k) h(n)
6.2 EVALUATION OF THE CONVOLUTION SUM

The convolution sum of two sequences can be evaluated in several ways. We describe two
time-domain methods in this chapter. Other methods will be described in future chapters
by relying on transform techniques.
135
SECTION 6.2
EVALUATION
OF THE
CONVOLUTION
SUM
136
6.2.1 Analytical Method
CHAPTER 6
LINEAR
CONVOLUTION
In this method, we simply employ the definition of the convolution sum of two sequences
to arrive at the result analytically.
Example 6.1 (Convolution of two sequences by analytical method)
Consider the sequences h(n) = 0.5n1 u(n 2) and x(n) = u(n + 1). Then, by definition, their
convolution can be found as follows:
y(n)
k=
k=
x(k)h(n k)
u(k + 1) [0.5nk1 u(n k 2)]
Now the step-sequence u(k + 1) is zero for k < 1, while the step-sequence u(n k 2) is zero
for k > n 2. This means that the product that appears inside the summation symbol will be zero
for all values of k in the interval k < 1 and k > n 2. Therefore, the limits of the summation
should go from k = 1 up to k = n 2 and the expression for y(n) reduces to
y(n)
n2
X
0.5nk1
k=1
Clearly, the summation will include nontrivial elements only for values of n such that n 2 1
or, equivalently, n 1. For n < 1, we get y(n) = 0 and for n 1 we have
y(n)
0.5
n2
X
k+1
k=1
0.5n 1 + 2 + 22 + 23 + . . . + 2n1
1 2n
0.5n
12
n
1
1
2
=
=
=
In summary, we find that

y(n) = 1
n
1
2
u(n 1),
for all n
where we added the step-sequence u(n 1) to enforce y(n) = 0 for n < 1.
6.2.2 Graphical Method
In this method, we evaluate the convolution sum x(n) h(n) graphically by applying the
steps outlined below; these steps are illustrated in the numerical example that follows:
(a) First, we plot the sequences h(k) k and x(k) k. Note that we are denoting the
independent variable by k now. Therefore, the horizontal axis will be the k axis.
(b) Then we plot h(k). In other words, we flip the sequence h(k) around the vertical
axis to obtain h(k).
(c) We subsequently compute the sequence x(k)h(k) by multiplying the sequences
x(k) and h(k) sample-by-sample. We add the samples of the resulting sequence
x(k)h(k). The resulting value would be y(0), namely, the value of the convolution
sum at time n = 0.
(d) Next, we shift h(k) by one time unit to the right in order to obtain h(1 k). We
again compute the product sequence x(k)h(1 k) and add its sample values. This
calculation provides y(1); the value of the convolution sum at time n = 1.
(e) Likewise, we shift h(k) by one time unit the the left to obtain h(1 k). We
compute the product sequence x(k)h(1 k) and add its sample values. This
calculation provides y(1); the value of the convolution sum at time n = 1
(f) We repeat the above procedure by shifting h(k) further to the right and further to
the left and computing the product sequences x(k)h(n k) each time, for positive
and negative n, and adding the resulting samples. This calculation provides the
values of y(n) for the various n.
Before illustrating the above procedure with an example, it is worth noting from the commutativity property of the convolution sum that, in the above graphical procedure, the roles
of h(k) and x(k) can be interchanged: it is irrelevant whether we flip h(k) and shift it or
whether we flip x(k) and shift it.
Example 6.2 (Convolution of two sequences by graphical method)

Let us evaluate the convolution of the following two sequences
n
2, 1 , 1, 2
0 , 1, 2
where we are using the box notation to indicate the location of the sample at time n = 0. The
sequences are illustrated in Fig. 6.4.
h(n)
x(n)
FIGURE 6.4
2
1
1
2
Two sequences x(n) and h(n) whose convolution we are evaluating graphically.
The first sequence has four samples with values

x(1) = 2, x(0) = 1, x(1) = 1, x(2) = 2
and all other samples are zero. Likewise, the second sequence has three samples with values
h(0) = 0, h(1) = 1, h(2) = 2
and all other samples are zero. We now follow the procedure outlined above in order to evaluate the
convolution of both sequences by means of the graphical method.
1. We first plot h(k) k and x(k) k. The result is shown in Fig. 6.5.
137
SECTION 6.2
EVALUATION
OF THE
CONVOLUTION
SUM
138
CHAPTER 6
h(k)
x(k)
LINEAR
CONVOLUTION
1
2
FIGURE 6.5 Plots of x(k) and h(k) for Example 6.2.
2. We then flip h(k) around the vertical axis to obtain h(k), namely,
n
h(k) = 2, 1 0
We further multiply the sequences x(k) and h(k) sample-by-sample. The result is shown
in Fig. 6.6. Adding the terms of the product x(k)h(k) gives y(0) = 2.
x(k)h(k)
h(k)
2 1
FIGURE 6.6 Plots of x(k)h(k) and h(k) for Example 6.2.

3. We now shift h(k) to the right by one unit of time and obtain
n
h(1 k) = 2, 1 , 0
We multiply this sequence by x(k) to obtain x(k)h(1 k). The result is shown in Fig. 6.7.
Adding the terms of the product x(k)h(1 k) gives y(1) = 3.
x(k)h(1 k)
h(1 k)
FIGURE 6.7 Plots of x(k)h(1 k) and h(1 k) for Example 6.2.
139
4. We shift h(k) to the right by one more unit of time to obtain

n
SECTION 6.2
2 , 1, 0
h(2 k) =
x(k)h(2 k)
h(2 k)
2
1

5. We shift h(k) to the right by three units of time and obtain
n
0 , 2, 1, 0
h(3 k) =
x(k)h(3 k)
h(3 k)
2
1

6. We shift h(k) to the right by four units of time and obtain
n
h(4 k) =
0 , 0, 2, 1, 0
7. Any further shifting to the right of the sequence h(k) gives a product x(k)h(n k) = 0 (for
n > 4). Hence,
y(n) = 0 for n > 4
8. We now shift h(k) to the left by one unit of time and obtain
n
h(1 k) = 2, 1, 0, 0
However, the product x(k)h(1 k) evaluates to zero and, therefore,
y(1) = 0
EVALUATION
OF THE
CONVOLUTION
SUM
140
x(k)h(4 k)
CHAPTER 6
LINEAR
CONVOLUTION
h(4 k)
4
2
2
1
In fact, any further shifting to the left of h(k) gives y(n) = 0 for n < 0.
n
9. In conclusion we obtain
y(n) =
2 , 3, 1, 0, 4
The result is illustrated in Fig. 6.11.
x(n) h(n)
4
3
2
1
1
2
FIGURE 6.11
Plot of the sequence that results from the convolution x(n)h(n) for Example 6.2.
6.3 APPLICATIONS
the context of some practical problems.
6.3.1 Echo Cancellation

Consider two real-valued causal sequences, x(n) and y(n). Their correlation (also called
cross-correlation) is denoted by rxy (n) and is defined as the sequence whose samples are
computed as follows:
rxy (n) =
x(k)y(k n)
(6.8)
k=
It is straightforward to verify that the sequence rxy (n) amounts to convolving x(n) with
the time-reversed sequence, y(n).
In this section, we are interested in causal sequences x(n) and y(n) of finite duration
N each. This means that the nonzero samples of x(n) and y(n) are assumed to occur over
the interval 0 n N 1. In this case, the definition of their cross-correlation sequence
reduces to
N
1
X
rxy (n) =
x(k)y(k n)
(6.9)
k=n
where the limits of the summation capture the nonzero samples of x(n) and y(n). Appealing to the graphical method for computing the convolution sum x(n) y(n), and using
the fact that the samples of x(n) exist over 0 n N 1 and the samples of y(n) exist
over (N 1) n 0, we can verify that the nonzero samples of rxy (n) will exist over
the interval (N 1)1 n N 1 (see also Prob. 6.27):
rxy (n) = x(n) y(n), (N 1) n N 1
(6.10)
When x(n) and y(n) are the same sequence, their correlation is called the auto-correlation
sequence of x(n) and is denoted by rx (n):
rx (n) = x(n) x(n),
(N 1) n N 1
(6.11)
Observe in particular that the zeroth term of the auto-correlation sequence of x(n) coincides with the energy value of the sequence:
rx (0) = Ex =
N
1
X
k=0
|x(k)|2
(6.12)
It can be further verified that the auto-correlation sequence, rx (n), is an even sequence and
always assumes its peak value at the location n = 0 (see the top plot of Fig. 6.13 for an
example):
rx (n) = rx (n) and max |rx (n)| = rx (0)
(6.13)
n
We now show how to employ the concepts of correlation and auto-correlation of sequences
to solve an echo cancellation problem.
Channel Probing or Training

Thus, assume that a known sequence x(n) is transmitted over a channel whose scaling
gain, , and delay, d, are unknown. The received signal at the other end of the channel is a
combination of both x(n) and its delayed version, say,
y(n) = x(n) + x(n d)
(6.14)
We say that the term x(n d) represents an echo of the original transmitted signal, x(n);
it has undergone a delay of d samples and scaling by (usually, is smaller than one).
The echo interferes with the transmitted signal, x(n), and the receiver ends up sensing a
combination of both x(n) and x(n d) see Fig. 6.12.
We select an integer N and assume it is large enough such that the nonzero samples
of all three sequences {y(n), x(n), x(n d)} can be assumed to exist within the interval
0 n N 1. We collect N samples of x(n) and y(n). Now given knowledge of
these N samples, we would like to use them to estimate and d. This set-up corresponds
141
SECTION 6.3
APPLICATIONS
142
Direct path:
scaling = 1
delay = 0
CHAPTER 6
LINEAR
CONVOLUTION
Echo path:
scaling =
delay = d
Received signal
x(n) + x(n d)
Source
x(n)
FIGURE 6.12 The echo signal interferes with the transmitted signal, x(n). The receiver senses
the combination y(n) = x(n) + x(n d).
to a situation when we are probing the channel with a training sequence x(n) in order to
estimate its gain and delay parameters.
Using the various properties of the convolution sum operation that were established in
this chapter we note that
ry (n)
y(n) y(n)
[x(n) + x(n d)] [x(n) + x(n d)]
so that
ry (n) = (1 + 2 ) rx (n) + rx (n + d) + rx (n d)
(6.15)
where we used both the distributivity property (6.5) of the convolution sum, as well as
property (6.7) pertaining to convolution with the unit-sample sequence. Indeed, note that
x(n d) x(n) =
=
=
=
[(n d) x(n)] x(n)
(n d) [x(n) x(n)]
(n d) rx (n)
rx (n d)
Likewise,
x(n) x(n d) =
=
=
=
x(n) [x(n) (n + d)]

[x(n) x(n)] (n + d)
rx (n) (n + d)
rx (n + d)
and, similarly,
x(n d) x(n d)
= (n d) [x(n) x(n)] (n + d)
= (n d) rx (n + d)
= rx (n)
It follows from the form of ry (n) in (6.15) that ry (n) has three peak values at the time
instants n = 0, n = d, and n = d (see the bottom plot in Fig. 6.13 for an example).
Therefore, the plot of ry (n) can be used to infer the value of d; it is the location of the
second peak of ry (n):
d = location of second peak of ry (n)
(6.16)
We can further evaluate the cross-correlation between the sequences x(n) and y(n) to
find
rxy (n)
=
=
x(n) y(n)
x(n) [x(n) + x(n d)]
That is,
rxy (n) = rx (n) + rx (n + d)
(6.17)
With the value of already determined, we can use the above relation to estimate from
=
rxy (0) rx (0)

rx (d)
(6.18)
This expression is in terms of the peaks values of rx (n) at n = 0 and n = d, and the peak
value of rxy (n) at n = 0.
Echo Cancellation
The initial probing stage allows us to identify the channel over which the training signal,
x(n), has been transmitted. With the channel parameters already identified, we can then
switch to the normal mode of operation when the receiver does not know the signal that
is being transmitted and would like to recover it. In other words, from knowledge of the
received data and the channel, namely, {y(n), a, d}, we would now like to recover x(n).
This operation amounts to using the channel parameters to clean the received sequence
y(n) from the interference caused by the echo signal, x(n d), and to generate a clean
version of x(n).
The echo cancellation process can be achieved in the time-domain as follows. Since we
are assuming that transmission starts at n = 0, then we know that x(n) is zero for n < 0.
It follows that the first d samples of x(n) and y(n) should coincide. In this way, we can
recover x(n) over 0 n d 1 as follows:
x(n) = y(n),
0nd1
(6.19)
To recover the values of x(n) for the time instants larger than or equal to d, we simply note
from (6.14) that
x(n) = y(n) x(n d), n d
(6.20)
Figure 6.13 illustrates an example with N = 400, d = 100, and = 0.5. The figure shows
the auto-correlations of two sequences x(n) and y(n), where x(n) has been generated
randomly. The peaks of both auto-correlation sequences have been normalized to one in the
plots for ease of display; their values are approximately rx (0) = 182.6 and ry (0) = 226.2.
The locations of the three peaks in ry (n) are indicated by the circles around them. It is
seen that the second peaks occur at locations n = 100, which allow us to identify d as
143
SECTION 6.3
APPLICATIONS
LINEAR
CONVOLUTION
d = 100. The un-normalized values of rxy (0) and ry (d) are approximately rxy (0) = 181.6
and ry (d) = 2. Using (6.18) we get
181.6 182.6
= 0.5
2
as expected.
Autocorrelation of x(n)
1
0.8
0.6
rx(n)
CHAPTER 6
0.4
0.2
0
400
300
200
100
0
100
n
Autocorrelation of y(n)
300
200
100
200
300
400
200
300
400
1
0.8
0.6
ry(n)
144
0.4
0.2
0
400
0
n
100
FIGURE 6.13 The top plot shows the auto-correlation of a randomly generated sequence of
duration N = 200 samples. The bottom plot shows the auto-correlation of the received sequence
y(n) assuming N = 400 and a channel with delay d = 100 and gain = 0.5. The peaks of both
auto-correlation sequences are normalized to one in the plots.
Practice Questions:
1. Show that for any sequence, x(n), its auto-correlation sequence, rx (n) is even and assumes
its peak value at n = 0.
2. Using y(n) = x(n) + ax(n d), establish that the auto-correlation of y(n) has three peaks
at n = 0, n = d, and n = d.
3. Assume x(n) = u(n) u(n 3), a = 0.5 and d = 5. Find the auto-correlation sequences
of x(n) and y(n). Find also the cross- correlation sequence of x(n) and y(n).
4. How is the value of ry (0) in (6.15) related to ?
5. Assume y(n) = 2x(n 1) + x(n d). How would you estimate and d from knowledge
of samples of {x(n), y(n)}?
6.3.2 Population Growth Management

In Sec. 4.11 we studied the problem of population growth and introduced the Malthusian
model (4.19), where the evolution of the population size was dictated by the birth (b%)
and death (d%) rates in the population. We considered a general scenario where the dynamics of the population growth can be influenced by adding or removing individuals at
arbitrary time instants, and not only at the initial time instant. The addition and removal
of individuals in a population is often the result of immigration and emigration acts. More
generally, targeted population management techniques are used to manage population sizes
in fisheries and other animal groups.
When management is desired, the evolution of the population size os governed by a
recursion of the form (4.23), namely,

d
b
y(n 1) + x(n), y(1) = 0
(6.21)
y(n) = 1 +
100 100
where x(0) = Po corresponds to the initial population size, and the sequence x(n) represents the input to the system at the various time instants. The above constant-coefficient
difference equation represents a causal LTI system with input sequence x(n) and output
sequence y(n).
Let us determine the impulse response sequence of system (6.21). To do so, we set
x(n) = (n) and y(n) = h(n) and use the difference equation to write

d
b
h(n 1) + (n), h(1) = 0
(6.22)
h(n) = 1 +
100 100
By iterating the recursion, we are able in this case to find a closed-form expression for h(n).
First note from the recursion that h(n) = 0 for all n < 0; this result is consistent with the
fact that we are dealing with a causal LTI system, and the impulse response sequences of
such systems must be causal sequences. Moreover, for n 0, we obtain from recursion
(6.22) the following sequence of results:
h(0) =
h(1) =
h(2) =
..
.
1

b
1+
100

b
1+
100
..
.

b
h(0) = 1 +
100

b
d
h(1) = 1 +
100
100
d
100
d
100
d
100
2
so that
h(n) =

n
b
d
1+
u(n)
100 100
(6.23)
Let us use this result to determine the response of the LTI system (6.21) to the case
when, starting from year one, the user adds a individuals to the population every odd year
and removes r individuals from the population every even year. The input sequence x(n)
under consideration can be captured by the following expression:

n
n1
ru
(6.24)
x(n) = (Po + r) (n) + a u
2
2
Observe that we are including two step-sequences to the expression for x(n).The sequence
u(n/2) is equal to one at all nonnegative even values of n (including n = 0) and is zero
elsewhere. For this reason, we need to add r to Po at n = 0 in order to eliminate the
contribution of r that comes from r u(n/2) at n = 0; this is because withdrawals start
145
SECTION 6.3
APPLICATIONS
146
CHAPTER 6
LINEAR
CONVOLUTION
occurring only at time n = 2 onwards. Likewise, the sequence u((n 1)/2) is equal to
one at all positive odd values of n and it helps us model the addition of a individuals at
these time instants.
To determine the response of the LTI system (6.21) to x(n) we resort to the convolution
sum of x(n) with h(n). First, let
=

d
b
1+
100 100
(6.25)
Now since the system is causal and the input sequence, x(n), is a causal sequence, we can
use (5.11) to write
n
X
y(n) =
x(k)h(n k)
(6.26)
k=0
n
X

k1
k
(Po + r) (k) + a u
ru
nk u(n k)
2
2
k=0
#
" n

X
k1
nk
n
(6.27)
(Po + r) + a
u
2
k=0
" n
#
X
k
nk
r
u
=
=
k=0
To evaluate the two sums that appear on the right-hand side, we note that their values
depend on whether n is even or odd. Thus, note that

n
X
k1
n1 + n3 + . . . + 2 + 1,
u
nk =
n1 + n3 + . . . + ,
2
n odd
n even
k=0
where in the first case (n odd) we are adding (n + 1)/2 terms of a geometric series with
ratio 2 and first term equal to one. In the second case of n even, we are adding n/2 terms
of a geometric series with ratio 2 and first term equal to . Using the result of Example
2.11 for the sum of a finite number of terms of a geometric series, we get
n+1
1 ( 2 ) 2
,
1 2

n
X
k1
nk =
u
n
k=0
1 ( 2 ) 2

,
1 2
Likewise,
n

n
X
k
+ n2 + . . . + ,
nk =
u
n + n2 + . . . + 2 + 1,
2
k=0
n odd
n even
n odd
n even
where in the first case (n odd) we are adding (n + 1)/2 terms of a geometric series with
ratio 2 and first term equal to . In the second case of n even, we are adding n/2 + 1
terms of a geometric series with ratio 2 and first term equal to 1. Using again the result
147
of Example 2.11 for the sum of a finite number of terms of a geometric series, we get
n+1
1 ( 2 ) 2
,
n odd

n
1 2
X
k
nk
=
u
n
k=0
1 ( 2 ) 2 +1
,
n even
1 2
SECTION 6.4
APPLICATIONS
Substituting these results into the convolution sum expression (6.26) for y(n) we arrive at
1 n+1
1 n+1
(Po + r) n + a
,
n odd
1 2
1 2
y(n) =
(6.28)
n
n+2
(Po + r) n + a
r
,
n even
1 2
1 2
These expressions describe the evolution of y(n) over n 0. Figure 6.14 illustrates the
evolution of the population as a function of time assuming a birth rate of 4%, a death rate
of 2%, addition of 3 individuals every odd year, and removal of 5 individuals every even
year.
birth rate=4%, death rate=2%, addition=3, and removal=5
125
y(n)
120
115
110
105
100
10
n (years)
12
14
16
18
FIGURE 6.14 Evolution of the population size over time assuming a birth rate of 4%, a death
rate of 2%, addition of 3 individuals every odd year, and removal of 5 individuals every even year.
Practice Questions:
1. Determine the response sequence, y(n), of the Malthusian system (6.21) when x(n) =
100(n) + 4(n 2) 2(n 3).
2. Assume Po = 100 individuals, b = 5%, d = 1%, a = 3 and r = 1. Find the population size
at years n = 1, 2, 3, 4, 5.
3. Repeat when a = 3 and r = 3.
148
6.4 PROBLEMS
CHAPTER 6
LINEAR
CONVOLUTION
Problem 6.1 Let y(n) = x(n) h(n). Express the following convolution sums in terms of the
sequence y(n):
(a) x(n) h(n 1).
(b) x(n 1) h(n).
(c) x(n 1) h(n 2).
(d) x(n) h(n 3).
Problem 6.2 Let y(n) = x(n) h(n). Express the following convolution sums in terms of the
sequence y(n):
(a) x(n + 2) h(n 2).
(b) x(n 3) h(n + 1).
(c) x(n) h(n 2).
(d) x(n) x(n 1) h(n) h(n + 2).
Problem 6.3 Evaluate the convolution sums:

1 n
u(n 1).
2

1 n
u(n) 2 u(n 1).
n
u(2n) 12 u(n).
(a) u(n)
(b)
(c)
Problem 6.4 Evaluate the convolution sums:

1 n1
u(n).
4

1 n
2) 4 u(n

1 n
u(n 1).
4
(a) u(n + 1)
(b) u(n +
(b) u(4n)
1).
Problem 6.5 Evaluate the convolution sum

n
1
4
n
1
2
u(n)
u(n)
using both the analytical and graphical methods. Compare your results.
n+1
1
4
n1
1
2
u(n 2)
u(n + 1)
n1
u(n + 2)
1
3
u(n 2)
n
u(n 2)
1
2
u(n)
149
Problem 6.9 Evaluate

n1
1
2
n
1
3
u(n)
SECTION 6.4
PROBLEMS
u(n 2) u(n + 1)
using the distributivity property of the convolution sum.

"
n+2
n+1
1
4
1
2
u(n)
n3
1
3
u(n 1)
u(n + 2)
using the distributivity property of the convolution sum.


1
1
(n + 1) (n)
2
3
n1
1
2
n
1
3
u(n)
u(n 2)

1
1
(n + 2) + (2n 2)
2
4
n+3
1
2
n
u(n)
1
3
u(n + 2)
Problem 6.13 Use the graphical method to evaluate the convolution sum shown in Fig. 6.15.
h(n)
x(n)
2
1
1
2
2
1
11
FIGURE 6.15 Convolution sum of two sequences for Prob. 6.13.
Problem 6.14 Use the graphical method to evaluate the convolution sum shown in Fig. 6.16.
150
CHAPTER 6
LINEAR
CONVOLUTION
h(n)
x(n)
2
1
1
2
2
1
11
FIGURE 6.16 Convolution sum of two sequences for Prob. 6.14.
Problem 6.15 A sequence x(n) is nonzero for values of n between 3 and 4 only, and a sequence
y(n) is nonzero for values of n between 7 and 9 only. Let z(n) = x(n) y(n). What can you say
about the sequence z(n)?
(a) It is equal to zero for n < 3 and n > 12.
(b) It can be nonzero at n = 0.
(c) It is nonzero at any point 3 n 12.
(d) None of the above.
Problem 6.16 A sequence x(n) is nonzero for values of n between 0 and 3 only, and a sequence
z(n) is nonzero for values of n between 1 and 5 only. If z(n) = x(n) y(n). What can you say
about the sequence y(n)?
(a) It is equal to zero for n < 1 and n > 5.
(b) It is zero at n = 1.
(c) It cannot be zero at n = 0.
(d) None of the above.
Problem 6.17 Consider the sequence x(n) shown in Fig. 6.17. The sequence is zero except at the
specified time instants. The amplitudes of the non-zero samples are either 1, 2, or 3.
x(n)
3
2
a) Define the sequence y(n) = u(n + 1) u(n 2). Compute the convolution x(n) y(n).
b) Define
151
n
1
2
h1 (n) =
SECTION 6.4
h(n)u(n)
PROBLEMS
where
1
3
x(n + 2) (n) + u(n 3)
2
2
Take h1 (n) to be the impulse response
of
an
LTI
system. What would the response of the
n
system be to the input sequence 13 u(n)?
h(n) =
Problem
6.18 Determine the output of an LTI system with impulse response sequence h(n) =

1 n
u(n)
when excited by each of the following input sequences:
2
(i) x(n) = u(n).
n
(ii) x(n) = 1, 0 , 1 .
(iii) x(n) =

1 n
3
n
u(n).
(iv) x(n) = 2 u(n).

Problem 6.19 Determine the convolution
[u(n) u(n L)] [u(n) u(n L)]
where L is a positive integer. Plot the resulting sequence for L = 5.
Problem 6.20 Determine the convolution
n
[u(n) u(n 5)]
1
2
u(n)
Problem 6.21 Let z(n) be such that z(n) x(n + 1) = x(n 2) y(n 1) for any x(n) and
y(n). Express the sequence z(n) in terms of y(n).
Problem 6.22 Assume x(n) h(n) = y(n) h(n). What can you say about the relation between
the sequences x(n) and y(n)?
Problem 6.23 Consider two possibly complex-valued sequences x(n) and h(n). Their crosscorrelation is the sequence rxh (n) whose samples are defined as follows:
rxh (n) =
k=
x(k)h (k n)
(a) Verify that rxh (n) = x(n) h (n).

(b) Use the graphical method to evaluate the correlation of the two sequences:
n
x(n) =
2, 1 , 1, 2
and
h(n) =
0 , 1, 2
Problem 6.24 The autocorrelation of a possibly complex-valued sequence x(n) is another sequence whose samples are defined as
rx (n) =
X
k=
x(k)x (k n) = x(n) x (n)
(a) Is rx (n) an even sequence? Is it conjugate symmetric?

(c) Find the autocorrelation of x(n) = (0.5)n u(n).
Problem 6.25 Let rx (n) denote the autocorrelation of a possibly complex-valued sequence x(n).
Show that rx (0) is equal to the energy of x(n).
152
CHAPTER 6
LINEAR
CONVOLUTION
Problem 6.26 Consider an even sequence x(n). Let h(n) = x(n) x(n). Show that h(0) is equal
to the energy of the sequence. How does h(n) relate to the autocorrelation sequence of x(n)?
Problem 6.27 Assume x(n) has nonzero samples only in the interval N1 n N2 . Likewise, assume h(n) has nonzero samples only in the interval M1 n M2 . All quantities
(N1 , N2 , M1 , M2 ) are positive integers. Generally, over what interval of time will the sequence
x(n) h(n) have nonzero samples? Prove your result and check it on a numerical example with
distinct values for the integers (N1 , N2 , M1 , M2 ).
Problem 6.28 Assume x(n) has nonzero samples only in the interval N1 n N2 . Generally,
over what interval of time will the following sequences have nonzero samples?
(a) r(n) = x(n) x(n).
(b) y(n) = x(n) x(n).
(c) y(n) = x(n) u(n).
Problem 6.29 The impulse response sequence of a causal LTI system is h(n) = u(n 2). Its
n2
response to an unknown input sequence x(n) is y(n) = 12
u(n 3). Find x(n).
Problem 6.30 Let h(n) denote the impulse response sequence of a causal LTI system. The ren1
sponse of the system to an unknown input sequence x(n) is y(n) = 13
u(n 2). What is the
response of the system to the input sequence x1 (n) defined by
x1 (n) = (n 2) + x(n 3) + x(n 1) y(n 3) + (n + 5)
n4
1
2
x(n + 1)?
Problem 6.31 A causal LTI system is described by the difference equation

y(n) =
1
y(n 1) + x(n 1)
4
Find its impulse response sequence. Find also the response to the input sequence shown in Fig. 6.18
using the convolution sum method.
x(n)
2
1
1
2
FIGURE 6.18 Input sequence x(n) for Prob. 6.31.

y(n) =
1
y(n 1) x(n + 2)
2
Find its impulse response sequence. Find also the response to the input sequence shown in Fig. 6.19
using the convolution sum method.
153
SECTION 6.4
x(n)
2
1
PROBLEMS
11

y(n) =
1
y(n 1) + x(n)
3
Find its response to the following input sequences

(a) x(n) = u(n).
(b) x(n) = u(n).
y(n) =
1
y(n 1) + x(n 1)
2
Find its response to the following input sequences

(a) x(n) = u(n).
(b) x(n) =

1 n
4
u(n).
CHAPTER
Homogeneous Difference Equations
In the earlier chapters we introduced several properties of discrete-time signals and systems (such as periodicity, causality, stability, linearity, and time-invariance). From this
chapter onwards, we start to develop tools for the analysis of discrete-time systems and,
in particular, LTI systems. These tools will enable us to answer questions such as how
to determine the response of a system to an input sequence in a more systematic manner
rather than continually resorting to convolution sum calculations or to iterating the respective difference equations. We shall not study general discrete-time systems. Instead, we
shall focus on the important subclass of systems that are described by constant-coefficient
difference equations.
Our objective in the present chapter, and in the following one, is to describe a procedure
for determining the response of systems described by constant-coefficient difference equations; the equations may or may not represent LTI systems. A useful first step towards this
objective is to understand how to solve homogeneous difference equations.
7.1 HOMOGENEOUS EQUATIONS

First-Order Equations
For motivation purposes, assume we are asked to identify a sequence y(n) that satisfies the
difference equation:
y(n) ay(n 1) = 0
(7.1)
for all n and for some given coefficient a (real or complex). This is a first-order difference
equation in the variable y(n). The equation is said to be homogeneous since its right-hand
side is zero and its left-hand side is a combination of y(n) and time-shifted versions of
y(n) (in this case, y(n 1)). The equation, as described, cannot be viewed as an inputoutput relation for a system since it does not specify an input sequence, x(n). Still, we
may interpret the equation as defining the response of the class of systems
{ y(n) ay(n 1) = x(n) }
to the zero input sequence, x(n) = 0. By determining all sequences {y(n)} that satisfy
the homogeneous equation (7.1) we would then be determining all possible responses of
the above class of systems to the zero input sequence.
Returning to (7.1), it is immediate to verify that the exponential sequence
y(n) = an
satisfies the homogeneous equation since

an a an1 = 0
Copyright
155
156
It is also easy to verify that any multiple of an is a solution as well, i.e., the choice
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
y(n) = Can , for any constant C
(7.2)
satisfies (7.1), where C can be real or complex-valued. We therefore find that even trivial
homogeneous equations of the form (7.1) admit an infinite number of solutions; one for
each choice of the constant C in (7.2). The same conclusion holds for homogeneous equations of higher-order, as we proceed to verify.
Higher-Order Equations
To begin with, an M -th order homogeneous equation is one of the form
y(n) + a1 y(n 1) + a2 y(n 2) + . . . + aM y(n M ) = 0
(7.3)
where the {ai } are scalar coefficients (real or complex; in general, they will be real-valued).
Observe that M delayed versions of y(n) appear in (7.3) and, hence, the equation is said
to be of order M due to the presence of the term y(n M ). Observe further that we
are normalizing the coefficient of y(n) to one while the coefficients for the time-delayed
versions of y(n) are denoted by {a1 , a2 , . . . , aM }. Our objective is to determine the form
of all sequences {y(n)} that satisfy the above homogeneous equation. The zero sequence,
y(n) = 0
for all n
is obviously one solution. We proceed to verify that we can find nontrivial solutions of the
form y(n) = n , for some nonzero number whose value can be real or complex. In other
words, we now verify that exponential sequences, n , are solutions of the homogeneous
equation (7.3) for some values of to be determined. To see that this is indeed the case,
we substitute the assumed form, y(n) = n , into the homogeneous equation (7.3) and find
that the scalar must satisfy the following relation for all n:
n + a1 n1 + . . . + aM nM = 0
(7.4)
This condition is equivalent to requiring to satisfy

nM M + a1 M1 + . . . + an1 + aM = 0
But since 6= 0 by assumption, we conclude that has to be a root of the following

algebraic equation (also known as the characteristic equation associated with (7.3)):
M + a1 M1 + a2 M2 + . . . + an1 + aM = 0
(7.5)
p() = M + a1 M1 + a2 M2 + . . . + an1 + aM
(7.6)
The polynomial
is known as the characteristic polynomial associated with the homogeneous equation (7.3).
Observe that the characteristic polynomial has the same order M as the homogeneous
equation. Moreover, the coefficient of M is equal to 1 while the coefficients for the other
decreasing powers of are equal to {a1 , a2 , . . . , aM }.
Now, it is a well-known result from algebra theory that every polynomial of order M
of the form (7.3) has M roots in the complex plane (some of them possibly repeated).
This result is known as the Fundamental Theorem of Algebra. Thus, let us denote these
roots by {1 , 2 , . . . , M }. The roots can be real or complex. When all the coefficients
{a1 , a2 , . . . , aM } are real-valued, then complex roots of p() = 0 can only occur in conjugate pairs. This statement means that if o is some complex root then so is its complex
conjugate, o . The following situations arise.
7.1.1 Distinct Roots

Assume first that all the roots { } of the characteristic equation (7.5) are distinct (real or
complex). It follows that we can find M solutions for the homogeneous equation (7.3),
each of the form
{ y(n) = n , 1 M }
There are in fact infinitely many solutions. It is straightforward to verify that any arbitrary
linear combination of the individual solutions {n } is also a solution of (7.3), say,
y(n) = C1 n1 + C2 n2 + . . . + CM nM
(7.7)
for arbitrary (real or complex) coefficients {C }.
7.1.2 Repeated Roots

Let us examine what happens when some of the roots of the characteristic equation (7.5)
are repeated. Assume initially that the characteristic equation has a nonzero double root at
some value 1 . We already know that the sequence y(n) = n1 satisfies the homogenous
equation (7.3). We now verify that because 1 is a double root, the sequence y(n) = nn1
also satisfies the homogeneous equation (7.3).
Proof: The fact that 1 is a double root of the characteristic equation (7.5) means that both the
characteristic polynomial (7.6) and its derivative should vanish at 1 :

p(1 ) = 0
dp()
= 0
d =1
and
That is,
M 1
2
M
+ a 2 M
+ . . . + aM 2 21 + aM 1 1 + aM = 0
1 + a 1 1
1
(7.8)
and
1
2
3
M M
+ a1 (M 1)M
+ a2 (M 2)M
+ . . . + 2aM 2 1 + aM 1 = 0
1
1
1
Since 1 is assumed to be nonzero, this second equality can be rewritten as

h
M
M 1
2
1
+ a 2 M
+ . . . + aM 2 21 + aM 1 1
1 M 1 + a 1 1
1
{z
=aM
by virtue of the first condition (7.8)

i
2
3
4
a 1 M
+ 2a2 M
+ 3a3 M
+ . . . + (M 2)aM 2 1 + (M 1)aM 1 = 0
1
1
1
so that the following relation holds:

1
2
3
a 1 M
+ 2a2 M
+ 3a3 M
+ . . . + (M 1)aM 1 1 + M aM = 0
1
1
1
(7.9)
157
SECTION 7.1
HOMOGENEOUS
EQUATIONS
158
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
We now use this fact to establish the desired result that nn

1 is a solution to the homogeneous equation
(7.3). For this fact to hold, we note by substituting nn
1 into the left-hand side of (7.3) that the
following equality must hold:
n1
nn
+ a2 (n 2)n2
+ . . . + aM (n M )nM
= 0
1 + a1 (n 1)1
1
1
This equality is equivalent to requiring

h
M 1
nnM
M
+ . . . + aM 1 1 + aM
1 + a 1 1
1
1
2
nM
a 1 M
+ 2a2 M
+ . . . + M aM
1
1
1
= 0
The above condition holds in view of the two equalities (7.8) and (7.9) shown above.
In summary, when n1 is a double root, then both n1 and nn1 are solutions of the homogeneous equation (7.3); it can be further verified that any linear combination of these two
solutions is also a solution, i.e.,
y(n) = [Co + C1 n] n1
(when 1 is a double root)
(7.10)
Similar arguments can be used for repeated roots of higher multiplicities. For example, if
1 is a root of multiplicity 3, then the sequences n1 , nn1 and n2 n1 are solutions of (7.3),
as well as any linear combinations of these terms, and so forth.
7.1.3 Complex Roots

When the coefficients {a1 , a2 , . . . , aM } are real-valued, complex roots of the characteristic
equation (7.5) can only occur in conjugate pairs. If {1 , 1 } represent a conjugate pair of
roots, then they contribute a term of the following form to the solution y(n):
y(n) = C1 n1 + C2 (1 )n
(general case)
(7.11)
for arbitrary complex numbers {C1 , C2 }. The above expression is simply a linear combination of the individual contributions n1 and (1 )n that originate from the modes {1 , 1 }.
When the terms of the sequence y(n) are required to be real-valued, then C2 and C1 must
be conjugate pairs as well, i.e., they should satisfy C2 = C1 . In this case, y(n) will be
expressed as the sum of two complex conjugate terms, in which case the sum assumes
real-values:
y(n) = C1 n1 + C1 (1 )n (real-valued sequence)
(7.12)
Proof: We already know that n
1 satisfies the homogeneous equation (7.3), namely,
n1
n
+ a2 n2
+ . . . + aM nM
= 0
1 + a 1 1
1
1
Conjugating both sides of the expression, and using the fact that the coefficients {a } are real-valued,
we find that
(1 )
+ a1 (1 )
n1
+ a2 (1 )
n2
+ . . . + aM (1 )
nM
= 0
so that the sequence (1 )n is also a solution to (7.3); this is a confirmation of the fact that complexroots must occur in conjugate pairs when the coefficients {a } are real-valued. It follows that any
linear combination of the form C1 n

1 + C2 (1 ) is also a solution. When C2 = C1 , the samples of
the sequence y(n) become real-valued, as can be seen from the following argument. If we express
C1 and 1 in polar forms, say, as
C1 = Aej ,
1 = ej
n
with {A, } real and {, } [, ], then the term C1 n
1 + C1 1 can be equivalently expressed
as the sinusoidal sequence
y(n) = 2An cos[n + ]
which is real-valued.
7.1.4 Solution Method
In summary, in order to determine nontrivial solutions for the homogeneous equation (7.3),
we proceed as follows:
(a) Solve the characteristic equation
M + a1 M1 + a2 M2 + . . . + aM1 + aM = 0
and find its M roots. The roots { } are called modes. Some modes may be realvalued and other modes may be complex-valued. Some modes may be simple and
other modes may have multiplicity larger than one.
(b) Every distinct root o (whether real or complex) contributes a term of the form Cno
to the solution y(n), for some arbitrary constant C.
(c) Every repeated root o , say with multiplicity m (and whether real or complex), contributes to the solution y(n) with a term of the form
[Co + C1 n + C2 n2 + . . . + Cm1 nm1 ]no
for some arbitrary constants {Co , C1 , . . . , Cm1 }
(d) When the coefficients {a1 , a2 , . . . , aM } are real-valued, complex roots must occur
in conjugate pairs. When the solution y(n) is required to be real-valued, then the
pair of roots {o , o } contributes a term of the following form to y(n):
Cno + C (o )n
for some arbitrary complex number C. This term can be expressed as the sinusoidal
sequence:
2An cos[n + ]
in terms of the real parameters {A, }, and the phases {, } [, ], that arise
from the polar representation of C and o :
C = Aej ,
o = ej
(e) All solutions y(n) to the homogeneous equation (7.3) are obtained by linearly combining all the terms contributed by the modes of its characteristic equation.
To complete the argument, we still need to show that the above construction provides all
nontrivial solutions of the homogeneous equation (7.3). This is indeed the case, but we
159
SECTION 7.2
HOMOGENEOUS
EQUATIONS
160
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
forgo a formal proof here. Instead, we shall focus henceforth on the important case in
which we desire to determine the solution to a homogeneous equation under the requirement that the solution should satisfy a given set of initial conditions. In this situation, we
are going to see that the homogeneous equation can only have a unique solution, and that
the above construction will lead to it.
7.2 HOMOGENEOUS EQUATIONS WITH INITIAL CONDITIONS

Consider again the difference equation (7.3) but assume now that we are given the values
of M initial conditions, say at times:
{y(1), y(2), . . . , y(M )}
(7.13)
We would like to determine the sequence (or sequences) y(n) that satisfy (7.3) and meet the
given initial conditions. Note that starting from the conditions (7.13), we can in principle
iterate recursion (7.3) and determine the values of y(n) for all n. For example, consider the
first-order equation (7.1) and assume we start from the initial condition y(0) = yo , namely,
y(n) = ay(n 1),
y(0) = yo
(7.14)
Iterating the recursion over n 1 we get

y(1) =
y(2) =
y(3) =
..
. =
ay(0) = yo a
ay(1) = yo a2
ay(2) = yo a3
..
.
Likewise, running the recursion backwards we get for n 1:

y(1) =
y(2) =
y(3) =
..
.
1
1
y(0) = yo
a
a
1
1
y(1) = yo 2
a
a
1
1
y(2) = yo 3
a
a
..
.
A pattern emerges and we can express the resulting sequence y(n) more compactly in the
form
y(n) = yo an
(7.15)
Observe that we are led to a well-defined and unique sequence, y(n). Note further that
this solution is a special case of the general expression (7.2); it corresponds to the special
choice C = yo . This choice of the constant C leads to the unique solution y(n) that passes
through the condition y(0) = yo .
More generally, it is obvious from the above argument that iterating a difference equation, as in (7.3), starting from a given set of initial conditions, as in (7.13), leads to a unique
sequence y(n) that satisfies both the homogeneous equation and the initial conditions. We
therefore conclude that every M th order homogeneous difference equation, such as (7.3),
with M initial conditions, has a unique solution y(n). Determining this solution by iterating the equation, as we did in the above first-order example, is not always feasible. To
find the solution in closed form, without resorting to exhaustive iteration of the difference
equation, we proceed to show how to employ the procedure just described in Sec. 7.1) by
considering several examples.
Example 7.1 (Distinct modes)
We wish to determine the unique solution of the homogeneous equation
y(n) y(n 1) 2y(n 2) = 0
that passes through the initial conditions
y(1) = 1,
y(2) = 2
The first step is to write down the characteristic equation and determine its modes. In this case, the
characteristic equation is given by
2 2 = 0
and it has two distinct modes at
= 2,
= 1
These modes contribute with individual terms of the form (2)n and (1)n to the solution y(n),
Accordingly, we know from the procedure described in Sec. 7.1, that all sequences that satisfy the
homogeneous equation are parameterized as follows:
y(n) = C1 2n + C2 (1)n
for arbitrary constants {C1 , C2 }. Each choice of {C1 , C2 } gives one possible solution sequence
y(n) that satisfies the homogeneous equation but not necessarily the assumed initial conditions. To
determine from among these solutions that sequence y(n) that satisfies y(1) = 1 and y(2) = 2,
we select the constants {C1 , C2 } in order to enforce the initial conditions. Using the initial values
y(1) = 1 and y(2) = 2, we find that {C1 , C2 } must satisfy the following system of linear
equations:
(
1 = C1 /2 C2
2 = C1 /4 + C2
Solving we get C1 = 4 and C2 = 1. Hence, the desired solution sequence is
y(n) = 4 2n + (1)n ,
for all n
It is straightforward to verify that this sequence satisfies the given initial conditions at times n = 1
and n = 2. Moreover, it also satisfies the homogeneous equation.
Example 7.2 (Repeated modes)

Consider now the homogeneous equation
y(n) 4y(n 1) + 4y(n 2) = 0
with initial conditions y(1) = 1 and y(2) = 0. The corresponding characteristic equation is
given by
2 4 + 4 = 0
with repeated modes that are equal to
= 2,
=2
161
SECTION 7.2
HOMOGENEOUS
EQUATIONS
WITH
INITIAL
CONDITIONS
162
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
Accordingly, from the procedure described in Sec. 7.1, we know that the form of the general solution
for the homogeneous equation is
y(n) = C1 2n + C2 n2n
We again use the initial conditions to arrive at the following linear system of equations in terms of
the unknowns {C1 , C2 }:
(
1 = C1 /2 C2 /2
0 = C1 /4 C2 /2
Solving we get C1 = 4 and C2 = 2. Hence, the desired solution sequence is

y(n) = 2n+2 + n2n+1 ,
for all n
Example 7.3 (Complex modes and a real solution sequence)
Consider the homogeneous equation

y(n) + y(n 2) = 0
with initial conditions y(1) = 1 and y(2) = 1. It is clear that if we iterate the recursion for
all values of n, the resulting samples of y(n) will all be real-valued. Therefore, we are seeking the
unique real-valued sequence y(n) that satisfies the homogeneous equation and passes through the
given initial conditions.
The corresponding characteristic equation is given by
2 + 1 = 0
with complex modes at
= j = ej/2 ,
= j = ej/2
Accordingly, from the procedure described in Sec. 7.1, we know that the form of the general solution
for the homogeneous equation is
y(n) = Cj n + C (j)n
We again use the initial conditions to arrive at the following linear system of equations in terms of
the unknown C and its complex conjugate:
(
1 = Cj C j
1 = C C
Solving we get
3
1
1
C = (1 + j) = ej 4
2
2
Hence, the desired solution sequence is
y(n)
=
=

n
n
1 3
1 j 3
+ ej 4 ej/2
e 4 ej/2
2
2
i
3
3
n
1 h j ( n
e 2 4 ) + ej ( 2 4 )
2
which, in view of Eulers relation (3.11), reduces to

y(n) = cos
n
2
4
163
Example 7.4 (Complex modes and a complex solution sequence)
SECTION 7.3
IMPULSE
RESPONSE
OF LTI
SYSTEMS
Consider the same homogeneous equation

y(n) + y(n 2) = 0
but with initial conditions y(1) = 0 and y(0) = j. Now, if we iterate the recursion for all values of
n, we find that the resulting sequence will have complex-valued samples. To determine the sequence,
we proceed as before but now express the general solution of the homogeneous equation in the form
y(n) = C1 j n + C2 (j)n
for two complex constant C1 and C2 that are not necessarily complex conjugates of each other.
We use the initial conditions to arrive at the following linear system of equations in terms of the
unknowns C1 and C2 :
(
j = C1 + C2
0 = jC1 + jC2
Solving we get
C1 = C2 = j/2
Observe that the coefficients {C1 , C2 } in this case are not complex conjugates of each other. Hence,
the desired solution sequence is
y(n)
j n
j
j +
(j)n
2
2
1
[1 + (1)n ] j n+1
2
1
[1 + (1)n ] ej 2 (n+1)
2
=
=
=
(
so that
y(n) =
0,
n
(1) 2 j,
n odd
n even
7.3 IMPULSE RESPONSE OF LTI SYSTEMS

One useful application of the solution method we just described for finding the homogeneous response of a constant-coefficient difference equation is that it can be used to
determine closed-form expressions for the impulse response sequence of LTI systems that
are described by such difference equations.
Example 7.5 (Distinct modes)
Consider a causal system that is described by the difference equation
y(n) + 2y(n 1) 8y(n 2) = 2x(n)
Let us assume the system is relaxed, so that the above input-output relation describes an LTI system.
We want to determine its impulse response sequence, i.e., the response to x(n) = (n). We move
slowly in this first example in order to highlight the main ideas.
Since the system is causal and LTI, we already know that its impulse response sequence, denoted
by h(n), should satisfy
h(n) = 0 for n < 0
(7.16)
164
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
Moreover, over n 0, the sequence h(n) should satisfy the same difference equation with x(n)
replaced by (n), i.e.,
h(n) + 2h(n 1) 8h(n 2) = 2(n),
n0
(7.17)
We want to solve this equation and determine h(n) over n 0.

Since the system is relaxed, it must hold that h(2) = 0 = h(1). This is because, by the
definition of a relaxed system, the output h(n) has to remain at zero as long as the input sequence
stays at zero. In the present situation, the input sequence is given by x(n) = (n), and it is zero for
all n < 0; it moves away from zero only at time n = 0.
Now note that the above difference equation for h(n) is not a homogeneous equation since it
contains the term (n) on the right-hand side. However, since (n) is zero for all n 1, we
conclude that the difference equation becomes homogeneous over the interval n 1 and can be
written as
h(n) + 2h(n 1) 8h(n 2) = 0, for n 1
This is a second-order homogeneous equation in h(n). In order to determine the sequence h(n), we
need to identify two initial conditions. Usually, but not necessarily, we identify these conditions at
the two time instants right before n = 1 (when the equation became homogeneous). These initial
conditions will in general reflect the impact of the unit-sample sequence, (n), that was present at
n = 0. We already know that h(1) = 0. For time n = 0, we use the difference equation (7.17)
and the fact that (0) = 1 to find that h(0) = 2.
Starting from the homogeneous equation (7.17), we reduced the problem to solving the secondorder homogeneous equation
h(n) + 2h(n 1) 8h(n 2) = 0,
h(1) = 0, h(0) = 2, n 1
(7.18)
with two initial conditions. The modes of the system can be easily found to be = 2 and = 4.
The general form of the homogeneous solution for all n is
h(n) = C1 2n + C2 (4)n
for some constants C1 and C2 that we need to determine. Using the initial conditions at times n = 0
and n = 1 we obtain the following linear system of equations in the unknowns {C1 , C2 }:
2 = C1 + C2 ,
0=
C1
C2
2
4
Solving we obtain C1 = 2/3 and C2 = 4/3. In this way, we arrive at the sequence
h(n) =
4
2 n
2 + (4)n
3
3
which satisfies the following homogeneous equation for all values of n (both positive and negative):
h(n) + 2h(n 1) 8h(n 2) = 0,
h(1) = 0, h(0) = 2
This is still not the sequence h(n) we are looking for. This is because the desired h(n) should be
zero for all n < 0 due to the assumed causality of the system. However, we can derive the desired
sequence by writing instead

h(n) =
4
2 n
2 + (4)n u(n)
3
3
(7.19)
where we introduced the the step-sequence to enforce the fact that h(n) = 0 for n < 0.
165
Example 7.6 (Mode cancelation)
SECTION 7.3
Let us now determine the impulse response sequence of a relaxed causal system that is described by
the difference equation
y(n) + 2y(n 1) 8y(n 2) = x(n) 2x(n 1)
This is again an LTI system. Since the system is causal and LTI, we already know that its impulse
response sequence, denoted by h(n), needs to satisfy
h(n) = 0
for n < 0
(7.20)
Moreover, h(n) satisfies the difference equation

h(n) + 2h(n 1) 8h(n 2) = (n) 2(n 1),
n0
(7.21)
We want to solve this equation and determine h(n) over n 0.

Since the input sequence is x(n) = (n), and the system is assumed to be relaxed, then we must
have h(1) = h(2) = 0. This is because, by the definition of a relaxed system, the output has
to remain at zero as long as the input stays at zero. In the present situation, the input sequence,
x(n) = (n), is zero for n < 0 and it moves away from zero only at time n = 0.
Now, the difference equation (7.21) for h(n) becomes homogeneous only for n 2. This is
because the input sequence combination
(n) 2(n 1)
becomes zero for all n 2. In this way, the difference equation becomes
h(n) + 2h(n 1) 8h(n 2) = 0
for n 2
In order to solve this second-order homogeneous difference equation, we need to determine two
initial conditions. We select the time instants n = 0 and n = 1 just prior to n = 2, when the
equation (7.21) becomes homogeneous. It follows from (7.21) that h(0) = 1 and h(1) = 4. We
are therefore reduced to solving the homogeneous equation
h(n) + 2h(n 1) 8h(n 2) = 0, h(0) = 1, h(1) = 4, for n 2
(7.22)
The general form of the homogeneous solution for all values of n is

h(n) = C1 2n + C2 (4)n
Using the initial conditions at times n = 0 and n = 1 we obtain the linear system of equations
1 = C1 + C2 ,
4 = 2C1 4C2
which gives C1 = 0 and C2 = 1. We thus arrive at the sequence

h(n) = (4)n
which satisfies the following homogeneous equation for all values of n (both positive and negative):
h(n) + 2h(n 1) 8h(n 2) = 0, h(0) = 1, h(1) = 4
This is still not the sequence h(n) we are looking for. This is because the desired h(n) should be
zero for all n < 0 due to the assumed causality of the system. We can derive the desired sequence
by writing instead
h(n) = (4)n u(n)
(7.23)
IMPULSE
RESPONSE
OF LTI
SYSTEMS
166
CHAPTER 7
Observe that in this case one of the modes does not appear in the expression for h(n) (since C1 = 0).
We therefore say that mode cancelation occurred.
HOMOGENEOUS
DIFFERENCE
EQUATIONS
Example 7.7 (Complex modes)

Let us now determine the impulse response sequence of the relaxed causal system
y(n) + 2y(n 1) + 2y(n 2) = x(n)
Since the system is causal and LTI, we already know that its impulse response sequence, denoted by
h(n), needs to satisfy
h(n) = 0 for n < 0
(7.24)
Moreover, h(n) satisfies the difference equation
h(n) + 2h(n 1) + 2h(n 2) = (n),
n0
(7.25)
We want to solve this equation and determine h(n) over n 0. As in the previous two examples,
the fact that the system is relaxed gives h(1) = h(2) = 0. Also, since (n) is zero for all n 6= 0,
the difference equation for h(n) becomes homogeneous over n 1:
h(n) + 2h(n 1) + 2h(n 2) = 0
for n 1
In order to solve this second-order homogeneous difference equation, we determine the initial conditions at times n = 1 and n = 0 from (7.25) as h(0) = 1 and h(1) = 0. We are thus reduced
to solving the homogeneous equation
h(n) + 2h(n 1) + 2h(n 2) = 0, h(1) = 0, h(0) = 1, n 1
(7.26)
The characteristic equation is given by

2 + 2 + 2 = 0
with complex roots at
= 1 + j =
2ej
3
4
= 1 j =
2ej
3
4
The general form of the homogeneous solution for all values of n is:
3
3
h(n) = C( 2)n ej 4 n + C ( 2)n ej 4 n
with C and its complex conjugate used as arbitrary constants. The initial conditions at times n = 0
and n = 1 lead to the equations
3
3
1 = C + C , 0 = 2Cej 4 + 2C ej 4
If we write C = a + jb, these equations collapse to 1 = 2a and a = b so that a = 1/2 and b = 1/2.
Hence,
1
2 j 4
e
C = (1 + j) =
2
2
And we conclude that the desired impulse response sequence is
h(n) = ( 2)

n+1
cos
3
(n + 1) u(n)
4
(7.27)
where we again added the step-sequence, u(n), to enforce the condition h(n) = 0 for n < 0.
167
7.4 STABILITY OF CAUSAL LTI SYSTEMS
SECTION 7.4
All three examples in Sec. 7.3 deal with causal LTI systems that are described by constantcoefficient difference equations, namely, with LTI systems that are described by equations
of the form:
M
N
X
X
y(n) =
ak y(n k) +
bk x(n k)
(7.28)
k=1
k=0
for some constant coefficients {ak , bk } and with time progressing forward. From the examples in Sec. 7.3, we observe the important fact that the impulse response sequence, h(n),
of every such system can always be expressed as a linear combination of sequences that
are defined by the modes of the system. Every mode o of multiplicity mo contributes to
the expression for h(n) up to mo terms that are of the form
n

o u(n), nno u(n), n2 no u(n), . . . , nmo 1 no u(n)
(7.29)
When all modes of the system are taken into account, the expression for h(n) will be a
linear combination of all such terms contributed by all modes. Assuming the system has a
total of L distinct modes { } with multiplicities {m } each, then the general expression
for h(n) would take the form:
h(n) =
L m
1
X
X
=1 m=0
Cm nm n u(n)
(7.30)
for some combination coefficients {Cm }. Usually, all the modes of the system would
appear in the expression for h(n), although sometimes mode cancelations may occur and
some of the coefficients Cm end up being zero, as was the case with Example 7.6.
Now recall from Sec. 5.3 that an LTI system is BIBO stable if, and only if, its impulse
response sequence is absolutely summable, namely, it should satisfy the condition
n=
|h(n)| < Bh <
for some finite positive scalar Bh . We would like to translate this condition into an equivalent statement in terms of the modes of the system. To do so, we first note that, for any o ,
each of the sequences below
n

o u(n), nno u(n), n2 no u(n), . . .
is known to be absolutely summable (also said to be absolutely convergent) if, and only if,
o is such that |o | < 1. That is, for any finite integer m 0, the following statement
holds:
X
|nm no | < |o | < 1
(7.31)
n=0
Proof: This argument can be skipped on a first reading (it requires familiarity with series and the
ratio convergence tests). One way to establish (7.31) is to invoke the so-called ratio test for checking
whether a series is absolutely convergent. Consider a generic sequence with terms {cn } for n 0.
The ratio test states that

cn+1

< 1 then
if lim
n
cn
X
n=0
|cn | <
STABILITY OF
CAUSAL LTI
SYSTEMS
168
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
In other words, if the limit of the ratio is strictly less than one, then the series is absolutely convergent.
Thus, consider the sequence {cn = nm n
o } for n 0 and for any finite integer m. It follows that

m
(n + 1)m n+1

n+1
o

lim
lim
= |o |
= |o | n
n
nm n
n
o
Therefore, the condition |o | < 1 is sufficient for the absolute convergence of the sequence {nm n
o}
over n 0. Conversely, assume that this sequence converges absolutely. Then this implies that its
individual terms tend to zero, namely,
lim nm n
o = 0
This is only possible if |o | < 1 so that the condition |o | < 1 is also necessary for absolute
convergence of the sequence.
Applying the result (7.31) to expression (7.30) for h(n), we conclude that h(n) will be
absolutely summable if, and only if, all modes } that appear in the expression for h(n)
have magnitudes strictly less than one; this is equivalent to saying that these modes should
lie inside the circle of unit radius in the complex plane. Observe that we are only requiring
the modes that appear in h(n) to lie inside the unit circle for stability to hold:
Causal LTI systems described
All modes { } that
by constant-coefficient difference
appear in h(n) satisfy
equations are BIBO stable

| | < 1
(7.32)
Proof: Assume initially that all modes that appear in expression (7.30) lie inside the unit circle,
| | < 1. Then we have
X
n=0
|h(n)|

L m
X
1

X
X

m
n
Cm n

n=0 =1
n=0 =1
m=0
L m
1
X
X
=1
<
m=0
X
L m
1
X
X
m=0
|Cm nm n
|
|Cm |
X
n=0
!
m
|n
n
|
since each of the sequences {nm n

} is absolutely convergent. It follows that h(n) is absolutely
summable and the system is BIBO stable. Conversely, assume that h(n) is absolutely summable.
This implies that the sequence h(n) tends to zero as n . This is only possible if all the that
appear in expression (7.30) lie inside the unit circle.
Example 7.8 (Stable LTI system)

Consider a causal system that is described by the constant-coefficient difference equation
y(n)
1
y(n 2) = x(n)
4
The modes of the system are the roots of the characteristic equation
2
1
=0
4
169
which are given by

1 = 1/2,
2 = 1/2
SECTION 7.5
Since both modes lie within the unit disc, the system is BIBO stable. This conclusion holds regardless
of the form of the input sequence that would appear on the right-hand side of the different equation.
In the above equation we have x(n). But had it been any combination of x(n) and delayed versions
of x(n), the same conclusion would still hold. For example, the following causal system
y(n)
1
y(n 2) = x(n) 2x(n 3)
4
is also BIBO stable for the same reason. This is because the modes of the system continue to be the
same and they lie inside the unit circle.
7.5 IMPULSES RESPONSE OF NON-LTI SYSTEMS

We can use the same technique described in Sec. 7.3 to find the response to the unit-sample
sequence, (n), of a constant-coefficient difference equation that is not necessarily relaxed
(and which therefore does not describe an LTI system).
Example 7.9 (Non-LTI system)
Consider a system that is described by the difference equation
y(n) + 2y(n 1) 8y(n 2) = x(n) 2x(n 1)
with initial conditions y(2) = 0 and y(1) = 1. We want to find its response to x(n) = (n) for
all n (i.e., for both n 0 and n < 0). This system is not relaxed. For example, even when the input
sequence is chosen to be x(n) = 0 for all n, the system will exhibit a nonzero output sequence y(n)
due to the initial condition y(1) = 1. Therefore, the given difference equation does not describe
an LTI system; recall that constant-coefficient difference equations need to be relaxed in order to
describe LTI systems.
Still, we can proceed to find the response of the system to the unit-sample sequence x(n) = (n).
Thus, note that the equation becomes homogeneous for n 2. If we denote the impulse response
sequence by h(n), then h(n) satisfies the relation
h(n) + 2h(n 1) 8h(n 2) = 0,
over n 2
with the same initial conditions h(2) = 0 and h(1) = 1, as the given system. To solve the above
homogeneous equation in h(n), we first need to propagate the initial conditions to the time instants
n = 0 and n = 1 to find h(0) = 1 and h(1) = 8; these initial conditions incorporate the effect of
the input combination (n) 2(n 1) at times n = 0 and n = 1. We are reduced t solving the
homogeneous equation
h(n) + 2h(n 1) 8h(n 2) = 0, h(0) = 1, h(1) = 8, for n 2
(7.33)
The general form of the homogeneous solution for all n is given by

h(n) = C1 2n + C2 (4)n
We now use the initial conditions at times n = 0 and n = 1 to determine the constants {C1 , C2 }
that would describe h(n) over the interval n 0. Doing so leads to the equations
1 = C1 + C2 ,
4 = C1 2C2
IMPULSE
RESPONSES OF
NON-LTI
SYSTEMS
170
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
so that C1 = 2/3 and C2 = 5/3. Therefore, the response of the system to x(n) = (n) is
h(n) =
2 n
5
2 (4)n ,
3
3
n0
(7.34)
We still need to find the form of h(n) over n < 0. For this purpose, we use instead the initial
conditions h(2) = 0 and h(1) = 1 (which do not incorporate the influence of the input excitation
(n)2(n1)) and solve for the constants {C1 , C2 }. Doing so leads to C1 = 2/3 and C2 = 8/3
and we conclude that the response of the system over n < 0 is given by
h(n) =
2 n
8
2 (4)n ,
3
3
n<0
(7.35)
7.6 COMPLETE RESPONSE OF LTI SYSTEMS

Clearly, the impulse response sequence of a system is most useful when the system is LTI
since it would then allow us to determine the response of the system to arbitrary input
sequences via convolution. Given that we now know how to find the impulse response
sequence of an LTI system that is described by a constant-coefficient difference equation,
we can therefore utilize the following two-step procedure to determine the response of such
systems to any other input sequence:
(a) First, determine the impulse response, h(n), as described in Sec. 7.3.
(b) Then, evaluate the convolution of h(n) with the given input sequence x(n) to find
the output sequence y(n).
Example 7.10 (Finding a complete response)

Let us determine the step-response of the relaxed and causal system that is described by the difference
equation
y(n) + 2y(n 1) 8y(n 2) = x(n) 2x(n 1)
Since the system is relaxed, the difference equation describes an LTI system. Hence, the stepresponse can be found by first determining the impulse response sequence and then convolving it
with the step sequence, u(n).
We already found the impulse response sequence of this system in Example 7.6, namely,
h(n) = (4)n u(n)
Now we convolve it with u(n) to find
y(n)
=
=
(4)n u(n) u(n)
X
k=
n
X
(4)k u(k)u(n k)
(4)k
k=0
Obviously, the sum evaluates to 0 when n < 0 so that y(n) = 0 for n < 0. On the other hand, the
sum contains non-trivial terms for n 0 in which case we get
1 (4)n+1
y(n) =
,
5
n0
We can combine the results for n < 0 and n 0 into a single expression and write
y(n) =

1
1 (4)n+1 u(n)
5
(7.36)
The fact that the sequence y(n) is zero for negative time in response to x(n) = u(n) also follows
from the fact that the system is relaxed and causal.
7.7 APPLICATIONS
the context of practical problems.
7.7.1 Carbon Dating

In chemical reactions, radioactivity refers to the change of one element into another mainly
through the emission of radioactive particles, such as (alpha) or (beta) particles. Alpha
particles consist of two protons and 2 neutrons each bound together (similar to Helium
nuclei) and beta particles consist of electrons. The half-life of a radioactive element is
defined as the time it takes for half of the sample size of the element to decay. One important radioactive element is carbon-14 (written as 14 C; its nucleus contains 6 protons and
8 neutrons). 14 C occurs in tiny traces in the environment and corresponds to only about
1 part per trillion of all the carbon present in the atmosphere; the other two isotopes of
carbon are far more abundant (with 12 C corresponding to 99% of all carbon and 13 C corresponding to the remaining 1%). All three isotopes of carbon have the same number of
protons (6 per atom) and differ in their number of neutrons (8, 7, and 6 in 14 C, 13 C, and
12
C, respectively).
FIGURE 7.1 Carbon dating is used to estimate the age of historical artifacts.
171
SECTION 7.7
APPLICATIONS
172
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
Although 14 C is not strongly radioactive, its level of decay is still detectable by modern
techniques to enable its use as the basis for the process known as carbon dating. By
measuring the level of radioactive activity in 14 C, it is possible to estimate the age of
historical remains for the following reason. During the photosynthesis process, plants
absorb CO2 from the atmosphere and incorporate a quantity of 14 C into their living tissues
at an amount that approximately matches the percentage of 14 C present in the atmosphere.
After a plant or an organism dies, the amount of 14 C continues to decay in the buried
matter without being replaced. By measuring the remaining radiation level in a historical
artifact, and comparing it with the radiation level in a living organism, it becomes possible
to approximate the age of the artifact see Fig. 7.1.6 The half-life of 14 C is T1/2 = 5730
years. This means that if we start with a certain amount of 14 C, then half of this amount
will be present 5730 years later. The amount of 14 C continues to be halved in this manner
every 5730 years.
Let y(0) = ro denote the radiation level of one gram of 14 C in a living material (a plant
or an organism) right before it dies. This radiation level can be measured in terms of the
number of atom disintegrations that occur per minute (dpm). It is know that the radiation
level of 14 C is in living organisms is about 14dpm. In writing y(0), we are using n = 0
to denote the origin of time. More generally, we write y(n) to denote the radiation level at
year n. Assume that, N years later, a piece of wood is recovered from an ancient tomb that
has been contaminated by the dead plant or organism. Assume further that we measure
the radiation of carbon at that point in time. Thus, let y(N ) = rN denote the radiation
level measured in one gram of 14 C found in the piece of wood. Given knowledge of ro and
T1/2 for 14 C, and measurement of rN , we would like to use this information to estimate
N (which in turn would help us determine the age of the piece of wood).
Let us write down a model for the radioactive decay of 14 C. Starting at n = 0, the
radiation level at any subsequent year, n, will be modeled, to a reasonable extent, by a
first-order difference equation of the form:

y(n 1), y(0) = ro , n 0
y(n) = 1
100
(7.37)
where the parameter % denotes the percentage by which the radiation level of 14 C decays
every year; this quantity is not known. However, the argument below will show that it can
be inferred from knowledge of T1/2 see (7.40). We are only interested in the evolution
of the radioactivity level over the interval of time n 0.
Recursion (7.37) is a homogeneous difference equation with a single mode at
=1
100
(7.38)
This mode controls the exponential rate of decay of the radioactivity level of 14 C. Indeed,
from the result (7.2), we already know that the form of the general solution of the difference
equation over n 0 is given by

n
, n0
y(n) = C 1
100
The constant C is determined from the initial condition y(0) = ro so that C = ro and

n
, n0
y(n) = ro 1
100
6 The
source for this image is Wikimedia Commons.
(7.39)
This recursion models how the radiation level of 14 C decays with time. We do not know the
value of but can determine it from knowledge of the half-time of 14 C. Since the radiation
level of 14 C is halved every T1/2 = 5730 years, we can set n = T1/2 and y(T1/2 ) = ro /2
into the above equation to get

T1/2
1
ro = ro 1
2
100
or, equivalently,
1
= 10
100
0
1
log(1/2)

A
T1/2
(7.40)
This relation shows how the mode of the difference equation (7.37) is determined by T1/2 ;
the relation can be used to determine from knowledge of T1/2 and vice-versa.
For carbon-dating the piece of wood, we would like to determine the value of N that
results in y(N ) = rN . Substituting into (7.39), and using (7.37), we get
rN
log(1/2) N

N
= ro 10 T1/2
ro 1
100
Solving for N we arrive at the following expression

N = 3.2193 T1/2 log
ro
rN
(7.41)
This result tells us how to estimate the age, N , of the artifact, from knowledge of {ro , rN , T1/2 }.
Let us use the following numerical values for the piece of wood:
T1/2 = 5730 years, ro = 14 dpm, rN = 11 dpm
Substituting into the expression for N gives
N = 2695 years old
(7.42)
Practice Questions:
1. For the same numerical values used above, at what rate does the carbon-14 isotope decay
per year?
2. Find the numerical value of the mode of the difference equation (7.37).
3. Carbon dating of a mummy resulted in a measurement of rN = 9.5 dpm. What is the age of
the mummy?
4. The radioactive element radon-222 decays at the rate of 16.8% per day. Find its half-life in
days.
7.7.2 Rabbit Population and Fibonacci Numbers

Let us start with a pair of rabbits and assume that each pair bears a new pair every month.
A month later, the new pair of rabbits becomes productive, and so on. We would like to
examine the evolution of the number of pairs of rabbits with time.
173
SECTION 7.7
APPLICATIONS
174
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
Let y(n) denote the number of rabbits at the start of month n, with n = 0 taken as the
origin of time when the original pair of rabbits is introduced. Assuming the original pair
of rabbits at n = 0 is young and becomes productive a month later at n = 1, we have
that y(0) = 1 and y(1) = 1. The pair of rabbits born at month n = 1 will only become
productive later at the start of month n = 2. Table 7.1 illustrates the evolution of the
rabbit population with the symbol denoting a productive pair of rabbits and the symbol
denoting an unproductive pair of rabbits.
TABLE 7.1 Evolution of the rabbit population over the first 5 months starting with one pair of
young rabbits at month n = 0.
y(n)
n=0
n=1
n=2
n=3
n=4
n=5
The resulting sequence of integers for y(n)

{1, 1, 2, 3, 5, 8, 13, 21, 34, 55, . . .}
(7.43)
are called the Fibonacci numbers, where each subsequent number is the sum of the previous
two numbers see Fig. 7.2.
FIGURE 7.2
The rabbit population evolves over consecutive generations according to the
sequence of Fibonacci numbers (7.43).
It can be seen that the quantity y(n) evolves with time according to the second-order difference equation
y(n) = y(n 1) + y(n 2), y(0) = 1, y(1) = 1, n 0
(7.44)
Let us determine a closed-form expression for y(n) using the techniques developed in this
chapter. The characteristic equation is
2 1 = 0
with modes at
1 5
1+ 5
, 1 =
1 =
2
2
The general form of the solution is therefore
"
!n
!n #
1+ 5
1 5
y(n) = C1
u(n)
+ C2
2
2
where we added the step-sequence, u(n), to enforce the condition y(n) = 0 for all n < 0.
We can determine the constants {C1 , C2 } from the initial conditions. Using y(0) = 1 =
y(1) we find that C1 and C2 should satisfy
1 5
1+ 5
+ C2
1 = C1 + C2 and 1 = C1
2
2
Solving we get
5 5
5+ 5
, C2 =
C1 =
10
10
so that
5+ 5
y(n) =
10
"
!n
5 5
1+ 5
+
2
10
!n #
1 5
u(n)
2
(7.45)
This result provides a closed-form expression for the Fibonacci numbers and, therefore,
for the number of rabbit pairs as a function of n. Despite the awkward looking expression
for y(n) above, the terms on the right-hand side always add up to an integer number.
Practice Questions:
1. Define the ratio r(n) = y(n)/y(n + 1). Show that it satisfies the recursion
r(n) =
1
,
1 + r(n 1)
r(0) = 1
2. Verify that the limit of r(n) as n is given by the so-called (inverse) golden ratio:
51
lim r(n) =
n
2
3. Consider a segment AB of length 1 and divide it into two segments AC of length x and CB
of length 1 x with AC denoting the longer segment. Determine the value of x such that the
ratio of the shorter segment to the longer segment equals the ratio of the longer segment to the
whole.
175
SECTION 7.8
APPLICATIONS
176
7.8 PROBLEMS
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
Problem 7.1 Find the modes of the LTI systems:

(a) y(n) + y(n 1) 5y(n 2) = x(n).
(b) y(n) = 4y(n 2) + x(n 1).

(c) y(n) 4y(n 2) = 2x(n).
Problem 7.2 Find the modes of the LTI systems:

(a) y(n) = y(n 2) + x(n) + x(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1).
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n).
Problem 7.3 Give an example of a causal LTI system whose modes are at 1 = 1/2, 2 = 1/3,
3 = 21 (1 + j), and 4 = 12 (1 j).
Problem 7.4 Give an example of a causal LTI system whose modes are at 1 = 1/4, 2 =
1
(1 + j), and 3 = 31 (1 j).
3
Problem 7.5 Describe all solutions to the following homogeneous equations:
(a) y(n) + y(n 1) 5y(n 2) = 0.
(b) y(n) = 4y(n 2).
(c) y(n) 4y(n 2) = 0.
Problem 7.6 Describe all solutions to the following homogeneous equations:

(a) y(n) = y(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 0.
(c) y(n) y(n 1) + y(n 2) y(n 3) = 0.
Problem 7.7 Determine the solution of each of the following homogeneous equations with initial
conditions:
(a) y(n) + y(n 1) 5y(n 2) = 0, y(0) = 0, y(1) = 1.
(b) y(n) = 4y(n 2), y(2) = 1, y(1) = 0.

(c) y(n) 4y(n 2) = 0, y(2) = 1, y(3) = 0.
conditions:
(a) y(n) = y(n 2), y(0) = 0, y(1) = 3.
(b) y(n) 6y(n 1) + 9y(n 2) = 0 y(1) = 2, y(0) = 1.
(c) y(n) y(n 1) + y(n 2) y(n 3) = 0 y(0) = 0, y(1) = 1, y(1) = 2.
conditions:
(a) y(n) + 9y(n 2) = 0, y(0) = 0, y(1) = 1.
(b) y(n) + 9y(n 2) = 0, y(0) = 0, y(1) = j.
conditions:
(a) y(n) + y(n 2) = 0, y(0) = 1, y(1) = 0.
(b) y(n) + y(n 2) = 0, y(0) = j, y(1) = 0.
Problem 7.11 Find the impulse-response sequences of the following causal LTI systems:
(a) y(n) + y(n 1) 5y(n 2) = x(n).
(b) y(n) = 4y(n 2) + x(n 1).

(c) y(n) 4y(n 2) = 2x(n).
Problem 7.12 Find the impulse-response sequences of the following causal LTI systems:
(a) y(n) = y(n 2) + x(n) + x(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1).
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n).
Problem 7.13 Find the step-response sequences of the following causal LTI systems:
(a) y(n) + y(n 1) 5y(n 2) = x(n).
(b) y(n) = 4y(n 2) + x(n 1).

(c) y(n) 4y(n 2) = 2x(n).
Problem 7.14 Find the step-response sequences of the following causal LTI systems:
(a) y(n) = y(n 2) + x(n) + x(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1).
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n).
Problem 7.15 Verify which each of the following causal LTI systems is BIBO stable:
(a) y(n) + y(n 1) 5y(n 2) = x(n).
(b) y(n) = 4y(n 2) + x(n 1).

(c) y(n) 4y(n 2) = 2x(n).
Problem 7.16 Verify which each of the following causal LTI systems is BIBO stable:
(a) y(n) = y(n 2) + x(n) + x(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1).
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n).
Problem 7.17 Assume the difference equation below describes a causal and relaxed system
y(n) +
5
1
y(n 1) y(n 2) = x(n)
6
6
where the time index runs forward.

(a) Find its modes.
(b) Find its impulse-response sequence.
(c) Find its response to x(n) =

1 n
2
n
u(n).
(d) Find its response to x(n) = 2 u(n).

Problem 7.18 Assume the difference equation below describes a non-causal and relaxed system
y(n) +
1
5
y(n 1) y(n 2) = x(n)
6
6
where the time index runs backwards.

(a) Find its modes.

1 n
2
n
u(n).
(d) Find its response to x(n) = 2 u(n).
177
SECTION 7.8
PROBLEMS
178
Problem 7.19 Consider the relaxed and causal system
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
5
1
y(n 1) y(n 2) = x(n 2)
6
6
y(n) +
(a) Is the system BIBO stable?

1 n
2
u(n 2).
Problem 7.20 A causal and relaxed system is described by the difference equation
y(n)
5
y(n 1) + y(n 2) = x(n 1) 2x(n 2)
2
Is the system BIBO stable? Find its step response.

y(n)
1
1
2
y(n 1)
y(n 2) +
y(n 3) = x(n)
3
12
12
(a) Find the modes of the system.

(b) Find its impulse response sequence. Is the system BIBO stable?
(c) Assume the initial conditions of the difference equation are set to y(3) = y(2) = 0 and
y(1) = 1. Find the response of the system when x(n) = 0 for all n.
7
2
y(n 1) + y(n 2) = x(n)
3
3
y(n)
(b) Find its impulse response sequence. Is the system BIBO stable?
(c) Assume the initial conditions are set to y(1) = and y(2) = . Find {, } for which
only one of the modes will be present in the response of the system to x(n) = 0 for all n.
Problem 7.23 Give an example of a causal and BIBO stable LTI system whose modes are at 1 =
1/2 and 2 = 2.
Problem 7.24 Give an example of a causal and BIBO stable LTI system whose modes are at 1 =
1/4, 2 = 1/3, and 3 = 2
Problem 7.25 Write a constant-coefficient homogeneous difference equation, with initial conditions, whose solution is given by
n
y(n) =
1
3
n+1
1
2
for all n.
n1
y(n) =
for all n.
1
3
n+2
1
2
n
2n
1
2
n
y(n) =
1
3
n
1
2
PROBLEMS

n+
cos
3
4
for all n.
n
y(n) =
n
1
3
1
2
sin
n+

4
for all n.
Problem 7.29 The impulse-response sequence of a causal and relaxed LTI system is given by
n1
h(n) =
1
3
n+2
1
2
u(n) +
n
u(n) 2n
1
2
u(n)
Use the techniques of this chapter to deduce a description for the system in terms of a constantcoefficient difference equation.
Problem 7.30 The impulse-response sequence of a causal and relaxed LTI system is given by
n1
h(n) =
1
3
n+2
u(n 3)
1
2
u(n 4)
Use the techniques of this chapter to deduce a description for the system in terms of a constantcoefficient difference equation.
Problem 7.31 Find the solution y(n) of the so-called Fibonacci difference equation
y(n) y(n 1) y(n 2) = 0
with y(1) = 0 and y(0) = 1.
Problem 7.32 Use the techniques of this chapter to find the solution of the following difference
equation
y(n) +
5
1
y(n 1) y(n 2) =
6
6
n
1
2
u(n), y(2) = 0, y(1) = 0, n 0
Problem 7.33 Find the response

n of the following causal and relaxed system when the input sequence is given by x(n) = 12 u(n 3):
y(n) +
1
5
y(n 1) y(n 2) = x(n 2) + (n 1)u(n), n 0
6
6
Problem 7.34 Find the response of the following causal and relaxed system when x(n) is the
sequence depicted in Fig. 7.3:
y(n) +
179
SECTION 7.8
5
1
y(n 1) y(n 2) = x(n 2) + (n 1)u(n), n 0
6
6
Problem 7.35 Given an example of a causal LTI system that is described by a constant-coefficient
difference equation with modes at 1 = 1/2, 2 = 1/3, 3 = 1/3, and 4 = 0.
difference equation with modes at 1 = 1/2, 2,3 = (1 j)/2, and 4 = 1/4.
180
CHAPTER 7
HOMOGENEOUS
DIFFERENCE
EQUATIONS
x(n)
4
3
2
1
1
2
FIGURE 7.3 Input sequence for Prob. 7.34.
difference equation with modes at 1 = 1/2, 2 = (1 + j)/3, and 3 = (2 + j)/4.
difference equation with modes at 1 = 1/2, 2 = (1 + j)/3, and 3 = (2 + j)/4.
difference equation with modes at 1 = 1/2, 2 = 1/4, and 3 = 1/3 and where mode cancellation occurs at 2 = 1/4.
difference equation with modes at 1 = 1/2, 2 = 1/3, and 3 = 1/4 and where mode
cancellation occurs at 2 = 1/3.
CHAPTER
ur objective from the discussions in Chapters 7 and 8 is to end up with a procedure

that would enable us to determine closed-form expressions for the solutions of constantcoefficient difference equations in response to certain types of input sequences without the
need for computing convolutions (as we did in Sec. 7.6). We shall henceforth focus on
the common situation of causal input sequences, whose samples are zero over n < 0, and
on the case of difference equations that run forward in time. For example, by the end of
the current chapter, we will be able to determine the solution over n 0 of a difference
equation of the form
y(n) 2y(n 1) + 4y(n 2) = x(n), y(1) = 1, y(2) = 0, n 0
(8.1)
with knowledge of both the initial conditions and the causal input sequence x(n). The technique described earlier in Sec. 7.6 does not apply to this case since the system described by
the above difference equation is not LTI any longer. In such a case, we would not be able
to determine the complete solution, y(n), by convolving x(n) with the impulse response
sequence of the LTI system that would correspond to the relaxed difference equation.
Thus, in this chapter, we shall motivate a systematic procedure for determining the
response of systems described by constant-coefficient difference equations. Alternative
procedures that employ transform techniques will be described in later chapters. While the
method of computation in this chapter involves straightforward calculations, what is likely
to confuse the reader are the various terms used to refer to different stages of the solution.
For example, by the end of this chapter, the reader will be exposed to each of the terms
listed in Table 8.1 and their meanings. The steps for finding each of the solutions listed in
Table 8.1 will be straightforward once the reader understands what each term means.
TABLE 8.1
Terminologies for important solutions associated with difference equations.

particular solution
homogeneous solution
complete solution
transient solution
steady-state solution
zero-input solution
zero-state solution
forced solution
unforced solution
natural solution
Before we proceed, we would like to re-emphasize that there is one important special
case for which we already know how to determine the complete solution of a constantcoefficient difference equation, as was described in Sec. 7.6. In that section we showed
that when the difference equation describes an LTI system, then we can first determine
the impulse response sequence of the system by solving a homogeneous equation and
subsequently convolve it with the given input sequence.
181
Copyright
182
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DIFFERENCE
EQUATIONS
In this chapter, however, we are interested in the more general case in which the difference equation need not describe an LTI system, for example, when the difference equation
has nonzero initial conditions. The techniques developed in this chapter can be applied to
both LTI and non- LTI systems. In the case of LTI systems, the techniques of this chapter
will provide an alternative to the two-step procedure of Sec. 7.6; in particular, they can
help avoid some of the effort that goes into computing the impulse response sequence and
the subsequent convolution operation.
In the first part of this chapter we develop a procedure for determining the complete response of constant-coefficient difference equations for a restricted (yet important) subclass
of input sequences. Towards the end of the chapter, we present a procedure for general
input sequences; the procedure will rely on the use of convolution and on the concept of
the zero-state response.
Useful Causal Input Sequences
For the most part in this chapter, we focus on causal input sequences of the four types
listed in Table 8.2. Later, when we introduce the z-transform technique starting in Ch. 9,
we shall be able to solve constant-coefficient difference equations for a broader set of input
sequences.
TABLE 8.2 Useful types of causal input sequences.
x(n) = Au(n)
x(n) = An u(n)
x(n) = A cos(o n)u(n)
x(n) = A sin(o n)u(n)
x(n) = An cos(o n)u(n)
x(n) = An sin(o n)u(n)
x(n) = Anp u(n)
x(n) = Anp n u(n)
step-sequence
exponential sequence
sinusoidal sequence
sinusoidal sequence
sinusoidal sequence with exponential modulation
sinusoidal sequence with exponential modulation
polynomial sequence, where p 0 is an integer
polynomial sequence with exponential modulation
8.1 PARTICULAR SOLUTION

We start by defining what is meant by a particular solution to a constant-coefficient difference equation, such as the one described by (8.1). Specifically, a particular solution
of a constant-coefficient difference equation is one sequence (from many others that are
possible; hence, the designation particular) that satisfies the difference equation for the
given input sequence and over an interval of time of the form n no , for some no 0.
No initial conditions enter into the determination of a particular solution.
We denote particular solutions by yp (n). Referring to (8.1), assume we select the input
sequence to be x(n) = (0.5)n u(n). Then, a particular solution would be any sequence
yp (n) that satisfies
yp (n) 2yp (n 1) + 4yp (n 2) = (0.5)n u(n)
over some interval n no to be specified. When the input sequence, x(n), belongs to the
class of sequences listed in Table 8.2, there is a useful procedure for determining yp (n) and
no . In each case, the particular solution is essentially assumed to have a similar form to
the input sequence and is selected according to the construction outlined in Table 8.3. For
example, the first line of the table states that if x(n) is a multiple of the step sequence, then
we assume a similar form for yp (n) for some constant K to be determined. Likewise, the
second line of the table states that if x(n) is an exponential sequence, then the same form
is selected for yp (n). On the other hand, when x(n) is a sinusoidal sequence, the third
line of table states that we should yp (n) as a combination of sine and cosine sequences
for some constants {K1 , K2 } to be determined. The next to last line of the table states
that we should select yp (n) to be a linear combination of powers of n when x(n) is of the
form np u(n), and so on. The multiplication constants K or {K } are determined from the
difference equation, as we illustrate in the sequel with several examples.
TABLE 8.3 Assumed forms for the particular solution.
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
Input sequence, x(n)
Particular solution, yp (n)
Au(n)
An u(n)
A cos(o n)u(n)
A sin(o n)u(n)
An cos(o n)u(n)
An sin(o n)u(n)
Anu(n)
An2 u(n)
Anp u(n)
Anp n u(n)
Ku(n)
Kn u(n)
[K1 cos(o n) + K2 sin(o n)]u(n)
[K1 cos(o n) + K2 sin(o n)]u(n)
n [K1 cos(o n) + K2 sin(o n)]u(n)
n [K1 cos(o n) + K2 sin(o n)]u(n)
[K1 n + K2 ]u(n)
[K1 n2 + K2 n + K3 ]u(n)
[K1 np + K2 np1 + . . . + Kp+1 ]u(n)
n [K1 np + K2 np1 + . . . + Kp+1 ]u(n)
Example 8.1 (Exponential sequence)

Let us determine a particular solution for the difference equation
y(n) + 2y(n 1) 8y(n 2) = 2x(n 1)
when x(n) = (0.5)n u(n) and time runs forward (i.e., the difference equation describes a causal
system). In this case, we use the second line of Table 8.3 and select an exponential form for the
particular solution, say,
yp (n) = K(0.5)n u(n)
for some constant K to be determined. Substituting this choice back into the difference equation we
find that the following relation must hold for all n in order for this selection of yp (n) to satisfy the
equation:
K(0.5)n u(n) + 2K(0.5)n1 u(n 1) 8K(0.5)n2 u(n 2) = 2(0.5)n1 u(n 1)
We need to determine the value of K. Thus, note that none of the terms in the above equation vanish
for n 2. Evaluating both sides of the equation at any n 2, we find that K must always satisfy
the relation:
K
+ K 8K = 1
4
which leads to K = 4/27. Therefore, the particular solution is
yp (n) =
4
(0.5)n , n 2
27
183
SECTION 8.1
PARTICULAR
SOLUTION
184
CHAPTER 8
SOLVING
DIFFERENCE
EQUATIONS
and this choice solution is valid over n 2 (so that no = 2). What this means is that the sequence
yp (n) so determined satisfies the difference equation for all values of n 2 and for the given
sequence, x(n) = (0.5)n u(n).
Example 8.2 (Polynomial sequence)

y(n) + 2y(n 1) = x(n)
when x(n) = nu(n) and time runs forward (i.e., the difference equation describes a causal system).
Since the input sequence is of polynomial type, then according to the last line in Table 8.3 we assume
a similar form for the particular solution, namely,
yp (n) = [K1 n + K2 ]u(n)
for some constants K1 and K2 to be determined. Substituting this selection into the difference
equation we find that the following relation must be satisfied in order for the choice of yp (n) to be a
viable solution:
K1 nu(n) + K2 u(n) + 2K1 (n 1)u(n 1) + 2K2 u(n 1) = nu(n)
Note that none of the terms in the above equality vanish for n 1. Evaluating at any n 1, we get
that K1 and K2 must satisfy
3K1 n + (3K2 2K1 ) = n
Equating the coefficients of powers of n on both sides we find that K1 = 1/3 and K2 = 2/9.
Therefore, the particular solution is given by

yp (n) =
2
1
n+
,
3
9
n1
and this choice is valid over n 1 and for the input sequence x(n) = nu(n).
Example 8.3 (Sinusoidal sequence)

y(n) + 2y(n 1) 8y(n 2) = 2x(n 1)
when x(n) = cos( 3 n)u(n) and time runs forward. In this case, we use the third line of Table 8.3
and select a particular solution of the form:
yp (n) = K1 cos

n u(n) + K2 sin

n u(n)
for some constants K1 and K2 to be determined. Substituting this choice back into the difference
equation we find that the following relation must hold for all n in order for this selection of yp (n) to
satisfy the equation:
K1 cos

n u(n) + K2 sin

n u(n) + 2K1 cos
(n 1) u(n 1) 8K1 cos
8K2 sin
(n 2) u(n 2) = 2 cos
2K2 sin
(n 1) u(n 1) +

(n 2) u(n 2)
(n 1) u(n 1)
Note that the step-sequences on both sides of the equality are nonzero for all n 2. Evaluating both
sides of the equality for any n 2 shows that K1 and K2 must satisfy:

K1 cos
n + K2 sin
n + 2K1 cos
(n 1) + 2K2 sin
(n 1)
3
3
3
3
8K1 cos
(n 2)
3
Using the trigonometric identities
8K2 sin
(n 2)
= 2 cos
(n 1)
cos(a b)
cos a cos b + sin a sin b
(8.2)
sin(a b)
=
=
sin a cos b cos a sin b
(8.3)
cos( a)
sin( a)
sin(a)
(8.5)
cos(a)
(8.4)
we can expand the terms of the previous equality and reduce it to the following equivalent form:

n
n
+ K2 sin
3
3

n
n
K1 cos
+ K1 3 sin
3
3
n
n
K2 sin
K2 3 cos
3
3
n
n
4K1 cos
4 3K1 sin
3
3
n
n
4K2 sin
+ 4 3K2 cos
3
3
n
n
cos
+ 3 sin
3
3
K1 cos
+
+
+
Grouping terms we have that K1 and K2 must satisfy

n

n
n
n
6K1 + 3 3K2 cos

3 3K1 + 2K2 sin
= cos
+ 3 sin
3
3
3
3
Equating the coefficients of cos() and sin() on both sides we find that K1 and K2 must satisfy the
relations
6K1 + 3 3K2 = 1 and 3 3K1 + 2K2 = 3
which lead to K1 = 11/15 and K2 = 3 3/5. Therefore, the particular solution is

yp (n) =

11
3 3
cos
n u(n) +
sin
n , n2
15
3
5
3
and this choice solution is valid over n 2.
Example 8.4 (Failure of procedure)
The procedure illustrated in the previous two examples can fail in some situations. Consider for
example the difference equation
y(n) 2y(n 1) + y(n 2) = x(n)
and let us try to determine its particular solution when x(n) = u(n). If we let yp (n) have the form
Ku(n) for some K, as suggested by Table 8.3, then the following equality must hold:
Ku(n) 2Ku(n 1) + Ku(n 2) = u(n)
185
SECTION 8.1
PARTICULAR
SOLUTION
186
CHAPTER 8
SOLVING
DIFFERENCE
EQUATIONS
This equation cannot be satisfied by any choice of K for any value of n 2 since we would get the
impossible relation
K 2K + K = 1
(impossible equality)
The difficulty in this case lies in the fact that the form we adopted for the particular solution, yp (n) =
Ku(n), also happens to be a solution to the homogeneous equation
y(n) 2y(n 1) + y(n 2) = 0
In such situations, the procedure we described can be adjusted in order to yield a correct particular
solution. However, we shall avoid pursuing such details here. This is because the z-transform technique that we shall introduce in later chapters will provide a powerful alternative method for solving
constant-coefficient different equations under varied conditions and for varied input sequences, not
just the ones listed in Table 8.3.
8.2 CHARACTERIZING ALL SOLUTIONS

Once we know how to find a particular solution for a constant-coefficient difference equation, we can then proceed to characterize all sequences that satisfy the same equation.
Actually, there are infinitely many such sequences and this is how we find them:
(a) First, we determine a particular solution, yp (n), by following the procedure explained in the previous section. As illustrated in the examples, the particular solution
will generally hold over an interval of the form n no , for some no 0.
(b) Next we determine the general form of the corresponding homogeneous solution.
Let us denote this form by yh (n). The homogeneous solution will satisfy the homogeneous equation for all n.
(c) If we now add the sequences yp (n) and yh (n), as
yc (n) = yp (n) + yh (n)
then yc (n) is a sequence that satisfies the same difference equation as the particular
solution. It will further hold, as we shall establish further ahead, that the sequences
{yc (n)} so constructed describe all possible solutions to the difference equation over
the interval n no .
The following example illustrates the procedure.
Example 8.5 (Describing all solutions)

Consider the difference equation
y(n) 0.5y(n 1) = u(n)
with time running forward. A particular solution of the form yp (n) = Ku(n) can be found by
determining the value of K from the equality:
Ku(n) 0.5Ku(n 1) = u(n)
This equality holds for all n 1 if we select K = 2. Moreover, we already know from Sec. 7.2 that
all solutions of the homogeneous equation
y(n) 0.5y(n 1) = 0
187
have the form

yh (n) = C(0.5)n ,
for arbitrary constants C
SECTION 8.3
CHARACTERIZING
ALL
SOLUTIONS
Therefore, all solutions that satisfy the original difference equation are given by
yc (n) = C(0.5)n + 2, n 1
Thus, observe that yc (n) describes a class of solutions: one sequence for every choice of C.
Let us now justify the following claim. Given a constant-coefficient difference equation,
the construction
yc (n) = yp (n) + yh (n)
(8.6)
describes all solutions to the equation for all n no , where n no is the range over
which the particular solution is valid.
Proof: Assume to the contrary that there is a solution to the difference equation over n no that is
not captured by the representation (8.6). Let us denote this solution by yc (n). Now pick any solution
yc,o (n) from the set (8.6), say,
yc,o (n) = yp (n) + yh,o (n)
for some homogeneous sequence yh,o (n). Then both sequences, yc,o (n) and yc (n), satisfy the
difference equation for n no . Consequently, their difference satisfies the homogeneous difference
equation for n no . Indeed, assume for illustration purposes, that the difference equation is secondorder and of the form
y(n) a1 y(n 1) a2 y(n 2) = x(n)
Then, since, by assumption, the sequence yc (n) satisfies the difference equation over n no we
have
yc (n) a1 yc (n 1) a2 yc (n 2) = x(n), n no
(8.7)
Likewise, since yc,o (n) satisfies the difference equation over n no , we must have
yc,o (n) a1 yc,o (n 1) a2 yc,o (n 2) = x(n),
n no
(8.8)
Subtracting (8.7) and (8.8) over n no we find that

[
yc (n) yc,o (n)] a1 [
yc (n 1) yc,o (n 1)] a2 [
yc (n 2) yc,o (n 2)] = 0, n no
so that the difference, yc (n) yc,o (n), satisfies the homogeneous equation
y(n) a1 y(n 1) a2 y(n 2) = 0,
n no
This argument can be easily extended to more general constant-coefficient difference equations.
Therefore, we must have that the difference, yc (n) yc,o (n), is some homogeneous solution and we
denote it by yh,1 (n):
yc (n) yc,o (n) = yh,1 (n), n no
It follows that we can express yc (n) over the interval n no in the following form:
yc (n)
yc,o (n) + yh,1 (n)
yp (n) + [yh,o (n) + yh,1 (n)]
188
CHAPTER 8
But since the sum of two homogeneous solutions is also a homogeneous solution, we conclude that
yc (n) belongs to the same set (8.6).
SOLVING
DIFFERENCE
EQUATIONS
8.3 FIRST METHOD FOR FINDING COMPLETE SOLUTIONS

The procedure described so far can be used to determine the complete response of systems
that are characterized by constant-coefficient difference equations with initial conditions.
These systems are not necessarily LTI. As indicated in the introduction of the chapter, we
are mainly interested in solving difference equations over n 0 in response to causal input
sequences. We proceed as follows:
(a) First, we determine the form of all solutions, yc (n), and the interval over which the
representation is valid, say over n no .
(b) Second, we propagate the given initial conditions up to the time instant no 1.
(c) We then determine the unknown constants in the description of the solution from
these initial conditions.
Example 8.6 (Finding the complete solution)

Consider again the system from Example 8.1, that is described by the difference equation
y(n) + 2y(n 1) 8y(n 2) = 2x(n 1)
albeit now with the initial conditions y(1) = 1 and y(2) = 0. Let us determine its complete
response when x(n) = (0.5)n u(n). Time is assumed to run forward. We proceed in steps.
1. We already know from Example 8.1 that a particular solution is given by
yp (n) =
4
(0.5)n ,
27
for n 2
2. The general form of the homogeneous solution is

yh (n) = C1 2n + C2 (4)n ,
for all n
since the modes of the system are 1 = 2 and 2 = 4, and where C1 and C2 are arbitrary
constants.
3. All solutions to the difference equation
y(n) + 2y(n 1) 8y(n 2) = 2x(n 1)
are therefore given by
y(n) = C1 2n + C2 (4)n
4
(0.5)n ,
27
for n 2
for some constants {C1 , C2 }.

4. We propagate the initial conditions up to the time instants just prior to n = 2, i.e., we determine y(0) and y(1). These values follow from the given initial conditions and from the
difference equation:
y(0) = 2, y(1) = 14
5. Using these initial conditions we find that the constants {C1 , C2 } must satisfy the linear
system of equations
4
2 = C1 + C2
,
27
ZERO-STATE
RESPONSE
2
14 = 2C1 4C2
27
which leads to C1 = 10/9 and C2 = 80/27.

6. Therefore, the complete solution of the system in response to x(n) = (0.5)n u(n) is given by
yc (n) =
10 n 80
4
2
(4)n
(0.5)n ,
9
27
27
n0
This solution is valid over n 0 and not just n 2 because we already enforced the initial
conditions at time instants n = 0 and n = 1.
8.4 ZERO-STATE RESPONSE

Now that we know how to determine the complete solution of a constant-coefficient difference equation, we move on to explain how to find two special responses of a system.
One is called the zero-state (or forced) response of the system and the other is called the
zero-input response of the same system.
By definition, the zero-state response of a system is the complete response of the system
when it is assumed to be relaxed or in zero initial state. In other words, the determination
of the zero-state response of the difference equation amounts to assuming that it describes
an LTI system by assuming that the system is relaxed and, accordingly, that the initial conditions are zero. We shall denote the zero-state response of a system by yzs (n).
Example 8.7 (Finding the zero-state response)
Consider again the system from Example 8.6:
y(n) + 2y(n 1) 8y(n 2) = 2x(n 1)
with initial conditions y(1) = 1 and y(2) = 0. Time is assumed to run forward. Let us
determine its zero-state response when x(n) = (0.5)n u(n). In order to do so, we need to assume
that the system is relaxed and use instead the alternative initial conditions
y(1) = 0,
y(2) = 0
We proceed in steps. The first four steps are similar to what we already did in Example 8.6.
1. We already know from Example 8.6 that the particular solution is given by
yp (n) =
4
(0.5)n ,
27
for n 2
2. The general form of the homogeneous solution is

yh (n) = C1 2n + C2 (4)n ,
for all n
since the modes of the system are 1 = 2 and 2 = 4, and where C1 and C2 are arbitrary
constants.
3. All solutions to the difference equation
y(n) + 2y(n 1) 8y(n 2) = 2x(n 1)
189
SECTION 8.4
190
are therefore described by
CHAPTER 8
SOLVING
DIFFERENCE
EQUATIONS
y(n) = C1 2n + C2 (4)n
4
(0.5)n ,
27
for n 2
for some constants {C1 , C2 }.

