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Deergha Rao
1. Introduction
K. Deergha Rao1
(1) Department of Electronics and Communication Engineering, Vasavi College of
Engineering (Affiliated to Osmania University), Hyderabad, Telangana, India
1.1 What is a Signal?
A signal is defined as any physical quantity that carries information and varies with time,
space, or any other independent variable or variables. The world of science and
engineering is filled with signals: speech, television, images from remote space probes,
voltages generated by the heart and brain, radar and sonar echoes, seismic vibrations,
signals from GPS satellites, signals from human genes, and countless other applications.
(1.1)
where ℜ is an operator.
Example 1.1 Consider the following signals x(t) and x i (t), i = 1,2,3. Express them using
only x(t) and its time-shifted, time-scaled, and time-inverted version.
Solution
Discrete-time signals are defined only at discrete times, and for these signals, the
independent variable n takes on only a discrete set of amplitude values as shown in Figure
1.5(b).
A signal that does not satisfy the above periodicity property is called an aperiodic
signal. The signal shown in Figure 1.6(b) is an example of an aperiodic signal.
Example 1.2 For each of the following signals, determine whether it is periodic or
aperiodic. If periodic, find the period.
(i) x(t) = 5 sin(2πt)
(iv)
(v)
Solution
(ii) It is periodic,
(iii) It is aperiodic,
(iv) It is periodic.
Example 1.3 If a continuous-time signal x(t) is periodic, for each of the following
signals, determine whether it is periodic or aperiodic. If periodic, find the period.
As x 1(t) is compressed version of x(t) by half, the period of x 1(t) is also compressed
by half.
Since x 2(t) = x 2(t + 2T ), x 2(t) is periodic with fundamental period 2T. As x 2(t) is
expanded version of x(t) by two, the period of x 2(t) is also twice the period of x(t).
Proposition 1.1 Let continuous-time signals x 1(t) and x 2(t) be periodic signals with
fundamental periods T 1 and T 2, respectively. The signal x(t) that is a linear combination
of x 1(t) and x 2(t) is periodic if and only if there exist integers m and k such that mT 1 = kT
2 and
(1.2)
The fundamental period of x(t) is given by mT 1 = kT 2 provided that the values of m and k
are chosen such that the greatest common divisor (gcd) between m and k is 1.
Example 1.4 For each of the following signals, determine whether it is periodic or
aperiodic. If periodic, find the period.
Determine whether x 3(t) = x 1(t)x 2(t) is periodic or aperiodic. If periodic, find the period.
Solution Decomposing signals x 1(t) and x 2(t) into sums of exponentials gives
Then,
It is seen that all complex exponentials are powers of e j(π/5). Hence, it is periodic.
Period is
Any signal can be expressed as sum of its even and odd parts as
(1.3)
(1.3a)
(1.3b)
(1.4)
(1.6)
(1.7)
Eqs. (1.6) and (1.7) are valid for no impulse or its derivative at the origin. These
properties are proved to be useful in many applications.
Example 1.6 Find the even and odd parts of x(t) = e j2t.
Solution From Eq. (1.3),
where
Example 1.7 If x e (t) and x o (t) are the even and odd parts of x(t), show that
(1.8)
Solution
Since 2
Example 1.8 For each of the following signals, determine whether it is even, odd, or
neither (Figure 1.8)
Solution By definition a signal is even if and only if x(t) = x(−t), while a signal is odd if
and only if x(t) = −x(−t).
(a) It is readily seen that x(t) ≠ x(−t) for all t and x(t) ≠ −x(−t) for all t; thus x(t) is
neither even nor odd.
(b) Since x(t) is symmetric about t = 0, x(t) is even.
Figure 1.9 (a) Causal signal, (b) noncausal signal, (c) anticausal signal
Example 1.9 Consider the following noncausal continuous-time signal. Obtain its
realization as causal signal.
Solution
(1.9)
(1.10a)
(1.10b)
Since the power is the time average of energy, the average power is defined as
. The signal x(t) expressed by Eq. (1.11), which is shown in
Figure 1.10(a), is an example of energy signal.
(1.11)
The signal x(t) shown in Figure 1.10(b) is an example of a power signal. The signal is
periodic with period 2. Hence, averaging x 2(t) over infinitely large time interval is the
same as averaging over one period, i.e., 2. Thus, the average power P is
Figure 1.10(b) Power signal
Thus, an energy signal has finite energy and zero average power, whereas a power
signal has finite power and infinite energy.
Example 1.10 Compute energy and power for the following signals, and determine
whether each signal is energy signal, power signal, or neither.
(iii)
(vi)
Solution (i)
The energy of the signal is infinite, and its average power is finite; x(t) is a power
signal.
The energy of the signal is finite, and its average power is zero; x(t) is an energy
signal.
(iii)
Using L’Hospital’s rule, we see that the power of the signal is zero. That is
The energy of the signal is infinite and its average power is zero; x(t) is neither energy
signal nor power signal.
Using L’Hospital’s rule, we see that the power of the signal is infinite. That is,
The energy of the signal is infinite and its average power is infinite; x(t) is neither
energy signal nor power signal.
The energy of the signal is infinite and its average power is finite; x(t) is a power
signal.
(vi)
.
The energy of the signal is infinite and its average power is finite; x(t) is a power
signal.
Example 1.11 Consider the following signals, and determine the energy of each signal
shown in Figure 1.11. How does the energy change when transforming a signal by time
reversing, sign change, time shifting, or doubling it?
Figure 1.11 Signals of example 1.11
Solution
The time reversal, sign change, and time shifting do not affect the signal energy.
Doubling the signal quadruples its energy. Similarly, it can be shown that the energy of k
x(t) is k 2 E x .
Proposition 1.2 The sum of two sinusoids of different frequencies is the sum of the
power of individual sinusoids regardless of phase.
Proof Let us consider a sinusoidal signal x(t) = Acos(Ωt + θ). The power of x(t) is given
by
(1.12)
Thus, a sinusoid signal of amplitude A has a power regardless of the values of its
frequency Ω and phase θ.
Now, consider the following two sinusoidal signals:
The first and second integrals on the right-hand side are the powers of the two
sinusoidal signals, respectively, and the third integral becomes zero since
Hence,
(1.13)
It can be easily extended to sum of any number of sinusoids with distinct frequencies
(1.14)
(1.15a)
(1.15b)
The delta function shown in Figure 1.14(b) can be evolved as the limit of the
rectangular pulse as shown in Figure 1.14(a).
(1.16)
As the width Δ → 0, the rectangular function converges to the impulse function δ(t)
with an infinite height at t = 0, and the total area remains constant at one.
Some Special Properties of the Impulse Function
Sampling property
If an arbitrary signal x(t) is multiplied by a shifted impulse function, the product is
given by
(1.17a)
(1.17b)
Shifting property
(1.18)
Scaling property
(1.19)
The unit impulse function can be obtained by taking the derivative of the unit step
function as follows:
(1.20)
The unit step function is obtained by integrating the unit impulse function as follows:
(1.21)
(1.22a)
(1.22b)
(1.23)
(1.24)
(1.25)
where both A and σ are real. If σ is positive, x(t) is a growing exponential signal. The
signal x(t) is exponentially decaying for negative σ. For σ = 0, the signal x(t) is equal to a
constant. Exponentially decaying signal and exponentially growing signal are shown in
Figure 1.18(a) and (b), respectively.
(1.26)
Hence
(1.26a)
(1.27)
(1.28)
Real sine function and real cosine function can be expressed by the trigonometric
identities as
Example 1.12 State whether the following signals are causal, anticausal, or noncausal.
(b) x(t) = tu(t) − t(u(t − 1) + e (1−t) u(t − 1))
Example 1.13 Determine and plot the even and odd components of the following
continuous-time signal
Solution
Example 1.14 Simplify the following expressions:
(a)
(b)
(c)
Solution
(a)
(b)
(c)
Figure 1.20 Exponentially damped sinusoidal signal with exponential parameter a = 0.4.
Example 1.17 Generate the following rectangular pulse function rect(t) using
MATLAB:
Solution The following MATLAB program generates the rectangular pulse function as
shown in Figure 1.22.
Figure 1.22 Rectangular pulse function
1.7.2 Filtering
Filtering is basically a frequency domain operation. Filter is used to pass certain band of
frequency components without any distortion and to block other frequency components.
The range of frequencies that is allowed to pass through the filter is called the passband,
and the range of frequencies that is blocked by the filter is called the stopband. A low-pass
filter passes all low-frequency components below a certain specified frequency Ω c , called
the cutoff frequency, and blocks all high-frequency components above Ω c . A high-pass
filter passes all high-frequency components above a certain cutoff frequency Ω c and
blocks all low-frequency components below Ω c . A band-pass filter passes all frequency
components between two cutoff frequencies Ωc1 and Ωc2 where Ωc1 < Ωc2 and blocks all
frequency components below the frequency Ωc1 and above the frequency Ωc2. A band-stop
filter blocks all frequency components between two cutoff frequencies Ωc1 and Ωc2 where
Ωc1 < Ωc2 and passes all frequency components below the frequency Ωc1 and above the
frequency Ωc2. Notch filter is a narrow band-stop filter used to suppress a particular
frequency, called the notch frequency.
1.7.4 Transformation
The transformation is the representation of signals in the frequency domain, and inverse
transform converts the signals from the frequency domain back to the time domain. The
transformation provides the spectrum analysis of a signal. From the knowledge of the
spectrum of a signal, the bandwidth required to transmit the signal can be determined. The
transform domain representations provide additional insight into the behavior of the signal
and make it easy to design and implement algorithms, such as those for filtering,
convolution, and correlation.
Figure 1.23 (a) Audio recording system, (b) the recorded voice signal “don’t fail me again”
The correlator in the user GPS receiver determines which codes are being received, as
well as their exact timing. When the received and receiver-generated sequences are in
phase, the correlator supplies the time delay. Now, the range can be obtained by
multiplying the time delay by the velocity of light. For example, assuming the time delay
as 3 ms (equivalent to 3 blocks of the C/A code of satellite 12), the correlation of satellite
12 producing a peak after 3 ms [Rao06] is shown in Figure 1.25.
Figure 1.26 (a) Schematic block diagram (b) tracking a mobile user of location-based mobile emergency service system
The processing of ECG signal yields information, such as amplitude and timing,
required for a physician to analyze a patient’s heart condition. Detection of R-peaks and
computation of R-R interval of an ECG record are important requirements of
comprehensive arrhythmia analysis systems. Heart rate is computed as
.
Figure 1.30 (a) MRI imaging system, (b) MRI image with brain tumor
1.9 Problems
(i)
(ii)
(iii)
(iv)
(vi)
(vii) x(t) = cos(2πt)u(t)
2. A periodic signal x 1(t) has a period 2, and another periodic signal x 2(t) has a period
3. Find the fundamental frequency and period for the signal y(t) = x 1(t) + x 2(t).
3. Classify the following continuous-time signals as even or odd signals or neither even
nor odd. Determine power and energy for each incase of power or energy signal.
6. Determine energy for each of the following signals and comment on the results.
7. What is the energy of the signal x(t) = cx(at − b), where a ≠ 0?
8. Verify that e −ct is neither energy nor a power signal for a complex value of c with
nonzero real part.
9. Show that the energy of x(t) ± y(t) is E x + E y , if x(t) and y(t) are orthogonal.
10. Derive an expression for the power of the following continuous-time signal x(t) = A 1
cos(Ω1 t + θ 1) + A 2 cos(Ω2 t + θ 2) for Ω1 = Ω2.
13. Find odd and even components for each of the following signals:
(a) x(t) = u(t)
(ii)
(iii)
(iv)
(v)
(vi)
1.10 MATLAB Exercises
2. Generate and plot each of the following continuous-time signals using MATLAB:
(i) x(t) = 10 sin(2πt) cos (πt − 4) for −10 ≤ t ≤ 10
Further Reading
1. Pierce, J.R., Noll, A.M.: Signals: The Science of Telecommunications. American Library, New Delhi (1960)
2. Lathi, B.P.: Linear Systems and Signals, 2nd edn. Oxford University Press, New York (2005)
3. Mandal, M., Asif, A.: Continuous and Discrete Time Signals and Systems. Cambridge University Press, Cambridge
(2007)
© Springer International Publishing AG, part of Springer Nature 2018
K. Deergha Rao, Signals and Systems, https://doi.org/10.1007/978-3-319-68675-2_2
This chapter presents time-domain analysis of continuous-time systems. It develops
representation of signals in terms of impulses . The notions of linearity , time-invariance,
causality , stability, memorability, and invertibility are introduced. It has shown that the
input-output relationship for linear time-invariant (LTI) continuous systems is described in
terms of a convolution integral. The differential equation representation of LTI continuous
systems and classical solutions of differential equations are also presented. Next, block-
diagram representation of LTI continuous-time systems is introduced. Furthermore, a brief
discussion on singularity functions is provided. Finally, the state-space representation of
continuous-time LTI systems is described.
(2.1)
(2.3)
Also, as Δ → 0, the summation approaches an integral, and the pulse approaches unit
impulse. Therefore, Eq. (2.3) can be rewritten as
(2.4)
(2.5a)
(2.5b)
Principle of homogeneity:
(2.6a)
(2.6b)
Principle of superposition:
(2.7)
Linearity :
(2.8)
(2.9)
Example 2.1 Check for linearity and time-invariance of the following system
Solution Linearity:
Let x 1(t) and x 2(t) be two distinct inputs applied to the system, then
For example, for an input x 1(t) = x(t − t 0), the output y1(t) can be written as
Solution Linearity:
Let x 1(t) and x 2(t) be two distinct inputs applied to the system, then
If an input equal to sum of the inputs ax 1(t), bx 2(t),x(t) = ax 1(t) + bx 2(t) is applied,
then
(i)
(iii)
(iv)
Solution
(i) Let x 1(t) and x 2(t) be two distinct inputs applied to the system, then
(ii)
Let x 1(t) and x 2(t) be two distinct inputs applied to the system, then
If an input equal to sum of the inputs ax 1(t), bx 2(t),x(t) = ax 1(t) + bx 2(t) is applied,
then
(iii) Let x 1(t) and x 2(t) be two distinct inputs applied to the system, then
(iv) Let x 1(t) and x 2(t) be two distinct inputs applied to the system, then
If an input equal to sum of the inputs ax 1(t), bx 2(t),
x(t) = ax 1(t) + bx 2(t) is applied, then
(ii)
Solution
For example, for an input x 1(t) = x(t − t 0), the output y 1(t) can be written as
For example, for an input x 1(t) = x(t − t 0), the output y 1(t) can be written as
For example, for an input x 1(t) = x(t − t 0), the output y 1(t) can be written as
For example, for an input x 1(t) = x(t − t 0), the output y 1(t) can be written as
Determine and sketch the response of the system to the following inputs:
(2.10)
If the input to the system is a unit step input u(t), then the system output is called the
step response s(t);that is,
(2.11)
(2.12)
From the linearity property of the system, Eq. (2.12) can be rewritten as
(2.13)
(2.14)
Thus, the output y(t) of a linear time-invariant system to an arbitrary input x(t) is
obtained in terms of the unit impulse input δ(t). Eq. (2.14) is referred to as the
convolutional integral and is denoted by the symbol * as
(2.15)
(2.16)
(2.17)
Then
(2.18)
(2.19)
(2.20)
(2.22)
(2.23)
(2.24)
(2.25)
(2.26)
(2.27)
(2.28)
(2.29)
The above right-hand side and left-hand side integrals are the same except for change
of the variables. Hence, the associative property is proved.
(2.30)
Proof By definition
(2.31)
(2.32)
Hence
(2.34)
and
(2.35)
(2.36)
Example 2.6 Determine the continuous-time convolution of x(t) and h(t) for the
following:
(v)
(i)
(ii)
Since u(t)*u(t) = tu(t),
(iii)
(iv)
Letting τ 1 = τ – 2,
(v)
Hence, y(t) is a shifted version of x(t) as shown in Figure 2.6.
(vii)
Example 2.7 Consider LTI system with the impulse response h(t); for an input x(t), the
output y(t) is as shown in Figure 2.7.
The output y 1(t) for the input x 1(t) = x(t) − x(t − 2) is given by
y 1(t) = y(t) − y(t − 2), which is shown in Figure 2.9.
Figure 2.9 Response y 1(t) to the input x 1(t)
Example 2.8 Consider a LTI system with input and output related through the equation
(ii) Determine the output y(t) of the system for the input
x(t) = u(t + 1) − u(t − 3).
Solution
(i)
Let τ 1 = τ − 3, then
Hence,
h(t) = e −(t − 3) u(t − 3)
(ii)
x(t − τ) and h(τ) are shown in Figure 2.10.
Figure 2.10 x(t − τ) and h(τ)
(i)
Solution
To compute y(t) = x(t) * h(t), first h(−τ) is to be obtained by inverting h(τ) about the
vertical axis. Then, the product of x(τ) and h(t − τ) is formed, point by point, and this
product is integrated to compute y(t). Thus, the overlap area between the rectangles
forming x(τ) and h(t − τ) is y(t).
Clearly, y(0) = 0 because there is no overlap between the rectangles forming x(τ) and
h(t − τ) at t = 0. For 0 < t < 8, there is overlap between the rectangles forming x(τ) and
h(t − τ). For t ≥ 8, there is no overlap, and hence, y(8) = 0. These are illustrated in Figure
2.11 with the final result for y(t). The shaded portion represents the overlap area of the
product x(τ) and h(t − τ).
Figure 2.11 Steps in the convolution and the final result
Example 2.10 Determine graphically y(t) = x(t) * h(t) for the following x(t) and h(t)
shown.
Solution
There is no overlap area between x(τ) and h(−τ) at t = 0, y(0) = 0.
For 0 < t < 6, there is overlap between the rectangles forming x(τ) and h(−τ). For t ≥ 6,
there is no overlap, and hence, y(6) = 0. These are illustrated in Figure 2.12 with the final
result for y(t). The shaded portion represents the overlap area of the products x(τ) and
h(t − τ).
Figure 2.12 Steps in the convolution and the final result
Example 2.11 Consider the RC circuit shown in Figure 2.13. Determine the V out (t) for V
in (t) = u(t − 1) − u(t − 2), and assume the time constant RC = 1 sec. Assume the capacitor
is initially discharged.
Letting we have
Example 2.13 If x(t) and h(t) are odd signals , then show that
y(t) = x(t) * h(t) is an even signal.
Solution y(t) = x(t) * h(t)
Hence,
The above equation can be rewritten as
Therefore,
Example 2.15 If the response of an LTI system to input x(t) is the output y(t), show that
the response of the system to is and using this result determines the impulse
response of an LTI system having the response y(t) = sin2t for an input x (t) = e −4tu(t).
Solution
From sampling property impulse function, it is known that x(t) δ (t) = x(0)δ(t).
Since e −4t δ(t) = e −0 δ(t) = δ(t), can be written as
Let x 1(t) = 4e −4t u(t), then by homogeneity , the corresponding output
As it is LTI system, if (x 1(t) + x 2(t)) is the input to the system, the corresponding
output is(y 1(t) + y 2(t))
since x 1(t) + x 2(t) = 4e −4t − 4e −4t + δ(t) = δ(t), the impulse response h(t) = y 1(t) + y
2(t) = 4 sin 2t + 2 sin 2t
Example 2.16 Consider a continuous-time LTI system with the unit step response s(t):
(i) Deduce that the response y(t) of the system to the input x(t) is
Thus,
Since y(t) = x(t) ∗ h(t) and if h(t) = δ(t) and y(t) = x(t) as x(t) ∗ δ(t) = x(t),thus, putting
h(t) = δ(t) in , it becomes
Since
(ii)
Since
Example 2.17 Consider h(t) be the triangular pulse and x(t) be the unit impulse train as
shown in Figure 2.16. Determine y(t) = x(t) * h(t) for T = 2.
Solution
(i) The following MATLAB program 2.1 is used to compute the convolution of x(t) and
h(t) of Example 2.9.
Program 2.1
The output y(t) = x(t) * h(t) of the above program is shown in Figure 2.18. It is
observed to be the same as that shown in Figure 2.11. Thus, it is verified.
Figure 2.18 Time versus y(t)
(ii) The following MATLAB program 2.2 is used to compute the convolution of x(t) and
h(t) of Example 2.10.
Program 2.2
The output y(t) = x(t) * h(t) of tie above program is shown in Figure 2.19 It is observed
to be the same as that shown in Figure 2.12. Thus, it is verified
Figure 2.19 Time versus y(t)
Example 2.19 Consider the RC circuit of Example 2.11 with time constant .
Determine the V out (t) using MATLAB for V in (t) = (u(t − 3) − u(t − 5)). Assume the
capacitor is initially discharged.
Solution The impulse response of the RC circuit is given by
The following MATLAB program 2.3 is used to compute the convolution of v in (t) and
h(t).
Program 2.3
The output V out (t) = V in (t) * h(t) of the above program is shown in Figure 2.20.
Figure 2.20 Time versus V out (t)
(2.37)
Figure 2.21 (a) Cascade connection of two systems. (b) Equivalent system
(2.38)
(2.39)
(2.41)
(2.42)
Figure 2.22 (a) Parallel connection of two systems. (b) Equivalent system
(2.43)
(2.44)
Example 2.20 An LTI system consists of two subsystems in cascade. The impulse
responses of the subsystems are, respectively, given by
Find the impulse response of the overall system.
Solution The overall impulse response of the system is given by
(2.45)
Example 2.21 Let y(t) be the periodic convolution of x 1(t) and x 2(t). Show that
Solution
(i)
Since x 2(t) is periodic with period T, x 2(t + T − τ) = x 2(t − τ)
(2.46)
(2.47)
Since the impulse response h(τ) = 0 for τ < 0 for a causal continuous-time system, the
output of a causal system can be expressed by the following convolution integral:
(2.48)
Example 2.22 Check stability of continuous-time system having the following impulse
responses:
(i)
Indicating that h(t) is absolutely integrable and, hence, h(t) is the impulse response of a
stable system ,
(ii)
Since e −τ|cos (2τ)| is exponentially decaying for 0 ≤ t ≤ ∞ , h(t) is absolutely
summable, and hence, h(t) is the impulse response of a stable system .
where N → ∞
Since h(t) is nonzero hence, h(t) is absolutely summable and, hence,
h(t) is the impulse response of an unstable system .
Example 2.23 Determine if each of the following system is causal or stable :
The process of recovering x(t) from x(t) * h(t) is called deconvolution as it corresponds
to reverse of the convolution operation.
The overall impulse response of the invertible system shown in Figure 2.23 is the
convolution of h(t) and g(t). It is required that the output of the invertible system is
equivalent to the input:
(2.50)
implying that
(2.51)
(2.52)
Hence, the input-output relation for the inverse system shown in Figure 2.24 is
(2.53)
Figure 2.24 Input-output relation for the inverse system
Example 2.24
(i) An echo of an auditorium can be modeled as a LTI system with an impulse response
consisting of a train of impulses:
However,
Implying that
Hence, g 0 = 1, g 1 = −a.
Example 2.25 Check y(t) = x(2t) for causality and invertibility.
Solution
At time t = 1
indicating that the value of y(t) at time t = 1 depends on x(t) at a time t = 2. Therefore,
y(t) = x(2t) is not causal.
y(t) is invertible;
(2.54)
where coefficients a n and b n are real constants. The order N refers to the highest
derivative of y(t) in Eq. (2.54). For example, consider the RC circuit considered in
Example 2.11, the input x(t) and the output y(t) = v o (t). If the current flowing through the
RC circuit is i(t), using Kirchhoff’s voltage law, we write
(2.55)
(2.56)
(2.57)
relating the voltage across the capacitor y(t) and the input x(t).
Example 2.26 Find the differential equation relating the current y(t) and the input voltage
x(t) for the RLC circuit shown in Figure 2.26 assuming R = 3 Ohms, L = 1 Henry, and
Example 2.27 Find the differential equation relating the input voltage and the
output voltage for the operational amplifier circuit shown in Figure 2.27.
Solution
which is rewritten as
(2.58)
where y c (t) is called the complementary solution and y p (t) is called the particular
solution . The complementary solution y c (t) is obtained by setting x(t) = 0 in Eq. (2.54).
Thus y c (t) is the solution of the following homogeneous differential equation
(2.59)
Example 2.28 Consider the RC circuit of Example 2.11 with time constant RC = 1 sec.
Determine the voltage across the capacitor for an input x(t) = e − 2 t u (t). Assume the
capacitor is initially discharged.
Solution As the time constant RC = 1, the input x(t) and the output of the RC
circuit are related by
The particular solution for the exponential input is of the form
yields
Thus, k = −1 and
Now,
at t = 0 y(0) = B – 1.