4. We propagate the initial conditions y(1) = 0 and y(2) = 0 to the time instants just prior
to n = 2. Using the given difference equation along with the relaxed initial conditions we get
y(0) = 0,
y(1) = 2
5. Using these initial conditions we find that the constants {C1 , C2 } must satisfy the linear
system of equations
0 = C1 + C2
4
,
27
14 = 2C1 4C2
2
27
which leads to C1 = 12/27 and C2 = 8/27.

6. Therefore, the zero-state response of the system is
yzs (n) =
12 n
8
4
2
(4)n
(0.5)n ,
27
27
27
n0
Superposition Principle for Zero-State Responses

An important property of the zero-state response of relaxed systems that are described by
constant-coefficient difference equations is that the zero-state response satisfies a superposition principle. Specifically, if we determine the zero-state responses of the system to
the input sequences x1 (n) and x2 (n) separately, then the zero-state response to the linear
combination ax1 (n) + bx2 (n) is the corresponding linear combination of the individual
zero-state responses to x1 (n) and x2 (n). This is simply because such relaxed systems are
linear.
Example 8.8 (Superposition property)

Consider a relaxed and causal system that is described by the difference equation
y(n) + 2y(n 1) 8y(n 2) = 2x(n 1)
and let us determine its response to the input sequence
x(n) = (0.5)n u(n) + 2u(n 1)
Obviously, since the system is relaxed, its response to the above input sequence coincides with the
zero-state response of the system. So let us determine the zero-state response.
First, note that the given sequence x(n) is a linear combination of two sequences. We already
know from Example 8.7, that the zero-state response of the system to the input sequence 0.5n u(n)
is
12 n
8
4
yzs,1 (n) =
2
(4)n
(0.5)n , n 0
27
27
27
Let us now determine the zero-state response to 2u(n 1). For this purpose, we proceed as in
Example 8.6.
191
To begin with, we know that the homogeneous solution has the form
SECTION 8.5
yh (n) = C1 2n + C2 (4)n ,
for all n
ZERO-INPUT
RESPONSE
A particular response to 2u(n 1) is assumed to be of the form

yp (n) = Ku(n)
In order determine the constant K we substitute yp (n) into the difference equation and find that it
must satisfy
Ku(n) + 2Ku(n 1) 8Ku(n 2) = 4u(n 2)
Evaluating both sides of the equality for any value of n 2 (for which none of the terms vanish), we
arrive at K = 4/5. This means that the complete solution of the system in response to 2u(n 1)
has the form
4
yc (n) = C1 2n + C2 (4)n ,
for n 2
5
Since the system is assumed to be relaxed, we now determine the constants {C1 , C2 } from the initial
conditions y(1) = 0 = y(2). We propagate these conditions up to n = 1 to get y(1) = 0 and
y(0) = 0. Solving for C1 and C2 we obtain
C1 =
2
2
and C2 =
3
15
Therefore, the zero-state response that corresponds to the input 2u(n 1) is

yzs,2 (n) =
2
4
2 n
2 +
(4)n ,
3
15
5
n0
It then follows from the superposition principle that the zero-state response of the system to
x(n) = (0.5)n u(n) + 2u(n 1)
is
yzs (n)
=
=
yzs,1 (n) + yzs,2 (n)

12
8
4
4
2
2
+
2n +
(4)n
(0.5)n , n 0
3
27
15
27
27
5
That is,
yzs (n) =
10 n
22
4
4
2 +
(4)n
(0.5)n , n 0
9
135
27
5
8.5 ZERO-INPUT RESPONSE
The zero-input response (also called natural or unforced) response of a system is simply the
homogeneous solution with the constants determined from the initial conditions. In other
words, we set the input sequence to zero (and, hence, the name zero-input). We denote the
zero-input response of a system by yzi (n).
192
CHAPTER 8
SOLVING
DIFFERENCE
EQUATIONS
Example 8.9 (Finding the zero-input solution)

Consider the same system from Example 8.6, namely,
y(n) + 2y(n 1) 8y(n 2) = 2x(n 1)
but with initial conditions y(1) = 1 and y(2) = 0. Time is assumed to run forward. We already
know that the general form of the homogeneous solution is
yh (n) = C1 2n + C2 (4)n ,
for all n
For the given initial conditions y(1) = 1 and y(2) = 0, we find that the constants {C1 , C2 }
must satisfy the linear system of equations
1=
C1
C2
,
2
4
0=
C1
C2
+
4
16
which leads to C1 = 2/3 and C2 = 8/3. Therefore, the zero-input response is

yzi (n) =
2 n 8
2 (4)n , n 0
3
3
Observe that we specified the time interval for yzi (n) as n 0 since we are focusing on responses
of systems over n 0.
It is easy to see from the above example that the zero-input response of a system that is
described by a constant-coefficient difference equation will always include a linear combination of the exponentials contributed by the modes of the system.
Superposition Principle for Zero-Input Responses
An important property of the zero-input response of systems that are described by constantcoefficient difference equations is that the zero-input response also satisfies a superposition
principle in relation to the initial conditions. Specifically, if we determine separately the
zero-input responses for two given sets of initial conditions that are specified at the same
time instants, then the zero-input response to a linear combination of these initial conditions is the corresponding linear combination of the individual zero-input responses.
Proof: We establish the statement for a second-order constant-coefficient difference equation with
two distinct nonzero modes, say and . The argument is general enough and conveys the main
idea.
For the second-order system, the zero-input response will be a linear combination of n and n ,
say
yzi (n) = C1 n + C2 n
Given initial conditions y(1) = a1 and y(2) = b1 , we can determine C1 and C2 by solving the
linear system of equations
2
3 2
32
3
1
1
a1
C1
6
7 6
76
7
4
5=4
54
5
1
1
b1
C2
2
2
| {z }
r1
{z
A
}|
{z
z1
This is a linear system of equations of the form Az1 = r1 , whose unique solution z1 gives us the
desired values for the constants {C1 , C2 } and, therefore, determines the first zero-input response,
which we denote by yzi,1 (n).
If we change the initial conditions to y(1) = a2 and y(2) = b2 , we obtain a new system of
linear equations with the same coefficient matrix, A, but with a different right-hand side vector r2 ,
say, Az2 = r2 , where now
2
6
4
a2
b2
7 6
5=4
| {z }
r2
1
2
1
2
{z
32
76
54
}|
C1
C2
{z
3
7
5
}
z2
The unique solution vector z2 gives us the values of the new constants {C1 , C2 } and it leads to a
second zero-input response, yzi,2 (n).
It is now easy to see that if we employ an initial condition vector r3 that is a linear combination
of the earlier initial condition vectors {r1 , r2 }, say,
r3 = 1 r1 + 2 r2
for some coefficients {1 , 2 }, then the new solution vector z3 to the linear system of equations
Az3 = r3
is given by the same linear combination of the corresponding {z1 , z2 }, i.e.,
z3 = 1 z1 + 2 z2
Consequently, the new zero-input sequence yzi,3 (n) is given by the same linear combination of the
corresponding sequences {yzi,1 (n), yzi,2 (n)},
yzi,3 = 1 yzi,1 (n) + 2 yzi,2 (n)
8.6 SECOND METHOD FOR FINDING COMPLETE SOLUTIONS

Using the concepts of the zero-state and zero-input responses, we can now describe a
second method for finding the complete solution of a system that is described by a constantcoefficient difference equation with initial conditions. Specifically, it always holds that the
complete response of such systems is the sum of the zero-input and zero-state responses,
yc (n) = yzi (n) + yzs (n)
(8.9)
This result suggests that in order to determine the complete solution, we simply determine
separately the zero-input response and the zero-state response and then add both responses.
This procedure decouples the effects of the initial conditions and the input sequence. When
we evaluate the zero-input response, we set the input sequence to zero and obtain the response of the system to the initial conditions. When, on the other hand, we evaluate the
zero-state response, we set the initial conditions to zero and evaluate the response of the
system to the input sequence.
Consider again the system
y(n) + 2y(n 1) 8y(n 2) = 2x(n 1)
193
SECTION 8.6
SECOND
METHOD
FOR FINDING
COMPLETE
SOLUTIONS
194
CHAPTER 8
with initial conditions y(1) = 1, y(2) = 0, and input sequence x(n) = (0.5)n u(n). Time is
assumed to run forward. We found earlier in Example 8.6 that the complete solution is given by
SOLVING
DIFFERENCE
EQUATIONS
yc (n) =
10 n 80
4
2
(4)n
(0.5)n ,
9
27
27
n0
We also found in Examples 8.7 and 8.9 that the zero-state and zero-input solutions of the system are
given by
yzs (n)
yzi (n)
8
4
12 n
2
(4)n
(0.5)n , n 0
27
27
27
2 n 8
2 (4)n , n 0
3
3
By adding these two components we easily find that the sum is identical to the expression for yc (n)
given above.
8.7 TRANSIENT AND STEADY-STATE RESPONSE

The transient response of a system is defined as that part of the complete response that
decays to zero as n approaches infinity. On the other hand, the steady-state response of the
system is that part of the complete solution that persists in the response as n . We
denote the steady-state response by yss (n) and the transient response by ytr (n). Furthermore, the steady-state value of a sequence, denoted by y(), refers to its limiting value as
n , i.e.,
y() = lim y(n)

(8.10)
n
Example 8.11 (Components of the complete solution)

Consider the same system of Example 8.11. Its complete response is given by
yc (n) =
4
10 n 80
2
(4)n
(0.5)n ,
9
27
27
n0
In this case, the steady-state response is

yss (n) =
10 n 80
2
(4)n ,
9
27
while the transient response is

ytr (n) =
4
(0.5)n ,
27
n0
Example 8.12 (Steady-state value of a sequence)

The steady-state value of the sequence
n
y(n) = 2u(n) +
1
2
u(n)
is easily seen to be
lim y(n) = 2.
since the exponential sequence dies out with time.
8.8 THIRD METHOD FOR FINDING COMPLETE SOLUTIONS
195
SECTION 8.8
The earlier discussions provided two methods for evaluating the complete responses of systems that are described by constant-coefficient difference equations. Both methods require
the input sequence to belong to one of the forms listed in Table 8.3 since both methods
involve a step that requires determining a particular solution:
1. First Method. Determine a particular solution, yp (n), and the general form of the
homogeneous solution, yh (n), in terms of arbitrary constants. Then, set yc (n) =
yp (n) + yh (n) and determine the unknown constants from the given initial conditions.
2. Second Method. Determine the zero-input response, yzi (n), by solving a homogeneous equation with the given initial conditions. Determine further the zerostate response by finding the complete solution of the relaxed system. Then, set
yc (n) = yzi (n)+yzs (n). The complete solution of the relaxed system is determined
by using the first method, through a combination of particular and homogeneous solutions.
Nevertheless, the second method described above in terms of the zero-state and zero-input
responses can be extended to more general input sequences (other than those listed in
Table 8.3) by noting that the zero-state response can be alternatively determined via a
convolution sum:
3. Third Method. Since finding the zero-state response requires that we assume that
the system is relaxed, then the difference equation in question will be describing an
LTI system. Hence, we can first determine the impulse response sequence of this
LTI system and then convolve it with the input sequence to determine the zero-state
response, yzs (n). This response is subsequently added to the zero-input response,
yzi (n), to obtain the complete solution, yc (n).

Consider again the system studied in Example 8.10, and which is described by the difference equation
y(n) + 2y(n 1) 8y(n 2) = 2x(n 1)
with initial conditions y(1) = 1 and y(2) = 0. Time is assumed to run forward. Let us determine
the complete response to x(n) = (0.5)n u(n) by means of the third method outlined above. We
already know from Example 8.10 that the complete solution is given by
yc (n) =
10 n 80
4
2
(4)n
(0.5)n ,
9
27
27
n0
We would like to arrive at this same expression by means of the third method. According to this
method, we proceed as follows:
1. We first determine the zero-input response of the system, which we already know from Example 8.9 to be:
2
8
yzi (n) = 2n (4)n , n 0
3
3
2. We next determine the impulse response sequence of the relaxed system, namely, we determine the solution of
h(n) + 2h(n 1) 8h(n 2) = 2(n 1)
THIRD
METHOD
FOR FINDING
COMPLETE
SOLUTIONS
196
CHAPTER 8
SOLVING
DIFFERENCE
EQUATIONS
with assumed initial conditions h(2) = 0 = h(1) due to relaxation. Relaxed systems
described by such constant-coefficient difference equations are LTI. Since the system is assumed be causal, we must have h(n) = 0 for n < 0. The above difference equation in h(n)
becomes homogeneous for n 2. Propagating the initial conditions up to n = 1 we have
h(0) = 0,
h(1) = 2
The modes of the system are 1 = 2 and 2 = 4. Therefore, the general form of the
impulse response sequence is
h(n) = C1 2n + C2 (4)n
We determine the constants {C1 , C2 } from the conditions h(0) = 0 and h(1) = 2. Doing so,
we obtain the linear system of equations
C1 + C2 = 0,
2C1 4C2 = 2
which leads to C1 = 1/3 and C2 = 1/3. Therefore, the impulse response sequence is given
by

1
1 n
h(n) =
2 (4)n u(n)
3
3
where we incorporated a step-sequence to enforce h(n) = 0 over n < 0.
3. We now determine the zero-state response of the system by convolving the input sequence,
x(n) = (0.5)n u(n), with the impulse response sequence, h(n). Specifically,
yzs (n) = x(n) h(n)
Obviously, since both sequences, x(n) and h(n) are causal, the resulting sequence yzs (n) is
also causal, i.e.,
yzs (n) = 0 for n < 0
To determine the samples of yzs (n) for n > 0 we evaluate the convolution sum:
yzs (n)
=
=
x(n) h(n)
X
k=
x(k)h(n k)
n
X
k=0
1 nX
2
3
n
(0.5)k
k=0
1 nk
1
2
(4)nk
3
3
k
1
4
X
1
1
(4)n
3
8
n

k
k=0
n+1
=
=
1
1 (1/8)n+1
1 n 1 (1/4)
2
(4)n
3
1 1/4
3
1 + 1/8
8
4
4 n
n
n
2
(4)
(0.5) , n 0
9
27
27
4. Finally, we add yzs (n) and yzi (n) to get

yc (n) =
10 n 80
4
2
(4)n
(0.5)n ,
9
27
27
n0
197
8.9 APPLICATIONS
SECTION 8.9
APPLICATIONS
In this section, we illustrate an application of some of the concepts covered in the chapter
in the context of macroeconomics and cell division in biology.
8.9.1 Macroeconomics Model

A simplified model for the national economy of a country can be motivated as follows. Let
y(n) denote the size of the national income evaluated during the accounting period n. This
income is the combined sum of three components:
(a) The amount of consumer expenditure through the purchase of goods. We denote
this quantity by C(n). It is assumed that C(n) proportional to the national income
during the preceding accounting period, say,
C(n) = y(n 1)
where is a positive variable smaller than one.
(b) The amount of private investments, which we denote by I(n). It is assumed that
I(n) is proportional to the increase in consumption of the current accounting period
relative to the previous period, i.e.,
I(n) = [C(n) C(n 1)] = [y(n 1) y(n 2)]
where is a positive constant.
(c) The amount of Government spending, which we will assume is normalized to one
unit.
Adding the three components (a)-(c), we find that the evolution of the national income is
governed by the second-order difference equation:
y(n) = (1 + )y(n 1) y(n 2) + 1, n 0
which we rewrite as
y(n) (1 + )y(n 1) + y(n 2) = 1, n 0
(8.11)
We set the origin of time n = 0 at the accounting period from which we start to assess the
evolution of the national income starting from some initial conditions, say, y(2) = 2 and
y(1) = 5/2. With this frame of reference, the value of one that appears on the right-hand
side of (8.11) can be taken to correspond to the step-sequence, u(n).
The evolution of y(n) with time is a function of the modes of the system (8.11), which
are in turn determined by the relative values of the parameters and . The characteristic
equation of the difference equation is
2 (1 + ) + = 0
and the modes are given by
=
(1 + )
p
2 (1 + )2 4
2
(8.12)
198
Note that complex modes are possible when the quantity below is negative
CHAPTER 8
SOLVING
DIFFERENCE
EQUATIONS
= 2 (1 + )2 4
When < 0, the evolution of y(n) with time will exhibit an oscillatory behavior. Table 8.4
lists the modes that correspond to three choices of the pair (, ). We now solve the
difference equation (8.11)for the three cases listed in the table.
TABLE 8.4
Modes of system (8.11) for three different choices of the parameters (, ).
8/9
5/9
1/2
1/2
5
1
2/3
5/3
1
(1 + j)
2
2/3
5/3
1
(1 j)
2
(a) = 8/9 and = 1/2. In this case, the national economy is modeled according to
the difference equation
4
4
y(n) y(n 1) + y(n 2) = u(n), y(2) = 2, y(1) = 5/2, n 0
3
9
We determine the complete solution by finding the zero-state and zero-input responses. To begin with, the system exhibits a double mode at = 2/3 (which
lies inside the unit circle). Therefore, the general form of the homogeneous solution
is
n
n
2
2
yh (n) = C1
+ C2 n
, for all n
3
3
For the given initial conditions we find that the constants C1 and C2 must satisfy the
relations
3
5
9
(C1 2C2 ) = 2 and
(C1 C2 ) =
4
2
2
which lead to C1 = 11/9 and C2 = 1/6. Therefore, the zero-input solution is given
by
n
n
1
2
11 2
+ n
, n0
yzi (n) =
9 3
6
3
Let us now determine the zero-state solution. We select a particular solution of the
form yp (n) = Ku(n) and substitute into the difference equation
4
4
Ku(n) Ku(n 1) + Ku(n 2) = u(n)
3
9
For any value of n 2, we find that K = 9 so that
yp (n) = 9u(n)
We already know that the general form of the homogeneous solution is
n
n
2
2
+ D2 n
, for all n
yh (n) = D1
3
3
for some constants D1 and D2 , which we now determine differently from the C1
and C2 in the zero-input case. Indeed, all solutions to the relaxed system
4
4
y(n) y(n 1) + y(n 2) = u(n)
3
9
are described by
n
n
2
2
+ D2 n
, n2
y(n) = 9u(n) + D1
3
3
We now propagate the initial conditions y(1) = 0 = y(2) to the time instants
n = 0 and n = 1 to find
y(0) = 1 and y(1) = 7/3
We use these values to solve for D1 and D2 and to arrive at the zero-state solution,
yzs (n). Doing so we find that D1 and D2 must satisfy
1 = 9 + D1
and
7
2
2
= 9 + D1 + D2
3
3
3
so that D1 = 8 and D2 = 2. Therefore, the zero-state response is

yzs (n) = 9u(n) 8
n
n
2
2
2n
, n2
3
3
Adding yzi (n) and yzs (n) we arrive at the desired complete solution, which describes the evolution of the national income as a function of time:
n
n

11
2
61 2
u(n)
n
yc (n) = 9
(8.13)
9 3
6
3
We observe in this case that the national income tends to the steady-state response
yss (n) = 9u(n)
as n
while the transient term below tends to zero

n

2
11
61
u(n)
+
n
ytr (n) =
9
6
3
(8.14)
(8.15)
(b) = 5/9 and = 5. In this case, the national economy is modeled according to the
difference equation
y(n)
10
25
y(n 1) + y(n 2) = u(n), y(2) = 2, y(1) = 5/2, n 0
3
9
Now the the system exhibits a double mode at = 5/3 (which lies outside the unit
circle). We can repeat a similar argument to the previous case and determine the
complete solution to arrive at
yc (n) =
9 1185
+
4
252
n
n
5
30
5
u(n)
+ n
3
7
3
(8.16)
199
SECTION 8.9
APPLICATIONS
200
We observe now that the national income grows unbounded rather fast with time.
CHAPTER 8
(c) = 1/2 and = 1. In this case, the national economy is modeled according to the
difference equation
1
y(n) y(n 1) + y(n 2) = u(n), y(2) = 2, y(1) = 5/2, n 0
2
Now the system exhibits two complex conjugate modes at
1
2 j
= (1 j) =
e 4
2
2
Both modes lie inside the unit circle. We can repeat a similar argument to the previous cases in order to determine the corresponding complete solution. We leave it
as an exercise to the reader. It will be observed in this case that the solution exhibits
decaying oscillatory behavior at the angular frequency = /4 and that it tends in
steady-state towards y(n) 2 as n .
Figure 8.1 plots the evolution of y(n) for the three cases considered above over the first 20
iterations. Only the first 6 iterations are shown for the second case with modes at = 5/3
since the solution grows fast.
=8/9, =1/2, double modes at =2/3
y(n)
10
5
0
10
15
n (years)
=5/9, =5, double modes at =5/3
20
y(n)
100
50
0
3
4
n (years)
=1/2, =1, complex modes at =0.5(1+j) and =0.5(1j)
2.5
y(n)
SOLVING
DIFFERENCE
EQUATIONS
2
1.5
10
n (years)
15
20
FIGURE 8.1 Plot of the time evolution of the national income as a function of time for three
cases involving modes inside the unit circle (top plot), modes outside the unit circle (middle plot),
and complex modes (bottom plot).
201
Practice Questions:
1. Determine the evolution of the national income when = 1/2 and = 1. Identify the
transient and steady-state components of the response.
2. For a fixed , describe all values of that would result in real modes. Describe all values of
that would result in modes that lie inside the unit circle.
3. For a fixed , describe all values of that would result in real modes.
4. Let = 8/9 and = 1/2. Assume the Government wishes to stimulate the economy by
injecting resources at time n = 5. Change the input signal from u(n) to u(n) + 0.25(n 5).
Determine the evolution of the national economy and compare it with (8.13). How do the
corresponding steady-state responses compare with each other?
8.9.2 Cell Division in Biology

We describe in this section a simplified mathematical model for the division of biological
cells as they evolve through three stages, starting from stem cells into progenitor cells and
then mature cells. The model adopted here can also be applied to other scenarios involving
multiple states and transitions between these states.
Stem cells have the ability to regenerate (or self-renew). They also have the ability to
divide into intermediate cells known as progenitor cells. This process of division is called
differentiation. Progenitor cells are more specialized than stem cells. They in turn can
differentiate to generate mature cells with highly specialized functions such as blood cells,
skin cells, or muscle cells. Progenitor cells lie in an intermediate state between stem cells
and mature cells. All three types of cells can also die.
We would like to examine the evolution of the number of cells in a tissue over time. To
do so, we let {s(n), p(n), q(n)} denote the number of stem, progenitor, and mature cells at
iteration n. We choose the time interval between two successive values of n to correspond
to the duration of time it takes for a stem cell to divide. We assume that progenitor cells
take P units of time to differentiate (with P 1) and that mature cells live for Q units of
time (with Q 1). Both P and Q can assume fractional values.
Stem Cell Population
Let s denote the fraction of stem cells that regenerate at each iteration. Let also s denote
the fraction of stem cells that differentiate into progenitor cells. The remaining fraction of
stem cells, given by 1 s s , is assumed to die see Fig. 8.2. Then, at time n, the
number of stem cells that are present is given by the following relation:
s(n) = 2s s(n 1), s(0) = so , n 0
(stem cells)
(8.17)
The factor of 2 is because each of the s s(n 1) cells divide into 2 cells during regeneration. The above recursion is a first-order homogeneous difference equation with an initial
condition, whose solution we immediately identify as the following exponential sequence:
s(n) = so (2s )n u(n)
(stem cell population)
(8.18)
SECTION 8.9
APPLICATIONS
202
CHAPTER 8
SOLVING
DIFFERENCE
EQUATIONS
progenitor cell
s
stem cell
mature cell
FIGURE 8.2 Stem cells regenerate and divide with probabilities s and s , respectively. The
remaining fraction of 1 s s dies. Likewise, progenitor cells regenerate and divide with
probabilities p and p , respectively. The remaining fraction of 1 p p dies.
Progenitor Cell Population

Likewise, let p denote the fraction of progenitor cells that regenerate every P units of
time. Let also p denote the fraction of progenitor cells that divide into mature cells every
P units of time. Since we are assuming that progenitor cells take P units of time to
regenerate, compared with one unit of time for the stem cells, we conclude that p /P is the
fraction (or probability) of progenitor cells that divide during one unit of time. Therefore,
we can express the number of progenitor cells that are present at time n by the following
relation:
p(n) =
2p
p(n 1) + 2s s(n 1), p(0) = po , n 0
P
(progenitor cells)
(8.19)
This relation takes into account the two sources of progenitor cells: self-renewal of a fraction of the cells (represented by p /P ) and differentiation by a fraction of the stem cells
(represented by s ).
If we substitute (8.18) into (8.19), we obtain
p(n) =
2p
p(n 1) + x(n), p(0) = po , n 0
P
(8.20)
with the exponential input sequence

x(n) = 2s so (2s )n1 u(n 1)
To solve the first-order difference equation (8.20), we can procedure in many ways, as explained in the body of this chapter. Here we illustrate the computation of the solution by
determining its zero-input and zero-state components and then combining them.
Zero-input component. We note that the general form of the homogenous solution of
(8.20) is
n

2p
, n0
ph (n) = C
P
for arbitrary constants C. We find C by using the initial condition ph (0) = p(0), which
gives C = po so that the zero-input component of the solution is given by

n
2p
pzi (n) = po
, n0
P
Zero-state component. To determine the zero-state solution of (8.20), we first determine a particular solution of the same form as x(n), namely,
pp (n) = K(2s )n u(n 1)
Plugging into (8.20) we find that K must satisfy
K(2s )n u(n 1) =
2p
K (2s )n1 u(n 2) + 2s so (2s )n1 u(n 1)
P
Selecting any value of n 2 and solving for K we find

K=
s s o
s Pp
so that the particular solution is given by

pp (n) =
s s o
n
(2s ) , n 2
s Pp
The zero-state solution then has the form

n
2p
s s o
n
pzs (n) = C
+
(2s ) , n 2
P
s Pp
where we added the contribution from the homogeneous component. To determine the
constant C we use the difference equation (8.20) to propagate the (now zero-state) initial
condition p(0) = 0 to time n = 1 to get p(1) = 2s so . Using this initial condition in the
above expression for pzs (n) gives
C=
Therefore,
s s o
s Pp
n

s s o
2p
n
yzs (n) =
, n1
(2s )
P
s Pp
Complete solution. Adding the zero-input and zero-state components we arrive at the
desired expression for the evolution of the population of the progenitor cells:

n
s s o
s
2p
s
o
n
p(n) =
, n0
p (2s ) + po p P
s
s
P
P
(8.21)
We see that the evolution of p(n) is controlled by the two modes: 1 = 2s and 2 =
2p /P . The first mode represents the dynamics of stem-cell differentiation, while the
second mode represents the dynamics of progenitor cell regeneration.
203
SECTION 8.9
APPLICATIONS
CHAPTER 8
SOLVING
DIFFERENCE
EQUATIONS
Mature Cell Population

Let denote the fraction of mature cells that die every Q units of time. Then /Q represents the fraction of mature cells that die during one unit interval. It follows that we can
express the number of mature cells that are present at time n by the following relation:
q(n) =

2p
q(n 1) +
p(n 1), q(0) = qo , n 0
1
Q
P
(mature cells)
(8.22)
This relation takes into account two effects: differentiation by progenitor cells (represented
by p /P ) and survival rate of mature cells (represented by 1 /Q).
Recursion (8.22) represents a first-order difference equation with input sequence given
by
2p
p(n 1)
x(n) =
P
We can use the expression for p(n) from (8.21) to determine a closed-form expression for
q(n) as a function of time by following the same procedure we used for p(n) in terms
of zero-state and zero-input components. We leave the details to the reader. Figure 8.3
illustrates the evolution of the population sizes of stem cells, progenitor cells, and mature
cells for the numerical values indicated in the caption of the figure.
mature cells
10
population size
204
progenitor cells
stem cells
4
10
50
100
n (iteration)
150
200
FIGURE 8.3 Evolution of the population of stem, progenitor, and mature cells in a simulation
that assumes s = 0.5, s = 0.3, p = 0.6, p = 0.3, = 0.1, P = 1.5, Q = 2, and po = 104
stem cells.
Practice Questions:
1. Determine a closed-form expression for q(n) by solving the difference equation (8.22).
2. What condition s should satisfy in order for the population of stem cells to remain stable at
so ?
3. Using s = 0.5, what condition should be satisfied in order for the steady-state value of the
progenitor cell population to be larger than the steady-state value of the stem cell population?
4. What is the steady-state value of the mature cell population? How does it compare to the
steady-state value of the progenitor cell population?
5. Assume a mutation happens that affects the value of and changes it to > . Will this
change affect the rate at which the number of mature cells evolves with time? Will the change
affect the steady-state population of mature cells?
8.10 PROBLEMS
Problem 8.1 Find, when possible, particular solutions to the following difference equations when
x(n) = u(n):
(a) y(n) + y(n 1) 5y(n 2) = x(n).
(b) y(n) = 4y(n 2) + x(n 1).

(c) y(n) 4y(n 2) = 2x(n).
x(n) = u(n 1) :
(a) y(n) = y(n 2) + x(n) + x(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1).
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n).
Problem 8.3
n Find, when possible, particular solutions to the following difference equations when
x(n) = 14 u(n):
(a) y(n) + y(n 1) 5y(n 2) = x(n).
(b) y(n) = 4y(n 2) + x(n 1).

(c) y(n) 4y(n 2) = 2x(n).
n
x(n) = n 14 u(n):
(a) y(n) = y(n 2) + x(n) + x(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1).
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n).
Problem 8.5
n Find, when
possible, particular solutions to the following difference equations when
x(n) = 14 cos 4 n u(n):
(a) y(n) + y(n 1) 5y(n 2) = x(n).
(b) y(n) = 4y(n 2) + x(n 1).

(c) y(n) 4y(n 2) = 2x(n).
Problem 8.6 Find, when

possible, particular solutions to the following difference equations when
x(n) = sin 4 (n 1) u(n 2):
(a) y(n) = y(n 2) + x(n) + x(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1).
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n).
Problem 8.7 Determine

the zero-state solutions of the following systems when the input sequence
n
is x(n) = n 14 u(n):
205
SECTION 8.10
PROBLEMS
206
CHAPTER 8
SOLVING
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EQUATIONS
(a) y(n) + y(n 1) 5y(n 2) = x(n), y(2) = 1, y(1) = 1.
(b) y(n) = 4y(n 2) + x(n 1), y(2) = 1, y(1) = 1.

(c) y(n) 4y(n 2) = 2x(n), y(2) = 1, y(1) = 1.
Problem 8.8 Determine the zero-state solutions of the following systems when the input sequence
is x(n) = n2 u(n 1):
(a) y(n) = y(n 2) + x(n) + x(n 2), y(2) = 1, y(1) = 1.
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1), y(2) = 1, y(1) = 1.
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n), y(2) = 1, y(1) = 1.
Problem 8.9 Determine

the zero-input solutions of the following systems when the input sequence
n
is x(n) = n 14 u(n):
(a) y(n) + y(n 1) 5y(n 2) = x(n), y(2) = 1, y(1) = 1.
(b) y(n) = 4y(n 2) + x(n 1), y(2) = 1, y(1) = 1.

(c) y(n) 4y(n 2) = 2x(n), y(2) = 1, y(1) = 1.
Problem 8.10 Determine the zero-input solutions of the following systems when the input sequence is x(n) = n2 u(n 1):
(a) y(n) = y(n 2) + x(n) + x(n 2), y(2) = 1, y(1) = 1.
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1), y(2) = 1, y(1) = 1.
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n), y(2) = 1, y(1) = 1.
Problem 8.11 Describe all solutions to the following systems

using the parametrization yc (n) =
n
yp (n) + yh (n), when the input sequence is x(n) = n 14 u(n):
(a) y(n) = y(n 2) + x(n) + x(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1).
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n).
Problem 8.12 Describe all solutions to the following systems using the parametrization yc (n) =
yp (n) + yh (n), when the input sequence is x(n) = n2 u(n 1):
(a) y(n) = y(n 2) + x(n) + x(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1).
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n).
Problem 8.13 Describe all solutions to the following systems

n using the parametrization yc (n) =
yzi (n) + yzs (n), when the input sequence is x(n) = n 14 u(n):
(a) y(n) + y(n 1) 5y(n 2) = x(n).
(b) y(n) = 4y(n 2) + x(n 1).

(c) y(n) 4y(n 2) = 2x(n).
Problem 8.14 Describe all solutions to the following systems using the parametrization yc (n) =
yzi (n) + yzs (n), when the input sequence is x(n) = n2 u(n 1):
(a) y(n) = y(n 2) + x(n) + x(n 2).
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1).
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n).
Problem 8.15 Determine the complete solution of the following systems

using the parametrization
n
yc (n) = yp (n) + yh (n), when the input sequence is x(n) = n 14 u(n):
(a) y(n) + y(n 1) 5y(n 2) = x(n), y(2) = 1, y(1) = 1.
(b) y(n) = 4y(n 2) + x(n 1), y(2) = 1, y(1) = 1.

(c) y(n) 4y(n 2) = 2x(n), y(2) = 1, y(1) = 1.
Problem 8.16 Determine the complete solution of the following systems using the parametrization
yc (n) = yp (n) + yh (n), when the input sequence is x(n) = n2 u(n 1):
(a) y(n) = y(n 2) + x(n) + x(n 2), y(2) = 1, y(1) = 1.
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1), y(2) = 1, y(1) = 1.
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n), y(2) = 1, y(1) = 1.
Problem 8.17 Determine the complete solution of the following systems

using the parametrization
n
yc (n) = yzs (n) + yzi (n), when the input sequence is x(n) = n 14 u(n):
(a) y(n) + y(n 1) 5y(n 2) = x(n), y(2) = 1, y(1) = 1.
(b) y(n) = 4y(n 2) + x(n 1), y(2) = 1, y(1) = 1.

(c) y(n) 4y(n 2) = 2x(n), y(2) = 1, y(1) = 1.
Problem 8.18 Determine the complete solution of the following systems using the parametrization
yc (n) = yzs (n) + yzi (n), when the input sequence is x(n) = n2 u(n 1):
(a) y(n) = y(n 2) + x(n) + x(n 2), y(2) = 1, y(1) = 1.
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1), y(2) = 1, y(1) = 1.
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n), y(2) = 1, y(1) = 1.
Problem 8.19 For each of the systems in Prob. 8.17 indicate how the zero-state and the zero-input
components of the solution will change if the initial conditions are changed to y(2) = 1 and
y(1) = 0. Find the new complete solution in each case.
Problem 8.21 For each of the systems in Prob. 8.17 indicate how the zero-state and
n the zero-input
components of the solution will change if the input sequence is instead x(n) = 14 u(n) while the
initial conditions stay the same.
components of the solution will change if the input sequence is instead x(n) = u(n) while the initial
conditions stay the same.
Problem 8.23 For each of the systems in Prob. 8.15 indicate how the particular and homogeneous
Problem 8.25 For each of the systems in Prob. 8.15 indicate how the particular and
n homogeneous
components of the solution will change if the input sequence is instead x(n) = 14 u(n) while the
initial conditions stay the same.
components of the solution will change if the input sequence is instead x(n) = u(n) while the initial
conditions stay the same.
207
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PROBLEMS
208
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DIFFERENCE
EQUATIONS
Problem 8.27 Determine the transient and steady-state components

of the complete response of
n
the following systems when the input sequence is x(n) = n 14 u(n):
(a) y(n) + y(n 1) 5y(n 2) = x(n), y(2) = 1, y(1) = 1.
(b) y(n) = 4y(n 2) + x(n 1), y(2) = 1, y(1) = 1.
(c) y(n) 4y(n 2) = 2x(n), y(2) = 1, y(1) = 1.
Problem 8.28 Determine the transient and steady-state components of the complete response of
the following systems when the input sequence is x(n) = n2 u(n 1):
(a) y(n) = y(n 2) + x(n) + x(n 2), y(2) = 1, y(1) = 1.
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1), y(2) = 1, y(1) = 1.
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n), y(2) = 1, y(1) = 1.
Problem 8.29 Determine the complete solution of the following systems when the input sequence
is
n
n
1
1
x(n) = n
u(n) +
u(n)
4
2
(a) y(n) + y(n 1) 5y(n 2) = x(n), y(2) = 1, y(1) = 1.
(b) y(n) = 4y(n 2) + x(n 1), y(2) = 1, y(1) = 1.
(c) y(n) 4y(n 2) = 2x(n), y(2) = 1, y(1) = 1.
Problem 8.30 Determine the complete solution of the following systems when the input sequence
is
n
1
u(n)
x(n) = n2 u(n 1) +
3
(a) y(n) = y(n 2) + x(n) + x(n 2), y(2) = 1, y(1) = 1.
(b) y(n) 6y(n 1) + 9y(n 2) = 2x(n 1), y(2) = 1, y(1) = 1.
(c) y(n) y(n 1) + y(n 2) y(n 3) = x(n), y(2) = 1, y(1) = 1.
Problem 8.31 Find the complete response of the causal system
y(n) =
1
y(n 1) + x(n) , y(1) = 1
2
in at least four different ways, when x(n) = u(n).

y(n) +
1
y(n 1) = x(n 1) , y(1) = 1
4
in at least four different ways, when x(n) =

1 n
2
u(n).
Problem 8.33 A causal system is described by the difference equation

y(n)
1
1
2
y(n 1)
y(n 2) +
y(n 3) = x(n)
3
12
12
with initial conditions

y(3) = y(2) = 0 and y(1) = 1
(a) Find the zero-state response for x(n) = u(n).
(b) Find the complete response of the system to x(n) = u(n). Identify both the transient and the
steady-state responses.
(c) Find the complete response of the system when x(n) = u(n) u(n L), for any finite
positive integer L.

y(n) +
3
1
y(n 1) + y(n 2) = 2
4
8
n
1
2
x(n 1)
with initial conditions

y(2) = 0,
y(1) = 1

1 n2
(a) Find the zero-state response for x(n) =
u(n).

1 n2
3
(b) Find the complete response of the system to x(n) =

and the steady-state responses.
u(n). Identify both the transient
(c) Find the complete response of the system when x(n) = u(n) u(n L), for any finite
positive integer L.
Problem 8.35 Consider a causal system that is described by the difference equation
y(n)
7
2
y(n 1) + y(n 2) = x(n) + x(n 1)
3
3
with initial conditions y(1)= 1 and y(2) = 0. Determine its responses to x(n) = u(n) +
n
1
u(n 2) and to x(n) = 12 u(n 1).
2
y(n) +
1
3
y(n 1) + y(n 2) = 2
4
8
n
1
2
x(n 1)
with initial conditions y(2) = 0 and y(1) = 1. Determine its response to x(n) = u(n 3) +
1
u(n 2).
4
Problem 8.37 The input sequence x(n) = u(n 1) is first processed by the causal system
y(n) = 2y(n 1) + x(n),
y(1) = 1
The result, y(n), is then processed by an LTI system with impulse response sequence
n
h(n) =
1 , 1
What is the output of this second system?

Problem 8.38 The input sequence x(n) =
y(n) =

1 n1
2
u(n 3) is first processed by the system
1
y(n 1) + x(n 2),
3
y(1) = 1
The result, y(n), is then processed by an LTI system with impulse response sequence
n
h(n) =
209
SECTION 8.10
1 , 0, 1
What is the output of this second system?

Problem 8.39
Consider the causal system
y(n) 2y(n 1) + y(n 2) = x(n) , y(2) = 1, y(1) = 0
Determine its complete response to x(n) = u(n).
Problem 8.40
Consider the causal system
y(n) 2y(n 1) + y(n 2) = x(n) , y(2) = 1, y(1) = 0
PROBLEMS
210
Determine its complete response to x(n) = cos
n
2
u(n).
CHAPTER 8
SOLVING
DIFFERENCE
EQUATIONS

y(n)
1
2
y(n 1) y(n 2) = x(n) , y(2) = 1, y(1) = 0
3
3
to the input sequence

1
x(n) = u(n 2) +
2
n1
1
4
u(n)
Check your answer.

y(n)
1
2
y(n 1) y(n 2) = x(n) , y(2) = 0, y(1) = 1
3
3
to the input sequence

x(n) =
n
u(n 2) + n2
2
n3
1
2
u(n)
Check your answer.

Problem 8.43 Consider the causal system
6y(n) = y(n 1) + y(n 2) + 3x(n) ,
(a) Find its zero-input response, yzi (n).
(b) Find its zero-state response, yzs (n), to x(n) =
y(1) = 0 , y(2) = 1

1 n1
4
u(n).
(c) Find its complete response.

(d) Find the same complete response by using instead the representation yc (n) = yh (n) + yp (n)
and by finding the constants that define the homogeneous solution from the initial conditions.
(e) Now assume the initial conditions are changed to y(1) = 1 and y(2) = 2. Which part of
the response will change, yzi (n) or yzs (n)? What would the new complete response be?
(f) Find the new complete response when the initial conditions are given by any linear combination of the above initial conditions.
(g) Assume now the system is relaxed. Find its impulse response sequence, h(n).
(h) Let x(n) be a sequence for which the convolution w(n) = x(n2)h(n) is known. Find the
complete response of the original system to x(n) in terms of w(n), assuming initial conditions
y(1) = 0 and y(2) = 1.
y(n) =
3
1
y(n 1) y(n 2) + x(n)
4
8

(b) Find all solutions to the homogeneous equation.
(c) Find particular solutions when
(c.1) x(n) = u(n).
(c.2) x(n) = nu(n).
(c.3) x(n) =

1 n
3
u(n).
(d) Find the complete solution in each of the above cases.

(e) Find the impulse response h(n) of the relaxed system. Is the relaxed system BIBO stable?
(f) Assume for the remainder of the problem that y(1) = 1 and y(2) = 1. Find the impulse response sequence h1 (n) of the system. Verify that the sequence {h1 (n)} is absolutely
summable. Is this sufficient to conclude that the system is BIBO stable?
(g) Find the zero-input response, yzi (n).
(h) Find the zero-state response, yzs (n), when x(n) =

1 n
3
u(n).
(i) Find the complete response of the system by combining the answers of parts (g) and (h).
(j) Find the steady-state response, yss (n).
(k) Find the complete solution of part (i) via the alternative expression y(n) = yp (n) + yh (n),
where the constants of the homogeneous part are determined from the initial conditions. Your
answer should match that of part (i).
(l) Find the complete response of the system to x(n) = u(n).
(m) Find the complete response of the system to the input sequence
"
1
1
x(n) =
1+
2
2
n1 #
1
3
u(n 2)
by using the results derived so far and the superposition principle.

(n) How does your answer to part (i) change if the initial conditions become y(1) = 0 and
y(2) = 1?
(o) Using y(1) = 0 and y(2) = 1, find the complete response of the system to the sequence
n
x(n) =
1 , 1, 0, 2
(p) For any given initial conditions {y(1), y(2)}, how would you determine the complete
response of the system to an arbitrary input sequence x(n) for which you dont know how to
find a particular solution?
Problem 8.45 Consider the difference equation
y(n) =
5
y(n 1) y(n 2) + x(n 1) 2x(n 2)
2
(a) Assume the recursion describes a system that is causal and relaxed. Is the system LTI? Find
its impulse response sequence and verify whether it is BIBO stable or not.
(b) Assume now that the initial conditions are y(2) = 0 and y(1) = 1. Is the system LTI?
Find its impulse response sequence.
(c) Find the step responses of the systems in parts (a) and (b).
y(n)
1
y(n 1) = x2 (n),
2
n0
with initial condition y(1) = 2, and where x(n) denotes the input sequence.
(a) Draw a block diagram representation for the system.
(b) Find the zero-input response of the system.
(c) Find the zero-state response of the system corresponding to x(n) = (1/2)n u(n 1).
(d) Find the complete response of the system. Verify that your solution satisfies the initial condition and the difference equation.
y(n) =
1
y(n 1) +
2
n1
1
3
x(n)u(n 1),
y(1) = 1
211
SECTION 8.10
PROBLEMS
212
(a) Find the complete response of the system when

n1
CHAPTER 8
SOLVING
DIFFERENCE
EQUATIONS
x(n) = u(n 2) +
1
4
u(n 3)
(b) What would the complete response be if the initial condition is changed from y(1) = 1 to
y(2) = 0?
Problem 8.48 A causal system is described by the difference equation:
y(n) =
3
1
y(n 1) +
y(n 2) + x(n 2) + 2x(n 3)
10
10
with initial conditions y(0) = a and y(1) = b, where a and b are some constants.
1. Assume a = b = 0 and the system is initially relaxed. Find the impulse response sequence of
the system.
2. Assume now a = 1 and b = 0. Find the complete response of the system when x(n) =
7u(n).
3. Find values for a, b, and K such that the complete response due to the input x(n) = Ku(n),
satisfies y(n) = 1 for n 2.
CHAPTER
z-Transform
ur discussion so far in the book has focused on studying signals and systems in the
time domain. Specifically, in the earlier chapters, we studied the properties of signals and
systems, and determined the responses of systems, by focusing on their descriptions in
the time domain. There is much more to be learned by studying signals and systems in
the transform and frequency domains. Apart from simplifying many of the calculations
that we performed before, such as solving difference equations and computing convolution
sums, the transform domain representation will enable us to get a deeper understanding of
the behavior of signals and systems by analyzing their frequency content as well. Before
plunging into a full-blown study of the frequency-domain characterization of signals and
systems, we will initially develop the ztransform technique in preparation for our discussions on the Discrete-Time Fourier Transform (DTFT) and the Discrete-Fourier Transform
(DFT).
9.1 BILATERAL Z-TRANSFORM

The bilateral (also called two-sided) ztransform of a sequence x(n) is denoted by X(z)
and is defined as the series
X(z) =
x(n)z n
(9.1)
n=
at all values of z in the complex plane where the series is well-defined, as explained in
greater detail in Sec. 9.2. For now, note that either a negative or positive power of z 1 is
associated with each term of the sequence and the result is summed to provide the function X(z). In this way, the ztransform maps a sequence x(n) into a function of the
complex-variable z. The z transform, X(z), then helps provide a compact representation of the entire sequence. Rather than work with the (possibly) infinitely-numbered
samples of x(n), it becomes more convenient to work with and manipulate algebraically
the function representation X(z). Several examples to this effect are given in the sequel.
If we expand the defining relation (9.1) for X(z) we see that
X(z) = . . . + x(2)z 2 + x(1)z + x(0) + x(1)z 1 + x(2)z 2 + . . .
so that samples of x(n) that exist at times n < 0 are multiplied by positive powers of z,
and samples of x(n) that exist at times n > 0 are multiplied by positive powers of z 1 .
Note that x(0) is not multiplied by any power of z or z 1 . The qualification bilateral
or two-sided refers to the fact that powers of both z and z 1 appear in the expression
for X(z); this is in contrast to the unilateral z-transform, which we shall encounter later
in Ch. 12. Nevertheless, we shall often use the shorter terminology z-transform to refer
213
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214
CHAPTER 9
z-TRANSFORM
to X(z). This is because it will usually be understood from the context that X(z) is the
bilateral transform.
Example 9.1 (Finite-duration sequence)

Consider the finite-duration sequence
x(n) = 2(n + 1) + 4(n) + 3(n 1)
The sequence consists of 3 nonzero samples (and, therefore, has finite duration), as illustrated in
Fig. 9.1.
x(n)
4
2
1
1
FIGURE 9.1 A finite-duration sequence x(n) with three samples.
Using (9.1), the corresponding z-transform is given by

X(z) = 2z + 4 + 3z 1
In general, the ztransform of a sequence is not defined for all values of z. For example, in this
particular example, the above X(z) is defined for all values of z in the complex plane with the
exception of the points z = 0 and ; the term involving z 1 is not defined at z = 0 and the term
involving z is not defined at .
9.2 REGION OF CONVERGENCE

The series (9.1), which defines the z-transform of a sequence, may or may not converge for
all values of z in the complex plane. In Example 9.1 we observed that the corresponding
X(z) did not exist at the points z = 0 and z = . In order to exclude such points, we
shall associate with every ztransform a so-called region of convergence (ROC, for short).
The ROC is defined as the set of all points z where the series X(z) converges.
The convergence properties of power series of the form (9.1) is a well-studied subject
in complex analysis. Appendix 9.A provides a summary of the main results in this respect
and some of the subtleties that arise. Based on the overview in the appendix, it is sufficient
for our treatment in this book to define the ROC of a ztransform as the set of all points z
215
that satisfy the following condition:
SECTION 9.2
ROC =

z C such that
n=

|x(n)z n | <
(9.2)
where the symbol C denotes the set of complex numbers. Actually, in our treatment, the
symbol C will refer to the extended set of complex numbers, which includes the points at
z = pm as well.
Definition (9.2) states that the ROC of X(z) consists of all points z in the complex
plane for which the sequence {x(n)z n } is absolutely summable, i.e., the sum of the
magnitudes of its terms is finite. When this happens, we say that the series (9.1) defining
X(z) converges absolutely. It follows that X(z) will be well-defined and assumes finite
values at all z ROC as shown by the following simple calculation:

X

X

n
x(n)z n < for all z ROC
x(n)z
|X(z)| =

n=
n=
We now distinguish between two kinds of sequences and comment on the nature of their
regions of convergence.
9.2.1 Finite-Duration Sequences

A finite-duration sequence, x(n), is defined as a sequence whose samples are equal to
zero for all values of n outside some finite interval [na , nb ]. Accordingly, a finite-duration
sequence has a finite number of nonzero samples within the interval [na , nb ]. As was illustrated by the solution of Example 9.1, the ztransform of a finite-duration sequence
generally involves either a finite number of powers of z or a finite number of powers of
z 1 or a finite number of powers of both z and z 1 (this last case was encountered in
Example 9.1). Subsequently, the ROC of a finite-duration sequence, x(n), is always the
entire complex plane except possibly the points z = 0 or z = as follows:
(a) The point z = 0 is excluded when x(n) is nonzero for some positive n (since then
X(z) will contain powers of z 1 ).
(b) The point z = is excluded when x(n) is nonzero for some negative n (since then
X(z) will contain powers of z).
REGION OF
CONVERGENCE
216
CHAPTER 9
z-TRANSFORM
Example 9.2 (Finite-duration sequences)

Table 9.1 lists three finite-duration sequences x(n) and the corresponding ztransforms and their
ROCs. In the first line, the ROC is the entire complex plane with the exception of point z = 0. In
the second line, the ROC is the entire complex plane with the exception of z = . In the third line,
the ROC is the entire complex plane with the exception of both z = 0 and z = . In the fourth line,
the ROC is the entire complex plane.
TABLE 9.1 Three finite-duration sequences and their z-transforms.
Sequence x(n)
x(n) = 2(n) 3(n 5)
x(n) = 4(n + 5) + 2(n)
x(n) = 4(n + 5) + 2(n) 3(n 5)
x(n) = 2(n)
z-transform X(z)
X(z) = 2 3z 5
X(z) = 4z 5 + 2
X(z) = 4z 5 + 2 z 5
X(z) = 2
ROC
z 6= 0
|z| <
0 < |z| <
zC
9.2.2 Infinite-Duration Sequences

An infinite-duration sequences is defined as a sequence with infinitely many nonzero samples. Such sequences can be divided into three categories see Fig. 9.2:
(a) A right-sided sequence x(n) is one for which x(n) = 0 for n < no for some finite
no . In other words, the nonzero samples of x(n) exist to the right of some finite time
instant no see Fig. 9.2. The value of no can be positive, negative, or zero. For
example, the sequence
x(n) = (0.5)n u(n + 3)
has infinite-duration. Moreover, it is a right-sided sequence since the samples of
x(n) are zero for n < 3.
(b) A left-sided sequence is one for which x(n) = 0 for n > no for some finite no . In
other words, the nonzero samples of x(n) exist to the left of some finite time instant
no see Fig. 9.2. The value of no can be positive, negative, or zero. For example,
the sequence
x(n) = (0.5)n u(n + 3)
has infinite-duration. Moreover, it is a left-sided sequence since the samples of x(n)
are zero for n > 3.
(c) A two-sided sequence is one that is neither right-sided nor left-sided. For example,
the sequence
x(n) = (0.5)n
is two-sided.
217
SECTION 9.2
REGION OF
CONVERGENCE
x(n)
right-sided sequence
n
no
x(n)
left-sided sequence
no
FIGURE 9.2 The top plot illustrates the domain of nonzero samples of right-sided sequences,
namely, to the right of some no . Likewise, the bottom plot illustrates the domain of of nonzero
samples of left-sided sequences, namely, to the left of some no .
It turns out that the ROC of infinite-duration sequences consists of discs or rings in the
complex domain as follows see Fig. 9.3:
(a) The ROC for right-sided sequences is always the exterior of a disc, namely, |z| > r
for some positive real number r.
(b) The ROC for left-sided sequences is always the interior of a disc, namely, |z| < r
for some positive real number r.
(c) The ROC for two-sided sequences is always a ring, namely, r1 < |z| < r2 , for some
positive real numbers {r1 , r2 }.
(d) In all three cases above, the points z = 0 or z = may or may not be included in
the ROC.
Proof: In order to justify the nature of the ROCs shown in Fig. 9.3 we proceed as follows. We first
express the complex argument z in polar form, say, z = rej . Then,
X
n=
|x(n)z n | =
X
n=
|x(n)|r n
It follows from this equality that if a value of z belongs to the ROC, then all values of z that lie
on the same circle of radius r also belong to the ROC. This observation establishes that the ROC
consists of a collection of concentric circles.
Now assume x(n) is a right-sided sequence with a possibly negative no . Then its ztransform
can be expressed as
X(z) =
1
X
n=no
x(n)z n +
X
n=0
x(n)z n
218
Im
CHAPTER 9
left-sided sequence
z-TRANSFORM
r
Re
right-sided sequence
Im
r
Re
Im
two-sided sequence
r2
r1
Re
FIGURE 9.3 The regions of convergence of left-sided sequences (top), right-sided sequences
(middle), and two-sided sequences (bottom).
When no is negative, the first sum converges for all z except at z = . When no is positive, the
first sum does not exist and z = is not excluded from the ROC. As for the second sum, assume a
point z of magnitude r belongs to the ROC and, hence,
X
n=0
|x(n)|r n <
Then by the previous argument, all points z of same magnitude r belong to the ROC. Now let z1 be
any point with |z1 | > r, say |z1 | = r1 . It follows that r1n < r n . Therefore,
X
n=0
|x(n)|r1n <
X
n=0
|x(n)|r n <
This means that all such z1 will also belong to the ROC. We therefore conclude that the ROC of a
right-sided sequence is necessarily the exterior of a disc.
Similarly, we can argue that the ROC of a left-sided sequence is the interior of a disc. As for a
two-sided sequence, we can express it as the sum of left- and right-sided sequences. Hence, its ROC
will be the intersection of the corresponding ROCs, which in general is a ring (when the intersection
exists).
219
9.3 EXPONENTIAL SEQUENCES
SECTION 9.3
We now illustrate the previous concepts by considering the important case of exponential
sequences.
Right-Sided Exponential Sequence
We start with a right-sided exponential sequence
x(n) = n u(n)
(9.3)
Figure 9.4 illustrates the behavior of x(n) for two choices of over the interval 0 n
10; the samples of the sequence are zero over n < 0. In one case, = 1/2 and the
sequence is seen to decay towards the value zero as n increases. In the other case, = 2
and the sequence is seen to grow unbounded as n increases.
n
x(n)=(1/2) u(n)
1
0.5
10
10
n
n
x(n)=2 u(n)
1000
500
4
n
FIGURE 9.4 The top plot shows the decaying exponential sequence x(n) = (1/2)n u(n), while
the bottom plot shows the growing exponential sequence x(n) = 2n u(n). Both sequences are shown
over the interval 0 n 10.
The ztransform of the exponential sequence (9.3) is given by

X(z) =
=
=
n u(n)z n
n=
X
n n
n=0
(z 1 )n
n=0
The terms (z 1 )n that appear in the above series are the terms of a geometric sequence
whose first element is equal to 1 and whose ratio is z 1 . Therefore, as long as |z 1 | <
EXPONENTIAL
SEQUENCES
220
1, the series converges to
CHAPTER 9
z-TRANSFORM
X(z) =
1
z
=
1 z 1
z
(9.4)
This means that the region of convergence is given by

ROC = { z C such that |z| > || }
We thus arrive at the following transform pair:
n u(n)
1
z
=
1 z 1
z
for |z| > ||
(9.5)
The sequence x(n) = n u(n) is a right-sided sequence. As we argued in the previous

section, the regions of convergence of such sequences are always the exterior of a disc, and
the above result is consistent with this property.
Left-Sided Exponential Sequence
Now consider the left-sided exponential sequence
x(n) = n u(n 1)
(9.6)
Figure 9.5 illustrates the behavior of x(n) for two choices of over the interval 10
n 1; the samples of the sequence are zero over n 0. In one case, = 1/2 and
the sequence is seen to grow unbounded as n decreases. In the other case, = 2 and the
sequence is seen to decay as n decreases.
n
x(n)=(1/2) u(n1)
0
500
1000
10
n
n
x(n)=2 u(n1)
0
0.25
0.5
10
5
n
FIGURE 9.5 The top plot shows the growing exponential sequence x(n) = (1/2)n u(n 1),
while the bottom plot shows the decaying exponential sequence x(n) = 2n u(n 1). Both
sequences are shown over the interval 10 n 1.
221
The z-transform of the left-sided exponential sequence (9.6) is given by
SECTION 9.3
X(z) =
n=
=
=
u(n 1)z
1
X
n z n
n=
X
1
EXPONENTIAL
SEQUENCES
z)n
n=1
1
z
1 1 z
1
1 z 1
z
z
=
=
=
as long as |1 z| < 1. This means that the region of convergence is now

ROC = { z C such that |z| < || }
and we arrive at the transform pair
n u(n 1)
z
1
=
1
1 z
z
for |z| < ||
(9.7)
Observe that the sequence x(n) = n u(n1) is a left-sided sequence. As we argued in

the previous section, the regions of convergence of such sequences are always the interior
of a disc and the result we arrived at is consistent with this conclusion.
Comparing (9.5) and (9.7), note the interesting fact that the two different sequences
x(n) = n u(n) and x(n) = n u(n 1) have the same z-transform albeit with different regions of convergence. This fact highlights the important point that knowledge of
the ztransform of a sequence alone does not uniquely identify the sequence unless more
information is provided such as the ROC of the transform.
Example 9.3 (Unit-step sequence)

The ztransforms of the unit-step sequences x(n) = u(n) and x(n) = u(n1) can be obtained
as special cases from (9.5) and (9.7) by setting = 1. Thus, note that
u(n)
1
z
=
1 z 1
z1
for |z| > 1
(9.8)
and
u(n 1)
1
z
=
1 z 1
z1
for |z| < 1
(9.9)
222
Example 9.4 (Two sequences)
CHAPTER 9
z-TRANSFORM
Let us determine the sequence x(n) whose ztransform is

X(z) =
z
z 1/2
We know from the results (9.5) and (9.7) that x(n) could be either
x(n) = (1/2)n u(n)
(a right-sided sequence)
or
x(n) = (1/2)n u(n 1)
(a left-sided sequence)
If we were additionally given the ROC of X(z), say as |z| > 1/2, then we would be able to conclude
that the actual sequence is the right-sided sequence
x(n) = (1/2)n u(n)
Two-Sided Exponential Sequence

Consider now the two-sided sequence
x(n) = n u(n) + n u(n 1)
(9.10)
Figure 9.6 illustrates the behavior of x(n) for two choices of and over the interval
10 n 10. In one case, we use = 1/2 and = 2 and the samples of the sequence
are seen to decay to zero as n approaches . In the other case, we use = 2 and
= 1/2 and the samples of the sequence are seen to grow unbounded as n approaches
.
n
x(n) = (1/2) u(n) + 2 u(n1)

1
0.5
0
10
0
n
10
10
x(n)=2n u(n) + (1/2)n u(n1)

1000
500
0
10
0
n
FIGURE 9.6
The top plot shows the decaying two-sided exponential sequence x(n) =
(1/2)n u(n) + 2n u(n 1), while the bottom plot shows the unbounded two-sided exponential
sequence x(n) = 2n u(n) + (1/2)n u(n 1). Both sequences are shown over the interval
10 n 10.
223
The ztransform of the two-sided exponential sequence (9.10) is given by
SECTION 9.4
X(z) =
=
n=
(z 1 )n +
n=0
=
=
PROPERTIES
OF THE
z-TRANSFORM
[n u(n) + n u(n 1)]z n
( 1 z)n
n=1
1
1
1 z 1
1 z 1
z
z
z z
as long as |z 1 | < 1 and | 1 z| < 1. This means that the region of convergence is now
given by
ROC = { z C such that || < |z| < || }
which is a nonempty set whenever || > ||; otherwise, the ztransform of the two-sided
sequence will not exist. Therefore, we arrive at the transform pair
x(n) = n u(n) + n u(n 1)
1
z
z
1
1 z 1
1 z 1
z z
for || < |z| < ||
(9.11)
The sequence x(n) = n u(n) + n u(n 1) is a two-sided sequence. As we argued in

the previous section, the regions of convergence of such sequences are always rings and
the above result is consistent with this conclusion.
9.4 PROPERTIES OF THE Z-TRANSFORM

The ztransform has several important properties that can be easily verified from its definition. A summary of these properties is given in Table 9.2 with the corresponding regions of
convergence. For example, the first two lines of the table start from two generic sequences
x(n) and y(n) with ROCs defined by
Rx = {r1 < |z| < r2 }
and
Ry = {r < |z| < r }
respectively, and then the subsequent lines provide the ztransforms of combinations and
transformations of these sequences along with the corresponding ROCs.
9.4.1 Linearity
Consider, for instance, the third line of the table. It states that the ztransform of a linear
combination of two sequences, namely, ax(n) + by(n), for any two scalars a and b, is
given by the same linear combination of their respective ztransforms, i.e.,
ax(n) + by(n) aX(z) + bY (z)
(9.12)
224
CHAPTER 9
z-TRANSFORM
But what about the ROC of the combination? Obviously, both X(z) and Y (z) need to exist
in order for the combination aX(z) + bY (z) to be well-defined. This means that all points
z Rx Ry should belong to the ROC of aX(z) + bY (z). The third line of the table
indicates, however, that the ROC can be larger than Rx Ry . This is because the points
z = 0 or z = may be included, as illustrated by Example 9.5.
TABLE 9.2 Properties of the z-transform.
Sequence
z -transform
ROC
1.
x(n)
X(z)
Rx = {r1 < |z| < r2 }
2.
y(n)
Y (z)
Ry = {r < |z| < r }
3.
ax(n) + by(n)
aX(z) + bY (z)
{Rx Ry } plus
possibly z = 0 or z =
linearity
Rx except possibly
z = 0 or z =
time-shifts
Property
x(n n0 )
z n0 X(z)
5.
an x(n)
X(z/a)
|a|r1 < |z| < |a|r2
exponential
modulation
6.
x(n)
X(1/z)
1/r2 < |z| < 1/r1
time reversal
7.
nx(n)
4.
dX(z)
dz
Rx except possibly
z = 0 or z =
linear modulation
8.
x (n)
[X(z )]
Rx
conjugation
9.
x(n) y(n)
X(z)Y (z)
{Rx Ry } plus
possibly z = 0 or z =
convolution
Proof: Let w(n) = ax(n) + by(n). Then

W (z)
w(n)z n
n=
[ax(n) + by(n)]z n
n=
x(n)z n + b
n=
y(n)z n
n=
aX(z) + bY (z)
for all values of z Rx Ry . The ROC of W (z) may include z = 0 or z = or both depending
on whether powers of z 1 or z disappear from the combination aX(z) + bY (z), as the next example
illustrates.
225
SECTION 9.4
Example 9.5 (Combining two sequences)
PROPERTIES
OF THE
z-TRANSFORM
Consider the sequences

x(n) = (n) 2(n 1) X(z) = 1 2z 1 with Rx = { z 6= 0 }
and
y(n) = 3(n+1)+5(n)+2(n1) Y (z) = 3z+5+2z 1 with Ry = { 0 < |z| < }
Consider now the linear combination w(n) = x(n) + y(n), which evaluates to
w(n) = 3(n + 1) + 6(n)
The ztransform of w(n) is given by
W (z) = 3z + 6 with ROC = { |z| < }
It is seen that the ROC of W (z) is larger than Rx Ry since
Rx Ry = { 0 < |z| < }
which excludes z = 0, while z = 0 is included in the ROC of W (z); this is because when x(n)
and y(n) are added together, the terms 2z 1 and 2z 1 that appear individually in X(z) and Y (z),
respectively, cancel each other from W (z).
Example 9.6 (Inverse-transformation)
Since different sequences can have identical z-transforms, it is therefore important to specify the
ROC of a z-transform in order to be able to recover the original sequence uniquely.
For example, let us determine the sequence x(n) whose ztransform is
X(z) =
z
z
+
z2
z+3
for the three possibilities of ROCs:

i) |z| > 3
ii) |z| < 2
iii) 2 < |z| < 3
iv) |z| > 2
To begin with, since X(z) is the sum of two first-order terms of the form z/(z 2) and z/(z + 3),
we note that there are four possible linear combinations of left- and right-sided sequences that could
have given rise to X(z):
x(n)
2n u(n) + (3)n u(n)
x(n)
2n u(n) (3)n u(n 1)
x(n)
x(n)
(right-sided sequence)
n
2 u(n 1) + (3) u(n)
2n u(n 1) (3)n u(n 1)
(two-sided sequence)
(two-sided sequence)
(left-sided sequence)
The ROC for the first possibility is |z| > 3, while the ROC for the second possibility is 2 < |z| < 3,
and the ROC for the fourth possibility is |z| < 2. The third possibility has an empty ROC and,
therefore, the sequence does not have a ztransform. We then conclude that:
1. For |z| > 3, the sequence is x(n) = 2n u(n) + (3)n u(n).
2. For |z| < 2, the sequence is x(n) = 2n u(n 1) (3)n u(n 1).
3. For 2 < |z| < 3, the sequence is x(n) = 2n u(n) (3)n u(n 1).
226
CHAPTER 9
4. For |z| > 2, there is no valid sequence since this is not a valid ROC. None of the choices:
right-sided, left-sided, two-sided, or finite-duration sequences, can lead to this ROC.
z-TRANSFORM
Example 9.7 (Poles)
In the previous example, the answer to the last part (namely, recognizing that there is no sequence
x(n) with the given X(z) and whose ROC is |z| > 2) can also be motivated as follows. Recall that
the ROC is the set of all points z in the complex plane where the z-transform, X(z), is well-defined.
Hence, for rational z-transforms, the ROC should exclude the poles of X(z), since a pole is defined
as a point where the rational function X(z) evaluates to . The poles of X(z) given in the previous
example, namely,
z
z
X(z) =
+
z2
z+3
are located at the points z = 2 and z = 3. The region |z| > 2 includes the pole at 3 and, therefore,
it cannot be a valid ROC!
9.4.2 Time Shifts

Consider now the fourth line in Table 9.2. It establishes the transform pair
x(n no )
z no X(z)
(9.13)
In other words, if the original sequence x(n) is shifted in time by an amount no (where
no can be positive or negative), then the corresponding ztransform is modified by multiplying it by z no . The ROC of the time-shifted sequence, x(n no ), will coincide with
the ROC of the original sequence x(n), with the exception of the points z = 0 or z = ,
which may be included or excluded as a result of the shift operation.
Proof: Let w(n) = x(n no ). Then
W (z)
w(n)z n
n=
n=
x(n no )z n
x(k)z (k+no ) ,
k=
no
using k = n no
!
x(k)z
k=
z no X(z)
for all values of z Rx . The ROC of W (z) may include or exclude the points z = 0 or z =
or both depending on whether powers of z 1 or z disappear from z no X(z), as the next example
illustrates.
227
Example 9.8 (Time-shifted exponential sequence)
SECTION 9.4
Property (9.13) is particulary useful in inverse-transform operations. Consider, for instance, the ztransform
1
, |z| > 0.5
X(z) =
z 0.5
with the ROC identified as the set of all points z in the complex plane satisfying |z| > 0.5. By
multiplying and dividing the given X(z) by an appropriate power of z, we can rewrite it as
X(z) =
z 1 z
= z 1
z 0.5
z
z 0.5
Now we know that the inverse transform of z/(z 0.5) is 0.5n u(n) over |z| > 0.5. By taking the
additional z 1 factor into account, and in view of the time-shift property (9.13), we then conclude
that
1
(0.5)n1 u(n 1)
z 0.5
Example 9.9 (A delayed exponential sequence)

Consider the exponential sequence
x(n) = (0.5)n+3 u(n + 3)
The samples of this sequence are nonzero for n 3; the sequence is obtained by shifting x (n) =
(0.5)n u(n) to the left by 3 units of time, i.e.,
x(n) = x (n + 3)
We already know what the ztransform of x (n) is, namely,
X (z) =
z
,
z 0.5
|z| > 0.5
Using property (9.13), we conclude that the ztransform of x(n) is

X(z) = z 3 X (z) =
z4
z 0.5
Observe, however, that the sequence x(n) contains samples that occur at negative time, namely,
x(3), x(2), and x(1). Therefore, using the definition (9.1) for the ztransform, we know that
X(z) = x(3)z 3 + x(2)z 2 + x(1)z +
x(n)z n
n=0
The three leading terms in the above series, namely,

x(3)z 3 + x(2)z 2 + x(1)z
show that the point z = should be excluded from the ROC. We therefore conclude that
X(z) =
z4
,
z 0.5
0.5 < |z| <
The fact that z = should be excluded from the ROC is also obvious from the expression for X(z);
observe that it is not defined at z = .
PROPERTIES
OF THE
z-TRANSFORM
228
Consider the fifth line in Table 9.2. It establishes the transform pair
CHAPTER 9
z-TRANSFORM
an x(n)
X(z/a)
(9.14)
In other words, if the original sequence x(n) is multiplied by the exponential sequence an ,
for some nonzero constant a, then the corresponding ztransform is modified by replacing
the independent variable z by z/a. The ROC of the exponentially-weighted sequence,
an x(n), is given by
ROC = {z C such that |a|r1 < |z| < |a|r2 }
We refer to the multiplication by the exponential sequence by saying that the original sequence is being exponentially weighted or modulated. The latter terminology of modulation is more appropriate when the scalar a is a complex number.
Proof: Let w(n) = an x(n). Then
W (z)
w(n)z n
n=
an x(n)z n
n=
x(n)(z/a)n
n=
x(n)(z )n ,
using z = z/a
n=
X(z )
for all values of z Rx .

Example 9.10 (Alternating signs)
Consider a sequence x(n) with generic ROC given by Rx = {r1 < |z| < r2 }. Let us determine the
ztransform of the sequence (1)n x(n). This sequence amounts to reversing the signs of all oddindexed samples of x(n). Using property (9.14) with a = 1, we find that the new ztransform is
given by
(1)n x(n) X(z),
with ROC = Rx
(9.15)
Example 9.11 (Sinusoidal sequences)
Let us determine the ztransform of the sinusoidal sequence

x(n) = cos(o n)u(n)
for some angular frequency o . For this purpose, we first invoke Eulers relation (3.11) to express
x(n) as the linear combination of two causal exponential sequences:
x(n) =
1 jo n
1
e
u(n) + ejo n u(n)
2
2
The first term on the right-hand side can be interpreted as an exponentially-weighted version of
u(n) with a = ejo . Likewise, the second term on the right-hand side can be interpreted as an
exponentially-weighted version of u(n) with a = ejo . Therefore, using the linearity property
(9.12), the exponential weighting property (9.14), and the ztransform of the unit-step sequence
from Example 9.3, namely,
u(n)
1
,
1 z 1
|z| > 1
we get
X(z)
=
=
1
1
1
+
2 1 (z/ejo )1
1 (z/ejo )1

1
1
1
1
+
, for |z| > 1
2 1 ejo z 1
2 1 ejo z 1
1
2
Combining terms we conclude that

cos(o n)u(n)
1 z 1 cos o
,
1 2z 1 cos o + z 2
for |z| > 1
(9.16)
z 1 sin o
,
1 2z 1 cos o + z 2
for |z| > 1
(9.17)
Similarly, we can verify that

sin(o n)u(n)
Example 9.12 (Exponential modulation of sinusoidal sequences)
Let us determine the ztransform of the sequence

x(n) = an cos(o n)u(n)
for some angular frequency o . We already know from the solution to Example 9.11 that
cos(o n)u(n)
Using property (9.14) we get
1 z 1 cos o
, for |z| > 1
1 2z 1 cos o + z 2
an cos(o n)u(n)
1 az 1 cos o
, for |z| > |a|
1 2az 1 cos o + a2 z 2
(9.18)
an sin(o n)u(n)
az 1 sin o
, for |z| > |a|
1 2az 1 cos o + a2 z 2
(9.19)
Likewise,
9.4.4 Time Reversal

Consider the sixth line in Table 9.2. It establishes the transform pair
x(n)
X(1/z)
(9.20)
In other words, if the original sequence x(n) is reversed in time (i.e., flipped around the
vertical axis), then the corresponding ztransform is modified by replacing the indepen-
229
SECTION 9.4
PROPERTIES
OF THE
z-TRANSFORM
230
CHAPTER 9
z-TRANSFORM
dent variable z by its inverse, 1/z. The ROC of the time-reversed sequence, x(n), is
given by
ROC = {z C such that 1/r2 < |z| < 1/r1 }
Proof: Let w(n) = x(n). Then
W (z)
w(n)z n
n=
x(n)z n
n=
x(k)z k ,
using k = n
k=
x(k)(1/z)k
k=
X(1/z)
for all values of 1/z Rx , i.e., r1 < 1/|z| < r2 or, equivalently,
1/r2 < |z| < 1/r1
Example 9.13 (Time-reversing an exponential sequence)

Consider the sequence studied in Example 9.9, namely,
x(n) = (0.5)n+3 u(n + 3)
X(z) =
z4
,
z 0.5
0.5 < |z| <
Reversing the sequence in time corresponds to replacing n by n. Thus, consider the sequence
x (n) = x(n) = (0.5)n+3 u(n + 3)
Using property (9.20), we find that the new ztransform is given by
X (z) = X(1/z) =
z 4
2z 3
=
,
0.5
z 2
z 1
0 < |z| < 2

Consider the seventh line in Table 9.2. It establishes the transform pair
nx(n)
dX(z)
dz
(9.21)
In other words, if the original sequence x(n) is modulated by the linear sequence n, then
the corresponding ztransform is obtained from the derivative of X(z) via multiplication
by z. The ROC of the linearly modulated sequence, nx(n), is Rx except possibly z = 0
or z = .
231
Proof: Let w(n) = nx(n) and recall first the definition of X(z):
X(z) =
SECTION 9.4
PROPERTIES
OF THE
z-TRANSFORM
x(n)z n
n=
for all values of z Rx . The series X(z) is absolutely summable over Rx . Thus, differentiating it
with respect to z we can write
dX(z)
dz
x(n)
n=
x(n)
n=
so that
z
dz n
dz
nz n1
z 2n
X
dX(z)
=
nx(n)z n = W (z)
dz
n=
And the ROC of W (z) coincides with the ROC of X(z) except possibly at z = 0 or z = .
Example 9.14 (Linearly-modulated exponential sequence)
Let us determine the ztransform of x(n) = nn u(n). We already know that

z
1
=
,
1 z 1
z
n u(n)
Therefore,
z
z
=
,
z
(z )2
for |z| > ||
z
z 1
=
,
(1 z 1 )2
(z )2
for |z| > ||
nn u(n) z
d
dz
In other words,
nn u(n)
for |z| > ||
(9.22)
In a similar vein we find that

nn u(n 1)
z 1
z
=
,
(1 z 1 )2
(z )2
for |z| < ||
(9.23)
Table 9.3 summarizes several of the transform pairs that have been encountered so far in
our exposition.
9.4.6 Complex Conjugation

Consider the eighth line in Table 9.2. It establishes the conjugation property:
x (n)
[X(z )]
(9.24)
In other words, if the samples of the time-domain sequence are complex conjugated, then
the corresponding ztransform is obtained by replacing z by z and subsequently conjugating X(z ). The ROC of x (n) continues to be Rx .
232
TABLE 9.3 Some useful z-transform pairs and their ROCs.
CHAPTER 9
z-TRANSFORM
Sequence
ROC
z-Transform
(n)
u(n)
z
z1
|z| > 1
n u(n)
z
z
|z| > ||
n u(n 1)
z
z
|z| < ||
nn u(n)
z
(z )2
|z| > ||
nn u(n 1)
z
(z )2
|z| < ||
complex plane
cos(o n)u(n)
z 2 z cos o
z 2 2z cos o + 1
|z| > 1
sin(o n)u(n)
z sin o
z 2 2z cos o + 1
|z| > 1
n cos(o n)u(n)
z 2 z cos o
z 2 2z cos o + 2
|z| > ||
n sin(o n)u(n)
z sin o
z 2 2z cos o + 2
|z| > ||
Proof: Let w(n) = x (n). Then

W (z)
w(n)z n
n=
x (n)z n
n=
"
x(n)(z )
n=
[X(z )]
for all z Rx .
In particular, note that if the sequence x(n) is real-valued, then it should hold that
X(z) = [X(z )]
(for real-valued sequences)
(9.25)
233
Example 9.15 (Illustration of conjugation property)
SECTION 9.4
PROPERTIES
OF THE
z-TRANSFORM
Consider the complex-valued sequence

x(n) = (n) + j(n 1)
Its ztransform is given by
X(z)
1 + jz 1
1 + j(1/z),
ROC = {entire complex plane except z = 0}
Now observe that if we replace z by z we get

X(z ) = 1 + j(1/z )
If we further conjugate X(z ) we obtain
[X(z )] = 1 j(1/z) = 1 jz 1
which is the ztransform of the complex conjugated sequence
x (n) = (n) j(n 1)

Consider the ninth line in Table 9.2. It establishes the transform pair
x(n) y(n)
X(z)Y (z)
(9.26)
In other words, convolution in the time domain amounts to multiplication in the transform
domain. The ROC of the linear convolution is Rx Ry plus possibly z = 0 or z = .
Proof: Let
w(n) = x(n) y(n) =
X
k=
x(k)y(n k)
234
Then,
CHAPTER 9
z-TRANSFORM
W (z)
w(n)z n
n=
n= k=
k= n=
x(k)y(n k)z n
n=
x(k)z
n=
x(k)z
k=
y(n k)z n z k
!
y(n k)z
(nk)
y(n )z
n A
n =
! 0
x(k)z
k=
n=
k=
y(n k)z n
x(k)z k
k=
x(k)
k=
x(k)y(n k)z n
using n = n k
y(n )z
n A
n =
X(z)Y (z)
for all z Rx Ry plus possibly z = 0 or z = .

Let us re-consider the two sequences x(n) and h(n) from Example 6.2 and proceed to evaluate their
linear convolution
n
o
n
o
y(n) = 2, 1 , 1, 2
0 , 1, 2
where we are using the box notation to indicate the location of the sample at time n = 0. The
sequences are reproduced in Fig. 9.7.
x(n)
h(n)
2
1
1
2
FIGURE 9.7 Two sequences x(n) and h(n) whose convolution we are evaluating by means of
the ztransform.
The ztransform of the sequence x(n) =
235
2, 1 , 1, 2 is given by
X(z) = 2z + 1 z 1 + 2z 2 ,
SECTION 9.4
PROPERTIES
OF THE
z-TRANSFORM
ROC = {0 < |z| < }
0 , 1, 2 is given by
and the ztransform of the sequence h(n) =
H(z) = z 1 + 2z 2 ,
ROC = {|z| > 0}
Multiplying both ztransforms we find that

Y (z) = 2 3z 1 + z 2 + 4z 4 ,
ROC = {|z| > 0}
In this case, the ztransform of the linear convolution sequence, y(n), only has a few terms in its
expansion and we can use the definition (9.1) of the ztransform to conclude that
y(0) = 2, y(1) = 3, y(2) = 1, y(3) = 0, y(4) = 4
and all other samples are zero. Observe how the result is obtained here more directly than using the
graphical method of Example 6.2.
Example 9.17 (Convolving two other sequences)

Let us now evaluate the convolution
n
y(n) = u(n)
1
2
u(n 1)
The first step is to determine the ztransforms of the sequences u(n) and 0.5n u(n 1). We already
know from Example 9.3 that the ztransform of the unit-step sequence is given by
U (z) =
z
,
z1
|z| > 1
In order to determine the ztransform of the sequence 0.5n u(n 1) we note that we can express it
as
n
n1
1
1 1
u(n 1) =
u(n 1)
2
2 2
where the sequence (0.5)n1 u(n 1) can be identified as a time-delayed version of 0.5n u(n) and,
hence,
n
z
1/2
1
1
=
, |z| > 1/2
u(n 1) z 1
2
2
z 1/2
z 1/2
It follows that the z-transform of y(n) is

Y (z) =
z
1/2
,
z 1 z 1/2
|z| > 1
with the ROC found by taking the intersection of the individual ROCs:
ROC = { |z| > 1 } { |z| > 1/2 } = { |z| > 1 }
In order to determine the sequence y(n) we need to inverse transform Y (z), i.e., we need to know
how to determine the time-domain sequence from its ztransform. We shall learn how to do this
reverse operation by means of partial fractions in Chapter 10. For now, it suffices to note that we can
express Y (z) as the sum of two rational functions as follows:
Y (z) =
1
1/2
,
z1
z 1/2
|z| > 1
236
Its inverse transform can be determined by noting that

CHAPTER 9
z-TRANSFORM
Y (z) = z 1
z
z1
1 1
z
2
z
z 1/2
|z| > 1
Now since the ROC is the outside of a disc, the inverse transform of both terms appearing in Y (z)
need to be right-sided sequences. More specifically, we have

z 1
1 1
z
2
z
z1
z
z 1/2

u(n 1)
1
2
n1
1
2
u(n 1)
so that
y(n)
u(n 1)
1
2
n1
1
2
n
1
2
u(n 1)
u(n 1)
9.5 EVALUATING SERIES

One useful application of the z-transform technique is the evaluation of series.
Example 9.18 (Evaluating a series)
Let us evaluate the series
S=
n
X
1
n=0
cos

For this purpose, we first note that S can be identified as the value at the point z = 2 of the ztransform of the sequence

n u(n)
x(n) = cos
3
Indeed, by definition,
X(z)
x(n)z n =
n=
cos

n=0
n z n
so that if z is set to z = 2 we get

X(z)|z=2 =
n
X
1
n=0
cos

= S
This conclusion obviously requires the point z = 2 to belong to the ROC of X(z). Now we know
from Example 9.4 that
cos

n u(n) X(z) =
1 0.5z 1
,
1 z 1 + z 2
and since z = 2 belongs to the ROC, we conclude that

S=
1 0.5z 1
= 0
1 z 1 + z 2 z=2
|z| > 1
Example 9.19 (A useful series)

Let us now evaluate the following series
S=
nn ,
n=0
|| < 1
Thus, note that S can be identified as the value at the point z = 1 of the z-transform of the sequence
x(n) = nn u(n)
Indeed, by definition,
X(z)
x(n)z n =
n=
nn z n
n=0
so that if z is set to z = 1 we get
X(z)|z=1 =
n=0
nn = S
This conclusion obviously requires the point z = 1 to belong to the ROC of X(z). Now we know
from Table 9.3 that
z
, |z| > ||
nn u(n) X(z) =
(z )2
and since z = 1 belongs to the ROC, we conclude that

S=
=
(z )2 z=1
(1 )2
Hence,
nn =
n=0
,
(1 )2
|| < 1
(9.27)
9.6 INITIAL VALUE THEOREM

Another useful property of the ztransform is that it allows us to determine the value of
a causal sequence x(n) at time 0 without the need to perform inverse z-transformation.
Thus, assume that x(n) is a causal sequence, namely,
x(n) = 0,
for n < 0
Then it holds that

lim X(z) = x(0)
where the point z = must belong to the ROC of X(z) for the limit to exist.
Proof: The result follows immediately from the relation
X(z) = x(0) + x(1)z 1 + x(2)z 2 + . . . ,
(9.28)
237
SECTION 9.6
INITIAL
VALUE
THEOREM
238
CHAPTER 9
z-TRANSFORM
where only negative powers of z are present due to the causality of x(n). Note that since x(n) is
causal, the ROC of X(z) is the outside of a disc. Moreover, since X(z) does not contain positive
powers of z, its ROC includes z = . In this way, the limit (9.28) is well-defined.
Example 9.20 (Initial value)

Consider the ztransform
X(z) =
0.5z
,
z 2 1.5z + 0.5
|z| > 1
Taking the limit as z we find that

lim X(z) = 0
so that x(0) = 0. We are therefore able to infer this result without determining x(n) from X(z).
Note that since the ROC is the outside of a disc, we know that x(n) is a right-sided sequence.
Subsequently, since the limit of X(z) as z is finite, we conclude that x(n) is necessarily a
causal sequence.
9.7 UPSAMPLING AND DOWNSAMPLING

We end this chapter by introducing briefly the notions of upsampling and downsampling of
a sequence, which will be useful in the study of multi-rate discrete-time systems in Chapters 2931. Multirate systems are systems that involve signals that are sampled at different
rates.
9.7.1 Upsampling
Starting from a sequence x(n), let us construct the sequence
n
,
x
L
y(n) =
0,
if
n
is integer
L
(9.29)
otherwise
We say that y(n) is obtained via time-expansion, which amounts to inserting L 1 zero
samples between two successive samples of x(n). We represent the upsampling of a
generic sequence x(n) in block diagram form as shown in Fig. 9.8.
x(n)
FIGURE 9.8
y(n)
Block diagram representation of upsampling a sequence x(n) by a factor L.
239
SECTION 9.7
Example 9.21 (Time expansion by a factor of 3)

Figure 9.9 illustrates the effect of time-expansion on a sequence x(n) using an upsampling factor of
L = 3. It is seen that the values of y(n) coincide with those of x(n/L) whenever n is a multiple of
3. It is also seen that L 1 = 2 zeros are inserted between successive samples of y(n).
x(n)
3
2
1
n
1 2 3
y(n) = x(n/3)
3
2
1
n
1 2 3 4 5 6 7 8 9
FIGURE 9.9
(bottom).
Upsampling a sequence x(n) (top) by a factor L = 3 to generate the sequence y(n)
The ztransforms of the sequences x(n) and y(n) in Fig. 9.8 are related as follows. Assume that the ROC of the sequence x(n) is given by
Rx = { z C such that r1 < |z| < r2 }
Then, it holds that
Y (z)
X zL
with ROC given by

1/L
Ry = { z C such that r1
(9.30)
1/L
< |z| < r2
(9.31)
UPSAMPLING
AND
DOWNSAMPLING
240
Proof: Using the definition (9.29), we have
CHAPTER 9
z-TRANSFORM
Y (z)
y(n)z n
n=
x(n/L)z n
n =
n/L integer
Now as n varies over the interval < n < , the ratio n/L covers all possible integer values
and, hence, all samples of the sequence x() enter into the second summation. Define the change of
variables m = n/L, whenever n/L is an integer. Then, we can write
Y (z)
x(m)z mL
m=
x(m)(z L )m
m=
X(z L )
for all values of z such that z L Rx .
9.7.2 Downsampling
Let us now consider a related operation known as downsampling. We motivate the discussion by considering first the sequence y(n) = x(2n). We represent the downsampling
operation of a generic sequence x(n) in block diagram form as shown in Fig. 9.10. The
sequence y(n) = x(2n) is shown in Fig. 9.11 starting from the same x(n) used in Example
9.21. It is seen that some samples of x(n) are now discarded. In this particular example,
every other sample of x(n) is discarded: x(0) and x(2) are maintained while x(1) and
x(3) are removed.
x(n)
FIGURE 9.10
y(n)
Block diagram representation of downsampling a sequence x(n) by a factor of 2.
241
SECTION 9.7
x(n)
UPSAMPLING
AND
DOWNSAMPLING
3
2
1
5 6
1 2 3
8 9
y(n) = x(2n)
3
2
1
2
1
3 4
FIGURE 9.11 Downsamping a sequence x(n) by a factor of 2. The dotted line are used to
illustrate which samples are maintained besides the sample at n = 0.
The ztransforms of the sequences x(n) and y(n) in Fig. 9.10 are related as follows.
Assume that the ROC of the sequence x(n) is given by
Rx = { z C such that r1 < |z| < r2 }
Then, it holds that

1
X(z 1/2 ) + X(z 1/2 )
2
x(2n)
(9.32)
with the ROC corresponding to the sequence y(n) = x(2n) given by

Ry = { z C such that r12 < |z| < r22 }
(9.33)
Proof: Introduce the sequence

b(n) =
1
[x(n) + (1)n x(n)]
2
Then b(n) = x(n) whenever n is even and b(n) = 0 whenever n is odd. Moreover, it also holds that
y(n) = b(2n). Now it follows from the properties of the ztransform that (recall Example 9.10):
B(z) =
1
[X(z) + X(z)]
2
242
and we can proceed to evaluate the ztransform of y(n) as follows:
CHAPTER 9
z-TRANSFORM
Y (z)
y(n)z n
n=
b(2n)z n
n=
b(k)z k/2 ,
using k = 2n
k =
k even
=
b(k)z k/2 ,
because for odd k we have b(k) = 0, by construction
k=
b(k)(z 1/2 )k
k=
=
=
B(z 1/2 )
i
1h
X(z 1/2 ) + X(z 1/2 )
2
for all values of z such that z 1/2 Rx .
M-fold Downsampling
More generally, we can perform M fold downsampling by considering
y(n) = x(M n)
(9.34)
for positive integers M see Fig. 9.12.
x(n)
FIGURE 9.12
y(n)
Block diagram representation of downsampling a sequence x(n) by a factor of M .
It holds in this general case that the ztransform of y(n) is related to the ztransform of
x(n) as follows:
Y (z) =

P
1 M1
k 1/M
X WM
z
M k=0
(9.35)
where WM denotes the M th root of unity, i.e.,
WM = ej2/M
(9.36)
and the ROC corresponding to y(n) is given by

Ry = { z C such that r1M < |z| < r2M }
(9.37)
243
Proof: Define the auxiliary sequence

(
SECTION 9.8
x(n),
0,
b(n) =
APPLICATIONS
n = multiple of M
otherwise
Then y(n) = b(M n). Moreover,

Y (z)
y(n)z n
n=
b(M n)z n
n=
b(k)z k/M ,
because b(k) = 0 unless k is a multiple of M
k=
b(k)(z 1/M )k
k=
B(z 1/M )
for all values of z such that z 1/M belongs to the ROC corresponding to the sequence b(n).
We still need to relate B(z) to X(z). For this purpose, we note that we can express b(n) in terms
of x(n) as follows:
"
b(n)
M 1
1 X kn
WM
x(n)
M
k=0
i
1 h
n(M 1)
n
2n
1 + WM
x(n)
+ WM
+ . . . + WM
M
This is because when n is a multiple of M , say n = mM , we get

kn
WM
=
ej2/M
kmM
= ej2km = 1
and, consequently,
b(n)
=
=
1
[1 + 1 + 1 + . . . + 1] x(n),
M
x(n)
when n is multiple of M
On the other hand, when n is not a multiple of M , the sum below evaluates to zero
h
n(M 1)
n
2n
1 + WM
+ WM
+ . . . + WM
= 0,
when n is not multiple of M
These facts justify the expression for b(n) in terms of x(n), namely,
b(n)
i
1 h
n(M 1)
n
2n
1 + WM
+ WM
+ . . . + WM
x(n)
M
Using the modulation property (9.14) of the ztransform we now get

B(z) =
for all values z Rx .
M 1
1 X
k
X(WM
z)
M k=0
244
9.8 APPLICATIONS
CHAPTER 9
z-TRANSFORM
TO BE ADDED
Practice Questions:
1.
2.
9.9 PROBLEMS
Problem 9.1 What is the ROC for the z-transform of x(n) = u(n + 3) u(n 3)?
Problem 9.2 What is the ROC for the z-transform of x(n) = u(n + 3)
Problem 9.3 Let
X(z) =

1 n
2
u(n 3)?
1
z 1
+
z 1/4
(z 1/3)2
Describe all sequences x(n) whose z-transforms coincide with X(z).

Problem 9.4 Let
X(z) =
z 1
z 2
2
(z 1/4)
z 1/2
Describe all sequences x(n) whose z-transforms coincide with X(z).

Problem 9.5 Find the ztransforms and ROCs of the following sequences:
(a) x(n) =
(b) x(n) =
(c) x(n) =
(d) x(n) =

1 n2
u(n 4).
2

1 n2
u(n).
2

1 n2
u(n 2).
n 2

2 1 n+2
u(n).
n 2
(a) x(n) = (n 1)

1 n
2
(b) x(n) =
(c) x(n) = n
(d) x(n) = n(n

1 n+1
2
u(n 1).
u(n 2).

1 n1
u(n).
2

1 n
1) 2 u(n).
(a) x(n) = n cos
(n
3
(b) x(n) = (n 2) cos

(c) x(n) = n

1 n
2
1) u(n 2).
sin
(n 1) sin 3 (n
3
n 3 u(n).
6
1) u(n 3).
(b) x(n) =
(c) x(n) =
n u(n 1).
6

n1
u(n).
n 12
sin 3 n 2
3

1 2n2
cos 3 n u(n 2).

2
(a) x(n) = cos
Problem 9.9 Determine the ztransforms of each of the following sequences and indicate their
regions of convergence:
(a) x(n) = nu(n 1).

(b) x(n) = 1 + n
|n|
(c) x(n) =
245
n2
SECTION 9.9
u(n 1).
PROBLEMS
, with > 0.
(d) The impulse response sequence of the relaxed causal system y(n) 34 y(n1)+ 18 y(n2) =
x(n 1).
Problem 9.10 Determine the ztransforms of each of the following sequences and indicate their
regions of convergence:
(a) x(n) = n2 u(n 1).
(b) x(n) = 1 n2n2 u(n 1).

(c) x(n) = |n| , with > 0.
(d) The impulse response sequence of the relaxed causal system y(n) 14 y(n1) 18 y(n2) =
2x(n 2).
Problem 9.11 Determine the ztransforms of the following sequences:
(a) x(n) = u(2n).
(b) x(n) = { 1 , 1, 1, 1, 1, 1, 1, 1, . . .}. That is, the samples of x(n) alternate between
1 and 1.
Problem 9.12 Determine the ztransforms of the following sequences:
(a) x(n) = u(2n 2).
(b) x(n) = { 1 , 1, 1, 1, 1, 1, 1, 1, . . .}. That is, the samples of x(n) alternate between
1 and 1.
Problem 9.13 Find the ztransform of the sequence
n
1
2
x(n) = nu(n 2) +
u(n + 1)
Problem 9.14 Find the ztransform of the sequence

x(n) = n2 u(n + 2) +
Problem 9.15 Let
X(z) =
n+2
1
4
u(n 1)
z
, |z| < 1/4
z 1/4
Can you use the initial value theorem to determine x(0)?

Problem 9.16 Let
X(z) =
Find x(0).
z 51
, |z| > 1/2
(2z 1/4)(3z 1/2)5 0
Problem 9.17 Consider the sequence x(n) shown in Fig. 9.13. Define
h(n) =
1
3
x(n + 2) (n) + u(n 3),
2
2
n
h1 (n) =
1
2
h(n)u(n)
Determine H(z), the ztransform of h(n), and indicate its ROC. Find also the ztransform of
h1 (n) and its ROC.
Problem 9.18 Consider the same sequence x(n) shown in Fig. 9.13. Define
h(n) =
1
x(n + 1) +
4
n
1
2
n1
u(n 2),
h1 (n) =
1
3
h(n)u(n 1)
246
x(n)
CHAPTER 9
z-TRANSFORM
3
2
Determine H(z), the ztransform of h(n), and indicate its ROC. Find also the ztransform of
h1 (n) and its ROC.
Problem 9.19 Let
n
x(n) =
1
2
2n
u(n 1),
1
3
h(n) =
u(n 3)
Use the ztransform technique to evaluate the following sequences:

(a) x(n) h(n).

1 n
h(n).
4

x(n) cos 4 n h(n).
n
x(n) h(n 1) 14 u(n).
(b) x(n 2)
(c)
(d)
Problem 9.20 Let

n
x(n) = n
1
3
n1
u(n),
h(n) =
1
4
u(n + 1)
Use the ztransform technique to evaluate the following sequences:

(a) x(n) h(n).

1 n1
h(n 2).
3

x(n) sin 3 n h(n 1).
n
x(n) h(n 2) 13 u(n).
(b) x(n 1)
(c)
(d)
Problem 9.21 Use the ztransform technique to evaluate the value at n = 0 of the convolution
sequence
"
n1
n+1 #
1
1
u(n)
u(n 2) 1 +
2
3
Problem 9.22 Use the ztransform technique to evaluate the value at n = 0 of the convolution
sequence
"
n+1
2n+2 #
1
1
u(n 1) n
u(2n)
3
2
Problem 9.23 Let x(n) = (0.5)n u(n 1). Find the ztransform of the sequence
y(n) =
n
X
m=
mx(m)
Problem 9.24 Let x(n) = n(0.5)n2 u(n + 1). Find the ztransform of the sequence
n
X
y(n) =
PROBLEMS
m=
Problem 9.25 Let
(m 1)x(m)
n1

1
n u(n 2)
cos
2
6
Find the z-transform and the corresponding ROCs of the following sequences:
x(n) = n
(a) x(2n).
(b) x(3n).
Problem 9.26 Let
n+1
1
3
x(n) = n
sin
(n 1) u(n)
Find the z-transform and the corresponding ROCs of the following sequences:
(a) x(2n 2).
(b) x(4n).
Problem 9.27 Let
n1
x(n) = n
1
2
cos

n u(n 2)
6
and assume x(n) is upsampled by a factor of 4 to generate the sequence y(n). Find Y (z) and its
ROC.
Problem 9.28 Let
n+1
x(n) = n
1
3
sin
(n 1) u(n)
and assume x(n) is upsampled by a factor of 3 to generate the sequence y(n). Find Y (z) and its
ROC.
Problem 9.29 Evaluate the series
X
k=0
k3
(k 1)
1
2
Problem 9.30 Evaluate the series

k+2
X
1
k=2
cos

3
Problem 9.31 Use the ztransform technique to evaluate the following sums:
(a)

1 n
.
3
n2
n=2
(b)
100
P

1 n
.
3
n=0
(c)
n=999

1 n
.
2
Problem 9.32 Use the ztransform technique to evaluate the following sums:
(a)
(n 1)
n=3
(b)
1000
P
n=100

1 n
.
4

1 n1
.
2
247
SECTION 9.9
248
CHAPTER 9
(c)
P
n=9999
n2

1 n1
.
2
z-TRANSFORM

y(n) y(n 1) +
1
y(n 2) = x(n 1),
4
y(1) = 0,
y(2) = 7/2
(a) Find its zero-input response.

(b) Find the ztransform of the zero-state response of the system when x(n) =

1 n+1
2
u(n).

y(n)
1
1
y(n 1) y(n 2) = x(n 1),
8
8
y(1) = 0,
y(2) = 1
(a) Find its zero-input response.

(b) Find the ztransform of the zero-state response of the system when x(n) = 41
n
u(n).
Problem 9.35 A causal system is described by the block diagram shown in Fig. 9.14 with x(n)
denoting the input sequence and y(n) denoting the output sequence. The initial state of the system
is y(2) = 0 and y(1) = 4/3.
x(n)
y(n)
3/4
1/8
z 1
FIGURE 9.14 Block diagram for the system of Prob. 9.35.

(b) Use the ztransform technique to find the complete response of the system when x(n) =
n u(n), where || < 1.
(c) Are there choices of for which at least one of the modes of the system is not excited (i.e.,
does not appear) at the output? Describe all such s.
(d) Find the energy of the output sequence when = 1/4.
(e) Which value of results in an output sequence with smallest energy?
Problem 9.36 A causal system is described by the block diagram shown in Fig. 9.15 with x(n)
denoting the input sequence and y(n) denoting the output sequence. The initial state of the system
is y(2) = 1 and y(1) = 0.
(b) Use the ztransform technique to find the complete response of the system when x(n) =
2n u(n 1), where || < 1.
(c) Are there choices of for which at least one of the modes of the system is not excited (i.e.,
does not appear) at the output? Describe all such s.
(d) Find the energy of the output sequence when = 1/4.
(e) Which value of results in an output sequence with smallest energy?
249
SECTION 9.9
x(n)
y(n)
1/8
1/4
z 1
FIGURE 9.15 Block diagram for the system of Prob. 9.36.
x(n)
x(n)
FIGURE 9.16
y(n)
w(n)
Two separate cascades of upsampling and downsampling blocks for Prob. 9.37.
Problem 9.37 Consider the two cascades shown in Fig. 9.16 where the order of thedownsampler
n
and upsampler blocks is switched in one case relative to the other. Let x(n) = 12 u(n). Find
Y (z) and W (z). Are the two cascades equivalent?
Problem 9.38 Consider
n the system shown in Fig. 9.17 with input x(n) and output y(n). Find
Y (z) when x(n) = 12 u(n).
x(n)
y(n)
z 1
FIGURE 9.17 Block diagram representation of the system for Prob. 9.38.
Problem 9.39 Assume x(n) is a real-valued and even sequence. Show that its ztransform satisfies X(z) = X(1/z).
Problem 9.40 Assume x(n) is a real-valued and even sequence. Show that if X(z) has a zero at
z = zo and a pole at z = po , then X(z) also has a zero at z = 1/zo and a pole at z = 1/po .
PROBLEMS
250
CHAPTER 9
z-TRANSFORM
Problem 9.41 Let x(n) = (1/2)n u(n 1). Find the ztransform of the sequence nx(n) +
x2 (n 2). Specify its region of convergence. Find also the energy of this sequence.
Problem 9.42 Let x(n) = (1/3)n2 u(n + 1). Find the ztransform of the sequence n2 x(n)
x2 (n). Specify its region of convergence. Find also the energy of this sequence.
Problem 9.43 Find the inverse ztransform of the following (non-rational) functions:
(a) X(z) = cos z.
(b) X(z) = sin z.
(c) X(z) = sin z cos z.
Problem 9.44 Find the inverse ztransform of the following (non-rational) functions:
(a) X(z) = cos2 z.
(b) X(z) = sin 2z.
(c) X(z) = sin(z/2).
Problem 9.45 Consider the following complex series expansion of the natural logarithm around
the point z = 1,
X
(1)n+1 n
z , |z| < 1
ln(1 + z) =
n
n=1
Use the result to determine the sequence x(n) whose ztransform is given by
X(z) = ln(1 + z 1 ),
|z| > ||
Problem 9.46 Find the inverse transform of

X(z) = ln(1 + z 1 ),
|z| > ||
by using the differentiation property of the ztransform.

Problem 9.47 Find the right-sided sequence whose ztransform is given by
1
X(z) = e 2z
Problem 9.48 Let X (z) denote the ztransform of the sequence x(n) = n u(n), for integers
1. Show that
dX1 (z)
X (z) = z
dz
Use this recursion to determine the ztransforms, and the corresponding ROCs, of the sequences
x2 (n), x3 (n), and x4 (n).
Problem 9.49 Find the inverse ztransform of
X(z) =
z 1

1 50
2
z 50
Problem 9.50 Find the inverse ztransform of

X(z) =
z 1
1+

1 48
3
z 60
Problem 9.51 Find the ztransform of x(n) = cosh(n).

Problem 9.52 Find the ztransform of x(n) = tanh(n).
Problem 9.53 Show that upsamplers are linear but time-variant systems.
251
Problem 9.54 Show that downsamplers are linear but time-variant systems.
SECTION 9.A
CONVERGENCE
OF POWER
SERIES
9.A APPENDIX: CONVERGENCE OF POWER SERIES

The convergence of sequences, series, and power series is a well studied problem in mathematical
and complex analysis. Here we summarize some of the main results in order to shed some light on
the convergence properties of the ztransform. For the benefit of the reader, we start from some
basic definitions.
Sequences. A sequence of numbers {an } is said to converge to some value a if, and only if, for any
> 0, there exists an integer N large enough such that
|an a| < for all n > N
(9.38)
A useful equivalent characterization of the convergence of a sequence is given by Cauchys criterion,

which states that the sequence {an } convergence to some number a if, and only if, for any > 0,
there exists an integer N large enough such that
|an am | < for all n, m > N
(9.39)
That is, the terms of the sequence get close to each other.
Series. A series is defined as the sum of an infinite number of terms, say
an
n=0
The sum may or may not converge. The convergence of a series is defined in terms of the convergence
of a partial sum sequence as follows. Let
Sm =
m
X
an
(9.40)
n=0
That is, Sm is the sum of the terms up to time m. Then, the series is said to convergence to some
value S if, and only if, the sequence of partial sums {Sm } converges to S.
Absolute convergence of series. A series is said to converge absolutely if
X
n=0
|an | converges
(9.41)
where the terms {an } are replaced by their magnitudes, {|an |}. A useful result is the fact that if a
series converges absolutely then the series is convergent.
X
n=0
|an | converges =
an converges
(9.42)
n=0
We therefore say that

absolute convergence = convergence
A useful test for absolute convergence is the ratio test. Let
= lim
|an+1 |
|an |
If < 1, then the series converges absolutely. If > 1, the series diverges. The case = 1 needs to
be studied separately and no general statement can be made beforehand.
252
CHAPTER 9
Conditional convergence of series. The converse statement is not true. There are series that con-
verge but are not absolutely convergent. For example, the series
z-TRANSFORM
X
(1)n+1
n=1
can be shown to converge; but is not absolutely convergent. Series of this kind, namely convergent
series that are not absolutely convergent are said to converge conditionally.
an converges while
n=0
n=0
|an | diverges
conditional convergence
(9.43)
Reordering. Consider a sequence {an } and reorder its terms into a new sequence {bn }. If the series
an
n=0
converges absolutely, then any reordering of the terms of the series will always result in the same
value, i.e.,
an =
n=0
bn
n=0
On the other hand, if a series converges conditionally, then reordering of its terms can lead to any
result. For example, for any real number , it can be shown that there always exists a reordering of
the sequence {an } into a new sequence {bn } such that the new series will evaluate to . Therefore,
there is ambiguity associated with conditional convergence, which makes it undesirable.
Power series. A power series is defined as a series of the form
X
n=0
an (z zo )n
where z is an arbitrary complex variable and zo is some given complex number. There are values of
z for which the series converges and other values of z for which the series diverges. The following
result is well-known for such power series. One of only three possibilities may occur:
1. The power series converges absolutely for all z, except possibly at z = .
2. The power series diverges for all z 6= zo .
3. There exists an r > 0 such that the power series converges absolutely for all |z zo | < r and
diverges for all |z zo | > r.
Under the third possibility, the series may diverge or converge for points z on the circle |z zo | = r.
However, if convergence occurs, it will be conditional in this case and, therefore, ambiguous. In all
other cases, the convergence of the power series is in the absolute (and desirable) sense. For this
reason, the region of convergence of a power series is defined as the set of all points z for which the
series converges absolutely:
(
ROC =
z C such that
X
n=0
)
n
|an (z zo ) | <
The value of the radius of convergence, r, can be determined from the expression
|an+1 |
1
= lim
n |an |
r
Two-sided power series. Consider now a power series of the form
X
n=
CONVERGENCE
OF POWER
SERIES
an (z zo )n
where the sum starts from n = . We can split the above series into two separate series:
S1
S2
X
n=0
an (z zo )n
1
X
n=
253
SECTION 9.1
an (z zo )n =
X
n=1
an (z zo )n
Observe that S1 involves positive powers of (z zo ) while S2 involves negative powers of (z zo ).

If we now apply the result regarding the convergence of power series, we find that S1 converges
absolutely for all points z satisfying |z zo | < for some > 0, while S2 converges absolutely
for all points z satisfying |z zo | > for some > 0. Therefore, the ROC of the two-sided series
will be of the general form
< |z zo | <
and the values of and are found as follows:
1
|an+1 |
= lim
,
n |an |
1
=
lim
|an+1 |
|an |
CHAPTER
10
Partial Fractions
nverse transformation is the process of recovering a sequence x(n) from knowledge of

its ztransform, X(z), and the corresponding ROC. In this chapter we describe one useful
technique for performing inverse transformation for ztransforms that are rational functions of z. The procedure is known as the partial fractions method and it is best explained
by means of examples.
10.1 RATIONAL TRANSFORMS

Many of the ztransforms that arise in discrete-time signal processing are rational functions of z (or z 1 ), say, of the form:
X(z) =
b0 + b1 z 1 + b2 z 2 + . . . + bq z q
a0 + a1 z 1 + a2 z 2 + . . . + ap z p
q
P
bk z k
k=0
p
P
ak z k
k=0
B(z)
A(z)
where B(z) and A(z) denote the numerator and denominator polynomials in z 1 and have
degrees q and p, respectively. Usually, the degree of the numerator is smaller than or equal
to the degree of the denominator, i.e., q p, in which case we say that X(z) is a proper
rational function in z 1 . When this is not the case, the rational function can be written in
the alternative form
R(z)
X(z) = Q(z) +
A(z)
where Q(z) is some polynomial in z 1 obtained by dividing B(z) by A(z) with remainder
R(z) (also a polynomial in z 1 ), namely,
B(z) = Q(z)A(z) + R(z)
and where R(z)/A(z) is a proper rational function in z 1 . For example, note that we can
write
2z 1
z 2
1
=
2z
+
X(z) =
1 21 z 1
1 21 z 1
255
Copyright
256
CHAPTER 10
PARTIAL
FRACTIONS
In several instances, it is convenient to express rational ztransforms in terms of powers

of z as opposed to z 1 . Thus, the following two representations are equivalent
X(z) =
2
2z 1
=
1 21 z 1
z
1
2
where the right-most expression is obtained by multiplying the numerator and denominator
of the middle expression by z. More generally, starting from a proper rational function of
the form
X(z) =
b(0) + b(1)z 1 + b(2)z 2 + . . . + b(q)z q

, qp
a(0) + a(1)z 1 + a(2)z 2 + . . . + a(p)z p
if we multiply the numerator and denominator polynomials by z p , we arrive at an equivalent expression in terms of powers of z:
X(z) =
b(0)z p + b(1)z p1 + b(2)z p2 + . . . + b(q)z pq

a(0)z p + a(1)z p1 + a(2)z p2 + . . . + a(p)
This expression will be a proper rational function in z since the degrees of its numerator
and denominator polynomials are both equal to p.
10.2 ELEMENTARY RATIONAL FRACTIONS

Now consider a ztransform X(z) that is a rational function of z (proper or not, i.e., the
degree of its numerator polynomial may be equal to, more than, or less than the degree
of its denominator polynomial). Starting from such a transform, the method of partial
fraction expansion expresses it as the sum of simpler ztransforms whose inversions are
immediate. The simpler transforms are usually of the form:
A
A
, ...
,
z (z )2
A,
(10.1)
for some constants {A, }. During the partial fraction expansion process, these terms
may appear multiplied by some powers of z or z 1 , which would be handled during the
inversion process by invoking the time-shift property of the ztransform. The main reason
for choosing to express X(z) as the sum of such terms is that the inverse transforms of
these terms are readily available, e.g., from the listing in Table 9.3, as we explain below.
Obviously, the inverse transforms will depend on the nature of the ROCs of the terms in
(10.1). We need to consider two situations.
Region of Convergence Outside the Disc (|z| > ||)
For regions of convergence of the form |z| > ||, the inverse transforms of (10.1) will
be right-sided sequences and we can invert each term to obtain the sequences shown in
Table 10.1. Thus, consider the term
A
,
(z )2
|z| > ||
We already know from Table 9.3 that the following transform pair holds:
nn u(n)
z
, |z| > ||
(z )2
257
Now note that the desired term, A/(z )2 , is related to the above transform via
SECTION 10.2
Elementary Rational
Fractions
A
A 1
z
=
z
(z )2
(z )2
Therefore, by using the time-shift and linearity properties of the ztransform from Table 9.3, we conclude that
A
, |z| > ||
(z )2
A (n 1)n2 u(n 1)
(10.2)
which is the third line in Table 10.1. We can justify the second line in the table by using a
similar argument.
TABLE 10.1
Inverse transforms of elementary functions in terms of right-sided sequences.
ztransform
A
A
z
A
(z )2
A
A
+
z
z
sequence
ROC
A(n)
complex plane
An1 u(n 1)
|z| > ||
A (n 1)n2 u(n 1)
|z| > ||
2 |A| ||n1 cos[(n 1) + ] u(n 1)
|z| > ||
Region of Convergence Inside the Disc (|z| < ||)

For regions of convergence of the form |z| < ||, the inverse transforms of terms of the
form (10.1) will be left-sided sequences and we can invert each term to obtain the sequences shown in Table 10.2. Thus, consider again the term
A
,
(z )2
|z| < ||
We already know from Table 9.3 that the following transform pair holds:
nn u(n 1)
z
, |z| < ||
(z )2
A (n 1)n2 u(n)
and, as before, we conclude that

A
, |z| < ||
(z )2
(10.3)
which is the third line in Table 10.2. Likewise, we can justify the second line in the table
by using a similar argument.
258
CHAPTER 10
TABLE 10.2
PARTIAL
FRACTIONS
Inverse transforms of elementary functions in terms of left-sided sequences.
ztransform
sequence
ROC
A(n)
complex plane
A
z
An1 u(n)
|z| < ||
A
(z )2
A (n 1)n2 u(n)
|z| < ||
A
A
+
z
z
2 |A| ||n1 cos[(n 1) + ] u(n)
|z| < ||
Complex Terms
Sometimes, during the process of partial fraction expansion, a complex conjugate pair
{, } arises and contributes to the expansion of X(z) with a sum of the form
A
A
+
z z
(10.4)
for some constant A and its complex conjugate, A . Thus, consider initially regions of
convergence of the form |z| > ||. In this case, using the second line of Table 10.1, the
inverse ztransform of the above sum is
[An1 + A (n1) ] u(n 1)
(10.5)
If we express the complex numbers {A, p} in polar forms, say,

A = |A|ej ,
= || ej
then
An1 + A (n1)
= |A|ej ||n1 ej(n1) + |A|ej ||n1 ej(n1)

= |A| ||n1 ej((n1)+) + ej((n1)+)
= 2 |A| ||n1 cos[(n 1) + ]
It follows that
A
A
2 |A| ||n1 cos[(n 1) + ] u(n 1) , |z| > ||
+
z z
(10.6)
A similar argument will show that for regions of convergence of the form |z| < ||, the
following conclusion holds:
A
A
2 |A| ||n1 cos[(n 1) + ] u(n) , |z| < ||
+
z z
(10.7)
10.3 PARTIAL FRACTIONS EXPANSION
259
SECTION 10.3
We can now proceed to explain the procedure for inverse transformation via partial fractions expansion. Thus, given a rational z-transform, X(z), and its ROC, the following
are general guidelines for its inversion by means of partial fractions. With time, as the
reader becomes more comfortable with the ztransform and its properties, the reader will
develop personal preferences and variations for some of the steps below.
(1) First, express the rational function X(z) in terms of positive powers of z in both the
numerator and the denominator. For example, starting from
X(z) =
1
1 2z 1
we rewrite it as
X(z) =
z
z2
after multiplying the numerator and denominator by z. Sometimes, it may be easier

to extract a negative power of z from X(z). For example, the transform
X(z) =
z 1 + 1
z+2
can be written as
X(z) = z 1
z+1
z+2
| {z }
X (z)
and we can proceed to invert X (z). If we succeed in inverse-transforming X (z),

then the inverse transform of X(z) is immediate since x(n) = x (n 1).
(2) In the sequel, we denote the transform that results from Step 1, in terms of positivepowers of z, by X (z). We next make sure that X (z) is strictly proper (i.e., that the
degree of its numerator is less than the degree of its denominator). If not, we can
divide the numerator by the denominator and proceed with the strictly proper part,
denoted by S(z). For example, given
X (z) =
z+1
z+2
we can write it as
X (z) = 1
1
z+2
| {z }
S(z)
where the strictly proper part is S(z) = 1/(z + 2). The additional terms that result
from this division (in the above example, it is only the term that is equal to 1), will
contribute a unit-sample sequence and possibly time-shifted versions of it. Thus, the
inverse transform of X (z) in the above example is one of two possibilities:
x (n)
x (n)
= (n) (2)n1 u(n 1)

= (n) + (2)n1 u(n)
when ROC = { |z| > 2 }

when ROC = { |z| < 2 }
PARTIAL FRACTIONS
EXPANSION
260
CHAPTER 10
PARTIAL
FRACTIONS
(3) More generally, when the denominator of S(z) is of higher-order, we determine the
roots of the denominator and use these roots to express S(z) as the sum of lowerorder terms:
(3.a) Each single root, say 1 , will contribute with a term of the form A/(z 1 ),
for some constant A to be determined.
(3.b) A double root at 2 will instead contribute with two terms of the form B/(z
2 ) and C/(z 2 )2 , for some constants {B, C}.
(3.c) For rational transforms S(z) with real coefficients, a complex pair at {, }
will contribute with two terms of the form D/(z ) and D /(z ), for
some constant D.
(3.d) We determine the coefficients (A, B, C, . . .) of the partial fraction expansion
by equating the numerators of both sides of the equality:
S(z) =
B
C
A
+
+
+ ...
z 1
z 2
(z 2 )2
Alternatively, the constant A can be determined from evaluating the product

S(z)(z 1 ) at z = 1 , i.e.,
A = S(z)(z 1 )|z=1
Likewise, C can be determined from

C = S(z)(z 2 )2 z=2
and so forth.
We illustrate the procedure with several examples.
Example 10.1 (Simple roots)

X(z) =
1 + z 1
,
1 1.5z 1 + 0.5z 2
with ROC = { |z| > 1 }
We proceed as follows in order to determine the sequence x(n):

1. We multiply both the numerator and the denominator of X(z) by z 2 to express X(z) in terms
of positive powers of z:
z2 + z
X(z) = 2
z 1.5z + 0.5
2. The resulting X(z) is not a strictly proper rational function. Dividing the numerator by the
denominator, we get
2.5z 0.5
X(z) = 1 + 2
z 1.5z + 0.5
|
{z
S(z)
where S(z) is strictly proper.
3. The roots of the denominator of S(z) are 1 and 0.5, which are simple roots. Therefore, we
can expand S(z) into partial fractions as follows:
A
B
S(z) =
+
z1
z 0.5
for some coefficients A and B to be determined. The value of A follows from:

2.5z 0.5
= 4
z 0.5 z=1
A = S(z)(z 1)|z=1 =
The value of B follows from
B = S(z)(z 0.5)|z=0.5 =
4. Therefore,
S(z) =
which we can also write as

S(z) = 4z 1
2.5z 0.5
= 1.5
z 1 z=0.5
1.5
4
z1
z 0.5
z
z1
1.5z 1
z
z 0.5
Using the second line of Table 10.1 we find that the inverse transform of S(z) over |z| > 1 is
given by
n1
1
u(n 1)
s(n) = 4u(n 1) 1.5
2
5. Consequently, the inverse ztransform of
X(z) = 1 + S(z)
over |z| > 1 is
"
x(n) = (n) +
n1 #
4 1.5
1
2
u(n 1)
Example 10.2 (Simple roots again)

Consider the same ztransform from the previous example,
X(z) =
1 + z 1
,
1 1.5z 1 + 0.5z 2
with ROC = { |z| > 1 }
We could have instead proceeded as follows. In step 1 of that example we obtained

X(z) =
z2
z2 + z
1.5z + 0.5
This transform is a proper rational function with z 2 +z in the numerator. We can alternatively express
X(z) as
X(z) = zX (z)
where
X (z) =
z+1
z 2 1.5z + 0.5
261
SECTION 10.3
PARTIAL
FRACTIONS
EXPANSION
262
CHAPTER 10
PARTIAL
FRACTIONS
is strictly proper. This amounts to extracting a z factor from the numerator of X(z). We can now
inverse transform X (z) to find x (n) and from it x(n).
1. We first expand X (z) into partial fractions:
X (z) =
A
B
+
z1
z 0.5
where the values of A and B follow from

A = X (z)(z 1)z=1 =

B = X (z)(z 0.5)z=0.5 =
2. Therefore,
z + 1
= 4
z 0.5 z=1

z + 1
= 3
z 1 z=0.5
X (z) =
4
3
z1
z 0.5
X(z) =
3z
4z
z1
z 0.5
and, hence,
3. The inverse ztransform of X(z) is then given by

n
x(n) = 4 3
1
2
u(n)
It is straightforward to verify that this expression for x(n) coincides with the one obtained in
the previous example.
Example 10.3 (Double roots)

Let us inverse transform
1
X(z) = 3 5 2
z 2 z + 2z

1
2
with ROC =
1
< |z| < 1
2
In this case, X(z) is already expressed in terms of positive powers of z and, moreover, it is a strictly
proper rational function. The denominator of X(z) has a simple root at z = 1/2 and a double root
at z = 1. Therefore, the partial fractions expansion of X(z) will be of the form
X(z) =
A
B
C
+
+
z1
(z 1)2
z
1
2
for some constants {A, B, C} to be determined. These constants can be obtained by comparing the
coefficients of the numerators on both sides of the equality:
1
z 3 25 z 2 + 2z
1
2
A(z 1)(z 21 ) + B(z 12 ) + C(z 1)2

(z 1)2 (z 12 )
(A + C)z 2 + ( 32 A + B 2C)z + ( A
2
5 2
1
3
z 2 z + 2z 2
B
2
+ C)
263
which leads to the linear system of equations

8
>
< A+C
>
:
SECTION 10.3
32 A + B
A
B2 + C
2
=
=
=
2C
PARTIAL
FRACTIONS
EXPANSION
0
0
1
The solution is given by A = 4, B = 2, and C = 4. Therefore,

X(z)
4
2
4
+
+
z1
(z 1)2
z

4z 1
z
z1
1
2
+ 2z 1
z
(z 1)2
+ 4z 1
z
z

1
2
1
< |z| < 1
2
Since the ROC is a ring, we find that the inverse transforms of the terms with poles at 1 lead to
left-sided sequences, while the inverse transform of the term with pole at 1/2 leads to a right-sided
sequence, namely,
z
z1
z
(z 1)2
z
z
1
2
u(n 1)
nu(n 1)
n
1
2
u(n)
and, consequently,
n1
x(n) = 4u(n) 2(n 1)u(n) + 4
1
2
u(n 1)
Example 10.4 (Complex roots)
Let us inverse transform

X(z) =
1
,
z2 + 1
with ROC =
|z| >
1
2
The denominator of X(z) has complex roots at j. Therefore, the partial fractions expansion of
X(z) will be of the form
A
A
X(z) =
+
z+j
(z j)
for some constant A and its complex conjugate, A , to be determined. These constants can be
obtained by comparing the coefficients of the numerators on both sides of the equality:
1
z2 + 1
=
=
A
A
+
z+j
(z j)
(A + A )z + j(A A)
z2 + 1
which leads to the equations

A + A = 0
and
j(A A) = 1
Solving for A and A we find that

A=
j
1
= ej 2
2
2
and
A =
j
1
= ej 2
2
2
264
Therefore,
CHAPTER 10
PARTIAL
FRACTIONS
X(z)
j/2
j/2
z+j
zj
Using (10.6) we arrive at

n1
x(n)
1
2
n1
1
2
cos
cos
(n 1) +
i
u(n 1)
2

n u(n 1)
10.4 INTEGRAL INVERSION FORMULA

There is a useful integral inversion formula that can be used to recover specific sample
values of a sequence x(n) for general ztransforms X(z), whether they are rational functions of z or not. The formula can be motivated as follows. Let X(z) denote a given
ztransform with the corresponding ROC. Consider the integral expression
I
1
z k dz
(10.8)
2j
H
where z is complex-valued and the notation means that the integration is carried over
any counter-clockwise contour around the origin and within the ROC of X(z). For our
purposes, it is sufficient to consider circular contours around the origin. So let z = rej
describe all points that lie on a circle of radius r with the phase varying between 0 and
2 see Fig. 10.1.
Im
Re
FIGURE 10.1 Points z lying on the circle of radius r in the complex plane.
Assume further that the value of r is such that all these points z lie inside the ROC of
X(z). Then
dz = jrej d
265
and, consequently, the contour integral (10.8) becomes
SECTION 10.4
1
2j
z k dz
1
2rk1
INTEGRAL
INVERSION
FORMULA
ej(1k) d
It is clear that the right-hand side evaluates to 1 when k = 1. On the other hand, when
k 6= 1, we get
1
2rk1
ej(1k) d
2
1
1
j(1k)
e

2rk1 j(k 1)
=0
0, k 6= 1
In other words,
1 H k
z dz =
2j
1, k = 1
0, k =
6 1
(10.9)
over circular contours around the origin. The result (10.9) is actually more general and is
not limited to circular contours; however, the argument is beyond the needs of our exposition. For the purposes of our discussion, it suffices to focus on circular contours.
Using the useful result (10.9), we can now proceed to develop an expression to recover
x(n) from X(z). We start from the definition of the ztransform
X(z) =
x(k)z k
k=
which converges absolutely for all points z in the ROC of X(z). Multiplying both sides of
the above equality by z n1 we get
X(z)z n1 = x(n)z 1 +
x(k)z nk1
k=,k6=n
Integrating over a circular contour around the origin and within the ROC of X(z) we get
1
2j
X(z)z n1dz
=
=

I
1
x(k)z nk1 dz
2j
k=,k6=n

I
I
X
x(k)
1
1
nk1
z dz +
z
dz
x(n)
2j
2j
1
2j
x(n)z 1 dz +
k=,k6=n
x(n) + 0
so that
x(n) =
1 H
X(z)z n1 dz
2j
(10.10)
We therefore arrive at a useful integral expression for recovering the sample x(n) by means
of a contour integral. Fortunately, evaluating the contour integral is a special case of a famous result in complex analysis known as the Cauchys Residue Theorem, which facilitates
determination of (10.10).
266
CHAPTER 10
PARTIAL
FRACTIONS
Cauchys Residue Theorem

Assume F (z) is a rational function of z, which is the case of interest for our treatment
in this book. Let {k } denote the poles of F (z), namely the points in the complex plane
where F (z) evaluates to . Let mk denote the multiplicity of pole k .
By definition, the residue of F (z) at any of its poles, say at z = k with multiplicity
mk , is computed as follows. Let
G(z) = F (z)(z k )mk
That is, we multiply F (z) by (z k )mk in order to obtain a function G(z) that does not
have a pole at k . Then

dmk 1
1

G(z)

m
1
(mk 1)! dz k
z=k
residue of F (z) at k =
(10.11)
in terms of the derivative of order (mk 1) of G(z). For example, if is a pole of order
1, then
residue of F (z) at = F (z)(z ) |z=
(pole of order 1)
On the other hand, if is a pole of order 2, then

d
2
residue of F (z) at =
F (z)(z )
dz
z=
(pole of order 2)
and so forth.
Cauchys residue theorem states that if we integrate the function F (z) over a counterclockwise contour, C, around the origin and within the ROC of F (z), then the result of the
integration is equal to the sum of the residues of F (z) at all poles that are inside C. More
explicitly,
P
1 H
F (z)dz =
(residues of F (z) at poles lying inside contour curve)
2j
(10.12)
Hence, the evaluation of the contour integral in (10.10) reduces to the evaluation of the
residues of the function X(z)z n1 in the region enclosed by the contour of integration and
we find that
x(n) =
(residues of X(z)z n1 at poles lying inside contour curve)
Example 10.5 (Evaluating residues)

z
over |z| > ||
z
We already know from Table 9.3 that the inverse transform is
X(z) =
x(n) = n u(n)
(10.13)
Let us use instead the result (10.13) to arrive at the expression for x(n). For any n 1, we start
from the function
zn
X(z)z n1 =
z
and note that it has a single pole at z = and possibly multiple poles at z = . Let us choose a
circular contour around the origin and within the ROC, namely, within |z| > ||. It is clear that this
contour encircles only the pole at z = . The residue at z = is easily seen to be

zn
zn
residue of
at =
(z )
= n
z
z
z=
Therefore, for all n 1, we get x(n) = n . For n = 0 we have

X(z)z n1 =
1
z
which has a single pole at z = and its residue is equal to

1
1
residue of
at =
(z )
= 1
z
z
z=
It follows that x(0) = 1, which can be written as 0 . Let us now examine the values of x(n) for
n < 0. Start with n = 1 so that
X(z)z n1 =
z 1
1
=
z
z(z )
and note that X(z)z n1 now has a single pole at z = and a single pole at z = 0; both poles are
encircled by a contour within the ROC. The residue at z = is easily seen to be

residue of

1
1
at =
(z )
= 1
z(z )
z(z )
z=
while the residue at z = 0 is

1
1
residue of
at 0 =
z)
= 1
z(z )
z(z )
z=0
Therefore, using (10.13) we conclude that

x(1) = 0
In a similar vein, we can find that x(n) = 0 for all n < 0. Consequently, putting the results together,
we arrive at
x(n) = n u(n)
as expected.
It is worth noting as well that relation (10.10) allows us to conclude that the following integral
result holds:
1 H zn
n =
(10.14)
dz
2j z
over any circular contour around the origin and within the region |z| > ||.
Therefore, as shown by the above example, a sequence x(n) can be recovered by evaluating the residues of X(z)z n1 within an appropriate circular contour region within the
ROC of X(z). This procedure may be useful in several instances, especially when we are
interested in the value of the sequence x(n) at a particular time instant n. In general, how-
267
SECTION 10.5
INTEGRAL
INVERSION
FORMULA
268
ever, inversion via (10.13) is not the most simple method to pursue.
CHAPTER 10
PARTIAL
FRACTIONS
10.5 APPLICATIONS
TO BE ADDED
Practice Questions:
1.
2.
10.6 PROBLEMS
Problem 10.1 Invert the transform
X(z) =
1
, ROC = {|z| > 1/5}
(z 1/8)(z + 1/5)

X(z) =
1
, ROC = {|z| < 1/8}
(z + 1/8)(z 1/4)

X(z) =
1
, ROC = {1/8 < |z| < 1/3}
(z + 1/8)2 (z + 1/3)

X(z) =
1
, ROC = {|z| > 1/2}
z 2 + 1/4

X(z) =
1
, ROC = {|z| < 1/3}
z 2 + 1/9

X(z) =
1
, ROC = {1/5 < |z| < 1/2}
(z 1/2)(z 2 + 1/25)
Problem 10.7 Given

X(z) =
1 2
z
3
Find x(n) when
+ z 1 + z
z 2 61 z 16
(a) ROC = {|z| > 21 }.

(b) ROC = {|z| < 13 }.
(c) ROC = { 13 < |z| < 12 }.
Problem 10.8 Given
X(z) =
z2
z 2
41 z
1
8
1
2
269
Find x(n) when
SECTION 10.6
(a) ROC = {|z| > 12 }.
PROBLEMS
(b) ROC = {|z| < 14 }.

(c) ROC = { 14 < |z| < 12 }.
Problem 10.9 Determine all possible sequences x(n) with ztransform
6z + 3 2z 1
6z 2 5z + 1
X(z) =

z 1
8z 2 z 1
X(z) =

X(z) =
12z 3 4z 2 + 12z 3
12z 2 7z + 1

1
X(z) =
12z 3 4z 2 + 12z 3
12z 2 7z + 1

1 2

X(z) =
z2
z 2
z+
1
2

1
z
X(z) =
1
3
z2
z 2
z +
1
2

1
X(z) =

1 3
3

1
X(z) =
Problem 10.17 Let
X(z) =
1
z
1
2

1 2
3
1
z
1
,
z2 + 1
1
2
|z| > 1
Decide whether x(n) is a causal sequence without inverting the transform.

Problem 10.18 Let
X(z) =
1
z+
1
3
1
,
z2 + 4
1
< |z| < 2
3
Decide whether x(n) is a causal sequence without inverting the transform.

Problem 10.19 Let
X(z) =
12z 3 4z 2 + 12z 3
, |z| > 1/2
12z 2 7z + 1
270
Use Cauchys residue theorem to evaluate x(99).
CHAPTER 10
PARTIAL
FRACTIONS
Problem 10.20 Let
12z 3 4z 2 + 12z 3
, |z| < 1/4
12z 2 7z + 1
Use Cauchys residue theorem to evaluate x(99).
X(z) =
Problem 10.21 Use the ztransform technique to evaluate the convolution

"
n1
1
2
u(n 2)
n+1 #
1+
1
3
u(n)
Problem 10.22 Use the ztransform technique to evaluate the convolution

"
n1
1
2
u(n + 3)
2n+1 #
1
3
u(n)
Problem 10.23 Consider the system

y(n) =
5
y(n 1) y(n 2) + x(n) ,
2
y(2) = 0, y(1) = 1
Find its complete response to the input sequence shown in Fig. 10.2 in two different ways:
(a) Using the ztransform technique.
(b) Without using the ztransform technique.
The input sequence is equal to one for all time instants n 0 except at n = 0, where it is zero,
and at n = 3, where it is equal to 1. The input sequence is also zero for n < 0. Simplify your
answers to parts (a) and (b) until they are identical. Plot the samples of the response sequence over
0 n 5.
x(n)
1
Problem 10.24 Consider the system

y(n)
1
1
y(n 1) y(n 2) = x(n 2) ,
4
8
y(2) = 1, y(1) = 0
Find its complete response to the same input sequence shown in Fig. 10.2 in two different ways:
(a) Using the ztransform technique.
(b) Without using the ztransform technique.
Simplify your answers to parts (a) and (b) until they are identical. Plot the samples of the response
sequence over 0 n 5.
Problem 10.25 Determine the sequence x(n) that is defined by
"
n2
x(n) =
1
2
u(n)
"
n+1
1
3
u(n 2)
n
(n + 1)
1
4
u(n + 1)
Problem 10.26 Determine the sequence x(n) that is defined by

"
2n
x(n) = n
1
2
"
n+1
1
3
u(n + 1)
"
u(2n)
3n
1
4
u(n 1)
y(n) =
5
1
y(n 1) y(n 2) + x(n 2),
6
6
y(2) = 0, y(1) = 1.
Determine its complete response to the sequence

n2
x(n) = (n 1)
1
4
u(n 1).
y(n) =
1
1
y(n 1) + y(n 2) + x(n 1),
4
8
y(2) = 1, y(1) = 0.
Determine its complete response to the sequence

2n2
1
3
x(n) = n
u(n 2).

y(n) y(n 1) +
1
y(n 2) = x(n),
4
Find its complete response to x(n) =

1 n
2
y(1) = 0,
y(2) = 4.
u(n 1).

y(n) + 2y(n 1) + 2y(n 2) = x(n),
Find its complete response to x(n) =

1 n
3
y(1) = 0,
y(2) = 1.
u(n 1).
Problem 10.31 Consider the ztransforms

X(z)
H(z)
1
1
,
z 1/2 z 1/3
0.5z 3
,
z 1/4
ROC = Rx
ROC = Rh
Let y(n) = x(n) h(n). Decide in each of the following cases whether y(n) is a causal sequence
by working with Y (z):
(a) Rx = {1/3 < |z| < 1/2} and Rh = {|z| > 1/4}.
(b) Rx = {|z| > 1/2} and Rh = {|z| > 1/4}.

(c) Rx = {|z| < 1/2} and Rh = {|z| < 1/4}.
Find y(0) in each case.
271
SECTION 10.6
PROBLEMS
272
Problem 10.32 Consider the ztransforms
CHAPTER 10
PARTIAL
FRACTIONS
X(z)
H(z)
1
1
,
(z 1/3)2 z 1/5
z 1
,
(z 1/6)2
ROC = Rx
ROC = Rh
Let y(n) = x(n) h(n). Decide in each of the following cases whether y(n) is a causal sequence
by working with Y (z):
(a) Rx = {1/5 < |z| < 1/3} and Rh = {|z| > 1/6}.
(b) Rx = {|z| > 1/3} and Rh = {|z| > 1/6}.

(c) Rx = {|z| < 1/3} and Rh = {|z| < 1/6}.
Find y(0) in each case.
CHAPTER
11
Transfer Functions
he ztransform is an important tool in the study of linear time-invariant (LTI) systems,

and also in the study of systems that may not be LTI by are still described by constantcoefficient linear difference equations. In this chapter, we focus on LTI systems and show
how the ztransform allows us to tackle several useful questions about such systems and
their behavior in a rather straightforward manner.
11.1 TRANSFER FUNCTIONS OF LTI SYSTEMS

To begin with, consider an LTI system with impulse response sequence h(n). Let H(z)
denote the ztransform of h(n):
H(z) =
h(n)z n
(11.1)
n=
over all values of z belonging to the corresponding ROC, namely, z Rh . We refer to

H(z) as the transfer function of the system. For example, the transfer function of an LTI
system with impulse response sequence
h(n) = n u(n)
is given by
H(z) =
z
,
z
|z| > ||
Thus, note that the impulse response sequence, h(n), and the transfer function, H(z), of
an LTI system determine each other uniquely.
The transfer function of an LTI system plays a critical role in characterizing the behavior
of the system, as the discussion in the current chapter will reveal. For most of our discussions, unless otherwise specified, we shall focus on LTI systems whose transfer functions
are rational functions in z or z 1 . As we are going to see, this is a rich class of systems
and it includes LTI systems that are described by constant-coefficient difference equations.
11.2 EIGENFUNCTIONS OF LTI SYSTEMS

Let us now select any point zo from within the ROC of the transfer function H(z), and
assume the LTI system is excited with the exponential input sequence
x(n) = zon ,
zo Rh
273

Copyright
274
CHAPTER 11
TRANSFER
FUNCTIONS
The value of zo may be real or complex. Since the system is LTI, its output sequence, y(n),
is obtained by convolving x(n) and h(n), i.e.,
y(n) =
=
k=
h(k)x(n k)
h(k)zonk
k=
"
zon
k=
n
zo H(zo )
h(k)zok
#
(11.2)
We therefore find that the output sequence y(n) is the same exponential sequence, zon , as
the input but scaled by the complex number H(zo ). The scaling factor is equal to the value
of the transfer function H(z) at the point z = zo see Fig. 11.1. For this reason, we
say that exponential sequences are eigenfunctions of LTI systems; an eigenfunction is a
sequence that is not modified by the system apart from some complex scaling. Here we see
that the exponential input sequence zon is regenerated at the output and suffers only scaling
by H(zo ).
zon
H(zo )zon
h(n)
FIGURE 11.1 When an LTI system is excited with the exponential sequence zon , the output is the
same exponential sequence but scaled by H(zo ).
Example 11.1 (Eigenfunction)

Consider the LTI system with transfer function
H(z) =
0.5z 2
,
z 2 1.5z + 0.5
|z| < 1/2
and assume it is excited by the exponential sequence

n
x(n) =
1
4
The point zo = 1/4 belongs to the ROC of H(z). Moreover,

H(1/4) = H(z)|z=1/4 = 1/6
275
so that the response of the system will be

y(n) =
n
SECTION 11.3
1
4
11.3 EVALUATION FROM DIFFERENCE EQUATIONS

The transfer function of an LTI system can be evaluated directly from knowledge of a
constant-coefficient difference equation describing the system, without the need to determine beforehand the corresponding impulse response sequence. The procedure is best
illustrated by means of an example.
Example 11.2 (Finding the transfer function)
y(n)
1
y(n 1) = x(n)
2
Since the system is relaxed, and since this is a constant-coefficient difference equation, we know that
the system is LTI. Let h(n) denote its impulse response sequence. In addition, because the system
is causal we know from the discussion in Sec. 5.2 that h(n) = 0 for n < 0 and, therefore, h(n) is a
right-sided sequence; its corresponding ROC has to be the outside of a disc.
Returning to the difference equation, we can evaluate the ztransforms of all sequences on both
sides of the equation, and use the linearity and time-shift properties of the ztransform, to obtain the
following algebraic equation:
1
Y (z) z 1 Y (z) = X(z)
2
Here, Y (z) denotes the ztransform of the sequence y(n) and X(z) denotes the ztransform of
the sequence x(n). Furthermore, z 1 Y (z) denotes the ztransform of the sequence y(n 1).
The sequences {x(n), y(n)} denote an arbitrary input-output pair satisfying the difference equation.
Let Rx and Ry denote the ROCs of X(z) and Y (z), respectively. The above algebraic equation
relating {Y (z), X(z)} will exist for all values z Rx Ry (i.e., for all values of z belonging to the
intersection of Rx and Ry ).
Thus, the key fact to note is that the original constant-coefficient difference equation has now
been transformed into a purely algebraic equation in the transform domain. The algebraic equation
can be solved to yield an expression for Y (z) in terms of X(z), namely,
Y (z)
1
=
X(z)
1 12 z 1
This ratio holds for any input-output pair {Y (z), X(z)}. For this reason, the ratio Y (z)/X(z), of
the output transform divided by the input transform, must be equal to the transfer function H(z)
of the LTI system. To see that this is indeed the case, assume x(n) = (n). Then, by definition,
y(n) = h(n). Hence, when X(z) = 1 we get Y (z) = H(z) and the ratio Y (z)/X(z) becomes
equal to H(z), namely,
1
z
H(z) =
=
1 1
1 2z
z 12
Now since h(n) is a right-sided sequence, it follows that the ROC of H(z) must be the outside of a
disc. Moreover, since the ROC of H(z) must exclude its pole located at z = 1/2, we conclude that
the ROC of H(z) should be given by |z| > 1/2. In summary, we arrive at
H(z) =
z
z
1
2
|z| > 1/2
EVALUATION
FROM
DIFFERENCE
EQUATIONS
276
CHAPTER 11
TRANSFER
FUNCTIONS
The above example illustrates one convenient method for determining the impulse response
sequence of an LTI system that is described by a constant-coefficient difference equation.
Specifically, we use the difference equation to determine the transfer function, H(z), and
then inverse transform H(z) to find h(n).
Example 11.3 (Finding the impulse response sequence)
Consider the same LTI system from the previous example, which is described by the relaxed equation
y(n)
1
y(n 1) = x(n)
2
We already determined its transfer function as

H(z) =
z
z
1
2
|z| > 1/2
The inverse transform is the impulse response sequence:

h(n) = (0.5)n u(n)
Example 11.4 (Causal LTI systems)
Recall from Sec. 5.2 that an LTI system is causal if, and only if, its impulse response sequence
satisfies h(n) = 0 for n < 0. It follows that h(n) is a right-sided sequence so that the expansion of
H(z) in terms of powers of z will have the form:
H(z) = h(0) + h(1)z 1 + h(2)z 2 + . . .
with only negative powers of z appearing in the expansion. Therefore, we conclude that an LTI
system is causal if, and only if, the ROC of its transfer function is the exterior of a disc, say, |z| > r
for some finite r; the point z = is included in the ROC since positive powers of z do not appear
in the expansion for H(z) due to the causality of h(n).
Example 11.5 (Stable LTI systems)

Recall from Sec. 5.3 that an LTI system is BIBO stable if, and only if, its impulse response sequence
is absolutely summable. That is, it must hold that
X
n=
|h(n)| <
In the transform domain, this condition is equivalent to requiring the ROC of H(z) to include the
unit circle,
LTI system is BIBO stable {|z| = 1} ROC
(11.3)
Indeed, for any point z such that |z| = 1, we have
X
n=
|h(n)z n | =
X
n=
|h(n)|
Now if |z| = 1 is a point in the ROC of H(z), then the term on the left-hand side converges absolutely
and, therefore, the impulse response sequence is absolutely summable.
EVALUATION
FROM
DIFFERENCE
EQUATIONS
For example, consider the LTI system with transfer function

H(z) =
1
,
(z 2)(z 0.5)
0.5 < |z| < 2
Since the ROC includes the unit circle, we conclude that the system is BIBO stable. Indeed, the
impulse response sequence of the system is
2
2
h(n) = (2)n1 u(n) (0.5)n1 u(n 1)
3
3
which is absolutely summable. Note that the system in question has two poles at z = 2 and z = 0.5;
it also has two zeros at z = .
Therefore, the conclusion about the BIBO stability of the system holds despite the fact that one of
the modes (or poles) lies outside the unit circle! This result does not contradict a statement we made
earlier in (7.32) requiring all modes of the impulse response sequence to lie inside the unit circle in
order for BIBO stability to hold. The statement given in (7.32) was specific to causal systems. In
our present example, we are dealing with a non-causal system (as evidenced from the fact that its
impulse response sequence is not zero for negative time or from the fact that the ROC of H(z) is a
ring).
Example 11.6 (Impulse response sequence)
We can also determine the impulse response sequence of an LTI system from knowledge of any inputoutput pair response (since any such pair of sequences determines the transfer function). Indeed,
given an input sequence, x(n), and the corresponding output sequence, y(n), we use their respective
ztransforms to determine the transfer function, H(z):
H(z) =
Y (z)
X(z)
and then inverse-transform the result using the proper ROC. For example, assume that we know that
the step response of a stable causal LTI system is (0.5)n u(n). Let us find its impulse response
sequence.
We thus have x(n) = u(n) and y(n) = (0.5)n u(n). Therefore,
Y (z) =
z
,
z 0.5
and
X(z) =
This leads to
z
,
z1
277
SECTION 11.4
|z| > 0.5

|z| > 1
z1
z
1
=
z 0.5
z 0.5
z 0.5
What about the ROC of H(z)? This can be determined from the statement that the system is both
causal and stable. Since H(z) has a pole at z = 0.5, we find that the ROC can be either |z| > 0.5 or
|z| < 0.5. However, the assumed stability of the system implies that the region of convergence must
include the unit circle. Moreover, the assumed causality of the system implies that the ROC must be
the exterior of a disc. Therefore, either condition, allows us to conclude that the ROC of H(z) must
be given by |z| > 0.5. Inverse-transforming H(z) then leads to
H(z) =
h(n) = 0.5n u(n) 0.5n1 u(n 1)
278
11.4 FINDING OUTPUT SEQUENCES
CHAPTER 11
TRANSFER
FUNCTIONS
We can use the transfer function of an LTI system to determine its response to arbitrary
input sequences. Thus, let x(n) denote the input sequence to an LTI system with impulse
response sequence h(n). We already know from Sec. 5.1 that the response sequence, say
y(n), can be determined via the convolution sum
y(n) = x(n) h(n) =
k=
x(k)h(n k)
If we, however, denote the z transforms of {x(n), h(n)} by {X(z), H(z)} and the corresponding ROCs by {Rx , Rh }, then from the convolution property (9.26) we know that
the ztransform of y(n) is given by
Y (z) = X(z)H(z)
(11.4)
The ROC of Y (z) consists of Rx Ry plus possibly the points z = 0 or z = . This result
suggests that the response of the LTI system can be determined via inverse-transformation
of the product X(z)H(z).
Example 11.7 (Evaluating the response sequence)
Consider again the same causal and relaxed system from Example 11.3 and let us determine its
response to the input sequence x(n) = u(n). We already know that the transfer function of the
system is given by
1
z
,
|z| >
H(z) =
2
z 12
On the other hand, the ztransform of the input sequence is
z
,
z1
X(z) =
|z| > 1
It follows that the ztransform of the output sequence is

Y (z) = X(z)H(z) =
z2
,
(z 0.5)(z 1)
|z| > 1
We can inverse-transform Y (z) by using the partial fractions method. Indeed, if we expand the
strictly proper function Y (z)/z 2 into partial fractions we obtain
Y (z)
2
2
=
z2
z1
z 0.5

so that
Y (z) = 2z
Consequently, we arrive at
z
z1
2z
z
z 0.5
y(n) = 2[1 (0.5)n+1 ]u(n + 1)
|z| > 1
11.5 FINDING DIFFERENCE EQUATIONS

Relation (11.4) also suggests a method for determining a description for an LTI system in
terms of a constant-coefficient difference equation from knowledge of its impulse response
sequence or, equivalently, its rational transfer function.
279
SECTION 11.5
FINDING
DIFFERENCE
EQUATIONS
Example 11.8 (Determining a difference equation)

H(z) =
z
,
z 1/2
|z| > 1/2
and let us determine an input-output description for the system in terms of a constant-coefficient
difference equation. Multiplying the numerator and denominator of H(z) by z 1 we obtain
H(z) =
1
,
1 12 z 1
|z| > 1/2
It is usually more convenient (though not necessary) to work with negative powers of z when determining difference equations. This observation explains why our first step involved multiplying the
numerator and denominator of H(z) by z 1 . Now, the fact that the ROC is |z| > 1/2 indicates
that the impulse response sequence, h(n), is a right-sided sequence. Then, for any input-output pair
{x(n), y(n)}, we know from (11.4) that the following relation must hold
Y (z)
z 1
=
X(z)
1 12 z 1
Cross-multiplying we get
Y (z) 1
1 1
= X(z)
z
2
and using the properties of the ztransform we arrive at the difference equation
y(n)
1
y(n 1) = x(n)
2
One question that arises is how the argument would be different had we started from the same transfer
function but with a different ROC, say,
H(z) =
z
,
z 1/2
|z| < 1/2
In this case, the impulse response sequence will need to be a left-sided sequence. However, if we
repeat the previous argument we arrive at the same difference equation for the system. So how do
we capture the fact that in the first case the difference equation should lead to a ROC that is equal to
|z| > 1/2, while in the second case the same difference equation should lead to a ROC that is equal
to |z| < 1/2?
The answer lies in the fact that in one case the impulse response sequence is right-sided while in
the second case it is left sided. Thus returning to the difference equation that we arrived at:
y(n)
1
y(n 1) = x(n)
2
we see that this relaxed equation describes a causal system if it is expressed in the following form
y(n) =
1
y(n 1) + x(n)
2
with time running forward. The causality of the system implies h(n) = 0 for n < 0 and, therefore,
h(n) would be right-sided, as desired. Indeed, since the system is relaxed and using x(n) = (n),
we find by iteration that
n
1
u(n)
h(n) =
2
280
CHAPTER 11
TRANSFER
FUNCTIONS
On the other hand, the same relaxed equation would describe a non-causal system if it is expressed
instead in the alternative form
y(n 1) = 2y(n) x(n)
with time running backwards. In this case, the resulting impulse response sequence will be left-sided
(and zero for n 0). Indeed, since the system is relaxed and using again x(n) = (n), we find by
iteration that
n
1
h(n) =
u(n 1)
2
Note that, in this case, we have h(n) = 0 for n 0 and we say that the system is strictly anti-causal.
In summary, we conclude that
H(z)
H(z)
z
,
z 1/2
z
,
z 1/2
1
y(n 1) + x(n)
2
|z| > 1/2
y(n) =
|z| < 1/2
y(n 1) = 2y(n) x(n)
Example 11.9 (Determining another difference equation)
Consider now an LTI system with transfer function

H(z) =
z+1
z 2 + 2z 3
difference equation. We are leaving the ROC of H(z) unspecified for now.
Multiplying the numerator and denominator of H(z) by z 2 we obtain
H(z) =
z 1 + z 2
1 + 2z 1 3z 2
Then, for any input-output pair {x(n), y(n)}, we know from (11.4) that the following relation must
hold
z 1 + z 2
Y (z)
=
X(z)
1 + 2z 1 3z 2
Y (z)[1 + 2z 1 3z 2 ] = X(z)[z 1 + z 2 ]
and using the properties of the ztransform we arrive at the following difference equation
y(n) + 2y(n 1) 3y(n 2) = x(n 1) + x(n 2)
As was explained in the previous example, we can now determine in which direction this difference
equation should run in accordance with the ROC of H(z). Thus, note that H(z) has two poles at
z = 1 and z = 3. Therefore, there are 3 possibilities of valid ROCs:
1. ROC = {|z| > 3}: In this case, the impulse response sequence needs to be right-sided and,
consequently, the difference equation should run forwards in time and represent a relaxed
causal system:
y(n) = 2y(n 1) + 3y(n 2) + x(n 1) + x(n 2)
In general, this will be the case of most interest for our studies: we start from a transfer
function description for a system and we arrive at a difference equation description that runs
forwards in time.
2. ROC = {|z| < 1}: In this case, the impulse response sequence needs to be left-sided and,
consequently, the difference equation should run backwards in time and represents a relaxed
281
anti-causal system:
y(n 2) =
SECTION 11.6
2
1
1
1
y(n) + y(n 1) x(n 1) x(n 2)
3
3
3
3
Poles, Zeros, and Modes
3. ROC = {1 < |z| < 3}: In this case, the impulse response sequence needs to be two-sided.
The question is how to express the difference equation in this case and how to determine a
suitable input-output representation for the system. To do so, we first use the partial fractions
expansion of H(z) to write
H(z) =
0.5
0.5
0.5z 1
0.5z 1
+
=
+
1
z+3
z1
1 + 3z
1 z 1
so that
Y (z) =
0.5z 1
0.5z 1
X(z)
X(z)
+
1 + 3z 1
1 z 1
{z
=Y1 (z)
{z
=Y2 (z)
In other words, the partial fraction expansion allows us to identify Y (z) as the result of combining the outputs {Y1 (z), Y2 (z)} of the two subsystems:
0.5z 1
1 + 3z 1
and
0.5z 1
1 z 1
respectively, namely,
Y (z) = Y1 (z) + Y2 (z)
Now, in view of the ROC in this case, we find that for the subsystem giving Y1 (z), its difference equation should run backwards in time. This is because the transfer function of this
subsystem is
0.5z 1
H1 (z) =
, |z| < 3
1 + 3z 1
and its ROC is of the form |z| < 3; this is the ROC for H1 (z) that would be consistent with
the overall ROC described by 1 < |z| < 3. Likewise, for the second subsystem giving Y2 (z),
its difference equation should run forwards in time. The transfer function for this subsystem
is
0.5z 1
H1 (z) =
, |z| > 1
1 z 1
In view of these remarks, starting from
Y1 (z) =
we write
y1 (n 1) =
0.5z 1
X(z)
1 + 3z 1
1
1
y1 (n) x(n 1)
3
6
(time running backwards)
and starting from

Y2 (z) =
we write
0.5z 1
X(z)
1 z 1
1
x(n 1) (time running forwards)
2
Combining y1 (n) and y2 (n) at any particular time instant n, we obtain the desired value for
y(n):
y(n) = y1 (n) + y2 (n)
y2 (n) = y2 (n 1) +
282
11.6 POLES, ZEROS, AND MODES
CHAPTER 11
TRANSFER
FUNCTIONS
The poles of a transfer function H(z) are the points in the extended complex plane where
H(z) = . In general, the poles are a subset of the roots of the denominator of H(z) since
cancellations can occur between the roots of the numerator and the denominator. Likewise,
the zeros of a transfer function H(z) are the points in the extended complex plane where
H(z) = 0. Again, the zeros are generally a subset of the roots of the numerator due to
the possibility of cancellations. Nevertheless, it always holds that the number of poles and
zeros of a transfer function, including those located at , should be equal:
number of poles = number of zero
(11.5)
Example 11.10 (Characteristic polynomial; poles and modes)

H(z) =
z+1
,
z 2 + 2z 3
|z| > 3
where the denominator is expressed in terms of positive powers of z. Note that the numerator polynomial is zero at z = 1 while the denominator polynomial is zero at z = 1 and z = 3. Therefore,
the numerator and denominator polynomials of H(z) do not share any common root and are said to
be coprime.
We already know from Example 11.9 that the system is described by the difference equation
y(n) = 2y(n 1) + 3y(n 2) + x(n 1) + x(n 2)
The corresponding homogeneous equation is given by
y(n) + 2y(n 1) 3y(n 2) = 0
and its characteristic polynomial is
p() = 2 + 2 3
The polynomial p() is also said to be the characteristic polynomial of the original systems complete
difference equation (and not only of its homogeneous part).
Now, observe that the polynomial that appears in the denominator of H(z) coincides with the
characteristic polynomial, p(), of the difference equation. Recall that the roots of the characteristic
polynomial are called the modes of the system, while the roots of the denominator of H(z) are
called the poles of the system. We therefore conclude that if no cancellations occur between the
numerator and the denominator polynomials of H(z) (i.e., if these polynomials are coprime), then the
denominator of H(z) (when written in terms of positive powers of z) coincides with the characteristic
polynomial of the difference equation. We also conclude that the modes of an LTI system can be
determined by evaluating the roots of the denominator of H(z).
The same conclusion would hold if the ROC of H(z) were instead |z| < 1 (i.e., for the case of
non-causal systems).
Example 11.11 (Cancellations)

Consider transfer function
z1
z 2 3z + 2
The denominator has two roots at z = 1 and z = 2; the former pole is canceled by the root of the
numerator at z = 1. Therefore, H(z) has a single pole at z = 2. The transfer function does not have
H(z) =
z zero at z = 1 due to the same cancellation, but it has a zero at z = . We thus note that H(z) has
the same number of zeros and poles (one of each in this case).
11.7 REALIZABLE LTI SYSTEMS

In practice we are generally interested in LTI systems whose transfer functions are realizable. This means that the systems need to be both BIBO stable and causal. The stability
property ensures that the system output remains bounded for bounded inputs. The causality
property ensures that the system output does not depend on future input samples.
Now recall that the stability of an LTI system requires the ROC of its transfer function,
H(z), to include the unit circle, |z| = 1. Likewise, the causality of an LTI system requires
its impulse response sequence, h(n), to be be causal:
h(n) = 0,
n<0
In other words, h(n) must be a right-sided sequence and, accordingly, the ROC of H(z)
must be the outside of a disc (including the point at z = ). Combining these conditions
allows us to conclude that an LTI system is realizable if, and only if, its ROC is of the form
ROC = {|z| > } for some 0 < 1 Realizable
(11.6)
Now since the ROC of a transfer function must exclude all its poles, we conclude that the
poles of any realizable H(z) must all lie inside the unit circle.
Example 11.12 (Realizable systems)

The system
z
, |z| > 0.5
z 0.5
is realizable since its ROC is of the required form (11.6). Note that H(z) has a singe pole at z = 1/2
and this pole lies inside the unit circle, as expected. On the other hand, the system
H(z) =
H(z) =
z
,
(z 0.5)(z 2)
|z| > 2
is not realizable since its ROC does not have the required form (11.6) for some 0 < 1.
What about the system
z
, |z| < 0.5?
H(z) =
z 0.5
Obviously, this system is not realizable since its ROC does not have the form (11.6). Note, however,
that the system has a pole at z = 1/2, which lies inside the unit circle. Therefore, having poles
inside the unit circle is not sufficient for a system to be realizable; it is only a necessary condition:
LTI system is realizable
Poles lie inside unit circle
poles lie inside unit circle

realizable LTI system
283
SECTION 11.8
REALIZABLE
LTI
SYSTEMS
284
11.8 SYSTEM INVERSION
CHAPTER 11
TRANSFER
FUNCTIONS
When an input sequence, x(n), is fed into an LTI system with transfer function H(z), the
result is an output sequence y(n) that is related to x(n) through the convolution sum
y(n) =
n=
x(k)h(n k)
in terms of the impulse response sequence, h(n), of the system. We say that the input
sequence is modified (or distorted) by the system and transformed from x(n) into y(n). In
the transform domain, the ztransforms of x(n) and y(n) are related via
Y (z) = H(z)X(z)
(11.7)
In several applications, we are interested in undoing the effect of the system on the input
sequence. This task can be accomplished by cascading a system G(z) in series with H(z)
in order to recover x(n), as illustrated in Fig. 11.2.
x(n)
H(z)
FIGURE 11.2
y(n)
x(n)
G(z)
The series cascade of an LTI system H(z) with its inverse system G(z).
The system G(z) acts on the sequence y(n) in order to recover the sequence x(n).
When this is possible, we say that G(z) is the inverse of H(z). In order to accomplish this
task, it must hold that
X(z) = G(z)Y (z)
Combining with (11.7), we find that the transfer functions H(z) and G(z) must satisfy the
relation
H(z)G(z) = 1
Obviously, for this equality to hold, the ztransforms H(z) and G(z) must have overlapping ROCs, i.e., there must exist some common region in the complex plane over which
both H(z) and G(z) are well defined. If this is the case, then an inverse system exists and
it is given by
1
G(z) =
(11.8)
H(z)
In particular, note that the zeros of H(z) become poles of G(z) and the poles of H(z)
become zeros of G(z).
One difficulty that arises in inverting LTI systems is that we are often interested in
realizable inverse systems, G(z). This condition requires the ROC of G(z) to be of the
form
ROC = {|z| > } for some 0 < 1
(11.9)
Since realizable inverses G(z) must have their poles inside the unit circle, we conclude
that a necessary condition for such inverses to exist is to require the zeros of H(z) to lie
inside the unit circle.
285
SECTION 11.8
Example 11.13 (Inversion of a stable and causal LTI system)
SYSTEM
INVERSION
Consider the system

z 0.25
, |z| > 0.5
z 0.5
The transfer function of the inverse system is given by
H(z) =
G(z) =
z 0.5
z 0.25
Now note that the ROC of the transfer function G(z) can be either |z| > 0.25 or |z| < 0.25. Since
H(z) and G(z) must have overlapping ROCs, we conclude that the inverse system of H(z) is
G(z) =
z 0.5
,
z 0.25
|z| > 0.25
Observe that this inverse is a realizable system.
Example 11.14 (Inversion of an unstable and noncausal LTI system)

Consider now the system
z 0.25
, |z| < 0.5
z 0.5
where we only changed the direction of the ROC relative to the previous example. This system is
unstable since its ROC does not include the unit circle; the system is also noncausal since its ROC is
the inside of a disc and, therefore, its impulse response sequence is left-sided. The transfer function
of the inverse system is given by
z 0.5
G(z) =
z 0.25
The ROC of the transfer function G(z) can be either |z| > 0.25 or |z| < 0.25. Both possibilities
lead to an ROC for G(z) that overlaps with the ROC of H(z). We therefore have two valid inverse
systems in this case:
z 0.5
, |z| > 0.25
G1 (z) =
z 0.25
or
z 0.5
G2 (z) =
, |z| < 0.25
z 0.25
However, only G1 (z) is a realizable inverse system.
H(z) =
Example 11.15 (Inversion of a noncausal but stable LTI system)

Consider now the system
z3
, |z| < 2
z2
This system is stable since its ROC includes the unit circle; the system is nevertheless noncausal
since its ROC is the inside of a disc and, therefore, its impulse response sequence is left-sided. The
transfer function of the inverse system is given by
H(z) =
G(z) =
z2
z3
286
CHAPTER 11
The ROC of the transfer function G(z) can be either |z| > 3 or |z| < 3. Since H(z) and G(z) must
have overlapping ROCs, we conclude that the inverse system of H(z) is
TRANSFER
FUNCTIONS
G(z) =
z2
,
z3
|z| < 3
The inverse system is stable since its ROC includes the unit circle; it is however noncausal.
11.9 APPLICATIONS
TO BE ADDED
Practice Questions:
1.
2.
11.10 PROBLEMS
Problem 11.1 Find the transfer functions, and the corresponding ROCs, of the LTI systems described by the following impulse response sequences:
(a) h(n) =

1 n1
2
(b) h(n) =

1 n1
(c) h(n) = n
(d) h(n) =
u(n 3).
u(n).

1 2n1
2

1 n1
2
u(n 2).
n
3
cos
u(n).
Problem 11.2 Find the transfer functions, and the corresponding ROCs, of the LTI systems described by the following impulse response sequences:
(a) h(n) =

1 2n+1
3
(b) h(n) =

1 n
3
(c) h(n) = (n 1)
(d) h(n) =

1 n1
3
u(n).
u(n + 2).

1 n3
3
sin
u(n 1).
n
3
2
3
u(n).
Problem 11.3 Find the impulse response sequences of the LTI systems with the following transfer
functions:
z2
(a) H(z) =
, |z| > 1/2.
(z 1/2)(z + 1/3)
1
, |z| < 1/2.
(b) H(z) = 2
(z + 1/4
(c) H(z) =
(d) H(z) =
(e) H(z) =
z + 1/3
,
(z 1/2)(z + 1/4)
z + 1/3
,
(z 1/2)(z + 1/4)
z + 1/3
,
(z 1/2)(z + 1/4)
|z| < 1/4.

1/4 < |z| < 1/2.
|z| > 1/2.
(f) H(z) =
z + 1/3
,
(z 2)(z + 4)
|z| < 2.
PROBLEMS
Problem 11.4 Find the impulse response sequences of the LTI systems with the following transfer
functions:
z 1
(a) H(z) =
, |z| > 1/4.
(z + 1/4)(z + 1/6)
(b) H(z) =
1 + z 1
,
(z 2 + 1/9
(c) H(z) =
z + 1/6
,
(z + 1/2)(z 1/8)
(d) H(z) =
(e) H(z) =
(f) H(z) =
|z| > 1/3.
z + 1/6
,
(z + 1/2)(z 1/8)
z + 1/6
,
(z + 1/2)(z 1/8)
z + 1/6
,
(z + 2)(z 8)
287
SECTION 11.10
|z| < 1/8.

1/8 < |z| < 1/2.
|z| > 1/8.
|z| > 8.
Problem 11.5 Find difference equations for the LTI systems described by the impulse response
sequences of Prob. 11.1.
Problem 11.6 Find difference equations for the LTI systems described by the impulse response
Problem 11.7 Find difference equations for the LTI systems described by the transfer functions of
Prob. ??.
Problem 11.8 Find difference equations for the LTI systems described by the transfer functions of
Prob. ??.
Problem 11.9 Find the poles, modes, and zeros of the LTI systems described by the impulse response sequences of Prob. 11.1.
Problem 11.10 Find the poles, modes, and zeros of the LTI systems described by the impulse
response sequences of Prob. 11.2.
Problem 11.11 Find the poles, modes, and zeros of the LTI systems described by the transfer
functions of Prob. 11.47.
Problem 11.12 Find the poles, modes, and zeros of the LTI systems described by the transfer
functions of Prob. 11.48.
Problem 11.13 For each of the LTI systems described by the impulse response sequences of
Prob. 11.1, determine whether it is stable? causal? realizable?
Problem 11.14 For each of the LTI systems described by the impulse response sequences of
Prob. 11.2, determine whether it is stable? causal? realizable?
Problem 11.15 For each of the LTI systems described by the transfer functions of Prob. 11.47,
determine whether it is stable? causal? realizable?
Problem 11.16 For each of the LTI systems described by the transfer functions of Prob. 11.48,
determine whether it is stable? causal? realizable?
Problem 11.17 Find the inverse of each of the LTI systems described by the impulse response
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FUNCTIONS
Problem 11.18 Find the inverse of each of the LTI systems described by the impulse response
Problem 11.19 Find the inverse of each of the LTI systems described by the transfer functions of
Prob. 11.47.
Problem 11.20 Find the inverse of each of the LTI systems described by the transfer functions of
Prob. 11.48.
Problem 11.21 Find the responses of the LTI systems described by the impulse response sequences
of Prob. 11.1 to the following input sequences
(a) x(n) =
(b) x(n) =

1 n2
u(n).
2

1 n
n 2 u(n
1).
Problem 11.22 Find the responses of the LTI systems described by the impulse response sequences
of Prob. 11.2 to the input sequences given in Prob. 11.21.
Problem 11.23 Find the responses of the LTI systems described by the transfer functions of Prob. 11.47
to the input sequences given in Prob. 11.21.
Problem 11.24 Find the responses of the LTI systems described by the transfer functions of Prob. 11.48
to the input sequences given in Prob. 11.21.
Problem 11.25 Find the transfer function of the moving average system
y(n) =
1
[x(n) + x(n 2)]
2
Is the system linear? causal? time-invariant? stable?

Problem 11.26 Find the transfer function of the moving average system with exponential weighting
M
1 X k
x(n k)
y(n) =
M +1
k=0
where || < 1. Is the system linear? causal? time-invariant? stable?

Problem 11.27 The transfer function of an LTI system is given by
H(z) =
z+3
(z 1/2)(z 2)(z 3)
Which of the following statements is correct?

(a) H(z) can be the transfer function of four different systems: one causal and BIBO stable, one
causal but not stable, one stable but not causal, and one neither causal nor stable.
(b) H(z) can be the transfer function of four different systems. Three of these systems are not
causal and three of them are not stable.
(c) H(z) can be the transfer function of four different systems. Since all the poles are positive
and real, all four systems are causal and BIBO stable.
(d) Since the system is LTI, H(z) uniquely determines the system.
Problem 11.28 Which statement is correct?

(a) If an LTI system is causal and BIBO stable, then all its poles must be inside the unit circle.
(b) If all the poles of an LTI system are inside the unit circle, then the system must be causal and
BIBO stable.
289
(c) Both (a) and (b) are correct.
SECTION 11.10
(d) Neither (a) nor (b) are correct.
PROBLEMS
Problem 11.29 Find the transfer function, and a difference equation description, for the LTI systems with the input-output pair:
n
1
2
x(n) =
n1
u(n),
y(n) =
1
3
u(n 2)
and with the properties below:

(a) System is stable and causal.
(b) System is stable and non-causal.
(c) System is unstable. Is it causal?
Problem 11.30 Find the transfer function, and a difference equation description, for the LTI systems with the input-output pair:
2n1
x(n) = n
1
2
n+1
u(n 1),
y(n) =
1
4
u(n 3)
and with the properties below:

(a) System is stable and causal.
(b) System is stable and non-causal.
(c) System is unstable. Is it causal?
Problem 11.31 True or False? An LTI system is realizable if, and only if, all its poles lie inside
the unit circle.
Problem 11.32 True or False? An LTI system has a realizable inverse if, and only if, all its zeros
lie inside the unit circle.
Problem 11.33 An input-output response pair of a relaxed causal and stable LTI system is given
by
n
n1
1
1
x(n) =
u(n), y(n)n
u(n 1)
2
2
(a) Determine the transfer function of the system and indicate its ROC.
(b) Determine the poles and zeros of the system.
(c) Determine a difference equation relating any input sequence x(n) to the corresponding output
sequence y(n).
(d) Draw a block diagram realization for the system using a minimum number of delay elements.
(e) If the system were not initially relaxed, but with initial conditions
n y(1) = 1 and y(k) = 0
for k < 1, what would have been its response to x(n) = 12 u(n)?
Problem 11.34 The transfer function of a stable and causal LTI system is
H(z) =
z
z
1
3
4z 1
1
16 z 3 (z 12 )2
(a) What is the region of convergence of H(z)? Justify your answer.

(b) Is the response to x(n) = 21n u(n) equal to y(n) = 12
without explicitly computing the response of the system.
n
H( 21 )u(n)? Justify your answer
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CHAPTER 11
TRANSFER
FUNCTIONS
(c) What are the poles of H(z)? Are all the zeros of H(z) finite? How many are there?
(d) Determine the response of the system to the input sequence x(n) =
1
2
(n)
1
4
(2n 4).
(e) Write down a constant-coefficient linear difference equation that relates the input and output
sequences of the above system.
Problem 11.35 Refer Fig. 11.3. Let S denote an LTI system with the input-output pair
n
x(n) =
1
2
n2
u(n 1),
u(n 3)
y(n)
x(n)
y(n)
1
3
y(n) =
x(n)
FIGURE 11.3 System configurations used for Prob. 11.35.
(a) If we apply y(n) to the input of S, what would the response of S be?
(b) If x(n) were observed at the output of S, which input sequence would have generated it?
Problem 11.36 Refer to Fig. 11.4. It shows a system with transfer function
H1 (z) =
1
z+
3
2
|z| < 3/2
cascaded with an unknown LTI system. The output of H1 (z) is multiplied by the sequence (1)n
and fed into the unknown system.
(1)n
1 n
2
u(n)
+
FIGURE 11.4
H1 (z)
LTI ?
1 n2
2
u(n 1)
A cascade of two systems with feedback interconnection for Prob. 11.36.
(a) Is the overall system LTI?

(b) Find the transfer function of the unknown LTI system.
Problem 11.37 An input-output response pair of a relaxed, causal, and stable first-order LTI system is given by
n2
n
1
1
u(n 1), y(n) = n
u(n 2)
x(n) =
4
4
Now assume that the system is not initially relaxed, but has initial condition y(1) = 2. Find its
complete response to the input sequence
x(n) = 0.5n u(n 2) + (n 5)
Plot the samples of y(n) for n = 6, 7, 8.
Problem 11.38 Find the inverse of the LTI system whose impulse response sequence is given by
h(n) = 0.5n u(n) 2(0.5)n1 u(n 1)
Problem 11.39 Consider a rational transfer function H(z) with a real-valued impulse response
sequence h(n). Show that the poles and zeros of H(z) occur in conjugate pairs. That is, if z = po is
a pole then so is z = po . Likewise, if z = zo is a zero then so is z = zo .
Problem 11.40 If H(z) is a rational transfer function that is causal and stable, what can you say
about H(z 1 ) and H(z)?
Problem 11.41 Consider a signal of the form x(n) = { 0 , 1, 1}. When the signal x(n) is
applied to a causal LTI system, the observed output is y(n). The odd part of y(n) is known to be
yo (n) = (1/4)n1 u(n 2) + (1/3)n u(n 1)
for n 0.
(a) Find the impulse response sequence of the system.

(b) Find the transfer function of the system.
(c) Find the energy of the impulse response sequence.
(d) Find the power of the impulse response sequence.
Problem 11.42 The even part of the impulse response sequence of a causal LTI system is given by
n
he (n) =
1
2
n2
1
4
u(n 1)
u(n)
(a) Find the transfer function of the system.

(b) Find the unit-step response of the system.
(c) Find a constant-coefficient difference equation describing the system.
(d) Draw a block diagram representation for the system.
Problem 11.43 Consider four LTI systems {S1 , S2 , S3 , S4 }. The following information is available:
(i) System S1 is described by the constant-coefficient difference equation
y(n)
1
1
y(n 1) y(n 2) = x(n)
6
6
with initial conditions y(2) = 0 and y(1) = 6.

(ii) System S2 is causal and BIBO stable with transfer function
H2 (z) =
1 z 1

z+
z 2 z + 14
1
4
(iii) System S3 has impulse response sequence
n1
h3 (n) =
1
2
u(n 2)
291
SECTION 11.10
PROBLEMS
292
CHAPTER 11
(iv) The output of the stable and causal system S4 in response to the input sequence u(n 1) is
"
TRANSFER
FUNCTIONS
y4 (n) =
1
4
3
n3 #
1
4
u(n 2)
The four systems are interconnected as shown in the block diagram of Fig. 11.5. All systems in the
figure are relaxed except for S1 . Systems S1 and S4 are connected in series. Additionally, a switch
and a nonlinear system (NL) are also shown in the figure. The input-output characteristics of the
nonlinear device (NL) is the following:
(
s(n) =
p(n),
0,
if p(n) > 2
otherwise
The switch is initially open and the signal y(n) is therefore disconnected from the lower part of
the circuit. At a large enough time instant No , the switch is closed and the signal y(n) then drives
the lower circuit. The value of No is such that the output of S1 can be assumed to have reached
steady-state. The input x(n) to the circuit is taken to be the step sequence, x(n) = u(n).
(a) Determine the sequence y(n) and its steady-state value.
(b) Determine the sequences q(n) and d(n).
(c) Determine the sequences p(n) and s(n).
(d) Determine the sequence v(n).
(e) Determine the sequence r(n).
(f) Determine the sequence w(n).
(g) How would your answers to the previous questions change if the initial conditions of S1 were
modified to y(2) = 6 and y(1) = 0?
x(n)
y(n)
S1
S4
r(n)
switch closed
at time No
q(n)
z 1
p(n)
+
NL
d(n)
s(n)
S2
S3
FIGURE 11.5 Block diagram for Prob. 11.43.
w(n)
v(n)
Problem 11.44 Consider the block diagram shown in Fig. 11.6. The input-output relation of system S1 is described by the difference equation
1
1
1
y1 (n) = y1 (n 1) + y1 (n 2) + x(n) x(n 1)
4
8
3
with initial conditions y1 (1) = and y1 (2) = . System S2 is LTI and its step-response is
given by the sequence

n
1
u(n)
hstep (n) = 2
2
The output of S2 is passed through a system that squares every sample and the resulting sequence is
subsequently multiplied by (1)n . The output of the overall block diagram is
y(n) = y1 (n) + y2 (n)
Find the values of the initial conditions and such that y(n) = 0 for all n 0 when x(n) = (n).
y1 (n)
S1
y(n)
x(n)
S2
()2
y2 (n)
(1)n
Problem 11.45 Consider the second-order LTI system that is described by the difference equation
y(n) + ay(n 2) = bx(n)
for some real scalar coefficient a. Find conditions on {a, b} for the system to be realizable.
Problem 11.46 Consider a second-order LTI system that is described by the difference equation
y(n) + a1 y(n 1) + a2 y(n 2) = bx(n)
for some real scalar coefficients {a1 , a2 , b}. Find conditions on {a1 , a2 , b} for the system to be
realizable. Plot the region in the plane a1 a2 that corresponds to realizable filters.
Problem 11.47 Consider the two cascades shown in Fig. 11.7 where the order of the downsampler
and upsampler blocks is switched in one case relative to the other. Find the transfer function of each
cascade and compare them.
Problem 11.48 Consider the system shown in Fig. 11.8 with input x(n) and output y(n). Find its
transfer function.
293
SECTION 11.10
PROBLEMS
294
CHAPTER 11
TRANSFER
FUNCTIONS
x(n)
x(n)
FIGURE 11.7
y(n)
w(n)
Two separate cascades of upsampling and downsampling blocks for Prob. 11.47.
x(n)
y(n)
z 1
FIGURE 11.8 Block diagram representation of the system for Prob. 11.48.
CHAPTER
12
In Chapters 7 and 8 we developed techniques for determining the complete solution of
constant-coefficient difference equations over the interval n 0. It turns out that the
ztransform provides a powerful (and often more general and more convenient) way for
solving these same difference equations. The purpose of this chapter is to illustrate how
to use the ztransform for such purposes, and to motivate the introduction of the unilateral ztransform, which is particularly suited for solving difference equations with initial
conditions.
12.1 Z-TRANSFORM AND DIFFERENCE EQUATIONS

We first explain how the ztransform can be used to determine the complete solution
of difference equations, and use the discussion to motivate the introduction of the more
convenient unilateral ztransform. This is best illustrated by means of an example.
Example 12.1 (Using the ztransform)

Consider the constant-coefficient difference equation
y(n) =
1
y(n 1) + x(n),
2
y(1) = 2
We would like to determine its complete solution over n 0 in response to the input sequence
x(n) = u(n). We already know from the discussion in Sec. 8.6 that the complete solution can be
represented in the form
y(n) = yzi (n) + yzs (n)
as the sum of the zero-input response and the zero-state response.
Now, the zero-input response , yzi (n), can be found by solving the homogeneous equation
y(n)
1
y(n 1) = 0,
2
y(1) = 2
The characteristic equation is given by
1
=0
2
so that the homogeneous solution has the form

n
yh (n) = C
1
2
for all n
295
Copyright
296
CHAPTER 12
UNILATERAL
z-TRANSFORM
Using the initial condition y(1) = 2 we conclude that the constant C is equal to 1 and, consequently, the zero-input solution over the desired interval n 0 is given by
n
yzi (n) =
1
2
n0
The zero-state response, on the other hand, can be obtained by determining the response of the
relaxed system
1
yzs (n) = yzs (n 1) + x(n)
(relaxed)
2
over n 0. Obviously, when the system is assumed to be relaxed, then the above difference equation
characterizes an LTI system. We can then determine the response to x(n) = u(n) by working in the
ztransform domain. Recall from Sec. 11.4 that the input-output transform pair of an LTI system is
related via the transfer function of the system, namely.
Yzs (z) = X(z)H(z)
In the current example, the transfer function is given by
H(z) =
1
,
z 1/2
|z| > 1/2
and the ROC is the exterior of a disc because the LTI system is causal (and, consequently, its impulse
response sequence is right-sided).
Using x(n) = u(n) or, equivalently,
X(z) =
we get
Yzs (z) =
so that
z
,
z1
|z| > 1
z
z
z
2z
=
+
,
z 1 z 1/2
z 1/2
z1
|z| > 1
n
1
2
yzs (n) = 2u(n)
u(n)
Combining this response with yzi (n) we arrive at

y(n) = yzi (n) + yzs (n) = 2u(n)
There is an alternative way for solving constant-coefficient difference equations over the
interval n 0 without the need to consider the zero-input and zero-state responses separately. This method relies on using the so-called unilateral z-transform.
12.2 UNILATERAL Z-TRANSFORM

The unilateral z-transform of a sequence x(n) is denoted by X + (z) and is defined by
X + (z) =
x(n)z n
(12.1)
n=0
Observe that the index of the sum runs over n 0 only. In other words, only samples of
x(n) over n 0 enter into the evaluation of X + (z). We say that only the causal part of
the sequence x(n) is used, and we ignore the samples that exist at negative time instants.
Clearly, for sequences that are zero for negative time, both the bilateral ztransform and
297
the unilateral ztransform coincide:
SECTION 12.2
X(z) = X + (z) for causal sequences, i.e., when x(n) = 0 for n < 0
(12.2)
On the other hand, for a generic sequence x(n), we shall denote its causal part by x+ (n),
namely, x+ (n) is a right-sided sequence that contains all the samples of x(n) for n 0
and is zero for n < 0 see Fig. 12.1:

x(n)
for n 0
+
x (n) =
(12.3)
0
otherwise
In other words,
x+ (n) = x(n)u(n)
(12.4)
where u(n) is the unit-step sequence. We therefore see that the unilateral ztransform of
a sequence x(n) coincides with the bilateral ztransform of its causal sequence, x+ (n).
We can denote this observation by the following notation
Z + [x(n)] = Z[x+ (n)] = Z[x(n)u(n)]
(12.5)
where Z refers to the bilateral z-transformation while Z + refers to the unilateral ztransformation.
x(n)
2
3
2
1
x+ (n)
2
3 2 1
FIGURE 12.1 A sequence x(n) and its causal part, x+ (n).
As was the case with bilateral ztransforms, we similarly associate regions of convergence (ROC) with unilateral ztransforms. Thus, the ROC of a unilateral ztransform is
UNILATERAL
z-TRANSFORM
298
CHAPTER 12
UNILATERAL
z-TRANSFORM
defined as the set of all values z in the complex plane for the which the series X + (z) is
absolutely summable:
ROC =
z C such that
n=0
|x(n)z
|<
(12.6)
Observe that the ROC of the unilateral ztransform of x(n) coincides with the ROC of the
bilateral ztransform of x+ (n):
ROC {Z + [x(n)]} = ROC {Z[x+ (n)]}
(12.7)
Note further that since the unilateral ztransform deals only with right-sided sequences,
its ROC will always be the exterior of a disc.
Example 12.2 (Unilateral and bilateral transforms)

The bilateral and unilateral z-transforms of the sequence x(n) = n u(n) coincide and are given by
X(z) =
z
= X + (z),
z
|z| > ||
On the other hand, the bilateral and unilateral z-transforms of the sequence x(n) = n+1 u(n + 1)
are different and are given by
X(z) =
z2
,
z
X + (z) =
z
,
z
|z| > ||
For the latter case, we simply note that the sample at n = 1 is ignored and for n 0, the samples
of x(n) can be described in terms of the following causal sequence
x+ (n) = ()n u(n)
We further note that x+ (n) is a causal sequence so that its bilateral ztransform coincides with its
unilateral ztransform and, hence,
X + (z) =
z
,
z
|z| > ||
Alternatively, we can evaluate X + (z) directly from first principles as follows:

X + (z)
=
=
=
X
n=0
X
n=0
x(n)z n
()n+1 u(n + 1)z n
()n+1 z n
n=0
n=0
1 z 1

1 n
299
provided that |z 1 | < 1. Therefore,

X + (z) =
z
,
z
SECTION 12.3
|z| > ||
12.3 PROPERTIES OF THE UNILATERAL Z-TRANSFORM

The unilateral ztransform has several important properties that can be easily verified by
invoking its definition. A summary of these properties is given in Table 12.1 with the
corresponding regions of convergence. For example, the first two lines of the table start
from two generic sequences x(n) and y(n) and the ROCs of their unilateral ztransforms,
denoted by
Rx+ = { |z| > r },
Ry+ = { |z| > r }
respectively. The sequences x(n) and y(n) need not be right-sided or causal; only in line
8 of the table (dealing with linear convolution), the sequences need to be causal. The subsequent lines in the table provide the unilateral ztransforms of combinations and transformations of these sequences, and their ROCs.
TABLE 12.1 Properties of the unilateral z-transform.

sequence
unilateral z -transform
ROC
1.
x(n)
X + (z)
Rx+ = { |z| > r }
2.
y(n)
Y + (z)
Ry + = { |z| > r }
3.
ax(n) + by(n)
aX + (z) + bY + (z)
{Rx+ Ry + } plus
possibly z = 0
linearity
property
4.
x(n 1)
z 1 X + (z) + x(1)
Rx+
time delay
5.
x(n + 1)
zX + (z) zx(0)
Rx+
time advance
6.
an x(n)
X + (z/a)
|z| > |a|r
7.
nx(n)
8.
x(n) and y(n) causal:

n
P
x(k)y(n k)
k=0
dX + (z)
dz
X (z)Y (z)
Rx+
exponential
modulation
linear modulation
{Rx+ Ry + } plus
possibly z = 0
convolution
PROPERTIES
OF THE
UNILATERAL
z-TRANSFORM
300
CHAPTER 12
UNILATERAL
z-TRANSFORM
12.3.1 Linearity
Consider, for instance, the third line of the table. It states that the unilateral ztransform
of a linear combination of two sequences, namely,
ax(n) + by(n)
for any two numbers a and b, is given by the same linear combination of their unilateral
ztransforms, i.e.,
ax(n) + by(n) aX + (z) + bY + (z)
(12.8)
But what about the ROC of the combination? Obviously, both X + (z) and Y + (z) need to
exist in order for the combination aX + (z) + bY + (z) to be well-defined. This means all
points z Rx+ Ry+ should belong to the ROC of aX + (z) + bY + (z). The ROC of the
combination can be larger than Rx+ Ry+ since the point z = 0 may be included as well.
W + (z)
=
=
=
w(n)z n
n=0
[ax(n) + by(n)]z n
n=0
x(n)z n + b
n=0
y(n)z n
n=0
aX + (z) + bY + (z)
for all values of z Rx+ Ry + . The ROC of W (z) may include z = 0, for example, when w(n)
evaluates to a constant sequence due to cancelations.
An alternative argument can be pursued by relying on the relation between the bilateral and unilateral ztransforms. Thus note that
Z + [ ax(n) + by(n) ]
(by definition)
Z [ ax(n)u(n) + by(n)u(n) ] ,
aZ [ x(n)u(n) ] + bZ [ y(n)u(n) ] ,
(by linearity)
aZ + [ x(n) ] + bZ + [ y(n) ] ,
(by definition)
Similar arguments, which exploit the relation between the unilateral and bilateral ztransforms, can
be used to establish the other properties in Table 12.1.
Consider the sequences

x(n) = (n) 2(n 1) X + (z) = 1 2z 1 with Rx+ = { z 6= 0 }
and
y(n) = 3(n + 1) + 5(n) + 2(n 1) Y + (z) = 5 + 2z 1 with Ry + = { z 6= 0 }
Consider now the linear combination w(n) = x(n) + y(n), which evaluates to
w(n) = 3(n + 1) + 6(n)
301
Its unilateral ztransform is given by
SECTION 12.3
W + (z) = 6 with ROC = { entire complex plane }
UNILATERAL
z-TRANSFORM
It is seen that the ROC of W + (z) is larger than Rx+ Ry + since

Rx+ Ry + = { z 6= 0 }
which excludes z = 0, while z = 0 is not excluded from the ROC of W + (z); this is because when
x(n) and y(n) are added together, the terms 2z 1 and 2z 1 in X + (z) and Y + (z) cancel each
other.
12.3.2 Time Shifts

Consider now the fourth line in Table 12.1. It establishes the transform pair
x(n 1)
z 1 X + (z) + x(1)
(12.9)
In other words, if the original sequence x(n) is shifted by one time instant to the right, then
the corresponding unilateral ztransform is modified by multiplying it by z 1 and adding
x(1)to the result. The ROC of the time-shifted sequence, x(n 1), will coincide with
the ROC of the original sequence x(n). Clearly, if the sequence x(n) is causal to begin
with (i.e., x(n) = 0 for n < 0), then the term x(1) would be zero and relation (12.9)
would simplify to
x(n 1)
z 1 X + (z)
(when x(n) is causal)
Proof: Let w(n) = x(n 1). Then

W + (z)
=
=
X
n=0
X
n=0
w(n)z n
x(n 1)z n
x(1) + x(0)z 1 + x(1)z 2 + x(2)z 3 + . . .
x(1) + z 1 x(0) + x(1)z 1 + x(2)z 2 + . . .
x(1) + z 1
x(1) + z 1 X + (z)
x(n)z n
n=0
for all values of z Rx+ .
Actually, the argument leading to (12.9) can be generalized for higher time-shifts, namely,
it is straightforward to verify that the following transform pair holds for all k 0:
x(n k) z k X + (z) + x(1)z k+1 + x(2)z k+2 + . . . + x(k)
(12.10)
where the ROC coincides with Rx+ . The case k = 1 leads to (12.9), while k = 2 leads to
x(n 2) z 2 X + (z) + z 1 x(1) + x(2)
(12.11)
302
CHAPTER 12
UNILATERAL
z-TRANSFORM
and so forth. Likewise, the following transform pair holds for all k 0:
x(n + k) z k X + (z) x(0)z k x(1)z k1 + . . . x(k 1)z
(12.12)
where the ROC coincides with Rx+ excluding possibly the point z = . In particular,
x(n + 1) zX + (z) zx(0)
2
(12.13)
x(n + 2) z X (z) z x(0) zx(1)
(12.14)
Example 12.4 (Two-sided sequence)

Consider the two-sided sequence
n
1
2
x(n) =
u(n + 2) (3)n u(n + 2)
The sequence x(n) consists of two sub-sequences. The sub-sequence

n
x1 (n) =
1
2
u(n + 2)
is exponentially decaying; it starts at n = 2 and is right-sided. On the other hand, the sub-sequence
x2 (n) = (3)n u(n + 2)
is also exponentially decaying; it starts at n = 2 and is left-sided. It follows that the causal part of
x(n) is given by
n
1
2
x+ (n) =
u(n) (n) 3(n 1) 9(n 2)
and, consequently, the unilateral ztransform of x(n) is given by

X + (z) =
z
z
1
2
1 3z 1 9z 2 ,
|z| > 1/2
where we used the fact that the sequences {(n), (n 1), (n 2)} are causal and, therefore,
their bilateral and unilateral ztransforms coincide.
Let us now use property (12.9) to deduce the unilateral ztransform of x(n1). Using x(1) =
5/3 we get
x(n 1)
=
=
z 1 X + (z) + x(1)

5
z
z 1
1 3z 1 9z 2 +
1
3
z2
5
1
z 1 3z 2 9z 3 +
3
z 12
with ROC given by |z| > 1/2.

Consider the fifth line in Table 12.1. It establishes the transform pair
an x(n)
X + (z/a)
(12.15)
In other words, if the original sequence x(n) is multiplied by the exponential sequence an ,
for some nonzero constant a, then the corresponding unilateral ztransform is modified
by replacing the independent variable z by z/a. The ROC of the exponentially-weighted
sequence, an x(n), is given by
ROC = {z C such that |z| > |a|r }

Proof: Let w(n) = an x(n). Then
W + (z)
n=0
n=0
n=0
w(n)z n
an x(n)z n
x(n)(z/a)n
x(n)(z )n ,
using z = z/a
n=0
+
X (z )
=
for all values of z Rx+ .
Example 12.5 (Alternating signs)

Consider a sequence x(n) and denote the ROC of its unilateral ztransform, X + (z), by Rx+ =
{|z| > r1 }. Let us determine the unilateral ztransform of the sequence (1)n x(n). This sequence
amounts to reversing the signs of all odd-indexed samples of x(n). Using property (12.15), we find
that
(1)n x(n) X + (z),
with ROC= Rx+
(12.16)

The unilateral ztransform of the sequence
x(n) = cos(o n)u(n)
coincides with its ztransform from Example 9.11 since the sequence is causal:
cos(o n)u(n)
1 z 1 cos o
,
1 2z 1 cos o + z 2
for |z| > 1
z 1 sin o
,
1 2z 1 cos o + z 2
for |z| > 1
Similarly, we can verify that

sin(o n)u(n)
303
SECTION 12.3
UNILATERAL
z-TRANSFORM
304
Example 12.7 (Exponential modulation of sinusoidal sequences)
CHAPTER 12
UNILATERAL
z-TRANSFORM
Likewise, the bilateral and unilateral ztransforms of the sequence

x(n) = an cos(o n)u(n)
coincide with each other since the sequence is causal. From Example 9.12 we therefore have
an cos(o n)u(n)
1 az 1 cos o
, for |z| > |a|
1 2az 1 cos o + a2 z 2
an sin(o n)u(n)
az 1 sin o
, for |z| > |a|
1 2az 1 cos o + a2 z 2
Likewise,

Consider the seventh line in Table 12.1. It establishes the transform pair
nx(n)
dX + (z)
dz
(12.17)
In other words, if the original sequence x(n) is multiplied by the linear sequence n, then
the corresponding unilateral ztransform is modified by replacing it by its derivative with
respect to z multiplied by z. The ROC of the linearly modulated sequence, nx(n), is
Rx+ .
Proof: Let w(n) = nx(n) and recall first the definition of X + (z):
X + (z) =
x(n)z n
n=0
for all values of z Rx+ . The series X + (z) is absolutely summable over Rx+ . Thus, differentiating
it with respect to z we can write
dX + (z)
dz
n=0
n=0
so that
z
x(n)

x(n)
dz n
dz
nz n1
z 2n
X
dX + (z)
=
nx(n)z n = W + (z)
dz
n=0
And the ROC of W + (z) coincides with the ROC of X + (z).
305
Example 12.8 (Linearly-modulated exponential sequence)
SECTION 12.3
The bilateral and unilateral ztransforms of the sequence x(n) = nn u(n) coincide since the
sequence is causal. We therefore conclude from Example 9.14 that
nn u(n)
z 1
z
=
,
(1 z 1 )2
(z )2
for |z| > ||
Consider the eighth line in Table 12.1, where it is now assumed that the sequences x(n)
and y(n) are causal, i.e.,
x(n) = 0 and
y(n) = 0
for n < 0
(12.18)
Then the following transform pair relation holds:

x(n) y(n)
X + (z)Y + (z)
(12.19)
In other words, convolution of two causal sequences in the time domain amounts to multiplication in the transform domain. The ROC of the linear convolution is Rx+ Ry+ plus
possibly z = 0.
Proof: Let
w(n)
=
=
x(n) y(n)
X
k=
n
X
k=0
x(k)y(n k)
x(k)y(n k)
where the last equality is because both sequences x(n) and y(n) are assumed to be causal (have zero
samples over n < 0). It follows that w(n) is also a causal sequence,
w(n) = 0
for
n<0
Therefore, the bilateral and unilateral ztransforms of w(n) coincide,

W + (z) = W (z)
Now, the bilateral ztransform for w(n) satisfies
W (z) = X(z)Y (z)
since w(n) is the linear convolution of x(n) and y(n). Moreover, since x(n) and y(n) are themselves
causal sequences, it holds that
X + (z) = X(z),
Y + (z) = Y (z)
We therefore conclude that

W + (z) = X + (z)Y + (z)
as desired, for all z Rx+ Ry + plus possibly z = 0.
UNILATERAL
z-TRANSFORM
306
CHAPTER 12
UNILATERAL
z-TRANSFORM
Let us evaluate the linear convolution of the two exponential sequences

n
1
2
x(n) =
n
u(n),
1
3
y(n) =
u(n)
Both sequences are causal and therefore their bilateral and unilateral ztransforms agree, namely,
X(z) = X + (z) =
z
z
Y (z) = Y + (z) =
z
z
and
1
2
1
3
|z| > 1/2
|z| > 1/3
Let w(n) = x(n) y(n). Then w(n) is also a causal sequence and
W (z) = W + (z) = X + (z)Y + (z) =
z
z
1
2
z
z
1
3
|z| > 1/2
Using partial fractions we can express W + (z) in the form

W + (z) =
B
A
+
z 21
z
1
3
where the constants {A, B} can be determined from the residue relations:

z 2
= 3/2
z 1/3 z=1/2
W + (z)(z 1/2)z=1/2 =
W + (z)(z 1/3)z=1/3 =
z 2
= 2/3
z 1/2 z=1/3
That is,
W + (z) =
3 1
z
2
"
so that
w(n) =
z
z 1/2
3
2
n1
1
2
2 1
z
3
2
3
z
z 1/3
|z| > 1/2
n1 #
1
3
u(n 1)
12.4 INITIAL AND FINAL VALUE THEOREMS

The initial value theorem of Sec. 9.6 still holds for the unilateral ztransform, especially
since the theorem is only valid for causal sequences. Thus, the unilateral ztransform also
allows us to recover the value of the original sequence x(n) at time 0 without the need
to perform inverse-transformation. Specifically, assume that x(n) is a causal sequence,
namely,
x(n) = 0, for n < 0
so that x+ (n) = x(n). Then, X(z) = X + (z) and, in view of the result we established
earlier in (9.28), it still holds that
lim X + (z) = x(0)
(12.20)
where the ROC of X + (z) needs to include the point z = so that the limit in (12.20) is
well-defined.
307
We now establish a second result known as the final value theorem. If
SECTION 12.4
INITIAL AND
FINAL VALUE
THEOREMS
lim x(N )
exists, then it holds that

lim x(N ) = lim (z 1)X + (z)
z1
(12.21)
The limits exist if the ROC of (z 1)X + (z) includes the unit circle.
Proof: We consider the unilateral ztransform of the sequence w(n) = x(n + 1) x(n):
W + (z)
X
n=0
[ x(n + 1) x(n) ]z n
N
X
lim
n=0
lim
[ x(n + 1) x(n) ]z
x(0) + (1 z 1 )x(1) + . . . + (1 z 1 )z (N1) x(N ) + x(N + 1)z N
Taking the limit of W + (z) as z 1 we find that

lim W + (z)
z1
lim [ x(0) + x(N + 1) ]
x(0) + lim x(N + 1)
x(0) + lim x(N )
On the other hand, using the properties of the unilateral ztransform we have that
W + (z)
=
=
zX + (z) zx(0) X + (z)
(z 1)X + (z) zx(0)
so that
lim W + (z) = lim (z 1)X + (z) x(0)
z1
z1
We arrived at two expressions for the limit of W + (z) as z 1. Equating the expressions we
conclude that the following result holds:
lim (z 1)X + (z) = lim x(N )
z1
Example 12.10 (Initial and final values)

Consider the unilateral ztransform
X + (z) =
0.5z
,
z 2 1.5z + 0.5
|z| > 1
Taking the limit as z we find that

lim X + (z) = 0
so that x(0) = 0. Likewise, taking the limit of (z 1)X + (z) as z 1 we get

lim (z 1)X + (z) = lim
z1
z1
0.5z
= 1
z 0.5
308
CHAPTER 12
so that x() = 1. Let us confirm these results by inverse transforming the strictly proper rational
function X + (z) by means of partial fractions, say
UNILATERAL
z-TRANSFORM
0.5
A
B
X + (z)
=
=
+
z
(z 1)(z 0.5)
z1
z 0.5
where A and B are found from the residue relations

0.5
(z 1)
(z 1)(z 0.5)
z=1

0.5
= 1
(z 0.5)
(z 1)(z 0.5)
z=0.5
= 1
Therefore,
X + (z) =
z
z1
and
z
z 0.5
|z| > 1
n
x(n) = 1
1
2
u(n)
Thus, note that x(0) = 0 and x() = 1, as expected.
12.5 SOLVING DIFFERENCE EQUATIONS

Let us now illustrate how the unilateral z-transform can be used to solve constant-coefficient
difference equations without the need to consider the zero-input and the zero-state responses separately, as was done in Example 12.1. We reconsider the same example.
Example 12.11 (Using the unilateral ztransform)
Consider the constant-coefficient difference equation

y(n)
1
y(n 1) = x(n), y(1) = 2
2
and let us again determine its response to the input sequence x(n) = u(n) whose unilateral ztransform
is
z
,
|z| > 1
X + (z) =
z1
Applying the unilateral ztransform to both sides of the difference equation, and using the time-shift
property from Table 12.1, we obtain
Y + (z)

1 1 +
z Y (z) + y(1) = X + (z)
2
Solving for Y + (z) we get

Y + (z) =
y(1)
z
2
z
1
2
+ X + (z)
z
z
1
2
Thus, note that the expression for Y + (z) consists of two terms: the first term on the right-hand side
contains the contribution of the initial condition y(1), and the second term on the right-hand side
contains the contribution of the input sequence, x(n). We therefore say that the first-term corresponds
to the zero-input response and the second-term corresponds to the zero-state response.
309
Using the values for X + (z) and y(1) = 2, we find that

Y + (z)
z
z
1
2
| {z }
z
z
z1z
Yzi (z)
{z
SECTION 12.7
APPLICATIONS
1
2
Yzs (z)
where we are identifying the transforms that correspond to the zero-input contribution and the zerostate contribution. Expanding the zero-state term into partial fractions gives
Y + (z)
=
=
z
2z
z
+
+
z1
z 12
z 21
2z
, |z| > 1
z1
By inverse transformation of Y + (z) we obtain a causal sequence and it is given by

y(n) = 2u(n)
12.6 APPLICATIONS
TO BE ADDED
Practice Questions:
1.
2.
12.7 PROBLEMS
Problem 12.1 Find the unilateral ztransforms and ROCs of the following sequences:
(a) x(n) =
(b) x(n) =
(c) x(n) =
(d) x(n) =

1 n2
u(n 4).
2

1 n2
u(n).
2

1 n2
n 2
u(n + 2).

n+2
u(n + 1).
n2 12
(a) x(n) = (n 1)

1 n
2
(b) x(n) =
(c) x(n) = n
(d) x(n) = n(n

1 n+1
2
u(n 1).
u(n 2).