Since the capacitor is initially discharged, y(0) = 0, and we obtain B = 1.
Hence, the voltage across the capacitor is given by
2.5.3 Linearity
The system specified by Eq. (2.54) is linear only if all of the initial conditions are zero.
For instance, in the Example 2.11, if the capacitor is not assumed to be discharged
initially, then
A linear system has the property that zero input produces zero output.
However, if we let x(t) = 0, then
(2.60)
(2.61)
2.5.4 Causality
A linear system described by Eq. (2.54) is causal when it is initially relaxed. It implies that
if x(t) = 0 for t ≤ t 0, then y(t) = 0 for t ≤ t 0, thus, the response for t > to with the initial
conditions
(2.62a)
(2.62b)
2.5.5 Time-Invariance
For a linear causal system, initial rest also implies time-invariance.
For example, consider the system described by
(2.63)
(2.64)
so that
(2.65)
and
(2.66)
Now, let x 2(t) = x 1(t − τ) and y 2 (t) be the corresponding response. From Eq. (2.64),
we get
(2.67)
(2.68)
and
(2.69)
Eqs. (2.68) and (2.69) are satisfied and thus the system is time invariant .
(2.70)
yields
Thus, k = −1 and
The particular solution h p (t) is zero since h p (t) cannot contain δ(t). Thus,
yields
But
yields
Thus, and
The particular solution h p (t) is zero since h p (t)) cannot contain δ(t). Thus,
The output response y(t) is determined and displayed by the MATLAB commands
y = dsolve(‘Dy+y=exp(−2*t)’, , ‘y(0)=0’, ‘t’);
disp ([‘y(t) = (‘,char(y), ‘)u(t) ‘ ]);
Example 2.31 Consider the RLC circuit of Example 2.26 and determine the current y(t)
for the input voltage x(t) = 10e−3t u(t) where the initial inductor current is zero and the
initial capacitor voltage is equal to 5 volts.
Solution The relation between the output y(t) and the input x(t) is related by
The impulse response and step response obtained from the above program are shown
in Figure 2.29(a) and (b), respectively.
Figure 2.29 (a) impulse response , (b) step response
Figure 2.30 Block-diagram representation of basic elements (a) adder, (b) multiplier, (c) integrator
Consider the system described by Eq. (2.54), which is repeated here for convenience
assuming M = N:
(2.71)
If it is assumed that the system is at rest, then the Nth integral of , and
the Nth integral of is . Hence, taking the Nth integral of Eq. (2.71), we
obtain the integral description of the system as
(2.72)
(2.73)
The direct form I and the direct form II implementations of Eq. (2.73) are shown in
Figure 2.31(a) and (b), respectively.
Figure 2.31 Block-diagram representation for continuous-time system described by integral Eq. (2.73) (a) direct form I
, (b) direct form II
Solution
Figure 2.32 Block diagram representation in direct form II for a second order continuous time system described by the
above differential equation is shown in Figure 2.32
(2.74)
The unit impulse response of the considered system is the derivative of the unit
impulse which is referred as the unit doublet u 1(t).
By the convolution representation of LTI systems, we represent
(2.75)
for any input signal x(t). Similarly, for an LTI system described by
(2.76)
we obtain
(2.77)
(2.79)
(2.80)
and hence the unit step function is the impulse response of an integrator, and we have
(2.81)
(2.82)
(2.83)
(i)
(ii)
Solution
(i)
(ii)
(2.84)
where y(t)) is the system output and ʊ(t) is the system input.
Define the following useful set of state variables
(2.85)
(2.86)
Rearranging Eq. (2.84) and using Eq. (2.86), we obtain
(2.87)
(2.88)
Denoting ,
(2.89a)
(2.89b)
(2.90)
(2.92)
(2.93)
where
Eqs. (2.92) and (2.93) are called N-dimensional state-space representation or state
equations of the system.
In general state equations of a system are described by
(2.94)
(2.95)
Solution: The order of the differential equation is three. Hence, the three-state variables
are
The first derivatives of the state variables are
Example 2.35 Obtain the state-space representation for the electrical circuit shown in
Figure 2.33 considering , i1, and as state variables and v c1 as output y(t).
implying that
Hence,
Example 2.36 Obtain state-space representation of the circuit shown in Figure 2.34
considering the current through the inductor and voltage across the capacitor as state
variables and voltage across the capacitor as the output y(t).
Since the voltage across the capacitor is equal to the voltage across the series inductor and
resistor branch,
we obtain
Hence
Therefore
(2.96)
(2.97)
where
2.9 Problems
(iv)
(v) y(t) = ln (x(t))
2. Determine which of the following systems are linear and which are nonlinear:
(i)
(ii)
(iii)
(iv)
3. An amplifier has an output y(t) = cos (ωt). If y(t) is limited by clipping resulting in
the clipped output y c (t), check for linearity and time-invariance of y c (t).
4. An input signal x(t) and two possible outputs of a linear time-invariant system are
shown in Figure P2.1. Which outputs are possible given that the system is linear and
time invariant?
6. Determine graphically the convolution y(t) = x(t) * h(t) for the following:
(i)
(ii)
(iii)
(iv)
(v)
9. An LTI system consists of two subsystems in cascade. The impulse responses of the
subsystems are, respectively, given by
10. If y(t) = x(t) * h(t) shows that the area of the convolution y(t) is the product of the
areas of the signals that are being convolved x(t) and h(t), that is,
11. Compute and sketch the periodic convolution of the square-wave signal x(t) shown
in Fig. P2.2 with itself.
Figure P2.2 Square wave signal of problem 11
(ii) Check for causality of the inverse of the LTI system if h(t) is causal .
15. Determine the response y(t) of the OP-Amp circuit shown in Figure P2.3 for an input
x(t) = u(t)
16. Determine the impulse response y(t) of the OP-Amp circuit shown in Figure P2.4.
Figure P2.4 OP-Amp circuit of problem 16
18. Draw block diagrams for direct form II implementation of the corresponding
systems
(i)
(ii)
20. Obtain the state-space representation for the electrical circuit shown
In Figure P2.5 considering i 1, , and as state variables and current through the
inductor as the output y(t)
2. Write a MATLAB program to compute the convolution of the input x(t) and the
impulse response h(t) shown in Figure M2.1.
4. Write a MATLAB program to find the impulse response h(t) and step response for a
system described by
Further Reading
1. Oppenheim, A.V., Willsky, A.S.: Signals and Systems. Prentice-Hall, Englewood Cliffs (1983)
2. Hsu, H.: Signals and Systems Schaum’s Outlines, 2nd edn. McGraw-Hill, New York (2011)
3. Kailath, T.: Linear Systems. Prentice-Hall, Englewood Cliffs (1980)
4. Zadeh, L., Desoer, C.: Linear System Theory. McGraw-Hill, New York (1963)
© Springer International Publishing AG, part of Springer Nature 2018
K. Deergha Rao, Signals and Systems, https://doi.org/10.1007/978-3-319-68675-2_3
The continuous-time signals and systems are often characterized conveniently in a
transform domain. This chapter describes the transformations known as Fourier series and
Fourier transform which convert time-domain signals into frequency-domain (or spectral)
representations. The frequency domain representation of continuous-time signals are
described along with the conditions for the existence of Fourier series for periodic signals
and Fourier transform for nonperiodic signals and their properties. Finally, the frequency
response of continuous-time systems is discussed.
(3.1)
The fundamental period T 0 is small minimum, positive nonzero value of T for which
Eq. (3.1) is satisfied, and is referred to as the fundamental angular frequency.
A sinusoidal signal x(t) = cos (ω 0 t) and the complex exponential signal are
the two examples of periodic signals.
The complex exponentials related harmonically are expressed by
(3.2)
The fundamental frequency of each of these signals is a multiple of Ω0, and hence
each is periodic with period T 0 .
Thus, a linear combination of complex exponentials related harmonically can be
written as
(3.3)
(3.4)
(3.5)
(3.6)
(3.7)
Hence,
(3.8)
(3.10)
Thus, the Fourier series of a periodic signal is defined by Eq. (3.3) referred to as the
synthesis equation and Eq. (3.10) as the analysis equation.
(3.11)
2. x(t) must have finite number of maxima and minima during any single period of it.
3. x(t) must have finite number of discontinuities in any finite interval of time and each
of these discontinuities being finite.
Proof Since , the Fourier coefficients a n of a periodic signal x(t) are given by
Letting τ = t − t 0
implies that Re(a n) = Re(a –n), i.e., the real part of a n is even, andIm(a n) = −(Im(a −n)),
i.e., the imaginary part of a n is odd.
If x(t) is a continuous-time real and even signal, i.e., x*(t) = x(t) x(t) = x(−t), it can be
easily shown that a n = a −n and .
Similarly, if x(t) is a continuous-time real and odd signal, i.e., x*(t) = x(t) x(t) = −x(−t),
it can be easily shown that a n = −a −n and .
Frequency Shifting Property If x(t) is a continuous-time periodic signal with period T 0
and the Fourier coefficients a n , then the Fourier series coefficients of the are
given by a n−K .
Proof The Fourier coefficients a n of a periodic signal x(t) are given by
Let d n be the Fourier coefficients of , then
Thus, it is proved.
Time Reversal Property If x(t) is a continuous-time periodic signal with the Fourier
coefficients a n , then the Fourier series coefficients of the x(−t) are given by a −n .
Time Scaling Property If x(t) is a continuous-time periodic signal with period T 0 and
the Fourier coefficients a n , then the Fourier series coefficients of the x(αt) α > 0 are given
by a n with period .
Proof Since , the Fourier coefficients a n of a periodic signal x(t) are given by
Letting τ = αt
Hence, .
Since
giving the Fourier series representation of . Comparing this with the Fourier
series representation of coefficients a n , it is clear that the Fourier coefficients of are
Integration Property If x(t) is a continuous-time periodic signal with period T 0 and the
Fourier coefficients a n , then the Fourier series coefficients of the ∫x(t) are given by
Since
giving the Fourier series representation of ∫x(t)dt. Comparing this with the Fourier
series representation of coefficients a n , it is clear that the Fourier coefficients of ∫x(t)dt
are
Periodic Convolution If x 1(t) and x 2(t) are two continuous-time signals with common
period T 0 and Fourier coefficients a n and b n , respectively, then the Fourier series
coefficients of the convolution integral of x 1(t) and x 2(t) are given by T 0 a n b n .
Proof The periodic convolution integral of two signals with common period T 0 is
defined by
(3.12)
(3.13)
(3.14)
(3.15)
(3.16)
Assuming that x(t) is complex valued x(t)x*(t) = |x(t)|2 and x*(t) can be expressed in
terms of its Fourier series as
(3.17)
(3.18)
(3.19)
Interchanging the order of integration, Eq. (3.19) can be rewritten as
(3.20)
Thus,
Conjugate x *(t)
Re[a n ]
jIm[a n ]
Frequency a n−K
shifting
Differentiation in time
Integration ∫x(t)dt
Half-Wave Symmetry
If the two halves of one period of a periodic signal are of identical shape, except that one
is the negation of the other, the periodic signal is said to have a half-wave symmetry.
Formally, if x(t) is a periodic signal with period T 0, then x(t) has half-wave symmetry if
Example 3.1 Prove that Fourier series representation of a periodic signal with half-wave
symmetry has no even-numbered harmonics.
Proof A periodic signal x(t) with half-wave symmetry is given by
For
Example 3.2 Find Fourier series of the following periodic signal with half-wave
symmetry as shown in Figure 3.1.
For odd n
Example 3.3 Find Fourier series of the following periodic signal with half-wave
symmetry (Figure 3.2)
Figure 3.2 Periodic signal with half-wave symmetry
For odd n
(i) x(t) is the sum of two periodic signals with periods and The ratio
is a rational number.
where
and a n = 0 for all other. It is noticed that for all n. Hence, x(t) is a real signal.
By Parseval’s theorem,
Example 3.5 Find the Fourier series coefficients for each of the following signals:
(i)
(ii)
Since Ω 0 = 2π
The magnitude spectrum and phase spectrum are shown in Figures 3.3 and 3.4,
respectively.
Since x(t) is a real valued, its magnitude spectrum is even and the phase spectrum is
odd.
Example 3.7
(i) Obtain x(t) for the following non-zero Fourier series coefficients of a continuous-time
real valued periodic signal x(t) with fundamental period of 8.