1 n1
u(n + 3).
2

1 n
1) 2 u(n + 2).
(a) x(n) = n cos
(n
3
(b) x(n) = (n 2) cos

(c) x(n) = n

1 n
2
sin
1) u(n + 2).
(n 1) sin 3 (n
3
n 3 u(n + 1).
6
1) u(n 3).
310
CHAPTER 12
UNILATERAL
z-TRANSFORM
n u(n 1).
6

1 n1
sin 3 n 2
n 2
u(n
3

1 2n2
cos 3 n u(n + 2).

2
(a) x(n) = cos

(b) x(n) =
(c) x(n) =
+ 3).
Problem 12.5 Determine the unilateral ztransforms of each of the following sequences and indicate their regions of convergence:
(a) x(n) = (n 1)u(n + 1).
(b) x(n) = 1 + n2 n2 u(n + 31).

(c) x(n) = |n| , with > 0.
(d) The impulse response sequence of the relaxed causal system y(n) 34 y(n1)+ 18 y(n2) =
x(n 1).
Problem 12.6 Determine the unilateral ztransforms of each of the following sequences and indicate their regions of convergence:
(a) x(n) = n2 u(n + 2).
(b) x(n) = 1 n2n2 u(n 1).

(c) x(n) = |n| , with > 0.
(d) The impulse response sequence of the relaxed causal system y(n) 14 y(n1) 18 y(n2) =
2x(n 2).
Problem 12.7 Find the unilateral ztransform of the sequence
x(n) = nu(n + 1) +
n
1
2
u(n + 3)
Problem 12.8 Find the unilateral ztransform of the sequence

x(n) = n2 u(n + 2) +
n+2
1
4
u(n 1)
Problem 12.9 Let

n
x(n) =
1
2
2n
u(n 1),
1
3
h(n) =
u(n 3)
Use the unilateral ztransform technique to evaluate the following sequences:

(a) x(n) h(n).
(b) x(n 2)

1 n
4
h(n).
(c) x(n) h(n 1)

1 n
4
u(n).
Problem 12.10 Let

n
x(n) = n
1
3
n1
u(n),
h(n) =
1
4
u(n 2)
Use the unilateral ztransform technique to evaluate the following sequences:

(a) x(n) h(n).
(b) x(n 1)
(c) x(n) sin
1 n1
h(n 2).
3
n h(n 1).
3

X + (z) =
1
(z 1/8)(z + 1/5)
311
What is its ROC? Determine the limiting value x() in two ways.
SECTION 12.7
PROBLEMS

X + (z) =
1
(z + 1/8)(z 1/4)
X + (z) =
1
(z + 1/8)2 (z + 1/3)
X + (z) =
1
z 2 + 1/4
X + (z) =
1
z 2 + 1/9
X + (z) =
1
(z 1/2)(z 2 + 1/25)
X + (z) =
1 2
z
3
+ z 1 + z
z 2 61 z 16
1
2
Determine the limiting value x() in two ways.

X + (z) =
z2
z 2
41 z
1
8
Determine the limiting value x() in two ways.

y(n) y(n 1) +
1
y(n 2) = x(n 1),
4
y(1) = 0,
y(2) = 7/2
Use the unilateral ztransform technique to determine its complete response when x(n) =

1 n+1
2
u(n).

y(n)
1
1
y(n 1) y(n 2) = x(n 1),
8
8
y(1) = 0,
y(2) = 1
Use the unilateral ztransform technique to determine its complete response when x(n) = 14
n
u(n).
312
CHAPTER 12
UNILATERAL
z-TRANSFORM
y(n) =
5
1
y(n 1) y(n 2) + x(n 2),
6
6
y(2) = 0, y(1) = 1.
Use the unilateral ztransform to determine its complete response to the sequence
n2
x(n) = (n 1)
1
4
u(n 1).
y(n) =
1
1
y(n 1) + y(n 2) + x(n 1),
4
8
y(2) = 1, y(1) = 0.
Use the unilateral ztransform to determine its complete response to the sequence
2n2
x(n) = n
1
3
u(n 2).

y(n) y(n 1) +
1
y(n 2) = x(n),
4
y(1) = 0,
Use the unilateral z-transform to find its complete response to x(n) =
y(2) = 4.

1 2n
2
u(n 1).

y(n) + 2y(n 1) + 2y(n 2) = x(n),
y(1) = 0,
Use the unilateral z-transform to find its complete response to x(n) =
y(2) = 1.

1 n
3
u(n 1).
Problem 12.25 Consider the constant-coefficient difference equation

y(n)
1
1
y(n 1) y(n 2) = x(n 1)
6
6
with initial conditions y(2) = 0 and y(1) = 6. Use the ztransform technique to find the
answers to parts (a)-(c):
(a) The zero-input response.
(b) The zero-state response to x(n) = u(n).
(c) The complete response using the unilateral ztransform.
(d) Check that your answer to part c) is the sum of the answers to parts (a) and (b).
(e) Now determine the answers to parts (a)-(c) by using the time-domain techniques you learned
earlier for solving constant-coefficient difference equations by working with the modes of the
system. Compare your answers to those obtained by using the ztransform technique.
n
(f) Find the response of the system to x(n) = 13 u(n 2) in three different ways.
Problem 12.26 A causal system is composed of the series cascade of two
n LTI subsystems with
n
impulse response sequences given by h1 (n) = 12 u(n) and h2 (n) = 13 u(n 1).
(a) Determine a description for the system in terms of a constant-coefficient difference equation.
Denote its input and output sequences by x(n) and y(n), respectively.
(b) Draw a block-diagram representation for the system using only two delay elements.
(c) Assume the system is relaxed, determine its impulse response and transfer function.
(d) Is the system stable? What are its modes?
(e) Determine an input sequence such that only the largest mode appears at the corresponding
output sequence.
(f) Assume the system is not relaxed, determine initial conditions y(1) and y(2) such that
only the smallest mode appears at the impulse response of the system.
CHAPTER
13
Discrete-Time Fourier Transform
In the earlier chapters we characterized the behavior of signals and systems both in the
time-domain and in the transform domain. For example, we studied properties of signals
and systems in the time-domain (such as periodicity, causality, stability, solutions of difference equations). We also studied signals and systems in the ztransform domain (such
as transform representations of signals and transfer functions for LTI systems). We related
both domains of studying systems and showed how to move back and forth between the
time domain and the transform domain.
In this chapter, we show how the transform domain representation of signals (in terms
of their ztransforms) and systems (in terms of their transfer functions) can be used to
motivate yet another useful characterization in the frequency domain. The concepts that
will be described in this chapter, and in subsequent chapters, will enable us to describe the
frequency content of a signal and the frequency response of a system. The first step towards
this goal is to introduce the Discrete-Time Fourier Transform (DTFT) of a sequence and to
study its properties.
13.1 DEFINITION OF THE DTFT

Consider an arbitrary sequence x(n) and let X(z) denote its bilateral ztransform:
X(z) =
x(n)z n
(13.1)
n=
Let ROC denote the region of convergence of X(z), namely, ROC is the set of all points z
in the complex plane where the defining series of X(z) converges absolutely:
ROC of X(z) =
X

n
x(n)z <
z C such that
(13.2)
n=
Assume, for the time being, that the ROC of X(z) includes the unit circle. That is, assume
it includes all points z satisfying |z| = 1. Then it must hold that the sequence x(n) itself
is absolutely summable, i.e.,
X
|x(n)| <
(13.3)
n=
For such sequences, we can evaluate X(z) at any point on the unit circle, i.e., at any point
z of the form z = ej for any angular frequency see Fig. 13.1. Usually, we limit to
the intervals w [, ] or [0, 2]. As varies over either interval, the variable ej
313
Copyright
314
covers the entire unit circle. If we substitute z in (13.1) by z = ej we get
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM
X(ej ) =
x(n)ejn
(13.4)
n=
The quantity X(ej ) so-defined is called the Discrete-Time Fourier transform (DTFT) of
the sequence x(n). We express the DTFT of x(n) as a function of ej and write X(ej ),
instead of X(), in order to emphasize the fact that the DTFT amounts to evaluating the
ztransform of x(n) on points that lie on the unit circle.
Im
z = ej
1
FIGURE 13.1
Re
Any point z on the unit circle can be expressed as z = ej for some [, ].
Observe that the DTFT of a sequence is a function of a real-valued variable . As explained in Sec. 3.3, the variable in the exponential sequence ejn that appears in (13.4)
plays the role of a frequency variable and it is measured in radians/sample. Moreover, although can in principle assume any value in the range (, ), as mentioned above,
we shall often limit to the 2-wide intervals [, ] or [0, 2] for reasons that will become clear as we progress in the discussion (in particular, we are going to see that X(ej )
is periodic in of period 2 and, therefore, it is sufficient to specify its behavior over a
2-wide interval).
The DTFT of an absolutely summable sequence x(n) can therefore be computed in one
of two ways:
(1) Find its ztransform, X(z), and replace the variable z by ej to get X(ej ).
(2) Evaluate X(ej ) directly the defining series (13.4).
Example 13.1 (Finite duration sequence)

x(n) = 0.5(n + 1) + (n) + 0.5(n 1)
It consists of three nonzero samples at time instants n = 1, 0, 1 see Fig. 13.2. Using the definition
(13.4) for the DTFT we get
315
SECTION 13.1
DEFINITION
OF THE
DTFT
x(n)
1
0.5
1
FIGURE 13.2 A sequence x(n) with 3 nonzero samples at n = 1, 0, 1.
X(ej ) = 0.5ej + 1 + 0.5ej

which is a well-defined function of [, ]. If we further call upon Eulers relation (3.11) we
can rewrite X(ej ) in the equivalent form
X(ej ) = 1 + cos(),
[, ]
Figure 13.3 shows a plot of X(ej ) over the interval [, ]. In this example, X(ej ) assumes
real values between 0 and 2. Observe further that the DTFT in this case is a continuous function of
.
2
1.8
1.6
1.4
X(e )
1.2
1
0.8
0.6
0.4
0.2
0
FIGURE 13.3
0
1
(radians/sample)
A plot of the DTFT X(ej ) = 1 + cos() over the interval [, ].
Example 13.2 (Unit-sample sequence)
The DTFT of the unit-sample sequence

x(n) = (n)
316
is obviously
X(ej ) = 1
CHAPTER 13
Here again, the DTFT is a real-valued function of ; it assumes the constant value 1 over [, ].
In this case, we say that the DTFT is flat over [, ] . Figures 13.4 and 13.5 illustrate this
situation.
x(n)
1
FIGURE 13.4 A plot of the unit-sample sequence x(n) = 1.
2
1.8
1.6
1.4
1.2
j
X(e )
DISCRETE-TIME
FOURIER
TRANSFORM
1
0.8
0.6
0.4
0.2
0
0
1
(radians/sample)
FIGURE 13.5 A plot of the DTFT X(ej ) = 1 over [, ] .
Example 13.3 (Delayed unit-sample sequence)

Consider the delayed unit-sample sequence
x(n) = (n no )
for some integer value no (it can be positive or negative). Using (13.4), the DTFT of x(n) is easily
seen to be
X(ej ) = ejno ,
[, ]
In this case, we find that the DTFT is a complex-valued function of . For each , the value of
X(ej ) is a complex number whose magnitude is one and whose phase is no .
Let us examine the phase of the DTFT more closely in this example. Assume no = 2 so that
X(ej ) = e2j and the phase is given by
X(ej ) = 2
It is seen that the phase of X(ej ) varies linearly. Plotting the phase variation over [, ] we see
in Fig. 13.6 that it decreases from 2 at = down to 2 at = .
phase plot
6
X(ej)
6
3
0
1
(radians/sample)
FIGURE 13.6 A plot of the phase of X(ej ) = ej2 over the interval [, ].
Recall, however, from the discussion in Sec. 2.1 that we can always limit the phase angle of a
complex number to the interval [, ]. If the phase angle of a number lies outside this interval,
then we can modify it by adding suitable integer multiples of 2 in order to replace the phase angle
by an equivalent angle representation within the interval [, ]. For example, the following two
polar representations
5
ej 3
and
ej 3
are equivalent: the first one has phase angle 5/3 (outside [, ]), while the second one has phase
angle /3 (inside [, ]). The equivalence between both representations can be seen from the
following manipulation
5
5
ej 3 = ej ( 3 2) = ej 3
since
ej2k = 1,
for any integer k
We use this explanation to motivate an alternative way to present the phase plot of a DTFT; we
shall adopt this alternative representation in all our subsequent discussions. Figures 13.7 and 13.8
show the plots that correspond to the situation x(n) = (n 2), where no = 2 for illustration
purposes. Observe that now we are using two plots in Fig. 13.8 to represent X(ej ): one plot is
used to illustrate its magnitude values over [, ] and a second plot is used to illustrate its
phase values over the same interval (when X(ej ) is real, we only need one plot, as was the case
with Figs. 13.3 and 13.5). Comparing with Fig. 13.6, we observe that the phase plot of X(ej ) in
Fig. 13.8 now exhibits an interesting behavior; each time the value of the phase angle is about to
leave the interval [, ], the angle is modified by 2 so that the plot remains within [, ]. For
example, consider the plot of the phase starting at = 0. As increases towards /2, the value of
the phase angle decreases towards . At w = /2, we add 2 to the phase angle so that its value
jumps to from where it starts to decrease again as continues to increase towards .
317
SECTION 13.1
DEFINITION
OF THE
DTFT
318
CHAPTER 13
x(n) = (n 2)
DISCRETE-TIME
FOURIER
TRANSFORM
FIGURE 13.7 A plot of the delayed unit-sample sequence x(n) = (n 2).
magnitude plot
phase plot
3
2
X(ej)
|X(ej)|
1.5
1
0.5
0
3
1
0
1
2
1
0
1
2
(rad/sample)
3
3
1
0
1
2
(rad/sample)
/2
FIGURE 13.8 A plot of the magnitude (left) and phase (right) of the complex-valued DTFT
X(ej ) = ej2 over [, ] .

Consider the exponential
x(n) = n u(n)
Using the definition (13.4), its DTFT is given by
X(ej ) =
X
n=0
(ej )n
which involves the sum of the terms of a geometric sequence with ratio ej and first term equal
to 1. Consequently,
1
,
|| < 1
X(ej ) =
1 ej
provided that || < 1. In other words, the DTFT of an exponential sequence exists for values of
that are inside the open unit disc.
Alternatively, we can arrive at the same conclusion by starting from the ztransform of the
exponential sequence. We already know from Sec. 9.3 that the ztransform of x(n) is given by
X(z) =
z
,
z
|z| > ||
with the ROC defined as the set of all points z satisfying |z| > ||. Replacing z by ej we obtain
the above DTFT. However, for this substitution to be valid we need to guarantee that the ROC of
X(z) includes the unit circle. This will be possible when || < 1 since then the condition |z| > ||
will include all points |z| = 1.
Thus, note that the DTFT of the exponential sequence n u(n) is defined only for || < 1. In
contrast, the ztransform of the same sequence is more general and is defined for any and for
values of z satisfying |z| > ||.
Similarly, consider the exponential sequence
x(n) = n u(n 1)
Using the definition (13.4), its DTFT is given by
X(ej )
=
=
1
X
(ej )n
n=
(1 ej )n
n=1
which involves the sum of the terms of a geometric sequence with ratio 1 ej and first term equal
to 1 ej . Consequently,
X(ej ) =
1
,
1 ej
|| > 1
provided now that || > 1.
Example 13.5 (Rectangular pulse)
Consider the rectangular pulse sequence defined as

(
x(n) =
1,
0,
0 n L1
otherwise
In other words, the sequence x(n) assumes the value 1 over the interval 0 n L 1 and is zero
elsewhere. We say that the width of the rectangular pulse is L samples. Using definition (13.4), the
319
SECTION 13.1
DEFINITION
OF THE
DTFT
320
DTFT of x(n) is given by
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM
X(ej )
L1
X
ejn
n=0
1 ejL
1 ej
(using the sum of geometric terms from Example 2.11)
=
=
ejL/2 ejL/2 ejL/2
(extracting the terms ejL/2 and ej/2 )

ej/2 (ej/2 ej/2 )
sin (L/2)
ej(L1)/2
(using Eurlers relation (3.12))
sin (w/2)
We thus arrive at the following important DTFT pair:

(
x(n) =
1,
0,
0nL1
otherwise
DTFT
X(ej ) =
8
>
< L,
j
>
: e
(L1)
2
sin
sin
when = 0 >
=
L
2
otherwise
The value of X(ej ) at = 0 is obtained by applying LHospitals rule to the ratio

sin (L/2)
sin (/2)
Specifically,
lim
sin (L/2)
sin (/2)
= lim
L
cos (L/2)
2
1
cos (/2)
2
= L
In this example, the magnitude of X(ej ) is given by

sin (L/2)

sin (/2)
|X(ej )| =
The phase of X(ej ), on the other hand, is dictated by the linear factor
(L 1)
2
and by the sign of the term

sin (L/2)
sin (/2)
When the above term is positive, the phase of X(ej ) is simply (L 1)/2. When the sign of the
above factor is negative then we need to correct the term (L 1)/2 by adding to it. Which
sign we choose for is not really relevant, except that it is customary to choose that sign for that
keeps the plot of the phase of X(ej ) within the interval [, ]. Figures 13.9 and 13.10 illustrate
this situation for the case L = 5.
13.2 UNIFORM CONVERGENCE

The DTFT of a sequence x(n) was motivated in Sec. 13.1 by specializing its ztransform
to the unit-circle, z = ej . A such, the sequence x(n) was required to be absolutely
summable, i.e.,
n=
|x(n)| <
(13.5)
>
;
321
SECTION 13.2
x(n)
UNIFORM
CONVERGENCE
FIGURE 13.9 A plot of the rectangular pulse x(n) with width L = 5.
phase plot
3
2
X(e )
|X(e )|
magnitude plot
5
2
1
0
1
2
1
0
1
2
(rad/sample)
3
3
sin( L/2)/sin(/2)
0
3
1
0
1
2
(rad/sample)
1
0
1
2
(rad/sample)
4.5
3
1.5
0
1.5
3
FIGURE 13.10
A plot of the magnitude (top left) and phase (top right) of DTFT of the
rectangular pulse of width L = 5. The bottom right plot shows the variation in the sign of the
ratio sin(L/2)/ sin(/2) over [, ] . Observe that whenever this ratio changes sign (from
positive to negative or from negative to positive), a factor of is added to the phase plot.
in order to ensure that the unit circle, |z| = 1, belongs to the ROC of X(z). The condition
of an absolutely summable sequence x(n) guarantees that the series (13.4) defining the
DTFT converges in a desirable sense known as uniform convergence. What this notion of
convergence means is the following. Let N > 0 be a finite integer and consider the partial
sum
XN (ej ) =
N
X
x(n)ejn
(partial sum)
(13.6)
n=N
This sum amounts to limiting the series (13.4) to the samples of x(n) that lie between
the time instants n = N and n = N . Then, the uniform convergence of the series
(13.4) defining X(ej ) means that, for large enough N , we can use XN (ej ) as a good
approximation for X(ej ). More explicitly, the following three properties hold:
322
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM
(a) First, uniform convergence means that no matter which value of we pick in the
interval [, ], then we can always find a small enough number > 0 and a large
enough integer No > 0 such that

XN (ej ) X(ej ) ,
for every N > No
(13.7)
In other words, the finite sum XN (ej ) is a good approximation for X(ej ) for
sufficiently large N . Even more importantly, the sum XN (ej ) is a good approximation for all [, ]. The qualification uniform in uniform convergence
is used to emphasize that the error bound is the same for all . Hence, at any ,
the approximation XN (ej ) will be close to X(ej ); it will not hold that at some
the approximation will be much worse than at other values of . Instead, the finite
sum approximation will be of similar uniform quality (and satisfy the same error
bound) for all [, ].
(2) Second, uniform convergence implies point-wise convergence at each . That is, it
holds that
lim XN (ej ) = X(ej )
(13.8)
1. Third, uniform convergence implies that X(ej ) is a continuous function of . This

is because, as indicated by property (13.7), the DTFT X(ej ) can be viewed as
the limiting function of the uniformly convergent sequence of continuous functions
in , {XN (ej )}. And it is known that when a sequence of continuous functions
converges uniformly, then their limit is a continuous function as well. Therefore,
we conclude that the DTFT, X(ej ), of an absolutely summable sequence, x(n), is
necessarily a continuous function of .

Consider the exponential sequence from Example 13.4, namely,
x(n) = n u(n)
We already know that this sequence is absolutely summable for || < 1 and, accordingly, its DTFT
converges uniformly and is given by
X(ej ) =
1
,
1 ej
|| < 1
For illustration purposes let us assume that is real-valued in this example. Consider now the finite
sum approximation
XN (ej )
UNIFORM
CONVERGENCE
N
X
x(n)ejn
n=N
N
X
n u(n)ejn
n=N
N
X
n ejn
n=0
N
X
ej
n
n=0
N+1
1 ej
1 ej
We can evaluate how close this approximation is to X(ej ) as follows:

j
j
XN (e ) X(e )
Thus, observe that the difference

ej N+1

1 ej
||N+1
|1 ej |
||N+1
|1 cos() + j sin()|
||N+1
(1 cos())2 + ( sin())2
1+
cos2 ()
||N+1
2 cos() + 2 sin2 ()
||N+1
1 + 2 2 cos()

j
j
XN (e ) X(e )
decays exponentially with N due to the factor ||N+1 in the numerator and, hence,
lim XN (ej ) = X(ej )
Moreover, the convergence is uniform as can be seen from the following argument. First note that
the function
1 + 2 cos()
is always positive since || < 1 and cos() assumes values between 1 and 1. Let > 0 denote
the minimum value of this function:
=
min
[,]
323
SECTION 13.2
1 + 2 cos()
For any desired error bound > 0, select No such that

||No +1 <
CHAPTER 13
Now since || < 1 and, usually, < 1, the above condition implies
DISCRETE-TIME
FOURIER
TRANSFORM
log ( )
log ||
No + 1 >
so that
||N+1 <
for any N > No . Consequently, in view of the definition of , we have

||N+1
= , for all N > No

<
1 + 2 cos()

j
j
XN (e ) X(e ) < ,
It follows that
for all N > No
and we conclude
that the convergence
of XN (e ) to X(e ) is uniform. Figure 13.11 plots the

difference XN (ej ) X(ej ) over the interval [, ] for increasing values of N and using
= 1/2. The plot on the left uses a linear scale while the plot on the right uses a dB scale and
displays the values of

20 log 10 XN (ej ) X(ej ) (dB)
It is seen that as N increases, the size of the difference decreases and the approximation becomes
better.
linear scale
x 10
dB scale
2.2
55
N=8
N=8
1.8
60
1.4
N=9
1.2
|XN(e )X(e )|
1.6
|X (ej)X(ej)| (dB)
324
1
0.8
0.6
65
N=10
70
N=10
75
N=11
0.4
N=11
N=12
N=12
0.2
3
N=9
1
0
1
2
(rad/sample)
80
3 2 1
0
1
2
(rad/sample)
FIGURE 13.11 A plot of the difference XN (ej ) X(ej ) for the exponential sequence
x(n) = (0.5)n u(n) over the interval [, ] and for increasing values of N . The plot on
the left uses a linear scale while the plot on the right uses a dB scale.
325
13.3 INVERSE DTFT
SECTION 13.3
There is a useful inversion formula that allows us to recover a sequence x(n) from knowledge of its DTFT. To see this, we start from the defining relation
X(ej ) =
x(k)ejk
(13.9)
k=
and assume, for the time being, that the sequence {x(n)} is absolutely summable so that
the above series converges uniformly to X(ej ).
We multiply both sides of (13.9) by ejn to get
X(ej )ejn =
x(k)ej(nk)
k=
Integrating over any interval of length 2, say, over [, ], we obtain

Z
1
2
X(e )e
jn
1
2
x(k)ejk
k=
x(k)
k=
1
2
ej(nk) d
(13.10)
where we exchanged the integration and summation signs on the right-hand side of the
above equality by virtue of the assumed uniform convergence of the series (13.9). Let us
now examine the integral expression on the right-hand side of (13.10). Assume initially
that k = n then, obviously,
1
2
ej(nk) d =
1
2
d = 1
On the other hand, when k 6= n, we have

1
2
ej(nk) d =
Therefore, it holds that

1
2

1
1
ej(nk)
= 0, k 6= n
2 j(n k)
=
j(nk)
d =
0
1
k 6= n
k=n
(13.11)
Substituting into the right-hand side of (13.10), we conclude that the following inversion
formula must hold
R
1
X(ej )ejn d
x(n) = 2
(13.12)
Actually, the same argument shows that the inversion formula holds when the integration
is carried over any 2-wide interval, and not only [, ]. For example, we can also write
x(n) =
1
2
X(ej )ejn d
(13.13)
INVERSE
DTFT
326
In order to emphasize this fact, we shall write more generically
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM
x(n) =
1
2
X(ej )ejn d
(13.14)
R
where the 2 under the integration symbol, , is used to refer to any 2long interval. The
above arguments show that it is sufficient for us to know X(ej ) over a 2-wide interval
in order to recover x(n).
Now, although expressions (13.12)(13.14) were derived under the assumption of an absolutely summable sequence x(n) and, hence, continuous X(ej ), we shall nevertheless
apply these inverse expressions to other more general sequences that may not be absolutely
summable for the reasons explained in the sequel starting in the next section. Loosely, this
is because the inversion formulas (13.12)(13.14) can be applied under weaker notions of
convergence for the series (13.9) than uniform convergence.
Example 13.7 (Low-pass DTFT)
Let us use the inversion formula (13.14) to recover the sequence x(n) whose DTFT over the interval
[, ] is given by
(
1,
0,
X(e ) =
|| < c
c ||
(13.15)
That is, the DTFT is equal to one over the interval [c , c ] and is zero elsewhere see Fig. 13.12.
In this case, we say that the DTFT is limited to a low-pass interval of angular frequencies; recall from
the discussion that led to Fig. 3.8 earlier that values of close to 0 are qualified as low frequencies
while values of close to are qualified as high frequencies. We shall have more to say about
this terminology in a later chapter.
X(ej )
(rad/sample)
FIGURE 13.12 A plot of the DTFT (13.15) over [, ].
Note that X(ej ) is not a continuous function of in this example because of the discontinuities
at = c . Therefore, the corresponding inverse sequence, x(n), cannot be absolutely summable
and the series (13.9) defining X(ej ) could not have converged uniformly. Still, we are going to use
the inversion formula (13.12) to find that
x(0)
1
2
d =
c
327
and, for n 6= 0,
x(n)
=
=
=
=
In summary, we get
x(n) =
SECTION 13.4
c
1
ejn d
2 c

1 jn c
1
e

2 jn
c
h
i
1
1
jc n
e
ejc n
2 jn
sin(c n)
, n 6= 0
n
8
< c /,
n=0
sin(c n)
: c
,
c n
n 6= 0
INVERSE
DTFT
(13.16)
where we are multiplying and dividing the expression for x(n) by c when n 6= 0 in order to present
it in a form that brings us closer to sinc functions, as discussed in the next example.
Example 13.8 (Sinc function)
The inverse transform in Example 13.7 is in terms of a common and useful function known as the
sinc function, which is defined as follows:
sinc() =
sin
(13.17)
where is generally a continuous variable. Using the sinc notation, the sequence (13.16) can be
rewritten in the form
c
x(n) =
sinc(wc n)
(13.18)
The function sinc() is such that it attains its maximum value of one at = 0, as can be seen by
applying LHospitals rule:
lim sinc() = lim
cos()
sin
= lim
= 1
0
Moreover, the function sinc() is equal to zero whenever the argument is an integer multiple of ,
namely,
sinc(k) = 0 for any integer k
We can use the sinc notation to define sequences. For example,
the notation sinc
n
4
refers to the sequence
sin(
n
4 )
n
4
Figure 13.13 shows a plot of the function sinc() for values of over the interval [20, 20]. The
same figure shows a plot of the sequence sinc(n/4) for values of n in the range 20 n 20;
this sequence evaluates to zero at multiples of 4.
328
continuoustime plot
1
CHAPTER 13
0.8
sinc()
0.6
0.4
0.2
0
0.2
20
15
10
0
5
discretetime plot
15
10
10
15
20
10
15
20
1
0.8
sinc( n/4)
DISCRETE-TIME
FOURIER
TRANSFORM
0.6
0.4
0.2
0
0.2
20
FIGURE 13.13
0
n
Plots of the function sinc() (top) and the sequence sinc
n
4
(bottom).
13.4 MEAN-SQUARE CONVERGENCE

The result of Example 13.7 raises an important issue. As anticipated, the resulting sequence
x(n), which was found to be
wc /,
n=0
wc sin(c n)
x(n) =
(13.19)
,
n 6= 0
c n
is not absolutely summable. Therefore, the series defining its DTFT in (13.4) does not
converge uniformly and one wonders about the interpretation of the transform pair obtained
in Example 13.7. Is it a valid transform pair?
There are many important sequences x(n) that are not absolutely summable. Strictly
speaking, for such sequences, their DTFTs cannot be defined as the power series
X(ej ) =
x(n)ejn
n=
since these series do not converge uniformly any more. However, there are weaker notions
of convergence that are useful for our purposes and which allow us to define the DTFT
even for sequences that are not absolutely summable. By resorting to these other notions
of convergence, instead of restricting ourselves to uniform convergence, we are able to
extend the definition of the DTFT to a larger class of sequences. This is what is happening
with the result of Example 13.7.
Thus, consider a sequence x(n) that has finite energy, as opposed to being absolutely
summable, namely, x(n) satisfies
n=
|x(n)|2 <
(13.20)
We say that x(n) is square-summable as opposed to absolutely summable. For such squaresummable sequences we will continue to associate a DTFT with them by calling upon
another notion of convergence called mean-square convergence, which is motivated as
follows. Consider again the partial sums
N
X
XN (ej ) =
x(n)ejn
(13.21)
n=N
for integers values N > 0. We say that this sequence of partial sums converges in the
mean-square sense if, and only if, there exists some function X(ej ) such that
lim
1
2

XN (ej ) X(ej )2 d
=0
(13.22)
In other words, the area under the square-error-curve over the interval [, ] should approach zero. This is a weaker notion of convergence than requiring XN (ej ) to converge
point-wise to X(ej ), as was guaranteed earlier by assuming uniform convergence of the
sequence of functions {XN (ej )} see (13.8). Mean-square convergence only guaran
2
tees that the area under the squared-error curve, XN (ej ) X(ej ) , converges to zero;
it does not guarantee that the error itself, XN (ej ) X(ej ), will converge to zero (since
there can exist, for example, some discrete frequency points at which both functions do
not converge to each other and yet the area under the square-error curve is zero).
The mean-square convergence condition (13.22) implies that there exists a large enough
No and a small enough such that
1
2

XN (ej ) X(ej )2 d < ,
for all N > No
(13.23)
This bound does not guarantee that at every the function XN (ej ) will be close to
X(ej ), as was the case with uniform convergence in (13.7); the function XN (ej ) can
be closer to X(ej ) at some values and further away from it at other values. But
the overall effect will be that the area under the square-error curve will be less than .
Mean-square convergence is a weaker notion of convergence than uniform convergence.
Specifically, uniform convergence implies mean-square convergence but the converse is
not necessarily true.
Now, it is known that for square-summable sequences, x(n), the partial sums {XN (ej )}
in (13.21) converge in the mean-square sense to a function X(ej ), which we shall therefore take to represent the DTFT of the sequence. Moreover, we can resort to the same
inverse expression (13.12) to determine which sequence can be associated with the given
DTFT, as was already illustrated in Example 13.7.
Example 13.9 (Finite energy sequence)
Consider again the sequence from Example 13.7, namely,
x(n) =
or, equivalently,
x(n) =
c
sinc(c n)
8
< wc /,
:
wc sin(c n)
,
c n
n=0
n 6= 0
329
SECTION 13.4
MEAN-SQUARE
CONVERGENCE
330
The sequence x(n) is square-summable since
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM
Ex
n=
|x(n)|2
X
sin2 (c n)
wc2
+2
2
2 n2
n=1
wc2
2
+ 2
2
<
X
1
n=1
n2
where we used the fact that the series {1/n2 } converges; actually,
X
1
n=1
n2
2
6
(13.24)
Therefore, the sequence x(n) is square-summable and its DTFT exists in the mean-square sense.
What this means is the following. If we form the partial sums
N
X
XN (ej ) =
n=N
c
sinc(c n) ejn
Then, as N , these partial sums will converge in the mean-square sense to the (low-pass)
function shown in Fig. 13.12 over [, ], namely,
(
1,
0,
X(e ) =

and
lim
1
2
|w| < wc
wc |w|

2

j
j
XN (e ) X(e ) d = 0
Example 13.10 (Gibbs phenomenon)
Let us continue to examine the partial sums of Example 13.9 and how they converge to X(ej ).
Thus note that
XN (ej )
N
X
n=N
c
sinc(c n) ejn
c
1 +
1 +
N
X
n=N,n6=0
sin(c n) jn A
e
c n
N
X
sin(c n) h
n=1
1 + 2
c n
N
X
sin(c n)
n=1
c n
jn
+ e
jn
cos(n)
Hence, XN (ej ) is a real-valued function of . It is also an even function of since XN (ej ) =

XN (ej ).
Figure 13.14 shows plots of XN (ej ) over the interval [, ], for increasing values of
N , superimposed on the rectangular pulse describing X(ej ) and using c = /2. For increasing
values of N , we expect XN (ej ) to provide increasingly better fits for X(ej ). The figure shows
that this is indeed the case. However, since the convergence is not uniform, there are some annoying discrepancies that persist in the form of ripples that occur around the discontinuities at c
regardless of the value of N . Moreover, the peak value of these ripples does not seem to decrease
with increasing N . This is known as Gibbs phenomenon (named after the mathematician J. Gibbs
who explained it around 1899). Gibbs showed that the size of the peak is independent of N (the
maximum value is 1.09, which corresponds to a 9% overshoot). Gibbs also showed that, for every
N , the value of XN (ej ) at each point of discontinuity is the average value of X(ej ) at the point.
In this example, we have two such points at c , with average value equal to 1/2.
The practical implication of these facts is that any truncated approximation of a discontinuous
pulse like X(ej ) will exhibit high frequency ripples near the discontinuities. This suggests that
a sufficiently large N should be used in order to guarantee that the total energy of the ripples is
sufficiently small.
N=1
N=3
0.5
0.5
0
2
N=5
N=10
0.5
0.5
0
2
N=20
N=50
0.5
0.5
0
2
0
(rad/sample)
0
(rad/sample)
FIGURE 13.14 Plots of X(ej ) and XN (ej ) for several values of N . The plots illustrate the
occurrence of Gibbs phenomenon.
Example 13.11 (Convergence in the distributional sense)

Let us determine the sequence x(n) whose DTFT over the interval [, ] is described by
X(ej ) = 2( o ),
[, ]
for some value o [, ]. The function (w) in the above expression denotes the impulse
function of a continuous variable . Recall that in continuous-time, the impulse function (t) is
defined by the following properties:
Z
(t)dt = 1,
f (t)(t to )dt = f (to )
331
SECTION 13.4
MEAN-SQUARE
CONVERGENCE
332
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM
for any function f (t) that is well defined at t = to . The second property is known as the sifting
property; it extracts the value of the function at the location of the impulse, t = to . Using the inverse
transform relation (13.14) we therefore get
Z
x(n)
1
X(ej )ejn d
2
Z
1
2( o )ejn d
2
=
=
ejo n
In other words, we arrive at the transform pair

ejo n
DTFT
2( o ),
[, ]
(13.25)
when o [, ]. Observe here that the sequence

x(n) = ejo n
is neither absolutely summable nor square-summable. Therefore, the corresponding DTFT series,
X(ej ) =
ejo n ejn =
n=
ej(o )n
n=
does not converge uniformly or even in the mean-square sense. In this case, we need a weaker sense
of convergence to justify the transform pair we arrived at. We shall not delve into the technical details
here except to say that the series
ej(o )n
n=
converges in a so-called distributional sense to another series that is given by

2
X
k=
That is,
j(o )n
(w o 2k)
= 2
n=
X
k=
(w o 2k)
(13.26)
Observe that the series on the right is periodic with period 2, and that the X(ej ) that we started
with corresponds to the value of this series over the interval [, ].
Let us now determine the sequence x(n) whose DTFT over the interval [, ] is described by
X(ej ) = [( o ) + ( + o )] ,
[, ]
for some o [, ]. Using the inverse transform relation (13.14) we find that
Z
x(n)
=
=
=
1
X(ej )ejn d
2
i
1 h jo n
e
+ ejo n
2
cos(o n)
333
TABLE 13.1 Some useful DTFT pairs over the interval [, ].

sequence x(n)
DTFT X(ej )
x(n) = (n)
X(ej ) = 1
x(n) =
1,
0,
0nL1
otherwise
X(e ) =
SECTION 13.5
INVERSE
DTFT BY
PARTIAL
FRACTIONS
8
< L,
: e
=0
(L1)
2
x(n) = n u(n), || < 1
X(ej ) =
1
1 ej
x(n) = n u(n 1), || > 1
X(ej ) =
1
1 ej
sin (L/2)
.
,
sin (/2)
otherwise
c
sinc(c n)
x(n) =
X(e ) =
x(n) = ejo n
X(ej ) = 2(w o )
x(n) = cos(o n), o [, ]
X(ej ) = [(w 0 ) + (w + 0 )]
x(n) = sin(o n), o [, ]
X(ej ) = j [(w 0 ) (w + 0 )]
1,
0,
|w| < wc
wc |w|
In other words, we arrive at the transform pair

cos(o n)
DTFT
[( o ) + ( + o )] ,
[, ]
(13.27)
where o [, ]. Likewise,
sin(o n)
DTFT
j [( o ) ( + o )] ,
[, ]
(13.28)
For ease of reference, Table 13.1 lists several transform pairs that were motivated in the
earlier discussions.
13.5 INVERSE DTFT BY PARTIAL FRACTIONS

When the DTFT of a sequence x(n) is a rational function of ej , we can invert it by
partial fractions just like we did for ztransforms. This procedure is best illustrated by
means of an example. Consider the DTFT
X(ej ) =
3 34 ej
1 56 ej + 16 e2j
334
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM
The denominator can be factored as

5
1
1 ej + e2j =
6
6

1 j
1 j
1 e
1 e
2
3
We can then determine constants A and B to satisfy the partial fractions expansion
3 34 ej
5 j
+ 16 e2j
6e
A
B
+
1 12 ej
1 31 ej
By comparing coefficients of powers of ej in the numerators on both sides of the above

equality we find that A = 1 and B = 2. Therefore,
3 34 ej
5 j
+ 16 e2j
6e
1
2
+
1 21 ej
1 13 ej
By inverse transforming we obtain

x(n) =
n
n
1
1
u(n)
+2
2
3
Alternatively, we can replace ej by z in the expression for X(ej ) and write first
X(z) =
3 34 z 1
1 56 z 1 + 61 z 2
We then choose a ROC for this z-transform that includes |z| = 1. In this case, the appropriate ROC should be |z| > 1/2 since the poles are at z = 1/2 and z = 1/3. We
subsequently invert X(z) by partial fractions by noting that
3 43 z 1
z
=
1 65 z 1 + 16 z 2
z
which leads to the same sequence, x(n), as above.
1
2
2z
z 31
335
13.6 APPLICATIONS
SECTION 13.7
APPLICATIONS
TO BE ADDED
Practice Questions:
1.
2.
13.7 PROBLEMS
Problem 13.1 Find the DTFT of
x(n) = j (n 1) j (n + 1)
Is X(ej ) real-valued? Plot its magnitude and phase over [, ].
x(n) = j (n 4) + (n 2) + (n + 2) j (n + 4)
Is X(ej ) real-valued? Plot its magnitude and phase over [, ].
Problem 13.3 Find the DTFT of the sequence x(n) = |n| where || < 1.
Problem 13.4 Find the DTFT of the sequence x(n) = |2n| where || < 1.
8
>
< 2,
x(n) =
1,
>
: 0,
n=0
1n5
otherwise
Plot its magnitude and phase over [, ].

8
>
1,
>
>
<
2,
x(n) =
>
1,
>
>
:
0,
n=0
n=1
2n6
otherwise

x(n) =
8
>
<
1,
1,
>
: 0,
n = 0, 1, 2
n = 3, 4, 5
otherwise

x(n) =
8
>
<
1,
1,
>
:
0,
n = 0, 2, 4
n = 1, 3, 5
otherwise
336
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM

Problem 13.9 Find the DTFTs of the following sequences:
(a) x(n) =
(b) x(n) =
(c) x(n) =

1 n1
u(n + 1).
2

2n1
1
u(n 1).
3

1 n2
4
u(n
1).
In each case, find expressions for the magnitude and phase of the DTFT.
(a) x(n) =
(b) x(n) =
(c) x(n) =

1 3n+2
u(n 3).
2

n+1
13
u(n + 1).

1 n2
u(2n).
4
(a) x(n) = cos
(b) x(n) = cos
(c) x(n) = cos
(d) x(n) = cos
n sin 3 n .
3
n 2
.
3
3
n + 2j sin
3

n cos 6 n .
3
n
6
In each case, plot the magnitude and phase of the DTFT.

(a) x(n) = cos
(b) x(n) = sin
(c) x(n) = sin
(d) x(n) = sin
(n 1) sin 4 (n
4
(n + 2) .
4

n + cos 3 n .
4

n sin 3 n .
4
1) .
In each case, plot the magnitude and phase of the DTFT.

(a) x(n) =
(b) x(n) =
(c) x(n) =

1 n3
u(n)
2

n1
1
u(n)
4

cos 4 n +

1 n
3
+ cos
u(n 1).
n
3
(n 1) + (n + 1).
(a) x(n) =
(b) x(n) =
(c) x(n) =

n+1
1 2n+2
u(n 1) + 16
u(2n 1).
4

1 n3
u(n + 2) + sin 3 (n 1) .
3

cos 4 (n + 1) + j (n 2) + j (n +
2).
Problem 13.15 Find the DTFTs of the sequences whose ztransforms are given below:
z
(a) X(z) =
, |z| > 1/2.
z 1/2
z 3
, |z| > 1/2.
z 1/2
z
1
, 1/2 < |z| < 3.

(c) X(z) =
z 1/2 z 3
(b) X(z) =
1
1
+
, |z| > 1/2.
z 1/2
z 1/3
Indicate in each case whether the DTFT is uniformly convergent.
(d) X(z) =
Problem 13.16 Find the DTFTs of the sequences whose ztransforms are given below:
z2
, |z| > 1/2.
z + 1/2
1
, |z| > 1/2.
(b) X(z) =
z(z 1/2)
(a) X(z) =
(c) X(z) =
z 1/3
1
, 1/2 < |z| < 2.

z 1/2 z 2
z 1
z2
+
, |z| > 1/3.
z 1/4
z + 1/3
Indicate in each case whether the DTFT is uniformly convergent.
(d) X(z) =
Problem 13.17 Verify whether each of the following sequences is absolutely summable or square
summable or both:
(a) x(n) =
(b) x(n) =
(c) x(n) =
(d) x(n) =
(e) x(n) =
(f) x(n) =

1 n
u(n).
2

n2
1
u(n
2
1 |n|
.
2
1
u(n).
n+1
sin
1).
n
3
.
cos 6 n
u(n).
n+1
Problem 13.18 Verify whether each of the following sequences is absolutely summable or square
summable or both:
(a) x(n) =
(b) x(n) =
(c) x(n) =

1 2n1
u(n).
3

1 n+1
u(2n
4
2
1 n
2
1).
1
(d) x(n) = 2
u(n).
n +1

sin n
(e) x(n) = 2 3 .
n +1

cos 6 n
u(n).
(f) x(n) =
n+1
Problem 13.19 Consider the DTFT function
(
j
H(e ) =
ej 2 ,
0,
|| < c
c ||
Can H(ej ) be the DTFT of the impulse response sequence of a stable LTI system?
Problem 13.20 Consider the DTFT function
(
j
H(e ) =
ej 3 ,
0,
|| > c
c ||
Can H(ej ) be the DTFT of the impulse response sequence of a stable LTI system?
337
SECTION 13.7
PROBLEMS
338
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM
(a) x(n) =
(b) x(n) =
sin
n
3
n
sin
.
(n
3
1)
.
n1

(a) x(n) =
(b) x(n) =
sin
(n
4
sin
n
6
2)
n+1
)
3
Problem 13.23 Invert the following DTFTs:

(a) X(ej ) = cos( 2 ).
(b) X(ej ) =
1
.
1 12 ej
(c) X(ej ) = cos( 3 ) sin( 3 ).
(d) X(ej ) = cos2 ( 6 ).
(e) X(ej ) = sin( 3 ) cos2 ( 4 ).

(a) X(ej ) = cos2 ( 2 ).
(b) X(ej ) =
1
.
1 + 13 ej
(c) X(ej ) = sin2 ( 3 ).

(d) X(ej ) = cos2 ( 6 ) + sin2 ( 4 ).
(e) X(ej ) = sin2 ( 3 ) cos( 4 ).
(a) X(ej ) =
(b) X(ej ) =
(c) X(ej ) =
8
>
< 1,
1,
>
: 0,
/4 /2
/2 < /4
otherwise
1,
>
:
0,
3/4
3/4
otherwise
8
>
< 1,
8
j 1
>
< 2e 2 ,
1
ej 3 ,
>
:
0,
/3 2/3
2/3 /3
otherwise
(a) X(ej ) =
8
>
1,
>
>
>
>
< 2,
1,
>
>
>
2,
>
>
:
0,
(b) X(ej ) =
8
>
< 1,
2,
>
:
0,
/4 /2
/2 3/4
/2 < /4
3/4 /2
otherwise
3/4
3/4
otherwise
(c) X(ej ) =
8
j 1
>
< 2e 2 ,
>
:
j 1
4
e
0,
339
0 /4
/2 /4
otherwise
SECTION 13.7
PROBLEMS
Problem 13.27 Use partial fractions to recover the sequence x(n) for the given DTFT:
X(ej ) =
1+
7 j
12
e
1 j
1 2j
e
12
e
12
Problem 13.28 Use partial fractions to recover the sequence x(n) for the given DTFT:
X(ej ) =
ej e2j
1 + 32 ej + 19 e2j
Problem 13.29 Consider the step-sequence x(n) = u(n). It is neither absolutely summable nor
square-summable.
(a) Determine a closed-form expression for the finite sum
N
X
XN (ej ) =
x(n)ejn
n=N
(b) Plot the magnitude and phase of XN (ej ) for increasing values of N , say N = 1, 2, 5, 10, 20, 70, 100.
Problem 13.30 Argue that the DTFT of the step sequence, x(n) = u(n), is given by
U (ej ) =
1
+
1 ej
X
k=
(w 2k)
Problem 13.31 Argue that the DTFT of the sequence x(n) = 1 for all n is given by
X(ej ) =
X
k=
2 ( 2k)
Problem 13.32 Show that an absolutely summable sequence always has finite energy.
Problem 13.33 Figure 13.15 shows the magnitude DTFT of a certain sequence x(n). Determine
the sequence x(n) for each of the phase plots shown in Fig. 13.16?
|X(ej )|
1
1/2
FIGURE 13.15
Plot of the magnitude DTFT of a sequence for Probs. 13.33 and 13.34.
340
X(ej )
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM
X(ej )
X(ej )
/2
FIGURE 13.16 Three phase plots for Prob. 13.33.
Problem 13.34 Consider the same magnitude plot of Fig. 13.15. Determine the sequence x(n) for
each of the phase plots shown in Fig. 13.17?
Problem 13.35 The DTFTs of two sequences {x(n), y(n)} are shown in Fig. 13.18. Determine
the sequences. Find also their energies.
Problem 13.36 The DTFTs of a sequence x(n) is shown in Fig. 13.19. Determine the energy of
the sequence.
341
SECTION 13.7
PROBLEMS
X(ej )
X(ej )
X(ej )
FIGURE 13.17 Three phase plots for Prob. 13.34.
X(ej )
2
4 8
(rad/sample)
(rad/sample)
Y (ej )
2
FIGURE 13.18 DTFT plot for Prob. 13.35.
342
CHAPTER 13
DISCRETE-TIME
FOURIER
TRANSFORM
|X(ej )|
1
1/2
X(ej )
/2
FIGURE 13.19
Plots of the magnitude and phase of the DTFT of the sequence for Prob. 13.36.
CHAPTER
14
Properties of the DTFT
he Discrete-Time Fourier Transform (DTFT) has several useful properties, which can
facilitate the evaluation of the DTFT without having to resort each time to evaluating the
defining series. This chapter establishes some of these properties and provides illustrative
examples.
14.1 PERIODICITY OF THE DTFT

We start from the definition of the DTFT of a sequence x(n), namely,
X(ej ) =
x(n)ejn
(14.1)
n=
and recall that the complex exponential sequence is periodic with period 2, i.e.,
ej = ej(+2)
It follows that X(ej ) is also periodic with period 2,
X(ej ) = X ej(+2)
(14.2)
It is for this reason that we have been displaying the magnitude and phase plots of the
DTFT over 2-wide intervals and, often, over the interval [, ].
Example 14.1 (Two equivalent plots)

We shall limit our representation of the DTFT of a sequence either to the interval [, ] or to the
interval [0, 2]. Fig. 14.1 illustrates several periods of a DTFT and the corresponding representations
over the intervals [, ] and [0, 2]. The two representations shown in the middle and bottom plots
over the 2long intervals are equivalent and they represent the same sequence x(n).
343
Copyright
344
CHAPTER 14
X(ej ) (multiple periods)
PROPERTIES
OF THE
DTFT
1
2
(rad/sample)
X(ej ) over [, ]
(rad/sample)
X(ej ) over [0, 2]
1
c
(rad/sample)
2 c
FIGURE 14.1
Multiple periods of a DTFT X(ej ) (top plot) followed by the equivalent
representations of the same DTFT over the periods [, ] (middle plot) and [0, 2] (bottom plot).
Example 14.2 (Finite duration sequence)

Let us reconsider the sequence of Example 13.1,
x(n) = 0.5(n + 1) + (n) + 0.5(n 1)
whose DTFT we already found to be
X(ej ) = 1 + cos()
Figures 14.2 and 14.3 display the sequence x(n) and its DTFT over the period [, ]. Since
X(ej ) is periodic of period 2, we illustrate in Fig. 14.4 several periods of X(ej ). The plot in
Fig. 14.3 shows only one period over the interval [, ]. However, we can as well display X(ej )
over another 2long interval, say over [0, 2]. This situation is illustrated in Fig. 14.5. The plots
in Figs. 14.3 and 14.5 represent the same DTFT because if these plots are repeated periodically, then
they would result in the same DTFTs as in Fig. 14.4.
14.2 USEFUL PROPERTIES

The DTFT shares several properties with the bilateral z-transform. A summary of these
properties is given in Table 14.1 further ahead. The first two lines of the table start from
345
SECTION 14.2
x(n)
USEFUL
PROPERTIES
1
0.5
2
1.8
1.6
1.4
X(e )
1.2
1
0.8
0.6
0.4
0.2
0
FIGURE 14.3
0
1
(radians/sample)
A plot of the DTFT X(ej ) = 1 + cos() over the interval [, ].
two generic sequences x(n) and y(n) and the subsequent lines provide the DFTT of combinations and transformations of these sequences.
14.2.1 Linearity
Consider, for instance, the third line of the table. It states that the DTFT of a linear combination of two sequences is given by the same linear combination of their DTFTs, namely,
ax(n) + by(n) aX(ej ) + bY (ej )

for any two scalars a and b.
(14.3)
346
X(e ) over the interval [4, 4]

2
CHAPTER 14
PROPERTIES
OF THE
DTFT
1.8
1.6
1.4
X(e )
1.2
1
0.8
0.6
0.4
0.2
10
0
(radians/sample)
10
FIGURE 14.4 A plot showing several periods of the DTFT X(ej ) = 1 + cos() over the
interval [4, 4].
2
1.8
1.6
1.4
X(e )
1.2
1
0.8
0.6
0.4
0.2
0
3
4
(radians/sample)
FIGURE 14.5 A plot of the DTFT X(ej ) = 1 + cos() over the interval [0, 2]. The
plots in Figs. 14.3 and 14.5 represent the same DTFT.

W (ej )
w(n)ejn
n=
[ax(n) + by(n)]ejn
n=
x(n)e
jn
n=
+ b
y(n)e
jn
n=
aX(ej ) + bY (ej )
347
TABLE 14.1 Several properties of the DTFT.
SECTION 14.2
sequence
DTFT
1.
x(n)
X(ej )
2.
y(n)
Y (ej )
3.
ax(n) + by(n)
aX(ej ) + bY (ej )
linearity
4.
x(n n0 )
ejn0 X(ej )
time-shifts
5.
ejo n x(n)
X(ej(o ) )
frequency shifts
6.
cos(o n)x(n)
1
X
2
sin(o n)x(n)
1
X
2j
7.
x(n)
X(ej )
8.
nx(n)
9.
x(n) y(n)
X(ej )Y (ej )
convolution
10.
x(n)y(n)
X(ej ) Y (ej )
multiplication
11.
x (n)
12.
property
ej(o ) + 21 X ej(+o )
ej(o )
1
X
2j
ej(+o )
1
2
n=
R
2
linear modulation
X(ej )]
x(n)y (n)
modulation
time reversal
dX(ej )
dw
conjugation
X(ej ) Y (ej )
Parsevals relation
Example 14.3 (Illustrating the linearity property)

Consider the sequence x(n) that is shown in the top plot of Fig. 14.6 and let us determine its DTFT.
We note that the sequence x(n) can be regarded as the sum of the two rectangular pulses shown in
the middle and bottom plots of the same figure,
x(n) = x1 (n) + x2 (n)
where x1 (n) has duration L = 2:
(
x1 (n) =
1,
0
0, n 1
otherwise
1,
0,
0n3
otherwise
and x2 (n) has duration L = 4:

(
x2 (n) =
USEFUL
PROPERTIES
348
x(n)
CHAPTER 14
PROPERTIES
OF THE
DTFT
2
1
1
x1 (n)
1
1
x2 (n)
1
1
FIGURE 14.6 The sequence x(n) in the top plot can be expressed as the sum of the two
rectangular pulses in the middle and bottom plots.
We already know from the discussion in Example 13.5 that the DTFTs of the rectangular pulses
x1 (n) and x2 (n) are given by
X1 (ej ) =
and
j
X2 (e ) =
8
< 2,
: e
when = 0
sin ()
,
.
sin 2
j
2
8
< 4,
: e
otherwise
when = 0
sin (2)
,
.
sin 2
j 3
2
otherwise
Using the trigonometric relation

sin(2) = 2 sin() cos()
we conclude that
(
j
X(e ) =
6,
2ej 2 cos
1 + 2ej cos()
=0
otherwise
Obviously, we could have also arrived at this result directly from the definition (13.4), which in this
case gives
X(ej ) = 2 + 2ej + ej2 + ej3
We instead opted to illustrate the linearity property (14.3). The magnitude and phase plots of the
resulting X(ej ) are displayed in Fig. 14.7 over [, ].
349
phase plot
1.5
0.5
X(e )
|X(ej)|
magnitude plot
3
2
USEFUL
PROPERTIES
0
0.5
1
1
3 2
FIGURE 14.7
from Fig. 14.6.
SECTION 14.2
1 0
1 2
(rad/sample)
1.5
3 2
1
0
1
2
(rad/sample)
The magnitude (top) and phase (bottom) plots of the DTFT of the sequence x(n)
14.2.2 Time Shifts

Consider now the fourth line in Table 14.1. It establishes the transform property
x(n no )
ejno X(ej )
(14.4)
In other words, if the original sequence x(n) is shifted in time by an amount no (where
no can be positive or negative), then the phase of the corresponding DTFT is modified by
the factor ejno . Observe that the magnitude of the DTFT is not modified since both
functions ejno X(ej ) and X(ej ) have the same magnitude for every . We therefore
say that shifting in the time-domain corresponds to phase change in the frequency domain
and vice-versa.
Proof: Let w(n) = x(n no ). Then
W (ej )
w(n)ejn
n=
n=
x(n no )ejn
x(k)ej(k+no ) ,
using k = n no
k=
=
=
jno
x(k)e
jk
k=
ejno X(ej )
Example 14.4 (Illustrating the time-shift property)

x(n) =

1
sinc
n
4
4
350
We already know from Table 13.1 that the DTFT of x(n) is given by
(
CHAPTER 14
PROPERTIES
OF THE
DTFT
1,
0,
X(ej ) =
|| 4
otherwise
The DTFT of x(n) is real-valued and it is shown in the top plot of Fig. 14.8. Now assume that we
shift x(n) by one unit sample to the right and consider the sequence
y(n) = x(n 1)
According to (14.4), the DTFT of y(n) is related to the DTFT of x(n) as follows:
Y (ej ) = ej X(ej )
where we are using no = 1. We therefore find that Y (ej ) has the same magnitude plot as X(ej ),
while the phase of Y (ej ) varies linearly according to
Y (ej ) = ,
|| /4
The magnitude and phase plots of Y (ej ) are shown in the bottom plots of Fig. 14.8.
X(ej )
1
(rad/sample)
|Y (ej )| = |X(ej )|
1
(rad/sample)
Y (ej ) =
/4
/4
(rad/sample)
FIGURE 14.8 Illustration of the time-shift property for the data in Example 14.4. The top plot
shows the DTFT of the sequence x(n) in the example, and the bottom plots show the magnitude and
phase plots of the DTFT of the sequence y(n) = x(n 1).
351
SECTION 14.2
14.2.3 Frequency Shifts
USEFUL
PROPERTIES
Consider now the fifth line in Table 14.1. It establishes the transform property
ejo n x(n)
X ej(o )
(14.5)
In other words, if the phase of the original sequence x(n) is modified by adding a linear
component to it, in the form of o n, then the corresponding DTFT is obtained by shifting
the DTFT of the original sequence by o . We therefore say that phase change in the timedomain corresponds to shifting in the frequency domain and vice-versa. This property is
the dual of the time-shift property.
Proof: Let w(n) = ejo n x(n). Then
W (ej )
w(n)ejn
n=
ejo n x(n)ejn
n=
x(n)ej(o )n
n=
X(ej(o ) )
Example 14.5 (Illustrating the frequency shift property)

Consider again the sequence
x(n) =

1
sinc
n
4
4
where
1,
0,
X(e ) =
|| /4
otherwise
The DTFT of x(n) is real-valued and is shown in the top plot of Fig. 14.9. Now consider the sequence
y1 (n)
ej 2 n x(n) = (j)n x(n)
According to (14.5), the DTFT of y1 (n) is related to the DTFT of x(n) as follows:

Y1 (ej ) = X ej ( 2 )
where we are using o = /2. We therefore find that Y1 (ej ) has the same plot as X(ej ) but is
shifted to the right and centered at the point = /2. This situation is illustrated in the center plot
of Fig. 14.9.
Consider further the sequence
y2 (n)
ejn x(n) = (1)n x(n)

Y2 (ej ) = X ej()
352
CHAPTER 14
PROPERTIES
OF THE
DTFT
where we are now using o = . We therefore find that Y2 (ej ) has the same plot as X(ej ) but is
shifted to the right by an amount that is equal to . However, since we are limiting the display of the
DTFT to the range [, ], then the portion of Y2 (ej ) that overflows beyond , appears on
the left-hand-side between [, 3
]. This situation is illustrated in the bottom plot of Fig. 14.9.
4
X(ej )
1
(rad/sample)
(rad/sample)
(rad/sample)
Y1 (ej )
1
3
4
Y2 (ej )
1
3
4
3
4
FIGURE 14.9 Illustration of the frequency-shift property for the data in Example 14.5. The top
plot shows the DTFT of the sequence x(n) in the example, and the bottom plots show the DTFTs of
y1 (n) = ej 2 n x(n) and y2 (n) = ejn x(n).
14.2.4 Modulation
Consider now the sixth line in Table 14.1. It establishes the transform property
cos(o n)x(n)
1
2
X ej(o )
+ X ej(+o )
(14.6)
In other words, if the sequence x(n) is modulated by a cosine sequence, then the DTFT
is scaled by 1/2 and shifted left and right to the locations o , which are the locations
dictated by the angular frequency of the sinusoidal sequence.
353
Proof: Let w(n) = cos(o n)x(n). Using Eulers relation (3.11) we have
SECTION 14.2
w(n) =
1 jo n
1
e
x(n) + ejo n x(n)
2
2
USEFUL
PROPERTIES
Invoking the linearity and frequency-shift properties (14.3) and (14.5) we conclude that
i
1h
X(ej(o ) ) + X(ej(+o ) )
2
W (ej ) =
Likewise, it holds that

sin(o n)x(n)

1
X ej(o ) X ej(+o )
2j
(14.7)
where the proof now requires that we employ the alternative form (3.12) for Eulers relation.
Example 14.6 (Illustrating the modulation property)
Consider again the sequence
x(n) =

1
sinc
n
4
4
where
j
X(e ) =
1,
0,
|| /4
otherwise
The DTFT of x(n) is real-valued and is shown in the top plot of Fig. 14.10. Now consider the
sequence
y1 (n)
cos

n x(n)
Y1 (ej ) =
1
1 j ( 2 )
+ X ej (+ 2 )
X e
2
2
where we are using o = /2. We therefore find that Y1 (ej ) is obtained from X(ej ) by shifting
the plot of the latter to the left and to the right by /2 and by scaling the magnitude by 1/2. This
situation is illustrated in the center plot of Fig. 14.10.
Consider further the sequence
y2 (n)
cos

n x(n)
Y2 (ej ) =
1 j ( 4 )
1
X e
+ X ej (+ 4 )
2
2
where we are now using o = /4. We therefore find that Y2 (ej ) is obtained from X(ej ) by
shifting the plot of the latter to the left and to the right by /4 and by scaling the magnitude by 1/2.
This situation leads to the bottom plot in Fig. 14.10.
Let us now consider the sequence
y3 (n)
cos

n x(n 1)
We already know from Example 14.4 how the DTFT of x(n 1) relates to the DTFT of x(n); this
is shown in the middle and bottom plots of Fig. 14.8. The sequence y3 (n) is obtained by multiplying
354
X(ej )
CHAPTER 14
PROPERTIES
OF THE
DTFT
(rad/sample)
(rad/sample)
(rad/sample)
Y1 (ej )
1/2
3
4
3
4
Y2 (ej )
1/2
FIGURE 14.10 Illustration of the modulation property for the data in Example 14.6. The top plot
shows the DTFT of the sequence x(n) in the example, and the bottom plots show the plots of the
DTFTs of y1 (n) = cos( 2 n)x(n) and y2 (n) = cos( 4 n)x(n).
x(n 1) by cos( 2 n). Therefore, the magnitude and phase plots of Y3 (ej ) are obtained by shifting
the magnitude and phase plots of the DTFT of x(n 1) to the right and to the left by /2, and by
scaling the magnitude plot by 1/2. This construction leads to the plots shown in Fig. 14.11.
14.2.5 Time Reversal

Consider the seventh line in Table 14.1. It establishes the transform property
x(n)
X(ej )
(14.8)
In other words, if the original sequence x(n) is reversed in time (i.e., flipped around the
vertical axis), then the corresponding DTFT is reversed in frequency.
355
SECTION 14.2
|Y3 (ej )|
USEFUL
PROPERTIES
1/2
3
4
3
4
(rad/sample)
(rad/sample)
Y3 (ej )
/4
3/4
/4
3
4
/4
FIGURE 14.11 Illustration of the modulation property for the data in Example 14.6. The figure
shows the magnitude and phase plots of the DTFT of the sequence y3 (n) = cos( 2 n)x(n 1).
Proof: Let w(n) = x(n). Then

W (ej )
w(n)ejn
n=
x(n)ejn
n=
x(k)ejk ,
k=
using k = n
X(ej )
Example 14.7 (Illustrating the time-reversal property)

x(n) =

1
sinc
n
4
4
where
1,
0,
X(ej ) =
|| /4
otherwise
The DTFT of x(n) is real-valued and is shown in the top plot of Fig. 14.12. We illustrated in the
middle and bottom plots of Fig. 14.8 the magnitude and phase plots of the delayed sequence x(n1).
y1 (n)
ej 8 n x(n 1)
356
CHAPTER 14
PROPERTIES
OF THE
DTFT
According to (14.5), the magnitude and phase plots of the DTFT of y1 (n) are obtained by shifting
the magnitude and phase plots of x(n 1) by /8 to the right. This situation is illustrated in the
bottom plots of Fig. 14.12.
X(ej )
1
(rad/sample)
(rad/sample)
|Y1 (ej )|
1
3
8
Y1 (ej ) = ( 8 )
/4
(rad/sample)
/4
FIGURE 14.12 Illustration of the frequency shift property for the data in xample 14.7. The top
plot shows the DTFT of the sequence x(n) in the example, and the bottom plots show the plots of
the DTFTs of y1 (n) = ej 8 n x(n 1).
Consider now the time-reversed version of y1 (n), namely,
y2 (n) = y1 (n) = ej 8 n x(n 1)

According to (14.8), the DTFT of y2 (n) is related to the DTFT of y1 (n) as follows:
Y2 (ej ) = Y1 (ej )
In other words, the magnitude and phase plots of Y1 (ej ) from Fig. 14.12 should be flipped around
the vertical axis. This step leads to the plots shown in Fig. 14.13.
357
SECTION 14.2
|Y2 (ej )|
USEFUL
PROPERTIES
3
8
(rad/sample)
Y2 (ej ) = +
/4
/4
(rad/sample)
FIGURE 14.13 Illustration of the time-reversal property for the data in Example 14.7. The figure
shows the magnitude and phase plots of the sequence y2 (n) = ej 8 n x(n 1).

[Linear Modulation] Consider the eighth line in Table 14.1. It establishes the transform
property
nx(n)
dX(ej )
d
(14.9)
In other words, if the original sequence x(n) is modulated by the linear sequence n, then
the corresponding DTFT is replaced by its derivative with respect to and multiplied by
j.
Proof: Let w(n) = nx(n) and recall first the definition of X(ej ):
X(ej ) =
x(n)ejn
n=
Differentiating with respect to we have

dX(ej )
d
X
n=
X
n=
x(n)
dejn
d
jnx(n) ejn
358
so that
CHAPTER 14
PROPERTIES
OF THE
DTFT
dX(ej )
=
d
nx(n)ejn = W (ej )
n=
Example 14.8 (Illustrating the linear modulation property)

1
sinc
n
4
4
We already know from Table 13.1 that the DTFT of x(n) is given by the rectangular pulse:
x(n) =
(
1,
0,
X(e ) =
|| /4
otherwise

y(n) = nx(n)
According to (14.9), the DTFT of y(n) is obtained by differentiating the DTFT of x(n) with respect
to and multiplying the result by j. This calculation leads to the expression

Y (ej ) = j +

j
4
4
In other words, Y (ej ) consists of two impulses located at = /4. This result is consistent with
the DTFT of a sinusoidal sequence from Table 13.1 where we have the transform pair
sin

j +

j
4
4
Indeed, observe that the expression for y(n) reduces to a sinusoidal sequence since
y(n)
=
=
=
=
nx(n)

1
n sinc
n
4
4
1 sin 4 n
n
4
n
4

1
sin
n
Fig. 14.14 illustrates the DTFT of y(n).

Consider the ninth line in Table 14.1. It establishes the transform property
x(n) y(n)
X(ej )Y (ej )
(14.10)
In other words, convolution in the time domain amounts to multiplication in the transform
domain.
Proof: Let
w(n)
=
=
x(n) y(n)
X
k=
x(k)y(n k)
359
SECTION 14.2
USEFUL
PROPERTIES
|Y (ej )|
1
(rad/sample)
(rad/sample)
Y (ej )
/2
/2
FIGURE 14.14 Illustration of the linear modulation property for the data in Example 14.8. The
top figure shows the magnitude plot of Y (ej ) and the bottom plot shows the corresponding phase
plot using the sequence y(n) = nx(n).
Then
W (ej )
w(n)ejn
n=
n= k=
k= n=
x(k)y(n k)ejn
n=
x(k)e
! 0
x(k)e
jk
jn
n=
k=
y(n k)e
jk
k=
x(k)
k=
x(k)y(n k)ejn
y(n k)e
jn jk
y(n )e
jn A
n =
using n = n k
X(ej )Y (ej )
Example 14.9 (Illustrating the linear convolution property)

Consider the two sequences
x1 (n) =

1
sinc
n ,
4
4
x2 (n) =

1 j 8 n
e
sinc
n
4
4
360
CHAPTER 14
The corresponding DTFTs are real-valued and are shown in the two top plots of Fig. 14.15. We are
interested in evaluating the linear convolution of x1 (n) and x2 (n), namely, the sequence
PROPERTIES
OF THE
DTFT
y(n) = x1 (n) x2 (n)

According to the property (14.10), the DTFT of y(n) is obtained by multiplying the DTFTs of x1 (n)
and x2 (n). Doing so results in the third plot in Fig. 14.15; the plot shows a rectangular pulse that
assumes the value one over the interval [/8, /4]. In order to recover the sequence y(n), we
can proceed, for example, by using the inversion formula
y(n)
=
=
=
=
=
=
=
1
2
1
2
1
2
1
2
1
2
1
2
Y (ej )ejn d
/4
ejn d
/8
1
jn
1
jn
1
jn
1
jn
/4

ejn
h
/8
n
j
4
ej 8 n
i
3
ej 16 n ej 16 n ej 16 n

3
n
16
ej 16 n 2j sin
3
ej 16 n sinc
16
3
n
16
Alternatively, and more directly, we can use the properties of the DTFT to receover y(n) from its
j
DTFT, Y (ej ). To do so, we observe that if we
Y (e
left by /16 we obtain a perfectly
shift
) to thej
3 3
centered rectangular pulse over the interval 16 , 16 . Let Y2 (e ) denote this new DTFT see
bottom plot of Fig. 14.15. Then

Y (ej ) = Y2 ej( 16 )
Now invoking the frequency-shift property (14.5) we conclude that
y(n) = ej 16 n y2 (n)
And we know from Table 13.1 that
y2 (n) =
so that
y(n) =
3
sinc
16
3
n
16
3
ej 16 n sinc
16
3
n
16
as expected.

Consider the tenth line in Table 14.1. It states that the DTFT of the product of two sequences is given by the expression below
x(n)y(n)
1
2
X(ej )Y (ej() )d
(14.11)
361
j
X1 (e )
SECTION 14.2
USEFUL
PROPERTIES
(rad/sample)
(rad/sample)
(rad/sample)
(rad/sample)
X2 (ej )
1
3
8
Y (ej )
1
Y2 (ej )
1
3
16
3
16
FIGURE 14.15 Illustration of the linear convolution property for the data in Example 14.9. The
top two plots show the DTFTs of the sequences x1 (n) and x2 (n). The last two plots show the DTFTs
of y(n) and y2 (n).
where the integration on the right-hand side is carried over an interval of width 2, say,
over [, ] or [0, 2].
Proof: Let
R(ej ) =
1
2
X(ej )Y (ej() )d
2
Using the inverse DTFT expression (13.14) we can recover the sequence r(n) as follows:
362
CHAPTER 14
PROPERTIES
OF THE
DTFT
r(n)
1
2
=

=

=

=
=
1
2
1
2
1
2
R(ej )ejn d
2
2
Z

j
X(e )Y (e
2 Z
j()
Z
X(ej )ejn
2 Z
Y (ej() )ej()n d d
2
Z
X(ej )ejn
Z
)d ejn d
Y (ej )ej n d d,
2

1
X(ej )ejn d
2 2
x(n)y(n)
1
2
(using = )
Y (ej )ej n d
2
as desired.
Evaluation of Continuous-Time Convolutions

Before proceeding in our discussions, it is important to comment on the nature of the
integral expression that appears on the right-hand side of (14.11); it has the form of a
convolution integral over the continuous variable with two distinctive features:
(a) First, the functions that are being convolved, namely, X(ej ) and Y (ej ), are periodic functions of with period 2.
(b) Second, the integration is being performed over an interval of length 2, say over
[, ] or [0, 2].
We refer to the convolution integral in (14.11) as circular convolution in order to to distinguish it from the operation of linear convolution, which is explained below. We denote the
circular convolution operation by the symbol and write
X(ej ) Y (ej ) =
1
2
X(ej )Y (ej() )d
(14.12)
Linear convolution in continuous-time is instead defined as follows. Consider two arbitrary (not necessarily periodic) signals, x(t) and h(t). Their linear convolution is the
function y(t) that results from the following operation:
y(t) = x(t) h(t) =
x( )h(t )d
(linear convolution)
(14.13)
which involves evaluating the area under the curve x( )h(t ) for every t. It is useful
to observe the analogy with the definition (5.6) of the convolution sum of two sequences,
x(n) and h(n), namely,
y(n) = x(n) h(n) =
k=
x(k)h(n k)
(linear convolution)
As was the case with convolution sums in Sec. 6.2, it is possible to interpret the linear
convolution integral (14.13) graphically. Specifically, the signal y(t) can be evaluated
graphically as follows (the steps are illustrated in Fig. 14.16):
(a) First, we plot the signals h( ) and x( ) . Note that we are denoting the
independent variable by . Therefore, the horizontal axis will be the axis.
[(b) Then we plot h( ). In other words, we flip the signal h( ) around the vertical axis
to obtain h( ).
(c) We subsequently compute the area under the product curve x( )h( ) to obtain
y(0):
Z
y(0) =
x( )h( )d
(d) Next, we shift h( ) by t units of time in order to obtain h(t ). We then compute
the area under the curve x( )h(t ) to find y(t):
Z
y(t) =
x( )h(t )d
and so on.
x( )
h( )
x( )
x( )
h( )
h(t )
t + o
FIGURE 14.16 Graphical evaluation of the linear convolution of two signals x(t) and h(t). The
top row shows the signals x( )) and h( ) . The middle row shows x( ) again and the last row
shows h( ) on the left and its shifted version, h(t ) on the right. The area that is common to
x( ) and h( ) on the left, as well as the area that is common to x( ) and h(t ) on the right, are
marked with dashed lines.
Now, note that the expression on the right-hand side of (14.11) cannot be interpreted as
the linear convolution of the signals X(ej ) and Y (ej ); this is because the integration
363
SECTION 14.2
USEFUL
PROPERTIES
364
CHAPTER 14
PROPERTIES
OF THE
DTFT
is limited to an interval of width 2 and the functions X(ej ) and Y (ej ) are both periodic of period 2. The same graphical construction described above to evaluate linear
convolutions can be used to evaluate circular convolutions as well. To do so, we simply
keep in mind during the shift operations that the functions X(ej ) and Y (ej ) are periodic and, therefore, when shifts occur, the entire functions are shifted (including all their
repeated periods) to the left or to the right. Subsequently, the area under the product curve
X(ej )Y (ej() ) is evaluated but only over a single period (as opposed to over the entire
interval (, )).
Example 14.10 (Modulation via convolution)

We can use the circular convolution property to re-derive the modulation result (14.6). Thus, recall
that the DTFT of cos o n over [, ] is given by
cos o n
( o ) + ( + o ).
Therefore,
Z
cos(o n)x(n)
1
[( o ) + ( + o )] X(ej() )d
2 2
1
1
X(ej(o ) ) + X(ej(+o ) )
2
2
Example 14.11 (Illustrating multiplication in the time domain)

x(n) =

1
sinc
n
4
4
and
y(n) =

1
sinc
n
2
2
The corresponding DTFTs are real-valued and are shown in the two top plots of Fig. 14.17. We are
interested in evaluating the DTFT of the product sequence, r(n) = x(n)y(n). According to (14.11),
the DTFT of r(n) is obtained by computing the circular convolution
R(ej ) =
1
2
X(ej )Y (ej() )d
2
We are going to evaluate this circular convolution in two ways: analytically by using the integral
expression and graphically.
Analytical method. Using the fact that X(ej ) = 1 over [/4, /4] and is zero elsewhere,
we get
R(ej )
1
2
/4
Y (ej() )d
/4
Now, note from Fig. 14.17 that the function Y (ej() ) is equal to one for all values of and that
satisfy

2
2
or, equivalently, for values of that lie inside the interval
+
2
2
365
We consider several possibilities according to how the boundaries /2 compare with /4:
1.
=
=
=
.
4
1 h
i
+
2 4
2
3
1
8
2
+ 2
1
d
2 /4
i
1 h
+ +
2
2
4
3
1
+
8
2
4 . In this case /4 and the expression for R(ej ) becomes

R(ej )
4. +
/4
=
=
1
2
In this case /4 and the expression for R(ej ) becomes

R(ej )
3.
USEFUL
PROPERTIES
4 . In this case /4 and the expression for R(ej ) becomes

R(ej )
2. +
SECTION 14.2
.
4
1
2
/4
d = 1/4
/4
In this case /4 and the expression for R(ej ) becomes

R(ej )
1
2
/4
d = 1/4
/4
In this case 3/4 and R(ej ) = 0.
5.
6. +
4 . In this case 3/4 and R(ej ) = 0.
.
4
The last row in Figure 14.17 shows the resulting R(ej ).

Graphical method. Evaluating R(ej ) graphically is far more immediate in this example. If we flip
Y (ej ) around the vertical axis we obtain the same plot back. If we now shift Y (ej ) to the left and
to the right and evaluate the common area with X(ej ) (and divide the result by 2), we can easily
deduce the form of R(ej ) shown in Fig. 14.17.
14.2.9 Conjugation
Consider the eleventh line in Table 14.1. It establishes the following property:
x (n)

X(ej )
(14.14)
That is, if the sequence x(n) is conjugated, which amounts to replacing its individual terms
by their complex conjugates, then the DTFT of x(n) is obtained by replaced by in
X(ej ) and then conjugating the result.
366
X(ej )
CHAPTER 14
PROPERTIES
OF THE
DTFT
(rad/sample)
(rad/sample)
(rad/sample)
Y (ej )
1
R(ej )
1/4
3
4
3
4
FIGURE 14.17 Illustration of the multiplication property for the data in Example 14.11. The top
two plots show the DTFTs of the sequences x(n) and y(n). The bottom plot shows the DTFT of
r(n) = x(n)y(n).
Proof: Let w(n) = x (n). Using the definition (13.4) of the DTFT we have
W (ej )
w(n)ejn
n=
x (n)ejn
n=
h
X
x(n)ejn
n=
"
=
h
x(n)e
n=
X(ej )
jn
Example 14.12 (Illustrating the conjugation property)

Consider the complex-valued sequence
x(n) = (n) + j(n 1)
367
By definition, its DTFT is given by
SECTION 14.2
X(ej )
x(0) + x(1)ej
=
=
1 + je
1 + j (cos() j sin())
USEFUL
PROPERTIES
(1 + sin()) + j cos()
Consider now the conjugated sequence

y(n) = x (n) = (n) j(n 1)
By definition, its DTFT is given by
Y (ej )
y(0) + y(1)ej
1 jej
=
=
1 j(cos() j sin())
(1 sin()) j cos()
Now note that if we simply conjugate the expression for X(ej ) we do not get Y (ej ) since
h
X(ej )
= (1 + sin()) j cos()
Instead, we first need to replace by in the expression for X(ej ) to get

X(ej )
(1 + sin()) + j cos()
(1 sin()) + j cos()
and then conjugate X(ej ) to arrive at

h
X(ej )
= (1 sin()) j cos()
which agrees with Y (ej ).
14.2.10 Real Sequences

For a real-valued sequence x(n), the magnitude and phase components of the DTFT have
the following properties:
|X(ej )| is an even function of
(14.15)
X(ej ) is an odd function of

In other words, the magnitude and phase plots satisfy the symmetry properties
|X(ej )| = |X(ej )|
and
X(ej ) = X(ej )
(14.16)
so that |X(ej )| is symmetric about the vertical axis and X(ej ) is symmetric about the
origin. This result suggests that for real sequences, it is sufficient to plot the magnitude
and phase of the DTFT over the smaller interval [0, ], since the plot over [, 0] can
be deduced from the symmetry properties. Likewise, the real and imaginary parts of the
368
DTFT satisfy similar symmetry properties:
CHAPTER 14
PROPERTIES
OF THE
DTFT
Real part of X(ej ) is an even function of

(14.17)
Imaginary part of X(ej ) is an odd function of
Proof: Using the definition (13.4) of the DTFT, and Eulers relation (3.9), we have
X(ej )
x(n)ejn
n=
n=
x(n)[cos(n) j sin(n)]
Therefore, since x(n) is real, the real and imaginary components of X(ej ) are given by
XR (ej )
n
X
x(n) cos(n)
n=
XI (ej )
n
X
x(n) sin(n)
n=
where XR and XI denote the real and imaginary components of X(ej ), respectively. It is now
clear that
XR (ej ) = XR (ej ) and XI (ej ) = XI (ej )
It follows that
|X(ej )| = |X(ej )|
X(ej ) = X(ej )
and
Example 14.13 (Illustrating the symmetry properties)

Consider the real-valued sequence
y(n) =

1
cos
n sinc
(n 1)
4
2
4
We already evaluated its DTFT in Example 14.6 (as the sequence y3 (n) in that example). The DTFT
of y(n) is reproduced in Fig. 14.18. It is seen that the magnitude plot is symmetric about the vertical
axis, while the phase plot is symmetric about the origin.
Consider the twelfth line in Table 14.1. It establishes the following equality, which is
known as Parsevals relation,
x(n)y (n) =
n=
1
2

X(ej ) Y (ej ) d
(14.18)
The sum on the left-hand side is in terms of products of time-domain samples of the
form x(n)y (n), which involve the conjugated terms of the sequence y(n). The integral expression on the right-hand side involves the area under the frequency-domain curve
369
|Y (ej )|
SECTION 14.2
USEFUL
PROPERTIES
1/2
3
4
3
4
(rad/sample)
(rad/sample)
Y (ej )
/4
3/4
/4
3
4
/4
FIGURE 14.18 Illustration of the symmetry properties for real-valued sequences as in Example
14.13. Note that the magnitude plot is an even function of while the phase plot is an odd function
of .

X(ej ) Y (ej ) over an interval of duration 2; this curve involves the conjugated
DTFT of the same sequence y(n) (and the area is normalized by 1/2). Therefore, Parsevals relation is an equality between a time-domain computation and a frequency-domain
computation; the relation allows us to move back and forth between the time and frequency
domains.
Note in particular the useful special case that arises when we select the sequence y(n)
to be x(n). In this case, Parsevals relation reduces to
|x(n)|2 =
n=
2
1 R
X(ej ) d
2
2
(14.19)
On the left-hand side we have the energy of the sequence x(n). We therefore find that
the energy of a sequence can be evaluated in the frequency domain by determining the
area under the curve |X(ej )|2 and normalizing the result by 2. It should be noted that
the quantity |X(ej |2 , which is equal to the square of the magnitude of the DTFT of the
sequence, is known as the spectrum of the sequence:
spectrum of x(n) = |X(ej )|2
(14.20)
Thus, we find that the energy of a sequence coincides with the normalized area under its
spectrum (the normalization is obtained by dividing by 2).
Proof: We now establish Parsevals relation (14.18). We already know from the complex conjugation property (14.14) that
h
i
y (n)
Y (ej )
Let r(n) denote the product sequence

r(n) = x(n)y (n)
370
and recall the definition (13.4) of the DTFT of a sequence, namely,
CHAPTER 14
PROPERTIES
OF THE
DTFT
R(ej ) =
r(n)ejn
n=
j
It follows that the value of R(e ) at = 0 is equal to the sum of the samples of the sequence r(n),
i.e.,

R(ej )
=0
r(n)
n=
This is a general and useful result. Applying this fact to the current context we see that we should
evaluate R(ej0 ) since
R(ej0 ) =
x(n)y (n)
n=
in view of the definition r(n) = x(n)y (n). Now we know from property (14.11) regarding the
multiplication of sequences in time that R(ej ) is given by the circular convolution
1
2
R(ej ) =
Therefore,
R(ej0 ) =
X(ej ) Y (ej(+))
1
2
X(ej ) Y (ej() )
and we arrive at the equality
x(n)y (n) =
n=
1
2
X(ej ) Y (ej() )
as desired.
Example 14.14 (Illustrating Parsevals relation)

x(n) = (n) (n 1)
and
y(n) = 2(n) + j(n 1)
with two samples each. In particular, the sequence y(n) is complex-valued. Obviously,
S
=
=
=
=
x(n)y (n)
n=
x(0)y (0) + x(1)y (1)
(1 2) + (1) (j)
2+j
Let us now arrive at this same result by means of Parsevals relation, which performs the calculations
in the frequency domain. First note that
X(ej ) = 1 ej ,
Y (ej ) = 2 + jej
371
and, hence,
S
1
2
X(ej ) Y (ej )
SECTION 14.2
USEFUL
PROPERTIES
1
(1 ej ) (2 + jej ) d
2
Z
1
(1 ej ) (2 jej )d, (the two terms j and ej are conjugated)
2
Z
1
(2 jej 2ej + j)d
2

1

(2 + j) ej 2jej
2
=
(2 + j) + 0 + 0
(2 + j)
=
=
=
=
as expected.
Example 14.15 (Evaluating integrals and series)

Consider the DTFT of the rectangular pulse as established in Example 13.5:
(
x(n) =
1,
0,
0nL1
otherwise
DTFT
X(ej ) =
8
>
< L,
>
: e
(L1)
2
sin
.
sin
when = 0
L
2
otherwise
According to Parsevals relation (14.19) the following equality holds:
X
n=
|x(n)|2 =
1
2

2

j
X(e ) d
Using the fact that x(n) is a rectangular pulse of width L we have that its energy evaluates to
|x(n)|2 =
n=
L1
X
(1)2 = L
n=0
At the same time, the spectrum of x(n) is given by

2
sin2

j
X(e ) =
2
sin
L
2
and we arrive at the following result:

1
2
sin2 (L/2)
d = L
sin2 (/2)
In other words, Parsevals relation provides a useful way for evaluating some integral expressions by
using the duality between the time and frequency domains.
In a similar vein, let us consider the sinc sequence studied in Example 13.7, namely,
x(n) =
8
< c /,
:
c sin(c n)
,
c n
n=0
n 6= 0
DTFT
X(e ) =
1,
0,
|| < c
c ||
372
The energy of the sequence x(n) is given by
CHAPTER 14
PROPERTIES
OF THE
DTFT
Ex
2
c
2
+ 2
2
n=,n6=0
2
2 sin2 ( n)
c
c
c2 n2
1
n=,n6=0
2
c
sin2 (c n) A
c2 n2
X
sin2 (c n)
c2 n2
n=1
At the same time, the spectrum of x(n) is given by

2

j
X(e ) = 1
so that
1
2
over (c , c )

2

j
X(e ) d =
1
2
d = c /
c
Using Parsevals relation (14.19) we arrive at the equality

2
c
or, more compactly,
+ 2
2
c
X
sin2 (c n)
n=1
c2 n2
n=1

X
sin2 (c n)
c2 n2
( c )
2c
In this case, Parsevals relation provides a useful way for evaluating some series expressions by using
again the duality between the time and frequency domains.
14.3 UPSAMPLING AND DOWNSAMPLING

We end this chapter by revisiting the discussion on upsampling and downsampling from
Sec. 9.7 and by examining the effect of these operations on the DTFTs of the original sequences.
14.3.1 Upsampling
Recall that starting from a sequence x(n), we may upsample it by a positive integer factor
L and define the sequence see Fig. 14.19:

x(n/L)
if n/L is integer
y(n) =
(14.21)
0
otherwise
This operation amounts to inserting L 1 zeros between successive samples of x(n).
We established earlier in Sec. 9.7 the ztransformation result

(14.22)
Y (z) X z L
so that if we replace z by ej we arrive at the DTFT result
Y (ej )
X ejL
(14.23)
373
SECTION 14.3
x(n)
y(n)
UPSAMPLING
AND
DOWNSAMPLING
FIGURE 14.19 Block diagram representation of upsampling by a factor of L.
We therefore conclude that the DTFT of the upsampled sequence is compressed in frequency by a factor of L. In order to illustrate this effect, we consider the case L = 2 and
refer to the DTFT X(ej ) that is shown in the top part of Fig. 14.20; the DTFT is displayed
over an extended range of frequencies in order to highlight the presence of the images that
are centered around 2 due to the 2periodicity of the DTFT. Observe how the DTFT
of the upsampled sequence, y(n), is compressed by a factor of 2 and, as a result, the images
that were originally centered around 2 are now centered around . In this way, the
DTFT of the upsampled sequence will exhibit new components within [, ] relative to
the DTFT of the original sequence.
X(ej )
1
2
(rad/sample)
(rad/sample)
Y (ej )
1
2
2c
c
2
FIGURE 14.20 Illustration of the effect of upsamling on the DTFT of a sequence for the case
L = 2. Observe how images are added within the range [, ] in the DTFT of the upsampled
signal, y(n). The dotted lines highlight the portions of the DTFTs that lie within the range [, ].
Example 14.16 (Illustrating upsampling)

x(n) = 2(n) + 2(n 1) + (n 2) + (n 3)
which was studied in Example 14.3 and shown in the top plot of Fig. 14.6. The magnitude and phase
plots of the DTFT X(ej ) were displayed in Fig. 14.7 over the interval [, ]. We now
374
upsample the sequence x(n) by a factor of L = 2 to generate
CHAPTER 14
PROPERTIES
OF THE
DTFT
y(0)
x(0) = 2
y(1)
y(2)
x(1) = 2
y(3)
y(4)
x(2) = 1
y(5)
y(6)
x(3) = 1
y(7)
y(8)
..
.
0
..
.
Figure 14.21 shows the original and upsampled sequences x(n) and y(n) over the interval 0 n
14. All other samples of both sequences are zero.
x(n)
2
7
n
y(n)
10
11
12
13
14
7
n
10
11
12
13
14
FIGURE 14.21 The original sequence x(n) (top) and the upsampled sequence y(n) by a factor
of L = 2 (bottom) over 0 n 14.
Figure 14.22 shows the DTFT of x(n) and y(n) over the range [3, 3]. In Fig. 14.23
we limit the plots to the interval [, ]. Thus, observe how the plots that correspond to the
upsampled sequence, y(n), are compressed relative to the original plots. In particular observe how,
due to the compression that occurs in the frequency domain, parts of the periodic images of X(ej )
that are centered around = 2 appear now within the range [, ] in the DTFT of Y (ej ).
Observe also that, in this example, the DTFT of x(n) extends between [, ] while its compressed

, L ], which for L = 2 corresponds to the interval [ 2 , 2 ].
image extends between [ L
14.3.2 Downsampling
1.5
0.5
3
2
3
0
3
6
(rad/sample)
magnitude plot
3
0
3
6
(rad/sample)
phase plot
3
0
3
6
(rad/sample)
1.5
0.5
Y(e )
|Y(e )|
UPSAMPLING
AND
DOWNSAMPLING
0.5
1.5
9 6
3
2
0
0.5
1
1
9 6
SECTION 14.3
1
0
9 6
375
phase plot
X(e )
|X(e )|
magnitude plot
3
0
3
6
(rad/sample)
1.5
9 6
FIGURE 14.22 The top figure shows the magnitude and phase plots of x(n) over [3, 3],
while the bottom figure shows the resulting magnitude and phase plots when the sequence x(n) is
upsampled by a factor of L = 2 to generate y(n).
Recall further that starting from a sequence x(n), we may downsample it by a positive
integer factor M and define the sequence see Fig. 14.24:
y(n) = x(M n)
This operation amounts to retaining only samples of x(n) that occur at multiples of M and
discarding all other samples.
We established earlier in Sec. 9.7 the ztransformation result
Y (z) =
M1

1 X
k 1/M
X WM
z
M
(14.24)
k=0
where WM denotes the M th root of unity, i.e.,

WM = ej2/M
(14.25)
If we replace z by ej we arrive at the DTFT result

Y (ej )
P j(2k)
1 M1
X e M
M k=0
(14.26)
We therefore conclude that the DTFT of the downsampled sequence is expanded in frequency by a factor M .
376
magnitude plot
5
0.5
X(e )
|X(e )|
1.5
3
2
3 2
1 0
1 2
(rad/sample)
magnitude plot
1.5
3 2
1 0
1 2
(rad/sample)
phase plot
1
0
1
2
(rad/sample)
1.5
0.5
Y(e )
0
0.5
1
|Y(e )|
PROPERTIES
OF THE
DTFT
phase plot
CHAPTER 14
3
2
0
0.5
1
1
3 2
1
0
1
2
(rad/sample)
1.5
3 2
FIGURE 14.23 The top figure shows the magnitude and phase plots of x(n) over [, ],
upsampled by a factor of L = 2 to generate y(n).
In the special case M = 2, the result specializes to

Y (ej ) =
=
=
i

1 h j/2
+ X ej/2
X e
2
i

1 h j/2
+ X ej(+2)/2
X e
2

i
1 h j/2
X e
+ X ej(/2+)
2
x(n)
(14.27)
y(n)
FIGURE 14.24 Block diagram representation of downsampling by a factor of M .
In order to illustrate the effect of downsampling in the frequency domain, we consider

the case L = 2 and refer to the DTFT X(ej ) that is shown in the top part of Fig. 14.25;
the DTFT is displayed over an extended range of frequencies in order to highlight the presence of the images that are centered around 2 due to the 2periodicity of the DTFT.
The DTFT of the downsampled sequence is obtained by using (14.27). Observe how the
DTFT of the downsampled sequence, y(n), is expanded by a factor of 2.
377
SECTION 14.4
X(ej )
APPLICATIONS
1
2
(rad/sample)
(rad/sample)
Y (ej )
1/2
2
c
2c
FIGURE 14.25 Illustration of the effect of downsampling on the DTFT of a sequence for the case
L = 2. The dotted lines highlight the portions of the DTFTs that lie within the range [, ].
Example 14.17 (Illustrating downsampling)

x(n) = 2(n) + 2(n 1) + (n 2) + (n 3)
which was studied in Example 14.3 and shown in the top plot of Fig. 14.6. The magnitude and phase
plots of the DTFT X(ej ) were displayed in Fig. 14.7 over the interval [, ]. We now
downsample the sequence x(n) by a factor of L = 2 to generate
y(0)
x(0) = 2
y(1)
x(2) = 1
y(2)
x(4) = 0
Figure 14.26 shows the original and the downsampled sequences x(n) and y(n) over the interval
0 n 8. All other samples of both sequences are zero.
Figure 14.27 shows the DTFT of x(n) and y(n) over the extended range [3, 3]. In
Fig. 14.28 we limit the plots to the interval [, ]. Thus, observe that since the DTFT of x(n)
extends over the entire range [, ], in this example we obtain interference among adjacent images
while forming the combination (14.27) to arrive at Y (ej ). For this reason, the DTFT of y(n) over
[, ] is not simply an expanded version of the DTFT of x(n) (as was the case with the illustration
in Fig. 14.25) but rather a distorted version of it.
14.4 APPLICATIONS
TO BE ADDED
Practice Questions:
1.
378
x(n)
CHAPTER 14
PROPERTIES
OF THE
DTFT
4
n
y(n)
4
n
FIGURE 14.26
over 0 n 8.
The original sequence x(n) (top) and the downsampled sequence y(n) (bottom)
2.
14.5 PROBLEMS
(a) x(n) =
n
3
sin
(b) y(n) = e
j
4
sin
(c) w(n) = ej 4 n
n
4
n
sin
.
n
6
.
n
(d) z(n) = x(n)y(n) from parts (a) and (c).
(e) r(n) = x(n)y (n) from parts (a) and (c).
(f) s(n) = x(n) y(n) from parts (a) and (c).
(a)
(b)
(c)
(d)
(n
3
1)
x(n) =
.
n1

sin 6 (n 1)
.
y(n) = ej 4
n1

sin 8 n
.
w(n) = ej 3 n
n
z(n) = x(n)y(n) from parts (a) and (c).
sin
(e) r(n) = x(n)y (n) from parts (a) and (c).

(f) s(n) = x(n) y(n) from parts (a) and (c).
1.5
0.5
3
2
SECTION 14.5
PROBLEMS
0
0.5
1
1
9 6
379
phase plot
X(e )
|X(e )|
magnitude plot
3
0
3
6
(rad/sample)
magnitude plot
1.5
9 6
3
0
3
6
(rad/sample)
phase plot
3
0
3
6
(rad/sample)
0.5
Y(e )
|Y(ej)|
2.5
2
1.5
1
9 6
3
0
3
6
(rad/sample)
0.5
9 6
FIGURE 14.27 The top figure shows the magnitude and phase plots of x(n) over [3, 3],
downsampled by a factor of L = 2 to generate y(n). In this example, since the DTFT of x(n)
extends over the entire range [, ], we find that interference occurs among adjacent images while
forming the combination (14.27) to arrive at Y (ej ).
Problem 14.3 Use Parsevals relation to determine the following quantities for the sequences in
Prob. 14.1:
(a)
(b)
(c)
(d)
(e)
(f)
n=
n=
n=
|x(n)|2 .
|w(n)|2 .
|z(n)|2 .
n=
z(n).
n=
r(n).
P
P
n=
|s(n)|2 .
Problem 14.4 Use Parsevals relation to determine the following quantities for the sequences in
Prob. 14.2:
(a)
(b)
(c)
(d)
(e)
(f)
n=
n=
n=
|x(n)|2 .
|w(n)|2 .
|z(n)|2 .
n=
z(n).
n=
r(n).
P
P
n=
|s(n)|2 .
Problem 14.5 Consider the same sequences given in Prob. 14.1. Find the DTFTs of the following
variations:
(a) x(2n).
380
magnitude plot
phase plot
6
5
0.5
|X(e )|
PROPERTIES
OF THE
DTFT
1.5
X(e )
CHAPTER 14
3
2
0
0.5
1
1
3 2
1
0
1
2
(rad/sample)
magnitude plot
1.5
3 2
1
0
1
2
(rad/sample)
phase plot
1
0
1
2
(rad/sample)
0.5
Y(e )
|Y(ej)|
2.5
2
1.5
1
3 2
1
0
1
2
(rad/sample)
0.5
3 2
FIGURE 14.28 The top figure shows the magnitude and phase plots of x(n) over [, ],
downsampled by a factor of L = 2 to generate y(n).
(b) y(2n).
(c) w(3n).
(d) z(2n).
(e) cos
n
3
n
x(2n).
(f) (1) y(2n).

(g)
sin
n
4
y(n).
variations:
(a) x(2n).
(b) y(2n).
(c) w(3n).
(d) z(2n).
(e) cos
n
3
n
x(2n).
(f) (1) y(2n).

(g)
sin
n
4
y(n).
variations:
(a) upsample x(n) by a factor of 2 to get x (n).
(b) upsample y(n) by a factor of 2 to get y (n).
(c) upsample w(n) by a factor of 2 to get w (n).
381
(d) upsample z(n) by a factor of 2 to get z (n).
SECTION 14.5
(e) x (n)y (n). Compare with (d).
PROBLEMS
variations:
(a) upsample x(n) by a factor of 2 to get x (n).
(b) upsample y(n) by a factor of 2 to get y (n).
(c) upsample w(n) by a factor of 2 to get w (n).
(d) upsample z(n) by a factor of 2 to get z (n).
(e) x (n)y (n). Compare with (d).
Problem 14.9 Let

x(n) = e
n
j
3
n
8
sin
Find the DTFTs of the following sequences:

(a) x(3n).
(b) (1)n x(2n).
(c) nx(2n).
(d) x(n).
(e) x(2n).
(f) (1)n+1 x(n 2) cos
(n
4
1) .
Problem 14.10 Let

x(n) = cos
sin
Find the DTFTs of the following sequences:
(n
8
1)
n1
(a) x(4n).
(b) (1)n2 x(3n).
(c) n2 x(2n).
(d) x(n + 1).

(e) x(3n).
(f) (1)n1 x(n + 2) sin
(n
2
1) .
Problem 14.11 Use the properties of the DTFT to establish the transform pair
(n + 1)n u(n),
|| < 1
1
(1 ej )2
Problem 14.12 Establish the validity of the following DTFT relation

x(2n + 1)
i
1h
X(ej ) X(ej()
2
Problem 14.13 Let x(n) = sinc(n/3). Plot the DTFT of

y(n) = (1)n x(n + 2) cos
Problem 14.14 Determine the DTFT of the sequence

x(n) = n sinc
(n 1)
4

3
382
CHAPTER 14
PROPERTIES
OF THE
DTFT
in two different ways: (a) by using the differentiation property of the DTFT and (b) by using the
linearity and time-shift properties of the DTFT.
Problem 14.15 Find and plot the DTFTs of the following sequences:
n
+ 18 sinc n
8
16
1

2 n
n
sinc
cos
2
8
4
(a) x(n) = 81 sinc

(b) y(n) =
cos
1
4
3n
16
sinc
n
4
(c) x(n)y(n).
Problem 14.16 Find and plot the DTFTs of the following sequences:
sin 2 n
.
(a) x1 (n) = (1)n
n

2
sin 8 n
.
(b) x2 (n) =
n
(c) ej 2 n x2 (n) x1 (n 2).

n
3
(d) x2 (n) cos
Problem 14.17 Let x(n) be a real-valued sequence. Show that its DTFT satisfies the symmetry
property
X(ej ) = X(ej() )
Problem 14.18 Show that the DTFT of the the sequence y(n) = x(n) x (n) is equal to the
spectrum of x(n), which is defined as
Y (ej ) = |X(ej )|2
Problem 14.19 Consider a sequence x(n) and let X(ej ) denote its DTFT. Establish the validity
of the symmetry properties listed in Table 14.2.
TABLE 14.2 Additional symmetry properties for the DTFT.
sequence x(n)
DTFT X (ej )
real-valued and even

imaginary and odd
real-valued and odd
imaginary and even
real-valued and even

real-valued and odd
imaginary and odd
imaginary and even
Problem 14.20 Can you express the DTFT of x(3n + 2) in terms of the DTFT of x(n)?
Problem 14.21 Figure 14.29 shows the magnitude and phase plots of the DTFT of a real sequence
x(n). Plot the DTFTs of the following sequences:
(a) cos
(b) cos
n
2
n
2
n
x(n).
x(n).
(c) (1) x(n).

(d) xe (n), even part of the sequence x(n).
(e) xo (n), odd part of the sequence x(n).
Determine also the energy of the sequence x(n).
Problem 14.22 The DTFT of a sequence x(n) is shown in Fig. 14.30. Answer the following
questions without determining x(n).
383
X(ej )
SECTION 14.5
PROBLEMS
2 4
(rad/sample)
(a) Find
(b) Find
n=
x(n).
n= (1)
x(n).
(c) Find the energies of x(n) and nx(n).

(d) Find x(0).
(e) Find
n=
x(n) cos( 4 n).
(f) Plot the DTFT of (1)n nx(n) sin
n
2
|X(ej )|
2
2
3
4
3
4
(rad/sample)
X(ej )
2
3
4
3
4
(rad/sample)
FIGURE 14.30 DTFT plot for Probs. 14.22, 14.23 and 14.24.
Problem 14.23 For the same DTFT in Fig. 14.30, determine x(n). Plot also the DTFT of x(n)
over the interval [0, 2] rather than [, ].
384
CHAPTER 14
PROPERTIES
OF THE
DTFT
Problem 14.24 Using the same DTFT in Fig. 14.30, plot the magnitude and phase responses of
the modified DTFT: X (ej ) = ej (2+ 3 ) X(ej ). Is the corresponding inverse transform x (n)
a real sequence?
Problem 14.25 Consider the sequence x(n) = (0.5)n u(n). Evaluate the following quantities
without finding X(ej ):
(a) X(ej0 ).
(b) X(ej ).
(c)
1
2
(d)
1
2
X(ej )d.
|X(ej )|2 d.
Problem 14.26 Find the sequence x(n) whose DTFT is given by

X(ej ) =
ej
1 21 ej50
Problem 14.27 Evaluate the following series by using properties of the DTFT:
X
cos
n
4
n2
n=1
and

X
sin
n=2
2
n
8
Problem 14.28 Evaluate the following integral and series using properties of the DTFT:
Z
sin (4) sin (3)

d
1 cos2 2
and

X
sin
n=1
3
n
4
Problem 14.29 Let x(n) = n u(n) with || < 1. Evaluate the following ratio by using the
properties of the DFTF:
P
2
n=0 n x(n)
P
n=0 x(n)
Problem 14.30 Let x(n) = 2n u(n 1) with || < 1. Evaluate the following ratio by using the
properties of the DFTF:
P
n2 x(n)
Pn=0
n=0 nx(n)
Problem 14.31 Consider the signal
x(n) =
cos[(1 2 )n] cos[(1 + 2 )n]

2 2 n2
where 1 = 3/4 and 2 = /2 (both measured in radians/sample).

(a) Plot the DTFT of x(n).
(b) Evaluate the sum S =
n= (1)
x(n).
CHAPTER
15
Frequency Response
equences and systems can be studied in the frequency domain as opposed to the time
domain and z-transform domains. In this chapter we explain what is meant by the frequency content of a sequence and the frequency response of an LTI system. In doing so,
it will become clear that the DTFT plays a pivotal role in characterizing the frequency
representations of signals and systems. Specifically, we shall see that the DTFT of a sequence conveys important information about the frequency content of the sequence, while
the DTFT of the impulse response sequence of a stable LTI system conveys important
information about the frequency response of the system.
15.1 FREQUENCY CONTENT OF A SEQUENCE

We first explain how the DTFT of a sequence, x(n), conveys information about the frequency content of the sequence. Thus, consider a sequence x(n) and let X(ej ) denote its
DTFT. Using the inversion formula (13.14) we know that x(n) and X(ej ) are related as
follows:
Z
1
X(ej )ejn d
(15.1)
x(n) =
2
We now approximate the integral expression on the right-hand side by means of a finite
sum. We divide the interval [, ] into 2N subintervals of width each where (see
Fig. 15.1):
=
2
=
radians
2N
N
(15.2)
and N is large enough for to be sufficiently small.
(rad/sample)
FIGURE 15.1
The interval [, ] is subdivided into small intervals of width each.
385
Copyright
386
Then we can write
CHAPTER 15
FREQUENCY
RESPONSE
x(n)
=
N
1 X
X(ejk )ejkn
2
k=N

N
X
1
jk
X(e
) ejkn
2
(15.3)
k=N
where and d in (15.1) has been replaced by k and , respectively.

Expression (15.3) indicates that the sequence x(n) can be approximated as a linear
combination of exponential sequences of the form ejkn ; each with angular frequency
k. The scaling coefficients of the linear combination are given by
1
X(ejk ), k = N, . . . , 0, . . . , N
2
(15.4)
Each of the factors in (15.4) provides an indication of the strength of the contribution of
the corresponding exponential sequence to the formation of x(n). We therefore say that
the DTFT of x(n) provides information about the frequency content of x(n): angular frequencies with large magnitude values |X(ej )| contribute more strongly to the sequence
x(n) than angular frequencies with smaller magnitude values |X(ej )|.
Example 15.1 (Illustrating the frequency content of a sequence)


1
sinc
n
4
4
We already know from Table 13.1 that the DTFT of x(n) is given by
x(n) =
(
X(e ) =
1,
0,
|| /4
otherwise
The DTFT of x(n) is real-valued and is shown in Fig. 15.2 over the range [, ]. We observe
that over this range of frequencies, only angular frequencies that lie within [ 4 , 4 ] contribute to the
formation of x(n), as is also evident from the inversion formula
x(n) =
1
2
/4
ejn d
/4
In this situation, all exponential sequences of the form ejn , for values of [ 4 , 4 ], contribute
at the same strength level towards x(n).
Now recall from the discussion that led to Fig. 3.8 that angular frequencies close to are termed
high frequencies, while angular frequencies close to 0 are termed low frequencies. We therefore say
that the sequence x(n) in this example is a low-frequency content sequence. This is because most of
its DTFT is concentrated closer to the lower frequency range.
Consider further the sequence y(n) whose DTFT is illustrated in Fig. 15.3 over the range [p, ].
It is seen from the figure that angular frequencies in the range [ 4 , 4 ] contribute equally to the
] contribute
frequency content of y(n). On the other hand, angular frequencies in the range [ 4 , 3
4
less heavily and with diminishing relevance to the frequency content of y(n); likewise for the angular
frequencies in the range [ 3
, 4 ]. We therefore say that the frequency content of the sequence
4
y(n) is concentrated in the range [ 3
, 3
], with higher emphasis for the range [ 4 , 4 ].
4
4
387
SECTION 15.1
X(ej )
FREQUENCY
CONTENT OF
A SEQUENCE
(rad/sample)
FIGURE 15.2 Frequency content of the sequence x(n) = 41 sinc
n
4
Y (ej )
1
3
4
3
4
(rad/sample)
FIGURE 15.3 Frequency content of a sequence y(n).

x(n) = n u(n),
|| < 1
According to Table 13.1, its DTFT is given by

X(ej ) =
1
1 ej
This DTFT is a complex-valued function of ; it has both real and imaginary parts. To determine
these parts, we assume for simplicity that is real-valued. Then we can write
X(ej )
=
=
=
=
1
1 ej
j
1 e
1 ej
j
1 e
1 + 2 2 cos
1 cos j sin
1 + 2 2 cos
1 cos
sin
j
1 + 2 2 cos
1 + 2 2 cos
so that the real and imaginary components of X(ej ) are given by

XR (ej ) =
1 cos
,
1 + 2 2 cos
XI () =
sin
1 + 2 2 cos
Consequently, the magnitude and phase components of X(ej ) are given by
CHAPTER 15
|X(ej )|
FREQUENCY
RESPONSE
|XR (ej )|2 + |XI (ej )|2

1
1 + 2 2 cos
=
=
and

X(ej )
arctan
arctan
XI (ej )
XR (ej )
sin
1 cos
The resulting plots are shown in Fig. 15.4 for the case = 1/2. It is seen that the exponential sequence x(n) = (0.5)n u(n) is mainly a low-frequency content signal since lower angular frequencies
contribute more heavily to the formation of x(n).
magnitude plot
phase plot
|X(ej)|
0.5
1.5
X(ej)
388
0.5
3 2
1
0
1
2
(rad/sample)
0.5
3 2
1
0
1
2
(rad/sample)
FIGURE 15.4 Plots of the magnitude and phase of the DTFT of x(n) = (0.5)n u(n).
15.2 FREQUENCY RESPONSE OF AN LTI SYSTEM

The DTFT is also useful in characterizing the so-called frequency response of stable LTI
systems. Thus, let h(n) denote the impulse response sequence of a stable LTI system. We
defined earlier in Sec. 11.1 the transfer function of the system as the ztransform of h(n),
namely,
P
H(z) =
(15.5)
h(n)z n
n=
over all values of z belonging to the corresponding ROC of H(z). Since the system is
assumed to be BIBO stable, its ROC must include the unit circle, |z| = 1. Therefore
evaluating H(z) at any point z = ej on the unit circle gives the DTFT of h(n):
H(ej ) =
h(n)ejn
(15.6)
n=
We refer to H(ej ) as the frequency response of the LTI system. Alternatively, we say
that the frequency response of a stable LTI system is the DTFT of its impulse response
sequence. As such, the frequency response and the transfer function of any stable LTI
system are related via
H(ej ) = H(z)|z=ej
(15.7)
The reason for the name frequency response is motivated as follows. Assume we excite the
LTI system with some exponential sequence, say,
x(n) = ejo n
(15.8)
at some angular frequency o [, ]. Then the resulting output sequence will be given
by the convolution sum:
y(n) =
=
k=
h(k)x(n k)
h(k)ejo (nk)
k=
= ejo n
"
h(k)ejo k
k=
= ejo n H(ejo )
(15.9)
This result shows that the same exponential sequence, ejo n , appears at the output of the
LTI system; albeit scaled by the value of the frequency response at = o see Fig. 15.5.
This conclusion is a special case of the result we obtained earlier in (11.2) while discussing
the concept of eigenfunctions of LTI systems. The value H(ejo ) represents the amount
of scaling that the LTI system performs on the exponential input sequence whose angular
frequency is o . For instance, if H(ej ) happens to be zero at = o , then the output
sequence, y(n), will be zero. Likewise, if H(ejo ) = 0.5ej 4 , then the output sequence,
y(n), will be obtained by scaling x(n) = ejo n by 1/2, and adding /4 to its phase so that
y(n) =
1 j (o n + 4 )
1
e
, when H(ejo ) = ej 4
2
2
ejo n
H(ej )
H(ejo ) ejo n
FIGURE 15.5 An exponential sequence at the angular frequency o is scaled by the value of the
frequency response at this same frequency, H(ejo ), as its passes through a stable LTI system.
We therefore say that the frequency response of a stable LTI system at any particular
angular frequency determines how the system responds to an exponential input sequence
at that same frequency. In general, the scaling factor H(ejo ) is complex-valued and,
accordingly, the input sequence x(n) = ejo n will not only have its magnitude modified
but its phase as well, as illustrated in the above example. Let us express H(ejo ) in polar
form as
jo
(15.10)
H(ejo ) = |H(ejo )| ejH(e )
389
SECTION 15.2
FREQUENCY
RESPONSE
OF AN LTI
SYSTEM
390
Then
CHAPTER 15
FREQUENCY
RESPONSE
y(n) = H(ejo n ) ejo n
jo
= |H(ejo )| ejH(e ) ejo n

jo
= |H(ejo )| ej (o n + H(e ))
(15.11)
That is, the response of the stable LTI system to ejo n is given by
ejo n
|H(ejo )| ej (o n+H(e
jo
))
(15.12)
so that both the magnitude and phase of the input sequence ejo n are modified.
In the sequel, we shall refer to the functions |H(ej )| and H(ej ) as the magnitude
and phase responses of the system:
|H(ej )|
j
H(e )
= magnitude response
(15.13)
= phase response
(15.14)
These functions are also generally specified over 2-wide intervals such as [, ].
Stable LTI Systems
We remark that the concept of a frequency response is being defined here for stable LTI
systems, i.e., for systems with absolutely summable impulse response sequences, h(n), for
which the DTFT in (15.6) converges uniformly. Although, as we saw earlier in Sec. 13.4,
the DTFT can be defined for sequences h(n) that are not necessarily absolutely summable
(e.g., square-summable sequences h(n) will do), these sequences do not correspond to
BIBO stable LTI systems. Most of our discussions will focus on stable LTI systems, although at times we shall consider square-summable impulse-response sequences as well
(e.g., when studying ideal filter responses see Example 15.5 ). It is worth noting here
that the concept of transfer functions of LTI systems is more general than the concept of
frequency responses since the former can be used to describe both stable and unstable LTI
systems.
Example 15.3 (Sinusoidal input)
Consider a stable LTI system with a real-valued impulse response sequence, h(n), and frequency
response, H(ej ). Assume the system is excited with the sinusoidal signal
x(n) = cos(o n + o )
Using Eulers relation we can write
x(n) =
1
1 j(o n+o )
e
+ ej(o n+o )
2
2
x(n) =
1 jo jo n
1
e e
+ ejo ejo n
2
2
or, equivalently,
(15.15)
391
It follows, by linearity and from (15.12), that the response of the system is given by
y(n)
jo
jo
1
1 jo jo n
)
e e
|H(ejo )| ejH(e ) + ejo ejo n |H(ejo )| ejH(e
2
2
This expression can be further simplified by using the fact that h(n) is real-valued. In this case, we
know that |H(ej )| is an even function of and H(ej ) is an odd function of (recall (14.17)).
That is, it holds that
|H(ejo )| = |H(ejo )|
H(ejo ) = H(ejo )
and
so that, by grouping terms in the expression for y(n) and applying Eulers relation again,
y(n) = |H(ejo )| cos o n + o + H(ejo )
(15.16)
In other words, we obtain the same sinusoidal sequence at the output of the system with its amplitude
scaled by |H(ejo )| and its phase adjusted by H(ejo ).
To illustrate this result, consider a stable LTI system with impulse response sequence
h(n) =
1
1
(n + 1) + (n 1)
2
2
Its frequency response is given by

H(ej ) = 0.5ej + 0.5ej = cos
The response of this system to the input sequence
x(n) = cos

is therefore

1
cos
n
2
3
since H(ej/3 ) = cos(/3) = 1/2. On the other hand, the response to
y(n) =
x(n) = cos n +
is

4

y(n) = cos n + +

5
= cos n +
4
4
since H(ej ) = cos() = 1 = ej . Finally, by linearity, the response to

x(n) = cos
is

n + cos n +

4

5
1
n + cos n +
y(n) cos
2
3
4
SECTION 15.2
FREQUENCY
RESPONSE
OF AN LTI
SYSTEM
392
CHAPTER 15
FREQUENCY
RESPONSE
Example 15.4 (Filtering)

We showed earlier in (15.3) that a sequence x(n) with DTFT X(ej ) can be approximated as a
linear combination of complex exponentials as follows:
x(n)
N
P
k=N
1
X(ejk ) ejkn
2
(15.17)
where = /N . Now consider a stable LTI system with frequency response H(ej ). Each input
complex exponential sequence of the form
ejkn ,
k = N, . . . , 0, . . . , N
generates a response of similar form when it is fed into H(ej ), namely,

ejkn
ejkn H(ejk )
If we express the frequency response at k in polar form, say,

jk
H(ejk ) = |H(ejk )| ejH(e
then we have that the exponential sequence ejkn is mapped into

ejkn
jk
|H(ejk )| ej (kn + H(e
))
Let us now examine what happens when the sequence x(n) is fed into the system H(ej ). Since
x(n) is approximated in (15.17) as a linear combination of the exponential sequences ejkwn , then
by invoking the linearity of the LTI system, we find that x(n) will be mapped into the output sequence:
y(n)
N
P
k=N
jk
)+X(ejk ))
|X(ejk )| |H(ejk )| ej (kn+H(e
2
(15.18)
where we introduced the polar representation
jk
X(ejk ) = |X(ejk )| eX(e
Construction (15.18) shows that the output sequence is composed of a combination of exponential
sequences whose magnitudes are scaled by the coefficients
|X(ejk )| |H(ejk )|
2
and whose phases are adjusted by the values
H(ejk ) + X(ejk )
We conclude that a stable LTI system modifies each frequency component, ejkn , of the input
signal via scaling and phase-change, and then combines all components together to arrive at y(n).
This construction provides a useful frequency-domain interpretation for the operation of a system
and we refer to the mapping from x(n) into y(n) as a filtering operation.
In order to illustrate this point, consider the sequence
x(n) =

1
sinc
n
2
2
whose DTFT is shown in the top plot of Fig. 15.6. Now assume we feed this sequence through an
LTI system whose frequency response is the one indicated in the middle plot of the same figure.
393
Obviously, the output sequence is given by the linear convolution
SECTION 15.2
y(n) = x(n) h(n)

and, as expected, the frequency content of y(n) is obtained from multiplying the individual DTFTs:
Y (ej ) = H(ej )X(ej )
The corresponding plot is the bottom plot in Fig. 15.6. In particular, observe that while the frequency
content of x(n) is flat over the range [ 4 , 2 ], these same frequency components in y(n) appear
with different scalings as evidenced by the linear inclination in the graph of Y (ej ) over [ 4 , 2 ].
We remark in passing that in this example, if desired, the sequence y(n) can be recovered from
the inversion formula
Z
1
y(n) =
Y (ej )ejn d
2
which can be evaluated by dividing the integral into several integrals over smaller intervals as follows:
y(n) =
1
2
/4
/2
3
4
1
2
ejn d +
/2
/4
3
4
(rad/sample)
(rad/sample)
(rad/sample)
ejn d +
1
2
/4
/4
jn
e d
2
and the evaluation completed to find y(n).
X(ej )
1
H(ej )
/2
3
4
3
4
Y (ej )
/2
/4
FIGURE 15.6
DTFT of the output sequence (bottom plot) for a stable LTI system whose
frequency response is described by the middle plot and input sequence is described by the top plot.
FREQUENCY
RESPONSE
OF AN LTI
SYSTEM
394
CHAPTER 15
FREQUENCY
RESPONSE
Example 15.5 (Low-pass filter)

Consider the DTFT shown in Fig. 15.7 over [, ]:
(
1,
0,
H(e ) =
|w| < wc
wc |w|
In the figure, the DTFT is equal to one over the interval [c , c ] and is zero elsewhere. This
DTFT cannot be the frequency response of a stable LTI system. Indeed, if the system were stable,
then its impulse response sequence would need to be absolutely summable. When this happens, the
DTFT would be uniformly convergent and, from the discussion in Sec. 13.2, the DTFT would need
to be a continuous function of . Since the DTFT in Fig. 15.7 is discontinuous at c we conclude
that the corresponding LTI system is not BIBO stable. Nevertheless, we shall refer to H(ej ) as
the frequency response of an ideal low-pass filter. This is because the frequency components of any
input sequence, x(n), that lie outside the range [c , c ] are filtered out by H(ej ) and will not
appear in the output sequence, y(n).
H(ej )
1
(rad/sample)
FIGURE 15.7 A plot of the DTFT for Example 15.5.
Example 15.6 (Steady-state response to sinusoidal excitations)

Let us examine the response of a system that is not LTI but is described by a constant-coefficient
difference equation. Thus, consider a causal system that is described by the difference equation
y(n) ay(n 1) = x(n),
n 0,
y(1) =
and assume |a| < 1. As we already know (see, e.g., the discussion in Sec. 4.9), this difference
equation does not describe an LTI system because of the initial condition.
We now excite the system with the exponential sequence
x(n) = ejo n u(n)
where the step sequence, u(n), ensures that the input sequence is applied over the interval n 0
over which the system is defined. We proceed to determine the response of the system. Recall from
Sec. 8.6 that the output sequence can be expressed as the sum of the zero-input response and the
zero-state response of the system, namely,
y(n) = yzi (n) + yzs (n)
Recall further that the zero-input response is described in terms of the modes of the system. In this
case, the system has a single mode at = a. Therefore, the zero-input response has the form
yzi (n) = Can
for some constant C to be determined in order to satisfy the initial condition yzi (1) = . This
leads to C = a so that
yzi (n) = an+1 , n 1
Let us now determine the zero-state response. To do so, we assume for this step that the system
is relaxed and, hence, LTI. Then, the zero-state response can be found by using the z-transform
representation:
Yzs (z) = H(z)X(z)
where
H(z) =
z
,
za
X(z) =
z
,
z ejo
and
Therefore,
|z| > |a|

|z| > 1
z2
,
(z a)(z ejo )
Using partial fractions expansion we obtain
Yzs (z) =
Yzs (z) =
z
a ejo
|z| > 1
a
ejo
za
z ejo
so that, by inverse transformation, the sequence yzs (n) over n 0 is given by

yzs (n) =
1
1
an+1 u(n)
ejo (n+1) u(n)
a ejo
a ejo
Combining with the result for yzi (n) we arrive at the desired expression for the output sequence over
n 0:
h
i
1
n+1
jo (n+1)
a
e
, n0
y(n) = an+1 +
a ejo
which is equivalent to

y(n) = +
1
an+1 u(n) + H(ejo )ejo n u(n)
a ejo
where
ejo
ejo a
Observe that the expression for y(n) consists of two terms. The first term on the right-hand side is
a transient term that dies out as n since |a| < 1. The second term on the right-hand side
determines the steady-state response of the system, namely,
H(ej ) = H(z)|z=ejo =
yss (n) = lim y(n) = H(ejo )ejo n

n
In comparison with (15.9), we now find that H(ejo )ejo n has the interpretation of being the steadystate response of the system after the transient component has died out.
For illustration purposes, the same conclusion can be obtained by resorting to the unilateral ztransform technique. Indeed, using y(1) = and starting from the given difference equation, we
can write
a
X + (z)
+
, |z| > |a|
Y + (z) =
1 az 1
1 az 1
Using
z
X + (z) =
,
|z| > 1
z ejo
395
SECTION 15.2
FREQUENCY
RESPONSE
OF AN LTI
SYSTEM
396
CHAPTER 15
FREQUENCY
RESPONSE
we get
z2
az
,
+
(z a)(z ejo )
za
which by partial fractions is equal to
Y + (z) =
z
Y (z) =
a ejo
z
z
za
z ejo
|z| > 1
az
za
It then follows by inverse transformation that

y(n) = an+1 +
h
i
1
an+1 ejo (n+1) ,
j
o
ae
n0
which is equivalent to

y(n) = +
1
an+1 + H(ejo )ejo n
a ejo
and the argument continues as before.
Example 15.7 (Geometric Interpretation)

It is useful to get some further insight into the frequency response of a stable LTI system by examining its pole-zero diagram. To see this, let us consider a simple example that is described by the
transfer function
z z1
H(z) =
z p1
with a zero at z1 and a pole at p1 . Since this is a rational transfer function, we know that the ROC is
either |z| > |p1 | or |z| < |p1 |. However, by the assumption of stability, the ROC must include the
unit circle. Therefore, if we assume that |p1 | < 1, then the ROC is given by |z| > |p1 |.
Im
ejo
x
ej1
y
p1
z1
Re
FIGURE 15.8 Vectors ~

x and ~
y for two different angular frequencies o and 1 .
Now for any value of in the range [, ], the complex number ej represents a point on the
unit circle at an angle relative to the positive horizontal axis. Let us pick any o and let ~
x and ~
y
denote the vectors connecting the points z1 and p1 to the point ejo , respectively. The frequency
response at ejo is then given by
ejo z1
|~
x|ej~x
H(ejo ) = jo
=
e
|~
y |ej~y
p1
so that
|H(ejo )| =
|~
x|
|~
y|
and
H(ejo ) = ~
x ~
y
In other words, the magnitude response at o is seen to be the ratio of the magnitude of ~
x to the
magnitude of ~
y . Likewise, the phase response at o is seen to be the phase of ~
x minus the phase of
~
y.
This construction suggests that the magnitude response of the LTI system will be relatively small
for points o that are close to the location of its zero, since in that case the vector ~
x will generally
have small magnitude compared to the vector ~
y. Likewise, the magnitude response will be large
for points o that are close to the location of the pole. This conclusion also holds for more general
transfer functions with multiple zeros and poles.
Decibel Plots
It is common to plot the magnitude response of an LTI system by using a log magnitude
scale known as decibels (abbreviated dB). In this case, the plot would show the quantity
20 log10 |H(ej )| (dB)
(15.19)
versus the frequency variable in radians/sample. Table 15.1. lists some of the correspondences between the linear and dB scales.
TABLE 15.1 Some values of |H(ej )| and their dB values.
|H(ej )|
1
2
2
1/ 2
1/2
10
dB value
0 dB
3 dB
6 dB
3 dB
6 dB
20 dB

Consider a stable LTI system with impulse response sequence
x(n) = n u(n),
|| < 1
From the argument in Example 15.2 we know that the magnitude response is given by
1
|H(ej )| = p
2
1 + 2 cos()
Figure 15.9 plots this magnitude response using both the linear scale and the dB scale for = 1/2.
397
SECTION 15.3
FREQUENCY
RESPONSE
OF AN LTI
SYSTEM
398
linear scale
dB scale
1.5
|H(e )|
FREQUENCY
RESPONSE
6
|H(ej)| (dB)
CHAPTER 15
4
2
0
2
0.5
3
1
0
1
2
(rad/sample)
4
3
1
0
1
2
(rad/sample)
FIGURE 15.9 Plots of the magnitude response of an LTI system using the linear scale (left) and
the dB scale (right) for H(ej ) = 1/(1 0.5ej ).
15.3 LINEAR TIME-INVARIANT SYSTEMS

As was the case with transfer functions in Chapter 11, the frequency response of LTI systems that are described by constant-coefficient difference equations can be deduced directly from the difference equations without determining first the corresponding impulse
response sequences.
Example 15.9 (Finding a frequency response)

y(n)
1
y(n 1) = x(n)
2
Since the system is relaxed, and since this is a constant-coefficient difference equation, we know
that the system is LTI. We also know that the system is stable and we evaluated its transfer function
earlier in Example 11.2:
1
H(z) =
, |z| > 1/2
1 12 z 1
The ROC is {|z| > 1/2} since it must be the exterior of a disc by causality, must include the unit
circle by stability, and must exclude the pole at z = 1/2. We can now find the frequency response of
the system by evaluating H(z) at z = ej , which results in
H(ej ) =
1
1 21 ej
Alternatively, we can proceed directly from the difference equation. We evaluate the DTFTs of
all terms on both sides of the equation, and use the properties of the DTFT, to obtain the following
algebraic equation:
1
Y (ej ) ej Y (ej ) = X(ej )
2
Here, Y (ej ) denotes the DTFT of the sequence y(n) and X(ej ) denotes the DTFT of the sequence
x(n). The sequences {x(n), y(n)} denote an arbitrary input-output pair satisfying the difference
equation. The useful fact to note is that the original constant-coefficient difference equation has now
been transformed into a purely algebraic equation in the transform domain. The algebraic equation
can be solved to yield an expression for Y (ej ) in terms of X(ej ), namely,
Y (ej )
1
=
X(ej )
1 21 ej
This ratio holds for any input-output pair {Y (ej ), X(ej )}. As such, we claim that the ratio
Y (ej )/X(ej ), of the output DTFT divided by the input DTFT, should coincide with the frequency
response H(ej ) of the LTI system. To see this, assume x(n) = (n) then, by definition, y(n) =
h(n). Hence, if X(ej ) = 1 then Y (ej ) = H(ej ). Substituting into the above relation gives
H(ej ) =
1
1 21 ej
as expected.
The above example suggests an alternative way for determining the impulse response sequence of an LTI system that is described by a constant-coefficient difference equation:
we use the difference equation to determine the frequency response, H(ej ), and then inverse transform H(ej ) to find h(n).
Example 15.10 (Finding an impulse response sequence)

Consider the same causal LTI system from the previous example, which is described by the relaxed
equation
1
y(n) y(n 1) = x(n)
2
We already determined its frequency response as
H(ej ) =
1
1 21 ej
The inverse DTFT is the impulse response sequence and, from Table 13.1, it is given by
h(n) = (0.5)n u(n)
Example 15.9 also suggests a method for determining a description for an LTI system in
terms of a constant-coefficient difference equation from knowledge of its impulse response
sequence or, equivalently, its frequency response.
399
SECTION 15.3
LINEAR
TIME
INVARIANT
SYSTEMS
400
Example 15.11 (Determining a difference equation)
CHAPTER 15
FREQUENCY
RESPONSE
Consider the causal LTI system with frequency response

H(ej ) =
1
1 21 ej
difference equation. We know that the DTFTs of any input-output pair {x(n), y(n)} should satisfy
the relation
Y (ej )
1
=
X(ej )
1 21 ej
Y (ej ) 1
1 j
= X(ej )
e
2
and using the properties of the DTFT we arrive via inverse transformation at the difference equation
y(n)
1
y(n 1) = x(n)
2
The system is assumed relaxed to ensure it is LTI. Also, the difference equation runs forward in time
to ensure causality.
We can also use the frequency response of an LTI system to determine its response to
arbitrary input sequences. Thus, let x(n) denote the input sequence to an LTI system
with impulse response sequence h(n). We already know from Sec. 5.1 that the response
sequence, say y(n), is determined via the convolution sum
y(n) = x(n) h(n) =
k=
x(k)h(n k)
(15.20)
so that from the convolution property (14.10) of the DTFT we have

Y (ej ) = X(ej )H(ej )
(15.21)
The result states that the response of the LTI system can be determined via inverse transformation of the product X(ej )H(ej ).
Example 15.12 (Evaluating the response sequence)
Consider again the same causal and relaxed system from Example 15.9 and let us determine its
response to the input sequence
n
x(n)
1
3
=
=
1
3
u(n 1)
n1
1
3
u(n 1)
We already know that the frequency response of the system is given by

H(ej ) =
1
1 21 ej
401
On the other hand, the DTFT of the input sequence is

X(ej )
1
3
ej
1 13 ej
SECTION 15.4
IDEAL
FILTERS
It follows that the DTFT of the output sequence is

j
Y (e ) = X(e )H(e ) =
1
1 j
1 2e
1 j
e
3
31 ej
We can inverse-transform Y (ej ) by using the partial fractions method (recall Sec. 13.5). We determine constants A and B to satisfy the partial fractions expansion
A
B
+
1 21 ej
1 31 ej
Y (ej ) =
By comparing coefficients of powers of ej in the numerators on both sides of the above equality
we find that A = 2 and B = 2. Therefore,
2
2
1 21 ej
1 31 ej
Y (ej ) =
By inverse transforming we obtain
n
y(n) = 2
1
2
n
1
3
u(n)
15.4 IDEAL FILTERS

We end the chapter by defining the class of ideal filters. As anticipated earlier in Example
15.5, the frequency responses outlined below will not correspond to stable LTI systems.
However, these filters generally serve as references for filter design and will be used extensively later in Chapters 2628.
Ideal Low-Pass Filter
An ideal low-pass filter is an LTI system whose frequency response is assumed to be of the
form:
j
Hlp (e ) =
Aejko ,
0,
|| c
otherwise
(15.22)
where c < is called the cutoff frequency. It is seen that the magnitude response is
constant and equal to A over the interval [c , c ], while the phase response is linear
over the same interval with slope dictated by the value of ko . The range of frequencies
[c , c ] is called the passband region, and the interval over which the magnitude
of the frequency response is zero is called the stopband region. We therefore find that an
ideal low-pass filter attenuates high frequency components and leaves intact, apart from a
delay, low frequency components. The frequency response of an ideal low-pass filter is
illustrated in Fig. 15.10.
402
CHAPTER 15
|Hlp (ej )|
FREQUENCY
RESPONSE
(rad/sample)
Hlp (ej ) = ko
(rad/sample)
slope ko
FIGURE 15.10 The magnitude and phase responses of an ideal low-pass filter assuming ko > 0.
The slope of the phase plot is ko over || c .
Example 15.13 (Impulse response of an ideal low-pass filter)

Let us determine the impulse response sequence of the ideal low-pass filter Hlp (ej ) defined by
(15.22). For this purpose, we recall from Table 13.1 the DTFT pair
sin c n
x(n) =
n
(
j
X(e ) =
1,
0,
|| c
otherwise
Therefore, by invoking the time-delay property (14.4) of the DTFT we conclude that
Hlp (ej )
h(n) =
A sin c (n ko )
(n ko )
Observe that h(n) is a noncausal sequence and, therefore, is physically unrealizable. Moreover, the
sequence h(n) is square-summable but not absolutely summable. Hence, the ideal low-pass filter is
not a BIBO stable system.
Ideal High-Pass Filter

In a similar vein, an ideal high-pass filter is an LTI system whose frequency response is
assumed to be of the form:

Aejko ,
c ||
Hhp (ej ) =
(15.23)
0,
otherwise
where c < is again called the cutoff frequency. It is seen that the magnitude response is
constant and equal to A over the intervals [, c] and [c , ], while the phase
response is linear over these intervals with slope dictated by the value of ko . The interval
|| c is called the passband region.
We therefore find that an ideal high-pass filter attenuates low frequency components
and leaves intact, apart from a delay, high frequency components. The frequency response
of an ideal high-pass filter is illustrated in Fig. 15.11.
|Hhp (ej )|
A
(rad/sample)
Hhp (ej ) = ko
(rad/sample)
slope ko
FIGURE 15.11 The magnitude and phase responses of an ideal high-pass filter assuming ko > 0.
The slope of the phase plot is ko over || c .
Ideal Band-Pass Filter

An ideal band-pass filter is an LTI system whose frequency response is assumed to be of
the form

Aejko ,
1 || 2
j
Hbp (e ) =
(15.24)
0,
otherwise
where {1 < } and {2 < } are called the cut-off frequencies. It is seen that the
magnitude response is constant and equal to A over the intervals [1 , 2 ] and
[2 , 1 ], while the phase response is linear over these intervals with slope dictated by
the value of ko . The interval 1 || 2 is called the passband region.
We therefore find that an ideal band-pass filter leaves intact, apart from a delay, frequency components that lie within its passband region. The frequency response of an ideal
band-pass filter is illustrated in Fig. 15.12.
Ideal Band-Stop Filter
Finally, an ideal band-stop filter is an LTI system whose frequency response is assumed to
be of the form

Aejko ,
|| 1 and 2 ||
j
Hbp (e ) =
(15.25)
0,
otherwise
where {1 < } and {2 < } are called the cut-off frequencies. It is seen that the
magnitude response is constant and equal to A over the intervals [1 , 1 ],
[, 2 ], and [2 , ], while the phase response is linear over these intervals with
403
SECTION 15.4
IDEAL
FILTERS
404
CHAPTER 15
FREQUENCY
RESPONSE
|Hbp (ej )|
A
(rad/sample)
Hbp (ej ) = ko
(rad/sample)
slope ko
FIGURE 15.12 The magnitude and phase responses of an ideal band-pass filter assuming ko > 0.
The slope of the phase plot is ko over 1 || 2 .
|Hbs (ej )|
A
(rad/sample)
Hbs (ej ) = ko
(rad/sample)
slope ko
FIGURE 15.13 The magnitude and phase responses of an ideal band-stop filter assuming ko > 0.
The slope of the phase plot is ko over the intervals || 1 and || 2 .
slope dictated by the value of ko . The frequency response of an ideal band-stop filter is
illustrated in Fig. 15.13.
405
SECTION 15.5
Example 15.14 (Location of poles and zeros)
REALIZABLE
FILTERS
From the earlier geometric interpretation of the frequency response of an LTI system in Example
15.7 we can conclude that the poles of an ideal low-pass filter should be located close to = 0,
while its zeros should be located close to = . Likewise, the poles of a high pass filter should
be located close to = , while its zeros should be located close to = 0.
15.5 REALIZABLE FILTERS

Ideal filters will serve as useful points of reference for practical filter design (as will be
discussed later at some length in Chapters 2628). In practice we are mainly interested
in designing filters that correspond to realizable LTI implementations. This means that
we would like the resulting filters to be LTI systems that are both stable and causal. The
stability property ensures that the filter output remains bounded for bounded inputs. And
the causality property ensures that the filter output does not depend on future input samples.
Now, recall that an LTI system is stable if, and only if, its impulse response sequence,
h(n), is absolutely summable. Recall further that the absolute summability of h(n) ensures
the uniform convergence of its DTFT, H(ej ). As such, and according to the discussion
in Sec. 13.2, the uniform convergence of H(ej ) implies that the frequency response of
stable LTI systems must be a continuous function of . Therefore, the ideal filter responses
described in the previous section for low-pass, high-pass, band-pass, and band-stop characteristics, cannot correspond to stable filters; this is because their ideal frequency responses
exhibit sharp discontinuous transitions.
Still, as we shall see in Chapters 2628, these ideal responses can serve as useful starting
points for designing implementable filters whose frequency responses will not be idea, but
will be good approximations for the ideal case. Figure 15.14 compares an ideal frequency
response with a realizable frequency response; the latter exhibits smooth transitions.
H(ej ) (ideal)
H(ej ) (approximation)
FIGURE 15.14 An illustration of an ideal low-pass filter response (left) and an approximate lowpass filter response (right) with smooth transitions around c .
Moreover, the causality requirement for realizable filters means that the impulse response sequence, h(n), should be causal, i.e., h(n) = 0 over n < 0. When this condition
is coupled with the stability requirement, then the two conditions (of causality and stability) translate into a requirement on the frequency response of a realizable system known as
the Paley-Wiener condition. The condition states that the frequency response, H(ej ), of
406
a realizable filter should satisfy:
CHAPTER 15
FREQUENCY
RESPONSE

R
ln |H(ej | d <
(Paley-Wiener condition)
(15.26)
That is, the integral of the absolute log of the magnitude response is bounded over the interval [, ]. It therefore follows from the Paley-Wiener condition that while the frequency
response can assume zero values at some isolated (discrete) frequencies, it cannot be zero
over a continuous range of frequencies. This is because the function ln |H(ej | would be
divergent over that range of frequency and the condition (15.26) will then be violated.
With regards to the transfer function, H(z), of a stable and causal LTI system we note
that stability means that the ROC must include the unit circle, |z| = 1, while causality
means that the ROC must be the outside of a disc (since h(n) is a right-sided sequence).
Therefore, the ROC of a realizable system must be of the form:
ROC = {|z| > } for some 0 < 1
(15.27)
Now since the ROC must exclude all poles of H(z), we conclude that all poles of a realizable H(z) must lie inside the unit circle.
15.6 APPLICATIONS
TO BE ADDED
Practice Questions:
1.
2.
15.7 PROBLEMS
Problem 15.1 Find the frequency components that are present in each of the following sequences:
(b) x(n) =
(c) x(n) =
(d) x(n) =
n + 2 cos 3 n
6

sin 6 n + 4 + 2 cos

sin2 6 n 4 .

sin4 6 n .
(a) x(n) = sin
n
3
Problem 15.2 Find the frequency components that are present in each of the following sequences:
(a) x(n) = sin
(b) x(n) = sin

(c) x(n) = cos2
(d) x(n) = cos4
n
6
cos
n
3
n+ 4
3
n 4 .
6
n .
6
.
2 cos
n
4
2
Problem 15.3 Determine the frequency response of the LTI system whose impulse response sequence is given by h(n) = (n10)+(n+10). Identify both the magnitude and phase components
of the frequency response over the range [, ].
Problem 15.4 Determine the frequency response of the LTI system whose impulse response sequence is given by h(n) = 1 + (n 1). Identify both the magnitude and phase components of the
frequency response over the range [, ].
Problem 15.5 Determine the frequency response of the LTI system whose impulse response sequence is given by h(n) = 1 + (n 2). Identify both the magnitude and phase components of the
frequency response over the range [, ].
Problem 15.6 Determine the frequency response of the LTI system whose impulse response sequence is given by h(n) = (n + 2) + 1 + (n 2). Identify both the magnitude and phase
components of the frequency response over the range [, ].
Problem 15.7 Figure 15.15 shows the frequency response of an LTI system.
(b) Is the system realizable?
(c) For each of the input sequences in Prob. 15.1, find the corresponding output sequence.
(d) Find the energy of the impulse response sequence of the system.
(e) Find the impulse response sequence of the system.
|H(ej )|
1
1/2
H(ej )
/2
FIGURE 15.15 Frequency response plot for Prob. 15.7.
Problem 15.8 Figure 15.16 shows the frequency response of an LTI system.
(b) Is the system realizable?
(c) For each of the input sequences in Prob. 15.2, find the corresponding output sequence.
(d) Find the energy of the impulse response sequence of the system.
407
SECTION 15.7
PROBLEMS
408
CHAPTER 15
|H(ej )|
FREQUENCY
RESPONSE
1/2
H(ej )
/2
FIGURE 15.16 Frequency response plot for Prob. 15.8.
(e) Find the impulse response sequence of the system.

Problem 15.9 Can an LTI system produce frequency components in the output sequence that are
not present in the input sequence? Explain or give a counter-example when necessary.
Problem 15.10 Can a nonlinear system produce frequency components in the output sequence that
are not present in the input sequence? Explain or give a counter-example when necessary.
Problem 15.11 Consider a causal LTI system that is described by the difference equation
y(n)
3
1
y(n 1) + y(n 2) = x(n 1)
4
8
(a) Find the transfer function of the system. Find its zeros and poles.
(b) Find the frequency response, H(ej ).
(c) Find the impulse response sequence, h(n).
(e) Find the steady-state response to x(n) =
(f) Find the steady-state response to x(n) =
(g) Find the steady-state response to x(n) =
n + 4 .
6

sin 3 n 8 .

sin2 6 n + 4 .

cos 6 n u(n).
(d) Find the steady-state response to x(n) = cos
Problem 15.12 Consider a causal LTI system that is described by the difference equation
y(n)
7
1
1
y(n 1) +
y(n 2) = x(n) + x(n 1)
12
12
2
(a) Find the transfer function of the system. Find its zeros and poles.
(b) Find the frequency response, H(ej ).
(c) Find the impulse response sequence, h(n).
(f) Find the steady-state response to x(n) =

(g) Find the steady-state response to x(n) =
SECTION 15.7
n 6 .
3
sin 4 n + 3 .

cos2 6 n 4 .

sin 3 n u(n).
(d) Find the steady-state response to x(n) = cos

(e) Find the steady-state response to x(n) =
409
PROBLEMS
Problem 15.13 Which of the following filters is realizable?

z
, |z| > 1/2.
(a) H(z) =
(z 1/2)(z 1/3)
z
(b) H(z) =
, 1/3 < |z| < 2.
(z 2)(z 1/3)
(c) Causal LTI system described by y(n) y(n 1) + 14 y(n 2) = x(n).
Problem 15.14 Which of the following filters is realizable?

z
(a) H(z) =
, |z| > 1/4.
(z + 1/4)(z 1/8)2
(b) H(z) =
z 10
, 1/2 < |z| < 4.
(z + 1/2)(z 4)
(c) Causal LTI system described by y(n) 34 y(n 1) + 18 y(n 2) = x(n) x(n 1).
Problem 15.15 Consider the LTI system whose impulse response sequence is given by h(n) =
(n 10) + (n + 10). Does it satisfy the Paley-Wiener condition?
Problem 15.16 Consider the LTI system whose frequency response is H(ej = 1 + e2j . Does
it satisfy the Paley-Wiener condition?
Problem 15.17 Given the four pole-zero distributions shown in Fig. 15.17, which ones correspond
to low-pass, high-pass, band-pass, or band-stop filters?
(a)
(b)
(c)
(d)
FIGURE 15.17 Four pole-zero distributions for Prob. 15.17.
Problem 15.18 Given the four pole-zero distributions shown in Fig. 15.18, which ones correspond
to low-pass, high-pass, band-pass, or band-stop filters?
Problem 15.19 Consider the frequency response
(
j
H(e ) =
j,
j,
0<
< < 0
410
CHAPTER 15
FREQUENCY
RESPONSE
(b)
(a)
(c)
(d)
FIGURE 15.18 Four pole-zero distributions for Prob. 15.18.
which corresponds to a 90o phase shifter. Show that the corresponding impulse response sequence
is given by
(
2/n,
n 6= 0
h(n) =
0,
otherwise
Is the phase shifter a stable system?
Problem 15.20 Consider an arbitrary phase shifter
(
j
H(e ) =
1 ,
2 ,
0<
< < 0
where 1 and 2 denote phase angles in radians. Find the corresponding impulse response sequence.
Problem 15.21 Find the impulse response sequence of the ideal high-pass filter (15.23).
Problem 15.22 Find the impulse response sequence of the ideal band-pass filter (15.24).
Problem 15.23 Find the impulse response sequence of the ideal band-stop filter (15.25).
Problem 15.24 Let h(n) denote the impulse response sequence of a low-pass filter with frequency
response H(ej ). Define h (n) = (1)n h(n).
(a) Verify that H (ej ) = H(ej() ).
(b) Conclude that the filter with impulse response sequence h (n) is of the high-pass type.
Problem 15.25 Find the frequencies that are present in the sequences x(3n), x(n/2), and x2 (n)
when x(n) = cos(o n).
Problem 15.26 Find the frequencies that are present in the sequences x(3n), x(n/2), and x2 (n)
when x(n) = cos2 (o n).
Problem 15.27 Give two examples of frequency responses that would generate the output sequence y(n) = ejn/8 when excited with x(n) = ejn/8 .
Problem 15.28 Find a constant-coefficient difference equation to describe an LTI system whose
impulse response sequence is given by
n1
h(n) =
1
2
What is the frequency response of the system?
sin
n
u(n 2)
Problem 15.29 A stable LTI ARMA system is described by the difference equation
411
SECTION 15.7
y(n) =
M
X
k=1
ak y(n k) +
N
X
k=0
PROBLEMS
bk x(n k)
Let h(n) denote the impulse response sequence of the system. Find the difference equation that
corresponds to the system with impulse response sequence h (n) = (1)n h(n).
Problem 15.30 Find a constant-coefficient difference equation to describe an LTI system whose
frequency response is H(ej ) = 1/ tan .
Problem 15.31 Figure 15.19 shows the interconnection of two LTI systems with frequency responses H(ej ) and G(ej ). The input sequence of the overall system is x(n) and the output
sequence is y(n). Show that the frequency response of the system mapping x(n) to y(n) is given by
G(ej )
1 G(ej )H(ej )
F (ej ) =
y(n)
x(n)
G(ej )
H(ej )
FIGURE 15.19 Feedback interconnection of two LTI systems.
Problem 15.32 Consider the block diagram shown in the top row of Fig. 15.20 and where the
transfer functions H(z) and G(z) are given by
H(z) =
1
z
1
2
G(z) = 1
1 1
z
2
These transfer functions denote stable and causal LTI systems. Let {Y (ej ), X(ej ), E(ej )} denote the DTFTs of the signals indicated in the figure. Let also H(ej ) and G(ej ) denote the
frequency responses of the above systems.
(a) The DTFTs of the signals {x(n), e(n)} are shown in the bottom rows of the same figure.
Compute the energies of these sequences. Compute also the signal-to-noise energy ratio at
the input of the system, which is defined as
SNR =
energy of x(n)
energy of e(n)
(b) Show that

Y (ej ) =
H(ej )G(ej )
1
X(ej ) +
E(ej ) = X (ej ) + E (ej )
1 G(ej )
1 G(ej )
{z
X (ej )
{z
E (ej )
412
CHAPTER 15
e(n)
FREQUENCY
RESPONSE
x(n)
H(z)
G(z)
y(n)
X(ej )
2
1

8 4
(rad/sample)
E(ej )
2
(rad/sample)
where X (ej ) refers to the contribution of the input signal x(n) at the output, while E (ej )
refers to the contribution of the interfering signal e(n) at the output.
(c) Compute the signal-to-noise energy ratio at the output of the system, which is defined as
SNR =
energy of x (n)
energy of e (n)
(d) Assume instead that

x(n) = cos

n ,
e(n) = sin
n+

6
Compute the steady-state response yss (n).

Problem 15.33 Consider an LTI system with a real-valued impulse response sequence h(n). Let
H(ej ) denote the frequency response of the system with real and imaginary parts HR (ej ) and
HI (ej ), respectively. Show that HR (ej ) is the DTFT of the even part of h(n), i.e.,
he (n)
HR (ej )
Problem 15.34 Consider an LTI system with a real-valued and causal impulse response sequence
h(n). The real-part of its frequency response is given by
HR (ej ) = 1 + cos() sin()
Determine h(n) and the complete frequency response.
Problem 15.35 Consider a stable and causal LTI system with a real-valued impulse response sequence, h(n). Let H(ej ) denote the frequency response of the system with real and imaginary
parts HR (ej ) and HI (ej ), respectively.
(a) Argue that H(ej ) is completely determined from knowledge of its real-part alone.
(b) Show that HI (ej ) is related to HR (ej ) through the following convolution relation:
HI (ej ) =
1
2
HR (ej )
d
tan
2
which has the form of a Hilbert transform.

Problem 15.36 Consider an FIR filter with a real-valued impulse response sequence h(n) of length
L, i.e., the nonzero samples of h(n) extend over the interval 0 n L 1. Assume L is odd and
that h(n) is anti-symmetric, namely, its samples satisfy
h(n) = h(L 1 n)
Show that the frequency response must satisfy
Z
H(ej )d = 0
Problem 15.37 A sixth-order comb filter is described by the constant-coefficient difference equation
y(n) = 6 y(n 6) + x(n) + x(n 6)
(a) Determine the filter transfer function.
(b) Determine the location of the zeros and poles of the filter.
(c) Plot the frequency response of the filter.
Problem 15.38 Find and plot the frequency response of the moving average system with exponential weighting
M
1 X k
y(n) =
x(n k)
M + 1 k=0
where 0 < < 1.
h(n) = 2n1 [u(n) u(n M )]
(a) Find its DTFT in two different ways: using the definition and the properties of the DTFT.
(b) What are the values of the integral expressions shown below?
Z
0
|H(ej )|2 d
and
H(ej )d
(c) Find the difference equation of a causal LTI system with impulse-response sequence h(n) and
draw a block diagram representation for it. Under what conditions on will the system be
stable?
(d) Find the steady-state response of the system to the input sequence
x(n) =
when =

1

u(n)
cos
n+
2
3
4
1 ej/6 .
2
(e) Plot the inverse DTFT of H(ej ) cos(M ) for = 1/2.
413
SECTION 15.7
PROBLEMS
414
CHAPTER 15
FREQUENCY
RESPONSE
Problem 15.40 The signal x(n) of Prob. 14.21 is fed into the system shown in Fig. 15.21. The
signal is first low-pass filtered through an ideal filter with cutoff frequency at /8 radians/sample.
The result is then modulated by cos( 8 n) and further scaled by 2(1)n . The display is supposed to
show the energy of the resulting sequence. What value would it show?
cos( n
8 )
x(n)
lowpass
filter
2(1)n
energy
measurement
display
Problem 15.41 Consider an LTI system with impulse response sequence given by
h(n) =
cos[(1 2 )n] cos[(1 + 2 )n]

2 2 n2
where 1 = 3/4 and 2 = /2, both measured in radians/sample.

(b) Determine and plot the frequency response of the system.
(c) Evaluate the following expressions:
(c.1)
(c.2)
n= (1)
n=
h(n)
)
h(n) cos( n
4
(d) Find the steady-state response of the system to the input sequence
x(n) =

1

sin
n+
4
2
4
(e) Plot the frequency response of the LTI system whose impulse response sequence is h(2n+1).
(f) Plot the DTFT of the response of the system when the input sequence is
x(n) =

(1)n
sin
n cos
n
n
3
4
CHAPTER
16
Minimum and Linear Phase Systems
In this chapter we invoke the frequency response characterization of LTI systems in order
to provide further insight into the behavior of such systems. We also introduce important
subclasses of stable LTI systems: linear phase systems, minimum-phase systems, and allpass systems. These subclasses of systems play an important role in the analysis, design,
and implementation of discrete-time filters.
16.1 GROUP DELAY

We start our exposition by defining the group delay of a stable LTI system. The group
delay is measured in units of time and is defined as the negative derivative of the phase
response of the system with respect to the angular frequency, , namely,
() =
dH(ej )
d
(16.1)
The group delay is a function o the angular frequency and we use the notation () to refer
to it. To illustrate the concept, consider an LTI system whose phase response depends in
an affine manner on , i.e.,
H(ej ) = a + b
(16.2)
for some constants a and b. Applying the definition (16.1), we see that the group delay in
this case assumes a constant value and is given by
() = a
(16.3)

h(n) = n u(n),
|| < 1
We already know from Example 15.2 that the magnitude and phase responses are given by
|H(ej )|
H(ej )
1
1 + 2 2 cos

sin
arctan
1 cos
415
Copyright
416
Differentiating the phase component with respect to leads to the group delay
CHAPTER 16
() =
cos 2
1 + 2 2 cos
Figure 16.1 shows the magnitude, phase, and group delay plots for the case = 1/2.
magnitude plot
phase plot
0.5
1.5
H(e )
|H(ej)|
0.5
3 2
1
0
1
2
(rad/sample)
0.5
3 2
1
0
1
2
(rad/sample)
group delay
1
0
1
2
(rad/sample)
0.5
()
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
3 2
FIGURE 16.1 Magnitude (top left), phase (top right), and group delay (bottom) plots for the
exponential sequence h(n) = (0.5)n u(n).

Consider the LTI system whose impulse response sequence, h(n), is the rectangular pulse
(
h(n) =
1,
0,
0nL1
otherwise
(16.4)
We know from the discussion in Example 13.5, and from Table 13.1, that its frequency response is
given by
8
< L,
=0
(16.5)
H(ej ) =
j(L1)/2 sin (L/2)
,
otherwise
.
: e
sin (/2)
Recall further that the phase of H(ej ) is given by (L 1)/2 up to a correction factor that depends on the sign of the ratio sin(L/2)/ sin(/2). No correction is needed when the ratio is positive, and a correction of is used when the sign is negative. Whether we use or in the latter
case is not relevant except to guarantee that the resulting value for the phase lies between [, ]
in accordance with our convention for plotting phase graphs (recall Example 13-3). Figures 16.2
and 16.3 illustrate these results for the case L = 5. Observe from the phase plot in the latter figure
that, in this example, a correction of + is made whenever the ratio sin(L/2)/ sin(/2) changes
sign.
417
SECTION 16.1
h(n)
GROUP
DELAY
FIGURE 16.2 A plot of the rectangular pulse h(n) with width L = 5.
phase plot
3
2
H(e )
3
2
1
0
1
2
1
0
1
2
(rad/sample)
3
3
sin( L/2)/sin(/2)
0
3
|H(e )|
magnitude plot
5
1
0
1
2
(rad/sample)
1
0
1
2
(rad/sample)
4.5
3
1.5
0
1.5
3
FIGURE 16.3 A plot of the magnitude (top left) and phase (top right) of the DTFT of the
rectangular pulse of width L = 5. The bottom right plot shows the variation in the sign of the
ratio sin(L/2)/ sin(/2) over [, ] . Observe that whenever this ratio changes sign (from
positive to negative or from negative to positive), a factor of is added to the phase plot.
We may express the phase of H(ej ) in the form

H(ej ) =
(L 1)
+ I()
2
where I() is the indicator function defined as follows:

(
I() =
0,
1,
when sin (L/2) / sin (/2) 0

otherwise
From the plot of sin(L/2)/ sin(/2) we see that I() is zero everywhere except when the ratio
is negative, where it will have the form of two rectangular pulses. It is evident from the definition
of I() and from Fig. 16.3 that the transitions in I() between the levels of 0 and 1 occur at the
angular frequencies where H(ej ) = 0. The derivative of I() with respect to is zero at all points
418
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
except at the frequencies where the vertical transitions occur. We shall ignore these transition
frequencies since they correspond to values where H(ej ) = 0. It then follows that the group delay
of the system under consideration is given by
() =
L1
= constant
2
(16.6)
16.2 LINEAR PHASE CHARACTERISTICS

We now provide a physical interpretation for the group delay of a system. To begin with,
the group delay, (), of an LTI system generally varies with frequency. The value of ()
at any particular angular frequency o provides a measure of the amount of delay that a
complex exponential signal at frequency o will undergo as it passes through the system
(assuming ejo n is not annihilated by H(ej ). This interpretation is straightforward to
examine in the case of systems with linear phase characteristics. Thus, consider a stable
LTI system with frequency response H(ej ) and assume its phase response is linear in ,
say, of the form
H(ej ) = no
(16.7)
for some no > 0. The corresponding group delay in this case will be
() = no = a constant
(16.8)
x(n) = ejo n
(16.9)
Now choose the input sequence

and apply it to the system. Then, according to the discussion in Sec. 15.2, the resulting
output sequence will be
y(n)
= ejo n H(ejo )
jo
= ejo n |H(ejo )| ejH(e
= ejo n |H(ejo )| ejno o

= ejo (nno ) |H(ejo )|
(16.10)
We see that, apart from scaling by |H(ejo )|, the input sequence ejo n appears at the
output of the system delayed by no units of time (which is the value of the group delay).
Observe further that if o were such that H(ejo ) = 0, then y(n) = 0 fr all n and we say
that the input sequence ejo n has been annihilated by the system. The same conclusion
will hold for any other choice of the angular frequency in this case.

Consider the LTI system of Example 16.2 with frequency response
H(ej ) =
8
< L,
: ej(L1)/2 .
=0
sin (L/2)
,
sin (/2)
otherwise
419
We already know that its group delay is given by
SECTION 16.2
() =
L1
= constant
2
LINEAR
PHASE
CHARACTERISTICS
Let us now select a nonzero angular frequency, o , where H(ejo ) 6= 0 (for example, select an o
o L/2)
> 0). Then, using (16.10), the output sequence that results in response to the
for which sin(
sin(o /2)
input sequence x(n) = ejo n will be
y(n)
ejo (n
L1
2
ejo (n
L1
2
) |H(ejo )|

) sin (o L/2)
sin (o /2)
Example 16.4 (Frequency-dependent group delay)

When the group delay of an LTI system varies with , then exponential sequences of different angular
frequencies will undergo different delays when they are processed by the system. Consider, for
example, the LTI system whose frequency response is illustrated in Fig. 16.4. The phase response
of the system consists of two linear regions with different slopes. One of the regions extends over
[a , a ], while the other region extends over [b , a ] [a , b ]. In the region
[a , a ], the phase variation is linear and passes through the origin so that it takes the form
H(ej ) = no ,
[a , a ]
where no > 0. Over the region [a , b ], the phase variation is affine in and takes the form
H(ej ) = n1 + b,
[a , b ]
for some nonzero constant b (since the line does not pass through the origin) and where n1 > 0.
Likewise, over the region [b , a ] we have
H(ej ) = n1 + b ,
[b , a ]
for some b 6= b. It follows from the expressions for H(ej ) that the group delay of the system has
two level values at no and n1 :
(
() =
no ,
n1 ,
[a , a ]
[b , a ] [a , b ]
Now choose an input sequence that is a combination of two exponential sequences, say,
x(n) = ejo n + ej1 n
(16.11)
where the angular frequency o lies within the range of frequencies where the slope is no , while
the angular frequency 1 lies within the range of frequencies where the slope is n1 (see Fig. 16.4).
Then, by linearity, and using (16.10), the response sequence will be a combination of the form
y(n) = Aejo (nno ) + Aejb ej1 (nn1 )
(16.12)
If the group delay were a constant for all , say, () = no , then the output sequence would have
been
h
i
y(n) = A ejo (nno ) + ej1 (nno )
In this second case, the exponential sequences would be delayed by the same amount and the output
sequence would consequently be a delayed version of the input sequence (16.11). However, when
420
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
|H(ej )|
A
(rad/sample)
(rad/sample)
H(ej )
b a
slope no
slope n1
()
no
n1
FIGURE 16.4
(rad/sample)
The magnitude and phase responses of an LTI system for Example 16.4.
the group delay is dependent on , as is the case with the situation illustrated in Fig. 16.4, then
the exponential sequences at the input will be delayed differently and the output sequence will be
a distorted version (and not simply a delayed version) of the input sequence, as shown by (16.12).
We see from this example that it is generally preferable for a system to have a constant group delay
(or, equivalently, linear phase characteristics). In the next section we examine in greater detail which
classes of FIR filters possess linear phase properties.
Example 16.5 (LTI systems with nonlinear phase characteristics)

Consider now a stable LTI system whose phase response exhibits some nonlinear dependency on .
For simplicity of notation, let us denote the phase response by () instead of H(ej ). We pick
some angular frequency o and linearize the phase response in the proximity of o , by means of a
Taylor series expansion, say, as:

() (o ) +
d
( o )
d =o
(16.13)
421
Using the definition of the group delay, we can write instead
SECTION 16.3
()
(o ) (o ) ( o )
(o ) + b
(16.14)
for some constant b that aggregates the other terms that depend on o . The linear approximation
(16.14) of the phase response is valid for values of that are sufficiently close to o . For this reason,
we can still interpret the group delay (o ) as the delay that an exponential sequence, ejo n , will
undergo when it passes through the system, i.e., x(n) = ejo n is transformed approximately to
y(n) |H(ejo )| ejb ejo (n (o ))
(16.15)
16.3 LINEAR PHASE FIR FILTERS

The LTI system studied in Example 16.2, with an h(n) that is described by a rectangular
pulse, is a finite-impulse response (FIR) filter. It was seen in that example that the filters
frequency response exhibits linear phase characteristics. More general FIR filters, other
than rectangular pulses, can also deliver (piecewise) linear phase characteristics as long
as their impulse response sequences satisfy certain symmetry properties. To motivate the
discussion, we start with a simple example.
Example 16.6 (A second-order FIR filter)

Consider a second-order FIR filter with transfer function
H(z) = 1 + 2z 1 + z 2
The corresponding impulse response sequence has duration L = 3 (and odd number of samples) and
is given by
h(n) = (n) + 2(n 1) + (n 2)
Note that the sequence h(n) is symmetric about n = 1. A sequence of algebraic manipulations
allows us to write the frequency response of the filter as follows:
H(ej )
1 + 2ej + ej2
(1 + ej2 ) + 2ej
ej (ej + ej ) + 2ej
2ej cos() + 2ej
2ej [1 + cos()]
Consequently, the associated magnitude and phase responses are

|H(ej )| = 2 |1 + cos() |
and
H(ej ) =
since 1 + cos() 0 for any . We therefore find that the phase response is linear, as illustrated in
Fig. 16.5.
LINEAR
PHASE
FIR
FILTERS
422
magnitude plot
CHAPTER 16
phase plot
3
2
H(ej)
3
|H(ej)|
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
2
1
0
FIGURE 16.5
1
0
1
2
3
0
2
(rad/sample)
0
2
(rad/sample)
The magnitude and phase responses of the FIR filter H(z) = 1 + z 1 + z 2 .
The argument used in Example 16.6 can be extended to more general FIR filters with symmetry properties in order to bring forth their linear phase properties. There are four types
of FIR filters to consider depending on the type of symmetry they exhibit and on the length
of their impulse response sequences (whether even or odd).
16.3.1 Type-I FIR Filters

Type-I FIR filters are characterized by causal and symmetric real-valued impulse response
sequences with an odd number of samples, L, i.e., h(n) satisfies
h(n) = h(L 1 n),
0 n L 1,
L odd
(Type I)
(16.16)
As illustrated in Fig. 16.6, the samples of h(n) for type-I filters are symmetric about the
point n = (L 1)/2. Let, for simplicity, Ls = (L 1)/2 denote the index of the point of
symmetry. Then the samples of h(n) that occur between 0 n Ls 1 coincide with
the samples that occur between Ls + 1 n L 1. We may therefore re-express the
symmetry property (16.16) in the equivalent form:
h(Ls 1 m) = h (m + Ls + 1) ,
for 0 m Ls 1
(16.17)
It can be verified that the frequency response of a type-I FIR filter takes the form:
H(ej ) = ej(L1)/2 ()
(Type I)
(16.18)
where () is a real-valued function given by
()
L1
2
g(0) =
g(m) =
g(m) cos(m)
m=0

h L1
2
2h m +
(16.19)

L1
2
, m = 1, 2, . . . , L1
2
so that the phase response of the filter is piece-wise linear and characterized by:
H(ej ) =
L1
2 ,
L1
2 ,
when () 0
when () < 0
(16.20)
423
SECTION 16.3
Type I FIR filter

L: number of samples
L odd
h(n) symmetric
n=0
n=L1
n=
L1
2
FIGURE 16.6 For a type-I FIR filter, the samples of h(n) are symmetric about n = (L 1)/2
and the number of samples, L, is odd. The samples to the left of the symmetry point n = (L 1)/2
coincide with the samples to the right of the symmetry point. The rectangular boxes are meant to
indicate the location of the samples and the dotted lines illustrate which samples relate to each other.
Proof: The frequency response of the filter (16.16) is given by:
H(ej )
LX
s 1
h(n)ejn + h (Ls ) ejLs +
n=0
L1
X
h(n)ejn
n=Ls +1
Introduce the change of variables m = n Ls 1 and apply it to the rightmost term of the above
expression. Recalling that L = 2Ls + 1, this step gives
H(ej )
LX
s 1
n=0
LX
s 1
h(m + Ls + 1)ej(m+Ls +1)
m=0
Using (16.17) we get

H(ej )
LX
s 1
n=0
LX
s 1
m=0
h(Ls 1 m)ej(m+Ls +1)
We now introdue another change of variables, n = Ls 1 m, and apply it to the last term to get
H(ej )
LX
s 1
n=0
LX
s 1
n=0
h(n)ej(n2Ls )
LINEAR
PHASE
FIR
FILTERS
424
It follows that
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
H(ej )
h (Ls ) ejLs +
LX
s 1
h(n) ejn + ej(n2Ls )
n=0
jLs
jLs
jLs
jLs
jLs
jLs
h (Ls ) + e
h (Ls ) +
jLs
LX
s 1
LX
s 1
h(n) e
jn
n=0
h(n) e
j(nLs )
n=0
h (Ls ) + 2
LX
s 1
n=0
h (Ls ) + 2
LX
s 1
n=0
h (Ls ) + 2
Ls
X
m=1
h (Ls ) + 2
Ls
X
+ e
j(n2Ls )
+ e
j(nLs )
!
i
h(n) cos[(n Ls )]
h(n) cos[(Ls n)]
h(Ls m) cos(m) ,
using m = Ls n
h(m + Ls ) cos(m) ,
using (16.16) (16.17)
m=1
If we now introduce the sequence

g(0) = h(Ls ),
g(m) = 2h(m + Ls ),
m = 1, 2, . . . , Ls
then we can write

H(ej )
ejLs
Ls
X
g(m) cos(m)
m=0
as desired.
Example 16.7 (Type-I filter)

The result of Example 16.6 is a special case of (16.18)(16.19) with
L = 3,
g(0) = 2,
g(1) = 2,
() = 2 + 2 cos
Moreover, in this case, () 0 for all and (16.20) trivializes to H(ej ) = .
16.3.2 Type-II FIR Filters

Type-II FIR filters are characterized by causal and symmetric real-valued impulse response
sequences with an even number of samples, L, i.e., h(n) satisfies
h(n) = h(L 1 n),
0 n L 1,
L even
(Type II)
(16.21)
In this case, as illustrated in Fig. 16.7, the samples of h(n) for type-II filters are symmetric
about the fractional point (L 1)/2. Let now Ls denote the integer value Ls = L/2.
Then the samples of h(n) that occur between 0 n Ls 1 coincide with the samples
that occur between Ls n L 1. We can therefore re-express the symmetry property

(16.21) in the equivalent form
h(Ls 1 m) = h (m + Ls ) ,
for 0 m Ls 1
(16.22)
Type II FIR filter

L even
h(n) symmetric
n=0
n=L1
L1
2
(fractional)
FIGURE 16.7 For a type-II FIR filter, the samples of h(n) are symmetric about the fractional
point (L 1)/2 and the number of samples, L, is even. The samples to the left of the symmetry
point coincide with the samples to the right of the symmetry point. The rectangular boxes are meant
to indicate the location of the samples and the dotted lines illustrate which samples relate to each
other.
As before, it can be verified (see Prob. 16.38) that the frequency response of a type-II FIR
filter takes the form:
H(ej ) = ej(L1)/2 ()
(Type II)
(16.23)
where () is now the real-valued function defined by
()
L/2
P

m 12

1 , m = 1, 2, . . . , L2
g(m) cos
m=1
g(m) = 2h m +
L
2
(16.24)
so that the phase response of the filter is piecewise linear and characterized by:
H(ej ) =
L1
2 ,
L1
2 ,
when () 0
when () < 0
(16.25)
425
SECTION 16.3
LINEAR
PHASE
FIR
FILTERS
426
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
Example 16.8 (Type-II filter)

Consider the FIR filter with transfer function
H(z) = 1 + 2z 1 + 2z 2 + z 3
The samples of the impulse response sequence satisfy the type-II condition (16.21) with L = 4.
Therefore, according to (16.23)(16.24), the corresponding frequency response is given by
3
H(ej ) = ej 2 ()
where
() = 4 cos

+ 2 cos
3
2
16.3.3 Type-III FIR Filters
Type-III FIR filters are characterized by causal and anti-symmetric real-valued impulse
response sequences with an odd number of samples, L, i.e., h(n) satisfies
h(n) = h(L 1 n),
0 n L 1,
L odd
(Type III)
(16.26)
As illustrated in Fig. 16.8, the samples of h(n) for type-III filters are anti-symmetric about
the point n = (L 1)/2 and the sample at n = (L 1)/2 must be zero. This is because
property (16.26) requires

L1
L1
h
= h
2
2
which is only possible when h(L 1/2) =. Let Ls = (L 1)/2 denote the index of
the point of symmetry. Then, we can express the anti-symmetry property in the equivalent
form:

h(Ls 1 m) = h (m + Ls + 1) , for 0 m Ls 1
(16.27)
h(Ls ) = 0
It can be verified (see Prob. 16.39) that the frequency response of a type-III FIR filter
has the form
H(ej ) = jej(L1)/2 ()
(Type III)
(16.28)
where () is a real-valued function defined by
()
(L1)/2
P
g(m) sin(m)

2h m + L1
, m = 1, 2, . . . , L1
2
2
m=1
g(m) =
(16.29)
H(ej ) =
L1
2 2 ,
L1
3
2 2 ,
when () 0
when () < 0
(16.30)
427
SECTION 16.3
LINEAR
PHASE
FIR
FILTERS
Type III FIR filter

L odd
h(n) anti-symmetric
n=L1
n=0
n=
L1
2
FIGURE 16.8 For a type-III FIR filter, the samples of h(n) are anti-symmetric about n = (L
1)/2 and the number of samples, L, is even. The samples to the left of the symmetry point n =
(L 1)/2 are the opposite of the samples to the right of the symmetry point. The sample at time
n = 0 is necessarily zero. The rectangular boxes are meant to illustrate the location of the samples
and the dotted lines indicate which samples relate to each other.
Example 16.9 (Type-III filter)

H(z) = 1 z 2
The samples of the impulse response sequence satisfy the type-III condition (16.26) with L = 3.
H(ej ) = ej ()
where
() = 2 sin()
16.3.4 Type-IV FIR Filters

Type-IV FIR filters are characterized by causal and anti-symmetric real-valued impulse
response sequences with an even number of samples, L, i.e., h(n) satisfies
h(n) = h(L 1 n),
0 n L 1,
L even
(Type IV)
(16.31)
As illustrated in Fig. 16.9, the samples of h(n) for type-IV filters are symmetric about
the fractional point (L 1)/2. Let Ls = L/2. Then, we can express the anti-symmetry
property in the equivalent form:
h(Ls 1 m) = h (m + Ls ) ,
for 0 m Ls 1
(16.32)
428
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
Type IV FIR filter

L even
h(n) anti-symmetric
n=L1
n=0
L1
2
(fractional)
FIGURE 16.9
For a type-IV FIR filter, the samples of h(n) are anti-symmetric about the
fractional point (L 1)/2 and the number of samples, L, is even. The samples to the left of the
symmetry point relate to the samples to the right of the symmetry point. The rectangular boxes are
meant to illustrate the location of the samples and the dotted lines indicate which samples relate to
each other.
It can be verified (see Prob. 16.40) that the frequency response of a type-IV FIR filter has
the form:
H(ej ) = jej(L1)/2 ()
(Type IV)
(16.33)
where () is the real-valued function defined by
()
L/2
P
m=1
g(m) =

m 12

1 , m = 1, 2, . . . , L2
g(m) sin
2h m +
L
2
(16.34)
H(e ) =
3
2
L1
2
L1
2
when () 0
(16.35)
when () < 0
Table 16.1 summarizes the results concerning the frequency responses of FIR filters of
types I, II, III, and IV.
429
TABLE 16.1 Types I through IV FIR filters with piecewise linear phase characteristics. The
impulse response sequence of each filter has samples over the range 0 n L 1.
Type
II
III
IV
Number of
h(n)
samples L
real-valued
odd
even
odd
even
H (ej )
e
symmetric
symmetric
anti-symmetric
anti-symmetric
j L1
2
j L1
2
() =
L1
2
g(m) cos(m)
g(0) = h
g(m) = 2h m +
() =
()
j ej
(L1)/2
P
m=0
L1
2
()
L1
2
L/2
P
L1
2
g(m) cos
() =
()
(L1)/2
P
L
2
L/2
P
g(m) sin(m)
g(m) = 2h m +
() =

1
2
m=1
()
m
m=1
g(m) = 2h m +
, m 6= 0
L1
2
g(m) sin
m=1
g(m) = 2h m +
L
2
m
1
2

Example 16.10 (Type-IV filter)

H(z) = 1 + 2z 1 2z 1 z 3
The samples of the impulse response sequence satisfy the type-IV condition (16.31) with L = 4.
3
H(ej ) = ej 2 ()
() = 4 sin

LINEAR
PHASE
FIR
FILTERS
()
j ej
where
SECTION 16.3
2 sin
3
2
430
Example 16.11 (FIR filters with piecewise linear phase characteristics)
Figures 16.1016.12 plot the impulse response sequences, magnitude responses, and phase responses,
of four FIR filters of types I, II, III, and IV. The impulse response sequences of the type I and III filters have 5 samples each, while the impulse response sequences of the type II and IV filters have 4
samples each. Observe the piecewise linear phase characteristics of the filters, as expected from the
prior results and analysis.
Type I (L=5)
Type II (L=4)
0.5
h(n)
0.75
h(n)
0.5
0.25
0.25
0
2
3
n
Type III (L=5)
0.5
0.5
0.25
0.25
0
0.25
0.5
FIGURE 16.10
h(n)
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
h(n)
CHAPTER 16
2
n
Type IV (L=4)
0
0.25
2
n
0.5
Examples of impulse response sequences of types I, II, III, and IV FIR filters.
2.4
SECTION 16.3
1.6
1.6
LINEAR
PHASE
FIR
FILTERS
0.8
0
3
1
0
1
2
(rad/sample)
Type III (L=5)
0.8
0
3
2.4
1.6
1.6
|H(ej)|
2.4
|H(e )|
431
Type II (L=4)
2.4
|H(ej)|
|H(e )|
Type I (L=5)
0.8
0
3
1
0
1
2
(rad/sample)
1
0
1
2
(rad/sample)
Type IV (L=4)
1
0
1
2
(rad/sample)
0.8
0
3
FIGURE 16.11 Magnitude responses of the FIR filters of Fig. 16.10 over the range [, ].
Observe how the magnitude responses of the filters of types III and IV are zero at = 0. Observe
also how the magnitude responses of the filters of types II and III are zero at = . These are
general properties and are proven in the sequel when the location of the zeros of FIR filters of types
I through IV are discussed.
16.3.5 Location of Zeros

The locations of the zeros of FIR filters of types I, II, III, and IV exhibit certain symmetry
propertiesas well. To begin with, for FIR filters of types I and II, we know from (16.16)
and (16.21) that
h(n) = h(L 1 n), 0 n L 1 (types I and II)
(16.36)
It follows that the transfer functions of these types of FIR filters can be written as
H(z) =
=
h(0) + h(1)z 1 + . . . + h(L 2)z (L2) + h(L 1)z (L1)
h(L 1) + h(L 2)z 1 + . . . + h(1)z (L2) + h(0)z (L1)
where the second expression can be readily identified as z (L1)H(z 1 ). Therefore, for
FIR filters of types I and II, it holds that their transfer functions satisfy the relation:
H(z) = z (L1) H(z 1 )
(types I and II)
(16.37)
Similarly, for FIR filters of types III and IV, we can verify that
H(z) = z (L1) H(z 1 )
(types III and IV)
(16.38)
It follows from the relations (16.37) and (16.38) that if z = zo is a zero of H(z) then
z = 1/zo is also a zero of the same transfer function. In other words, the zeros of H(z)
432
Type I (L=5)
Type II (L=4)
3
H(e )
0
1
2
0
1
2
3
3
1
0
1
2
(rad/sample)
Type III (L=5)
3
3
H(e )
H(ej)
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
H(ej)
CHAPTER 16
0
1
1
0
1
2
(rad/sample)
Type IV (L=4)
1
0
1
2
(rad/sample)
0
1
2
2
3
3
1
0
1
2
(rad/sample)
3
3
FIGURE 16.12 Phase responses of the FIR filters of Fig. 16.10 over the range [, ].
Observe how the filters exhibit piecewise linear phase characteristics.
occur in reciprocal pairs. Moreover, since h(n) is real-valued, if z = zo is a complexvalued zero of H(z), then its complex conjugate point, z = zo , should also be a zero of
H(z). This is because H(z) is a polynomial in z 1 and it is well-known that the roots of
polynomials with real coefficients occur in complex conjugate pairs. Consequently, for a
complex-valued zero z = zo , the points z = 1/zo , z = zo , and z = 1/zo are also zeros of
H(z).
As for the occurence of zeros at the special points z = 1, we can invetigate this
possibility by evaluating the relations (16.37) and (16.38) at z = 1. Thus, note from
(16.37) that filters of types I and II satisfy
H(1) = (1)(L1) H(1)
(types I and II)
(16.39)
This equality implies that the point z = 1 must be a zero when L is even because only
then the identify H(1) = H(1) can be satisfied. We therefore conclude that type-II
FIR filters must have zeros at z = 1. Now since 1 = ej , we find that

H(ej )= = 0
(type II)
(16.40)
so that the frequency reponse, H(ej ), of a type-II filter is zero at = . Likewise, we

note from (16.38) that filters of types III and IV satisfy
H(1) = (1)(L1)H(1) (types III and IV)
(16.41)
This equality implies that the point z = 1 must be a zero when L is odd and we conclude
that type-III FIR filter must have zeros at z = 1. Moreover, since 1 = ej , we find that

H(ej )= = 0
(type III)
(16.42)
so that the frequency response of a type-III filter is zero at = . Relation (16.38) also
implies that filters of types III and IV satisfy
H(1) = H(1)
(types III and IV)
(16.43)
so that these filtes must have a zero at z = 1 or, equivalently,

H(ej )=0 = 0
(types III and IV)
(16.44)
The plots in Fig. 16.11 illustrate these properties. Table 16.2 summarizes the results concerning the location of zeros of FIR filters of types I, II, III, and IV. Figure 16.13 illustrates
the typical locations of the zeros of these filters.
TABLE 16.2 Location of zeros of types I through IV FIR filters with linear phase characteristics.
The impulse response sequence of each filter has samples over the range 0 n L 1.
Number of
h(n)
Type
samples L
real-valued
odd
II
z = zo
symmetric
z = +1
?
z = 1
?
even
symmetric
zero
{zo , 1/zo , 1/zo } are also zeros
III
odd
anti-symmetric
zero
zero
IV
even
anti-symmetric
zero
is a general zero
16.4 ALL-PASS SYSTEMS

We move on to describe two other subclasses of stable LTI systems with rational transfer
functions, namely, all-pass systems and minimum-phase systems. It will be seen that the
group delay of minimum-phase systems has a useful property. It will also be seen that general rational and stable LTI systems can always be decomposed into a cascade combination
of all-pass and minimum-phase systems.
An all-pass system is defined as a causal and stable (i.e., realizable) LTI system whose
rational transfer function, H(z), satisfies the condition
H(z) [H(1/z )]
= 1
(16.45)
where the term [H(1/z )] amounts to replacing the argument z by 1/z and conjugating
the result. In particular, if we set z = ej we find that

[H(1/z )] z=ej = H(1/ej ) = H(ej ) H (ej )
(16.46)
so that the frequency response of an all-pass system must satisfy

|H(ej )|2 = 1
(16.47)
433
SECTION 16.4
ALL-PASS
SYSTEMS
434
CHAPTER 16
Im
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
1/zo
zo
1/z1
z1
Re
zo
1/zo
FIGURE 16.13 Location of zeros of FIR filters of types I, II, III, and IV. If zo is a complex zero,
then zo , 1/zo , and 1/zo are also zeros. If z1 is a real zero, then 1/z1 is also a zero. Also, the point
z = 1 is a zero for types III and IV and the point z = 1 is a zero for types II and III.
|H(ej )|
1
FIGURE 16.14
range.
(rad/sample)
The magnitude response of an all-pass LTI system is flat over the entire frequency
We therefore say that an all-pass system has unit magnitude response over the entire frequency range [, ], as illustrated in Fig. 16.14.
The requirement of causality translates into requiring the impulse-response sequence of
the all-pass system to be a right-sided sequence. It follows that the ROC of H(z) must
be the outside of a circular region. Moreover, the requirement of BIBO stability translates
into requiring the ROC of H(z) to include the unit circle see Fig. 16.15. Collecting these
observations together, we conclude that the poles of all-pass systems must lie strictly inside
435
SECTION 16.4
ALL-PASS
SYSTEMS
Im
ROC
Re
FIGURE 16.15 The ROC of all-pass rational transfer functions is the outside of a circular region
that includes the unit circle. All poles must lie strictly inside the unit circle
the unit circle.
Useful Application
One useful application of all-pass systems is the following. Consider a discrete-time system with transfer function G(z) and let us cascade it with an all pass system, H(z), as
shown in Fig. 16.16. The magnitude response of the cascade will coincide with the magnitude response of the original system since
|G(ej ) H(ej )| = |G(ej )| |H(ej )| = |G(ej )|
On the othre hand, the phase response of the cascade is given by

G(ej ) H(ej ) = G(ej ) + H(ej )
(16.48)
(16.49)
We therefore see that cascading G(z) with an all-pass system maintains the magnitude
response of G(z) unchanged but modifies its phase response. This result indicates that
all-pass systems can be used to adjust the phase response and group delay characteristics
of other systems without modifying their magnitude responses. For instance, if a system
G(z) has some undesirable phase response, then cascading it with an all-pass system can
help address this deficiency by adjusting the phase response of the cascade and bringing it
closer to the desired response.
16.4.1 First-Order All-Pass Sections

The simplest examples of all-pass systems are
H(z) = 1
(16.50)
436
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
x(n)
G(z)
H(z)
y(n)
all-pass
FIGURE 16.16 The cascade of a stable LTI system, G(z), with an all-pass system, H(z), has the
same magnitude response as the original system, G(z). In this way, the all-pass system can be used
for phase and group delay compensation.
and
H(z) = z d (pure delays with integers d > 0)
(16.51)
More generally, it follows from the normalization requirement (16.45) that if H(z) has a
pole at z = a, then H(z) must have a zero at z = 1/a so that the condition
H(a)[H(1/a )] = 1
is possible. Therefore, a first-order all-pass rational transfer function must be of the form:
H(z) = ej
z 1 a
j 1 a z
=
e
,
1 az 1
za
|a| < 1, |z| > |a|
(16.52)
where ej is a unit-magnitude scaling complex coefficient; usually, we have = 0. In

this first-order case, the transfer function has a single pole inside the unit circle at the
point z = a, and a single zero outside the unit circle at the point z = 1/a . The case
a = 0 reduces to H(z) = ej z 1 , which has a pole at z = 0 and a zero at z = .
The magnitude response of (16.52) can be easily seen to be one over the entire frequency
range. Indeed,
|H(ej )| =
=
=
=
=
=

1 a ej

|ej | j
e a

j ej a
e

ej a

j ej a

e
ej a
j

e
a

ej a
j

(e a)

ej a
1
since the expressions in the numerator and denominator are complex conjugates of each
other. The phase response of (16.52), on the other hand, can be verified to be
H(ej ) = 2 arctan
r sin( )
1 r cos( )
(16.53)
where we introduced the polar representation of a as a = rej . Figure 16.17 displays the
magnitude and phase responses of the first-order all-pass section (16.52) for a = 1/2 and
= 0.
magnitude plot
phase plot
1.5
H(e )
|H(e )|
2
1
0
2
0.5
3
1
0
1
2
(rad/sample)
1
0
1
2
(rad/sample)
FIGURE 16.17 The magnitude and phase responses of the first-order all-pass section (16.52) for
a = 1/2 and = 0.
The group delay of (16.52) is obtained by differentiating the phase response (16.53)
with respect to to arrive at
() =
1 r2
1 + r2 2r cos( )
(16.54)
Note that since r < 1, it holds that the group delay is necessarily positive at all . Figure 16.18 plots the group delay of the first-oder section (16.52) for = 0 and a = 1/2
(i.e., r = 1/2 and = 0).
group delay
3
()
2.5
2
1.5
1
0.5
3
FIGURE 16.18
0
(rad/sample)
Group delay of the first-order all-pass section (16.52) for a = 1/2 and = 0.
437
SECTION 16.4
ALL-PASS
SYSTEMS
438
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
16.4.2 Second-Order All-Pass Sections

The series cascade of two first-order all-pass sections results in a second-order all-pass
transfer function. If we define
1 a1 z 1 a2 z
, |a|1 < 1, |a|2 < 1, |z| > max{|a1 |, |a2 |}

z a1
z a2
(16.55)
for some complex scalar ej (usually, = 0), then H(z) is also all-pass since

1 a ej 1 a2 ej

= 1
|H(ej )| = |ej | j 1
e a1 ej a2
H(z) = ej
Furthermore, if we expand the product on the right-hand-side of (16.55), it can be seen that
H(z) can be expressed in the alternative form:
H(z) = ej
2 z 2 + 1 z + 1
z 2 + 1 z + 2
(second order)
(16.56)
where the coefficients {1 , 2 } are related to the poles {a1 , a2 } via

1 = (a1 + a2 ),
2 = a1 a2
(16.57)
Expression (16.56) brings to light how the coefficients of the polynomials in the numerator
and denominator of an all-pass system relate to each other. To highlight this fact, let us
introduce the polynomials
A(z) =
A# (z) =
Then it is easy to see that
z 2 + 1 z + 2
2 z 2 + 1 z + 1
(denominator polynomial)
(numerator polynomial)

1
A (z) = z A
z
#
(16.58)
(16.59)
(16.60)
and the coefficients of A# (z) are obtained by conjugating the coefficients of A(z) and
reversing their order. Moreover, it is worth noting that that the two roots of the denominator
polynomial A(z) must lie inside the unit circle since A(z) is the product of two first-order
polynomials:
A(z) = (z a1 )(z a2 )
(16.61)
and both a1 and a2 lie inside the unit circle. Likewise, the roots of the numerator polynomial A# (z) must lie outside the unit circle since A# (z) is the product of the two first-order
polynomials
A# (z) = (1 a1 z)(1 a2 z)
(16.62)
and both 1/a1 and 1/a2 lie outside the unit circle.
439
SECTION 16.4
16.4.3 Higher-Order All-Pass Sections
ALL-PASS
SYSTEMS
More generally, recall again from the normalization requirement (16.45) that if H(z) has
a pole at z = a, then H(z) must have a zero at z = 1/a . Therefore, a rational all-pass
transfer function of order N that satisfies (16.45) must have the form
H(z) = ej
1 aN z
1 a1 z 1 a2 z
...
z a1
z a2
z aN
(16.63)
for some poles {a1 , a2 , . . . , aK }, including poles at zero, and for some complex scalar
ej (usually, = 0). We see that the above H(z) can be obtained by cascading multiple
first-order sections. If we let A(z) denote the denominator polynomial of H(z), then the
same argument used in the previous section will show that H(z) can be written as
H(z) = ej
A# (z)
,
A(z)
ROC = {|z| > }, for some 0 < 1
(16.64)
where A(z) is now a polynomial of degree N in z:

A(z) = z N + 1 z N 1 + 2 z N 2 + . . . + N
(16.65)
with roots inside the unit circle. And A# (z) is the conjugate reversal polynomial of A(z)
given by
A# (z) =
=

1
zN A
z
N z N + . . . + 2 z 2 + 1 z + 1
(16.66)
That is, A# (z) is obtained by conjugating the coefficients of A(z) and reversing their
order. Note again the useful property that if A(z) has a zero at zo , then A# (z) has a zero
at 1/zo . In particular, since all the zeros of A(z) lie inside the unit circle, then the zeros of
A# (z) must lie outside the unit circle.
Moreover, since H(z) in (16.63) is the product of elementary first-order all-pass sections, we conclude that the phase response of H(z) is the sum of the phase responses of
the individual sections, say,
H(ej ) = + H1 (ej ) + H2 (ej ) + . . . + HN (ej )
where each Hk (z) denotes
Hk (z) =
1 ak z
z ak
(16.67)
(16.68)
Accordingly, differentiating both sides of (16.67), we find that the group delay of H(z) is
given by
() = 1 () + 2 () + . . . + N ()
(16.69)
in terms of the individual group delays. We remarked earlier following (16.54) that k () >
0 for all . It follows that general all-pass systems of the form (16.63) have positive group
delays as well:
() > 0
(16.70)
440
Example 16.12 (All-pass section with real h(n))
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
Consider a third-order all-pass system with transfer function H(z) and real-valued impulse response
sequence, h(n). Since h(n) is real-valued, it follows from the complex conjugation property (9.25)
of the ztransform that H(z) should satisfy
H(z) = [H(z )]
Therefore, if H(z) has a pole at z = a, then H(z) has a pole at z = a as well. In other words,
when h(n) is real-valued, the poles of H(z) must occur in complex conjugate pairs. Thus, let
{a1 , a1 , a2 } denote the poles of the third-order transfer function H(z) where {a1 , a1 } correspond
to the conjugate pair and a2 is real. Since H(z) is an all-pass system, then H(z) must have zeros at
the locations {1/a1 , 1/a1 , 1/a2 }. Moreover, all poles of H(z) must lie inside the unit circle. We
conclude that H(z) may be written as
H(z) =
1 a1 z 1 a1 z 1 a2 z
z a1
z a1
z a2
|a1 | < 1,
|a2 | < 1
Figure 16.19 illustrates the conjugate reciprocal symmetry properties of the poles and zeros of the
third-order all-pass system.
Im
1/a1
1/a2
a2
a1
Re
a1
1/a1
FIGURE 16.19 The plot illustrates the conjugate reciprocal symmetry of the poles and zeros of a
third-order all-pass system with a real-valued impulse response sequence.
16.5 MINIMUM PHASE SYSTEMS

The other subclass of systems that we wish to highlight corresponds tothe so-called minimum phase systems. A rational minimum-phase system is a stable and causal LTI system
whose inverse is also a stable and causal LTI system. In other words, a minimum-phase
system is a realizable system whose inverse is also realizable. Accordingly, the inverse of a
minimum-phase system is itself minimum-phase. Let us examine the implications of these
requirements on the pole-zero distribution of minimum-phase systems.
Let H(z) denote the transfer function of a stable and causal LTI system. We know from
Sec. 11.8 that its inverse is defined by the transfer function G(z) that is given by
1
G(z) =
H(z)
(16.71)
and whose ROC is chosen such that the ROCs of both G(z) and H(z) have overlapping
regions. Now, the realizability of H(z) implies that its poles lie inside the unit circle. Likewise, the realizability of G(z) implies that its poles should lie inside the unit circle. But
since the poles of G(z) coincide with the zeros of H(z), we conclude that the poles and
zeros of a minimum-phase system should both lie inside the unit circle.
Characterization of minimum-phase systems. A realizable rational LTI system is
minimum-phase (and therefore admits a realizable inverse) if, and only if, all its poles and
zeros lie inside the unit circle.
Proof: We argued before the statement that if H(z) is realizable and minimum-phase then its zeros
and poles must lie inside the unit circle. Conversely, assume the zeros and poles of a realizable
system lie inside the unit circle and let us show that it has to be minimum-phase (i.e., it must have a
realizable inverse).
To begin with, the realizability of the system H(z) implies that its ROC has the form
ROC = {|z| > } for some 0 < 1
Let G(z) = 1/H(z) denote its inverse. The zeros and poles of G(z) then lie inside the unit circle
as well. Let |zmin | and |zmax | denote the smallest and largest magnitudes of the poles of G(z) (i.e.,
zeros of H(z)). Since, by assumption, all poles of G(z) lie inside the unit circle, we have that
|zmin | < 1 and |zmax | < 1. Then, the ROC of G(z) can be either |z| > |zmax | or |z| < |zmin |.
Only the choice |z| > |zmax | leads to a realizable system G(z) whose ROC overlaps with the ROC
of H(z). We therefore find that the inverse of H(z) is given by
G(z) =
1
,
H(z)
|z| > |zmax |
and this system is realizable since its ROC has the form
ROC = {|z| > } for some 0 < 1
Characterization of the ROCs of minimum-phase systems. The ROCs of a

minimum-phase system and its minimum-phase inverse are of the forms
ROC of H(z) = {|z| > }
ROC of G(z) = {|z| > }
for some 0 < 1

for some 0 < 1
(16.72)
(16.73)
Example 16.13 (Non-minimum-phase system)