(ii) Consider a continuous periodic signal with the following magnitude spectra shown
in Figure 3.5. Find the DC component and average power of the signal.
Solution (i)
Example 3.8 Which of the following signals cannot be represented by the Fourier series?
(i) x(t) = 4 cos(t) + 6 cos(t)
(ii) x(t) = 3 cos(πt) + 6 cos(t)
(iii) x(t) = cos(t) + 0.75
(iv) x(t) = 2 cos(3πt) + 3 cos(7πt)
(v) x(t) = e −|t| sin (5πt)
Solution
The ratio is not a rational number. Hence, x(t) is not a periodic signal.
(v) Due to decaying exponential function, it is not periodic. So Fourier series cannot be
defined for it.
Hence, (ii) and (v) cannot be represented by Fourier series . Since the remaining three
are periodic; they can be represented by Fourier series.
Example 3.9 Find the Fourier series of a periodic square wave with period T 0 defined
over one period by
For n ≠ o,
Since ,
Example 3.10 Find the Fourier series of the periodic signal shown in Figure 3.6.
For n=0,
For n ≠ o,
(3.21)
(3.22a)
(3.22b)
(3.23a)
(3.23b)
b n = 0 for all n.
If x(t) is an odd periodic signal, thena 0 = 0;a n = 0 for all n
(3.24)
Therefore, for every even signal b n = 0. Hence, Fourier series of an even signal
contains DC term and cosine terms only. Fourier series of an odd signal contains sine
terms only.
Example 3.11 Find the trigonometric Fourier series representation of the periodic signal
shown in Figure 3.7 with A = 3 and period T 0 = 2π.
Example 3.12 Find the trigonometric Fourier series representation of the periodic
triangle wave shown in Figure 3.8 with A = 2 period T 0 = 2.
The periodic triangle wave x(t) defined over one period with A=2 is
Since x(t) has odd symmetry, a 0 = 0 and a n = 0
Example 3.13 Find the trigonometric Fourier series representation of the following
periodic signal with period T 0 = 2.
For n = 1,
For n = 2,
For n = 3
For n = 4,5,6,… .
Example 3.14 If the input to the half-wave rectifier is an AC signal x(t) = cos(2πt), find
trigonometric Fourier series representation of output signal of the half-wave rectifier.
Solution The output y(t) of half-wave rectifier is
The sinusoidal input and output of half-wave rectifier are shown in Figure 3.9.
Figure 3.9 Input and output of half-wave rectifier
Since y(t) is a real and even function, its Fourier coefficients are real and even. The
period T 0=1.
For n = 0,
For n≠0
Figure 3.10 (a) Nonperiodic signal (b) periodic signal obtained from (a)
(3.25a)
(3.25b)
Since for and also since x(t) = 0 outside this interval , so we have
(3.26)
Define
(3.27)
Then
(3.28)
(3.29)
(3.30)
Eq. (3.27) is referred to as the Fourier transform of x(t), and Eq. (3.30) is called the
inverse Fourier transform .
(3.31)
2. x(t) must have a finite number of maxima and minima within any finite interval.
3. x(t) must have a finite number of discontinuities within any finite interval, and each of
these discontinuities is finite.
Although the above Dirichlet conditions guarantee the existence of the Fourier
transform for a signal, if impulse functions are permitted in the transform, signals which
do not satisfy these conditions can have Fourier transforms.
Example 3.15 Determine x(0) and X(0) using the definitions of the Fourier transform and
the inverse Fourier transform
Solution By the definition of Fourier transform , we have
(3.32)
implying that the Fourier transform of the unit impulse contribute equally at all
frequencies
Example 3.16 Find the Fourier transform of x(t) = e −bt u(t) for b > 0.
Solution
(i)
By the sampling property of the impulse function, we have
Hence,
Thus
Therefore,
(ii)
Example 3.19 Find the Fourier transform of the rectangular pulse signal shown in Figure
3.11
Solution
Hence,
Example 3.20 Consider the Fourier transform X(jΩ) of a signal shown in Figure 3.12.
Find the inverse Fourier transform of it.
Solution
Solution
Letting
Letting
Since
Substituting
(3.33)
Since
Symmetry for Real Valued Signal If x(t) is a continuous-time real valued signal with
Fourier transform X(jΩ), then
(3.34)
implying that the real part is an even function of Ω and the imaginary part is an odd
function of Ω.
For real x(t) in polar form
indicating that the magnitude is an even function of Ω and the phase is an odd function
of Ω.
Symmetry for Imaginary Valued Signal If x(t) is a continuous-time imaginary valued
signal with Fourier transform X(jΩ), then
(3.35)
Proof
Since x(t) is purely imaginary, we get x(t) = −x*(t), and we get
(i) If x(t) is a continuous-time real valued and has even symmetry, then
(3.36a)
(ii) If x(t) is a continuous-time real valued and has odd symmetry, then
(3.36b)
Proof Since x(t) is real x(t) = x*(t) and x(t) has even symmetry x(t) = x(−t), we get
(3.37)
Proof
Hence,
Proof
(3.39)
Thus,
Hence,
Thus,
Therefore,
since
(ii)
Since
(i)
(ii)
Solution (i)
Hence,
(ii)
(3.41)
Example 3.27 Find the Fourier transform of the following continuous-time signal:
Solution For
For
For
Thus
for
Integration If x(t) is a continuous-time signal with Fourier transform X(jΩ), then the
Fourier transform of the is given by
(3.42)
(3.43)
(3.44)
Assuming that x(t) is complex value x(t)x ∗(t) = |x(t)|2 and x*(t) can be expressed in
terms of its Fourier transform as
(3.45)
(3.46)
(3.47)
Interchanging the order of integration , Eq. (3.47) can be rewritten as
(3.48)
Thus,
Example 3.28 Consider a signal x(t) with its Fourier transform given by
can be written as
where
Since
By linearity property,
(3.49)
Hence, convolution of two sequences x 1(t) and x 2(t) in the time domain is equal to the
product of their frequency spectra.
Proof By the definition of convolution integral ,
(3.50)
(3.51)
(3.52)
(3.53)
Thus
Example 3.29 Determine the Fourier transform of the triangular output signal y(t) of an
LTI system as shown in Figure 3.13 .
Figure 3.13 Triangular output signal
By definition,
Example 3.30 Consider an LTI continuous-time system with the impulse response
. Find the output y(t) of the system for an input
Solution Since
shown as
which is depicted as
Therefore,
Duality property
For a given Fourier transform pair
By interchanging the roles of time and frequency, a new Fourier transform pair is
obtained as
For example, the duality exists between the Fourier transform pairs of Examples 3.19
and 3.20 as given by
(3.54)
This can be easily proved by dual property. Eq. (3.54) is called the modulation
property since the multiplication of two signals often implies amplitude modulation .
Example 3.31 Find the Fourier transform of
By modulation property
By convolution property
There is no overlap between the two transforms X 1(jΩ) and X 2(jΩ), and hence
Therefore,
Solution Since the Fourier transform of sinc (2t) is , using Parseval’s theorem ,
Example 3.34 Consider a signal x(t) with its Fourier transform given by
Hence,
Symmetry
Re[X(jΩ)
jIm[X(jΩ)]
Integration
Table 3.3 Basic Fourier transform pairs
δ(t) 1
1 2πδ(Ω)
2πδ(Ω−Ω0)
sin(Ω0t) π[δ(Ω−Ω 0 )−δ(Ω−Ω 0 )]
δ(t−t 0)
u(t)
Sgn(t)
(3.55)
(3.56)
(3.57)
(3.58)
(3.59)
(3.60)
(3.61)
Example 3.36 Find the frequency response H(jΩ) of the following circuit
Solution
(3.63)
(3.63a)
(3.65)
(3.66)
(3.67)
(3.68)
Eq. (3.61) implies that the AM shifts the message signal so that it is centered at ±Ωc.
The message signal m(t) can be recovered if Ωc >Ωm so that the replica spectra do not
overlap. The AM modulation in frequency domain is illustrated in Figure 3.14.
Amplitude Demodulation
The message signal m(t) can be extracted by multiplying the AM signal s(t) by the same
carrier c(t) and passing the resulting signal through a low-pass filter as shown in Figure
3.15.
(3.69)
3.6 Problems
1. Find the exponential Fourier series representation for each of the following signals:
(i) x(t) = cos(Ω0 t)
(iii)
(iv) x(t) = sin2(t)
(vi)
3. If x 1 (t) and x 2(t) are periodic signals with fundamental period T 0, find the Fourier
series representation of x(t) = x 1(t)x 2(t).
5. Find the Fourier series of a periodic signal x(t) with period 3 defined over one period
by
6. Find the Fourier series of a periodic signal x(t) with period 6 defined over one period
by
15. Find the Fourier transform of the complex sinusoidal pulse given by
16. Find Fourier transform of the following signal shown in Figure P3.5
Figure P3.5 Signal x(t)
17. Consider the following two signals x(t) and y(t) as shown in Figure P3.6 . Determine
Fourier transform y(t) using the Fourier transform of x(t), time shifting property, and
differentiation property
20. Consider the following communication system shown in Figure P3.7 to transmit two
signals simultaneously over the same channel.
Figure P3.7 Communication system
Plot the spectra of x(t), y(t), and z(t) for given the following spectra of the two input
signal shown in Figure P3.8.
Further Reading
1. Lanczos, C.: Discourse on Fourier Series. Oliver Boyd, London (1966)
2. Körner, T.W.: Fourier Analysis. Cambridge University Press, Cambridge (1989)
3. Walker, P.L.: The Theory of Fourier Series and Integrals. Wiley, New York (1986)
4. Churchill, R.V., Brown, J.W.: Fourier Series and Boundary Value Problems, 3rd edn. McGraw-Hill, New York (1978)
5. Papoulis, A.: The Fourier Integral and Its Applications. McGraw-Hill, New York (1962)
6. Bracewell, R.N.: Fourier Transform and Its Applications, rev, 2nd edn. McGraw-Hill, New York (1986)
7. Morrison, N.: Introduction to Fourier Analysis. Wiley, New York (1994)
8. Lathi, B.P.: Linear Systems and Signals, 2nd edn. Oxford University Press, New York (2005)
9. Oppenheim, A.V., Willsky, A.S.: Signals and Systems. Englewood Cliffs, NJ, Prentice- Hall (1983)
© Springer International Publishing AG, part of Springer Nature 2018
K. Deergha Rao, Signals and Systems, https://doi.org/10.1007/978-3-319-68675-2_4
4. Laplace Transforms
K. Deergha Rao1
(1) Department of Electronics and Communication Engineering, Vasavi College of
Engineering (Affiliated to Osmania University), Hyderabad, Telangana, India
The Laplace transform is a generalization of the Fourier transform of a continuous time
signal. The Laplace transform converges for signals for which the Fourier transform does
not. Hence, the Laplace transform is a useful tool in the analysis and design of continuous
time systems. This chapter introduces the bilateral Laplace transform , the unilateral
Laplace transform , the inverse Laplace transform, and properties of the Laplace transform
. Also, in this chapter, the LTI systems, including the systems represented by the linear
constant coefficient differential equations , are characterized and analyzed using the
Laplace transform. Further, the solution of state-space equations of continuous time LTI
systems using Laplace transform is discussed.
(4.1)
The complex variable s is of the forms = σ + jΩ, with a real part σ and an imaginary
part Ω. The Laplace transform defined by Eq. (4.1) is called as the bilateral Laplace
transform.
The difference between Eqs. (4.1) and (4.2) is on the lower limit of the integration. It
indicates that the bilateral Laplace transform depends on the entire signal, whereas the
unilateral Laplace transform depends on the right-sided signal , i.e., x (t) = 0 for t < 0.
(4.3)
(4.4)
(4.5)
(4.7)
The right hand side of Eq. (4.7) is the Fourier transform of x(t)e −σt. Thus, the Laplace
transform can be interpreted as the Fourier transform of x(t) after multiplication by a real
exponential signal.
(4.8)
where N(s) is the numerator polynomial and D(s) represents the denominator polynomial.
The Eq. (4.8) is referred to as rational. The roots of the numerator polynomial are referred
to as zeros of X(s) = 0 because for those values of s, X(s) becomes zero. The roots of the
denominator polynomial are called the poles of X(s), as for those values of s, X(s) = ∞. A
rational Laplace transform can be specified by marking the locations of poles and zeros by
x and o in the s-plane, which is called as pole-zero plot of the Laplace transform. For a
signal, the Laplace transform converges for a range of values of s. This range is referred to
as the region of convergence (ROC) , which is indicated as shaded region in the pole-zero
plot.