The system
(z 0.9)(z 0.8)
, |z| < 0.5
(z 0.5)(z 0.6)
is not minimum phase. Although its poles and zeros lie inside the unit circle, the system is nevertheless unstable since the ROC does not include the unit circle.
H(z) =
441
SECTION 16.5
MINIMUM
PHASE
SYSTEMS
442
Example 16.14 (Pole and zero distribution)
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
Consider the system

(z 0.9)(z 0.8)
,
(z 0.5)(z 0.6)
This system is realizable since its ROC has the form
H(z) =
|z| > 0.6
ROC = {|z| > } for some 0 < 1

where = 0.6. Note further that the system has two zeros at z = 0.9 and z = 0.8, and two poles at
z = 0.5 and z = 0.6. Thus, the zeros and poles of H(z) lie inside the unit circle and, according to
the characterization of minimum-phase systems, we conclude that H(z) is minimum-phase. Let us
now determine its causal and stable inverse.
Note first that the transfer function of the inverse system is given by
G(z) =
(z 0.5)(z 0.6)
(z 0.8)(z 0.9)
The ROC of G(z) can be either |z| > 0.9 or |z| < 0.8. Both possibilities lead to an ROC for G(z)
that overlaps with the ROC of H(z). We therefore have two valid inverse systems in this case:
G1 (z) =
(z 0.5)(z 0.6)
,
(z 0.8)(z 0.9)
|z| > 0.9
G2 (z) =
(z 0.5)(z 0.6)
,
(z 0.8)(z 0.9)
|z| < 0.8
or
However, only G1 (z) is a realizable inverse system. We therefore say that the realizable inverse of
the realizable system H(z) is the system G1 (z).
Example 16.15 (Non-minimum-phase system)

The transfer function
z2
, |z| > 1/4
z 1/4
does not correspond to a minimum-phase system since it has a zero at z = 2, which lies outside the
unit circle.
H(z) =
16.6 FUNDAMENTAL DECOMPOSITION

Not every stable and causal system is all-pass. Likewise, not every stable and causal system
is minimum-phase. However, every stable and causal system can be expressed as the product of a minimum-phase system and an all-pass system. To see this, consider an arbitrary
irreducible rational transfer function of a causal and stable LTI system, say,
H(z) =
N (z)
,
D(z)
ROC = {|z| > } for some 0 < 1
(16.74)
where the roots of the denominator D(z) lie inside the unit circle (by virtue of the causality
and stability assumptions). On the other hand, the roots of N (z) may lie either inside or
outside the unit circle. If all of them lie inside the unit circle, then we would already be
dealing with a minimum-phase system. So assume that N (z) has at least one zero outside
443
the unit circle. We can then factor N (z) as the product of two polynomials, say
SECTION 16.6
N (z) = N1 (z)N2 (z)
(16.75)
with N1 (z) having all its zeros inside the unit circle and N2 (z) having all its zeros outside
the unit circle. Then we can write
H(z) =
N1 (z)N2 (z) N2# (z)

#
D(z)
N2 (z)
(16.76)
which is also equivalent to the decomposition

H(z) =
N2 (z)
N1 (z)N2# (z)
#
D(z)
N (z)
{z
} | 2{z }
|
minimum-phase
(16.77)
all-pass
The transfer function

Hmin (z) =
N1 (z)N2# (z)
,
D(z)
ROC = {|z| > } for some 0 < 1
is minimum-phase since the zeros of its numerator and denominator polynomials all lie
inside the unit circle. Likewise, the transfer function
Hap (z) =
N2 (z)
N2# (z)
ROC = {|z| > } for some 0 < 1
is all-pass since its poles lie inside the unit circle and its magnitude response evaluates to
one over the entire frequency range. More compactly, we write (16.77) as
H(z) = Hmin (z) Hap (z)
(16.78)
Example 16.16 (Decomposition of systems)

Consider the causal and stable LTI system with transfer function
H(z) =
z4
,
z 1/2
|z| > 1/2
Its fundamental decomposition takes the form

H(z) =
1
z4
1 4z 1 4 z
1 4z
z 1/2 1 4z
z 1/2 z 1/4
so that

1 4z
Hmin (z) =
, |z| > 1/2
z 1/2
and
Likewise, consider the alternative causal and stable LTI system with transfer function
H(z) =
1
,
z 1/2
1 14 z
Hap (z) =
, |z| > 1/4
z 1/4
|z| > 1/2
FUNDAMENTAL
DECOMPOSITION
444
Its fundamental decomposition takes the form
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
H(z) =

so that
z
z 1
z 1/2

Hmin (z) =
z
, |z| > 1/2
z 1/2
Hap (z) = z 1 , z 6= 0
and
16.6.1 Minimum Group Delay Property

The result (16.78) reveals that a transfer function H(z) has the same magnitude response
as its minimum-phase component, Hmin (z),
|H(ej )| = |Hmin (ej )|
(16.79)
H(ej ) = Hmin (ej ) + Hap (ej )
(16.80)
Moreover, since
we conclude that the respective group delays satisfy the relation
H () = min () + ap ()
(16.81)
However, we argued earlier (16.70) that all-pass systems have positive group delays so that
ap () > 0. We conclude that
H () > min ()
(16.82)
so that among all systems with the same magnitude response, the minimum-phase system
is the one with the smallest group delay!
Minimum group delay property. Among all realizable LTI rational systems with the
same magnitude response, the minimum-phase system has the smallest group delay.
Example 16.17 (Illustrating the group delay property)

Consider the causal and stable system
H(z) =
1
,
z 1/2
|z| > 1/2
We already know from Example 16.16 that this system is not minimum phase since it includes a zero
at . Its fundamental decomposition is given by
H(z) =
z
z 1/2
|{z}
z 1
| {z }
allpass
minimumphase
Thus, the systems

H(z) =
1
, |z| > 1/2
z 1/2
and
Hmin (z) =
z
,
z 1/2
|z| > 1/2
have the same magnitude response. However, only Hmin (z) is minimum phase. Figure 16.20
compares the magnitude responses of both systems, as well as their group delays, over the interval [, ]. The group delay of Hmin (z) is evaluated by specializing the expression derived
earlier in Example 16.1 for the case = 1/2:
min () =
0.5 cos 0.25

1.25 cos
Likewise, the group delay of H(z) is evaluated by using

H () = min () + ap ()
Thus, note that Hap = z 1 so that
Hap (ej ) =
Differentiating with respect to gives
ap () =
Therefore,
H () =
dHap (ej )
= 1
d
1 0.5 cos
1.25 cos
magnitude plot
1.8
1.8
1.6
1.6
|Hmin(e )|
|H(e )|
magnitude plot
1.4
1.2
1.4
1.2
0.8
0.8
1
0
1
2
(rad/sample)
1
0
1
2
(rad/sample)
group delay
2
H(z)
()
1.5
1
0.5
(z)
min
0
3
0
(rad/sample)
FIGURE 16.20 Magnitude responses of H(z) = 1/(z 0.5) and Hmin (z) = z/(z 0.5) (top)
and their respective group delays (bottom). Observe that the group delay of Hmin (z) is lower than
that of H(z).
16.6.2 Minimum Energy Delay Property

Minimum phase systems exhibit another useful property in relation to all realizable LTI
rational systems with the same magnitude response, namely, they have the smallest energy
delay. To explain what this means, let hmin (n) denote the impulse response sequence of
a minimum phase system Hmin (z) and let h(n) denote the impulse response sequence of
445
SECTION 16.6
FUNDAMENTAL
DECOMPOSITION
446
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
an arbitrary realizable LTI system H(z). We assume that H(z) is obtained from Hmin (z)
by multiplying it by an all-pass transfer function. Then, obviously, both systems have the
same magnitude response,
|Hmin (ej )| = |H(ej )|
(16.83)
Now we want to compare the energies of the corresponding impulse response sequences
and establish that the following result holds:
n
P
k=0
|h(k)|2
n
P
k=0
|hmin (k)|2 ,
for any n 0
(16.84)
The result means that the energy in the samples of hmin () up to time n will always exceed
the energy in the samples of any other possible sequence h() up to the same time instant.
Proof: Let us estbalish first an auxiliary result. Consider a first-order all-pass (and therefore, causal
and stable) section of the form

Hap (z) =
z 1 a
1 a z
=
,
za
1 az 1
|a| < 1,
|z| > |a|
and let x(n) denote an arbitrary causal input sequence and y(n) the corresponding causal output
sequence, as illustrated in the top row of Fig. 16.21. The bottom row of the figure splits the implementation of the all-pass section into the series cascade of two subsections and denotes the output of
1/(1 az 1 ) by the causal sequence w(n).
x(n)
x(n)
z 1 a
1az 1
y(n)
y(n)
w(n)
1
1az 1
z 1 a
FIGURE 16.21 A causal sequence x(n) is fed into an all-pass section (top); the same section
is implemented as the series cascade of two sub-sections (bottom) and the intermediate signal is
denoted by w(n).
Using the ztransform notation we have

W (z) = X(z)
1
1 az 1
and
Y (z) = (z 1 a )W (z)
and
Y (z) = (z 1 a)W (z)
or, equivalently,
X(z) = W (z)(1 az 1 )
These relations express X(z) and Y (z) in terms of the intermediate variable W (z). Transforming
back to the time domain we find that the sequences x(n) and y(n) are related to w(n) through the
relaxed difference equations
x(n)
y(n)
w(n) aw(n 1)
a w(n) + w(n 1)
It follows that
n
X
k=0
|x(k)|2
n
X
k=0
|y(k)|2
n
X
k=0
|w(k) aw(k 1)|2

n
X
(1 |a|2 )
(1 |a|2 ) |w(n)|2
k=0
|a w(k) + w(k 1)|
|w(k)|2 |w(k 1)|2
k=0
n
X
since |a| < 1. We therefore conclude that the partial energies of the input and output sequences (up
to time n) of an all-pass section satisfy the relation
n
X
k=0
|y(k)|2
n
X
k=0
|x(k)|2
(16.85)
Now consider an arbitrary stable and causal system H(z) and introduce its fundamental decomposition
H(z) = Hmin (z)Hap (z)
where Hmin (z) is the minimum-phase component and Hap (z) is an all-pass system. The above
fundamental decomposition can be interpreted as applying the sequence hmin (n) into the all-pass
system, Hap (z), and generating the sequence h(n), as shown in the top row of Fig. 16.22. The
bottom row in the same figure implements the all-pass system as a cascade of elementary first-order
all-pass sections, say Hap,i (n) for i = 1, 2, 3, . . . , m.
hmin (n)
hmin (n)
Hap,1 (z)
w1 (n)
Hap (z)
Hap,2 (z)
w2 (n)
h(n)
Hap,m (z)
h(n)
FIGURE 16.22 The fundamental decomposition can be interpreted as applying the sequence
hmin (n) into the all-pass system and generating the sequence h(n) (top). The bottom figure
implements the all-pass system as a cascade of elementary first-order all-pass sections.
447
SECTION 16.6
FUNDAMENTAL
DECOMPOSITION
448
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
We can now apply the conclusion (16.85) to each elementary all-pass section. Thus, note that
from the first section we get
n
X
k=0
n
X
|w1 (k)|2
k=0
|hmin (k)|2
and from the second section we obtain

n
X
k=0
Consequently,
n
X
k=0
n
X
|w2 (k)|2
k=0
n
X
|w2 (k)|2
k=0
|w1 (k)|2
|hmin (k)|2
Continuing in this manner we arrive at the desired conclusion:

n
X
k=0
n
X
|h(k)|2
k=0
|hmin (k)|2 ,
for any n 0
Example 16.18 (Illustrating the energy delay property)
Consider the causal and stable system

H(z) =
1
,
z 1/2
|z| > 1/2
We already know from Example 16.16 that this system is not minimum phase since it includes a zero
at . Its fundamental decomposition is given by
z
z 1/2
H(z) =
|{z}
z 1
| {z }
allpass
minimumphase
Thus, the systems

H(z) =
1
, |z| > 1/2
z 1/2
Hmin (z) =
and
z
,
z 1/2
|z| > 1/2
have the same magnitude response. However, only Hmin (z) is minimum phase. The corresponding
impulse response sequences are
n1
h(n)
1
2
n
u(n 1),
hmin (n) =
1
2
u(n)
Therefore,
Eh (n)
n
X
k=0
|h(k)|2 =
n k1
X
1
k=1
n1
X
m
1
2
m=0
1 0.5n
= 2(1 0.5n )
1 0.5
Likewise,
Ehmin (n)
n
X
k=0
|hmin (k)|
n k
X
1
k=0
1 0.5n+1
= 2(1 0.5n+1 )
1 0.5
449
and we see that

Ehmin (n) > Eh (n)
as expected.
SECTION 16.8
APPLICATIONS
16.7 APPLICATIONS
TO BE ADDED
Practice Questions:
1.
2.
16.8 PROBLEMS
Problem 16.1 Given H(z) = ej 3 z 4 , find its group delay and the response of the system to
x(n) = ej 3 n + ej 6 n .
Problem 16.2 Given H(ej ) = 1 + e2j , find its group delay and the response of the system to
x(n) = ej 4 n + ej 2 n .
Problem 16.3 Given H(ej ) = cos(), find its group delay and the response of the system to
x(n) = sin 6 n .
Problem 16.4 Given H(ej) = ej + e3j , find its group delay and the response of the system
to x(n) = (1)n + cos 6 n .
Problem 16.5 Given H(z) =

sin2 6 n .
Problem 16.6 Given H(z) =

cos2 6 n + 4 .
z 2
,
z1/3
find its group delay and the response of the system to x(n) =
z2
,
z1/2
find its group delay and the response of the system to x(n) =
Problem 16.7 Classify each of the following FIR filters as type-I, II, III, IV, or none:
(a) H(z) =
1
2
+ 2z 1 + 21 z 2 .
(b) H(z) = 21 + 2z 1 + 12 z 2 .
(c) H(z) =
(d) H(z) =
1
2
1
2
+ 21 z 2 .
+ 2z 1 + 2z 2 + z 3 .
(e) H(z) = 21 + 2z 1 2z 2 + 12 z 3 .
(f) H(z) = 21 + 2z 1 2z 2 12 z 3 .
Problem 16.8 Classify each of the following FIR filters as type-I, II, III, IV, or none:
(a) H(z) =
(b) H(z) =
1
3
1
4
2z 1 + 31 z 2 .
+ z 1 41 z 2 .
(c) H(z) = 1 + z 4 .
(d) H(z) =
1
4
+ 2z 1 2z 2 z 3 .
450
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
(e) H(z) = 31 + 2z 1 + 2z 2 + 13 z 3 .
(f) H(z) = 41 + 2z 1 2z 2 + 14 z 3 .
Problem 16.9 Determine the zero locations for each of the FIR transfer functions in Prob. 16.7.
Problem 16.10 Determine the zero locations for each of the FIR transfer functions in Prob. 16.8.
Problem 16.11 Determine the frequency responses for each of the FIR transfer functions in Prob. 16.7.
Problem 16.12 Determine the frequency responses for each of the FIR transfer functions in Prob. 16.8.
Problem 16.13 Let H(z) denote the transfer function o a type-III FIR filter with one zero at zo =
1 j
e 4 . What is the smallest order H(z) with this property? Find H(z) assuming the energy of its
2
impulse response sequence is normalized to one.
Problem 16.14 Let H(z) denote the transfer function o a type-II FIR filter with one zero at zo =
1 j
e 4 . What is the smallest order H(z) with this property? Find H(z) assuming the energy of its
2
impulse response sequence is normalized to one.
Problem 16.15 Determine a third-order type-IV FIR filter with a zero at zo = 1/2.
Problem 16.16 Determine the value of the smallest order that types-I and III FIR filters can assume
when they have zeros at the locations indicated below:
(a) zo = 1/2.
(b) z = 12 ej 4 .
(c) zo = 1/4, z1 = 2, and z2 = 12 ej 3 .

Problem 16.17 Give transfer functions of two second-order all-pass systems with poles at a1 =
1/2 and a2 = 1/3.
Problem 16.18 Give the expression of a third-order all-pass transfer function with poles at a1 =
1/2, a2 = 1/3, and a3 = 1/4. Find its group delay. Find also the associated third-order polynomials
A3 (z) and A#
3 (z).
Problem 16.19 Determine a difference equation describing a third-order all-pass transfer function
with poles at a1 = 1/2, a2,3 = 31 (1 j).
Problem 16.20 Give the expression of a fourth-order all-pass transfer function with poles at a1 =
1/2, a2 = 1/3, and a3,4 = 12 (1 j). Find its group delay. Find also the associated fourth-order
polynomials A3 (z) and A#
3 (z).
Problem 16.21 Let
H(z) =
(a) Determine its group delay, ().
1 + 14 z 1 + 13 j 1 21 j
z + 41 z 13 j z + 21 j
(b) Determine the corresponding third-order polynomials A3 (z) and A#

3 (z).
(c) Determine a difference equation describing the system.
Problem 16.22 Let
H(z) =
1 21 e 4
1 12 z 1 21 ej 3
1
1 j
z 2
z 2e 3
z 21 ej 4
(a) Determine its group delay, ().
(b) Determine the corresponding third-order polynomials A3 (z) and A#

3 (z).
(c) Determine a difference equation describing the system.
Problem 16.23 What is the smallest-order all-pass system that satisfies the following two condi
tions: (a) h(n) is real-valued and (b) H(z) has a pole at 21 ej 4 . Is H(z) unique? Give one valid
expression for H(z).
Problem 16.24 What is the smallest-order all-pass system that satisfies the following two condi
tions: (a) h(n) is real-valued and (b) H(z) has poles at 21 ej 4 and 31 ej 3 . Is H(z) unique? Give
one valid expression for H(z).
Problem 16.25 Which of the following LTI systems are minimum-phase?
z 31 z 41

, |z| > 1/2
(a) H(z) =
z 21 z 81
z 31 z 41

, |z| < 1/8.
(b) H(z) =
z 21 z 81
(c) H(z) =
(d) H(z) =
z 31 z 41

, 1/8 < |z| < 1/2.
z 21 z 81
z
z
1
2
1
3
1
8
, |z| > 1/2.

z 41 z 61

, |z| > 1/3
(a) H(z) =
z 31 z 51
(b) H(z) =
(c) H(z) =
(d) H(z) =
z 41 z 61

, |z| < 1/5.
z 31 z 51
z 41 z 61

, 1/5 < |z| < 1/3.
z 31 z 51
z
z
1
3
1
4
1
5
, |z| > 1/3.

1
(a) H(z) =
, |z| > 1/2.
z 12
(b) H(z) =
(c) H(z) =
z 2
, |z| > 1/2.
z 12
z2
, |z| > 1/2.
z 12

1
(a) H(z) =
2 , |z| > 1/2.
z 2 + 14
(b) H(z) =
(c) H(z) =
z 2
z2 +
z2
z2 +
,
1 2
4
|z| > 1/2.
,
1 2
4
|z| > 1/2.
Problem 16.29 Determine, when they exist, realizable inverses for the systems in Prob. 16.25.
Problem 16.30 Determine, when they exist, realizable inverses for the systems in Prob. 16.26.
Problem 16.31 Find the fundamental decomposition of the following causal and stable systems
into all-pass and minimum-phase components:
451
SECTION 16.8
PROBLEMS
452
(a) H(z) =
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
(b) H(z) =
(c) H(z) =
(d) H(z) =
z3

, |z| > 1/2.
1
z 41
2
z(z 3)

, |z| > 1/2.
z 21 z 41
z 1 (z 3)

, |z| > 1/2.
z 21 z 41
1
1
2
1
4
, |z| > 1/2.
Problem 16.32 Find the fundamental decomposition of the following causal and stable systems
into all-pass and minimum-phase components:
z2
(a) H(z) =
2
, |z| > 1/2.
z 21
z + 61
(b) H(z) =
(c) H(z) =
(d) H(z) =
z(z 2)

1 2
2
1
z
z
z+
(z 2)

1 2
2

1 2
2
1
6
z+
1
6
z+
1
6
, |z| > 1/2.

, |z| > 1/2.
, |z| > 1/2.
Problem 16.33 Consider the causal and stable system

H(z) =
z 1
z 14
Let Hmin (z) be its minimum-phase component.

(a) Find h(n) and hmin (n).
(b) Find the energies of h(n) and hmin (n). Compare the energies.
(c) Compare the group delays of H(z) and Hmin (z).
(d) Find the steady-state response of each of the systems to x(n) = ej 3 n . Compare the energies
of the responses.
Problem 16.34 Consider the causal and stable system
H(z) =
1
z+
1
3
Let Hmin (z) be its minimum-phase component.

(a) Find h(n) and hmin (n).
(b) Find the energies of h(n) and hmin (n). Compare the energies.
(c) Compare the group delays of H(z) and Hmin (z).
(d) Find the steady-state response of each of the systems to x(n) = ej 4 n . Compare the energies
of the responses.
Problem 16.35 Prove that the product of two minimum-phase transfer functions is also minimum
phase.
Problem 16.36 Find a minimum-phase system with squared magnitude response given by

2
25
26 5 cos

j
H(e ) =
16
17 8 cos
Problem 16.37 Derive the frequency response expression (16.23) for a type II FIR filter.
Problem 16.38 Consider the frequency response of a type-II FIR filter from Table 16.1.
(a) Establish the trigonometric identity

cos
1
2

+ cos
m+
1
2
PROBLEMS

= 2 cos

cos(m)
(b) Use part (a) to show that H(ej ) can be expressed in the alternative form
H(ej ) = ej
L1
2
cos
L/2
X
c(m) cos(m)
m=0
for some coefficients c(m), m = 0, 1, . . . , L/2.

(c) Relate the coefficients {g(m)} and {c(m)}.
Problem 16.39 Consider the frequency response of a type-III FIR filter from Table 16.1.
(a) Establish the trigonometric identity
sin [(m + 1)] sin [(m 1)] = sin() cos(m)
H(ej ) = ej (
L1
+
2
2
(L1)/2
) sin ()
c(m) cos(m)
m=0
for some coefficients c(m), m = 0, 1, . . . , (L 1)/2.
Problem 16.40 Consider the frequency response of a type-IV FIR filter from Table 16.1.
(a) Establish the trigonometric identify

sin
1
m+
2

1
m
2
sin

= 2 sin

cos(m)
H(ej ) = ej (
453
SECTION 16.8
L1
+
2
2
X
) sin
c(m) cos(m)
2
m=0
L/2
for some coefficients c(m), m = 0, 1, . . . , L/2.

Problem 16.41 Consider the first-order all-pass section (16.52). Show that its impulse response
sequence is given by
h(n) = a (n) + (1 |a|2 )an1 u(n 1)
Problem 16.42 Consider the first-order all-pass section (16.52). Find the energy of its impulse
response sequence.
Problem 16.43 The impulse response of a causal and stable LTI system S is given by
n
h(n) =
1
2
u(n)
Find the impulse response of another causal and stable LTI system L with a zero at the point z = 3
and such that the magnitude responses of both S and L are identical. How do the phase responses
compare to each other?
454
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
Problem 16.44 The impulse response of a causal and stable LTI system S is given by
n1
h(n) = n
1
3
u(n 2)
Find the impulse response of another causal and stable LTI system L with a zero at the point z = 2
and such that the magnitude responses of both S and L are identical. How do the phase responses
compare to each other?
Problem 16.45 The DTFT of a sequence x(n) has the triangular form shown in Fig. 16.23 over
the interval [, ]. The sequence x(n) is transmitted over a channel (or system) that consists of a
series cascade of three relaxed sub-systems: an FIR system with transfer function z 2 21 z 1 , an
LTI system with frequency response H(ej ), and a single pole IIR system. The output of the last
subsystem is further modulated by the sequence (1)n in order to generate y(n).
H(ej )
x(n)
1/2
z 2 12 z 1
y(n)
z 1
(1)n
1/2
X(ej )
(rad/sample)
FIGURE 16.23 A sequence x(n) with the indicated DTFT, X(ej ), is transmitted over the
system depicted in top part of the figure for Prob. 16.45.
(a) Plot |Y (ej )| over [, ]. Is y(n) a real sequence? Hint. Recall the form of a first-order
all-pass filter.
(b) How would your answer to part (a) change if the roles of H(ej ) and X(ej ) are interchanged?
(c) How would your answer to part (a) change if the modulator is moved from the end to the
beginning of the system? That is, x(n) is first modulated by (1)n before being fed into the
cascade of three sub-systems. [The output of the last sub-subsystem now becomes y(n).]
(d) Determine the energy of the impulse response sequence of the overall system shown in the
figure.
(e) Is the system time-invariant? linear? causal?
(f) Give an example of a DTFT plot, X(ej ), that would result in a zero output sequence y(n).
Problem 16.46 Consider the block diagram shown in Fig. 16.24 where x(n) and y(n) are the input
and output sequences, respectively, and H1 and H2 are two LTI systems with frequency responses
H1 (ej ) and H2 (ej ). In the upper branch, the sequence x(n) is first multiplied by (1)n , filtered
by H1 , and then multiplied by (1)n . In the lower branch, the sequence x(n) is filtered by H2 .
The output sequence y(n) is the sum of the sequences obtained at the outputs of both branches. Let
H(ej ) denote the frequency response of the overall system, i.e.,
Y (ej ) = H(ej )X(ej )
for any input-output pair {x(n), y(n)}.
(1)n
(1)n
H1 (ej )
y(n)
x(n)
+
H2 (ej )
(a) Prove that H(ej ) is equal to

h
H(ej ) = H1 ej() + H2 (ej )

(b) Plot H(ej ) (both magnitude and phase) in the range [, ] when H1 is an ideal low-pass
filter with cutoff frequency /4 and unit magnitude in the passband (including the frequencies
/4 and /4), while H2 is a low-pass filter with cutoff frequency /3 and unit magnitude
in the passband (including the frequencies /3 and /3). Both filters have linear phases in
their passbands with group delays that are equal to 2.
(c) Assume H1 is the same as above, while H2 is now a high-pass filter with cutoff frequency
at /2, unit magnitude in the pass band, and group delay that is equal to 4. Determine the
steady-state response of the system to

x(n) = cos
3
n+
4
3
Problem 16.47 Consider the feedback configuration shown in Fig. 16.25. The impulse response
sequence, h(n), of a causal system is indicated by the rectangular box. The value of the sample at the
time instant n = 1 is unknown and denoted by , which can assume an arbitrary real value contrary
to what the figure might suggest.
(a) Find the transfer function of the discrete-time system indicated by the rectangular box, i.e.,
find G(z). Indicate the corresponding region of convergence. What is the order of the system?
Is it stable? Is it minimum phase? What are the poles of G(z)? Do they depend on the value
of ? What are the zeros of G(z)? Do they depend on ?
(b) Find the transfer function, H(z), of the feedback system that maps x(n) to y(n). How do
the zeros of H(z) compare to the zeros of G(z)? What about the poles? What is the order
of H(z)? Find an for which H(z) is unstable and verify your answer. Is there any for
which H(z) is minimum phase?
(c) Find a set of conditions that should satisfy if H(z) were to be stable and verify whether a
solution exists.
455
SECTION 16.8
PROBLEMS
456
CHAPTER 16
MINIMUM
AND
LINEAR
PHASE
SYSTEMS
h(n)
y(n)
x(n)
+
FIGURE 16.25 A feedback configuration for Prob. 16.47.
Problem 16.48 Consider an elementary causal and stable all-pass function with a real pole at a,
B(z) =
z 1 a
,
1 az 1
|a| < 1
(a) Argue that B(ej ) is a monotonically decreasing function that starts at B(ej0 ) = 0 and
attains B(ej ) = . That is, the change in phase as goes from 0 to is .
(b) Now consider an all-pass function of order M with real poles, viz., a product of M elementary
sections as follows
A(z) =
z 1 aM
z 1 a1 z 1 a2
...
1
1
1 a1 z 1 a2 z
1 aM z 1
Prove that the phase response A(ej ) is also a monotonically decreasing function that starts
at A(ej0 ) = 0 and attains A(ej ) = M . That is, the change in phase as goes from
0 to is M . In particular, what is the value of A(ej )?
(c) Consider the filter structure shown in Fig. 16.26 where A(z) is chosen as a second-order
(M = 2) all-pass function as above. The transfer function from x(n) to y(n) is denoted by
G(z) and is equal to
1
G(z) = [A(z) + 1]
2
Argue that there should exist an angular frequency 0 < 0 < such that G(ej0 ) =
G(ej ) = 1 and G(ej0 ) = 0. Remark. This filter structure can be used to remove the
single-frequency component at 0 from the input signal x(n), by properly choosing a1 and
a2 to satisfy A(ej0 ) = .
x(n)
A(z)
y(n)
+
FIGURE 16.26 A digital notch filter for Prob. 16.48.
CHAPTER
17
he discrete-time Fourier transform (DTFT) is a useful frequency-domain representation

that associates a function X(ej ) of with every sequence x(n). The angular frequency
is a continuous real-valued variable that assumes values over a 2-wide interval such
as [, ] or [0, 2]. Thus, note that while the sequence x(n) is defined only for discrete
(integer) values of n, its DTFT is defined over continuous (real) frequencies . Storing
and manipulating such frequency representations by digital processors can be problematic unless an analytical expression is available for X(ej ) in terms of a finite number of
parameters.
These remarks serve as motivation for introducing the Discrete Fourier Transform (DFT),
where the qualification discrete-time in the DTFT is now replaced by the qualification
discrete in the DFT. The DFT will associate with every sequence x(n) another sequence
in the frequency domain, denoted by X(k), with discrete frequencies indexed by the variable k. In this way, both x(n) and X(k) end up being sequences with a finite number
of samples. By dealing with discretized sequences in the time and frequency domains,
the signals become amenable to manipulations (such as storage and processing) that are
particularly suited for digital computing devices.
17.1 MOTIVATION
Consider a sequence x(n) and let X(ej ) denote its DTFT,
X(ej ) =
x(n)ejn
(17.1)
n=
We already know that the DTFT is periodic with period 2. We can therefore focus our
attention on a 2long interval, say, either [0, 2] or [, ]. As the discussion
will reveal, when studying the DFT of a sequence it is more convenient to focus on the
interval [0, 2] as opposed to the interval [, ], which was our practice in the
previous chapters.
Thus, consider the interval [0, 2] and assume we divide it into N smaller sub-intervals
of width 2/N each, as illustrated in Fig. 17.1. This construction results in N discrete
angular frequencies at the endpoint locations of the sub-intervals:
0,
2 4 6
2(N 1)
,
,
, ...,
N N N
N
(17.2)
We represent these discrete frequencies as
k =
2k
, k = 0, 1, . . . , N 1
N
(17.3)
457

Copyright
458
CHAPTER 17
DISCRETE
FOURIER
TRANSFORM
Note that we are excluding the discrete frequency that corresponds to the location = 2
(which would result from setting k = N ); this is because the value of X(ej ) at = 2
coincides with the value of X(ej ) at = 0 due to the 2-periodicity of the DTFT. That
is, the information that is represented by X(ej ) at = 2 is already represented by the
value of X(ej ) at = 0 and we can therefore ignore it.
2
N
(rad/sample)
N sub-intervals of
width 2/N each
N 1
k axis
N discrete points
FIGURE 17.1 Dividing the interval [0, 2] into N subintervals of width 2/N each gives
rise to the N discrete frequency points defined by (17.3) with indices k = 0, 1, . . . , N 1.
Now consider the DTFT of x(n) over the one-period interval [0, 2]. We sample
X(ej ) every 2/N radians/sample at the discrete frequency locations (17.3) to obtain the
N samples

X(k) = X(ej )= 2k , k = 0, 1, . . . , N 1
(17.4)
N
This construction is illustrated in Fig. 17.2; sampling one period of X(ej ) results in N
samples X(k).
Obviously, since the DTFT is periodic of period 2, had we sampled it instead at all
multiples of 2/N , and not only over k = 0, 1, . . . , N 1, then we would have obtained
a sequence X(k) that is periodic of period N (instead of only the N samples represented
by (17.4)), namely,

X(k) = X(ej )= 2k , k = . . . , 2, 1, 0, 1, 2, . . .
(17.5)
This construction is illustrated in Fig. 17.3.

The DFT sequence X(k) defined via (17.5) inherits the periodic nature of X(ej ) since
its samples repeat themselves every period N , i.e.,
X(k) = X(k + N ),
for all integer k
(17.6)
However, just as was the case with the study of the DTFT transform, it is sufficient for analysis and design purposes to focus only on one period of X(k), as was already anticipated
in (17.4).
459
j
SECTION 17.1
X(e )
MOTIVATION
(rad/sample)
X(k)
k
k=0
k =N 1
FIGURE 17.2 Sampling the DTFT of a sequence x(n) every 2/N radians/sample over the
interval [0, 2] results in N samples X(k) for k = 0, 1, . . . , N 1.
X(ej )
(rad/sample)
X(k)
k
k = (N 1)
k=0
k =N 1
FIGURE 17.3 Sampling the periodic DTFT of a sequence x(n) over the entire line at multiples
of 2/N results in a periodic sequence X(k) of period N .
Example 17.1 (Sampling the DTFT of a sequence)

x(n) = 0.5(n + 1) + (n) + 0.5(n 1)
460
whose DTFT we already found to be
CHAPTER 17
X(ej ) = 1 + cos()
Figures 17.4 and 17.5 display the sequence x(n) and its DTFT over the interval [0, 2].
x(n)
1
0.5
2
1.8
1.6
1.4
1.2
j
X(e )
DISCRETE
FOURIER
TRANSFORM
1
0.8
0.6
0.4
0.2
0
FIGURE 17.5
3
4
(rad/sample)
A plot of the DTFT X(ej ) = 1 + cos() over the interval [0, 2].
Let us now sample the DTFT of x(n) by using two different choices for N , say, N = 3 and
N = 4. In the first case, we divide the interval [0, 2] into 3 subintervals with discrete frequencies
at the locations
2 4
0,
,
(N = 3)
3
3
while in the second case we divide the same interval into 4 subintervals with discrete frequencies at
the locations
3
0, , ,
(N = 4)
2
2
461
The samples of X(k) that arise from choosing N = 3 are
SECTION 17.1
X(0)
X(1)
1 + cos(0) = 2

2
= 0.5
1 + cos
3

X(2)
1 + cos
4
3
MOTIVATION
= 0.5
while the samples of X(k) that arise from choosing N = 4 are

X(0)
X(1)
X(2)
X(3)
1 + cos(0) = 2

1 + cos
= 1
2
1 + cos () = 0

3
1 + cos
=1
2
These samples are illustrated in Figs. 17.6 and 17.7. We therefore see that different choices for N
lead to different DFT sequences. That is why we usually quality the DFT by explicitly mentioning
the value of N that it relates to. We usually use the terminology N point DFT to emphasize that
the DFT is being computed based on N points. In the current example, we evaluated 3point and
4point DFTs.
X(ej) and its samples X(k) for k=0, 1, 2
2
X(0)
1.8
1.6
1.4
X(e )
1.2
1
0.8
X(1)
0.6
X(2)
0.4
0.2
0
3
4
(rad/sample)
FIGURE 17.6 A plot of the DTFT X(ej ) = 1 + cos() over the interval [0, 2] and its
sampled version, X(k) for k = 0, 1, 2. The samples are computed at intervals that are multiples of
2/3.
462
X(ej) and its samples X(k) for k=0, 1, 2, 3

2
CHAPTER 17
X(0)
DISCRETE
FOURIER
TRANSFORM
1.8
1.6
1.4
X(e )
1.2
X(1)
X(3)
0.8
0.6
0.4
0.2
X(2)
0
3
4
(radians/sample)
FIGURE 17.7 A plot of the DTFT X(ej ) = 1 + cos() over the interval [0, 2] and its
sampled version, X(k) for k = 0, 1, 2, 3. The samples are computed at intervals that are multiples
of /2.
17.2 RELATION TO ORIGINAL SEQUENCE

The sequence X(k) in (17.4) was obtained by sampling the DTFT, X(ej ), of the sequence x(n) at N discrete frequencies, k , for k = 0, 1, . . . , N 1. Two questions are in
order and deserve closer investigation.
Question A: Is it possible to obtain the N values of X(k) directly from the time-domain
sequence, x(n), without having to go through the intermediate step of determining X(ej )
first and then sampling it?
Answer: The answer will be in the affirmative. As we are going to see, the procedure will
generally involve transforming x(n) into a periodic sequence xp (n) and then using xp (n)
to determine the values of X(k) see (17.18) further ahead.
Question B: Given the N point DFT values, X(k) over k = 0, 1, . . . , N 1, is it possible to recover the original time-domain sequence x(n) from these values (just like it was
possible to recover x(n) from knowledge of its DTFT, X(ej ))?
Answer: The answer will be negative unless the sequence x(n) is causal with finiteduration L N . Recall that we can always recover a sequence x(n) from knowledge
of its DTFT over [0, 2] by means of the inversion formula
x(n) =
1
2
X(ej )ejn d
(17.7)
However, by sampling X(ej ) at every 2/N radians/sample, and by retaining the sample
values X(k) for k = 0, 1, . . . , N 1, some information is generally lost. The discussion
in the sequel will clarify for which sequences, x(n), the DFT can still be used to recover
the sequence. The discussion will also reveal for which sequences x(n), the DFT will lead
to loss of information due to aliasing-in-time (in which case, x(n) will not be recoverable
from its DFT).
Determining the DFT Directly from the Time-Domain Sequence

We address the first question, namely, the task of evaluating the sequence X(k) directly
from x(n). For this purpose, we start from the definition of the sequence X(k) for all
values of k, as given by (17.5):

X(k) = X(ej )= 2k , k = . . . , 2, 1, 0, 1, 2, . . .
(17.8)
As noted earlier, we are mainly interested in the N sample values X(k) over the interval
k = 0, 1, . . . , N 1. Nevertheless, to arrive at an expression for these values in terms
of the samples of x(n), it is useful to examine the samples X(k) over all possible integer
values of k (both within the range 0 k N 1 and outside it).
To begin with, we substitute into (17.8) the definition of X(ej ) to get
X(k) =
x(n)e
jn
n=
x(n)ej
2k
N n
n=
= 2k
N
k = . . . , 2, 1, 0, 1, 2, . . .
(17.9)
where has been replaced by the sample value 2k/N . In expression (17.9), each sample
2k
x(n) is multiplied by the exponential sequence ej N n . Let us examine the summation
more closely and find out what is happening with the particular samples of x(n).
Thus, note that the sample x(0), at time n = 0, inside the summation (17.9) is multiplied
by 1 since

2k
= 1
(17.10)
ej N n
n=0
Likewise, for any value of n that is an integer multiple of N , say n = mN , we find that
x(mN ) is also multiplied by the same value 1 since
ej
2k
N n
= 1,
when n = mN
(17.11)
We therefore say that the sample n = 0 and all other samples of x(n) at multiples of N
are processed equally by the summation (17.9).
Now, consider the sample x(1), at time n = 1, inside the summation (17.9). It is
multiplied by

2k
2k
= ej N
(17.12)
ej N n
n=1
Likewise, for any value of n that is an integer multiple of N away from 1, say n = mN +1,
we also find that x(mN + 1) will be multiplied by the same factor ej2k/N since
ej
2k
N n
= ej
2k
N
when n = mN + 1
(17.13)
We therefore say that the sample n = 1 and all other samples of x(n) that are at multiples
of N away from n = 1 are processed equally by the summation (17.9).
463
SECTION 17.2
RELATION TO
ORIGINAL
SEQUENCE
464
CHAPTER 17
DISCRETE
FOURIER
TRANSFORM
The argument continues similarly for all other samples of x(n) at the time instants
n = 2, 3, . . . , N 1. Consider, for example, the sample x(N 1) at time n = N 1
inside the summation (17.9). This sample is multiplied
ej
2k
N n
n=N 1
= ej
2k(N 1)
N
For any value of n that is an integer multiple of N away from (N 1), say n = mN +(N
1), the corresponding sample x(n) will also be multiplied by the same factor ej2k(N 1)/N
since
2k(N 1)
2k
,
when n = mN + N 1
(17.14)
ej N n = ej N
We therefore say that the sample n = N 1 and all other samples of x(n) that are at
multiples of N away from n = N 1 are processed equally by the summation (17.9).
By examining what happens to the samples of x(n) over n = 0, 1, . . . , N 1, we have
also been able to deduce what happens to all other samples of x(n) for all other values of
n. This is because these other samples are located at multiples of N away from any of the
initial samples at n = 0, 1, . . . , N 1. If we therefore group together all samples of x(n)
that are multiplied by the same factor, we can equivalently rewrite the summation (17.9)
into a sum of N separate rows as follows:
X(k) = [. . . + x(N ) + x(0) + x(N ) + x(2N ) + . . .] +

[. . . + x(N + 1) + x(1) + x(N + 1) + x(2N + 1) + . . .] ej
2k
N
j 4k
N
+
[. . . + x(N + 2) + x(2) + x(N + 2) + x(2N + 2) + . . .] e
..
.
2k(N 1)
[. . . + x(1) + x(N 1) + x(2N 1) + x(3N 1) + . . .] ej N
Each row in the above expression contains a sum of terms multiplied by a particular factor.
There are N such factors and they are given by
ej
2kn
N
, n = 0, 1, . . . , N 1
(17.15)
Let us denote the sum of terms in the first row by xp (0):
xp (0) = . . . + x(N ) + x(0) + x(N ) + x(2N ) + . . .
(17.16)
Likewise, let us denote the sum of terms in the other rows by

xp (1) =
xp (2) =
..
. =
xp (N 1) =
. . . + x(N + 1) + x(1) + x(N + 1) + x(2N + 1) + . . .

. . . + x(N + 2) + x(2) + x(N + 2) + x(2N + 2) + . . .
..
.
(17.17)
. . . + x(1) + x(N 1) + x(2N 1) + x(3N 1) + . . .
Then, we can re-express X(k) as

X(k) = xp (0) + xp (1)ej
2k
N
+ xp (2)ej
4k
N
+ xp (N 1)ej
2k(N 1)
N
465
which we write more compactly as
SECTION 17.2
RELATION TO
ORIGINAL
SEQUENCE
X(k) =
NP
1
xp (n)ej
2k
N n
k = . . . 2, 1, 0, 1, 2 . . .
n=0
(17.18)
This result tells us how to obtain the samples X(k) directly from the time-domain sequence
x(n). We first use x(n) to form the N samples xp (n), n = 0, 1, . . . , N 1, and then use
these samples to evaluate the X(k) as in (17.18).
Expressions (17.16)(17.17) show that the samples xp (n) can be evaluated through the
following simple construction. We shift the sequence x(n) to the left and to the right by
multiples of N and then add the shifted sequences. The resulting sequence will be periodic
with period N :
xp (n) =
x(n + N )
(17.19)
When we evaluate the N point DFT of x(n) by means of the relation (17.18), we only
need to use the N samples of xp (n) over the period n = 0, 1, . . . , N 1.
Example 17.2 (Evaluating the DFT from the original time sequence)
x(n) = 0.5(n + 1) + (n) + 0.5(n 1)
and select N = 4. Figure 17.8 shows x(n) and several shifted versions of it to the left and to the
right by multiples of N = 4 samples. It is seen in this case that the value of N is large enough
so that when we sum the shifted versions of x(n) their samples will not interfere with each other.
Sometimes, however, the width of x(n) and the chosen value of N will be such that the various
shifted versions of x(n) will have samples at common locations, and these samples end up interfering
with each other. We illustrate this situation later in Example 17.4.
If we now add up x(n) and all its shifted versions we obtain the sequence xp (n) that is shown
in Fig. 17.9. Observe that xp (n) is periodic and has period N = 4. The samples of xp (n) over
n = 0, 1, 2, 3 are given by
xp (0)
xp (1)
0.5
xp (2)
xp (3)
0.5
Observe further that the samples of the original sequence x(n) over the same time interval are
x(0)
x(1)
0.5
x(2)
x(3)
466
x(n)
CHAPTER 17
DISCRETE
FOURIER
TRANSFORM
1
0.5
x(n 4)
1
0.5
x(n + 4)
1
0.5
x(n 8)
1
0.5
8
FIGURE 17.8 A sequence x(n) and several shifted versions of it to the left and to the right by
multiples of N = 4 samples.
Therefore, the samples of xp (n) and x(n) need not coincide with each other over the interval n =
0, 1, . . . , N 1.
Now using the just determined values for xp (n), n = 0, 1, 2, 3, in expression (17.18) for X(k)
we find that
X(0)
3
X
n=0
xp (n) = 2

3
2
X(1)
1 + 0.5 ej 2 + ej
X(2)
1 + 0.5 ej + ej3 = 1 1 = 0
X(3)
1 + 0.5 ej

3
2
+ ej
9
2
=1+0=1

=1+0 =1
and we arrive at the same values that were illustrated earlier in Fig. 17.7.
467
SECTION 17.2
xp (n)
RELATION TO
ORIGINAL
SEQUENCE
FIGURE 17.9 The periodic sequence xp (n) that is obtained from adding all the shifted versions
of the sequence x(n) shown in the first row of Fig. 17.8.
Example 17.3 (Constructing the periodic sequence)

Figure 17.10 illustrates a causal sequence, x(n), with nonzero samples over n = 0, 1, 2,:
x(n) = 0.5(n) + (n 1) + 0.5(n 2)
Select again N = 4. The same figure shows x(n) and shifted versions of it to the left and to the right
by multiples of N = 4 samples. It is seen in this case that the value of N is again large enough so
that when we sum the shifted versions of x(n) their samples will not interfere with each other.
If we now add up x(n) and all its shifted versions we obtain the sequence xp (n) that is shown
in Fig. 17.11. Observe again that xp (n) is periodic with period N = 4. Observe further that the
samples of xp (n) over n = 0, 1, 2, 3, now coincide with the samples of x(n):
xp (0)
xp (1)
xp (2)
xp (3)
=
=
=
=
0.5
1
0.5
0
=
=
=
=
x(0)
x(1)
x(2)
x(3)
Therefore, in this case, we can work directly with the samples of x(n) to obtain X(k) without
the need to evaluate xp (n) first. The fact that the samples of xp (n) and x(n) coincide over n =
0, 1, . . . , N 1, is due to two factors:
1. The sequence x(n) is causal (exists for n 0) and has finite-duration.
2. The duration L of x(n) satisfies L N . That is, the value of N is large enough so that no
interferences happen when the sequence x(n) and all its shifted versions are summed up. In
this way, the samples of x(n) over n = 0, 1, . . . , N 1, remain intact.
Example 17.4 (Aliasing in time)

Figure 17.12 illustrates the same causal sequence x(n) as in the previous example, with nonzero
samples over n = 0, 1, 2,:
x(n) = 0.5(n) + (n 1) + 0.5(n 2)
However, we now select N = 2. The same figure shows x(n) and several shifted versions of it
to the left and to the right by multiples of N = 2 samples. It is seen in this case that x(n) and
x(n 2) share a sample at location n = 2. Likewise, x(n) and x(n + 2) share a sample at location
n = 0. A similar observation holds for all other shifted sequences. Therefore, when we sum these
468
x(n)
CHAPTER 17
DISCRETE
FOURIER
TRANSFORM
1
0.5
x(n 4)
1
0.5
x(n + 4)
1
0.5
A causal sequence x(n) and shifted versions of it to the left and to the right by
FIGURE 17.10
N = 4 samples.
xp (n)
1
FIGURE 17.11 The samples of the periodic sequence xp (n) and the causal sequence x(n)
coincide over n = 0, 1, 2, 3.
sequences together, interferences will occur at the common sample locations. We say that aliasing
in time occurs. For instance, if we now add x(n) and all its shifted versions we obtain the sequence
xp (n) that is shown in Fig. 17.13. Observe that in this example xp (n) assumes the value 1 for all n;
the samples of xp (n) over n = 0, 1, are given by
xp (0)
xp (1)
469
x(n)
SECTION 17.2
RELATION TO
ORIGINAL
SEQUENCE
1
0.5
x(n 2)
1
0.5
x(n 4)
1
0.5
x(n + 2)
1
0.5
FIGURE 17.12 A causal sequence x(n) and several shifted versions of it to the left and to the
right by multiples of N = 2 samples.
It is seen that the value of xp (n) at n = 0 is a distorted version of the value of x(n) at n = 0; the
distortion is due to the interference that occurs between the samples of x(n) and x(n + 2) at n = 0.
Using the just determined values for xp (n), n = 0, 1, in the expression (17.18) for X(k) we find
that
X(0)
1
X
xp (n) = 2
n=0
X(1)
1 + ej = 0
Example 17.5 (Different time sequences can lead to the same periodic sequence)
From the result (17.18) we see that in order to find the DFT of a sequence x(n) we first embed
it into a periodic sequence xp (n) and then use (17.18) to determine the values of X(k) for k =
0, 1, . . . , N 1.
470
xp (n)
CHAPTER 17
DISCRETE
FOURIER
TRANSFORM
FIGURE 17.13 The samples of the periodic sequence xp (n). Aliasing in time occurs, for
example, at n = 0 and n = 2. The aliasing is due to the interference between the samples of
x(n) and x(n + 2) at time n = 0, and between the samples of x(n) and x(n 2) at n = 2.
Now, if the sequence x(n) happens to have a finite duration L, and if the value of N that we
choose for the DFT is larger than or equal to L, then the samples of xp (n) in the interval 0 n
N 1 will coincide with the samples of x(n) but the exact locations of the samples may in general
be different. This situation was encountered in Figs. 17.8 and 17.9 where we had
xp (0)
xp (1)
0.5
xp (2)
xp (3)
0.5
x(1)
0.5
x(0)
x(1)
0.5
x(2)
and
The sample values are the same but they appear over different intervals: n [0, 3] for xp (n) and
n [1, 2] for x(n). Can we use the samples of xp (n) to recover the exact locations of the samples
of x(n)? The answer is negative. This is because different sequences x(n) can lead to the same
values for xp (n) over 0 n 3.
For example, consider the sequence
x (n) = x(n 4)
The sequence x(n 4) is shown in the second row of Fig. 17.8. Now take the sequence x (n) and
shift it to the left and to the right by multiples of N = 4 and add up all resulting sequences. The
corresponding periodic sequence will continue to be the same xp (n) that we obtained for x(n) and
which was shown in Fig. 17.9. Thus, given the samples of xp (n) over n = 0, 1, 2, 3, we cannot tell
whether they were generated from x(n) or x (n).
However, assume the following two conditions are satisfied simultaneously:
1. The original sequence x(n) has some finite-duration L satisfying L N .
2. The sequence x(n) is causal, x(n) = 0 for n < 0.
That is, the values of x(n) exist over 0 n L 1 and L N . Then, x(n) and xp (n) will
coincide over n = 0, 1, . . . , N 1, and they define each other uniquely:
x(n) = xp (n) for 0 n N 1
Obviously, if N is smaller than L, then aliasing in time occurs while forming xp (n) and the samples
of x(n) cannot be recovered completely from those of xp (n), as was illustrated in Example 17.4.
17.3 DISCRETE FOURIER TRANSFORM

The discussion in the previous section shows that the N point DFT of a sequence x(n) is
obtained in two steps as follows:
(a) First, embed x(n) into a periodic sequence xp (n) of period N using (17.19). Specifically, shift x(n) to the left and to the right by multiples of N and add up all sequences
to obtain xp (n). Keep the samples of xp (n) that lie within the period 0 n N 1.
(b) Compute the N point DFT samples by using the relation
X(k) =
N
1
X
xp (n)ej
2k
N n
n=0
k = 0, 1, . . . , N 1
(17.20)
This step results in N values X(k), k = 0, 1, . . . , N 1.

Causal and Finite-Duration Sequences
However, the presentation in Examples 17.217.5 showed that when the sequence x(n)
is causal and has duration L N , then the samples of x(n) and xp (n) coincide over
0 n N 1. In this case, we can evaluate the N point DFT directly from x(n) as
follows:
X(k) =
NP
1
x(n)ej
2k
N n
n=0
k = 0, 1, . . . , N 1
(17.21)
where we are replacing xp (n) by x(n). For convenience, the upper index in the summation
is kept as N 1 rather than L 1 even though x(n) = 0 for n L. This is done in order
to emphasize that we are dealing with an N point DFT.
The case of a causal and finite-duration sequence x(n) with L N is the situation
that we encounter most frequently in practice. It is for this reason that the N point DFT
of a sequence is often defined directly as in (17.21); our discussion is more general and
explains what happens when the sequence x(n) is not causal or has duration L > N .
It should be understood though that the DFT sequence so defined (whether by using
xp (n) or x(n)) is a periodic sequence of period N , and that the definitions (17.20)(17.21)
provide the values of X(k) over a single period, namely, over k = 0, 1, . . . , N 1. The
fact that the DFT is periodic of period N was illustrated in Fig. 17.3 and can be easily seen
from the above defining relations as well. For example, using (17.21) we get
X(k + N ) =
N
1
X
n=0
x(n)ej
2(k+N )
n
N
N
1
X
x(n)ej
2k
N n
= X(k)
(17.22)
n=0
In addition, it should also be understood that the N point DFT is in effect the transform
of a periodic sequence xp (n), which is obtained by periodically repeating x(n) every N
samples and summing the repeated sequences as in (17.19).
471
SECTION 17.3
DISCRETE
FOURIER
TRANSFORM
472
CHAPTER 17
DISCRETE
FOURIER
TRANSFORM
Example 17.6 (Unit-sample sequence)

Consider the unit-sample sequence
x(n) = (n)
This is obviously a causal and finite-duration sequence (with duration L = 1). The N point DFT
of x(n) is therefore given by expression (17.21):
X(k)
N1
X
x(n)ej
2k n
N
(n)ej
2k n
N
n=0
N1
X
n=0
1,
k = 0, 1, . . . , N 1
In other words,
x(n) = (n)
DFT
X(k) = 1,
k = 0, 1, . . . , N 1
Figure 17.14 shows x(n) and its N point DFT X(k). In this example, the samples X(k) are realvalued and, therefore, only their amplitudes need to be displayed.
X(k)
x(n)
1
N 1
FIGURE 17.14 Plots of the unit-sample sequence (left) and its N point DFT (right).

Consider now the rectangular pulse
(
x(n) =
1,
0,
0 n L1
otherwise
(17.23)
with finite-duration L. We select an integer N L and evaluate the N point DFT of x(n) using
expression (17.21):
X(k)
N1
X
x(n)ej
DISCRETE
FOURIER
TRANSFORM
2kn
N
n=0
L1
X
ej
2kn
N
1 ej
2kL
N
(a geometric sum)
n=0
=
=
=
1 ej
ej
kL
N
ej
k
N
2k
N
ej
kL
N
ej
ejk(L1)/N
k
N
ej
ej
kL
N
X(k) =
: e
k
N
sin (kL/N )
,
sin (k/N )
8
< L,
so that
k = 0, 1, . . . , N 1
k=0
jk(L1)/N
sin (kL/N )
,
sin (k/N )
k = 1, . . . , N 1
(17.24)
Recall that we determined the DTFT of the rectangular pulse x(n) earlier in Example 13.5, where
we found that
sin (L/2)
X(ej ) = ej(L1)/2
, [0, 2]
(17.25)
sin (w/2)
As expected, we see that the DFT samples X(k) in (17.24) correspond to sampling the DTFT (17.25)
at the locations = 2k/N .
In particular, observe that if we choose N = L, then the Lpoint DFT of the rectangular pulse
trivializes to
(
L,
k=0
(17.26)
X(k) =
0,
k = 1, . . . , L 1
In other words,
X(k) = L (k),
k = 0, 1, . . . , L 1 (Lpoint DFT)
(17.27)
The sequence X(k) = L (k) may not covey much information about the DTFT of x(n); however,
it is sufficient to fully recover x(n) (as will be seen by using the inverse DFT expression of the next
section see Example 17.9)! This situation is illustrated in Fig. 17.15.
x(n)
X(k)
L
1
1
FIGURE 17.15
(right).
L1
473
SECTION 17.3
Plots of a rectangular pulse of duration L samples (left) and its Lpoint DFT
Let us now plot the DFT of the rectangular pulse for values of N larger than L. For illustration
purposes, we first reproduce in Fig. 17.16 the magnitude and phase plots of the DTFT of x(n), which
were derived earlier in Example 13.5; here we are showing the plots over the interval [0, 2].
474
CHAPTER 17
Figures 17.17 and 17.18 show the magnitude and phase plots of the DFT of the rectangular pulse for
L = 5 for both cases of N = 32 and N = 16.
DISCRETE
FOURIER
TRANSFORM
magnitude plot
5
|X(ej)|
4
3
2
1
0
3
4
(rad/sample)
phase plot
3
4
(rad/sample)
X(ej)
2
1
0
1
2
3
FIGURE 17.16 Plot of the magnitude (top) and phase (top) of DTFT of a rectangular pulse of
width L = 5 over [0, 2].
Example 17.8 (2-point DFT)

Consider a sequence x(n) with L = 2 samples, x(0) and x(1). Its 2point DFT is given by
X(k)
1
X
x(n)ej
2 kn
2
n=0
x(0) + x(1)ejk , k = 0, 1
so that the two DFT coefficients are

X(0)
x(0) + x(1)
X(1)
x(0) x(1)
Note that the DFT coefficients in this case are obtained by simply adding and subtracting the samples of x(n). We shall call upon this result later in Chapter 20 when we develop the Fast Fourier
Transform (FFT) see Example 19.2.
475
32point DFT
SECTION 17.4
INVERSE
DFT
|X(k)|
4
3
2
1
0
10
10
15
k
phase plot
20
25
30
20
25
30
X(k)
2
1
0
1
2
15
k
FIGURE 17.17 Plot of the magnitude (top) and phase (bottom) of the 32point DFT of a
rectangular pulse of width L = 5.
17.4 INVERSE DFT

Let us now address the question of recovering the original time-domain sequence x(n)
from knowledge of its N point DFT. Thus consider a causal sequence x(n) with duration
L N . Its N point DFT is given by
X(k) =
N
1
X
x(m)ej
2k
N m
m=0
k = 0, 1, . . . , N 1
(17.28)
where we are now using the symbol m to denote the time variable. Multiplying both sides
of the above equality by ej2kn/N we get
X(k)ej
2k
N n
N
1
X
x(m)ej
2k
N (mn)
(17.29)
m=0
Summing both sides over k = 0, 1, . . . , N 1 gives

N
1
X
X(k)ej
2k
N n
N
1 N
1
X
X
x(m)ej
2k
N (mn)
k=0 m=0
k=0
N
1
X
m=0
x(m)
N
1
X
k=0
j 2k
N (mn)
(17.30)
476
16point DFT
CHAPTER 17
DISCRETE
FOURIER
TRANSFORM
|X(k)|
4
3
2
1
0
10
15
10
15
k
phase plot
3
X(k)
2
1
0
1
2
3
0
5
k
FIGURE 17.18 Plot of the magnitude (top) and phase (bottom) of the 16point DFT of a
rectangular pulse of width L = 5.
where we switched the order of the summations since x(m) is independent of k. Now note
the useful identity:
NP
1
ej
2k
N
k=0
N,
0,
if = 0, N, 2N, . . .
otherwise
(17.31)
In other words, the sum evaluates to N for values of that are multiples of N (this conclusion is obvious), and the sum is equal to zero otherwise. To justify this latter conclusion,
consider, for example, the case = 1. Then the sum becomes
N
1
X
ej
2k
N
k=0
which corresponds to adding N complex numbers on the unit circle, and which are located
at the angles
2 4 6
2(N 1)
0,
,
,
, ...,
N N N
N
These numbers cancel each other and the sum evaluates to zero, as illustrated in Fig. 17.19
for the cases N = 3 and N = 4.
Applying the identity (17.31) to the sum that appears in (17.30) we have
N
1
X
k=0
ej
2k
N (mn)
N,
0,
if (m n) = 0, N, 2N, . . .
otherwise
(17.32)
477
N =3
N =4
Im
Im
12
INVERSE
DFT
Re
FIGURE 17.19
and N = 4.
SECTION 17.4
Re
Location of the samples of the sequence ej2k/N on the unit circle for N = 3
Now recall that both m and n vary over the range 0 n N 1. Therefore, the condition
(m n) = 0, N, 2N, . . .
can only be satisfied when m n = 0 and, hence, m = n. Returning to (17.32) we
conclude that
N
1
X
j 2k
N (mn)
k=0
N
0
when m = n
for all m 6= n over 0 m N 1
(17.33)
Consequently, the expression on the right-hand side of (17.30) collapses to

N
1
X
x(m)
m=0
N
1
X
j 2k
N (mn)
k=0
= N x(n)
(17.34)
and we arrive at the desired inverse DFT (IDFT) relation:

x(n) =
1
1 NP
2k
X(k)ej N n ,
N k=0
n = 0, 1, . . . , N 1
(17.35)
Aliasing in Time
The derivation of the inverse DFT expression (17.35) assumed a causal and finite-duration
sequence x(n) with L N for which the DFT is computed via (17.28). What if the
original sequence x(n) is not causal or its duration exceeds N , in which case aliasing in
time may occur? In this case, we would form first the corresponding periodic sequence
xp (n) and use it instead to determine the N point DFT via
X(k) =
N
1
X
n=0
xp (n)ej
2k
N n
k = 0, 1, . . . , N 1
(17.36)
478
CHAPTER 17
DISCRETE
FOURIER
TRANSFORM
The same argument given above will then indicate that the inverse DFT formula will allow
us to recover the samples of xp (n), as opposed to those of x(n), as follows:
xp (n) =
N 1
2k
1 X
X(k)ej N n ,
N
n = 0, 1, . . . , N 1
k=0
(17.37)

Consider the Lpoint DFT
X(k) = L(k),
k = 0, 1, . . . , L 1
and let us determine its inverse DFT. Using the relation (17.35) we have
x(n)
L1
2k
1 X
X(k)ej L n
L k=0
L1
2k
1 X
L(k)ej L n
L
1,
k=0
n = 0, 1, . . . , L 1
We find that the time-sequence is the rectangular pulse of width L, as anticipated earlier.
17.5 VECTOR REPRESENTATION

Let WN denote the N th root of unity, i.e., the complex numberi.e.,
WN
= ej N

2
2
j sin
= cos
N
N
(17.38)
(17.39)
Then evaluating the nk-th power of WN gives

(WN )nk = ej
2k
N n
so that the definition (17.21) for the N point DFT of a causal and finite-duration sequence,
x(n), with L N , can be expressed as
X(k) =
N
1
X
x(n)WNnk ,
n=0
k = 0, 1, . . . , N 1
(17.40)
Observe from (17.40) that for each k, the value of X(k) can be evaluated as the inner
product of two vectors:
h
WNk
WN2k
WN3k
(N 1)k
. . . WN
(17.41)
479
and
x(0)
x(1)
x(2)
..
.
x(N 1)
SECTION 17.5
VECTOR
REPRESENTATION
(17.42)
In this way, if we collect the N DFT coefficients into a column vector as well, say, as
X(0)
X(1)
X(2)
(17.43)
..
.
X(N 1)
then expression (17.40) gives the vector relation:
1
1
1
X(0)
1
2
W
W
X(1)
N
N
X(2)
1
WN2
WN4
=
.
..
.
..
.
..
|
X(N 1)
{z
}
XN
(N 1)
WN
x(0)
x(1)
x(2)
..
.
2(N 1)
WN
1
WN3
WN6
...
1
(N 1)
. . . WN
2(N 1)
. . . WN
..
.
3(N 1)
. . . WN
W
{z N
FN
(N 1)2
(17.44)
x(N 1)
{z
}
xN
If we introduce the N 1 vectors XN and xN , as defined above, as well as the N N

matrix FN , then we can write (17.44) more compactly as
XN = FN xN
(17.45)
This relation states that the N point DFT of the sequence x(n) can be evaluated by transforming the vector xN into the vector XN by means of the so-called N N DFT matrix
FN whose (m, k)-th entry is given by
[FN ]mk = ej
2mk
N
(N N )
(17.46)
The matrix FN has a useful property. First, note that FN is a matrix with complex-valued
entries. Thus, let FN denote the matrix that is obtained by complex conjugating the entries
of FN followed by transposing the matrix. For example, let N = 4. Then
1
1
1
1
3
1 ej 2
ej ej 2
F4 =
1 ej ej2 ej3
3
9
1 ej 2 ej3 ej 2
480
CHAPTER 17
DISCRETE
FOURIER
TRANSFORM
If we complex-conjugate each entry of F4 we get the matrix
1
1
1
1
3
1 ej 2
ej ej 2
j
j2
j3
1 e
e
e
3
9
1 ej 2 ej3 ej 2
In order to arrive at F4 we transpose the above complex-conjugated matrix. However,

since the matrix is already symmetric we get that it agrees with F4 and we therefore
1
1
1
1
3
1 ej 2
ej ej 2
F4 =
1 ej ej2 ej3
9
3
1 ej 2 ej3 ej 2
Now it follows from the identity (17.32) that the matrix FN satisfies
FN FN = N IN = FN FN
where IN denotes the N N identity matrix
1
IN =
..
(N N )
(17.47)
(17.48)
That is, the products FN FN and FN FN are diagonal matrices with diagonal entries equal
to N .
Likewise, the inverse DFT operation (17.35) can be expressed in matrix form as follows:
xN =
1
F XN
N N
(17.49)
in terms of the matrix FN . We can derive this result by either following the derivation that
led to (17.44) or by starting from (17.45) and using (17.47). Indeed, multiplying both sides
of (17.45) by FN from the left we get
FN XN = FN FN xN = N xN
| {z }
=N IN
from which (17.49) follows.
17.6 APPLICATIONS
TO BE ADDED
Practice Questions:
1.
2.
481
17.7 PROBLEMS
SECTION 17.7
PROBLEMS
Problem 17.1 Find the 4point DFTs of the sequences defined below over the interval 0 n 3
(all other samples are zero):
n
(a) x(n) =
1 , 1, 1, 1 .
(b) y(n) = (1)n x(n).
(c) y(n) = ej 2 n x(n).

n
2
(d) y(n) = cos
x(n).
In each case, plot |X(k)| and X(k).

Problem 17.2 Find the 6point DFTs of the sequences defined below over the interval 0 n 5
(all other samples are zero):
n
(a) x(n) =
1 , 0, 1, 0, 1, 0 .
(b) y(n) = ejn x(n).
(c) y(n) = ej 3 n x(n).
(d) y(n) = sin
2
n
3
x(n).
In each case, plot |X(k)| and X(k).

Problem 17.3 Let x(n) = (n + 2) (n) + (n 2).
(a) Determine its DTFT.
(b) Sample the DTFT to obtain the 6point DFT of x(n).
(c) Obtain the same 6point DFT using (17.20).
(d) Obtain the same 6point DFT using (17.21). Does aliasing in time occur?
Problem 17.4 Let x(n) = (n + 1) + 2(n) (n 1).
Problem 17.5 Let x(n) = (n + 3) (n) + (n 3).
Problem 17.6 Let x(n) = (n + 2) + (n 2).
Problem 17.7 Find the inverse DFTs of the DFT sequences defined below over one period, 0
k 3:
n
(a) X(k) =
1 , 1, 1, 1 .
(b) Y (k) = (1)n X(k).
482
CHAPTER 17
DISCRETE
FOURIER
TRANSFORM
(c) Y (k) = ej 2 k X(k).
k
2
(d) Y (k) = cos
X(k).
In each case, plot |x(n)| and x(n).

Problem 17.8 Find the inverse DFTs of the DFT sequences defined below over one period, 0
k 6:
n
(a) X(k) =
1 , 0, 1, 0, 1, 0 .
(b) Y (k) = ejk X(k).
(c) Y (k) = ej 3 k X(k).
k
3
(d) Y (k) = sin
X(k).
In each case, plot |x(n)| and x(n).

Problem 17.9 Determine the 4point DFT of the rectangular pulse of width L = 5 from Example
17.7. Compare the answer with the 4point DFT of a rectangular pulse of width L = 4. Explain the
results.
Problem 17.10 Determine the 4 and 6point DFT of the following pulse
8
>
>
>
<
2,
1,
x(n) =
>
0,
>
>
:
1,
n=0
n=1
n=2
n=3
Problem 17.11 Let x(n) be a rectangular pulse of duration 4 (L = 4).

(a) Determine its 8point DFT using (17.20).
(b) Determine its 8point DFT using (17.45).
Problem 17.12 Let X(k) = { 1, , 2, 1, 1}.
(a) Find its inverse DFT using (17.35).
(b) Determine its 8point DFT using (17.49).
Problem 17.13 Determine and plot the N point DFT of the following sequences, which are all
limited to the interval 0 n N 1:
(a) x(n) = (n).
(b) x(n) = u(n) u(n N ).

(c) x(n) = cos
2nk0
N
, k0 < N .
Problem 17.14 Determine and plot the N point DFT of the following sequences, which are all
limited to the interval 0 n N 1:
(a) x(n) = (n n0 ) + (n N + n0 ), n0 < N .
(b) x(n) = (n n0 ) + (n N + n0 ), n0 < N .

(c) x(n) = sin
2nk0
N
, k0 < N .
Problem 17.15 Determine the N point DFT of the sequences

(a) x(n) = cos2 (n).
(b) x(n) = sin2 (n).

Problem 17.16 Determine the N point DFT of the sequences
(a) x(n) = sin(2n) cos2 (n).
(b) x(n) = cos(n) sin2 (n).
483
Problem 17.17 Determine the N point DFT of the sequence

x(n) = cos(o n),
SECTION 17.7
0nN 1
PROBLEMS
Simplify the result when o is a multiple of 2/N .

Problem 17.18 Determine the N point DFT of the sequence
x(n) = sin3 (o n),
0nN 1
Simplify the result when o is a multiple of 2/N .

Problem 17.19 Figure 17.20 shows the magnitude and phase components of the DTFT of a sequence x(n). Determine the 4 and 8point DFTs of the sequence.
|X(ej )|
1
1/2
X(ej )
/2
FIGURE 17.20
Magnitude and phase plots of the DTFT of a sequence x(n) for Prob. 17.19.
Problem 17.20 Figure 17.21 shows the magnitude and phase components of the DTFT of a sequence x(n). Determine the 4 and 8point DFTs of the sequence.
Problem 17.21 Figure 17.22 shows a 4point DFT. Give two sequences x(n) whose 4point
DFTs agree with the figure.
Problem 17.22 Figure 17.23 shows a 4point DFT. Give two sequences x(n) whose 4point
DFTs agree with the figure.
Problem 17.23 Consider two real-valued sequences x1 (n) and x2 (n), both of finite-duration N .
Define y(n) = x1 (n) + jx2 (n). Determine the N points DFTs X1 (k) and X2 (k) in terms of the
N -point DFT Y (k). Remark. This problem shows how a complex-valued N point DFT can be
used to determine two N point DFTs of real-valued sequences.
Problem 17.24 Consider two periodic sequences x(n) and y(n) of periods Nx and Ny , respectively. Define w(n) = x(n) + y(n).
(a) Show that w(n) is periodic with period Nx Ny .
484
CHAPTER 17
|X(ej )|
DISCRETE
FOURIER
TRANSFORM
1/2
X(ej )
/2
FIGURE 17.21
Magnitude and phase plots of the DTFT of a sequence x(n) for Prob. 17.20.
X(k)
1
1/2
1/2
2
1/2
FIGURE 17.22 4point DFT for Prob. 17.21.
(b) Determine the N point DFT W (k) in terms of the N point DFTs X(k) and Y (k).
Problem 17.25 Express the N point DFT of cos2 (o n) x(n) in terms of the N point DFT of
x(n).
Problem 17.26 Express the N point DFT of sin2 (o n) x(n 2) in terms of the N point DFT
of x(n).
Problem 17.27 Express the N point inverse DFT of (1)k cos2 (o k) X(k) in terms of the
N point inverse DFT of X(k).
2
Problem 17.28 Express the N point inverse DFT of ej N k sin2 (o k) X(k) in terms of the
N point inverse DFT of X(k).
485
SECTION 17.7
X(k)
PROBLEMS
1/2
1
2
3
1/2
FIGURE 17.23 4point DFT for Prob. 17.22.
Problem 17.29 Let X(k) denote the N point DFT of a sequence x(n). Let y(n) denote the
N point DFT (not inverse DFT) of the sequence X(k). Let Y (k) denote the N point DFT of the
sequence y(n). Let w(n) denote the N point DFT (not inverse DFT) of the sequence W (k). Use
(17.45) to relate the sequences w(n) and x(n).
Problem 17.30 Consider a finite-duration sequence x(n), defined over 0 n L 1. Let X(k)
denote its N point DFT, with L N . Let also X(z) and X(ej ) denote its ztransform and
discrete-time Fourier transform (DTFT), respectively.
(a) Show that X(z) and X(k) are related as follows:
X(z) =
(b) Conclude that
X(ej ) =

N1
X(k)
1 z N X
2k
N
1 ej N z 1
k=0

N1
X(k)
1 ejN X
2k
N
1 ej( N )
k=0
(c) Show further that the above expression is equivalent to

X(ej ) =
N1

2k
1 X
X(k)R ej ( N )
N
k=0
where R(e ) is the DTFT of a rectangular pulse of width N , i.e.,

R(ej ) = ej(
N 1
2
) sin (N/2)
sin (/2)
Remark. The expression in part (c) provides an interpolation formula that allows us to recover the
DTFT of x(n) from its DFT sequence. We associate with each sample X(k) the sinc-like function
R(ej ), which is centered at the location of X(k). We subsequently combine the contributions from
all samples. Note that R(ej ) is periodic with period 2.
Problem 17.31 Consider the signal x(n) = |n| , where || < 1.
(a) Compute the DTFT of x(n).
(b) Let X(k) denote the 4-point DFT of x(n). Compute X(k) for k = 0, 1, 2, 3.
(c) Compute the first 4 values (n = 0, 1, 2, 3) of the periodic sequence x1 (n) =
r=
|n4r| .
CHAPTER
18
Properties of the DFT
he Discrete Fourier Transform (DFT) has several useful properties. This chapter establishes some of these properties and provides illustrative examples. Thus, consider a causal
sequence x(n) with duration L N . According to the definition (17.21), the N point
DFT of x(n) is given by
X(k) =
N
1
X
x(n)ej
2k
N n
n=0
k = 0, 1, . . . , N 1
(18.1)
This expression takes N samples of x(n) (appended with zero samples if L < N ) and
transforms them into N samples of X(k).
The case of causal and finite-duration sequences, x(n), is the situation that we encounter
most frequently in practice. Remember, however, that we explained in Sec. 17.3 explained
the procedure that we should follow to evaluate the N point DFT of a general sequence
x(n), which may neither be causal nor have finite-duration. Specifically, we follows these
two steps:
(a) We first embed x(n) into a periodic sequence, xp (n), of period N using (17.19).
This task is achieved by shifting x(n) to the left and to the right by multiples of N
and by adding up all sequences to obtain xp (n).
(b) We subsequently compute the N point DFT samples by using the relation
X(k) =
N
1
X
xp (n)ej
n=0
2k
N n
k = 0, 1, . . . , N 1
(18.2)
with x(n) in (18.1) replaced by xp (n) in (18.2). This step uses only the N samples
of xp (n) that are within the interval 0 n N 1. The calculation results in N
values of X(k), k = 0, 1, . . . , N 1.
Nevertheless, we shall assume from now on that we are dealing with sequences x(n) that
are causal and have finite-duration L N . Whenever this is not the case, we simply
replace x(n) by the periodic sequence xp (n).
487
Copyright
488
18.1 PERIODICITY OF THE DFT
CHAPTER 18
PROPERTIES
OF THE
DFT
Assume we evaluate the DFT of x(n) in (18.1) for all values of k and not only over k =
0, 1, . . . , N 1, namely,
X(k) =
N
1
X
x(n)ej
2k
N n
k = . . . , 2, 1, 0, 1, 2, . . .
n=0
(18.3)
Then, it can be easily verified that the DFT sequence X(k) is periodic with period N , i.e.,
X(k + N ) = X(k)
for all integers k
(18.4)
This property was illustrated earlier in the construction shown in Fig. 17.3, and was also
established in (17.22).
Proof: Note that the complex exponential sequence ej2kN/n is periodic with period N over the
variable k since
2(k+N )
2k
j 2k
n
N
ej N n = |ej2n
= ej N n
{z } e
=1
Accordingly, we get
X(k + N )
N1
X
x(n)ej
2(k+N )
n
N
n=0
N1
X
x(n)ej
2(k)
n
N
n=0
X(k)
Figure 18.1 illustrates the periodicity of the N point DFT; the figure plots the samples of a
real-valued DFT. It is because of the periodicity property (18.4) that we usually display the
magnitude and phase plots of the DFT over a one period interval. This interval is generally
chosen as 0 k N 1.
Example 18.1 (Periodicity in time and frequency)
Consider the causal and finite-duration sequence (for which L = 4):
x(n) = (n) + 0.5(n 1) + 0.5(n 3)
Evaluating the 4point DFT of x(n) we get
X(0)
3
X
n=0
x(n) = 2

3
2
X(1)
1 + 0.5 ej 2 + ej
X(2)
1 + 0.5 ej + ej3 = 1 1 = 0
X(3)
1 + 0.5 ej

3
2
+ ej
9
2
=1+0=1

=1+0 =1
489
SECTION 18.2
X(k) (periodic)
USEFUL
PROPERTIES
k
k = (N 1)
k=0
k =N 1
X(k) (one period)
N 1
FIGURE 18.1 The N point DFT is periodic of period N . The top figure shows the periodic
structure of the DFT while the bottom figure shows one period of the DFT.
Recall that this 4point DFT is also the DFT of the periodic sequence, xp (n), which we obtain from
x(n) by shifting the samples of x(n) to the left and to the right by multiples of N = 4 and adding
all shifted sequences. The periodic sequence, xp (n), and the corresponding periodic 4point DFT
sequence, X(k), are shown in Fig. 18.2.
Therefore, we can view the N point DFT of a sequence as a transform that associates a
periodic sequence X(k) with a periodic sequence xp (n); both sequences have period N .
Due to this periodicity, we often limit ourselves to working with samples that lie within
one period of each sequence, namely, the N samples of X(k) and xp (n) that lie within the
intervals 0 n, k N 1.
18.2 USEFUL PROPERTIES

The DFT shares several properties with the DTFT. A summary of these properties is given
in Table 18.1. For example, the first two lines of the table start from two generic causal
sequences x(n) and y(n) of finite duration L N , and the subsequent lines provide the
DFT of combinations and transformations of these sequences.
18.2.1 Linearity
Consider the third line of the table. It states that the N point DFT of a linear combination
of two sequences is given by the same linear combination of their N point DFTs, i.e.,
ax(n) + by(n) aX(k) + bY (k)
(18.5)
490
x(n)
CHAPTER 18
PROPERTIES
OF THE
DFT
xp (n)
1
X(k)
2
FIGURE 18.2 A causal and finite-duration sequence x(n) with L = 4 is shown in the top plot.
Its periodic embedding, xp (n), is shown in the middle plot and the resulting 4point periodic DFT
sequence X(k) is shown in the bottom plot. Note in this case that the samples of x(n) and xp (n)
agree over the one-period interval 0 n N 1.
for any scalars a and b.

W (k)
N1
X
w(n)ej
2k n
N
n=0
N1
X
[ax(n) + by(n)]ej
n=0
N1
X
x(n)e
n
j 2k
N
n=0
2k n
N
+ b
N1
X
y(n)e
n
j 2k
N
n=0
aX(k) + bY (k)
491
TABLE 18.1 Several properties of the N point DFT. The variables n and k lies within the
intervals 0 n, k N 1.
Causal sequences
of duration L N
N point DFT
1.
x(n)
X(k)
2.
y(n)
Y (k)
3.
ax(n) + by(n)
aX(k) + bY (k)
4.
x[(n no ) mod N ]
ej
5.
ej
6.
cos
7.
2ko
n
N
linearity
X(k)
circular time shift
X[(k ko ) mod N ]
circular frequency shift
1
1
X[(k ko ) mod N ] + X[(k + ko ) mod N ]
2
2
modulation
x(n mod N )
X(k mod N )
time reversal
8.
x (n)
X (k mod N )
conjugation in time
9.
x (n mod N )
X (k)
conjugation in frequency
10.
x(n) y(n)
X(k)Y (k)
circular convolution
11.
x(n)y(n)
1
(X(k) Y (k))
N
product of sequences
P
1 N1
X(k)Y (k)
N k=0
Parsevals relation
12.
N1
P
x(n)
2no
k
N
Property
2ko
n
N
x(n)
x(n)y (n)
n=0

Consider the sequence x(n) that is shown in the top plot of Fig. 18.3 and let us determine its 4point
DFT.
While we can evaluate the 4point DFT of x(n) directly from the definition (18.1), we instead
appeal to the linearity property (18.5) to illustrate it. Thus, note that the sequence x(n) can be
regarded as the sum of the two rectangular pulses shown in the middle and bottom plots of Fig. 18.3:
x(n) = x1 (n) + x2 (n)
where x1 (n) has duration L = 2:
(
x1 (n) =
1,
0,
0n1
otherwise
1,
0,
0n3
otherwise
and x2 (n) has duration L = 4:

(
x2 (n) =
SECTION 18.2
USEFUL
PROPERTIES
492
x(n)
CHAPTER 18
PROPERTIES
OF THE
DFT
2
1
1
x1 (n)
1
1
x2 (n)
2
1
1
FIGURE 18.3 The sequence x(n) in the top plot can be expressed as the sum of the two
rectangular pulses in the middle and bottom plots.
We already know from the discussion in Example 17.7, that the 4point DFTs of the rectangular
pulses x1 (n) and x2 (n) are given by
8
< 2,
X1 (k) =
: e
k=0
jk/4
sin (k/2)
,
sin (k/4)
k = 1, 2, 3
and
X2 (k) = 4(k),
k = 0, 1, 2, 3
since the duration of x2 (n) agrees with the value of N . We conclude that
8
< 6,
X(k) =
: e
k=0
jk/4
sin (k/2)
,
sin (k/4)
k = 1, 2, 3
In other words,
X(0)
X(1)
X(2)
1j =
X(3)
1+j =
2ej/4
2ej/4
493
The magnitude and phase plots of the resulting 4point DFT are shown in Fig. 18.4.
SECTION 18.2
USEFUL
PROPERTIES
|X(k)|
6
4
2
2
X(k)
1
FIGURE 18.4 The magnitude (top) and phase (bottom) plots of the 4point DFT of the sequence
x(n) from Fig. 18.3.
18.2.2 Circular Time Shifts

Consider now the fourth line in Table 18.1. It relates to circular shifts in the time domain
and their effect on the DFT of a sequence. Before establishing the property, we need to
explain what circular shifts are and motivate the need for their use in the context of the DFT.
Definition
Consider a causal sequence x(n) with duration L N . We focus on the samples within
the window 0 n N 1. The samples from n = L to n = N 1 will be equal to zero
when L N . One such sequence is displayed in Fig. 18.5 (top plot) and will be used to
illustrate circular shifts.
When we delay the sequence x(n) by one sample, the traditional shift operation simply
displaces all samples of x(n) to the right by one position; this operation results in the
sequence x(n 1). The situation is illustrated in the bottom plot of Fig. 18.5. Observe, for
example, how a new zero sample moves into location n = 0 and that the zero sample that
used to occur at location n = 5 in x(n) has now moved to location n = 6.
A circular operation, on the other hand, is an operation that keeps the samples of x(n)
pinned within the window 0 n N 1. When we shift the samples of x(n) circularly
to the right by one sample, the right-most sample at n = 5 that is about to leave the
window from the right is wrapped around and moved back inside the window at location
n = 0. Figure 18.6 illustrates the result of three successive circular shifts to the right of
the sequence x(n). Observe how the samples leaving the window 0 n N 1 from
the right are always wrapped around and stay always within the same window.
In a similar fashion, Fig. 18.7 illustrates the result of three successive circular shifts of
the same sequence x(n) albeit to the left. Now, the samples leaving the window 0 n
N 1 from the left are wrapped around and enter the window from the right.
494
CHAPTER 18
PROPERTIES
OF THE
DFT
x(n)
1
x(n 1) (traditional shift)

1
FIGURE 18.5 Consider a sequence x(n) of length L = 4 and choose N = 4. The samples of
x(n) lie within the window 0 n 5. The two zero samples at n = 4 and n = 5 are indicated
explicitly with bullets. Different colors are used for the three bullets at zero in order to facilitate the
tracking of their movement as circular shifts occur.
How do Circular Shifts Arise?

The main reason why circular shifts arise in the context of the N point DFT of a sequence
is because standard shifts of the periodic sequence xp (n), correspond to circular shifts of
the original sequence, x(n). Figure 18.8 shows a causal sequence x(n) (top plot) with
L = 4 samples along with its periodic embedding, xp (n), with period N = 4 (middle
plot). The samples of xp (n) within 0 n 3 are marked with a dotted box around them.
The bottom plot shows the sequence xp (n) but shifted to the right by one sample. This is
the traditional shift operation and all samples of xp (n) move to the right by one place. If
we examine the samples that lie within the window 0 n 3 in xp (n 1) we find that
they could have been obtained by circularly shifting to the right the samples of x(n).
Notation
To denote a circular shift of k samples, we employ the notation
x [(n k) mod N ]
in terms of the modulo N operation, which is defined as follows. Given two integer numbers, m and N , the notation
m mod N
refers to the remainder, r, that is obtained when dividing m by N . For example,
11 mod 6
4 mod 6
495
SECTION 18.2
USEFUL
PROPERTIES
x(n 1) (circular shift)

1

1

1
FIGURE 18.6
Three successive circular shifts to the right of the sequence x(n) of Fig. 18.5.
The result of the modulo operation is always an integer in the interval 0 r N 1.

When m is larger than N , we subtract from m sufficient multiples of N until the remainder
lies within this interval. Thus, note that
22 = (3 6) + 4
(18.6)
22 mod 6 = 4
(18.7)
so that
Sometimes, we shall encounter dividend values, m, that are negative integers. In such
cases, we add to m sufficient multiples of N until the remainder lies within the interval
0 r N 1. For example
11 + 2 6 = 1
(18.8)
so that
11 mod 6 = 1
(18.9)
Using the definition of the modulo operation, let us now examine why an operation of the
form
x [(n 1) mod N ]
496
CHAPTER 18
PROPERTIES
OF THE
DFT
x(n + 1) (circular shift)

1

1

1
FIGURE 18.7
Three successive circular shifts to the left of the sequence x(n) of Fig. 18.5.
corresponds to a circular shift to the right by one position. Indeed, for illustration purposes,
select N = 6 and let y(n) denote the resulting sequence
y(n) = x [(n 1) mod 6]
Then, the samples of y(n) over 0 n 5 are given by
y(0)
y(1)
y(2)
y(3)
y(4)
y(5)
=
=
=
=
=
=
x(1 mod 6)
x( 0 mod 6)
x( 1 mod 6)
x( 2 mod 6)
x( 3 mod 6)
x( 4 mod 6)
=
=
=
=
=
=
x(5)
x(0)
x(1)
x(2)
x(3)
x(4)
and it is seen that the sample of y(n) at time n = 0 coincides with the sample of x(n) at
time n = 5, as expected from a circular shift of x(n) by one position to the right.
497
SECTION 18.2
xp (n)
USEFUL
PROPERTIES
xp (n 1) (traditional shift)
1
FIGURE 18.8 A periodic sequence xp (n) with period N = 4 (top plot). The samples of xp (n)
within 0 n 3 are marked with a dotted box around them. The bottom plot shows the same
sequence but shifted to the right by one sample. This is the traditional shift operation and all samples
of xp (n) move to the right by one place.
Example 18.3 (Modulo operation)

Let N = 6 and no = 2. Let us examine the range of values that the modulo operation
(n no ) mod N
assumes as n varies over the range 0 n N 1. Let
r = (n 2) mod 6
Then simple calculations reveal that
n=0
r=4
n=1
r=5
n=2
r=0
n=3
r=1
n=4
r=2
n=5
r=3
In other words, the remainder r assumes all values in the interval 0 r N 1 as n varies over
the same interval.
Let us repeat the same example for some value of no that is larger than N , say no = 9. Thus, let
r = (n 9) mod 6,
0n5
498
Again, some simple calculations reveal that
CHAPTER 18
n=0
PROPERTIES
OF THE
DFT
r=3
n=1
r=4
n=2
r=5
n=3
r=0
n=4
r=1
n=5
r=2
and we again see that the remainder r assumes all values in the interval 0 r N 1 as n varies
over the same interval.
Circular Time-Shift Property

Let us now return to the fourth line in Table 18.1. It establishes the transform property:
x[(n no ) mod N ]
ej
2no
N
X(k)
(18.10)
In other words, if the original sequence x(n) is circularly shifted in time by no samples (where no may be a positive or negative integer; it may also be smaller or larger
than N in magnitude), then the phase of the corresponding DFT sequence is modified
by the factor ej2no k/N . Observe that the magnitude of the DFT is not modified since
ej2no k/N X(k) and X(k) have the same magnitude for every . We therefore say that
circular shifts in the time-domain correspond to phase change in the frequency domain.
Proof: For any integer n in the interval 0 n N 1, let us express n no in the form
n no = aN + r
for some integer a and where r denotes the result of the modulo operation:
(n no ) mod N = r
The values of a and r vary with the value of n. So, strictly speaking, we should write a(n) and r(n)
instead of a and r to emphasize their dependency on n. However, this level of detail is unnecessary
for our argument and is omitted for simplicity of notation.
The values of r span the interval 0 r N 1 as n varies over the same interval. Thus, let
w(n) = x[(n no ) mod N ]. Then
W (k)
N1
X
w(n)ej
2k n
N
n=0
N1
X
n=0
=
=
x[(n no ) mod N ] ej
no
j 2k
N
j 2k
no
N
j 2k
no
N
ej
2k n
o
N
N1
X
n=0
N1
X
2k n
N
x[(n no ) mod N ] e
!
x(r) e
(aN+r)
j 2k
N
x(r) e
j 2k
r
N
r=0
N1
X
(nno )
j 2k
N
, using r = (n no ) mod N
r=0
X(k)
where in the third equality we multiplied and divided by the factor ej2kno /N , and in the fourth
equality we replaced (n no ) by aN + r and then used the fact that
ej
2k (nn )
o
N
ej
2k (aN+r)
N
= ej
2k r
N
Example 18.4 (Illustrating the circular time-shift property)

Consider the sequence x(n) that is shown in the top plot of Fig. 18.9. We already evaluated its
4point DFT in Example 18.2 and found that
X(0)
X(1)
X(2)
X(3)
2ej/4
2ej/4
The magnitude and phase plots of this DFT were shown in Fig. 18.4. Now assume that we circularly
shift x(n) to the right by one sample and consider the resulting sequence
y(n) = x[(n 1) mod 4]
According to property (18.10), the 4point DFT of y(n) is related to the 4point DFT of x(n) as
follows:
k
Y (k) = ej 2 X(k), 0 k 3
Consequently, using the values for X(k), we get
Y (0)
Y (1)
Y (2)
Y (3)
X(0) = 6
ej 2 X(1) =
2ej3/4
ej X(2) = 0
3
ej 2 X(3) = 2ej5/4 = 2ej3/4
Figures 18.9 and 18.10 illustrate the sequences x(n) and y(n) and their respective 4point DFTs.
499
SECTION 18.2
USEFUL
PROPERTIES
500
x(n)
CHAPTER 18
PROPERTIES
OF THE
DFT
2
1
1
y(n) = x[(n 1) mod 4]

2
1
1
FIGURE 18.9 The sequence x(n) in the top plot is circularly shifted by one sample to the right
in the bottom plot.
18.2.3 Circular Frequency Shifts

Consider now the fifth line in Table 18.1. It establishes the transform property:
ej
2ko
N
x(n)
X[(k ko ) mod N ]
(18.11)
The result states that if the phase of the original sequence is modified by adding a linear
component to it, in the form of 2nko /N , then the corresponding DFT is obtaining by
circularly shifting the DFT of the original sequence by ko samples. We therefore say that
phase change in the time-domain corresponds to circular shifts in the frequency domain
and vice-versa. This property is the dual of the circular time-shift property (18.10).
Proof: For any integer k in the interval 0 k N 1, let us express k ko in the form
k ko = aN + r
for some integer a and where r denotes the result of the modulo operation
(k ko ) mod N = r
Again, the values of a and r vary with k. So, strictly speaking, we should write a(k) instead of a and
r(k) instead of r to emphasize their dependency on k. However, this level of detail is unnecessary
for our argument and is omitted for simplicity of notation.
The values of r span the interval 0 r N 1 as k varies over the same interval. Thus, let
w(n) = ej
2ko
n
N
x(n)
501
X(k)
|X(k)|
SECTION 18.2
USEFUL
PROPERTIES
6
4
2
2
1
2
2
|Y (k)|
Y (k)
6
3
4
4
2
1
1
k
3
4
FIGURE 18.10 The magnitude and phase plots of the 4point DFTs of the sequences x(n) and
y(n) = x[(n 1) mod 4] from Fig. 18.9.
Then
W (k)
N1
X
w(n)ej
2k n
N
n=0
N1
X
ej
2n k
o
N
n=0
N1
X
x(n) ej
0k N 1

x(n)
ej
2k n
N
2(kko )
n
N
n=0
N1
X
x(n) ej
2n (aN+r)
N
x(n) ej
2n r
N
x(n) ej
2n [(kk )
o
N
n=0
N1
X
n=0
N1
X
mod N]
n=0
X[(k ko ) mod N ]
502
Example 18.5 (Illustrating the circular frequency-shift property)
CHAPTER 18
PROPERTIES
OF THE
DFT
Consider again the sequence x(n) that is shown in the top plot of Fig. 18.3. We already evaluated its
X(0)
X(1)
X(2)
X(3)
2ej/4
2ej/4
The magnitude and phase plots of this DFT were shown in Fig. 18.4. Now let N = 4 and ko = 1
and introduce the sequence
y(n) = ej
2nko
N
x(n) = ej
n
2
x(n)
whose samples are related to those of x(n) as follows:

y(0)
y(1)
y(2)
y(3)
=
=
=
=
x(0)
ej/2 x(1)
ej x(2)
ej3/2 x(3)
=
=
=
=
2
2ej/2
ej
ej/2
All other samples of y(n) are equal to zero.