(4.9)
Then, the line ℜe(s) = σ 1 is in the ROC . For ℜe(s) = σ 2 also to be in the ROC, it is
required that
(4.10)
Hence,
(4.11)
The RHS of Eq. (4.11) is bounded and hence the LHS. Thus, the ℜe(s) > σ 1 must also
be in the ROC. Similarly, if σ 2 < σ 1 , it can be shown that is absolutely
integrable. Hence, the ROC is the entire s-plane.
Property 4: If ROC of a right-sided signal contains the line ℜe(s) = σ 1, then
ℜe(s) > σ 1 will also be in the ROC for all values of s.
Proof For a right-sided signal, x(t) = 0 prior to some finite time t 1 as shown in Figure
4.2
Figure 4.2 Right-sided signal
(4.12)
(4.13)
Proof For left-sided signal, x(t) = 0 after some finite time t 2 as shown in Figure 4.3. This
can be proved easily with the same argument and intuition for the property 4.
Property 6: If ROC of a two-sided signal contains the line ℜe(s) = σ 0 , then ROC
will contain a strip, which includes the line.
Proof A two-sided signal is of infinite duration for both t > 0 and t < 0 as shown in
Figure 4.4(a)
Figure 4.4 (a) Two-sided signal (b) Right-sided signal . (c) Left-sided signal . (d) ROC of the two-sided signal
Let us choose an arbitrary time t 0 that divides the signal into as sum of right-sided
signal and left-sided signal as shown Figure 4.4(b) and (c). The Laplace transform of x(t)
converges for the values of s for which both the right-handed signal and left-handed signal
converge. It is known from property 4 that the ROC of Laplace transform of right-handed
signal X r (s) consists of a half plane ℜe(s) > σ r for some value σ r ; and from property 5, it
is known that X r (s) consists of a half plane ℜe(s) > σ l for some value σ l . Then the
overlap of these two half planes is the ROC of the two-sided signal x(t) as shown in Figure
4.4(d) with the assumption that σ r < σ l . If σ r is not less than σ l , then there is no overlap.
In this case, X(s) does not exist even X r (s) and X l (s) individually exist.
(4.14)
Applying the inverse Fourier transform on the above relationship, we obtain
(4.15)
(4.16)
(4.17)
(4.18)
(4.19)
(4.20)
The result concerning the ROC follows directly from the theory of complex variables
concerning the convergence of a sum of two convergent series.
Time Shifting If x(t) is a signal with Laplace transform X(s) and ROC R, then for any
constant t 0 ≥ 0, the Laplace transform of x(t – t 0) is given by
(4.22)
Substituting τ = t – t 0,
(4.23)
Shifting in the s-domain If x(t) is a signal with Laplace transform X(s) and ROC R,
then the Laplace transform of the signal is given by
(4.24)
(4.25)
Time Scaling If x(t) is a signal with Laplace transform X(s) and ROC R, then the
Laplace transform of the x(at) for any constant a, real or complex, is given by
(4.26)
(4.27)
(4.28)
Differentiation in the Time Domain If x(t) is a signal with the Laplace transform X (s)
and ROC R, then
(4.29)
Then,
(4.30)
From the above expression, it can be stated that the inverse Laplace transform of sX(s)
is .
Differentiation in the s-Domain If x(t) is a signal with the Laplace transform X(s), then
(4.31)
Proof From the definition of the Laplace transform,
(4.32)
with ROC = R
Division by t
If x(t) is a signal with Laplace transform X(s), then
(4.33)
(4.34)
(4.35)
(4.36)
Integration If x(t) is a signal with Laplace transform X(s) and ROC R, then
(4.37)
Then,
Convolution in the Time Domain If x 1(t) and x 2(t) are two signals with Laplace
transforms X 1(s) and X 2(s) with ROCs R 1 and R 2, respectively, then
(4.38)
(4.39)
(4.40)
Let t 1 = t – τ; dt 1 = dt;
(4.41)
(4.42)
(4.43)
(4.44)
(4.45)
(4.46)
where
(4.47)
This integral converges for ℜe(s − p) > σ 2 . By substituting Eq. (4.47) in Eq. (4.46),
yields the following proving the property
(4.48)
The above properties of the Laplace transform are summarized in Table 4.1.
Table 4.1 Some properties of the Laplace transform
(4.49)
Consider
(4.50)
(4.51)
(4.52)
Consider
(4.53)
(4.54)
(4.55)
Applying this second time yields the unilateral Laplace transform of as given by
(4.56)
Similarly, applying this for third time yields the unilateral Laplace transform
(4.57)
Continuing this procedure for the nth time, the unilateral Laplace transform of is
given by
(4.58)
Example 4.1 Determine whether the following Laplace transforms correspond to the
even time function or odd time function and comment on the ROCs .
(a)
(b)
(c) Comment on the ROCs
Solution
(a)
Hence, the corresponding x(t) is an odd function.
(b)
Hence, the corresponding x(t) is an even function.
(c) The ROCs for (a) and (b) are shown in Figure 4.5(a) and (b), respectively. From
Figure 4.5(a) and (b), it can be stated that for the time function to be even or odd, the
ROC must be two sided.
Example 4.2 A real and even signal x(t) with its Laplace transform X(s) has four poles
with one pole located at , with no zeros in the finite s-plane and X(0) = 16. Find X(s).
Solution Since X(s) has four poles with no zeros in the finite s-plane, it is of the form
As x(t) is real, the poles of X(s) must occur as conjugate reciprocal pairs.
Hence, and X(s) becomes
Since x(t) is even, the X(s) also must be even, and hence the poles must be symmetric
about the jΩ axis. Therefore, .
Thus,
We obtain
Hence,
(4.59)
Proof To prove the theorem, let the following integral first be evaluated by using
integration by parts
(4.60)
(4.61)
As the integration is independent of s, the calculation of the limit and the integration
can be permuted provided that the integral converges uniformly. If {x(t)} exists, then
(4.62)
(4.63)
(4.64)
(4.65)
Again one can permute the sequence of determining the limit and the integration
provided the integral converges. The result is
(4.66)
Therefore,
(4.68)
(4.69)
(4.70)
(4.71)
(4.72)
(i)
Because u(−t) = 1 for t < 0 and u(−t) = 0 for t > 0,
(ii)
Because u(−t) = 1 for t < 0 and u(−t) = 0 for t > 0,
Similarly, for n = 3, .
Therefore,
Example 4.8 Find the Laplace transform of x(t) = cos ωt u(t).
Solution
Therefore,
Solution
Solution
(b) For which of the following values of a does the Fourier transform of x(t) e −αt
converge?
(i) α = 1
(ii) α = 2.5
(a) The Fourier transform of the signal does not converge as x(t) is not absolutely
integrable due to the rising exponentials.
and its pole-zero plot and ROC are as shown in Figure 4.6.
Determine the system output y(t) for all t if the input x(t) is given by x(t) = e −3t/2 + 2e
−t for all t.
Using linearity and superposition, it can be recognized that if x(t) = e −3t/2 + 2e −t, then
So that
Hence,
(4.73)
with the order of N(s) is less than the order of the denominator polynomial.
The poles p 1, p 2, … .., p n are distinct.
The rational Laplace transform X(s) can be expanded using partial fraction expansion
as
(4.74)
The coefficients k1, k2, … ., kn are called the residues of the partial fraction expansion.
The residues are computed as
(4.75)
With the known values of the coefficients k1, k1, … ., kn inverse transform of each
term can be determined depending on the location of each pole relative to the ROC .
(4.76)
with multiplicity r poles at s = p 1.
The X(s) with multiple poles can be expanded as
(4.77)
The coefficients k2, … ., kn can be computed using the residue formula used in Section
4.6.1. The residues k11, k12, …, klr are computed as
(4.78)
(4.79)
(4.80)
and so on.
Example 4.16 Find the time function x(t) for each of the following Laplace transform
X(s)
(a)
(b)
(c)
(d)
Solution
(a)
Using partial fraction expansion,
Thus,
δ(t) 1 All s
δ(t − t 0) All s
ℛe(s) > 0
ℛe(s) > − α
(b)
Using partial fraction expansion,
Thus,
Hence,
(d)
Using partial fraction expansion,
Thus,
Solution
Example 4.18 Determine x(t) for the following conditions if X(s) is given by
The ROC is to the right of the rightmost pole as shown in Figure 4.7(a).
Figure 4.7 (a) ROC of right-sided x(t) (b) ROC of left-sided x(t) (c) ROC of both-sided x(t)
The ROC is to the left of the leftmost pole as shown in Figure 4.7(b)
(i)
(ii)
Solution
(i)
(ii)
where N is the vector of the numerator polynomial coefficients in decreasing order and
the vector D contains the denominator polynomial coefficients in decreasing order.
Example 4.20 Find the inverse Laplace transform of the following using MATLAB
Solution The following MATLAB statements are used to find the Laplace transform of
given X(s)
coefficients of the numerator polynomial in decreasing order
After getting the partial fraction expansion of X(s), the following MATLAB
statements are used to obtain x(t), that is, the inverse Laplace transform of X(s).
syms s t
X=2/(s+3)-1/(s+2);
ilaplace(X)
Execution of the above three MATLAB statements gives
(4.81)
(4.82)
indicating the Laplace transform of the output signal y(t) is the product of the Laplace
transforms of the impulse response h(t) and the input signal x(t). The transform H(s) is
called the transfer function or the system function and expressed as
(4.83)
The roots of the denominator of the transfer function are called poles. The roots of the
numerator are called zeros. The places where the transfer function is infinite (the poles)
determine the region of convergence .
Example 4.21 Obtain the system function of the following Sallen-Key low-pass filter
circuit.
Solution For the circuit shown in Figure 4.8, the following relations can be established:
Figure 4.8 Sallen-Key low-pass filter circuit
Substituting the above Eq. for , the input-output relation can be written as
Thus,
(4.84)
The Laplace transform of the impulse response is known as the system function, which
can be written as
(4.85)
A continuous-time LTI system is stable if and only if the transfer function has ROC
that includes the imaginary axis (the line in complex where the real part is zero).
Causal LTI System
A continuous time LTI system is causal if its output y(t) depends only on the current and
past input x(t) but not the future input. Hence, h(t) = 0 for t < 0.
For causal system, the system function can be written as
(4.86)
If s = σ + jΩ and h(t)e −σt being absolutely integrable for convergence of H(s) leads to
the following convergence condition,
(4.87)
Any large value of σ satisfies the above equation . Thus, the ROC is the region to the
right of a vertical line that passes through the point ℜe(s) = σ as shown in Figure 4.9.
Figure 4.9 ROC of a causal LTI system
In particular, if H(s) is rational, , then the system is causal if and only if its
ROC is the right-sided half plane to the right of the rightmost pole and the order of
numerator N(s) is no greater than that of the denominator D(s), so that the ROC is a right-
sided plane without any poles (even at s → ∞).
Stable and Causal LTI System
As the ROC of a causal system is to the right of the rightmost pole and for a stable system,
the rightmost pole should be in the left half of the s-plane and should include the jΩ axis;
all the poles of a system should lie in the left half of the s-plane (the real parts of all poles
are negative, ℜe(s p ) < 0 for all s p ) for a system to be causal and stable as shown in
Figure 4.10.
Show the pole-zero locations of the system and ROCs for the following:
Since the ROC of a causal system is to the right of the rightmost pole, the pole-
zero plot and ROC of the system are shown in Figure 4.11.
Figure 4.12 Pole-zero plot and ROC of the noncausal, stable system
(c) Since the system is neither causal nor stable, the ROC should not include the
imaginary axis and not to the right of the rightmost pole. Hence, the pole-zero plot
and the ROC are shown in Figure 4.13.
Figure 4.13 Pole-zero plot and ROC of the system neither causal nor stable
(4.88)
Taking the Laplace transform of both sides of equation repeated use of the
differentiation property and linearity property , we obtain
(4.89)
(4.90)
The pole-zero plot for the system function is shown in Figure 4.14.
(iii) For the system to be neither causal nor stable , the ROC is ℜe(s) < − 2.
Hence,
Solution
Step 1: Laplace transform both sides of given differential equation is
Comparing the numerator polynomial with the numerator polynomial of Y(s) of step 2,
we get
Thus,
Example 4.25 Find the solution of the following differential equation using Laplace
transform
Solution
Step 1: Laplace transform both sides of given differential equation is
Step 2: Simplifying the expression for Y(s),
Comparing the numerator polynomial with the numerator polynomial of Y(s) of step 2,
we get
Example 4.26 Consider the following RLC circuit with R = 5 ohms, L = 1h, and
C = 0.25 f.
(a) Determine the differential equation relating and .
Solution
(a) By applying Kirchhoff’s voltage law, we can arrive at the following differential
equation:
, , .
Thus,
Example 4.27 Consider the following parallel RLC circuit with R=1 ohm, L=1 h.
(b) Obtain zero-state response for I L (t) using Laplace transform for I s (t) = e −3t u(t).
(c) Obtain zero-input response for I L (t) using Laplace transform with I L (0) = 1 (Figure
4.16).
Solution
Applying the unilateral Laplace transform to the above differential equation, we obtain
(c) For the zero-input response, I s (t) = 0 and given that I L (0) = 1, we have to find the
solution of the following differential equation for zero-input response
Thus,
Example 4.29 Find the solution of linear differential equation considered in Example
4.25 using MATLAB
Solution The following MATLAB statements are used to find the solution of the
differential equation considered in Example 4.25
Execution of the above MATLAB statements gives the solution of the differential
equation as
(4.91)
and from convolution property of the Laplace transform, the associated system function is
(Figure 4.17)
(4.92)
Figure 4.17 Series combination of two LTI systems
(4.93)
and from linearity property of the Laplace transform , the associated system function is
(Figure 4.18)
(4.94)
(4.95)
Let us define the following basic elements for addition, multiplication, differentiation,
and integration in the s-domain (Figure 4.19)
Figure 4.19 Block-diagram representation basic elements (a) adder, (b) multiplier, (c) differentiator, (d) integrator
The block-diagram representation of the above system function can be obtained as the
interconnection of these basic elements similar to the block-diagram representation of
differential equations in the time domain carried out in Section 2.6 of Chapter 2. The
block-diagram representation of the system function is shown in Figure 4.20.
Example 4.30 Is the system represented by the following block-diagram stable ? (Figure
4.21)
Figure 4.21 Block-diagram representation of a second-order system function
Solution Let the signal at the bottom node of the block diagram be denoted by E(s).
Then we have the following relations
From the system function H(s) found in the previous part, we see that the poles of the
system are at s = −1. Since the system is given to be causal , and the rightmost pole of the
system is left of the imaginary axis, the system is stable .
(4.96)
(4.97)
(4.98)
(4.100a)
(4.100b)
(4.101)
(4.102)
(4.103)
Thus,
(4.104)
Example 4.31 Consider the electrical circuit given in Example 2.36. Find if
under an initially relaxed condition.
Solution
Since the circuit is initially relaxed, e At X(0) = 0.
Therefore,
Since
Hence,
4.11 Problems
4. Show that the ROC for the Laplace transform of a noncausal signal is the region to
the left of a vertical line in the s-plane.
5. Find Laplace transform of a periodic signal with period T, that is, x(t+T) = x(t).
6. By first determining x(t), verify the final value theorem for the following with
comment
Find the impulse response and region of convergence and the value of α for the
system to be causal and stable.
9. Find the solution of the following differential equation using Laplace transform
12. Determine the differential equation characterizing the system represented by the
following block diagram
13. Consider the following state-space representation of a system. Determine the system
output y(t) with the initial state condition .
Further Reading
1. Doetsch, G.: Introduction to the theory and applications of the Laplace transformation with a table of Laplace
transformations. Springer, New York (1974)
2. LePage, W.R.: Complex variables and the Laplace transforms for engineers. McGraw-Hill, New York (1961)
3. Oppenheim, A.V., Willsky, A.S.: Signals and systems. Prentice-Hall, Englewood Cliffs (1983)
4. Hsu, H.: Signals and systems, 2nd edn. Schaum’s Outlines, Mc Graw Hill (2011)
5. Kailath, T.: Linear systems. Prentice-Hall, Englewood Cliffs (1980)
6. Zadeh, L., Desoer, C.: Linear system theory. McGraw-Hill, New York (1963)
© Springer International Publishing AG, part of Springer Nature 2018
K. Deergha Rao, Signals and Systems, https://doi.org/10.1007/978-3-319-68675-2_5
5. Analog Filters
K. Deergha Rao1
(1) Department of Electronics and Communication Engineering, Vasavi College of
Engineering (Affiliated to Osmania University), Hyderabad, Telangana, India
Filtering is an important aspect of signal processing. It allows desired frequency
components of a signal to pass through the system without distortion and suppresses the
undesired frequency components. One of the most important steps in the design of a filter
is to obtain a realizable transfer function H(s), satisfying the given frequency response
specifications. In this chapter, the design of analog low-pass filters is first described.
Second, frequency transformations for transforming analog low-pass filter into band-pass ,
band-stop , or high-pass analog filters are considered. The design of analog filters is
illustrated with numerical examples. Further, the design of analog filters using MATLAB
is demonstrated with a number of examples. Also, the design of special filters by pole and
zero placement is illustrated with examples.
(5.1)
Hence,
(5.2)
(5.3)
(5.4)
Thus,
(5.5)
The filter bandwidth is proportional to Ω c and the width of the main lobe is
proportional to . The impulse response becomes narrow with increase in the bandwidth.
High-Pass Filter
The following system (Figure 5.3) is generally used to obtain high-pass filter from a low-
pass filter . The frequency response of an ideal analog high-pass filter H HP (Ω) that passes
a signal for |Ω| > Ω c can be expressed by
(5.6)
and is shown in Figure 5.4. The frequency Ω c is called the cutoff frequency
From Figure 5.3, the frequency response of the ideal high-pass filter can also be expressed
as
(5.7)
Therefore, the impulse response of an ideal high-pass filter is given by the inverse
Fourier transform of Eq. (5.7).
Hence, the impulse response of the ideal high pass filter is given by
(5.8)
Band-Pass Filter
The frequency response of band-pass filter can be expressed by
(5.9)
From Figure 5.5, the frequency response of the ideal band-pass filter can be expressed as
(5.10)
Thus, the impulse response of an ideal band-pass filter is
(5.11)
Band-Stop Filter
The band-stop filter can be realized as a parallel combination of low-pass filter with cutoff
frequency Ωc1 and high-pass filter cutoff frequency Ωc2. The frequency response of band-
stop filter can be expressed by
(5.12)
(5.13)
Figure 5.6 System with summation of high-pass filter and low-pass filter
(5.14)
(5.15)
(5.16)
(5.17)
For Ω = Ω s , the stopband attenuation is
(5.18)
(5.19)
(5.20)
(5.21)
(5.22)
(5.23)
Since the order N must be an integer, the value obtained is rounded to the next higher
integer. This value of N is used in either Eq. (5.19) or Eq. (5.20) to determine the 3 dB
cutoff frequency Ω c . In practice, Ω c is determined by Eq. (5.20) that exactly satisfies
stopband specification at Ω c , while the passband specification is exceeded with a safe
margin at Ω p . We know that |H(jΩ)|2 may be evaluated by letting s = jΩ in H(s)H(−s),
which may be expressed as
(5.24)
(5.25)
where
(5.26)
Since |s k | = 1, we can conclude that there are 2N poles placed on the unit circle in the
s-plane. The normalized transfer function can be formed as
(5.27)
where p l for l = 1, 2,.., N are the left half s-plane poles . The complex poles occur in
conjugate pairs.
For example, in the case of N = 2, from Eq. (5.26), we have
Hence,
and
In the case of N = 3,
Hence
and
The following MATLAB Program 5.1 can be used to obtain the Butterworth
normalized transfer function for various values of N.
Program 5.1 Analog Butterworth Low-Pass Filter Normalized Transfer Function
N=input(‘enter order of the filter ’);
[z,p,k] = buttap(N)% determines poles and zeros
disp(‘Poles are at’);disp(p);
[num,den]=zp2tf(z,p,k);
%Print coefficients in powers of s
disp(‘Numerator polynomial’);disp(num);
disp(‘Denominator polynomial’);disp(den);
sos=zp2sos(z,p,k);%determines coefficients of second order
sections
The normalized Butterworth polynomials generated from the above program for typical
values of N are tabulated in Table 5.1.
Table 5.1 List of normalized Butterworth polynomials
N Denominator of H N (s)
1 S + 1
3 (s + 1)(s 2 + s + 1)
The magnitude response of the normalized Butterworth low-pass filter for some
typical values of N is shown in Figure 5.9. From this figure, it can be seen that the
response monotonically decreases both in the passband and the stopband as Ω increases.
As the filter order N increases, the magnitude responses both in the passband and the
stopband are improved with a corresponding decrease in the transition width. Since the
normalized transfer function corresponds to Ω c = 1, the transfer function of the low-pass
filter corresponding to the actual Ω c can be obtained by replacing s by (s/Ω c ) in the
normalized transfer function.
Figure 5.9 Magnitude response of typical Butterworth low-pass filter
Example 5.1 Design a Butterworth analog low-pass filter with 1 dB passband ripple,
passband edge frequency Ω p = 2000π rad/sec, stopband edge frequency Ω s = 10,000π
rad/sec, and a minimum stopband ripple of 40 dB.
Solution Since α s = 40 dB, α p = 1 dB, Ω p = 2000π, and Ω s = 10,000π,
(5.28)
(5.29)
(5.30)
(5.31)
(5.33)
(5.34)
Substituting Eq. (5.32) for ε in the above equation and solving for N, we get
(5.35)
The poles of the normalized Type 1 Chebyshev filter transfer function lie on an ellipse
in the s-plane and are given by
(5.36)
(5.37)
(5.38)
where
(5.39a)
and
(5.39b)
As an illustration, consider the case of N = 2 with a passband ripple of 1 dB. From Eq.
(5.32), we have
Hence
Therefore, from (5.39a), the poles of the normalized Chebyshev transfer function are
given by
Hence
Thus for N = 2, with a passband ripple of 1 dB, the normalized Chebyshev transfer
function is
Similarly for N = 3, for a passband ripple of 1 dB, we have
Thus,
Hence, the normalized transfer function of Type 1 Chebyshev low-pass filter for N=3
is given by
The following MATLAB Program 5.2 can be used to form the Type1 Chebyshev
normalized transfer function for a given order and passband ripple.
Program 5.2 Analog Type 1 Chebyshev Low-Pass Filter Normalized Transfer Function
N=input(‘enter order of the filter’);
Rp=input(‘enter passband ripple in dB’);
[z,p,k] = cheb1ap(N,Rp)% determines poles and zeros
disp(‘Poles are at’);disp(p);
[num,den]=zp2tf(z,p,k);
%Print coefficients in powers of s
disp(‘Numerator polynomial’);disp(num);
disp(‘Denominator polynomial’);disp(den);
The normalized Type 1 Chebyshev polynomials generated from the above program for
typical values of N and passband ripple of 1 dB are tabulated in Table 5.2.
Table 5.2 List of normalized Type 1 Chebyshev transfer functions for passband ripple = 1 dB
N Denominator of H N (s) H 0
1 S + 1.9652 1.9652
The typical magnitude responses of a Type 1 Chebyshev low-pass filter for N = 3, 5, and 8
with 1 dB passband ripple are shown in Figure 5.10. From this figure, it is seen that Type
1 Chebyshev low-pass filter exhibits equiripple in the passband with a monotonic decrease
in the stopband.
Figure 5.10 Magnitude response of typical Type 1 Chebyshev low-pass filter with 1 dB passband ripple
Example 5.2 Design a Type 1 Chebyshev analog low-pass filter for the specifications
given in Example 5.1.
Solution Since α s = 40 dB, α p = 1 dB, Ω p = 2000π, and Ω s = 10,000π,
Since the order of the filter must be an integer, we choose the next higher integer value 3
for N. The normalized Type 1 Chebyshev low-pass filter for N = 3 with a passband ripple
of 1 dB is given from Table 5.2 as
(5.40)
The order N can be determined using Eq. (5.35). The Type 2 Chebyshev filter has both
poles and zeros , and the zeros are on the jΩ axis. The normalized Type 2 Chebyshev low-
pass filter , or the normalized inverse Chebyshev filter (normalized to Ω s = 1), may be
formed as
(5.41)
where
(5.42a)
(5.42b)
(5.42c)
(5.42d)
(5.42e)
(5.42f)
For example, if we consider N = 3 with a stopband ripple of 40 dB, then from (5.42e),
Hence,
Hence
Thus, from (5.42b), the poles are
Therefore, the normalized Type 2 Chebyshev low-pass filter for N = 3 with a stopband
ripple of 40 dB is given by
The following MATLAB Program 5.3 can be used to form the Type2 Chebyshev
normalized transfer function for a given order and stopband ripple.