According to property (18.11), the 4point DFT of y(n) is related to the 4point DFT of x(n)
as follows:
Y (k) = X[(k 1) mod 4]
In other words, the DFT of x(n) is circularly shifted to the right by one sample. Figures 18.11
and 18.12 illustrate the sequences x(n) and y(n) and their respective 4point DFTs. Note that since
the samples of y(n) are now complex-valued, we are plotting both the magnitude and phase plots of
y(n).
x(n)
2
1
1
|y(n)|
y(n)
3
1
FIGURE 18.11 The phase of the sequence x(n) in the top plot is modified to yield the sequence
y(n) = ejn/2 x(n) in the bottom plot.
503
X(k)
|X(k)|
SECTION 18.2
USEFUL
PROPERTIES
6
4
2
2
1
2
2
|Y (k)|
Y (k)
6
4
2
1
2
1
y(n) = ejn/2 x(n) from Fig. 18.11.
18.2.4 Modulation
Consider now the sixth line in Table 18.1. It establishes the transform property:
1
1
2
2
(18.12)
In other words, if the sequence x(n) is modulated by a cosine sequence, then its N point
DFT is scaled by 1/2 and circularly shifted to the left and to the right by ko samples.
cos
2ko
N n
x(n)
Proof: Let
w(n) = cos
2ko
n x(n)
N
Using Eulers relation (3.11) we have

w(n) =
2n
1 j 2n
1
e N ko x(n) + ej N ko x(n)
2
2
Invoking the linearity and frequency-shift properties (18.5) and (18.11) we conclude that
W (k) =
1
1
2
2
504
Likewise, it holds that
CHAPTER 18
PROPERTIES
OF THE
DFT

2n
ko x(n)
N
1
1
X[(k ko ) mod N ]
X[(k + ko ) mod N ]
2j
2j
(18.13)
where the proof now requires that we employ the alternative form (3.12) of Eulers relation.
sin
Example 18.6 (Illustrating the modulation property)

X(0)
X(1)
X(2)
X(3)
2ej/4
2ej/4
The magnitude and phase plots of this DFT were shown in Fig. 18.4. Now let N = 4, ko = 1, and
introduce the sequence

y(n) = sin

2ko
n
x(n)
n x(n) = sin
N
2
whose samples are related to those of x(n) as follows:

y(0)
y(1)
y(2)
y(3)
=
=
=
=
0

sin 2 x(1)
sin()x(2)

sin 3
x(3)
2
=
=
=
2
0
1
All other samples of y(n) are equal to zero. Obviously, the sequence y(n) is rather trivial, with only
two nonzero samples, and we can evaluate its 4point DFT directly from the definition (18.1). Here,
however, we would like to illustrate the use of the modulation property (18.12).
According to (18.12), the 4point DFT of y(n) is related to the 4point DFT of x(n) as follows:
Y (k) =
1
1
X[(k 1) mod 4]
X[(k + 1) mod 4]
2j
2j
In other words, the DFT of x(n) should be circularly shifted to the right and to the left by one sample
and the results should be combined after scaling by 1/2j.
Let us consider first the operation
Z1 (k) =
1
X[(k 1) mod 4]
2j
and let us evaluate the samples of Z1 (k) in terms of those of X(k) over 0 k 3. Thus, note that
the factor 1/2j can be expressed in polar form as
1
1
= j/2 = ej/2
2j
2
Using the values of X(k) we then get
Z1 (0)
Z1 (1)
Z1 (2)
Z1 (3)
1
X(1
2j
1
X(0
2j
1
X(1
2j
1
X(2
2j
=
=
=
=
mod 4)
mod 4)
mod 4)
mod 4)
=
=
=
=
1 j/2
e
X(3)
2
1 j/2
e
X(0)
2
1 j/2
e
X(1)
2
= 21 ej/2 X(2)
505
2 j/4
e
2
3ej/2
2 j3/4
e
2
=
=
=
=
SECTION 18.2
USEFUL
PROPERTIES
Likewise, consider the second operation

1
X[(k + 1) mod 4]
2j
Z2 (k) =
Using the values of X(k) we get
Z2 (0)
Z2 (1)
Z2 (2)
Z2 (3)
=
=
=
=
1
X(1
2j
1
X(2
2j
1
X(3
2j
1
X(4
2j
mod
mod
mod
mod
4)
4)
4)
4)
1 j/2
e
X(1)
2
1 j/2
e
X(2)
2
1 j/2
e
X(3)
2
1 j/2
e
X(0)
2
=
=
=
=
=
=
=
=
2 j3/4
e
2
2 j/4
e
2
j/2
3e
Now using
Y (k) = Z1 (k) Z2 (k)
we arrive at
Y (0)
Y (1)
Y (2)
=
=
Y (3)
2
ej/4 ej3/4
2
3ej/2
0

j3/4
2
e
ej/4
2
j/2
0 3e
=
=
=
=
2
2
j/2
3e

2
2
2
j/2
3e
ej
3ej/2
Figures 18.13 and 18.14 illustrate the sequences x(n) and y(n) and their respective 4point DFTs.
18.2.5 Circular Time Reversal
The circular time reversal of a sequence is defined by the operation

x(n mod N )
where, as usual, the result of the modulo operation lies within the interval 0 r N 1.
Consider, for example, N = 4. Then the samples of the sequence
y(n) = x(n mod N )

are related to the samples of x(n) as follows:
y(0) = x(0 mod 4) = x(0)
y(1) = x(1 mod 4) = x(3)
y(2) = x(2 mod 4) = x(2)
y(3) = x(3 mod 4) = x(1)
Observe in particular that the sample at location n = 0 continues to be x(0).
One convenient way to arrive at the circular time-reversal of a sequence x(n) is to
start by placing markers on a circle going from one sample to another, say in a counterclockwise direction, from sample x(0) to x(1) to x(2) to x(3), and so forth, as illustrated in
Fig. 18.15 for the case of a sequence with four samples. Then, to obtain the samples for the
circularly time-reversed sequence, y(n), we simply start from x(0) and visit the samples in
506
x(n)
CHAPTER 18
PROPERTIES
OF THE
DFT
2
1
1
y(n) = sin
n
n
2
x(n)
2
1
3
1
FIGURE 18.13 The sequence x(n) in the top plot is modulated to yield the sequence y(n) =
sin(n/2)x(n) in the bottom plot.
the opposite direction (i.e., in the clockwise direction). In this example, we would end up
visiting x(0), x(3), x(2) and x(1), in that order, and these samples constitute the samples
of the sequence y(n). Figure 18.16 illustrates the circular time-reversal operation on a
sequence x(n). The samples of x(n) are color-coded to facilitate tracking their location.
Now consider the seventh line in Table 18.1. It establishes the transform property:
x(n mod N )
X(k mod N )
(18.14)
In other words, if the original sequence x(n) is reversed in time circularly, then the corresponding N point DFT is also reversed in frequency circularly.
Proof: Let
r = n mod N
and write
n = aN + r
for some integer a. The values of r span the interval 0 r N 1 as n varies over the same
interval. Now consider
w(n) = x(n mod N )
507
SECTION 18.2
X(k)
|X(k)|
USEFUL
PROPERTIES
6
4
2
2
|Y (k)|
Y (k)
1
1
k
2
y(n) = sin(n/2) x(n) from Fig. 18.13.
Then
W (k)
N1
X
w(n)ej
2k n
N
n=0
N1
X
0k N 1
x(n mod N ) ej
2k n
N
n=0
N1
X
x(r) ej
2k (aN+r)
N
x(r) ej
2k r
N
r=0
N1
X
using n = aN + r
r=0
N1
X
x(r) ej
2(kN )
r
N
r=0
N1
X
x(r) ej
replacing k by k N
2(N k)
r
N
r=0
=
=
X(N k)
X(k mod N )
where the last equality is obviously true for all k in the range 1 k N 1 [with regards to k = 0,
we recall that the DFT sequence, X(k), has period N and, hence, X(N ) = X(0)].
508
x(1)
CHAPTER 18
PROPERTIES
OF THE
DFT
x(0)
x(2)
x(3)
FIGURE 18.15 Markers are placed on a circle and the 4 samples of a sequence x(n) are visited
in a counter-clockwise direction. By reading the samples in the opposite clockwise direction, we
obtain the samples of the circularly time-reversed sequence y(n) = x(n mod 4).
Example 18.7 (Illustrating the circular time-reversal property)

X(0)
X(1)
X(2)
X(3)
2ej/4
2ej/4
The magnitude and phase plots of this DFT were shown in Fig. 18.4. Now consider the time-reversed
sequence
y(n) = x(n mod 4)
According to property (18.14), the 4point DFT of y(n) is related to the 4point DFT of x(n) via
the relation
Y (k) = X(k mod 4)
Figure 18.17 illustrates the effects of the circular time-reversal operation on x(n) and the resulting
magnitude and phase plots of Y (k).
18.2.6 Complex Conjugation in Time and Frequency

Consider now rows eight and nine from Table 14.1. They deal with the complex conjugation property of sequences, both in time and frequency. The results establish the following
transform property:s:
x (n)
X (k mod N ) (conjugation in time)
(18.15)
and
x (n mod N )
X (k) (conjugation in frequency)
(18.16)
509
SECTION 18.2
x(n)
USEFUL
PROPERTIES
x(n)
(traditional time-reversal)
3 2 1
x(n mod 4)
(circular time-reversal)
FIGURE 18.16 A sequence x(n) is reversed in time by flipping its samples around the vertical
axis (middle plot). The sequence x(n) is circularly reversed in time by moving the samples of x(n)
back to lie within the interval 0 n N 1 (bottom plot); this can be accomplished by shifting
the samples of x(n) circularly to the right N times. The operation that goes from the top plot to
the bottom plot is called circular time-reversal. Note that the sample at time n = 0 remains intact.
In the above relations, the notation x (n) refers to complex conjugating the term x(n).
Clearly, when x(n) is real-valued for some n, then x (n) = x(n) at that value of n.
Likewise, X (k) denotes the complex conjugation of X(k).
Result (18.16) states that if the samples of the time-domain sequence, x(n), are complex
conjugated, then the samples of X(k) are complex conjugated and, in addition, they are
reversed circularly in frequency. A similar statement follows from (18.16). Specifically,
if the samples of the time-domain sequence, x(n) are complex-conjugated and reserved
circularly in time, then the samples of X(k) are complex-conjugated.
510
y(n) = x(n mod 4)
CHAPTER 18
PROPERTIES
OF THE
DFT
2
1
1
|Y (k)|
6
4
2
2
Y (k)
3
1
FIGURE 18.17 The top plot shows the sequence that is obtained by circularly time-reversing the
sequence x(n) from Fig. 18.3. The middle and bottom plots show the corresponding magnitude and
phase plots of time-reversed sequence, y(n) = x(n mod 4).
Proof: Let w(n) = x (n). Then

W (k)
N1
X
w(n)ej
2k n
N
0k N 1
n=0
N1
X
x (n) ej
2k n
N
n=0
Conjugating both sides of the above equality gives

W (k)
N1
X
x(n) ej
2k n
N
x(n) ej
2n (kN)
N
n=0
N1
X
n=0
N1
X
x(n) ej
2n (Nk)
N
n=0
=
=
X(N k)
X(k mod N )
since X(k) has period N , and where in the second equality we replaced k by (k N ) since, for any
k,
e
j 2n
(kN)
N
j2n
j 2kn
N
= e
j 2kn
N
Therefore, conjugating again we get

W (k) = X (k mod N )
as desired. A similar argument establishes (18.16). Let now w(n) = x (n mod N ). Then
W (k)
N1
X
w(n)ej
2k n
N
n=0
N1
X
0k N 1
x (n mod N ) ej
2k n
N
n=0
Conjugating both sides of the above equality gives

W (k)
N1
X
x(n mod N ) ej
2k n
N
n=0
N1
X
x(r) ej
2k (aN+r)
N
x(r) ej
2k r
N
r=0
N1
X
using n = aN + r
r=0
X(k)
and, therefore, by conjugating again,

W (k) = X (k)
as desired.
Example 18.8 (Illustrating the complex conjugation property)

Consider the following sequence with complex-valued sample values,
y(n) = 2(n) + 2ej/2 (n 1) (n 2) + ej/2 (n 3)
We encountered this sequence earlier in Example 18.5. The magnitude and phase plots of the sequence y(n) were shown in Fig. 18.11, while the magnitude and phase plots of the 4point DFT
Y (k) were shown in Fig. 18.12. These plots are grouped together in Fig. 18.18.
Let N = 4 and introduce the sequences
z(n) = y (n)
and
w(n) = y (n mod 4)
We are interested in determining the 4point DFTs of z(n) and w(n) in terms of Y (k). According
to (18.15) and (18.16) we get
Z(k) = Y (k mod 4)
and
W (k) = Y (k),
0k3
Using these relations, we display in Figs. 18.19 and 18.20 the magnitude and phase plots of Z(k)
and W (k).
511
SECTION 18.2
USEFUL
PROPERTIES
512
CHAPTER 18
PROPERTIES
OF THE
DFT
|y(n)|
y(n)
3
1
|Y (k)|
Y (k)
6
4
2
1
2
1
FIGURE 18.18 The top row shows the magnitude and phase plots of the sequence y(n) from
Example 18.8, while the bottom row shows the magnitude and phase plots of the corresponding
4point DFT, Y (k).
18.2.7 Circular Convolution

When we deal with the DFT, it is useful to introduce a new notion of convolution known
as circular convolution as opposed to linear convolution, which was studied extensively
earlier in Chapters 5 and 6. The linear convolution operation was seen to be a fundamental tool in the study of linear time-invariant (LTI) systems; it enabled us to evaluate the
response of an LTI system to an arbitrary input sequence through the computation of the
linear convolution of the input sequence with the impulse response sequence of the system.
In this section, we introduce another convolution operation known as circular convolution, and then show that the DFT technique provides an efficient way for evaluating circular convolutions. We further explain that linear convolutions can be evaluated by means of
circular convolutions. In this way, we end up with efficient ways for computing linear convolutions as well. So let us start by defining what the circular convolution of two sequences
is. It will become apparent that the steps involved in computing the circular convolution of
two sequences are similar to the steps involved in computing the standard linear convolution of these same sequences except that traditional time-shift and time-reversal operations
are now replaced by circular shift and circular reversal operations.
Definition
Consider two causal and finite-duration sequences, {x(n), h(n)}, of equal length N so
that their nonzero samples exist over the interval 0 n N 1. If any of the sequences
happens to have length less than N then we pad it with zeros and bring the length up to
N ; it is important that both sequences have the same length N for circular convolutions.
The circular convolution of x(n) and h(n) is then defined as another causal sequence, say,
513
SECTION 18.2
|z(n)|
USEFUL
PROPERTIES
z(n)
1
1
2
3
|Z(k)|
Z(k)
6
4
2
1
FIGURE 18.19 These plots illustrate complex conjugation in the time domain. The top row
shows the magnitude and phase plots of the sequence z(n) = y (n) from Example 18.8, while the
bottom row shows the magnitude and phase plots of the corresponding 4point DFT, Z(k).
y(n), also of length N , and is denoted by the notation:

y(n) = x(n) h(n)
(circular convolution)
(18.17)
with the symbol used as opposed to the symbol that we use for linear convolutions
(recall (5.6)). The samples of y(n) over the interval 0 n N 1 are evaluated as
follows:
y(n) =
NP
1
x(m)h[(n m) mod N ] , n = 0, . . . , N 1
(18.18)
m=0
Comparing this expression with the definition (5.11) for the linear convolution of causal
sequences we see similarities in the expressions albeit with three striking differences:
(a) First, the index of the sequence h() in (18.18) is (n m) mod N (using a modulo
operation) and not (n m) alone. In this way, the time index of h() in (18.18) will
always be a value r within the range 0 r N 1.
(b) The time-shifts and the time-reversals that are involved in evaluating the term h[(n
m) mod N ] are all circular in nature.
(c) The circular convolution sequence, y(n), has N sample values and, therefore, all
three sequences, {x(n), h(n), y(n)}, have the same duration. Recall that the linear
514
|w(n)|
CHAPTER 18
PROPERTIES
OF THE
DFT
w(n)
2
1
|W (k)|
W (k)
6
4
2
1
FIGURE 18.20 These plots illustrate complex conjugation in the frequency domain. The top row
shows the magnitude and phase plots of the sequence w(n) = y (4 n) for Example 18.8, while
the bottom row shows the magnitude and phase plots of the corresponding 4point DFT, W (k).
convolution of two sequences, x(n) and h(n) of length N each, can be as long as
2N 1 samples.
Graphical Method of Evaluation
As was the case with linear convolutions, the circular convolution of two sequences can
also be evaluated graphically by applying the steps outlined below; these steps are illustrated in the numerical example that follows:
(a) First, we plot the sequences h(m) m and x(m) m over the interval 0 m
N 1. Note that we are denoting the independent variable by m. Therefore, the
horizontal axis will be the m axis.
(b) Then we plot h(m mod N ). In other words, we circularly reverse the sequence
h(m).
(c) We subsequently compute the sequence x(m)h(m mod N ) by multiplying the
sequences x(m) and h(m mod N ) sample-by-sample. We add the samples of the
product sequence x(m)h(m mod N ). The resulting value would be y(0), namely,
the value of the circular convolution sum at time n = 0.
(d) Next, we circularly shift h(m mod N ) by one time unit to the right in order to
obtain h[(1 m) mod N ]. We again compute the product sequence x(m)h[(1
m) mod N ] and add its sample values. This calculation provides y(1); the value of
the circular convolution sum at time n = 1.
(e) We repeat the above procedure by circularly shifting h(m mod N ) further to the
right and computing the product sequences x(m)h(n m mod N ) each time, and
adding the resulting samples. This calculation provides the values of y(n) for the
various values of n inside the interval 0 n N 1.
Example 18.9 (Circular convolution of two sequences)

Let us evaluate the circular convolution the following two sequences:
n
x(n) =
1 , 2, 0.5
and
h(n) =
1 , 1, 2
for N = 3, where we are using the box notation to indicate the location of the sample at time n = 0.
The sequences are illustrated in the left column of Fig. 18.21.
y(m)
x(m)
3.5
3
2
1/2
h(m mod 3)
h(m)
2
h[(1 m) mod 3]
h[(2 m) mod 3]
2
2
1
1
FIGURE 18.21 The figure illustrates the steps involved in evaluating the circular convolution of
the sequences x(n) and h(n) from Example 18.9. The figure shows the plots of x(m), h(m), and the
resulting circular convolution y(m). The figure also shows circularly shifted and reversed versions
of h(m).
1. We first circularly reverse h(m) to obtain h(m mod 3). This leads to
n
h(m mod 3) =
1 , 2, 1
515
SECTION 18.2
USEFUL
PROPERTIES
516
and the resulting sequence is shown in the second row of Fig. 18.21.
CHAPTER 18
PROPERTIES
OF THE
DFT
2. We then multiply the samples of x(m) and h(m mod 3) and add the products to obtain
y(0) = 1 + 4 + 0.5 = 3.5
3. We now shift h(m mod 3) circularly to the right by one sample to obtain
n
h[(1 m) mod 3] =
1 , 1, 2
This sequence is shown in the last row of Fig. 18.21. Multiplying the samples of x(m) and
h[(1 m) mod 3] and adding the products we obtain
y(1) = 1 2 + 1 = 0
4. We further shift h[(1 m) mod 3] circularly to the right by one step to obtain
n
h[(2 m) mod 3] =
2 , 1, 1
This sequence is shown in the last row of Fig. 18.21. Multiplying the samples of x(m) and
h[(2 m) mod 3] and adding the products we obtain
y(2) = 2 + 2 0.5 = 3.5
The top row of Fig. 18.21 plots the resulting sequence y(m) m.
Circular Convolution Property

Consider the tenth line in Table 18.1. It considers two causal and finite-duration sequences,
x(n) and y(n), both with duration L N . The lengths of the sequences are extended to
N samples each by padding a sufficient number of zeros. The circular convolution of x(n)
and y(n) will again be a causal sequence with N samples. The tenth line of Table 18.1
relates the N point DFT of the circular convolution to the individual N point DFTs of
the sequences, namely,
x(n) y(n) X(k)Y (k)
(18.19)
In other words, the N point DFT of the circular convolution sequence is obtained by multiplying, sample-by-sample, the individual N point DFTs.
Proof: For any integers n and m, let r denote the result of the following modulo operation
r = (n m) mod N
Obviously, r varies within the range 0 r N 1. Moreover, we can write
n m = aN + r
for some integer a and residual r (that are dependent on n and m). Let also
w(n) = x(n) y(n) =
N1
X
m=0
x(m)y[(n m) mod N ]
517
Then
W (k)
N1
X
SECTION 18.2
w(n)ej
2k n
N
n=0
N1
X N1
X
n=0 m=0
N1
X N1
X
m=0 n=0
N1
X
x(m)y[(n m) mod N ]ej
2k n
N
x(m)y[(n m) mod N ]ej
2k n
N
N1
X
x(m)
m=0
N1
X
n=0
x(m)e
y[(n m) mod N ]e
N1
X
m
j 2k
N
m=0
N1
X
n=0
x(m)e
N1
X
j 2k
m
N
m=0
N1
X
x(m)e
N1
X
m
j 2k
N
N1
X
y[(n m) mod N ]e
x(m)ej
2k m
N
y(r)e
y(r)e
r
j 2k
N
N1
X
(nm)
j 2k
N
j 2k
(aN+r)
N
r=0
m=0
!
n
j 2k
N
r=0
m=0
USEFUL
PROPERTIES
0k N 1
y(r)ej
2k r
N
r=0
X(k)Y (k)
Example 18.10 (Illustrating the circular convolution property)

Let reconsider the two sequences of Example 18.9, namely,
n
x(n) =
1 , 2, 0.5
and
h(n) =
1 , 1, 2
with N = 3. The circular convolution of these two sequences was computed by means of the
graphical method in that example. The result was
n
y(n) =
3.5 , 0, 3.5
That is,
y(0)
7/2
y(1)
y(2)
7/2
Let us first evaluate the 3point DFT of y(n) directly from the definition and then we compare the
result with the one obtained from application of the convolution property (18.19).
Using the definition (18.1), with N = 3, we have
Y (k) =
2
X
n=0
y(n)ej
2k n
3
k = 0, 1, 2
518
Therefore,
CHAPTER 18
PROPERTIES
OF THE
DFT
Y (0)
2
X
y(n) = 7
n=0
Y (1)
=
=
=
=
=
Y (2)
=
=
=
=
y(0) + y(1)ej
4
3
8
3
+ y(2)ej

1
7
3
7
j
+
2
2
2
2

7
1j 3
4
7 j/3
e
2

Y (k) =

Y (k) =
4
3
Therefore, we find that
i.e.,
y(0) + y(1)ej 3 + y(2)ej

4
7
7
+ ej 3
2
2

7
7
1
3
+
+j
2
2
2
2

7
1+j 3
4
7 j/3
e
2
7,
7,
7 j/3 7 j/3
e
, e
2
2

7

7
1+j 3 ,
1j 3
4
4
Let us now employ the convolution property (18.19) to arrive at the same conclusion. To do so, we
first need to determine the 3point DFTs of x(n) and h(n). This can be done by resorting again to
the definition (18.1). Indeed, for X(k) we have
X(k) =
2
X
n=0
x(n)ej
2k n
3
k = 0, 1, 2
519
Therefore,
X(0)
2
X
SECTION 18.2
USEFUL
PROPERTIES
x(n) = 7/2
n=0
X(1)
=
=
=
=
=
X(2)
=
=
=
=
=
That is,
x(0) + x(1)ej
+ x(2)ej
4
3
1 j 4
e 3
2

1
1
1
3
3
1 + 2 j
+
+j
2
2
2
2
2

1
3
1 + 1 j 3 + + j
4
4

1
1 + j3 3
4
1 + 2ej
2
3
x(0) + x(1)ej
4
3
+ x(2)ej
8
3
1 j 8
e 3
2

1
1
1
3
3
1 + 2 +j
+
j
2
2
2
2
2

1
3
1 + 1 + j 3 + j
4
4

1
1 j3 3
4
1 + 2ej
7/2 ,
X(k) =
2
3
4
3

1
1
1 + j3 3 ,
1 j3 3
4
4
Likewise, for H(k) we get

H(k) =
2
X
h(n)ej
2k n
3
k = 0, 1, 2
n=0
Therefore,
H(0)
2
X
h(n) = 2
n=0
H(1)
h(0) + h(1)ej
1 + ej
=
=
H(2)
1 + ej
That is,
4
3
4
3
+ h(2)ej
8
3
+ 2ej 3

1
1
3
3
+ 2 j
1 + + j
2
2
2
2

1
5+j 3
2

H(k) =
4
3
2
3
h(0) + h(1)ej
+ h(2)ej
+ 2ej 3

3
3
1
1
1 + j
+ 2 +j
2
2
2
2

1
5j 3
2

2
3
2,

1
1
5j 3 ,
5+j 3
2
2
520
Multiplying the samples of X(k) and H(k) sample-by-sample leads to the following values

CHAPTER 18
PROPERTIES
OF THE
DFT
X(k)H(k) =
7,

7

7
1+j 3 ,
1j 3
4
4
which agree with the samples of the 3point DFT Y (k) that were computed earlier directly from
the definition of the DFT.

Consider the eleventh line in Table 18.1. It again considers two causal and finite-duration
sequences, x(n) and y(n), both with duration L N . The lengths of the sequences are
extended to N samples each by padding a sufficient number of zeros. The result in the
table states that the N point DFT of the element-wise product of two sequences is given
by the expression below:
x(n)y(n)
1
X(k) Y (k)
N
(18.20)
in terms of the circular convolution of the sequences X(k) and Y (k).

Proof: For any integers k and m, let us express
(k m) = aN + r
for some integer a and where r denotes the result of the modulo operation:
r = (k m) mod N
We already know that r varies over the interval 0 r N 1 as k and m vary over the same
interval. Now let W (k) denote the scaled circular convolution:
W (k) =
1
X(k) Y (k),
N
k = 0, 1, . . . , N 1
Writing down the definition (18.18) of circular convolution in this case we have
W (k)
N1
1 X
X(m)Y [(k m) mod N ]
N m=0
Using the inverse DFT relation (17.35) we can recover the sequence w(n) as follows:
521
w(n)
SECTION 18.2
N1
2k
1 X
W (k)ej N n ,
N
k=0
N1
X
N1
2k
1 X
X(m)Y [(k m) mod N ] ej N n
N m=0
1
N
N1
1 X
X(m)
N 2 m=0
N1
X
N1
1 X
X(m)
N 2 m=0
N1
X
N1
1 X
X(m)
N 2 m=0
N1
X
=
=
k=0
USEFUL
PROPERTIES
n = 0, 1, . . . , N 1
Y [(k m) mod N ]e
k=0
n
j 2k
N
Y (r)e
2(m+aN +r)
n
N
Y (r)e
2(m+r)
n
N
r=0
r=0
N1
2m
1 X
X(m)ej N n
N m=0
using k m = aN + r
N1
2r
1 X
Y (r)ej N n
N r=0
x(n)y(n)
as desired.
Example 18.11 (Illustrating multiplication in time)

Consider again the two sequences from Examples 18.9 and 18.10, namely,
n
x(n) =
1 , 2, 0.5
and
h(n) =
1 , 1, 2
with N = 3. The corresponding 3point DFTs are given by

X(k) =
7/2 ,

and
H(k) =

1
1
1 + j3 3 ,
1 j3 3
4
4

1
1
2,
5j 3 ,
5+j 3
2
2
Let y(n) denote the element-wise product of both sequences so that

n
y(n) =
1 , 2, 1
Using definition (18.1), with N = 3, we can evaluate the 3point DFT of y(n) directly as follows:
Y (k) =
2
X
n=0
y(n)ej
2k n
3
k = 0, 1, 2
522
so that
CHAPTER 18
PROPERTIES
OF THE
DFT
Y (0)
2
X
y(n) = 2
n=0
Y (1)
Y (2)
y(0) + y(1)ej
1 + 2ej
y(0) + y(1)ej
1 + 2ej
4
3
2
3
4
3
+ y(2)ej
8
3
4
3
+ ej 3

1
1
3
3
+ 2 +j
+ j
2
2
2
2

1
3
+ 1 + j 3 + j
2
2

5j 3

1
2
+ y(2)ej
+ ej 3

3
3
1
1
+ 2 j
+ +j
2
2
2
2

3
1
+ 1 j 3 + + j
2
2

5+j 3

1
2
2
3
Therefore, we find that

Y (k) =
2,

1
1
5+j 3 ,
5j 3
2
2
Let us now evaluate the circular convolution of X(k) and H(k) in order to arrive at the same result
for Y (k), as suggested by property (18.20). To do so, we employ the graphical method of evaluation.
Let W (k) denote the circular convolution of the sequences X(k) and H(k):
W (k) = X(k) H(k)
1. We first circularly reverse H(m) to obtain H(m mod 3). This leads to

H(m mod 3) =
2,

1
1
5+j 3 ,
5j 3
2
2
2. We then multiply the samples of X(m) and H(m mod 3) and add the products to obtain
W (0) = 6
3. We now shift H(m mod 3) circularly to the right by one sample to obtain

H[(1 m) mod 3] =

1
21 5 j 3 , 2,
5+j 3
2
Multiplying the samples of X(m) and H[(1 m) mod 3] and adding the products we obtain
W (1) =

3
5+j 3
2
4. We further shift H[(1 m) mod 3] circularly to the right by one step to obtain

H[(2 m) mod 3] =

1
12 5 + j 3 ,
5j 3 , 2
2
Multiplying the samples of X(m) and H[(2 m) mod 3] and adding the products we obtain
W (2) =
That is,
W (k) =
6,

3
5j 3
2
USEFUL
PROPERTIES

3
3
5+j 3 ,
5j 3
2
2
According to property (18.20), the desired sequence Y (k) is related to W (k) via
Y (k) =
1
W (k),
3
k = 0, 1, 2
If we divide the samples of W (k) by 3 we obtain the samples for Y (k), as expected.

Consider the twelfth line in Table 18.1. It again considers two causal and finite-duration
sequences, x(n) and y(n), both with duration L N . The lengths of the sequences are
extended to N samples each by padding a sufficient number of zeros. The result in the
table is known as Parsevals relation and it states that
NP
1
n=0
x(n)y (n)
1
1 NP
X(k)Y (k)
N k=0
(18.21)
We can regard the computation on the left-hand side as the inner product of the samples of
x(n):
{ x(0), x(1), x(2), . . . , x(N 1) }
(18.22)
with the conjugated samples of y(n):
{ y (0), y (1), y (2), . . . , y (N 1) }
(18.23)
At the same time, we can regard the computation on the right-hand side of (18.21) as the
scaled inner product of the samples of X(k):
{ X(0), X(1), X(2), . . . , X(N 1) }
(18.24)
with the conjugated samples of Y (k):

{ Y (0), Y (1), Y (2), . . . , Y (N 1) }
(18.25)
Therefore, Parsevals relation is essentially a statement that computation of inner products

can be performed either in the time-domain (by using the sequence samples) or in the
frequency domain (by using the DFT samples). Note in particular the special case that
arises when we select y(n) = x(n); in this case, Parsevals relation reduces to
NP
1
n=0
|x(n)|2
1
1 NP
|X(k)|2
N k=0
523
SECTION 18.2
(18.26)
On the left-hand side we have the energy of the sequence x(n). We therefore find that the
energy of a sequence can be evaluated in the frequency domain as well, by evaluating the
524
CHAPTER 18
PROPERTIES
OF THE
DFT
energy of its N point DFT and scaling the result by 1/N .

Proof: We already know from the complex conjugation property (18.16) that
y (n)
Y (k mod N )
Let w(n) denote the product sequence

w(n) = x(n)y (n)
We also know from property (18.20) regarding the multiplication of sequences in time that the
N point DFT, W (k), is given by
W (k)
1
X(k) Y (k mod N ),
N
N1
1 X
X(m)Y [(k m) mod N ]
N m=0
k = 0, 1, . . . , N 1
(18.27)
(18.28)
Now recall definition (18.1) of the N point DFT of a sequence, namely,

W (k) =
N1
X
w(n)ej
2k n
N
k = 0, 1, . . . , N 1
n=0
It follows that the value of W (k) at k = 0 is equal to the sum of the samples of the sequence w(n),
i.e.,
W (0) =
N1
X
w(n)
(18.29)
n=0
This is a general and useful result. Applying this fact to the current context we have that
W (0) =
N1
X
x(n)y (n)
(18.30)
n=0
in view of the definition w(n) = x(n)y (n). At the same time, from (18.28) we have
W (0)
N1
1 X
X(m)Y (m)
N m=0
(18.31)
Equating with (18.30)we arrive at the desired conclusion that

N1
X
n=0
x(n)y (n)
N1
1 X
X(m)Y (m)
N m=0
Example 18.12 (Illustrating Parsevals relation)

Consider again the two sequences from Examples 18.9 and 18.10, namely,
n
x(n) =
1 , 2, 0.5
and
h(n) =
1 , 1, 2
with N = 3. The corresponding 3point DFTs are given by

X(k) =
7/2 ,

1
1
1 + j3 3 ,
1 j3 3
4
4
and
525

1
1
2,
5j 3 ,
5+j 3
2
2
H(k) =
SECTION 18.4
APPLICATIONS
Now note that

2
X
n=0
x(n)h (n) = (1 1) + (2 1) + (0.5 2) = 2
At the same time
7
2
2
2
X
X(k)H (k) =
k=0

i

i
1 h
1 h
+
1 + j3 3 5 j 3
+
1 j3 3 5 + j 3
= 6
8
8
so that
2
1X
X(k)H (k) = 2
3
k=0
as desired.
18.3 APPLICATIONS
TO BE ADDED
Practice Questions:
1.
2.
18.4 PROBLEMS
Problem 18.1 Use the linearity property of the DFT to determine the 4point DFT of the following sequences whose nonzero samples occur only over the range 0 n 3; all other samples are
zero:
(a) x(n) = cos
n
2
+ sin
(b) x(n) = cos (n) + sin
n .
2
n .
2
(c) x(n) = cos (n) + sin (n).

(d) x(n) = cos2 (n).
Problem 18.2 Use the linearity property of the DFT to determine the 4point DFT of the following sequences whose samples are nonzero only over the range 0 n 3:
(a) x(n) = cos
(b) x(n) = cos
n
3
n
4
+ sin
sin
(c) x(n) = cos (n) sin
(d) x(n) = sin2 (n).
n .
4
n .
6
n .
3
526
CHAPTER 18
PROPERTIES
OF THE
DFT
Problem 18.3 Plot |X(k)| and X(k) for each of the DFTs in Prob. 18.1.
Problem 18.4 Plot |X(k)| and X(k) for each of the DFTs in Prob. 18.2.
Problem 18.5 Find the 4point DFTs of the following sequences whose samples are nonzero only
over the range 0 n 3:
(n 1) .
4

cos 4 (n 1) mod 4 .

cos2 4 (n + 1) mod 4 .

sin 4 (n 1) mod 4 .
(a) x(n) = cos

(b) x(n) =
(c) x(n) =
(d) x(n) =
Problem 18.6 Find the 4point DFTs of the following sequences whose samples are nonzero only
over the range 0 n 3:
(a) x(n) = cos
(b) x(n) = cos

(c) x(n) = sin
(n
4

3

4
(n
4
1) sin
(n 1) mod 4 .
(n + 1) cos2

4
1) .
(n + 1) mod 4 .
(d) x(n) = sin2 4 (n 1) mod 4 .

Problem 18.7 Find the 4point DFT of the following sequence whose samples over 0 n 3
are described below (all other samples are zero):
x(n) = cos

+ sin
(n 3) mod 4
sin2

Plot |X(k)| and X(k).

Problem 18.8 Find the 4point DFT of the following sequence whose samples over 0 n 3
are described below (all other samples are zero):
x(n) = cos2
Plot |X(k)| and X(k).

sin
(n + 1) mod 4
1 , 2, 1, 1/2 .
(a) Find the 4point DFT of x(n).

(b) Find the 4point DFT of x((n + 2) mod 4).
(c) Find the 4point DFT of x((n 1) mod 4).
(d) Find the 4point DFT of x(n 1).
(e) Find the 4point DFT of x(n + 2).

n
1 , 0, 1/2, 2 .
(a) Find the 4point DFT of x(n).

(b) Find the 4point DFT of x((n + 1) mod 4).
(c) Find the 4point DFT of x((n 2) mod 4).
(d) Find the 4point DFT of x(n 2).
(e) Find the 4point DFT of x(n + 1).

n
1 , 2, 1, 1/2 .
(a) Find the 4point DFT of (1)n x(n).
(b) Find the 4point DFT of ej
3 n
2
x(n).
sin2

(c) Find the 4point DFT of ej 3 n cos
n
3
527
x(n).
SECTION 18.4
PROBLEMS
1 , 0, 1/2, 2 .
(a) Find the 4point DFT of (1)n x(n).
(b) Find the 4point DFT of ej 4 n x(n).
(c) Find the 4point DFT of ej 2 n sin
n
3
x(n).
Problem 18.13 Find the 4point DFT of the following sequences:

(a) x(n) = (1)n cos
(b) x(n) = (1)n
(c) x(n) = (1)n
n .
3

cos 3 (n 2) mod 4 .

cos2 3 (n + 1) mod 4 .
Problem 18.14 Find the 4point DFT of the following sequences:

(a) x(n) = (1)n cos 3 n mod 4 .
(b) x(n) = ej 2 n cos2
(n
3
2) mod 4 .
(c) x(n) = (1)n ej 2 n cos2
(n
3
+ 1) mod 4 .
Problem 18.15 The magnitude and phase components of a sequence x(n) are given by
n
|x(n)| =
1 , 1, 2, 3/2 ,
x(n) =
/2 , , 0, /4
Determine and plot the following DFTs:

(a) 4point DFT of x(n).
(b) 4point DFT of x (n).
(c) 4point DFT of (1)n x(n).
(d) 4point DFT of cos

(e) 4point DFT of cos
(f) 4point DFT of cos
(g) 4point DFT of cos
(h) 4point DFT of cos
n
2
n
2
n
2
n
2
n
2
x (n).
x(n).
x(n mod 4).
x (n mod 4).
x ((n 1) mod 4).
Problem 18.16 The magnitude and phase components of a sequence x(n) are given by
n
|x(n)| =
1/2 , 0, 1, 2 ,
x(n) =
/3 , /6, /6, /2
Determine and plot the following DFTs:

(a) 4point DFT of x(n).
(b) 4point DFT of x (n).
(c) 4point DFT of (1)n x(n).
(d) 4point DFT of cos

(e) 4point DFT of cos
(f) 4point DFT of cos
(g) 4point DFT of cos
(h) 4point DFT of cos
n
2
n
2
n
2
n
2
n
2
x (n).
x(n).
x(n mod 4).
x (n mod 4).
x ((n 1) mod 4).
Problem 18.17 Compute the circular convolution

cos
2n
N
cos
4n
N
528
CHAPTER 18
PROPERTIES
OF THE
DFT
where both sequences are limited to the interval 0 n N 1.

Problem 18.18 Compute the circular convolution

sin2
2n
N
(1)n cos
4n
N
where both sequences are limited to the interval 0 n N 1.

Problem 18.19 Evaluate the expression
S=
N1
X
ej
2n
N
cos
n=0
2ko n
N
cos
2k1
N
where ko and k1 are distinct integers in the interval 0 k N 1.

Problem 18.20 Evaluate the expression
S=
N1
X
ej
2n
N
n=0
sin
2ko n
N
cos2
2k1
N
where ko and k1 are distinct integers in the interval 0 k N 1.

Problem 18.21 What is the N point DFT of
y(n) = x(n) + x((n N ) mod N )
in terms of the N point DFT of x(n)?
Problem 18.22 What is the N point DFT of

y(n) = cos
2ko
n x(n) + (1)n x((n N ) mod N )
N
in terms of the N point DFT of x(n) and where ko lies in the interval 0 ko N 1?
Problem 18.23 Consider a sequence x(n) of length N . We embed x(n) into two new sequences
of lengths 2N each in the following manner:
x1 (n)
{ x(0), x(1), . . . , x(N 1), 0, 0, . . . , 0 }

|
{z
x2 (n)
zeros
{ 0, 0, . . . , 0, x(0), x(1), . . . , x(N 1) }

|
{z
zeros
That is, the first N samples of x1 (n) coincide with those of x(n) while the last N samples are zeros.
Likewise, the first N samples of x2 (n) are zero while the last N samples coincide with those of
x(n). Both x1 (n) and x2 (n) are defined over the interval 0 n 2N 1. Let X1 (k) and X2 (k)
denote the 2N -point DFTs of the sequences x1 (n) and x2 (n), respectively.
(a) Show that X1 (k) and X2 (k) are related as follows:
(
X2 (k) =
X1 (k),
X1 (k),
k even
k odd
for k = 0, 1, . . . , 2N 1.
(b) If X(k) denotes the N -point DFT of x(n), how do you recover the N values of X(k) from
the 2N values of X1 (k)?
Problem 18.24 Consider a sequence x(n) of length N , where N is even. We embed x(n) into
two new sequences of lengths 2N each in the following manner:
x1 (n)
{0, 0, . . . , 0, x(0), x(1), . . . , x(N 1), 0, 0, . . . , 0 }

|
{z
N/2
x2 (n)
{z
N/2
{ 0, 0, . . . , 0, x(0), x(1), x(2), x(3), . . . , x(N 1) }

|
{z
zeros
That is, the leading and trailing N/2 samples of x1 (n) are zero, while the first N samples of x2 (n)
are zero. Additionally, the samples of x(n) in the sequence x2 (n) are multiplied by 1 in an
alternating manner. The sequences x1 (n) and x2 (n) are defined over the interval 0 n 2N 1.
Let X1 (k) and X2 (k) denote the 2N -point DFTs of x1 (n) and x2 (n), respectively.
(a) Relate the samples of the sequences X1 (k) and X2 (k).
(b) Let X(k) denote the N -point DFT of x(n). Show that X(k) can be obtained directly from
the samples of X1 (k) without using any further DFT or inverse DFT operations. Repeat in
terms of the samples of X2 (k).
Problem 18.25 Consider a causal sequence x(n) of length N and let X(k) denote its N -point
DFT. Define the extended sequences
x1 (n)
x2 (n)
{x(0), x(1), . . . , x(N 1), x(0), x(1), . . . , x(N 2), x(N 1)}
{x(0), x(1), . . . , x(N 1), x(N 1), x(N 2), . . . , x(1), x(0)}
(a) Determine the 2N -point DFT of x1 (n) in terms of X(k).

(b) Determine the even-indexed terms of the 2N -point DFT of x2 (n) in terms of X(k).
Problem 18.26 Consider a sequence x(n) of length N and let X(k) denote its N point DFT. Let
y(n) further denote the N point DFT of the sequence X(k). Show that y(n) = N x(n mod N ).
That is, show that y(n) is the circular reversal of x(n) (multiplied by N ).
Problem 18.27 Consider a sequence x(n) of length N , where N is even. What is the result of the
following succession of operations on x(n)?
(1)k
(1)n
DFT
DFT
DFT
x(n) ?
That is, x(n) is first transformed by an N point DFT, the result is modulated by the sequence
(1)k , transformed by a second N point DFT, modulated again by (1)n , and transformed one
more time. Answer the question for a generic sequence x(n). Then plot the resulting sequences at
all intermediate steps when N = 4 and x(n) = cos(n) for n = 0, 1, 2, 3.
Problem 18.28 Let X(k) denote the N point DFT of a causal finite-duration sequence x(n),
0 n N 1. Find the N point DFT (not inverse DFT) of the sequence X(k), 0 k N 1.
How does the result compare to x(n)?
Problem 18.29 Let X(k) denote the N point DFT of a causal finite-duration sequence x(n),
0 n N 1. Find the inverse DFT of Re(X(k)) in terms of the sequence x(n) and its complex
conjugate.
Problem 18.30 Consider the sequence x(n) whose DTFT is shown in Fig. 18.22. Define the
sequences
r(n) =
x(n + 16m),
m=
z(n) =
x(n + 8m)
m=
(a) Determine the 16-point DFT of the circular convolution r(n) r(n).
(b) Determine the 8-point DFT of z 2 (n).
(c) Determine the values of x(1), r(1), and z(1).
529
SECTION 18.4
PROBLEMS
530
CHAPTER 18
PROPERTIES
OF THE
DFT
X(ej )
4
2 4
(rad/sample)
FIGURE 18.22 Sequence x(n) for Prob. 18.30.
CHAPTER
19
Computing Linear Convolutions
he circular convolution property (18.19) shows that the DFT can be used to evaluate the
circular convolution of two causal sequences. Specifically, if we multiply the respective
N point DFTs and inverse transform the result, then we arrive at the circular convolution
sequence:
x(n) y(n) = IDFT { X(k)Y (k) }
(19.1)
The purpose of the discussion in this chapter is to show how how the DFT can also be
used to evaluate the linear convolution of two sequences. The main motivation for doing
so is that linear convolutions are fundamental in the operation of LTI systems and, more
importantly, the DFT will provide an efficient way for their evaluation (as we show later in
Chapter 20).
19.1 RELATING LINEAR AND CIRCULAR CONVOLUTIONS

To motivate our discussion, we start by examining the exact relationship between linear and
circular convolutions for two arbitrary sequences. Thus, consider two causal sequences,
x(n) and h(n), of same length N , i.e., their nonzero samples exist over the interval 0
n N 1. Let yc (n) denote their circular convolution, which is again an N point
sequence:
yc (n) =
=
x(n) h(n) (circular convolution)

n
X
x(m)h[(n m) mod N ], 0 n N 1
(19.2)
m=0
Likewise, let y(n) denote the linear convolution of the same two sequences. Then, y(n)
has length 2N 1 and its samples are given by (recall (5.11)):
y(n) =
=
x(n) h(n) (linear convolution)

n
X
x(m)h(n m), 0 n 2N 1,
(19.3)
m=0
We are interested in examining the relation between the sequences yc (n) and y(n).
Let yp (n) denote the sequence that results from the periodic repetition of y(n) as follows:
X
y(n N )
(19.4)
yp (n) =
=
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CHAPTER 19
COMPUTING
LINEAR
CONVOLUTIONS
That is, the sequence y(n) is repeated at multiples of N and all shifted sequences are
combined to yield yp (n); this is the same construction we encountered earlier in (17.19)
while embedding a sequence x(n) into its periodic extension xp (n). Then we claim that
the circular convolution sequence, yc (n), agrees with the samples of yp (n) over the interval
0 n N 1:
yc (n) = yp (n),
0nN 1
(19.5)
Proof: Let X(k), H(k), and Yc (k) denote the N point DFT coefficients of the sequences x(n),
h(n), and yc (n), respectively. Then we know from the circular convolution property (18.19) that
Yc (k) = X(k)H(k),
0k N 1
Let further X(ej ), H(ej ), and Y (ej ) denote the DTFTs of the sequences x(n), h(n), and y(n),
respectively. Then we also know from the linear convolution property (14.10) that
Now recall that the DFT coefficients X(k) and H(k) are obtained by uniformly sampling the corresponding DTFTs of x(n) and h(n):

X(k) = X(ej )
= 2k
N
H(k) = H(ej )
= 2k
N
k = 0, 1, , . . . , N 1
The element-wise product of the DFTs X(k) and Y (k) then provides N samples of the DTFT of
y(n). That is,
Yc (k) provides N samples of Y (ej ) at k =
2k
,
N
k = 0, 1, . . . , N 1
However, we already know from the discussion in Sec. 17.2 that the N point DFT coefficients
Yc (k) are the DFT coefficients of the periodic sequence
yp (n) =
X
=
y(n N )
Accordingly,
yc (n) = yp (n),
0 n N 1
19.2 COMPUTING LINEAR CONVOLUTIONS VIA THE DFT

We now explain how relation (19.5) can be exploited to evaluate the linear convolution of
two sequences by means of the DFT. At the onset, note that since the linear convolution
sequence y(n) has length 2N 1, it is expected that aliasing in time will occur when the
sequence is repeated at multiples of N to form the periodic sequence, yp (n). In this way,
the samples of yp (n) over the interval 0 n N 1, which are of interest in (19.5),
will generally suffer from aliasing and may not allow us to recover the desired samples
of y(n). Nevertheless, as we now argue, it is still possible to use the DFT to evaluate the
linear convolution of two sequences if we use sufficient zero-padding in order to avoid the
occurrence of aliasing in time when yp (n) is formed.
Specifically, consider two causal sequences x(n) and h(n) of lengths Nx and Nh , respectively. That is, the domain of x(n) is 0 n Nx 1 and the domain of h(n) is
0 n Nh 1; the lengths of the sequences can now be arbitrary and they do not need
to agree anymore. The linear convolution of x(n) and h(n) is the sequence y(n) defined
by
n
X
x(m)h(n m)
(19.6)
y(n) = x(n) h(n) =
m=0
The sequence y(n) will have length Nx + Nh 1. Let us choose an integer N that satisfies
N Nx + Nh 1
(19.7)
Motivated by the discussion so far, to evaluate y(n) by means of the DFT technique we
proceed as follows:
(a) We first extend each sequence by padding enough zeros and obtain two new sequences x (n) and h (n) of same length N each, say,
sequence x (n) = { x(0), x(1), . . . , x(Nx 1), 0, . . . , 0}
{z
}
|
samples between n = 0 and N 1
(19.8)
sequence h (n) = { h(0), h(1), . . . , h(Nh 1), 0, . . . , 0}

{z
}
|
samples between n = 0 and N 1
(19.9)
It is usually desirable to pad enough zeros to obtain a length N that is a power of 2;

we select N as the closest power of 2 that satisfies (19.7). For example, if Nx = 12
and Nh = 7, then N can be selected as N = 32 = 25 . This is desirable because
fast algorithms for evaluating the DFT, such as the Fast Fourier Transform (FFT) of
Chapter 20, exploit the power-of-two property very effectively. Padding additional
zeros to ensure a value of N that is a power-of-two does not affect the final result; it
will nevertheless enable more efficient computational schemes.
(b) We next evaluate the circular convolution of the extended sequences x (n) and h (n):
y (n) = x (n) h (n)
(19.10)
The resulting sequence y (n) will have length N ; its domain will be 0 n N 1.
(c) We now argue that the samples of y (n) that lie within the interval 0 n Nx +
Nh 1 will coincide with the samples of the desired linear convolution sequence
y(n) from (19.6):
y (n) = y(n),
n = 0, 1, 2, . . . , Nx + Nh 1
(19.11)
Proof: By padding x(n) and h(n) with zeros in order to attain length N , we do not change the
DTFTs of these sequences. That is, the DTFTs of x(n) and x (n) are identical since
X (ej ) =
N
X
x (n)ejn =
n=0
Nx
X
x(n)ejn = X(ej )
n=0
Likewise, the DTFTs of h(n) and h (n) are identical since

H (ej ) =
N
X
n=0
h (n)ejn =
Nh
X
n=0
h(n)ejn = H(ej )
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SECTION 19.2
Computing Linear
Convolutions via the DFT
534
CHAPTER 19
Now, from the linear convolution property (14.10) of the DTFT, we know that the DTFT of y(n) is
related to the DTFTs of x(n) and h(n) via the relation
COMPUTING
LINEAR
CONVOLUTIONS

We therefore conclude initially that it also holds that
Y (ej ) = X (ej )H (ej )
in terms of the DTFTs of the extended sequences.
To proceed, let X (k) and H (k) denote the N point DFTs of x (n) and h (n), respectively.
Then recall that these DFT coefficients are obtained by uniformly sampling the DTFTs of x (n)
and h (n), respectively, at the N points k = 2k/N for k = 0, 1, . . . , N 1. Let Y (k) =
X (k)H (k). Then
Y (k) provides N samples of Y (ej ) at k =
2k
,
N
k = 0, 1, . . . , N 1
The N DFT coefficients Y (k) are sufficient to uniquely recover the linear convolution sequence
y(n). This is because y(n) is a causal sequence of length L = Nx + Nh 1 and L N . When
we evaluate its N point DFT, no aliasing in time occurs over the interval 0 n N 1 (recall
the discussion following (17.20)). We conclude that Y (k) should coincide with the N point DFT
Y (k) of the linear convolution sequence y(n):
Y (k) = Y (k),
k = 0, 1, . . . , N 1
By inverse transforming Y (k) and keeping its samples over 0 n Nx + Nh 1, we can recover
the desired linear convolution samples y(n). Using (17.35) we can write
y(n) =
N1
2k
1 X
Y (k)ej N n ,
N k=0
n = 0, 1, . . . , Nx + Nh 1
where we only recover the samples up to time Nx + Nh 1 and where we use the N point DFT
coefficients Y (k) that result from the circular convolution of x (n) and y (n).
Example 19.1 (Illustrating computation of linear convolutions)

Consider the two sequences from Examples 18.9 and 18.10, namely,
n
x(n) =
1 , 2, 0.5
and
h(n) =
1 , 1, 2
Let y(n) denote the sequence that results from the linear convolution of x(n) and h(n). Evaluating
y(n) by means of the graphical method of Sec. 6.2 leads to the following result:
n
y(n) =
1 , 1, 7/2, 9/2, 1
with all other samples equal to zero. Let us now arrive at the same result by employing the DFT.
It is seen that the lengths of the sequences x(n) and h(n) are Nx = 3 and Nh = 3, respectively.
The closest power-of-two to Nx + Nh 1 = 5 is
N =8
535
We pad enough zeros to attain length N = 8 samples for each sequence, and define
n
x (n)
h (n)
SECTION 19.3
BLOCK
CONVOLUTION
METHODS
1 , 2, 0.5, 0, 0, 0, 0, 0
1 , 1, 2, 0, 0, 0, 0, 0
Let y (n) denote the circular convolution of x (n) and h (n). Evaluating y (n) by means of the
graphical method of Sec. 18.2 leads to the following result:
y (n) =
1 , 1, 7/2, 9/2, 1, 0, 0, 0
with all other samples equal to zero. It is seen that the first 5 samples of y (n) coincide with the
samples of y(n), as expected from (19.11).
Alternatively, we can evaluate the circular convolution sequence y (n) by means of the DFT and
its inverse transformation. In this case, we first use the definition (18.1) to evaluate the 8point
DFTs of X (k) and H (k) as follows:
X (k)
7
X
x (n)ej
2k n
8
x (n)ej
k n
4
k = 0, 1, . . . , 7
n=0
7
X
n=0
1 + 2ej
k
4
1 j k
e 2
2
and
H (k)
7
X
h (n)ej
2k n
8
h (n)ej
k n
4
k = 0, 1, . . . , 7
n=0
7
X
n=0
1 + ej
k
4
+ 2ej
k
2
Once the samples {X (k), H (k)} are determined for k = 0, 1, . . . , 7, we multiply them elementwise to obtain the DFT coefficients of the circular convolution y (n), i.e.,
Y (k) = X (k)H (k),
k = 0, 1, . . . , 7
We finally inverse transform Y (k) to obtain y (n) using the IDFT expression (17.35):
y (n) =
7
2k
1 X
Y (k)ej 8 n ,
8 k=0
n = 0, 1, . . . , 7
Clearly, in this particular example, evaluating the circular convolution y (n) in the time domain is
much more immediate.
19.3 BLOCK CONVOLUTION METHODS

The computation of the linear convolution of two sequences can be demanding if one of
the sequences is long since it would then involve the evaluation of long DFTs. There are
methods that segment the long sequence into smaller sequences and transform the problem into the computation of several smaller linear convolutions. These methods are called
536
block convolution methods and they are of two kinds: overlap-add and overlap-save.
CHAPTER 19
COMPUTING
LINEAR
CONVOLUTIONS
19.3.1 Overlap-Add Convolution Method

The overlap-add method is simpler to motivate and describe. Consider a causal sequence
x(n) whose length Nx is a multiple of some positive integer M :
{ x(n), 0 n Nx 1, Nx = pM }
(19.12)
If this is not the case, then the length of x(n) can be extended by padding a sufficient
number of trailing zeros. We partition the sequence x(n) into p segments of length M
each. Figure 19.1 illustrates this segmentation for the case p = 4. Four segments are
shown and they are denoted by xm (n), for m = 0, 1, 2, 3.
x(n)
FIGURE 19.1
x0 (n)
x1 (n)
x2 (n)
M
x3 (n)
A sequence x(n) is segmented into shorter sequences of length M each.
Each of the sequences xm (n) is causal, has M nonzero samples, and is defined as
follows:

x(n + mM ),
n = 0, 1, . . . , M 1
xm (n) =
(19.13)
0,
otherwise
In other words, each sequence xm (n) extracts a block of M samples from x(n) and shifts
it down to the origin of time, n = 0. For example, assuming p = 4 segments, M = 3
samples per block, and
n
o
x(n) =
1 , 1, 2, 3, 2, 4, 10, 0, 6, 4, 3, 5
we get four sub-sequences of duration M = 3 each:
o
n
1 , 1, 2
x0 (n) =
n
o
x1 (n) =
3 , 2, 4
o
n
10 , 0, 6
x2 (n) =
n
o
x3 (n) =
4 , 3, 5
where we are using the box notation to indicate the location of the sample at time n = 0.
All other samples in the sequences are zero. We can reconstruct the original sequence x(n)
from its segments, xm (n), as follows:
x(n) =
p1
P
m=0
xm (n mM )
(19.14)
That is, the sub-sequences are shifted to the right by multiples of M samples and added
together. For the example under consideration, we have
x(n) = x0 (n) + x1 (n M ) + x2 (n 2M ) + x3 (n 3M )
Now consider another causal sequence h(n) of length Nh . We are interested in evaluating
the linear convolution of x(n) with h(n), namely, the sequence y(n) that is given by
y(n) = x(n) h(n) =
n
X
k=0
x(k)h(n k)
Using the representation (19.14), and the distributivity property (6.5) of linear convolutions, we find that
y(n) = x(n) h(n)
=
p1
X
m=0
p1
X
m=0
xm (n mM )
h(n)
[xm (n mM ) h(n)]
This result is equivalent to saying that

y(n) =
p1
P
ym (n mM )
(19.15)
m=0
where the sub-sequences ym (n) are the result of convolving each of the xm (n) with h(n):
ym (n) = xm (n) h(n) =
n
P
k=0
xm (k)h(n k)
(19.16)
According to (19.15), the sub-sequences ym (n) are shifted by multiples of M and added
together to yield the linear convolution sequence, y(n) see Fig. 19.2. Since each subsequence ym (n) has length M + Nh 1 and these sub-sequences are shifted by multiples
of M , then overlaps occur between successive shifted sub-sequences before addition; the
occurrence of the overlap motivates the designation overlap-add method for this technique.
Summary of the Overlap-Add Method for Evaluating Linear Convolutions

(a) Partition the sequence x(n) into p non-overlapping segments of size M each, as
defined by (19.13).
(b) Compute the linear convolutions ym (n) = xm (n) h(n), as defined by (19.16).
These convolutions can be computed using one of your preferred methods, e.g.,
either directly in the time-domain by using (19.16) or indirectly in the frequency
domain by using the DFT technique of Sec. 19.2.
(c) Concatenate the convolution sequences ym (n) as indicated by (19.15).
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SECTION 19.3
BLOCK
CONVOLUTION
METHODS
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CHAPTER 19
COMPUTING
LINEAR
CONVOLUTIONS
x(n)
x0 (n)
x1 (n)
x2 (n)
y(n)
y0 (n)
M
x3 (n)
y1 (n)
y2 (n)
M + Nh 1
y3 (n)
FIGURE 19.2 The sub-sequences ym (n) are generated by convolving the corresponding subsequences xm (n) with h(n). The resulting ym (n) are then shifted by multiples of M and added
together to yield the linear convolution sequence, y(n). Each sub-sequence ym (n) has length M +
Nh 1. Therefore, overlaps occur between the successive shifted sub-sequences before addition,
which motivates the designation overlap-add method.
Example 19.2 (Illustrating the overlap-add method)

n
x(n) =
1 , 1, 2, 3, 2, 4, 10, 0, 6, 4, 3, 5
with Nx = 12 samples. Let M = 3 samples per block and p = 4 blocks. The resulting subsequences are
n
x0 (n)
x1 (n)
x2 (n)
x3 (n)
n
n
n
1 , 1, 2
3 , 2, 4
o
10 , 0, 6
4 , 3, 5
where we are using the box notation to indicate the location of the sample at time n = 0. Let us
employ the overlap-add method to evaluate the convolution of x(n) with the following sequence
h(n) = (n) + 0.5(n 1)
539
We first linearly convolve each of the sub-sequences xm (n) with h(n) to get
n
y0 (n)
x0 (n) h(n) =
y1 (n)
x1 (n) h(n) =
y2 (n)
x2 (n) h(n) =
y3 (n)
x3 (n) h(n) =
n
n
n
SECTION 19.3
BLOCK
CONVOLUTION
METHODS
1 , 0.5, 1.5, 1
3 , 3.5, 3, 2
o
10 , 5, 6, 3
4 , 1, 6.5, 2.5
Then
y(n)
3
X
m=0
ym (n 3m)
1 , 0.5, 1.5, 2, 3.5, 3, 12, 5, 6, 1, 1, 6.5, 2.5
19.3.2 Overlap-Save Convolution Method
Let us now motivate an alternative block convolution method known as the overlap-save
method, which is more demanding to describe.
Consider again causal sequences x(n) and h(n) of lengths Nx and Nh , respectively,
with Nx > Nh . The linear convolution of x(n) and h(n) is the sequence y(n) given by
y(n) = x(n) h(n) =
n
X
k=0
x(k)h(n k)
(19.17)
and whose length is equal to Nx + Nh 1.

Useful Property of Circular Convolution
We first derive a useful property of circular convolutions, which will serve as the basis for
the derivation of the overlap-save method.
Assume, for the sake of argument, that we extend the length of h(n) to Nx through
sufficient zero padding, and that we perform the Nx point circular convolution of x(n)
with the extended h(n):
yc (n) = x(n) h(n),
0 n Nx 1
(19.18)
The sequence yc (n) has length Nx . We already know from the discussion in Sec. ?? that
the sequence yc (n) is related to the periodic embedding yp (n) of y(n) by repeating y(n)
every Nx samples (since Nx is the length of the circular convolution) and adding together
all shifted sequences:
X
y(n Nx )
(19.19)
yp (n) =
=
Specifically, it holds that
yc (n) = yp (n),
0 n Nx 1
(19.20)
Let us examine the samples of yp (n) over the interval 0 n Nx 1. Figure 19.3
illustrates the sequence y(n) and two of its shifted versions, y(nNx) and y(n+Nx); these
are the only shifted sequences that interfere with the samples of y(n) over 0 n Nx 1
while forming yp (n).
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CHAPTER 19
COMPUTING
LINEAR
CONVOLUTIONS
Nx + Nh
y(n)
y(n + Nx )
y(n Nx )
Nh
Nh
Nx
FIGURE 19.3 When the sequence y(n) is shifted to the left by Nx samples and to the right by
Nx samples, overlaps occur at the leading and the trailing Nh samples.
Observe that aliasing in time occurs over the leading interval 0 n Nh 1, and over
the trailing interval Nx n Nx + Nh 1. However, the samples of y(n) that lie within
the interval
Nh n Nx 1
are not subjected to aliasing and remain intact while forming yp (n). We then conclude
from (19.20) that the samples of yc (n) that occur within Nh n Nx 1 coincide with
the samples of y(n) over the same interval:
yc (n) = y(n),
Nh n Nx 1
(19.21)
This discussion leads to the following observation, which will be exploited to motivate and
develop the overlap-save block convolution method.
Useful conclusion. The Nx point circular convolution of two sequences, x(n) and
h(n), allows us to recover a portion of their linear convolution without any distortion. The
portion that is recovered is the one that lies within the interval Nh n Nx 1, where
Nx and Nh are the lengths of the sequences x(n) and h(n), respectively, and it is assumed
that Nx > Nh .
Overlap-Save Computations
We now use the above conclusion to derive the overlap-save method for the evaluation of
linear convolutions. Thus, consider again a causal sequence x(n) whose length Nx is a
multiple of some positive integer M . If this is not the case, then the length of x(n) can be
extended by padding a sufficient number of leading zeros.
We partition the sequence x(n) into a sufficient number of overlapping segments of
length M + Nh each in order to cover the entire samples of x(n). Compared with the
overlap-add method, observe that the segments are now overlapping and have length M +
Nh each. Figure 19.4 illustrates the overlap-save segmentation for the case p = 4. Four
segments are shown and they are denoted by xm (n), for m = 0, 1, 2, 3.
541
SECTION 19.3
x(n)
BLOCK
CONVOLUTION
METHODS
x
0 (n)
x
1 (n)
zero
padding
x
2 (n)
x
3 (n)
M + Nh
FIGURE 19.4 The sequence x(n) is partitioned into overlapping sub-sequences x

m (n) of length
M + Nh each. The leading Nh samples of x
m (n) overlap with the trailing Nh samples of the
preceding sequence x
m1 (n). The first sequence, x
0 (n) is padded with Nh leading zeros.
The first sub-sequence, x0 (n), has a leading block of Nh zeros followed by the first M
samples of x(n), i.e.,
x
0 (n) =
0,
x(n Nh ),
0 n Nh 1
Nh n M + Nh 1
The second sub-sequence, x

1 (n), has the same Nh trailing samples of x0 (n) and the second block of M samples from x(n):
x1 (n) =
x
0 (n + M ),
x(n + M Nh ),
0 n Nh 1
Nh n M + Nh 1
The third sub-sequence, x

2 (n), has the same Nh trailing samples of x
1 (n) and the second
block of M samples from x(n):
x
2 (n) =
x1 (n + M ),
x(n + 2M Nh ),
0 n Nh 1
Nh n M + Nh 1
and so on. We continue this construction until any additional segment will consist solely
of zero samples (see Example 19.3 further ahead).
We assume the length of h(n) is extended to M + Nh via zero padding. We then
start by evaluating the circular convolution of x0 (n) with h(n). The result will be an
(M + Nh )point sequence y0 (n):
y0 (n) = x
0 (n) h(n)
(19.22)
The first Nh samples of this convolution are aliased and are discarded. On the other hand,
and according to the useful property (19.21) of circular convolutions, the samples within
the range Nh n Nh + M 1 are not subject to distortion and coincide with the
samples of the linear convolution of x
0 (n) and h(n) over the same range:
y0 (n) = x
0 (n) h(n),
over Nh n Nh + M 1
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CHAPTER 19
COMPUTING
LINEAR
CONVOLUTIONS
However, because of the padding of Nh leading zeros in x

0 (n), we find that the samples
of the linear convolution x
0 (n) h(n) over the interval Nh n Nh + M 1 coincide
with the samples of the desired linear convolution x(n) h(n) over 0 n M 1
see Fig. 19.5. In other words, we conclude that the samples of the computed circular
convolution, y0 (n), in (19.22) allow us to identify the first M samples of the desired linear
convolution sequence y(n):
y(n) = y0 (n + Nh ),
0nM 1
(19.23)
M + Nh
M + Nh
x
0 (n)
h(n)
zero
padding
zero
padding
Nh
y0 (n)
discard
coincide with samples of

y(n) over 0 n M 1
FIGURE 19.5 The sub-sequence x

0 (n) is circularly convolved with h(n) to generate sequence
y0 (n) with M +Nh samples. The first Nh samples of y0 (n) are discarded and the remaining samples
coincide with the samples of the linear convolution of x(n) and h(n) over 0 n M 1.
Before we proceed, let us examine this result more closely. Let x0 (n) denote the trailing M entries of x
0 (n); these are the M samples from x(n) that belong to x
0 (n). That
is, we are excluding the block of leading zeros from x
0 (n) and denoting the remaining
sequence by x0 (n) see Fig. 19.6. The above discussion indicates that the samples of
y0 (n) between Nh n M + Nh 1 coincide with the first M samples of the linear
convolution x0 (n) h(n):
x0 (n) h(n) = y0 (n + Nh ),
over 0 n M 1
However, the linear convolution of x0 (n) and h(n) is a sequence of length M +Nh 1. We
therefore still need to determine the trailing Nh 1 samples of this linear convolution. It is
for this reason that the subsequent sequence x
1 (n) is defined with an overlapping segment
of length Nh with the prior sequence x
0 (n). The samples in this overlapping segment are
the ones from x
o (n) that contribute to the evaluation of the missing Nh 1 samples.
So let us now evaluate the circular convolution of the second sequence, x
1 (n), with
h(n). The result will be an (M + Nh )point sequence y1 (n):
y1 (n) = x
1 (n) h(n)
543
SECTION 19.3
M + Nh
M + Nh
x
0 (n)
x0 (n)
h(n)
h(n)
Nh
M + Nh 1
?
First M samples
of y(n) already computed
Trailing Nh 1 samples
of y(n) are still missing
FIGURE 19.6 Linearly convolving the sequence x0 (n) of M samples with the sequence h(n) of
Nh samples results in a sequence of length M + Nh 1. The circular convolution of x
0 (n) and
h(n) already provides the first M samples of x0 (n) h(n). We still need to identify the last Nh
samples.
The first Nh samples of this convolution are aliased and are discarded. On the other hand,
the samples within the range Nh n Nh + M 1 are not subject to distortion and
coincide with the samples of the linear convolution of x(n) and h(n) over M n
2M 1:
y(n) = y1 (n + Nh M ), M n 2M 1
(19.24)
and so forth.
Summary of the Overlap-Save Method for Evaluating Linear Convolutions
(a) Partition the sequence x(n) into (p + 1) overlapping segments of size M + Nh each,
as illustrated in Fig. 19.4. The leading Nh samples of each sub-sequence xm (n)
overlap with the trailing Nh samples of the preceding sub-sequence x
m1 (n). The
first sub-sequence x
0 (n) is padded with Nh leading zeros.
(b) Compute the circular convolutions
ym (n) = xm (n) h(n)
which result in sub-sequences ym (n) of length M + Nh each.
BLOCK
CONVOLUTION
METHODS
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CHAPTER 19
COMPUTING
LINEAR
CONVOLUTIONS
(c) Discard the leading Nh samples of the sub-sequences ym (n) to generate the subsequences ym (n) of length M each:
ym (n) =
ym (n + Nh ),
0,
0nM 1
otherwise
(d) Concatenate the sub-sequences ym (n) to generate the desired linear convolution sequence y(n)as follows:
y(n) =
p1
X
m=0
ym (n mM )
Example 19.3 (Illustrating overlap-save method)

Let us consider the same sequence x(n) as Example 19.2:
n
x(n) =
1 , 1, 2, 3, 2, 4, 10, 0, 6, 4, 3, 5
with Nx = 12 samples. Let M = 3 samples per block. We want to employ the overlap-save method
to evaluate the convolution of x(n) with the following sequence
h(n) = (n) + 0.5(n 1)
We now need p = 5 blocks to cover the sequence with the corresponding sub-sequences generated
as follows:
n
x
0 (n)
x
1 (n)
x
2 (n)
x
3 (n)
x
4 (n)
n
n
n
n
0 , 0, 1, 1, 2
1 , 2, 3, 2, 4
o
2 , 4, 10, 0, 6
0 , 6, 4, 3, 5
o
3 , 5, 0, 0, 0
where we are using the box notation to indicate the location of the sample at time n = 0.
We first convolve circularly each of the sub-sequences, xm (n), with h(n) to get
n
y0 (n)
x
0 (n) h(n) =
y1 (n)
x
1 (n) h(n) =
y2 (n)
x
2 (n) h(n) =
y3 (n)
x
3 (n) h(n) =
y4 (n)
x
4 (n) h(n) =
n
n
n
n
1 , 0, 1, 0.5, 1.5
3 , 0.5, 2, 3.5, 3
o
1 , 3, 12, 5, 6
2.5 , 6, 1, 1, 6.5
o
3 , 6.5, 2.5, 0, 0
545
Discarding the leading Nh = 2 samples of the sequences ym (n) we get

n
y0 (n)
y1 (n)
y2 (n)
y3 (n)
y4 (n)
n
n
n
n
1 , 0.5, 1.5
SECTION 19.5
APPLICATIONS
2 , 3.5, 3
o
12 , 5, 6
1 , 1, 6.5
o
2.5 , 0, 0
Then,
y(n)
3
X
m=0
ym (n 3m)
1 , 0.5, 1.5, 2, 3.5, 3, 12, 5, 6, 1, 1, 6.5, 2.5
19.4 APPLICATIONS
TO BE ADDED
Practice Questions:
1.
2.
19.5 PROBLEMS
Problem 19.1 Let
x(n) =
-1 , 1, 0, 2 ,
h(n) =
1 , 2, 1/2, 0
(a) Find x(n) h(n) directly from the definition of the linear convolution.
(b) Find x(n) h(n) by using instead the circular convolution.
Problem 19.2 Let
x(n) =
-1 , 2, 1, 1, 2, 0, 1 ,
h(n) =
1 , 1, 0, 1
Problem 19.3 Given the sequences
x1 (n) = (n) +
1
1
(n 1) and x2 (n) = (n) + (n 2)
2
2
Compute the linear convolution of x1 (n) and x2 (n) in the following different ways:
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CHAPTER 19
COMPUTING
LINEAR
CONVOLUTIONS
(a) Using the graphical method.

(b) Using ztransforms.
(c) Using discrete-time Fourier transforms (DTFTs).
(d) Using circular convolution.
(e) Using discrete Fourier transforms (DFTs).
Problem 19.4 Given the sequences
x1 (n) = (n 1) (n 3) and x2 (n) = (n) (n 1) + (n 2)
Compute the linear convolution of x1 (n) and x2 (n) in the following different ways:
(a) Using the graphical method.
(b) Using ztransforms.
(c) Using discrete-time Fourier transforms (DTFTs).
(d) Using circular convolution.
(e) Using discrete Fourier transforms (DFTs).
Problem 19.5 Let
x(n) =
-1 , 1, 0, 2, 1/2, 3, 1 ,
h(n) =
1 , 0, 2
(c) Find x(n) h(n) using the overlap-add method.
(d) Find x(n) h(n) using the overlap-save method.
Problem 19.6 Let
x(n) =
2 , 1, 1, 1/2, 3 ,
h(n) =
1 , 0, 2, 1
(c) Find x(n) h(n) using the overlap-add method.
(d) Find x(n) h(n) using the overlap-save method.
Problem 19.7 Consider a causal sequence x(n) of length 5. The values of its samples at time
instants 0, 1, 3, and 4 are shown in Fig. 19.7, except for the sample at time instant n = 2, whose
value is unknown. The values of the other samples are either 0, 1, or 2. The coefficients of the
2-point DFT of x(n) satisfy
X(0) + X(1) = 2
(a) Can you recover the value of the unknown sample of x(n)? If so, what is its value? If not,
explain why?
(b) Determine X(k) X(k). That is, find the circular convolution of X(k) with itself.
(c) Find also X(k) X(k) by using circular convolution.
Problem 19.8 Consider two causal sequences x(n) and h(n) of duration N each. Let y(n) denote
their N point circular convolution. Show that the samples of the sequences {x(n), h(n), y(n)} can
be related in vector form as follows:
yN = H N x N
547
SECTION 19.5
x(n)
PROBLEMS
?
2
1
1
FIGURE 19.7 Sequence x(n) for Prob. 19.7.
where the N 1 vectors yN and xN are

2
yN
6
6
6
=6
6
6
4
y(0)
y(1)
y(2)
..
.
y(N 1)
7
7
7
7
7
7
5
6
6
6
=6
6
6
4
and
xN
x(0)
x(1)
x(2)
..
.
x(N 1)
3
7
7
7
7
7
7
5
and the N N matrix HN is called circulant and has the following form:
2
HN
6
6
6
=6
6
6
4
h(0)
h(1)
h(2)
..
.
h(N 1)
h(N 1)
h(0)
h(1)
..
.
h(N 2)
h(N 2)
h(N 1)
h(0)
..
.
h(N 3)
...
...
...
..
.
...
h(1)
h(2)
h(3)
3
7
7
7
7
7
7
5
h(0)
Observe that each row of HN is obtained by shifting the row above it circularly to the right by one
position.
Problem 19.9 Consider the circulant matrix HN of Prob. 19.8. Let FN denote the N N DFT
matrix defined by (17.46). Show that the product FN HN FN is a diagonal matrix. Remark: This
result shows that every circulant matrix is diagonalizable by the DFT matrix.
CHAPTER
20
Fast Fourier Transform
ne of the main advantages of working with the DFT in discrete-time signal processing
is that efficient methods exist for the evaluation of the DFT and its inverse. These methods
are known generally by the name Fast Fourier Transforms or FFTs for short. We saw in the
previous chapter that the DFT is useful in evaluating linear convolutions. Therefore, the
Fast Fourier Transform (FFT) will provide efficient ways to evaluate linear convolutions as
well.
20.1 COMPUTATIONAL COMPLEXITY

To motivate the FFT, we start by examining the computational cost involved in evaluating
an N point DFT. Thus, let x(n) denote a causal sequence of duration N and introduce its
N point DFT sequence, X(k):
X(k) =
NP
1
x(n)ej
2k
N n
n=0
k = 0, 1, . . . , N 1
(20.1)
The sequence x(n) can be recovered from its DFT through the inverse operation:
x(n) =
1
1 NP
2k
X(k)ej N n ,
N k=0
n = 0, 1, . . . , N 1
(20.2)
For convenience of notation, we let WN denote the N th root of unity, i.e.,

2
WN = ej N
(20.3)
Then evaluating the nk-th power of WN gives

nk
(WN )
= ej
2k
N n
so that definition (20.1) for the N point DFT of x(n) can be re-expressed in terms of WN
as
NP
1
X(k) =
(20.4)
x(n)WNnk , k = 0, 1, . . . , N 1
n=0
Likewise, the inverse DFT relation (20.2) can be re-expressed in terms of WN as

x(n) =
1
1 NP
X(k)WNnk ,
N n=0
n = 0, 1, . . . , N 1
(20.5)
549
Copyright
550
The complex number WN defined by (20.3) satisfies several useful relations such as:
CHAPTER 20
FAST
FOURIER
TRANSFORM
(WN ) = WN/q ,
q+ N
2
(WN )
= (WN ) ,
(WN )
qN/2
= (1)q
(20.6)
for any integer q.

Proof: Indeed, note that
(WN )q
q
= ej
ej N
q+ N
2
(WN )
j 2
(q+ N
)
N
2
(WN )qN/2
ej N q
N
2
= e
2q
N
= e
2q
j N
2
j N/q
= WN/q
ej = ej
2q
N
= WNq
= ejq = (1)q
We shall exploit the above properties while deriving efficient methods for evaluating the
DFT. Now refer to expression (20.4) and observe that the evaluation of each coefficient
X(k) generally involves computing N complex multiplications and (N 1) complex
additions (especially when the samples of x(n) are complex-valued themselves). Each
complex addition refers to the addition of two complex numbers, which involves two real
additions since
(a + jb) + (c + jd) = (a + c) j(b + d)
(20.7)
Likewise, each complex multiplication refers to the multiplication of two complex numbers
and involves four real multiplications and two real additions since
(a + jb) (c + jd) = (ac bd) + j(ad + bc)
(20.8)
This complexity translates into an overall cost of N (N 1) complex additions and N 2

complex multiplications to evaluate all N DFT coefficients, X(k). We therefore say that
the evaluation of the N point DFT of a sequence through the standard definition (20.4)
has complexity of the order of 2N 2 complex operations (additions and multiplications),
written as:
N -point DFT requires O(2N 2 ) complex operations
(20.9)
where the notation O(2N 2 ) signifies of the order of 2N 2 . A similar complexity figure
holds for the inverse DFT operation.
Example 20.1 (Cost of 1024-point DFT)

The evaluation of a 1024-point DFT requires
(1024)2 = 1, 048, 576
complex multiplications
(1024) (1023) = 1, 047, 552
complex additions
In other words, it is necessary that we perform of the order of 2 million complex operations. To have
an idea of what this cost entails, consider a discrete-time processor operating at the rate of 1GHz.
Assume further that each complex operation requires one clock cycle, namely, 109 sec. We then
find that the evaluation of the 1024 DFT coefficients would necessitate approximately 2.1 msec.
The computational cost of the DFT translates into a demand on the amount of time that
is necessary to evaluate its coefficients. We now describe efficient methods for evaluating the same coefficients by resorting to divide-and-conquer strategies that lead to Fast
Fourier Transform (FFT) techniques. There are several variants of the FFT algorithm. We
limit ourselves to the so-called radix-2 decimation-in-time and decimation-in-frequency
versions, which are widely used. Other FFT variants essentially share the same divideand-conquer strategies.
20.2 DECIMATION-IN-TIME FFT

Assume the length N is a power of 2, say N = 2p for some positive integer p. This
requirement is not restrictive since we can always pad the sequence x(n) with additional
trailing zeros in order to meet the condition. The padding of zeros does not alter the DTFT,
X(ej and, therefore, does not influence the values of the DFT coefficients, X(k).
Since N is even, we can split the sequence, x(n), into two smaller sequences of duration
N/2 each. In one sequence we group the even-indexed samples of x(n) and in the other
sequence we group the odd-indexed samples of x(n), say,
xe (n)
xo (n)
o
x(0) , x(2), x(4), . . . , x(N 4), x(N 2)
o
x(1) , x(3), x(5), . . . , x(N 3), x(N 1)
(20.10)
(20.11)
In (20.10)(20.11), we are using the box notation to indicate the location of the sample of
index n = 0 in both sequences xe (n) and xo (n). Figure 20.1 illustrates this construction
for a particular sequence x(n) of duration N = 8.
Let Xe (k) and Xo (k) denote the N2 point DFT of xe (n) and xo (n), respectively,
N
2
Xe (k)
1
X
kn
xe (n)WN/2
n=0
N
2
1
X
n=0
kn
, k = 0, 1, . . . , (N/2) 1
x(2n)WN/2
(20.12)
N
2
Xo (k)
1
X
kn
xo (n)WN/2
n=0
N
2
1
X
n=0
kn
x(2n + 1)WN/2
, k = 0, 1, . . . , (N/2) 1
(20.13)
Note that we have N/2 coefficients Xe (k) and N/2 coefficients Xo (k). Recall that although we are limiting the sequences to the interval 0 k N2 1, the DFT coefficients
Xe (k) and Xo (k) are actually periodic sequences with period N/2. Therefore, when necessary, the coefficients Xe (k) and Xo (k) over the extended interval 0 k N 1 are
551
SECTION 20.2
DECIMATIONIN-TIME FFT
552
CHAPTER 20
x(n)
FAST
FOURIER
TRANSFORM
2
1
4
xo (n)
xe (n)
x(0)
x(6)
x(3)
x(2)
x(5)
x(7)
2
3
x(4)
x(1)
FIGURE 20.1 The sequence x(n) of duration N = 8 (top) is decimated into two smaller
sequences, xe (n) and xo (n), of duration 4 samples each (bottom).
found from
{Xe (k), k = 0, 1, . . . , N 1} =
and

N
N
1 , Xe (0), Xe (1), . . . , Xe
1
Xe (0) , Xe (1), . . . , Xe
2
2
|
{z
} |
{z
}
one period with N2 samples

a second period with N2 samples
{Xo (k), k = 0, 1, . . . , N 1} =

N
N
Xo (0) , Xo (1), . . . , Xo
1 , Xo (0), Xo (1), . . . , Xo
1
2
2
|
{z
} |
{z
}
N
N
one period with 2 samples
a second period with 2 samples
where the box notation is used to indicate the location of the sample at bin k = 0.
Given the coefficients Xe (k) and Xo (k) over the extended interval 0 n N 1, we
now verify that the N point DFT of x(n) can be determined from Xe (k) and Xo (k) as
553
follows:
SECTION 20.2
X(k)
N
1
X
x(n)WNnk
(20.14)
n=0
x(n)WNkn +
n=even
n=odd
N
2
1
X
N
2
x(2m)WNk2m
m=0
1
X
N
2
km
x(2m)WN/2
WNk
m=0
k(2m+1)
x(2m + 1)WN
m=0
N
2
1
X
x(n)WNkn
{z
Xe (k)
1
X
km
x(2m + 1)WN/2
m=0
{z
Xo (k)
(20.15)
In other words, we find that

X(k) = Xe (k) + WNk Xo (k) , k = 0, 1, . . . , N 1
(20.16)
By further using the identify

k+ N
2
WN
= WNk
(20.17)
and the fact that Xe (k) and Xo (k) are periodic with period N/2, relation (20.16) for X(k)
can be rewritten in the equivalent form:
X(k)
X k+
N
2
Xe (k) + WNk Xo (k) , 0 k
N
2
Xe (k) WNk Xo (k) , 0 k
N
2
(20.18)
This result shows that the determination of the N point DFT coefficients X(k) can be
alternatively achieved as follows:
(a) We split the original sequence x(n) into two sub-sequences, xe (n) and xo (n), of
duration N/2 each. One sequence contains the even-indexed samples of x(n) and
the other sequence contains the odd-indexed samples of x(n).
(b) We determine the N2 point DFTs Xe (k) and Xo (k), k = 0, 1, . . . , N/2 1. The
evaluation of each of these DFTs requires

O 2 (N/2)2 = O N 2 /2 complex operations
(c) We multiply the N/2 coefficients Xo (k) by WNk , k = 0, 1, . . . , N/2 1. This step
requires N/2 complex multiplications.
(d) We add and subtract the N/2 samples of the sequences {Xe (k), WNk Xo (k)} to generate X(k). This step requires N complex additions.
The total computational cost adds up to O(N 2 ) complex operations, down from the earlier
figure of O(2N 2 ) operations in (20.9) when the N point DFT is computed directly from
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CHAPTER 20
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FOURIER
TRANSFORM
the defining relation (20.4). We therefore find that by decimating the sequence in time into
two smaller sequences, we are able to reduce the cost by a factor of 2.
The same decimation procedure can now be applied to the evaluation of the N2 point
DFTs Xe (k) and Xo (k) by splitting each of the sequences xe (n) and xo (n) into two
smaller sequences and computing their respective DFTs. For instance, we split xe (n)
into two smaller sequences of duration N/4 each: one of the sequences contains the evenindexed samples of xe (n) and the other sequence contains the odd-indexed samples of
xe (n), say,
n
o
x(0) , x(4), x(8), . . . , x(N 6), x(N 2)
o
n
x(2) , x(6), x(10), . . . , x(N 8), x(N 4)
=
xee (n)
xeo (n)
(20.19)
(20.20)
We then evaluate the N4 point DFTs of xee (n) and xeo (n), denoted by Xee (k) and Xeo (k),
respectively, and combine them to obtain the N2 -point DFT Xe (k):
Xe (k)
Xe k +
N
4
k
= Xee (k) + WN/2
Xeo (k) , 0 k
N
4
k
= Xee (k) WN/2
Xeo (k) , 0 k
N
4
(20.21)
Likewise, we split xo (n) into two smaller sequences of duration N/4 each: one of the
sequences contains the even-indexed samples of xo (n) and the other sequence contains the
odd-indexed samples of xo (n), say,
n
o
x(1) , x(5), x(9), . . . , x(N 5), x(N 1)
o
n
x(3) , x(7), x(11), . . . , x(N 7), x(N 3)
xoe (n) =
xoo (n) =
(20.22)
(20.23)
We subsequently evaluate the corresponding N4 -point DFTs, denoted by Xoe (k) and Xoo (k),
respectively, and combine them to obtain the N2 point DFT Xo (k):
Xo (k)
Xo k +
N
4
k
Xoe (k) + WN/2
Xoo (k) , 0 k
N
4
k
Xoe (k) WN/2
Xoo (k) , 0 k
N
4
(20.24)
Observe that now we need to evaluate 4 DFTs of order N/4 each.

The decimation process can be repeated again and applied to each of the sequences
{xee (n), xeo (n), xoe (n), xoo (n)}
In this way, the
N
4 point
DFTs
{Xee (k), Xeo (k), Xoe (k), Xoo (k)}
would be computed in terms of N8 point DFTs and so on. Starting with a duration N that
is a power of 2, say N = 2p , then the decimation process can be repeated p = log2 (N )
times until we collapse to a stage that requires the evaluation of 2point DFTs only. This
construction is best illustrated by means of an example (see below). Figure 20.2 helps
illustrate how the samples of an 8-point sequence , x(n), are decimated into the sequences
555
xe (n)
x(n)
SECTION 20.2
xee (n)
xeo (n)
xoe (n)
xoo (n)
xo (n)
FIGURE 20.2 A representation of the decimation process that takes the samples of an 8point
sequence x(n) and divides them into smaller sequences {xee (n), xeo (n), xoe (n), xoo (n)} of size
2 samples each. Within each sequence, the even-indexed samples and the odd-indexed samples are
denoted by same colored dots.
xe (n) and xo (n) and the subsequent sequences {xee (n), xeo (n), xoe (n), xoo (n)} of size 2
samples each.
Example 20.2 (8point DFT via decimation-in-time)

Consider an 8-point sequence
n
x(n) =
x(0) , x(1), x(2), x(3), x(4), x(5), x(6), x(7)
By splitting it into even and odd-indexed samples we obtain the two subsequences:
xe (n)
xo (n)
{x(0), x(2), x(4), x(6)}
{x(1), x(3), x(5), x(7)}
By further splitting each sub-sequence into even and odd-indexed samples we obtain the four subsequences:
xee (n)
xeo (n)
xoe (n)
xoo (n)
{x(0), x(4)}
{x(2), x(6)}
{x(1), x(5)}
{x(3), x(7)}
556
We are therefore reduced to computing 2point DFTs. Using the result of Example 17.8 we have
CHAPTER 20
FAST
FOURIER
TRANSFORM
Xee (0)
x(0) + x(4)
Xee (1)
x(0) x(4)
Xeo (0)
x(2) + x(6)
Xeo (1)
x(2) x(6)
Xoe (0)
x(1) + x(5)
Xoe (1)
x(1) x(5)
Xoo (0)
x(3) + x(7)
Xoo (1)
x(3) x(7)
Observe that eight additions are involved in the calculations in this first stage. We now combine the
above 2point DFTs to obtain the 4point DFTs of the sequences xe (n) and xo (n):
Xe (0)
Xee (0) + Xeo (0)
Xe (1)
Xe (2)
Xee (1) + W41 Xeo (1)
Xe (3)
Xo (0)
Xoe (0) + Xoe (1)
Xo (1)
Xo (2)
Xoe (0) + W41 Xoe (1)
Xo (3)
Xee (0) Xeo (0)
Xee (1) W41 Xeo (1)
Xoe (0) Xoe (1)
Xoe (0) W41 Xoe (1)
Observe that eight complex additions and four complex multiplications are again involved in this
second stage. Finally, we combine the 4point DFTs to obtain the desired 8-point DFT of x(n):
X(0)
Xe (0) + Xo (0)
X(1)
X(2)
Xe (1) + W81 Xo (1)
X(3)
X(4)
X(5)
X(6)
X(7)
Xe (2) + W82 Xo (2)
Xe (2) + W83 Xo (3)

Xe (0) Xo (0)
Xe (1) W81 Xo (1)
Xe (2) W82 Xo (2)
Xe (2) W83 Xo (3)
We again note that eight complex additions and six complex multiplications are involved in this
third stage. Note that 3 stages are all we need in this example to arrive at the desired DFT, and
3 = log2 (8). Figure 20.3 shows a flow diagram representation for the above calculations, where the
dark circles represent adders.
Computational Cost
Observe from Fig. 20.3 that the basic building block for the implementation of the decimationin-time FFT is the so-called butterfly section shown in Fig. 20.4: its input values are two
557
SECTION 20.2
x(0)
x(4)
Xe (0)
Xee (1)
Xe (1)
Xeo (0)
X(0)
X(1)
x(2)
x(6)
Xee (0)
Xe (2)
X(2)
Xe (3)
Xeo (1)
1
Xoe (0)
x(1)
X(3)
W41
Xo (0)
X(4)
x(5)
Xoe (1)
Xo (1)
Xoo (0)
x(3)
Xo (2)
1
x(7)
W41
X(5)
X(6)
W82
W83
Xo (3)
Xoo (1)
1
W81
X(7)
FIGURE 20.3 Flowgraph representation of the 8point decimation-in-time FFT from Example
20.2. The dark circles in the figure represent adders.
scalars a and b and its output values are given by

A
B
= a + bW
= a bW
(20.25)
(20.26)
in terms of a complex scaling coefficient W . Each butterfly transformation requires two

complex additions and one complex multiplication by W . For simplicity, we shall say that
each butterfly transformation requires three complex operations (additions and multiplications together). Observe further that the implementation of an N point decimation-intime FFT requires log2 (N ) stages with N/2 butterflies in each stage. Accordingly, the
computational complexity of the FFT implementation is
3
N
log2 (N ) = 1.5N log2 (N ) complex operations
2
It follows from this analysis that

N -point decimation-in-time FFT requires O (1.5N log2 N ) complex operations
(20.27)
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CHAPTER 20
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A = a + Wb
B = a Wb
FIGURE 20.4 Elementary butterfly section for decimation-in-time FFT implementations where
W is a complex scalar factor.
Example 20.3 (Cost of 1024-point DFT)

Let us reconsider the evaluation of the 1024-point DFT of Example 20.1 by using the decimation-intime FFT implementation. The computational complexity would be of the order of
1.5 1024 log 2 (1024) = 1.5 1024 10 = 15, 360 complex operations
Considering again a discrete-time processor operating at the rate of 1GHz and assuming that each
complex operation requires one clock cycle, we find that the evaluation of the 1024 DFT coefficients
in this manner would now necessitate approximately 15.3 sec; a significant improvement over 2.1
msec.
Ordering of Samples
We observe from Example 20.2 and Fig. 20.3 that the DFT coefficients X(k) appear in
their natural order on the right-hand side of the figure, while the samples of input sequence,
x(n), appear shuffled on the left-hand side of the same figure. The order of the input
samples can be inferred from the following simple rule. We express the time indices of
x(n) in binary format and then reverse the binary representation. For example, for N = 8
we obtain the construction shown in Table 20.1.
TABLE 20.1 Ordering of the input samples, x(n), in decimation-in-time FFT.
natural order
binary format
reversed order
after shuffling
0
1
2
3
4
5
6
7
000
001
010
011
100
101
110
111
000
100
010
110
001
101
011
111
0
4
2
6
1
5
3
7
Evaluating the Inverse DFT through Decimation-in-Time FFT

The same decimation-in-time procedure can be used to compute the IDFT (20.5). Thus,
given an N point DFT sequence X(k), the inverse DFT sequence x(n) can be determined
by using the same structure shown in Fig. 20.3 for the case N = 8 with the following
adjustments:
(a) The coefficients WNk are replaced by their complex conjugate values, WNk .
(b) The samples on the left-hand side of the figure would be the shuffled DFT coefficients:
{X(0), X(4), X(2), X(6), X(1), X(5), X(3), X(7)}
(c) The samples on the right-hand side of the figure should be scaled by 1/N and would
coincide with the input samples in their natural order:
{x(0), x(1), x(2), x(3), x(4), x(5), x(6), x(7)}
The procedure is illustrated in Fig. 20.5 for the case N = 8.
1/N x(0)
X(0)
X(4)
1/N
1/N x(2)
X(2)
X(6)
x(1)
1
1/N x(3)
W41
1/N x(4)
X(1)
1
X(5)
1/N
W81
1/N x(6)
X(3)
X(7)
x(5)
W82
W83
1/N x(7)
W41
FIGURE 20.5 Flowgraph representation of the 8point decimation-in-time inverse FFT. The
dark circles in the figure represent adders.
559
SECTION 20.3
DECIMATIONIN-FREQUENCY
FFT
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20.3 DECIMATION-IN-FREQUENCY FFT
CHAPTER 20
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TRANSFORM
We now describe another efficient implementation of the DFT by relying on decimation

in frequency rather than in time. In this variant, the order of the input samples is kept
unchanged while the order of the DFT coefficients is shuffled. The algorithm is motivated
as follows.
Consider again a causal sequence x(n) of duration N . We split x(n) into two subsequences, x (n) and xr (n), where x (n) consists of the leading N2 samples of x(n) and
xr (n) consists of the trailing N2 samples of x(n):
o
n
x(0) , x(1), . . . , x ((N/2) 1)
(20.28)
x (n) =
n
o

xr (n) =
x N2 , x ((N/2) + 1) , . . . , x(N 1)
(20.29)
We can also write
x (n) =
xr (n) =
x(n),
x n+
N
2
n = 0, 1, . . . , (N/2) 1
, n = 0, 1, . . . , (N/2) 1
(20.30)
Figure 20.6 illustrates this construction for a particular sequence x(n) of duration N = 8.
x(n)
2
1
4
x (n)
2
xr (n)
x(3)
x(0)
2
x(2)
x(6)
x(5)
x(7)
x(1)
x(4)
FIGURE 20.6 The sequence x(n) of duration N = 8 (top) is decimated into two smaller
sequences, x (n) and xr (n), of duration 4 samples each (bottom).
We further introduce two sequences x1 (n) and x2 (n), also of duration N/2 each and
which are defined in terms of x (n) and xr (n) as follows:
x1 (n) =
x2 (n) =
x (n) + xr (n),
n = 0, 1, . . . , (N/2) 1
[x (n) xr (n)]WNn , n = 0, 1, . . . , (N/2) 1
(20.31)
That is, x1 (n) is obtained by adding the first N2 samples of x(n) to the trailing N2 samples
of x(n). Likewise, x2 (n) is obtained by subtracting the same two sets of samples and
scaling the result by the complex factor
WNn = ej
for each n. Let X1 (k) and X2 (k) denote the

x2 (n), respectively,
2n
N
N
2 -point
N
2
X1 (k)
1
X
DFTs of the sequences x1 (n) and
x1 (n)ej N/2 kn
(20.32)
n=0
N
2
X2 (k)
1
X
x2 (n)ej N/2 kn
(20.33)
n=0
We now verify that the N point DFT of x(n) can be expressed in terms of the
DFTs, X1 (k) and X2 (k). Indeed, note that
X(k) =
N
1
X
N
2 point
x(n)WNkn
n=0
N
2
1
X
x(n)WNkn +
n=0
x(n)WNkn
n= N
2
N
2
1
X
N
1
X
x (n)WNkn +
N k/2
WN
n=0
N
2 1
X
xr (n)WNkn , k = 0, 1, . . . , N 1
n=0
Now using the fact that

N k/2
WN
= (1)k
(20.34)
we obtain the equivalent expression
N
2
X(k) =
1
X

x (n) + (1)k xr (n) WNkn , k = 0, 1, . . . , N 1
n=0
(20.35)
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SECTION 20.3
FFT
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CHAPTER 20
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TRANSFORM
We can therefore split X(k) into even and odd-indexed samples and note that these samples
coincide with the DFT coefficients X1 (k) and X2 (k) introduced earlier in (20.32)(20.33):
N
2
X(2k) =
1
X
[x (n) + xr (n)] WN2kn
n=0
N
2
1
X
kn
x1 (n)WN/2
n=0
X1 (k), k = 0, 1, . . . , (N/2) 1
N
2
X(2k + 1) =
1
X
n=0
(20.36)
(2k+1)n
[x (n) xr (n)] WN
N
2
1
X
n=0
kn
[x (n) xr (n)] WNn WN/2
N
2
1
X
kn
x2 (n)WN/2
n=0
X2 (k), k = 0, 1, . . . , (N/2) 1
(20.37)
In summary, we have shown the following so far:

(a) Starting with a sequence x(n), we split it into two sequences of duration N/2 each.
The first sequence, x (n), contains the leading N/2 entries of x(n) and the second
sequence, xr (n), contains the trailing N/2 entries of x(n).
(b) We then add the sequences x (n) and xr (n) to generate x1 (n). We also subtract the
sequences and multiply each difference by WNn to generate the sequence x2 (n).
(c) We then evaluate the N2 -point DFTs of x1 (n) and x2 (n). The DFT X1 (k) provides
the even-indexed coefficients of the desired N point DFT, X(k), while the DFT
X2 (k) provides the odd-indexed coefficients of X(k).
The same decimation procedure can now be applied to the evaluation of N2 point DFTs
X1 (k) and X2 (k) by splitting each of the sequences x1 (n) and x2 (n) into two smaller
sequences and computing their respective DFTs. For instance, we split x1 (n) into two
smaller sequences of duration N/4 each: one sequence contains the leading N/4 entries
of x1 (n) and the other sequence contains the trailing N/4 entries of x1 (n), say

N
1
(20.38)
x1 (n) =
x1 (0) , x1 (1), x1 (2), . . . , x1
4

N
N
N
x1r (n) =
x1 N4 , x1
+ 1 , x1
+ 2 , . . . , x1
1
4
4
2
(20.39)
We then introduce the sequences
x11 (n) =
x12 (n) =
x1 (n) + x1r (n),

n = 0, 1, . . . , N/4 1
n
[x1 (n) x1r (n)] WN/2
, n = 0, 1, . . . , N/4 1
(20.40)
and evaluate the N4 point DFTs of x11 (n) and x12 (n), denoted by X11 (k) and X12 (k),
respectively. The DFT X11 (k) provides the even-indexed coefficients of X1 (k) and the
DFT X12 (k) provides the odd-indexed coefficients of X1 (k):
X1 (2k) = X11 (k),
X1 (2k + 1) = X12 (k),
0 k N/4 1
0 k N/4 1
(20.41)
(20.42)
Likewise, we split x2 (n) into two smaller sequences of duration N/4 each: one sequence contains the leading N/4 entries of x2 (n) and the other sequence contains the
trailing N/4 entries of x2 (n), say

N
x2 (0) , x2 (1), x2 (2), . . . , x2
1
(20.43)
x2 (n) =
4

N
N
N
+ 1 , x2
+ 2 ,...,x
1
x2r (n) =
x2 N4 , x2
4
4
2
(20.44)
We then introduce the sequences
x21 (n) =
x22 (n) =
x2 (n) + x2r (n),

n = 0, 1, . . . , N/4 1
n
[x2 (n) x2r (n)] WN/2
, n = 0, 1, . . . , N/4 1
(20.45)
and evaluate the N4 point DFTS of x21 (n) and x22 (n), denoted by X21 (k) and X22(k),
respectively. The DFT X21 (k) provides the even-indexed coefficients of X2 (k) and the
DFT X22 (k) provides the odd-indexed coefficients of X2 (k):
X2 (2k) = X21 (k),
X2 (2k + 1) = X22 (k),
0 k N/4 1
(20.46)
0 k N/4 1
(20.47)
Observe that now we need to evaluate 4 DFTs of order N/4 each.