Program 5.3 Analog Type 2 Chebyshev Low-Pass Filter Normalized Transfer Function
N=input(‘enter order of the filter’);
Rs=input(‘enter stopband attenuation in dB’);
[z,p,k] = cheb2ap(N,Rs);% determines poles and zeros
disp(‘Poles are at’);disp(p);
[num,den]=zp2tf(z,p,k);
%Print coefficients in powers of s
disp(‘Numerator polynomial’);disp(num);
disp(‘Denominator polynomial’);disp(den);
The normalized Type 2 Chebyshev transfer functions generated from the above program
for typical values of N with a stopband ripple of 40 dB are tabulated in Table 5.3.
Table 5.3 List of normalized Type 2 Chebyshev transfer functions for stopband ripple = 40 dB
Order N H N (s)
The typical magnitude response of a Type 2 Chebyshev low-pass filter for N = 4 and 7
with 20 dB stopband ripple is shown in Figure 5.11. From this figure, it is seen that Type 2
Chebyshev low-pass filter exhibits monotonicity in the passband and equiripple in the
stopband.
Figure 5.11 Magnitude response of typical Type 2 Chebyshev low-pass filter with 20 dB stopband ripple
Example 5.3 Design a Type 2 Chebyshev low-pass filter for the specifications given in
Example 5.1.
Solution The order N is chosen as 3, as in Example 5.2, since the equation for order
finding is the same for both Type 1 and Type 2 Chebyshev filters. The normalized Type 2
Chebyshev low-pass filter for N = 3 with a stopband ripple of 40 dB has already been
found earlier and is given by
For Ω s = 10,000π, the corresponding transfer function can be obtained by substituting
s = (s/Ω s ) = (s/10000π) in the above expression for H N (s). Thus, the required filter
transfer function is
(5.43)
where U N (x) is the Jacobian elliptic function of order N and ε is a parameter related to the
passband ripple. In an elliptic filter, a constant k, called the selectivity factor, representing
the sharpness of the transition region is defined as
(5.44)
A large value of k represents a wide transition band, while a small value indicates a
narrow transition band.
For a given set of Ω p , Ω s , α p , and α s , the filter order can be estimated using the
formula
(5.45)
(5.47)
(5.48)
The following MATLAB Program 5.4 can be used to form the elliptic normalized
transfer function for given filter order and passband ripple and stopband attenuation. The
normalized passband edge frequency is set to 1.
Program 5.4 Analog Elliptic Low-Pass Filter Normalized Transfer Function
N=input(‘enter order of the filter’);
Rp=input(‘enter passband ripple in dB’);
Rs=input(‘enter stopband attenuation in dB’);
[z,p,k] = ellipap(N,Rp,Rs)% determines poles and zeros
disp(‘Poles are at’);disp(p);
[num,den] =zp2tf(z,p,k);
%Print coefficients in powers of s
disp(‘Numerator polynomial’);disp(num);
disp(‘Denominator polynomial’);disp(den);
The normalized elliptic transfer functions generated from the above program for typical
values of N and stopband ripple of 40 dB are tabulated in Table 5.4.
Table 5.4 List of normalized elliptic transfer functions for passband ripple = 1 dB and stopband ripple = 40 dB
Order N H N (s)
4
5
The magnitude response of a typical elliptic low-pass filter is shown in Figure 5.12,
from which it can be seen that it exhibits equiripple in both the passband and the stopband.
Figure 5.12 Magnitude response of typical elliptic low-pass filter with 1 dB passband ripple and 30 dB stopband ripple
Example 5.4 Design an elliptic analog low-pass filter for the specifications given in the
Example 5.1.
Solution
and
and hence
Choose N = 3. Then, for N = 3, a passband ripple of 1 dB, and a stopband ripple of
40dB, the normalized elliptic transfer function is as given in Table 5.4. For Ω p = 2000Π,
the corresponding transfer function can be obtained by substituting s = (s/Ω p ) = (s/2000π)
in the expression for H N (s). Thus, the required filter transfer function is
(5.49)
(5.50)
The magnitude responses of a third-order Bessel filter and Butterworth filter are
shown in Figure 5.13 and the phase responses of the same filters with the same order are
shown in Figure 5.14. From these figures, it is seen that the magnitude response of the
Bessel filter is poorer than that of the Butterworth filter, whereas the phase response of the
Bessel filter is more linear in the passband than that of the Butterworth filter.
Figure 5.13 Magnitude response of a third-order Bessel filter and Butterworth filter
Figure 5.14 Phase response of a third-order Bessel filter and Butterworth filter
The following MATLAB program is used to generate the magnitude and phase responses
for these specifications.
Program 5.5 Magnitude and Phase Responses of Analog Filters of Order 8 with a
Passband Ripple of 1 dB and a Stopband Ripple of 35 dB
clear all;clc;
[z,p,k]=buttap(8);
[num1,den1]=zp2tf(z,p,k);[z,p,k]=cheb1ap(8,1);
[num2,den2]=zp2tf(z,p,k);[z,p,k]=cheb2ap(8,35);
[num3,den3]=zp2tf(z,p,k); [z,p,k]=ellipap(8,1,35);
[num4,den4]=zp2tf(z,p,k);
omega=[0:0.01:5];
h1=freqs(num1,den1,omega);h2=freqs(num2,den2,omega);
h3=freqs(num3,den3,omega);h4=freqs(num4,den4,omega);
ph1=angle(h1);ph1=unwrap(ph1);
ph2=angle(h2);ph2=unwrap(ph2);
ph3=angle(h3);ph3=unwrap(ph3);
ph4=angle(h4);ph4=unwrap(ph4);
figure(1),plot(omega,20*log10(abs(h1)),‘-’);hold on
plot(omega,20*log10(abs(h2)),‘—’);hold on
plot(omega,20*log10(abs(h3)),‘: ’);hold on
plot(omega,20*log10(abs(h4)),‘-.’);
xlabel(‘Normalized frequency’);ylabel(‘Gain,dB’);axis([0 5
-80 5]);
legend(‘Butterworth’,‘Chebyshev Type 1’,‘Chebyshev Type
2’,‘Elliptic’);hold off
figure(2),plot(omega,ph1,‘-’);hold on
plot(omega,ph2,‘—’);hold on
plot(omega,ph3,‘: ’);hold on
plot(omega,ph4,‘-.’)
xlabel(‘Normalized
frequency’);ylabel(‘Phase,radians’);axis([0 5 -8 0]);
legend(‘Butterworth’,‘Chebyshev Type 1’,‘Chebyshev Type
2’,‘Elliptic’);
The magnitude and phase responses for the above specifications are shown in Figure 5.15.
The magnitude response of Butterworth filter decreases monotonically both in passband
and stopband with wider transition band. The magnitude response of the Chebyshev Type
1 exhibits ripples in the passband, whereas the Chebyshev Type 2 has approximately the
same magnitude response to that of the Butterworth filter. The transition band of both the
Type 1 and Type 2 Chebyshev filters is the same, but less than that of the Butterworth
filter. The elliptic filter exhibits an equiripple magnitude response both in the passband
and the stopband with a transition width smaller than that of the Chebyshev Type 1 and
Type 2 filters. But the phase response of the elliptic filter is more nonlinear in the
passband than that of the phase response of the Butterworth and Chebyshev filters. If
linear phase in the passband is the stringent requirement, then the Bessel filter is preferred,
but with a poor magnitude response.
Figure 5.15 A comparison of various types of analog low-pass filters : (a) magnitude response and (b) phase response
Another way of comparing the various filters is in terms of the order of the filter
required to satisfy the same specifications. Consider a low-pass filter that meets the
passband edge frequency of 450 Hz, stopband edge frequency of 550 Hz, passband ripple
of 1 dB, and stopband ripple of 35 dB. The orders of the Butterworth, Chebyshev Type 1,
Chebyshev Type2, and elliptic filters are computed for the above specifications and listed
in Table 5.5. From this table, we can see that elliptic filter can meet the specifications with
the lowest filter order.
Table 5.5 Comparison of orders of various types of filters
Filter Order
Butterworth 24
Chebyshev Type 1 9
Chebyshev Type 2 9
Elliptic 5
Figure 5.16 Low-pass to low-pass frequency transformation . (a) Prototype Low-pass filter frequency response. (b)
Low-pass filter frequency response
Also, the transfer function for these filters is related to the corresponding
normalized low-pass transfer function H N (s) by
(5.51b)
(5.51c)
where is the cutoff frequency of the desired Butterworth filter and is given by Eq.
(5.19). For similar reasons, the transfer function for the Type 2 Chebyshev filter is
related to the normalized transfer function H N (s) by
(5.51d)
Figure 5.17 Low-pass to high-pass frequency transformation . (a) Prototype low-pass filter frequency response. (b)
High-pass filter frequency response
Let the passband edge frequencies of the prototype low-pass and the desired high-pass
filters be Ω p = 1 and , as shown in Figure 5.17. The transformation from prototype low
(5.52a)
and the desired transfer function is related to the low-pass transfer function H N (s)
by
(5.52b)
The above equations (5.52a) and (5.52b) hold for all filters except for Butterworth and
Type 2 Chebyshev filter. For Butterworth
(5.53a)
(5.53b)
(5.53c)
and
(5.53d)
Example 5.5 Design a Butterworth analog high-pass filter for the following
specifications:
Passband edge frequency: 30.777 Hz
Stopband edge frequency: 10 Hz
Passband ripple: 1 dB
Stopband ripple: 20 dB
Solution For the prototype analog low-pass filter , we have
Substituting these values in Eq. (5.23), the order of the filter is given by
Hence, we choose N = 3. From Table 5.1, the third-order normalized Butterworth low-
pass filter transfer function is given by
From the above transfer function, the analog transfer function of the high-pass filter
can be obtained by substituting
Example 5.6 Design a Butterworth analog high-pass filter for the specifications of
Example 5.5 using MATLAB
Solution The following MATLAB code fragments can be used to design H HP (s)
[N,Wn]=buttord(1,3.0777,1,20,‘s’);
[B,A]=butter(N,Wn,‘s’);
[num, den]=1p2hp(B,A,3.0777);
The transfer function H LP (s) of the analog low-pass filter can be obtained by displaying
numerator and denominator coefficient vectors B and A and is given by
The transfer function H HP (s) of the analog high-pass filter can be obtained by
displaying numerator and denominator coefficient vectors num and den and is given by
the lower stopband edge frequency, and the upper stopband edge frequency of the
desired band-pass filter. Let us denote by B p the bandwidth of the passband and by
the geometric mean between the passband edge frequencies of the band-pass filter , i.e.,
(5.54a)
(5.54b)
Figure 5.18 Low-pass to band-pass frequency transformation . (a) Prototype low-pass filter frequency response. (b)
Band-pass filter frequency response
(5.55)
and
(5.57)
In order to satisfy the stopband requirements and to have symmetry of the stopband
edges in the low-pass filter, we choose Ω s to be themin{|A 1|, |A 2|}. Thus, the spectral
transformation (5.55) leads to the following design equations for the normalized low-pass
filter (except in the case of the Type 2 Chebyshev filter)
(5.58a)
where A 1 and A 2 are given by (5.56) and (5.57), respectively, and the desired high-
pass transfer function can be obtained from the normalized low-pass transfer
function H N (s) using (5.55). In the case of the Type 2 Chebyshev filter, the equation
corresponding to (5.58a) is
(5.58b)
Example 5.7 Design a Butterworth IIR digital band-pass filter for the following
specifications:
Lower passband edge frequency: 41.4 Hz
Upper passband edge frequency: 50.95 Hz
Lower stopband edge frequency: 7.87 Hz
Upper stopband edge frequency: 100 Hz
Passband ripple: 2 dB
Stopband ripple: 10 dB
Solution We have
For the prototype analog low-pass filter , Ω p = 1, Ω s = min {|A 1|, |A 2|} = 8.26; α p
= 2 dB α s = 10 dB
Substituting these values in Eq. (5.23), the order of the filter is given by
Let us choose N = 1
The transfer function of the first-order normalized Butterworth low-pass filter is given
by
To arrive at the analog transfer function of the band-pass filter , variable s in the above
normalized transfer function is to be replaced by
Example 5.8 Design a band-pass Butterworth filter for the specifications of Example 5.7
using MATLAB
Solution The following MATLAB code fragments can be used to design H BS (s):
Bandwidth = bw = 50.95–41.4 = 9.55; .