The decimation procedure can be repeated again and applied to each of the sequences
{x11 (n), x12 (n), x21 (n), x22 (n)}
In this way, the
N
4 point
DFTs
{X11 (k), X12 (k), X21 (k), X22 (k)}
would be computed in terms of N8 point DFTs and so on. The entire process will again
require log2 N stages of decimation with computational complexity given by
N -point decimation-in-frequency FFT requires O (1.5N log2 N ) complex operations
(20.48)
Example 20.4 (8point DFT via decimation-in-time)

Consider an 8-point sequence
n
x(n) =
x(0) , x(1), x(2), x(3), x(4), x(5), x(6), x(7)
563
SECTION 20.3
FFT
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CHAPTER 20
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TRANSFORM
We first determine the sub-sequences {x1 (n), x2 (n)} of duration 4 each:

x1 (0)
x(0) + x(4)
x1 (1)
x(1) + x(5)
x1 (2)
x(2) + x(6)
x1 (3)
x(3) + x(7)
x(0) x(4)
x2 (0)
x2 (1)
x2 (2)
x2 (3)
[x(1) x(5)] W81
[x(2) x(6)] W82
[x(3) x(7)] W83
We then determine the sub-sequences {x11 (n), x12 (n), x21 (n), x22 (n)} of duration 2 each:
x11 (0)
x1 (0) + x1 (2)
x11 (1)
x1 (1) + x1 (3)
x12 (0)
x12 (1)
x1 (0) x1 (2)
[x1 (1) x1 (3)] W41
x21 (0)
x2 (0) + x2 (2)
x21 (1)
x2 (1) + x2 (3)
x22 (0)
x22 (1)
x2 (0) x2 (2)
[x2 (1) x2 (3)] W41
We evaluate the 2point DFTs of the sequences {x11 (n), x12 (n), x21 (n), x22 (n)}:
X11 (0)
x11 (0) + x11 (1)
X11 (1)
x11 (0) x11 (1)
X12 (0)
x12 (0) + x12 (1)
X12 (1)
x12 (0) x12 (1)
X21 (0)
x21 (0) + x21 (1)
X21 (1)
x21 (0) x21 (1)
X22 (0)
x22 (0) + x22 (1)
X22 (1)
x22 (0) x22 (1)
565
and then map these coefficients into the corresponding coefficients X(k):
SECTION 20.3
X(0)
X11 (0)
X(4)
X11 (1)
X(2)
X12 (0)
X(6)
X12 (1)
X(1)
X21 (0)
X(5)
X21 (1)
X(3)
X22 (0)
X(7)
X22 (1)
FFT
Figure 20.7 shows a flow diagram representation for the above calculations, where the dark circles
represent adders.
x(0)
x(1)
x1 (0)
x11 (0)
x1 (1)
x11 (1)
X(0)
x1 (2)
x(2)
x12 (0)
X(2)
x1 (3)
x(3)
x12 (1)
1
x(4)
W41
x2 (0)
x21 (0)
x2 (1)
x21 (1)
X(4)
X(6)
X(1)
1
x(5)
1 W81
x2 (2)
x(6)
x(7)
W82
W83
x22 (0)
X(5)
X(3)
x2 (3)
x22 (1)
1
W41
X(7)
FIGURE 20.7 Flowgraph representation of the 8point decimation-in-frequency FFT. The dark
circles in the figure represent adders.
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CHAPTER 20
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Computational Cost
Observe from Fig. 20.7 that the basic building block for the implementation of decimationin-frequency FFT is again a butterfly section; albeit one with the transformation shown in
Fig. 20.8: its input values are two scalars a and b and its output values are given by
A =
B =
a+b
(a b)W
in terms of a complex scaling coefficient W . Each butterfly transformation requires two

complex additions and one complex multiplication. Observe further that the implementation of an N point decimation-in-frequency FFT requires log2 (N ) stages with N/2
butterflies in each stage. Accordingly, the computational complexity of the FFT implementation is
3
N
log2 (N ) = 1.5N log2 (N ) complex operations
2
which is of the same order as the decimation-in-time implementation:

N -point decimation-in-time FFT requires O (1.5N log2 N ) complex operations
A= a+b
B = (a b)W
FIGURE 20.8 Elementary butterfly section for decimation-in-frequency FFT implementations

where W is a complex scalar factor.
Evaluating the Inverse DFT through Decimation-in-Frequency FFT

The same decimation-in-frequency procedure can be used to compute the IDFT (20.5).
Thus, given an N point DFT sequence X(k), the inverse DFT sequence x(n) can be
determined by using the same structure shown in Fig. 20.7 for the case N = 8 with the
following adjustments:
(a) The coefficients WNk are replaced by their complex conjugate values, WNk .
(b) The samples on the left-hand side of the figure would be the DFT coefficients in
natural order:
{X(0), X(1), X(2), X(3), X(4), X(5), X(6), X(7)}
(c) The samples on the right-hand side of the figure should be scaled by 1/N and would
coincide with the input samples in a reshuffled order:
{x(0), x(4), x(2), x(6), x(1), x(5), x(3), x(7)}
567
The procedure is illustrated in Fig. 20.9 for the case N = 8.
SECTION 20.4
APPLICATIONS
1/N x(0)
X(0)
1/N
X(1)
1/N
X(2)
1
1/N
X(3)
W41
x(4)
x(2)
x(6)
1/N x(1)
X(4)
1
1/N
X(5)
W81
1/N x(3)
X(6)
X(7)
x(5)
W82
W83
1/N
W41
x(7)
FIGURE 20.9 Flowgraph representation of the 8point decimation-in-frequency inverse FFT.

The dark circles in the figure represent adders.
Relation between Decimation in Time and in Frequency

By comparing the flowgraphs of Figs. 20.3 and 20.9 for decimation-in-time FFT and
decimation-in-frequency inverse FFT, we observe a relationship between both diagrams.
If we start with the flowgraph of a decimation-in-time FFT and perform the following
sequence of operations, we arrive at the implementation for the decimation-in-frequency
inverse FFT:
(a) We invert the direction of flow of the graph and interchange the input and output
nodes.
(b) We invert the phase factors, WNk .
(c) We scale the output by 1/N .
Likewise, by comparing the flowgraphs of Figs. 20.5 and 20.7 for decimation-in-time inverse FFT and decimation-in-frequency FFT, we observe a similar relationship between
both diagrams. If we start with the flowgraph of a decimation-in-frequency FFT and perform the same sequence of operations as above, we arrive at the implementation for the
decimation-in-time inverse FFT.
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20.4 APPLICATIONS
CHAPTER 20
FAST
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TRANSFORM
TO BE ADDED
Practice Questions:
1.
2.
20.5 PROBLEMS
Problem 20.1 Let x(n) = { 1+j , 1, ej 3 , 2}.

(a) Determine the 4point DFT of x(n) using the definition (20.1).
(b) How many real additions and real multiplications are needed to evaluate the DFT of part (a)?
Count additions and multiplications separately.
(c) Draw a decimation-in-time FFT structure to evaluate X(k). How many real additions and
real multiplications are needed to evaluate the DFT coefficients in this structure?
(d) Draw a decimation-in-frequency FFT structure to evaluate X(k). How many real additions
and real multiplications are needed to evaluate the DFT coefficients in this structure?
Problem 20.2 Let x(n) = { -1 , 0, 1 j, ej 2 , 2}.
(a) Determine the 8point DFT of x(n) using the definition (20.1).
(c) Draw a decimation-in-time FFT structure to evaluate X(k). How many real additions and
real multiplications are needed to evaluate the DFT coefficients in this structure?
(d) Draw a decimation-in-frequency FFT structure to evaluate X(k). How many real additions
Problem 20.3 Let X(k) = { 1+j , 1, 1, 1 j}.
(a) Determine the inverse 4point DFT of X(k) using the definition (20.2).
(c) Draw an inverse decimation-in-time FFT structure to evaluate x(n). How many real additions
(d) Draw an inverse decimation-in-frequency FFT structure to evaluate the x(n). How many real
additions and real multiplications are needed to evaluate the DFT coefficients in this structure?
Problem 20.4 Let x(n) = { -1 , j, j, 0, 1}.
(a) Determine the inverse 8point DFT of X(k) using the definition (20.2).
(c) Draw an inverse decimation-in-time FFT structure to evaluate x(n). How many real additions
(d) Draw an inverse decimation-in-frequency FFT structure to evaluate x(n). How many real
additions and real multiplications are needed to evaluate the DFT coefficients in this structure?
569
Problem 20.5 Consider the two sequences

x1 (n) = (n) +
1
1
(n 1) and x2 (n) = (n) + (n 2)
2
2
Explain how you would evaluate the linear convolution of x1 (n) and x2 (n) by employing a decimationin-time FFT. Draw a complete diagram with all necessary weights and input and output signals
clearly indicated. The output nodes of your diagram should be the samples of the linear convolution
sequence.
Problem 20.6 The samples of a finite-duration sequence h(n) are nonzero only over 0 n 2
and they are equal to the samples of the periodic sequence
hp (n) =
X
k=
n5k
1
2
u(n 5k)
over the same interval of time.

(a) Determine the samples of h(n).
(b) Draw the diagram of the smallest radix-2 decimation-in-time FFT that computes a DFT of
h(n).
(c) Draw a radix-2 decimation-in-frequency FFT that computes the 4point DFT of the sequence
x(n) = (n) (n 2)
(d) Explain how you would combine the FFT diagrams of parts (b) and (c) in order to obtain at
the output nodes the values of the circular convolution
y(n) = { 1 , 0, 1, 0} {h(n), 0}
Show all connections. Determine also the sequence y(n).
Problem 20.7 Consider the problem of finding the circular convolution of two complex sequences,
each having 2m entries.
(a) How many complex multiplications and additions are required to directly convolve the sequences without using any DFTs?
(b) Assuming that a real addition and a real multiplication take about the same computation time,
for what values of m, less computation time will be required to perform the convolution as
above, rather than computing the FFT of each sequence, multiplying the DFT transforms, and
performing an inverse FFT?
Problem 20.8 Let x(n) and h(n) denote two real-valued causal sequences of duration 1024 and
128, respectively. Let y(n) denote the linear convolution of x(n) and y(n).
(a) How many real additions and multiplications are required to evaluate the samples of y(n)
directly from the definition of the linear convolution.
(b) How many real additions and multiplications are required to evaluate the samples of y(n) by
using the overlap-add method of Sec. 19.3, where the required smaller linear convolutions are
computed again from the definition.
(c) How many real additions and multiplications are required to evaluate the samples of y(n)
by using the overlap-save method of Sec. 19.3 where the required circular convolutions are
computed directly from the definition.
(d) How many real additions and multiplications are required to evaluate the samples of y(n)
by using the overlap-save method of Sec. 19.3 where the required circular convolutions are
computed by means of decimation-in-time FFTs.
Problem 20.9 Express the DFT of the 9point sequence {x(0), x(1), . . . , x(8)} in terms of the
DFTs of the 3point sequences:
{x(0), x(3), x(6)},
{x(1), x(4), x(7)},
{x(2), x(5), x(8)}
SECTION 20.5
PROBLEMS
590
21.4 APPLICATIONS
CHAPTER 21
SPECTRAL
RESOLUTION
TO BE ADDED
Practice Questions:
1.
2.
21.5 PROBLEMS
Problem 21.1 Consider the truncated sequence
xt (n) = cos(o n) + cos(1 n),
0nN 1
where N = 64. Assume
9
and 1 =
4
32
both in radians/sample. Let Xt (ej ) denote the DTFT of xt (n) and let X(k) denote the 64point
DFT of xt (n).
o =
(a) What is the spectral resolution of the DTFT in this case?

(b) What is the spectral resolution of the DFT in this case?
(c) What is the separation between the sinusoidal frequencies?
(d) Explain why the 64point DFT can still resolve the sinusoidal components even though the
DTFT cannot and the DFT coefficients are obtained from sampling the DTFT?
Problem 21.2 Show that the DTFT of the Kaiser window (21.6) is given by
q
W (ej ) =
N sinh
Io ()
where the hyperbolic sine function sinh is defined by

sinh(x) =
ex + ex
2

N 2
2

N 2
2
CHAPTER
22
Sampling
We indicated earlier in Chapter 1 that sequences often arise from the process of sampling continuous-time signals. In the previous chapters we studied sequences and their
properties in the time and frequency domains rather extensively. We also studied how
systems operate on sequences to transform them from one form to another.
In the current chapter we examine the sampling process more closely. Our objective is
to close the gap between the continuous-time signal, x(t), and its discrete-time counterpart,
x(n). An important question to address is whether the original continuous-time signal x(t)
can be fully recovered from knowledge of its sampled version, x(n), alone. For example,
are there conditions on how small or how large the sampling period, Ts , should be so that
one can fully recover x(t) from knowledge of its samples? As we shall see, this question
is answered by Nyquists sampling theorem.
22.1 SAMPLING PROCESS

If x(t) is a continuous-time signal, sampling it every Ts units of time results in the sequence
x(nTs ). In this construction, only values of x(t) at multiples of Ts are retained in the
sampling process and the other values of x(t) are ignored see Fig. 22.1. Usually, the
compact notation x(n) is used instead of x(nTs ) to refer to the resulting sequence with the
letter Ts dropped. We thus write x(n) to refer to x(nTs ):
x(n) = x(t)|t=nTs
(22.1)
The quantity
Fs = 1/Ts
(22.2)
is called the sampling frequency and it is measured in samples/second or Hertz (abbreviated

as Hz). The value of Fs indicates how many samples, x(n), are generated per second by
the sampling process.
Example 22.1 (Sampling frequency)
A continuous-time signal x(t) is sampled at the rate of 107 samples per second. This means that ten
million samples are generated for every duration of one second of the signal so that
Fs = 10 MHz
It also means that the sampling period is
Ts =
1
= 0.1 s
Fs
591

Copyright
592
CHAPTER 22
SAMPLING
x(t)
x(n)
x(t)
Ts
FIGURE 22.1 Sampling a continuous-time signal, x(t), at multiples of the sampling period, Ts ,
generates a sequence x(n).
so that the interval of time separating two successive samples is 0.1s. Assume that each sample is
digitized, say, as explained earlier in Sec. 1.2, and represented by 8 bits. Knowing that every 8 bits
correspond to one byte and every 1024 bytes correspond to 1KB (kilo byte), then every segment of
duration equal to one second of the signal requires
storage space
107 bytes
107 /1024 9.8MB
It is seen that the higher the sampling frequency, Fs , the more storage space is needed to store the
digitized samples of x(n).
Generally, given a continuous-time signal, x(t), the more samples we keep of it (i.e.,
the higher the sampling frequency, Fs ) the more information we have about the signal.
However, it is often desirable to reduce the number of samples that we generate in order to
save both on the storage requirement and on the computational effort that is subsequently
needed to process the samples. Intuition suggests that if the underlying signal x(t) exhibits
fast variations, then it may be necessary to sample it at a higher rate in order to capture
the fast signal variations with greater fidelity. If, on the other hand, the signal x(t) varies
slowly then we should be able to capture its behavior well with a smaller number of samples. These observations suggest that the choice of the sampling frequency, Fs , should be
related to the frequency content of x(t) and whether x(t) has high frequency components
or not. The Nyquists sampling theorem will provide an explicit condition on how low the
sampling frequency, Fs , can be without the risk of losing critical information about the
underlying continuous-time signal.
In order to introduce Nyquists result, we first review briefly some material about the
frequency representation of continuous-time signals, x(t). Only those properties that are
necessary for our arguments in this chapter are presented here. In the process of deriving the sampling theorem, we shall also highlight several useful connections between
the frequency-domain representation of continuous-time signals and the frequency-domain

representation of their discrete-time versions.
22.2 FOURIER TRANSFORM

Consider a continuous-time signal, x(t). Under some relatively mild conditions, its continuoustime Fourier Transform (FT) is defined as follows:
X(j) =
x(t)ejt dt
(Fourier transform)
(22.3)
in terms of an integral over the range t (, ). The continuous variable is called

the angular frequency and it is measured in radians/second. The variable assumes values
over the entire real axis, (, ). The Fourier transform, X(j), is in general a
complex-valued function of ; it admits both a magnitude component and a phase component.
Observe that we are denoting the argument of the FT by j instead of simply ; while
the justification for this choice of notation is unnecessary for our presentation here, it is
nevertheless useful to mention in passing that the notation is the result of specializing the
so-called Laplace transform, X(s), of the signal x(t) to the imaginary axis by setting the
complex variable s to the choice s = j:
X(s) =
x(t)est dt
(Laplace transform)
(22.4)
We shall have more to say about the Laplace transform later in Sec. 27.1.
It is instructive for the reader to note the analogy between the FT of a signal x(t) and
the DTFT of a sequence x(n). The latter was defined earlier in (13.4) as
X(ej ) =
x(n)ejn
(22.5)
n=
in terms of a frequency variable [, ] that is measured in radians/sample. The

reason for using the notation X(ej ) instead of X() was meant to highlight the fact that
the DTFT of x(n) can generally be obtained by evaluating the corresponding ztransform
on the unit circle (by setting z = ej ). We shall comment later on the exact relation
between X(j) and X(ej ) when the sequence x(n) is obtained from sampling x(t). For
now, we continue with the discussion of the FT (22.3).
Given the Fourier transform of a signal x(t), we can recover the signal via inverse
transformation as follows:
x(t) =
R
1
X(j)ejt d
2
(22.6)
It is again instructive to note the analogy with the inverse DTFT expression (13.14) for
sequences, namely,
Z
1
x(n) =
X(ej )ejn d
(22.7)
2 2
where the integral is carried over a 2long interval, such as [, ] or [0, 2].
593
SECTION 22.2
FOURIER
TRANSFORM
594
CHAPTER 22
SAMPLING
Consider a continuous-time rectangular pulse of unit amplitude and width that is centered around
the origin, t = 0 see Figure 22.2. We shall use the notation

to refer to the pulse. That is,

(

2 < t <
otherwise
1,
0,
t
FIGURE 22.2 A rectangular pulse of amplitude one and width ; the pulse is centered around the
origin of time and extends between t = 2 and t = 2 .
Using definition (22.3) we can evaluate the Fourier transform of the rectangular pulse as follows:
Z
X(j)
x(t)ejt dt
/2
ejt dt
/2
=
=
=
=
=
/2
1 jt
e

j
/2

1 j/2
e
ej/2
j
sin(/2)
/2
sin(/2)
/2
sinc(/2)
in terms of the sinc function, which was defined earlier in Example 13.8. Therefore, we arrive at the
transform pair:
FT
,
sinc
=0
6= 0
(22.8)
In other words, the Fourier transform of a rectangular pulse is a sinc function. Recall that a sinc
function assumes zero values whenever its argument is a multiple of . In the current example, this
means that the Fourier transform of the rectangular pulse will evaluate to zero at values that satisfy
595
SECTION 22.2
= m,
2
FOURIER
TRANSFORM
for any nonzero integer m
Figure 22.3 illustrates the Fourier transform of the rectangular pulse of width = 2.
Fourier transform
2
amplitude
1.5
0.5
20
15
10
5
0
5
(radians/second)
10
15
20
FIGURE 22.3 The Fourier transform of the rectangular pulse (t/2), which has amplitude 1,
width 2 and is centered around the origin of time, t = 0. The Fourier transform is displayed over the
range [20, 20].
Example 22.3 (Sinc function)

Consider now the continuous-time function

x(t) = sinc
t
To
8
< 1,
t=0
sin(t/To )
,
:
t/To
t 6= 0
This is a sinc function that attains the value of one at t = 0 and evaluates to zero whenever t satisfies
t
= m,
To
for any nonzero integer m
That is, x(t) is zero at all nonzero multiples of To ,

t = mTo ,
m 6= 0
Figure 22.4 plots the function x(t) over the interval t [5, 5] for the case To = 1.
We now verify that the following Fourier transform pair holds
(
1,
sinc(t/To ),
t=0
t 6= 0
FT
To
(22.9)
596
x(t)=sinc( t)
1
CHAPTER 22
SAMPLING
0.8
amplitude
0.6
0.4
0.2
0.2
5
0
t (sec)
FIGURE 22.4 A plot of the sinc function x(t) = sinc(t) over the interval [5, 5]. The function
attains the value of one at t = 0 and crosses zero at integer values of t.
where
2
To
In other words, the Fourier transform is a rectangular pulse in the frequency domain with amplitude
To , centered at the origin, = 0, and located between o /2 and o /2 see Fig. 22.5.
o =
To
To
To
2o
o
2
(radians/sec)
FIGURE 22.5 A rectangular pulse in the frequency domain with amplitude To and width o ;
the pulse is centered around the origin of frequency. This pulse is the Fourier transform of the sinc
function, x(t) = sinc(t/To ).
The above transform pair can be established by employing the inverse Fourier transform formula
(22.6), namely,
x(t)
1
2
To
2
X(j)e
jt
FOURIER
TRANSFORM
o /2
o /2
ejt d

To 1 jt o
e
2 jt
o /2
/2

To 1 jo t/2
e
ejo t/2
2 jt
sin(t/To )
, using o = 2/To
t/To
=
=
as desired.
Example 22.4 (Impulse function)

Consider the signal
x(t) = (t T )
(22.10)
which is defined in terms of the continuous-time delta function. Recall that in continuous-time, the
impulse function (t) is defined by the following properties:
Z
(t)dt = 1,
f (t)(t to )dt = f (to )
(22.11)
for any function f (t) that is well defined at t = to . The second property is known as the sifting
property; it extracts the value of the function at the location of the impulse function, t = to .
Using the definition (22.3), and the above sifting property, we can evaluate the Fourier transform
of (t T ) as follows:
Z
X(j)
=
=
=
x(t)ejt dt
(t T )ejt dt
jT
Therefore, we arrive at the Fourier transform pair

FT
(t T ) ejT
(22.12)
Example 22.5 (Train of impulses)

Consider now the function
x(t) =
X
n=
597
SECTION 22.2
(t nTo )
in terms of continuous-time impulse functions, (t). The signal x(t) so defined is referred to as
a train of impulses of unit amplitude; it is a periodic signal of period To and consists of impulses
598
CHAPTER 22
SAMPLING
located at t = nTo for all integers n. It turns out that the Fourier transform of the train of impulses
is another train of impulses in the frequency domain with period o = 2/To and amplitude o ,
namely,
P
n=
(t nTo ) o
( ko ) ,
o = 2/To
(22.13)
k=
Figure 22.6 illustrates the signal x(t) and its Fourier transform.
x(t) =
(t nTo )
n=
3To 2To To
To
2To
3To
X(j) = o
( ko )
n=
3o
2o
2o
3o
(radians/sec)
FIGURE 22.6 A periodic train of impulses in the time domain (top) with period To . Its Fourier
transform is another train of impulses in the frequency domain with amplitude o and period o =
2/To (bottom).
One way to justify the result (22.13) is as follows. Since the Fourier transform of (t to ) is
ejto , from the linearity of the Fourier transform we have
X(j) =
ejnTo
(22.14)
k=
Now recall that in Prob. 13.31 we argued that the DTFT of the sequence y(n) = 1 for all n is given
by
Y (ej ) =
n=
ejn =
k=
2 ( 2k)
Comparing with (22.14), we note that we can interpret expression (22.14) for X(j) as the DTFT
of the unit sequence evaluated at = To , i.e.,
X(j) =
n=

jn
e

= Y (ej )|=To
=To
Consequently,
X(j) = 2
X
k=
(To 2k)
Let o = 2/To . Then, from the scaling property of the delta function, namely, (at) = a1 (t), we
arrive at
X(j) = o
k=
( ko )
as desired.
Example 22.6 (Modulation property)

Consider a signal x(t) with Fourier transform X(j). It holds that
ejo t x(t) X (j( o ))
(22.15)
That is, if the time-domain signal is multiplied by a complex exponential signal at frequency = o ,
where o can be positive or negative, then its Fourier transform is shifted to the right (when o > 0)
or to the left (when o < 0) by an amount equal to |o | see Fig. 22.7.
X(j)
(rad/sec)
Y (j)
B + o
B + o (rad/sec)
FIGURE 22.7 Illustration of the modulation property for a positive value of o . The figure shows
the Fourier transform of x(t) (top) and the Fourier transform of y(t) = ejo t x(t) (bottom).
599
SECTION 22.2
FOURIER
TRANSFORM
600
Proof: Using the definition (22.3) we have

Z
CHAPTER 22
SAMPLING
Y (j)
x(t)ejo t ejt dt
=
=
x(t)ej(o )t dt
X(j( o ))
Example 22.7 (Symmetry property)
Consider a real-valued signal x(t) with Fourier transform X(j). Then the following symmetry
relations hold
|X(j)| = |X(j)|
X(j) = X(j)
and
(22.16)
That is, the magnitude response is an even function of while the phase response is an odd function
of .
Proof: Using the definition (22.3) and Eulers relation (3.9), we have
Z
X(j)
x(t)ejt dt
x(t) [cos(t) j sin(t)] dt
Therefore, since x(t) is real, the real and imaginary components of X(j) are given by
Z
XR (j) =
x(t) cos(t)dt,
XI (j) =
x(t) sin(t)dt
where XR and XI denote the real and imaginary components of X(j), respectively. It is now clear
that
XR (j) = XR (j) and XI (j) = XI (j)
which leads to the desired result since
|X(j)| =
|XR (j)|2 + |XI (j)|2 ,
X(j) = arctan
XI (j)
XR (j)
Example 22.8 (Scaling property and bandlimited signals)

The Fourier transform of a continuous-time signal satisfies the scaling property
x(at)
1
X
|a|
j
a
(22.17)
where a is a nonzero number. Before establishing the result, we note that this property has a useful
interpretation. For scalars |a| > 1, the graph of x(at) is compressed in relation to the graph of x(t),
601
SECTION 22.2
x(t)
FOURIER
TRANSFORM
t1
to
x(2t)
to
2
t21
x(t/2)
2t1
2to
FIGURE 22.8 Illustration of the compression and expansion behavior that results from replacing
x(t) (top) by x(2t) (middle) and x(t/2) (bottom).
while for |a| < 1, the graph of x(at) is expanded in relation to the graph of x(t) see Fig. 22.8. A
similar interpretation holds for functions in the frequency domain.
By examining the scaling property (22.17) we see that whenever compression occurs in the time
domain, it leads to expansion in the frequency domain and vice-versa. For instance, note that as the
duration of a time-domain signal decreases (say by compressing its time-scale), then the domain of
its Fourier transform expands. It follows that if a signal x(t) has finite duration, then its Fourier
transform cannot be limited in bandwidth. By the same token, if a signal has finite bandwidth, then
it should have infinite duration.
Proof: Let y(t) = x(at). Assume initially that a > 0. Then, using the definition (22.3) we have
Z
Y (j)
=
=
=
=
x(at)ejt dt
x( )ej /a d /a,
1
a
1
X
a
x( )ej /a d
j
a
using = at
602
Assume now that a < 0. Then
CHAPTER 22
SAMPLING
Y (j)
=
=
=
=
x(at)ejt dt
1
a
1
X
|a|
x( )ej /a d /a,
x( )ej /a d
using = at
j
a
which establishes (22.17).
Table 22.1 summarizes some of the properties of the Fourier transform for ease of reference.
TABLE 22.1 Several properties of the Fourier Transform.
Signal
Fourier Transform
x(t)
X(j)
y(t)
Y (j)
ax(t) + by(t)
aX(j) + bY (j)
linearity
x(at)
1
X( j
)
a
|a|
scaling
X(t)
2 x(j)
duality
x(t to )
ejto X(j)
time-shifts
ejo t x(t)
X(j jo )
modulation
Rt
1
X(j) + X(0) (j)
j
integration
j X(j)
differentiation
X(j)Y (j)
convolution
1 R
X(j)Y (j j)d
2
multiplication
x( )d
dx(t)/dt
R
x()y(t )d
x(t)y(t)
Property
22.3 LINEAR CONVOLUTION

Consider two continuous-time functions x(t) and h(t). Their linear convolution is the
function y(t), denoted by
y(t) = x(t) h(t)
603
and which is obtained as follows:
SECTION 22.3
y(t) =
x()h(t )d
(22.18)
It is again instructive for the reader to note the analogy with the discrete-time linear convolution of two sequences x(n) and h(n), which we defined earlier in (5.6) as
y(n) = x(n) h(n) =
k=
x(k)h(n k)
Just as was the case with discrete-time, the continuous-time convolution of two signals admits several useful properties both in the time-domain and in the frequency domain. Here
we highlight only three of these properties since they are of immediate interest to our discussions on sampling.
Convolving with an Impulse Function

The first property we are interested in is the fact that
x(t) (t T ) = x(t T )
(22.19)
In other words, convolving a signal x(t) with a continuous-time impulse function at t = T

results in shifting the signal into the location of the impulse see Fig. 22.9.
Proof: Let
y(t) = x(t) (t T )
Then, according to the definition of the linear convolution,
Z
y(t)
=
=
x()(t T )d
x(t T )
where in the second equality we used the sifting property (22.11) of impulse functions. The property
states that we should evaluate x() at the location of the impulse function (t T + ); this impulse
occurs at location = t T .
Convolution in Time
The second property we are interested in pertains to the Fourier transform of the convolution of two signals. The following transform pair relation holds:
FT
x(t) h(t) X(j)H(j)
(22.20)
In other words, the Fourier transform of the convolution of two signals is given by the
product of the individual Fourier transforms.
Proof: Let
y(t) = x(t) h(t)
LINEAR
CONVOLUTION
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CHAPTER 22
SAMPLING
x(t) (t T )
x(t)
FIGURE 22.9 Convolving a signal x(t) with an impulse located at t = T , namely, (t T ),

results in shifting the signal into the location of the impulse function.
Then, according to the definition of the linear convolution,

Z
y(t)
x()h(t )d
Using definition (22.3) of the Fourier transform we have

Z
Y (j)
=
=
y(t)ejt dt
Z
=
=
x()h(t )d ejt dt
x()
Z
Z
x()
Z
h(t )ej(t) dt ej d

h()e
Z
x()ej d
d ej d,
using = t
h()ej d
X(j)H(j)
Example 22.9 (Convolution with a train of impulses)

Consider a signal x(t) and let us convolve it with a train of impulses, say
y(t) = x(t)
X
n=
(t nTo )
The train of impulses is periodic with period To . The above convolution therefore amounts to repeating x(t) periodically every To units of time.
According to the convolution property (22.20), the Fourier transform of the signal y(t) is obtained
by multiplying the Fourier transform of x(t) with the Fourier transform of the train of impulses. We
605
already stated in Example 22.5 the transform pair
X
n=
(t nTo ) o
X
k=
( ko ) ,
SECTION 22.3
LINEAR
CONVOLUTION
o = 2/To
which determines the Fourier transform of the train of impulses. Combining this result with the
convolution property (22.20) we get
Y (j) = X(j) o
X
k=
( ko )
Note that the effect of multiplying the function X(j) by a periodic train of impulses is to extract
samples from X(j) at the locations of these impulses. Thus, the above relation for Y (j) can be
rewritten as
!
Y (j) = o
k=
X(jko ) ( ko )
where the amplitudes of various impulse functions are now modulated by X(j). The result of this
construction is illustrated in Fig. 22.10. We therefore say that multiplication by a train of impulses in
the time domain corresponds to sampling the Fourier transform of the signal in the frequency domain
and scaling the result by o . We also say that periodic repetition in the time domain amounts to
sampling in the frequency domain.
X(j)
1
(radians/sec)
Y (j)
o
(radians/sec)
FIGURE 22.10 Convolving a signal x(t) with a periodic train of impulses of period To in the
time domain corresponds to sampling its Fourier transform at multiples of o = 2/To and scaling
the result by o . The top plot shows an example of a Fourier transform X(j) and the bottom plot
shows the resulting Y (j). The amplitude values of 1 and o at = 0 are meant to illustrate the
scaling that occurs when going from X(j) to its sampled version Y (j).
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CHAPTER 22
SAMPLING
Example 22.10 (Low-pass filtering)

Consider a signal x(t) and assume we convolve it with the signal
1
sinc
h(t) =
To
t
To
8
< 1/To ,
t=0
1 sin(t/To )
,
:
To t/To
t 6= 0
That is, let us study the signal

y(t) = x(t) h(t)
The signal h(t) is a sinc function and it was studied in Example 22.3 where we showed that its
Fourier transform is the rectangular pulse

H(j) =
This pulse is shown in the middle row of Fig. 22.11. We see that H(j) extends between o /2
and o /2 and is zero elsewhere. We normalized the definition of h(t) above by 1/To so that the
amplitude of H(j) is set to one.
Using the convolution property (22.20) we know that the Fourier transform of the convolution
signal y(t) is given by
Y (j) = X(j) H(j)
We therefore see that the effect of convolving with the sinc function h(t) is to limit the frequency
components in X(j) to the range [ 2o , 2o ]. We say that H(j) performs low-pass filtering
since it retains the low frequency components of X(j). This situation is illustrated by means of an
example in Fig. 22.11. The plot in the top row shows a Fourier transform that extends beyond the
range [ 2o , 2o ]. The middle row shows the Fourier transform of the sinc function h(t) where the
value of o defines To through the relation o = 2/To . The last row of the figure shows the result
of the convolution of x(t) and h(t). It is seen that the parts of the Fourier transform of x(t) that lie
outside the frequency range of the low-pass transform H(j) are removed.
Multiplication in Time is Convolution in Frequency

The third property we are interested in pertains to the Fourier transform of the product of
two signals. The following transform pair relation holds:
x(t)h(t)
R
1
1
X(j)H(j j)d
X(j) H(j) =
2
2
(22.21)
In other words, the Fourier transform of the product of two signals is given by the (scaled)
linear convolution of the individual Fourier transforms.
Proof: Let Y (j) denote the following scaled convolution:
Y (j) =
1
2
X(j)H(j j)d
607
SECTION 22.3
X(j)
LINEAR
CONVOLUTION
(radians/sec)
o
2
2o
H(j) =
To
2o
(radians/sec)
o
2
Y (j)
(radians/sec)
o
2
2o
FIGURE 22.11 Convolving a signal x(t) with the sinc function h(t) = 1/To sinc(t/To ) results
in a new signal y(t) whose Fourier transform is limited to the range [ 2o , 2o ]. The top row of the
figure shows a Fourier transform that extends beyond the range [ 2o , 2o ]. The middle row shows
the Fourier transform of the sinc function h(t) = 1/To sinc(t/To ) where the value of To is related
to o via o = 2/To . The last row shows the result of the convolution of x(t) and h(t).
and let us verify that its inverse Fourier transform coincides with the product x(t)h(t). Indeed, using
the inverse transformation relation (22.6) we have
y(t)
=
=
1
2
1
2
=

=

=
=
1
2
1
2
1
2
Y (j)ejt d

x(t)h(t)
1
2
X(j)H(j j)d ejt d

X(j)ejt d

1
2
1
2
X(j)ejt d
1
2
X(j)e
jt

H(j j)ej()t d

H(j)e
jt
d ,

using =
H(j)ejt d
608
CHAPTER 22
SAMPLING
Example 22.11 (Multiplication by a train of impulses)

Consider a signal x(t) and let us multiply it by a train of impulses, say
!
y(t) = x(t)
n=
(t nTo )
The train of impulses is periodic with period To . The above multiplication therefore amounts to
sampling x(t) in the time domain at multiples of To . According to the convolution property (22.21),
the Fourier transform of the signal y(t) is obtained by convolving the Fourier transform of x(t) with
the Fourier transform of the train of impulses, and scaling the result by 1/2. We already stated in
Example 22.5 the transform pair
X
n=
(t nTo ) o
X
k=
( ko ) ,
o = 2/To
which gives the Fourier transform of the train of impulses. Combining this result with the convolution
property (22.21) we get
1
Y (j) =
X(j)
2
X
k=
( ko )
Recall further from property (22.19) that convolving a signal with an impulse function results in
shifting the signal to the location of the impulse function. This property of convolution holds regardless of whether we are convolving functions in the time domain or the frequency domain. In the
above expression for Y (j), we are convolving X(j) with several delayed impulse functions and,
consequently, the relation for Y (j) can be rewritten as
1
Y (j) =
To
X
k=
X(j jko )
This expression reveals that the Fourier transform of Y (j) will consist of the Fourier transform of
X(j) repeated periodically every o radians/sec and scaled by 1/To . The result of this construction
is illustrated in Fig. 22.12. We therefore say that multiplication by a train of impulses in the time
domain corresponds to periodically repeating the Fourier transform of the signal in the frequency
domain and scaling the result by 1/To . We also say that sampling in the time domain corresponds to
periodic repetition in the frequency domain.
22.4 LINEAR TIME-INVARIANT SYSTEMS

Our presentation on sampling will benefit from properties of systems in continuous-time.
These properties are analogous to what we have seen in discrete-time in Chapters 4 and 5,
and we shall therefore be brief in reviewing them here.
Systems. A continuous-time system is defined as a mapping (or transformation) between
an input signal, x(t), and an output signal, y(t) see Fig. 22.13. What makes a system
special is that the input signal, x(t), uniquely defines the output signal, y(t).
Causality. A continuous-time system is causal if its output at time t depends only on
present and past values of the input signal. In other words, y(t) depends only on x( ) for
609
X(j)
SECTION 22.4
LINEAR
TIME
INVARIANT
SYSTEMS
(radians/sec)
Y (j)
1/To
(radians/sec)
FIGURE 22.12 Multiplying a signal x(t) with a periodic train of impulses of period To in the
time domain corresponds to repeating its Fourier transform periodically at multiples of o = 2/To
and scaling the result by 1/To . The top plot shows an example of a Fourier transform X(j) and
the bottom plot shows the resulting Y (j). The amplitude values of 1 and 1/To at = 0 are meant
to illustrate the scaling that occurs when going from X(j) to Y (j).
input signal
system
S
x(t)
output signal
y(t)
FIGURE 22.13 A continuous-time system S maps an input signal, x(t), into an output or
response signal, y(t).
t. Otherwise, the system is noncausal.

Stability. A continuous-time system is bounded-input bounded-output (BIBO) stable if
every bounded input signal, x(t), yields a bounded output signal y(t). By definition, a
bounded signal x(t) is one for which all its values are bounded by some finite positive
number Bx :
|x(t)| Bx < for all t
(22.22)
Time-Invariance. A continuous-time system is time-invariant if a time delay (or advance) in the input signal yields an identical delay (or advance) in the output signal. In
other words, if
y(t) = S[x(t)]
describes a system, then it should hold that
y(t ) = S[x(t )],
for any real
(22.23)
610
CHAPTER 22
SAMPLING
Linearity. A continuous-time system is linear if it satisfies the superposition principle,

which states that
ay1 (t) + by2 (t) = S[ax1 (t) + bx2 (t)]
(22.24)
for any constants {a, b}. In other words, the response of the system to any linear combination of input signals is the same linear combination of the corresponding output signals.
To be more precise, we should also require the superposition property to hold for a combination of an infinite number of input signals.
Impulse Response. The impulse response of a system is denoted by h(t) and is defined
as the response of the system to the impulse function, x(t) = (t), i.e.,
h(t) = S[(t)]
(22.25)
Linear Time-Invariance. The input-output relation of linear-time-invariant (LTI) continuoustime systems is fully characterized by the convolution integral
y(t) =
x( )h(t )d
(22.26)
In other words, for any input signal, x(t), the corresponding output signal, y(t), can be
determined by convolving x(t) with the impulse response, h(t). This is because we can
express a signal x(t) through the integral representation
x(t) =
x( )(t )d
so that, by linearity,
S[x(t)]
Z

= S
x( )(t )d
Z
=
x( )S [(t )] d
Z
=
x( )h(t )d
as desired. The Fourier transform of h(t) is called the frequency response of the LTI system.
Stability and Causality of LTI Systems. A continuous-time LTI system is stable if,
and only if, its impulse response is absolutely integrable:
LTI system is stable
|h(t)|dt <
(22.27)
Likewise, a continuous-time LTI system is causal if, and only if, its impulse response is a
causal signal, i.e.,
LTI system is causal
h(t) = 0 for all t < 0
(22.28)
The proofs of the above two statements are analogous to what was done in discrete-time in
Secs. 5.2 and 5.3 and are therefore omitted.
22.5 NYQUIST RATE FOR BASEBAND SIGNALS

The results of Examples 22.922.11 provide the necessary building blocks for the derivation of Nyquists sampling theorem. The theorem will give a lower bound on how small
the sampling frequency, Fs , can be so that we can still recover the original signal x(t) from
its sampled version x(n).
We start our exposition with a generic continuous-time signal x(t) and assume that its
Fourier transform, X(j), is bandlimited to B radians/second. That is, X(j) lies within
the range [B, B]. Such signals are called baseband signals. In practice, the condition
usually means that the portions of X(j) that exist outside the range [B, B] are negligible. This situation is illustrated in Fig. 22.14; the curves shown in the figure are for
illustration purposes only and they do not correspond to an actual Fourier transform pair.
Remark. Recall from the result of Example 22.8 that a signal that is bandlimited in frequency must
necessarily have infinite-duration in the time-domain. We shall ignore this technical detail in our presentation and assume that the time-domain signal decays sufficiently fast so that it can be assumed
to be reasonably confined to a finite duration.
x(t)
X(j)
1
t (s)
(rad/s)
FIGURE 22.14 A signal x(t) in the time domain (left) and its assumed Fourier transform (right),
where is measured in radians/second and t is measured in seconds. The amplitude value of one
is meant as a reference point to illustrate the scaling that occurs as we go through the sampling
procedure in future plots. It should be noted that the curves shown in this figure are for illustration
purposes only; they do not correspond to an actual Fourier transform pair.
Sampling in Time
Now given x(t), we sample it every Ts seconds. The sampling process corresponds to
multiplying x(t) by a periodic train of impulses of period Ts . This operation generates a
sampled signal, which we shall denote by xs (t) see Fig. 22.15:
!
X
xs (t) = x(t)
(t nTs )
n=
n=
x(nTs )(t nTs )
(22.29)
611
SECTION 22.5
NYQUIST
RATE
FOR
BASEBAND
SIGNALS
612
CHAPTER 22
SAMPLING
This situation is analogous to the case that was discussed in Example 22.11. We find that
xs (t) consists of a train of impulses that are modulated by the amplitude of x(t).
xs (t)
x(t)
xs (t)
x(t)
X
3Ts 2Ts Ts
Ts
2Ts
3Ts
FIGURE 22.15 Multiplying a continuous-time signal, x(t), by a train of impulses results in a

sampled signal xs (t).
According to the convolution property (22.21), and as was already explained in the
aforementioned example, the Fourier transform of the signal xs (t) is obtained by convolving the Fourier transform of x(t) with the Fourier transform of the train of impulses. We
already stated in Example 22.5 the transform pair
!
X
X
(t nTs ) s
( ks ) ,
s = 2/Ts
n=
k=
which gives the Fourier transform of the train of impulses. Combining this result with the
convolution property (22.21) we get
!
X
1
X(j) s
( ks )
Xs (j) =
2
k=
!
X
1
X(j jks )
(22.30)
=
Ts
k=
That is, the Fourier transform of Xs (j) will consist of the Fourier transform of X(j)
repeated periodically every s radians/second and scaled by 1/Ts . The result of this construction is illustrated in Fig. 22.16 for the same signals of Fig. 22.14.
Useful conclusion: We therefore say that sampling in time corresponds to periodically repeating the Fourier transform of the signal in the frequency domain every s radians/second and scaling the result by 1/Ts .
Nyquists Condition
If we examine the Fourier transform of the sampled signal, Xs (j), in Fig. 22.16, we notice that several images (or bands) of the original Fourier transform X(j) are generated;
these images are centered at multiples of s . Let us focus on the part that corresponds to
613
x(t)
SECTION 22.5
X(j)
NYQUIST
RATE
FOR
BASEBAND
SIGNALS
B (rad/s)
t(s)
(t nTs )
n=
t(s)
Ts
repeat periodically
every s radians/second
and scale the result by 1/Ts
xs (t)
t(s)
Xs (j)
1/Ts
B s
B
B + s
(rad/sec)
B + s
FIGURE 22.16 Multiplying a signal x(t) with a periodic train of impulses of period Ts in the
time domain corresponds to repeating its Fourier transform periodically at multiples of s = 2/Ts
and scaling the result by 1/Ts . The top plot shows an example of a Fourier transform X(j) and the
bottom plot shows the resulting Xs (j). The amplitude values of 1 and 1/Ts at = 0 are meant to
illustrate the scaling that occurs when going from X(j) to Xs (j). The second row in the figure
shows the periodic train of impulses in the time domain and the third row shows the sampled signal
xs (t).
the original transform, and which stretches from B to B, and on the image that is to the
right of it, and which stretches from B + s to B + s see Fig. 22.17. In the figure
we are using markers to indicate the boundaries between adjacent images. It is seen that
two adjacent images will not interfere with each other if the value of s is chosen such that
B < B + s
or, equivalently,
s > 2B
(22.31)
614
CHAPTER 22
SAMPLING
In this case, the portion of the Fourier transform Xs (j) that lies within the signal bandwidth [B, B] will only be a scaled version (scaled by 1/Ts ) of the original Fourier transform X(j).
Nyquists Sampling Theorem: If the continuous-time signal x(t) is bandlimited, say,
X(j) = 0
for
|| > B
(22.32)
then no overlaps occur in the Fourier transform of the sampled signal, xs (t), if the sampling rate is larger than twice the signal bandwidth.
The result (22.31) is known as Nyquists sampling condition. It provides a lower limit
on the sampling frequency, namely, the sampling frequency needs to be at least twice as
large as the bandwidth of the original signal. This minimum sampling frequency is usually
called the Nyquist rate or Nyquist frequency. We are going to see that when Nyquists
condition (22.31) is satisfied, it is possible to recover the original signal x(t) from its
sampled version xs (t) (and, hence, from the corresponding sequence x(n)).
X(j)
1
B (rad/s)
Xs (j)
1/Ts
B s
B
B + s
(rad/sec)
B + s
FIGURE 22.17 Overlap does not occur between adjacent images in the Fourier transform of
Xs (t) if, and only if, the sampling frequency s guarantees that B < B + s , which is equivalent
to requiring s > 2B. The markers in the figure indicate the boundaries between adjacent images.
Figure 22.18 illustrates what happens when the sampling frequency is smaller than 2B,
i.e., when it violates Nyquists condition. In this case, the images are not separated enough
and overlap occurs between adjacent images in the Fourier transform of Xs (j). It is
seen that the portion of the Fourier transform Xs (j) that lies within the signal bandwidth
[B, B] is now distorted relative to the original Fourier transform X(j).
615
SECTION 22.5
X(j)
NYQUIST
RATE
FOR
BASEBAND
SIGNALS
B (rad/s)
Xs (j)
1/Ts
s B
(rad/sec)
B + s
FIGURE 22.18 Overlap occurs between adjacent images in the Fourier transform of Xs (t) when
the sampling frequency s violates Nyquists condition and is smaller than 2B.
Example 22.12 (Speech signals)

Speech signals are generally bandlimited at B = 4KHz. That is, the Fourier transform of a speech
signal usually contains significant frequency components inside the range [4, 4] KHz. Therefore,
Nyquists rate for sampling speech signals is
Fs = 2 4 = 8 KHz
This means that employing 8000 samples per second satisfies Nyquists condition (22.31). This
sampling rate is used by most telephony systems.
Example 22.13 (Audio signals)
Audio signals are usually bandlimited at B = 20KHz. These signals are sampled at Fs = 48 KHz
in professional audio systems and at Fs = 44.1 KHz in compact disc (CD) storage systems. Both
sampling rates meet Nyquists condition (22.31).
Signal Reconstruction
Assume the sampling frequency, s , is large enough and satisfies Nyquists condition
(22.31), i.e.,
s 2B
so that
s
B
2
616
CHAPTER 22
SAMPLING
Then no overlap occurs between adjacent bands in the Fourier transform of Xs (j). We
now show how the original signal x(t) can be recovered from its sampled version xs (t)
(and, consequently, from the corresponding sequence x(n)).
To achieve this objective, we call upon the result of Example 22.10. Thus, consider the
sinc function

1,
t=0
t
sin(t/Ts )
h(t) = sinc
(22.33)
=
,
t 6= 0
Ts
t/Ts
The signal h(t) was studied in Example 22.3 where we showed that its Fourier transform
is the rectangular pulse:

Ts ,
2s < < 2s
=
H(j) = Ts
(22.34)
0,
otherwise
s
This pulse is shown in the third row of Fig. 22.19. We see that H(j) extends between
s /2 and s /2 and is zero elsewhere. Its amplitude is equal to Ts .
We now multiply the Fourier transforms Xs (j) and H(j) and denote the result by
Y (j) = Xs (j) H(j)
(22.35)
The effect of this multiplication is to limit the frequency components in Y (j) to the range
[ 2s , 2s ]. This situation is illustrated in the last row of Fig. 22.19. The plot in the top row
shows the original Fourier transform X(j), which is bandlimited to B radians/second.
The second row shows the Fourier transform of the sampled signal, xs (t): it consists of
periodic repetitions of the transform of x(t) scaled by 1/Ts . The third row shows the
low-pass Fourier transform of h(t) stretching over the frequency range [ 2s , 2s ] with
amplitude Ts . Multiplying Xs (j) and H(j) extracts that portion of Xs (j) that lies
within the same frequency range [ 2s , 2s ]. The result is Y (j) and is shown in the last
row of the figure; note that Y (j) and X(j) agree with each other.
The discussion therefore shows that
X(j) = Xs (j) H(j) = Y (j)
(22.36)
so that X(j) can be recovered from Xs (j) by multiplying the latter by the reconstruction filter H(j). Using the convolution property (22.20) we can translate this result into
the time domain and write
x(t) = xs (t) h(t)
(22.37)
That is, the original signal x(t) can be recovered from xs (t) by convolving the latter with
the sinc function h(t). Let us examine this result more closely.
Using expressions (22.29) and (22.33) for xs (t) and h(t), respectively, and substituting
into the above expression for x(t) we obtain
x(t)
=
=
xs (t) h(t)
!

X
t
x(nTs )(t nTs ) sinc
T
s
n=
which shows that x(t) is obtained by convolving the sinc function with a periodic train of
impulses. Using property (22.19) about convolution with impulse functions we arrive at
the following useful expression for x(t) in terms of the sampled signal xs (t):
x(t) =
x(nTs ) sinc
n=
(tnTs )
Ts
(22.38)
617
SECTION 22.5
X(j)
NYQUIST
RATE
FOR
BASEBAND
SIGNALS
B (rad/s)
Xs (j)
1/Ts
2s
(rad/sec)
s
2
H(j)
Ts
2s
s
2
(rad/sec)
Y (j) = X(j)
1
B (rad/s)
FIGURE 22.19 The top row shows the original Fourier transform X(j), while the second row
shows the Fourier transform of the sampled signal xs (t): it consists of periodic repetitions of the
transform of x(t) scaled by 1/Ts . The third row shows a low-pass Fourier transform stretching over
the frequency range [ 2s , 2s ] with amplitude Ts . Multiplying Xs (j) and H(j) extracts that
portion of Xs (j) that lies within the same frequency range [ 2s , 2s ], and which agrees with the
original Fourier transform X(j).
This result shows that x(t) consists of the sum of modulated sinc functions that are centered at multiples of Ts and scaled by x(nTs ). The construction (22.38) is illustrated in
Fig. 22.20.
Recall that the sinc function

t
sinc
Ts
is one at t = 0 and is zero at multiples of Ts . Therefore, the reconstruction procedure

(22.38) for recovering x(t) from xs (t) consists of associating with each sampling instant
a sinc function whose amplitude is the sample value x(nTs ) and is centered at nTs . The
618
CHAPTER 22
SAMPLING
sinc functions do not interfere with each other at the sampling instants nTs since they all
evaluate to zero at these points, except the function that is centered at nTs .
xs (t) sinc
t
Ts
x(0)
x(Ts )
x(2Ts )
Ts 2Ts
FIGURE 22.20 Convolving xs (t) with the sinc function sinc(t/Ts ) results in placing a sinc at
every multiple of Ts and scaling it by the amplitude of the sample at that location, x(nTs ). A sinc
function at location nTs evaluates to zero at all other multiples of Ts . The combination of all sinc
functions enables the reconstruction of the values of x(t) for all t.
Aliasing
When the sampling frequency, s , does not satisfy Nyquists condition (22.31), we find
that overlap occurs between the bands in the Fourier transform of Xs (j), as was illustrated in Fig. 22.18. We say that aliasing occurs. As a result, multiplying Xs (j) by
the low-pass Fourier transform H(j) will not allow us to recover X(j). The resulting Fourier transform, Y (j), will be different from X(j), as illustrated in the steps of
Fig. 22.21.
Often in practice, the Fourier transform of the original signal x(t) is not bandlimited to
some sharp frequency range [B, B]. Instead, the transform may extend beyond this range
and have frequency components of negligible amplitude outside [B, B]. For this reason,
it is common practice to first pass the continuous-time signal x(t) through a so-called antialiasing filter, a(t), before sampling it. That is, x(t) is convolved with a(t) to generate an
intermediate signal z(t):
z(t) = x(t) a(t)
The purpose of the anti-aliasing filter a(t) is to limit the bandwidth of z(t) to the range
[B, B]. The filter a(t) is generally a low-pass filter whose Fourier transform leaves
unaltered frequency components within the range [B, B] and significantly attenuates frequency components outside this range. In this way, when z(t) is sampled at some suitable
rate satisfying s > 2B, then aliasing will not occur see Fig. 22.22.
619
X(j)
SECTION 22.5
NYQUIST
RATE
FOR
BASEBAND
SIGNALS
B (rad/s)
Xs (j)
1/Ts
s B
2s
(rad/sec)
B
s
2
H(j)
Ts
2s
s
2
(rad/sec)
Y (j) = X(j)
1
2s
(rad/s)
B
s
2
FIGURE 22.21 When aliasing occurs, multiplying Xs (j) by the low-pass Fourier transform
H(j) does not recover X(j). Instead a distorted Fourier transform is obtained whose inverse
transform will not lead to x(t) but to some other signal.
Zero-Order-Hold Reconstruction
In practice, it is not possible to implement the low-pass filtering operation (22.35). There
are several ways to illustrate the impracticality of this solution. Note, for example, from
the reconstruction formula (22.38) that the value of x(t), at any time t, depends on samples
x(nTs ) that occur before and after time t since n runs over the entire range, < n < .
In this way, the transformation (22.38) is a non-causal operation and the value of x(t)
cannot be generated in real-time by relying solely on samples, x(nTs ), that occur prior
to time t; one needs to wait for future samples of x(nTs ) as well. Another reason for
the impracticality of the reconstruction procedure (22.38) is that the ideal low-pass filter
H(j) is not implementable because it corresponds to an unstable and noncausal filter.
620
CHAPTER 22
SAMPLING
sample at nTs
where s > 2B
x(t)
z(t)
a(t)
zs (t)
Anti-aliasing lter.
It limits the signal
bandwidth to [B, B]
FIGURE 22.22 The anti-aliasing filter a(t) is a low-pass filter whose purpose is to limit the
bandwidth of the original signal x(t) to the range [B, B] radians/second before sampling at the
rate of s > 2B radians/second.
Specifically, note that its impulse response function, h(t), which is given by (22.33), is not
absolutely integrable and is noncausal (since it is nonzero for t < 0).
For this reason, the sinc function h(t) in (22.33) is usually replaced by the so-called
zero-order-hold (ZOH) function:
hzoh (t) =
1,
0,
0 t Ts
otherwise
(22.39)
which corresponds to a rectangular pulse within the interval [0, Ts ] see Fig. 22.23.
hzoh (t)
Ts
FIGURE 22.23 Impulse response function of the zero-order-hold operation.
The Fourier transform of hzoh (t) can be obtained by noticing that hzoh (t) can be expressed in terms of a regular rectangular pulse of width Ts and shifted to the right by Ts /2,
namely,
!
t T2s
(22.40)
hzoh (t) =
Ts
Thus, in a manner similar to Example 22.2 we can find that
Hzoh (j) = ej
Ts
2
Ts
sin(Ts /2)
= ej 2 Ts sinc
/2
Ts
2
(22.41)
621
The magnitude and phase plots of Hzoh (j) are shown in Fig. 22.24 over the range
[20, 20] radians/second and for the case Ts = 1. Observe that the main lobe in the
magnitude response of Hzoh (j) extends between s and s . The dotted lines in the
top plot of Fig. 22.24 indicate the frequency response of the ideal low-pass filter (22.34)
with magnitude scaled to one; the response extends between 2s and 2s .
Magnitude plot
1
Ideal lowpass filter
|Hzoh(j)|
0.8
0.6
ZOH filter
0.4
0.2
20
15
10
5
0
5
(radians/second)
Phase plot
10
15
20
10
15
20
3.14
3
(j)
zoh
2
3
20
15
10
5
0
5
(radians/second)
FIGURE 22.24 Magnitude (top) and phase (bottom) responses of the ZOH filter Hzoh (j) in
(22.41) for Ts = 1 over the range [20, 20].
Applying the ZOH filter Hzoh (j) to the sampled signal xs (t) results in
Xzoh (j) = Xs (j)Hzoh (j)
where we are denoting the result of the ZOH operation by Xzoh (j). The above transformation replaces the ideal low-pass filtering operation represented by (22.36). The resulting
time-domain equivalent of (22.38) then becomes
xzoh (t)
= xs (t) hzoh (t)

!
X
=
x(nTs )(t nTs )
n=
n=
x(nTs )
(n+1)Ts
2
Ts
t T2s
Ts
!
(22.42)
The form of the sequence xzoh (t) is indicated in Fig. 22.25; it consists of rectangular pulses
s
for all integers n. It is seen
of amplitudes x(nTs ) and centered at the locations t = (n+1)T
2
that xzoh (t) provides a staircase approximation for the original signal x(t).
We therefore conclude that the zero-order-hold operation, while practical and implementable, it nevertheless does not lead to perfect reconstruction of the original signal x(t).
SECTION 22.5
NYQUIST
RATE
FOR
BASEBAND
SIGNALS
622
CHAPTER 22
SAMPLING
xs (t)
xzoh (t)
xzoh (t)
xs (t)
hzoh (t)
Ts
FIGURE 22.25 Convolving the sampled signal xs (t) with the zero-order-hold impulse function,
hzoh (t), results in a staircase approximation for the original signal x(t).
In order to get the signal xzoh (t) closer to the signal x(t), the ZOH operation is usually
followed by another filtering (compensation) operation denoted by Hcomp (j), say,
X (j) = Xs (j)Hzoh (j)Hcomp (j)
where the purpose of Hcomp (j) is to compensate for the non-ideal frequency characteristics of Hzoh (j) see Fig. 22.26.
xs (t)
Hcomp (j)
ZOH
x (t) x(t)
FIGURE 22.26
The zero-order-hold (ZOH) operation is followed by a compensation filter
Hcomp (j) in order to approximate the ideal reconstruction procedure that results in x(t).
Ideally, we would like the combination of Hzoh (j) and Hcomp (j) to approximate
the ideal low-pass filter, H(j), given by (22.34), within the range [ 2s , 2s ], namely,
Hzoh (j)Hcomp (j) Ts
Ts ,
0,
2s < <
otherwise
s
2
Using expression (22.41) for Hzoh (j) we conclude that the compensation filter should
approximate the following frequency response:
( T
s
Ts /2
ej 2 sin(T
,
2s < < 2s
s /2)
Hcomp (j)
0,
otherwise
Figure 22.27 displays the magnitude responses of the zero-order-hold filter, Hzoh (j),
and the desirable compensation filter, Hcomp (j), for the case Ts = 1. Since it is difficult
to implement compensation filters with sharp transitions at s /2, the filter is usually
replaced by an approximation with a wider transition band as illustrated in Fig. 22.28. The
effect of the wider transition band on the reconstruction process would be negligible when
the sampling frequency, s , is sufficiently larger than the Nyquist frequency, 2B, so that
the original signal, x(t), does not have relevant frequency components around s /2.
Magnitude plots
1.5
|Hzoh(j)|
and
|H comp(j)
Hcomp(j)
Hzoh(j)
0.5
s/2 s/2
0
20
15
10
5
0
5
(radians/second)
10
15
20
FIGURE 22.27 Magnitude responses of the zero-order-hold filter (solid line) and the desirable
compensation filter (dotted line) using Ts = 1.
Hcomp (j)
Hcomp (j)
2s
1
s
2
2s
s
2
FIGURE 22.28
Desired magnitude response of the compensation filter (left) and an
approximation for it with wider transition bands (right).
22.6 SAMPLING OF BANDPASS SIGNALS

When the original signal x(t) is not baseband, there are situations where the signal can be
fully recovered even when the sampling rate is below Nyquists condition as the ensuing
analysis demonstrates. We distinguish between real-valued and complex-valued bandpass
signals and provide illustrative examples.
623
SECTION 22.6
SAMPLING OF
BANDPASS
SIGNALS
624
CHAPTER 22
SAMPLING
22.6.1 Complex-valued Bandpass Signals

To begin with, consider a signal x(t) whose Fourier transform, X(j), is nonzero over
the frequency range [1 , 2 ], as shown in Fig. 22.29. We say that x(t) is a bandpass
signal and, actually, x(t) is a complex-valued bandpass signal. This is because if x(t)
were real-valued then from the property (22.16), the magnitude response of its Fourier
transform should be an even function of , which is not the case here. Now since the
highest frequency is 2 , the Nyquist condition (22.31) states that the signal x(t) can be
recovered from sampling it at any rate s satisfying s > 22 . However, since x(t) is
bandlimited to the range [1 , 2 ], we shall now verity that for such bandpass signals
it is possible to sample x(t) at a lower rate than dictated by Nyquists condition without
any loss in information. Specifically, we shall argue that s need only satisfy the condition
s > 2 1
(22.43)
where the difference 2 1 amounts to the total signal bandwidth.

To see that this is indeed the case, we start by introducing the frequencies
B =
2 1
2
o =
and
2 + 1
2
(22.44)
where B denotes half the width of the bandwidth of the signal, and o is the midpoint of
the same bandwidth.
X(j)
(rad/s)
1 +2
2
FIGURE 22.29 The Fourier transform of a complex-valued bandpass signal x(t) is assumed to
be nonzero over the range [1 , 2 ].
Let
y(t) = ejo t x(t)
(22.45)
That is, y(t) is obtained by multiplying x(t) by the complex exponential signal ejo t . It
is clear that y(t) and x(t) determine each other uniquely since given y(t) we can find x(t)
through the relation
x(t) = ejo t y(t)
(22.46)
Therefore, if we can show how to recover y(t) from the samples of x(t), then we would
also arrive at a procedure to recover x(t) from these same samples.
Using the modulation property (22.15) we find that the Fourier transform of y(t) is
related to the Fourier transform of x(t) as follows:
Y (j) = X(j( + o ))
which amounts to shifting X(j) to the left by o radians/second. Doing so results in the
Fourier transform shown in Fig. 22.30. Observe that y(t) is now a baseband signal and its
Fourier transform extends between B and B radians/second.
Y (j)
2
1 +
2
1 +2
2
(rad/s)
FIGURE 22.30 Fourier transform of the signal y(t) = ejo t x(t).
Assume the signal y(t) is sampled at some rate s radians/second. Then Nyquists
condition (22.31) for perfect reconstruction of y(t) from its samples, y(nTs ), requires that
we select s such that
s > 2B = 2 1
(22.47)
In this case, we can recover y(t) from its samples:
y(nTs ) = ejo nTs x(nTs )
(22.48)
through the reconstruction formula (22.38):

y(t) =
=
n=
n=
y(nTs ) sinc
(t nTs )
Ts
ej(1 +2 )nTs /2 x(nTs ) sinc
(t nTs )
Ts
Consequently, using relation (22.45) between x(t) and y(t) we conclude that, as long as
the sampling frequency satisfies:
s > 2 1
(complex-valued bandpass signals)
(22.49)
the signal x(t) can be recovered from its samples through the reconstruction procedure
x(t) =
ej(1 +2 )(tnTs )/2 x(nTs ) sinc
n=
(tnTs )
Ts
(22.50)
625
SECTION 22.6
SAMPLING OF
BANDPASS
SIGNALS
626
CHAPTER 22
SAMPLING
Example 22.14 (Sampling complex-valued bandpass signals)

The Fourier transform of a complex-valued bandpass signal x(t) is nonzero over the range 10KHz <
f < 12KHz. The highest frequency in x(t) is 12KHz and, therefore, according to Nyquists condition (22.31), the signal x(t) can be recovered from its samples, x(n), for any sampling rate, Fs , that
exceeds 24 KHz.
However, the result (22.49) suggests that for such bandpass signals we can sample x(t) at much
lower rates and still recover it fully from its samples. In this case we have B = 1KHz and, therefore,
the signal can be sampled at 2KHz without loss of information.
22.6.2 Real-valued Bandpass Signals

Consider now a real-valued signal x(t) whose Fourier transform, X(j), is nonzero over
the frequency range 1 < |f | < 2 , i.e., over [2 , 1 ] and [1 , 2 ] as shown in
Fig. 22.29. We say that x(t) is a real-valued bandpass signal. The fact that x(t) is realvalued implies, from the symmetry property (22.16), that the magnitude response |X(j)|
must be an even function of , while the phase response X(j) must be an odd function
of .
Now since the highest frequency is 2 , Nyquists condition (22.31) states that the signal
x(t) can be recovered from sampling it at any rate s satisfying s > 22 . However, we
shall argue that for real-valued bandpass signals of this kind it is possible at times to sample
the signal at lower rates than required by Nyquist condition without any loss in information.
To see that this is indeed the case, we start by introducing the frequencies
B = 2 1
and
o =
2 + 1
2
(22.51)
We distinguish between two cases depending on whether 2 is a multiple of B or not.
X(j)
1
2
1 +
2
1 +2
2
FIGURE 22.31 The Fourier transform of a real-valued bandpass signal x(t) is assumed to be
nonzero over the ranges [1 , 2 ] and [1 , 2 ].
627
Upper frequency 2 is a multiple of B

Assume the signal x(t) is sampled at some rate s radians/second to obtain
xs (t) =
SECTION 22.6
SAMPLING OF
BANDPASS
SIGNALS
x(nTs )(t nTs )
(22.52)
As we already know, the Fourier transform of xs (t) is obtained by repeating the Fourier
transform of x(t) periodically every s radians/second and scaling the result by 1/Ts .
When 2 is a multiple of B, then the interval between 2 and 2 will correspond to an
integer multiple of 2B. In this case, if we select the sampling frequency as
s = 2(2 1 ),
Ts =
2
s
(real-valued bandpass signals)
(22.53)
then when the Fourier transform of x(t) is repeated every s radians/second, the resulting Fourier transform will be filled with adjacent non-overlapping images as shown in
Fig. 22.32.
Xs (j)
1/Ts
1 o 2
FIGURE 22.32 The Fourier transform of the sampled signal xs (t) when 2 is a multiple of B
and the signal x(t) is sampled at 2 = 2B radians/second.
Since no aliasing occurs, the signal x(t) can be recovered from the sampled signal xs (t)
through bandpass filtering, i.e.,
X(j) = Xs (j)H(j)
where the ideal reconstruction filter H(j) is chosen as
Ts ,
1 < < 2
H(j) =
Ts ,
2 < < 1
0,
otherwise
(22.54)
Note that H(j) can be expressed in the form of two rectangular pulses of width B =
2 1 each and centered at o :

t + o
t o
+ Ts
(22.55)
H(j) = Ts
B
B
628
CHAPTER 22
SAMPLING
so that, by inverse transformation, the corresponding impulse response function is

t
t
h(t) = ejo t sinc
+ ejo t sinc
Ts
Ts

t
= 2sinc
cos(o t)
(22.56)
Ts
Consequently, the reconstruction formula is given by
x(t) =
x(nTs ) sinc
n=
(t nTs )
Ts
cos(o (t nTs ))
(22.57)
Example 22.15 (Sampling a real-valued bandpass signal)

Figure 22.33 displays the Fourier transform of a real-valued bandpass signal x(t). The frequency
content of the signal is nonzero over the range 9 || 12 radians/second. The highest frequency
in the signal is 2 = 12 radians/second and, therefore, according to Nyquists condition (22.31),
the signal x(t) can be recovered from its samples, x(n), for any sampling rate, s , that exceeds 24
radians/second. However, the current situation corresponds to B = 3 radians/second and the higher
frequency, 2 = 12, is a multiple of B. The result (22.53) then suggests that for such bandpass
signals we can sample x(t) at much lower rates and still recover it fully from its samples. In this
case, the signal can be sampled at s = 2B = 6 radians/second without loss of information.
The middle plot in the same figure shows the Fourier transform of Xs (j), which is obtained by
repeating the Fourier transform of x(t) at every s = 6 radians/second. It is seen that the successive
images do not overlap with each other. The dotted lines in the figure indicate the location of the
ideal bandpass reconstruction filter that would recover x(t) from xs (t). Note, however, that the
result (22.53) does not state that any s that is larger than 2(2 1 ) will do! Indeed, the last
plot in Figure 22.33 illustrates the Fourier transform that would result for xs (t) when the sampling
rate is s = 7 radians/second. In this case, aliasing occurs and it is not possible to recover x(t)
from xs (t). In other words, expression (22.53) simply provides the value of the smallest possible
sampling frequency.
Upper frequency 2 is not a multiple of B

Let us now consider the case when the upper frequency, 2 is not a multiple of B. In
this case, if we continue to sample the signal x(t) at the rate s = 2B, then aliasing in
frequency will occur, as illustrated in Fig. 22.34
In order to avoid aliasing in frequency, we can adjust the value of B to a new larger value
Ba such that 2 is a multiple of Ba . The smallest such Ba can be selected as follows. Let
2 = mB + r
where m is a positive integer and r is the remainder of dividing 2 by B, and choose
Ba = 2 /m
(22.58)
Obviously, Ba B and 2 is a multiple of Ba . This choice of Ba corresponds to assuming

that the bandwidth of x(t) lies within the expanded range 2 Ba < || < 2 see
Fig. 22.35.
629
SECTION 22.6
X(j)
SAMPLING OF
BANDPASS
SIGNALS
12
s = 6
12
12
12
Xs (j)
1/Ts
12
s = 7
Xs (j)
1/Ts
12
FIGURE 22.33 The Fourier transforms of a real-valued bandpass signal x(t) (top), and its
sampled versions, xs (t), for sampling frequencies s = 6 radians/second (middle) and s = 7
radians/second (bottom). The dotted lines in the middle plot illustrate the location of the ideal
reconstruction bandpass filter that allows recovering x(t) from xs (t). Observe how aliasing in
frequency occurs in the bottom plot.
Now, the same argument that was used in the case when 2 is a multiple of B will show
that if we sample x(t) at the rate
s = 2Ba ,
Ts =
2
s
(22.59)
then the signal x(t) be recovered from the sampled version xs (t) through bandpass filtering, namely,
X(j) = Xs (j)H(j)
where now
Let
Ts ,
H(j) =
T ,
s
0,
a =
2 Ba < < 2
2 < < 2 + Ba
otherwise
(22.60)
22 Ba
2
(22.61)
630
CHAPTER 22
SAMPLING
Xs (j)
1/Ts
1 o 2
FIGURE 22.34 The Fourier transform of the sampled signal xs (t) when 2 is not a multiple of
B and the signal x(t) is still sampled at 2 = 2B radians/second. We now find that aliasing occurs.
X(j)
1
2
2 Ba
2 + Ba
a
FIGURE 22.35 The Fourier transform of the real-valued bandpass signal x(t) is assumed to
extend over the ranges [2 , 2 + Ba ] and [2 Ba , 2 ].
and note that the ideal reconstruction filter, H(j), can be expressed in the form of two
rectangular pulses of width Ba and centered at a :
H(j) = Ts
t a
Ba
+ Ts
t + a
Ba
(22.62)
so that, by inverse transformation, the corresponding impulse response function is

h(t) =
=
ja t
t
Ts
+ e
sinc

t
2sinc
cos(a t)
Ts
ja t
sinc
t
Ts

(22.63)
Consequently, the reconstruction formula is given by

x(t) =
x(nTs ) sinc
n=
(t nTs )
Ts
cos(a (t nTs ))
(22.64)
631
SECTION 22.7
Example 22.16 (Sampling a real-valued bandpass signal)

Figure 22.36 displays the Fourier transform of a real-valued bandpass signal x(t). The frequency
content of the signal is nonzero over the range 7 || 11 radians/second. The highest frequency
in the signal is 2 = 11 radians/second and, therefore, according to Nyquists condition (22.31),
the signal x(t) can be recovered from its samples, x(n), for any sampling rate, s , that exceeds 22
radians/second. However, the current situation corresponds to B = 4 radians/second and the higher
frequency, 2 = 11, is not a multiple of B. Note though that
2 = 2 B + 3
so that m = 2. Select
Ba = 2 /m = 11/2 = 5.5
The result (22.53) then suggests that the signal can be sampled at s = 2Ba = 11 radians/second
without loss of information, which is much lower than the Nyquist rate.
The middle plot in the same figure shows the Fourier transform of Xs (j), which is obtained by
repeating the Fourier transform of x(t) at every s = 8 radians/second. It is seen that the successive
images overlap with each other. The bottom plot of the same figure shows the Fourier transform of
xs (t) when x(t) is sampled at s = 11 radians/second.The dotted lines in the figure indicate the
location of the ideal bandpass reconstruction filter that would recover x(t) from xs (t).
Example 22.17 (Sampling real-valued bandpass signals)

Consider a real-valued signal whose Fourier transform is nonzero over the range
10KHz < |f | < 12KHz
Then B = 2KHz and the higher frequency of 12KHz is a multiple of B. Therefore, this signal can
be sampled at the rate of Fs = 2B = 4KHz without loss of information.
Now assume instead that the Fourier transform of the signal is nonzero over the range
10KHz < |f | < 13KHz
Then B = 3KHz and the higher frequency of 13KHz is not a multiple of B. Note though that
13 = 4 B + 1
so that we can define
Ba = F2 /4 = 13/4 = 3.25KHz
Then the signal can be sampled at the rate of Fs = 2Ba = 6.5KHz without loss of information.
22.7 RELATION OF FOURIER TRANSFORM TO THE DTFT

The discussion in the previous section on the sample procedure, and the constructions that
are associated with it in the frequency domain, can be used to highlight useful relations between the Fourier transform (FT) of continuous-time signals and the discrete-time Fourier
transform (DTFT) of discrete-time signals.
Thus, recall that we started with a generic continuous-time signal x(t) with Fourier
transform, X(j), as was illustrated in Fig. 22.14. We subsequently sampled x(t) to
RELATION
OF FOURIER
TRANSFORM
TO THE DTFT
632
CHAPTER 22
X(j)
SAMPLING
12
12
12
12
Xs (j)
s = 8
1/Ts
12
s = 11
Xs (j)
1/Ts
12
FIGURE 22.36 The Fourier transforms of a real-valued bandpass signal x(t) (top), and its
sampled versions, xs (t), for sampling frequencies s = 8 radians/second (middle) and s = 11
radians/second (bottom). The dotted lines in the bottom plot illustrate the location of the ideal
reconstruction bandpass filter that allows recovering x(t) from xs (t). Observe how aliasing in
frequency occurs in the middle plot.
obtain
xs (t) =
n=
x(nTs )(t nTs )
(22.65)
and we argued that the Fourier transform of the sampled signal, xs (t), consists of the
Fourier transform of X(j) repeated periodically every s radians/second and scaled by
1/Ts , namely,
!
X
1
Xs (j) =
X(j jks )
(22.66)
Ts
k=
This construction was illustrated in Fig. 22.16.

There is an alternative expression for the Fourier transform of xs (t), which is useful for
the purposes of the discussion in this section. To arrive at this expression we start from
(22.65) and recall from Example 22.2 the Fourier transform pair
(t nTs ) ejnTs
(22.67)
Using this result in (22.65), and invoking the linearity property of the Fourier transform,
we get
X
(22.68)
Xs (j) =
x(nTs )ejnTs
n=
Let us write x(nTs ) simply as x(n), which has been our notation for the terms of a sequence all along, and let us also introduce a convenient change of variables as follows
= Ts
(frequency normalization)
(22.69)
This transformation maps the frequency variable to a normalized frequency variable .

Note in particular that the value
= s radians/second (sampling frequency)
is mapped to the normalized value
= s T s =
2
Ts = 2 radians/sample
Ts
(22.70)
That is,
= s (radians/second) = 2 (radians/sample)
(22.71)
Note further that the units for are radians/sample. Using the normalized frequency , we
find that expression (22.68) for Xs (j) becomes
Xs (j) =
x(n)ejn
n=
We readily recognize the expression on the right-hand side as the DTFT of the sequence
x(n), and which we denoted earlier by X(ej ). Therefore, we conclude that the following
relation holds:
Xs (j) = X(ej )
(22.72)

X(e ) = Xs j
Ts
(22.73)
or, equivalently,
j
Observe that both Xs (j) and X(ej ) are periodic functions: the former is periodic with
period s while the latter is periodic with period 2. Obviously, the values of both periods
are related through the normalization = Ts .
This discussion shows that the DTFT of a sequence x(n) is nothing but the original
Fourier transform X(j) repeated every s rad/s (and, hence, periodic) and scaled by
1/Ts . Moreover, since we normalize the frequency axis so that the sampling frequency
= s is mapped to = 2, then the DTFT becomes periodic with period 2: the
s -periodicity in the -domain is transformed into a 2-periodicity in the -domain see
633
SECTION 22.7
RELATION
OF FOURIER
TRANSFORM
TO THE DTFT
634
Xs (j)
CHAPTER 22
SAMPLING
1/Ts
(rad/sec)
(rad/sample)
X(ej )
1/Ts
BTs
BTs
FIGURE 22.37 The Fourier transform (FT) of the sampled signal, xs (t), and the discrete-time
Fourier transform (DTFT) of the corresponding sequence, x(n), coincide apart from a normalization
of the frequency variable through the relation = Ts , where the sampling frequency s is mapped
to 2.
Fig. 22.37. One useful consequence of using the normalized frequency variable, , is that
frequency plots now become independent of the value of the actual sampling rate, s .
Example 22.18 (Normalized frequency)
A discrete-time system processes data that has been sampled at the rate of 8 KHz. A tone at 60Hz in
the continuous-time domain would correspond to the following normalized frequency in the discretetime domain:
=
=
Ts
2 60/8000
3/200
0.0471 radians/sample
Example 22.19 (Discrete-time filtering)

The frequency response of a discrete-time filter is shown in Fig. 22.38. The filter operates at 16KHz.
A 3KHz tone is sampled and fed through the filter. We would like to determine the attenuation that
the tone will undergo as it is processed by the filter. To do so, we first map the frequency of the tone
to the normalized domain as follows:
=
=
=
Ts
2 3000/16000
3/8 radians/sample
Therefore, the frequency of the tone is viewed by the discrete-time system as = 3/8. From the
frequency response in Fig. 22.38 we find that this particular frequency is attenuated by 0.85.
H(ej )
0.85
/2
/2
(rad/sample)
3/8
FIGURE 22.38 The frequency response of a discrete-time filter operating at 16KHz.
22.8 RELATION OF FOURIER TRANSFORM TO THE DFT

We can also comment on the relation between the Fourier transform (FT) of a signal, x(t),
and the discrete Fourier transform (DFT) of the corresponding sequence, x(n). Thus, recall
that the N point DFT of x(n) is obtained by sampling its DTFT, X(ej ), at multiples of
2/N radians, i.e.,

X(k) = X(ej )= 2k
N
Using the transformation = Ts , and the correspondence between X(ej ) and Xs (j)
from (22.73), we conclude that the samples of the N point DFT correspond to sampling
Xs (j) in the domain at multiples of s /N rad/s, namely,
s
X(k) = Xs j k
N
(22.74)
Let us pursue this relation further in the time domain. Thus, let Xr (j) denote the
Fourier transform that results from sampling Xs (j) at multiples of s /N , namely,
!

X
s
(22.75)
Xr (j) = Xs (j)
k
N
k=
The above operation amounts to multiplying Xs (j) by a periodic train of impulses with
period s /N . We listed earlier in Example 22.5 the Fourier transform of a periodic train
of impulses, which in the current case translates into
!

X
X
s
s
,
s = 2/Ts
(t nN Ts )
k
N
N
n=
k=
Using the convolution property (22.20), which states that multiplication in the frequency
domain amounts to convolution in the time domain, we find that the inverse transform of
635
SECTION 22.8
RELATION
OF FOURIER
TRANSFORM
TO THE DFT
636
Xr (j) is the signal xr (t) given by
CHAPTER 22
SAMPLING
xr (t)
= xs (t)
=
N Ts
N
s
(t N Ts )
!
xs (t N Ts )
(22.76)
That is, the sampled signal xs (t) is repeated periodically every N Ts seconds and the results
are added and scaled by N Ts /2. Obviously, aliasing in time will occur (i.e., adjacent
repetitions of the signal xs (t) will interfere with each other) if the period N Ts is not
larger than the duration of the signal xs (t). Thus, we shall assume that the duration of the
original signal x(t) is reasonably within N Ts seconds. This construction is illustrated in
Fig. 22.39. Note that the samples of the periodic signal xr (t) are scaled versions of the
periodic sequence xp (n) that was constructed earlier in (17.19) while studying the DFT in
Sec. 17.2.
In order to determine the sequence xr (n) that corresponds to xr (t) we proceed in two
equivalent ways. Doing so will further clarify the relation between the Fourier transform
and the DFT in the time domain. To begin with, replacing with /Ts (and, accordingly,
s with 2) in (22.75), we obtain a periodic function in of period 2, i.e.,

!

= Xs j
Xr j
Ts
Ts
Ts
N
k=
!

X
2k
(22.77)

= T s Xs j
Ts
N
k=
where in the second equality we used the normalization property (at) = a1 (t). The
function Xr (j Ts ) so obtained is periodic with period 2 and it can be viewed as the DTFT
of some sequence, xr (n), to be determined. We shall denote this DTFT by the standard
notation Xr (ej ) and use (22.73) to rewrite the above equality in the equivalent form:
!

X
2k
j
j
(22.78)
Xr (e ) = Ts X(e )

N
k=
To arrive at xr (n), we now compute the inverse DTFT of Xr (ej ), and use the convolution
property (14.10) of the DTFT (which states that multiplication in the frequency domain
amounts to convolution in the time domain). Indeed, note first that the inverse DTFT of
the train of impulses in (22.78) is given by
!

Z 2
N 1
X
1
Ts X j 2kn
k2
(22.79)
ejn d =
e N
Ts

2 0
N
2
k=0
k=
where the sum on the right-hand side is over the interval 0 k N 1 because the
integration on the left-hand side is over the interval [0, 2]; therefore, the integration
extracts only the dirac impulses that are located within [0, 2]. Returning to (22.78) and
inverse-transforming both sides with the help of (22.79) we obtain
!
N 1
Ts X j 2kn
xr (n) = x(n)
(22.80)
e N
2
k=0
637
SECTION 22.8
RELATION
OF FOURIER
TRANSFORM
TO THE DFT
Xs (j)
1/Ts
2s
(rad/sec)
(rad/sec)
s
2
Xr (j)
1/Ts
B
s
N
xs (t)
1
t(s)
xr (t)
N Ts /2
N Ts
N Ts
t(s)
Ts
FIGURE 22.39 Sampling the Fourier transform Xs (j) every s /N radians/second results in
the sampled Fourier transform Xr (j) shown in the second row. The corresponding action in the
time domain is to repeat the sampled signal xs (t) every N Ts seconds and to scale the result by
N Ts /2.
Now recall that the sum of exponential sequences that appears in the above expression
reduces to the following:
N
1
X
ej
2kn
N
k=0
N,
0,
n = 0, N, 2N, . . .
otherwise
(22.81)
That is,
N
1
X
k=0
ej
2kn
N
= N
(n N )
(22.82)
638
CHAPTER 22
SAMPLING
in terms of shifted versions of the unit-sample sequence, (n). Substituting into (22.80)
we find that
P
xr (n) = Ns
(22.83)
x(n N )
=
This means that, apart from scaling by N/s , the sequence xr (n) is obtained by repeating the sequence x(n) periodically every N samples and adding the results. Obviously,
aliasing in time will occur (i.e., adjacent repetitions of the sequence x(n) will interfere
with each other) if the duration of x(n) is larger than N samples. We shall therefore assume that the duration of the original continuous-time signal, x(t), is reasonably within
N Ts seconds to avoid the possibility of aliasing when forming xr (n). Note further that the
sequence xr (n) in (22.83) is a scaled version of the periodic sequence, xp (n), which we
constructed earlier in (17.19) while studying the DFT of x(n) in Sec. 17.2, namely,
xp (n) =
x(n N )
(22.84)
Now, there is an alternative way to arrive at an expression for xr (n) from (22.78). We
rewrite (22.78) as
!
2k
X
2k
j
j N
(22.85)
Xr (e ) = Ts

X e
N
k=
where the term involving X(ej ) has been moved inside the summation. Subsequently,
the inverse DTFT of Xr (ej ) can also be found as follows:
xr (n)

2k
2k
j N

ejn d
X e
N
0
N 1
Ts X j 2k j 2kn
X e N e N
2
k=0
#
" N 1

2kn
ks
1 X
N
ej N
Xs j
s
N
N
1
= Ts
2
=
(22.86)
k=0
Comparing (22.86) with (22.83) and (22.84) we conclude that

xp (n) =
1
N
NP
1
k=0
j 2kn
s
Xs j k
e N
N
(22.87)
which again confirms that the coefficients of the N point DFT of x(n) are given by
s
Xs (j k
N ).
22.9 SPECTRAL RESOLUTION

The discussion on the relation between the Fourier transform of x(t) and the N point
DFT of the corresponding sequence x(n) allows us to make the identification

ks
X(k) = Xs j
N
(22.88)
We therefore conclude that the separation between two DFT coefficients, which is equal to
2/N in the normalized frequency domain (see Sec. 21.3), translates into a separation of
s
radians/second
N
(22.89)
in the regular frequency domain. For this reason, we say that the frequency (or spectral)
resolution that is provided by an N DFT is given by
frequency resolution in radians/second =
s
2
=
N
N Ts
(22.90)
In Hertz, we obtain
frequency resolution in Hertz =
Fs
1
=
N Ts
N
(22.91)
where
N Ts duration in seconds of data segment used by the DFT
Therefore, the longer the duration of a signal segment in time the better the resolution in
frequency that is provided by the DFT. Observe that the resolution is not a function of N
alone but rather of the product N Ts .
Example 22.20 (Zero padding)

A speech signal is bandlimited to 4KHz. We can therefore sample it at 10KHz and satisfy Nyquists
condition. We would like to determine the amount of samples that are needed and the size of the
DFT to use in order to achieve spectral resolution of 10Hz.
First note that the duration of the segment of speech that we need to collect is inversely related to
the desired spectral resolution and therefore,
duration of speech segment needed =
1
= 0.1 second
10
Since the sampling rate is Fs = 10KHz (i.e., 10000 samples per second), a segment of duration 0.1
second would correspond to
N = 10000 0.1 = 1000 samples
The closet power-of-2 for N is to select N = 1024 and to compute a 1024point DFT.
What if we are unable to collect 0.1 second of speech data but, say, only 0.05 second? This
duration would result in only 500 samples available and we need 1000 samples to attain the desired
frequency resolution. We can take the available 500 data points and pad them with zeros to increase
the length of the data up to N = 1024 points. Doing so improves the spectral resolution.
This step can be justified as follows. Assume we return to the construction illustrated in Fig. 22.39
and reduce the separation s /N between two successive samples of Xr (j) in the second row of
the figure (for example, by increasing the value of N to N > N but keeping s fixed); in this way,
the frequency resolution s /N is enhanced. Then the effect on xr (t) in the last row of the figure is
to separate the images further apart from each other and to add additional zero samples in between
639
SECTION 22.10
SPECTRAL
RESOLUTION
640
xr (t)
CHAPTER 22
SAMPLING
N Ts /2
t(s)
N Ts
Ts
xr (t)
N Ts /2
N Ts
t(s)
Ts
FIGURE 22.40 Sampling the Fourier transform Xs (j) in Fig. 22.39 at a finer scale every
s /N radians/second (where N > N ) results in repeating xs (t) every N Ts ; the images in the
time domain are further apart repeating every N Ts seconds (bottom) rather than every N Ts seconds
where N < N (top).
see Fig. 22.40. In other words, the data segment that is used by the DFT of order N will have
additional zeros padded to its end relative to the data segment that is used by the DFT of order N .
22.10 APPLICATIONS
TO BE ADDED
Practice Questions:
1.
2.
22.11 PROBLEMS
Problem 22.1 Find the Fourier transform of the following signals

(a) x(t) = cos(o t)

(b) x(t) = cos(o t) sinc
t
Ts
t
Ts
(c) x(t) = cos(o (t 1))
.

t2
Ts
(d) x(t) =
(e) x(t) =
P

t
(f) x(t) =
2t
Ts
n=
641
(t nTs ).
SECTION 22.11
(t nTs ).
n=
PROBLEMS
Problem 22.2 Find the Fourier transform of the following signals

(a) x(t) = sin(o t)
t1
Ts
(b) x(t) = cos2 (o t) sinc
.
t
Ts
(d) x(t) =
(e) x(t) =
t1
(f) x(t) =
P
n=
3(t2)
Ts
(c) x(t) = sin(o (t + 1))

t2
n=
t2
Ts
(t nTs ).
(t nTs ).
Problem 22.3 Find the energies of the signals in Prob. 22.1.

Problem 22.4 Find the energies of the signals in Prob. 22.2.
Problem 22.5 Compute the convolution

t
2
t1
4
(a) Using the definition of linear convolution.

(b) Using properties of the Fourier transform.
Problem 22.6 Compute the convolution

cos(2t)
t
2
t+1
4
(a) Using the definition of linear convolution.

(b) Using properties of the Fourier transform.
Problem 22.7 Figure 22.41 shows the Fourier transform of an input signal, x(t), and the frequency
response of an LTI filter, H(j). Determine the output signal y(t) and its energy. Is the filter stable?
Problem 22.8 Figure 22.42 shows the Fourier transform of an input signal, x(t), and the frequency
response of an LTI filter, H(j). Determine the output signal y(t) and its energy. Is the filter stable?
Problem 22.9 Consider the Fourier transform shown in Fig. 22.43.
(a) What is Nyquists rate for this signal?
(b) Set s = 10 radians/sec. Draw Xs (j).
(c) Set s = 6 radians/sec. Draw Xs (j). Does aliasing occur?
(d) What is the energy of x(t)?
(e) What is the energy of the sequences x(n) from parts (b) and (c)?
(a) What is Nyquists rate for this signal?
(b) Set s = 6 radians/sec. Draw Xs (j).
(c) Set s = 3 radians/sec. Draw Xs (j). Does aliasing occur?
642
CHAPTER 22
X(j)
SAMPLING
(radians/sec)
H(j)
1
(radians/sec)
B
2
B2
FIGURE 22.41 Fourier transform of the input signal (top plot) and the frequency response of an
LTI system (bottom plot) for Prob. 22.7.
X(j)
1
(radians/sec)
H(j)
1
B B2
B
2
(radians/sec)
FIGURE 22.42 Fourier transform of the input signal (top plot) and the frequency response of an
LTI system (bottom plot) for Prob. 22.8.
(d) What is the energy of x(t)?

(e) What is the energy of the sequences x(n) from parts (b) and (c)?
(a) What is the smallest sampling frequency s for no aliasing to occur?
(b) What is the energy of x(t)?
(c) Set s = 3 radians/sec. Draw Xs (j).
(d) For part (c), what is the energy of the resulting sequence, x(n).
(e) Set s = 1 radians/sec. Draw Xs (j).
643
SECTION 22.11
PROBLEMS
X(j)
1
(radians/sec)
FIGURE 22.43 Fourier transform for Prob. 22.9.
X(j)
1
(radians/sec)
X(j)
2
1
4 5 6
(radians/sec)
(f) For part (e), what is the energy of the resulting sequence x(n).
Problem 22.12 Repeat Prob. 22.11 for Fig. 22.46.
Problem 22.13 Repeat Prob. 22.11 for Fig. 22.47.
Problem 22.14 Repeat Prob. 22.11 for the transform X (j) = ej/2 X(j).
Problem 22.15 Consider the Fourier transform shown in Fig. 22.48. Select s = 6 radians/sec.
644
CHAPTER 22
X(j)
SAMPLING
1
4 5 6
6 5 4
(radians/sec)
X(j)
2
1
7
5 4
4 5
(radians/sec)
(a) Draw Xs (j).

(b) Draw X(ej ).
(c) Draw Xr (ej ).
X(j)
2
2 1
1 2
(radians/sec)
Problem 22.16 Consider the Fourier transform shown in Fig. 22.49. Select s = 8 radians/sec.
(a) Draw Xs (j).
(b) Draw X(ej ).
(c) Draw Xr (ej ).
Problem 22.17 A continuous-time signal is sampled at the rate of 1GHz. Each sample is quantized
to 16 bits. How many bytes of memory are necessary to record one hour of the signal?
645
SECTION 22.11
PROBLEMS
X(j)
2
(radians/sec)
Problem 22.18 A CD audio signal is band-limited to about 22KHz. If the signal is sampled at
twice the Nyquist rate, and if each sample is quantized to 12 bits, determine the resulting bit rate
measured in bits per second. How much memory, measured in mega bits, would you need to store
one minute of a CD audio signal at this sampling rate?
Problem 22.19 Establish Parsevals relation for continuous-time signals,
|x(t)|2 dt =
R
1
|X(j)|2 d
2
Problem 22.20 Establish the Fourier transform property

dx(t)
dt
jX(j)
Problem 22.21 Establish the Fourier transform property

x(2t)
1
X
2
j
2
Problem 22.22 The Fourier transform of a signal x(t) is zero for || > B radians/second. The
signal is sampled at the rate of s > 2B. Let Ed denote the energy of the resulting discrete-time
sequence,
Ed =
n=
|x(n)|2
Likewise, let Ec denote the energy of the continuous-time signal,

Z
Ec =
|x(t)|2 dt
Show that Ec = Ts Ed where Ts = 2/s .

Problem 22.23 Find the frequency response of the LTI system described by the differential equation
y(t) + 3y(t)
+ 2y(t) = x(t 1)
where y and y denote the second and first-order derivatives of y(t). Find also the impulse response
of the system.
Problem 22.24 Find the frequency response of the LTI system described by the differential equation
3
1
y(t) + y(t)
+ y(t) = cos(2t) x(t)
2
2
646
Find its impulse response as well.
CHAPTER 22
SAMPLING
Problem 22.25 What is the Nyquist rate for the signal

x(t) = cos(2F1 t) + sin(2F2 t)
where F1 = 100 Hz and F2 = 125 Hz.
Problem 22.26 The maximum frequency that is present in a baseband signal x(t) is 500 Hz. What
is the minimum sampling frequency that can be used to avoid aliasing? The sampled signal is to be
processed by a discrete-time filter to remove the frequency content in the range 50 70Hz. What is
the impulse response sequence of the ideal discrete-time filter that achieves this task?
Problem 22.27 A continuous-time speech signal x(t) is contaminated with a unit amplitude interfering tone at 3000Hz. The speech signal, which is assumed to be band-limited at 4 KHz, is sampled
at the rate of 12000 samples per second. The resulting discrete-time sequence is then processed by
the LTI system:
1
y(n) = y(n 1) + x(n 1)
2
(a) By how much is the 3000Hz interference attenuated at the output of the discrete-time filter?
Find the steady-state response of the filter to the unit amplitude tone at 3000Hz.
(b) Consider a first-order LTI system of the form
y(n) = ay(n 1) + x(n)
for some value of a. Design a such that the amplitude of the 3000Hz interference is reduced
by at least 25%.
Problem 22.28 A causal
system is composed of the series cascade of two LTI subsystems
with
n
n
impulse responses 12 u(n) and 13 u(n 1). A tone at 1KHz is attenuated by 2/ 21 (i.e., by
approximately 43.6). Can you tell what the sampling rate is?
Problem 22.29 Let o = 2 radians/second and o = 13 radians/second. Consider the signal

x(t) = sinc
o t
2
cos o t
Find the smallest frequency at which x(t) can be sampled without loss of information.
Problem 22.30 Assume the Nyquist rate of a signal x(t) is s radians/second. What is the Nyquist
rate of the following transformations of the signal:
(a) x(t).
(b) x(2t).
(c) x(t/2).
(d) x2 (t).
Problem 22.31 Assume the Nyquist rate of a signal x(t) is s radians/second. What is the Nyquist
rate of the following transformations of the signal:
(a) dx(t)/dt.
(b) ejo t x(t).
(c) x(t) x(t).
(d) cos(o t) x(t).
Problem 22.32 The DTFT of a sequence x(n) is shown in Fig. 22.50. If x(n) was obtained by
sampling a signal x(t) at 20KHz, can you determine the bandwidth of x(t)?
647
X(ej )
SECTION 22.11
PROBLEMS
2 4
(rad/sample)
FIGURE 22.50 DTFT of the sequence x(n)for Prob. 22.32.
Problem 22.33 Consider a discrete-time processor operating at the rate of 1GHz and assume that
each complex operation (addition or multiplication) requires one clock cycle. It is desired to sample
a continuous-time signal, x(t), and to process the resulting sequence, x(n), by an FIR filter whose
impulse response sequence has L nonzero coefficients. What is the maximum bandwidth that the
signal x(t) can have in terms of L and the DSP clock rate? Assume L = 124 and find the resulting
numerical value.

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DISCRETE-TIME

PROCESSING AND FILTERING

Electrical Engineering Department

The notes are complemented by an interactive website at http://www.ee.ucla.edu/dsplab

Please email typos and suggestions to sayed@ee.ucla.edu.

Convolution Sum for LTI Systems

Impulse Response Sequence

Properties of the Convolution Sum

Homogeneous Difference Equations

Solving Difference Equations

Transfer Functions of LTI Systems

Initial Value Theorem

z-Transform and Difference Equations

Discrete-Time Fourier Transform

Properties of the DTFT

Definition of the DTFT

Periodicity of the DTFT

Frequency Content of a Sequence

Minimum and Linear Phase Systems

Discrete Fourier Transform

Properties of the DFT

16.3.4 Type-IV FIR Filters

Periodicity of the DFT

Computing Linear Convolutions

Relating Linear and Circular Convolutions

Fast Fourier Transform

Realizations of FIR Filters

Design of FIR Filters

Design of Analog Filters

Practical Filter Specifications

Design of IIR Filters

Relating Laplace and z-Transforms

Analysis Filter Bank

Sampling Rate Conversion

Probability Density Functions

Mean and Variance

Estimation Without Observations

Wiener Smoothing Problem

Example 1.1 (Moving cart)

Example 1.2 (Two-dimensional image)

Example 1.3 (Students in a course)

Example 1.4 (Satellite orbiting the Earth)

1.2 CLASSIFICATION OF SIGNALS

(used for nonnegative amplitudes)

100, 101, 110, 111

(used for negative amplitudes)

x(n) falls within the interval [

Example 1.5 (3-bit quantization)

lies within the interval

lies within the interval

lies within the interval

x(n) = x(t)|t=nT = x(nT )

1.5 SIGNAL PROCESSING

A system processes an input signal and transforms it into an output signal.

1.7 DSP TECHNOLOGY

1.8.1 Voiced and Unvoiced Speech

FIGURE 1.10 A representation of the vocal tract.

The classification of speech into voiced and unvoiced segments is accomplished by

Likewise, the frame of index k = 1 contains the samples

and, more generally, the frame of index k contains the samples

threshold = voiced frame

< threshold = unvoiced frame

1.8.2 Estimation of DC Levels

where we are labeling the students from n = 0 up to n = N 1. The measurements r(n)