[N,Wn]=buttord(1,8.26,2,10,‘s’);
[B,A]=butter(N,Wn,‘s’);
[num, den]=1p2bp(B,A, 45.9271,9.55);% [num, den]=1p2bp(B,A,
Ω o , bw);
The transfer function H LP (s) of the analog low-pass filter can be obtained by displaying
numerator and denominator coefficient vectors B and A. It is given by
The transfer function H BP (s) of the analog band-pass filter can be obtained by
displaying numerator and denominator coefficient vectors num and den. It is given by
(5.59)
where is the geometric mean between the stopband edge frequencies of the band-stop
filter , i.e.,
(5.60)
Figure 5.19 Low-pass to band-stop frequency transformation. (a) Prototype low-pass filter frequency response. (b)
Band-stop filter frequency response
frequency for the normalized low-pass filter . Now, we transform the lower
stopband edge frequency to the stopband edge frequency Ω s of the normalized low-
pass filter; hence,
(5.61a)
where is the bandwidth of the stopband. Also, the upper stopband edge
frequency is transformed to
(5.61b)
(5.61c)
(5.62a)
and
(5.62b)
(5.63a)
where
(5.63b)
(5.63d)
Example 5.9 Design an analog band-stop Butterworth filter with the following
specifications :
Lower passband edge frequency: 22.35 Hz
Upper passband edge frequency: 447.37 Hz
Lower stopband edge frequency: 72.65 Hz
Upper stopband edge frequency: 137.64 Hz
Passband ripple: 3 dB
Stopband ripple: 15 dB
Solution From Eq. (5.63b), we have
Now using (5.63a), we get the specifications for the normalized analog low-pass filter
to be
Substituting these values in Eq. (5.23), the order of the filter is given by
Solving for Ω c , we get Ω c = 1.1818. The analog transfer function of the low-pass
filter is obtained from H N (s) by substituting
To arrive at the analog transfer function of the band-stop filter , we use, in the above
expression, the low-pass to band-stop transformation given by (5.63c), namely,
to obtain
Example 5.10 Design a band-stop Butterworth filter for the specifications of Example
5.9 using MATLAB
Solution The following MATLAB code fragments can be used to design H BS (s):
Bandwidth = bw = 447.37 – 22.35; .
[N,Wn]=buttord(1,6.5397,3,15,‘s’);
[B,A]=butter(N,Wn,‘s’);
[num, den]=1p2bs(B,A,100,425.02);% [num, den]=1p2bs(B,A, Ω o
, bw);
The transfer function H LP (s) of the analog low-pass filter can be obtained by displaying
numerator and denominator coefficient vectors B and A and is given by
The transfer function H BS (s) of the analog band-stop filter can be obtained by
displaying numerator and denominator coefficient vectors num and den and is given by
(5.64)
with placement of poles as shown in Figure 5.20(a). The magnitude response of the
system with pole locations shown in Figure 5.20(a) is given by
(5.65)
and is shown in Figure 5.20(b), and its phase response is shown in Figure 5.20(c)
Figure 5.20 (a) Pole locations of H(s). (b) Magnitude response. (c) Pole locations of H(s)
(5.66)
with placement of zeros as shown in Figure 5.21(a). The magnitude response of the
system with zeros locations shown in Figure 5.21(a) is given by
(5.67)
and is shown in Figure 5.21(b), and its phase response is shown in Figure 5.21(c)
Figure 5.21 (a) Zero locations of H(s). (b) Magnitude response. (c) Phase response
As such a filter does not have unity gain at zero frequency. The notch will not be
sharp. By placing two poles close to the two zeros on the semicircle as shown in Figure
5.22(a), the notch can be made sharp with unity gain at zero frequency as shown in Figure
5.22(b).
Figure 5.22 (a) Placing two poles close the two zeros on the semicircle. (b) Magnitude response of (a)
5.5 Problems
1. Test the impulse response of an ideal low-pass filter for the following properties:
(i) Real valued
(ii) Even
(iii) Causal
2. Consider the first-order RC circuit shown in the figure below
(i) Determine H(Ω), the transfer function from vs to vc. Sketch the magnitude and
phase of H(Ω).
(a) Draw the corresponding RC circuit and determine H(Ω), the transfer
function from v to vs. Sketch the magnitude and phase of H(Ω).
3. Design a continuous time low-pass filter with the following transfer function
If H(Ω) is an ideal band-pass filter , determine for what values of α, it will act as
an ideal band-stop filter .
5. Design an elliptic analog high-pass filter for the specifications of Example 5.5
Further Reading
1. Raut, R., Swamy, M.N.S.: Modern Analog Filter Analysis and Design: a Practical Approach. Springer, WILEY-
VCH Verlag & Co. KGaA, Weinheim, Germany (2010)
2. Antoniou, A.: Digital Filters: Analysis and Design. McGraw Hill Book Co., New York (1979)
3. Parks, T.W., Burrus, C.S.: Digital Filter Design. Wiley, New York (1987)
4. Temes, G.C., Mitra, S.K. (eds.): Modern Filter Theory and Design. Wiley, New York (1973)
5. Vlach, J.: Computerized Approximation and Synthesis of Linear Networks. Wiley, New York (1969)
6. Mitra, S.K.: Digital Signal Processing. McGraw-Hill, New York (2006)
7. Chen, C.T.: Digital Signal Processing, Spectral Computation and Filter Design. Oxford University Press,
NewYork/Oxford, UK (2001)
© Springer International Publishing AG, part of Springer Nature 2018
K. Deergha Rao, Signals and Systems, https://doi.org/10.1007/978-3-319-68675-2_6
Discrete-time signals are obtained by the sampling of continuous-time signals . Digital
signal processing deals basically with discrete-time signals, which are processed by
discrete-time systems. The characterization of discrete-time signals as well as discrete-
time systems in time domain is required to understand the theory of digital signal
processing. In this chapter, time-domain sampling and the fundamental concepts of
discrete-time signals as well as discrete-time systems are considered. First, the sampling
process of analog signals is described. Next, the basic sequences of discrete-time systems
and their classification are emphasized. The input-output characterization of linear time-
invariant (LTI) systems by means of convolution sum is described. Further, sampling of
discrete-time signals is introduced. Finally, the state-space representation of discrete-time
LTI systems is described.
(6.1)
where
(6.1a)
x p (t) is the impulse train with the amplitudes of the impulses equal to the samples of x
a (t) at intervals T, 2T, 3T, … .
(6.2)
A sequence of samples x(n) is obtained from an analog signal x a (t) according to the
relation
(6.3)
In Eq. (6.2), T is the sampling period , and its reciprocal, F T = 1/T is called the
sampling frequency , in samples per second. The sampling frequency F T is also referred
to as the Nyquist frequency.
Sampling Theorem
The sampling theorem states that an analog signal must be sampled at a rate at least twice
as large as highest frequency of the analog signal to be sampled. This means that
(6.4)
where f max is maximum frequency component of the analog signal. The frequency 2f max
is called the Nyquist rate .
For example, to sample a speech signal containing up to 3 kHz frequencies, the required
minimum sampling rate is 6 kHz, that is, 6000 sample per second. To sample an audio
signal having frequencies up to 22 kHz, the required minimum sampling rate is 44 kHz,
that is, 44000 samples per second.
A signal whose energy is concentrated in a frequency band range f L < |f| < f H is often
referred to as a band-pass signal. The sampling process of such signals is generally
referred to as band-pass sampling. In the band-pass sampling process, to prevent aliasing
effect, the band-pass continuous-time signal can be sampled at sampling rate greater than
twice the highest frequency (f H ):
(6.5)
(6.6)
Consider that the highest frequency contained in the signal is an integer multiple of the
bandwidth that is given as
(6.7)
(6.8)
(6.9)
The possible outputs of the quantizer (i.e., the quantization levels) are indicated by
where L stands for number of intervals into which the
signal amplitude is divided. For uniform quantization,
(6.10)
where Δ is the quantizer step size.
The coding process in an ADC assigns a unique binary number to each quantization
level. For L levels, at least L different binary numbers are needed. With word length of n
bits, 2 n distinct binary numbers can be represented. Then, the step size or the resolution of
the A/D converter is given by
(6.11)
Figure 6.3 (a) Quantizer, (b) mathematical model, (c) power spectral density of quantization noise
(6.13)
The effect of the additive quantization noise on the desired signal can be quantified by
evaluating the signal-to-quantization noise (power) ratio (SQNR) that is defined as
(6.14)
(6.15a)
On the other hand, a signal x(n) is called anti-symmetric if it follows the condition
(6.15b)
(6.16)
(i)
(iii)
(iv)
Solution
(i) The value of ω 0 in x(n) is . Since is a rational number, it is periodic
discrete-time sequence. The fundamental period of x(n) is given by For
m = 1, . Hence, is periodic with fundamental period N = 8,
(ii) x(n) = sin 2n is aperiodic because ω 0 N = 2N = 2πm is not satisfied for any integer
value of m in making N to be an integer.
(iii) is periodic and cos 2n is aperiodic. Since the sum of periodic and aperiodic
signals is aperiodic, the signal is aperiodic.
(6.17)
(6.18a)
The signal is referred to as an energy signal if the total energy of the signal satisfies
the condition 0 < E < ∞. It is clear that for a finite energy signal, the average power P is
zero. Hence, an energy signal has zero average power. On the other hand, if E is infinite,
then P may be finite or infinite. If P is finite and nonzero, then the signal is called a power
signal. Thus, a power signal is an infinite energy signal with finite average power.
The average power of a periodic sequence x(n) with a period I is given by
(6.18b)
The energy E is finite and the average power P is zero. Hence, the signal x(n) = a n
u(n) is an energy signal for |a| < 1.
The energy E is infinite and also the average power P is infinite. Hence, the signal
x(n) = a n u(n) is neither an energy signal nor a power signal for |a| > 1.
Example 6.3 Determine whether the following sequences
(i) x(n) = e –n u(n), (ii) x(n) = e n u(n), (iii) x(n) = nu(n), and (iv) x(n) = cosπn u(n)
are energy or power signals or neither energy nor power signals.
Solution
The energy E is finite and the average power P is zero. Hence, the signal x(n) = e −n
u(n) is an energy signal.
(ii) x(n) = e +n u(n). Therefore, E and P are given by
The energy E is infinite and also the average power P is infinite. Hence, the signal
x(n) = e n u(n) is neither an energy signal nor a power signal.
The energy E is infinite and also the average power P is infinite. Hence, the signal
x(n) = nu(n) is neither an energy signal nor a power signal.
The energy E is not finite and the average power P is finite. Hence, the signal x
(n) = cos πnu(n) is a power signal.
(6.19)
where ℜ is an operator.
(6.20)
(6.21)
where a is an arbitrary constant and y 1(n) and y 2(n) are the responses of the system when
x 1(n) and x 2(n) are the respective inputs. Equations (6.20) and (6.21) represent the
homogeneity and additivity properties, respectively.
The above two conditions can be combined into one representing the principle of
superposition as
(6.22)
(i)
(i) The outputs y 1(n) and y 2(n) for inputs x 1(n) and x 2(n) are, respectively, given by
The output y(n) due to an input x(n) = ax 1(n) + bx 2(n) is then given by
(ii) The outputs y 1(n) and y 2(n) for inputs x 1(n) and x 2(n) are given by
The output y(n) due to an input x(n) = ax 1(n) + bx 2(n) is then given by
(iii) The outputs y 1(n) and y 2(n) for inputs x 1(n) and x 2(n), respectively, are given by
The output y(n) due to an input x(n) = ax 1(n) + bx 2(n) is then given by
Hence, the system y(n) = x(n − n 0) is linear.
Time-Invariant Systems
A time-invariant system (shift invariant system) is one in which the internal parameters do
not vary with time. If y 1(n) is output to an input x 1(n), then the system is said to be time
invariant if, for all n 0, the input sequence x 1(n) = x(n − n 0) produces the output sequence
y 1(n) = y(n − n 0), i.e.,
For example, for an input x 1(n) = x(n − n 0), the output y1(n) can be written as
Solution For an inputx 1(n) = x(n − n 0), the output y1(n) of the compressor system can
be written as
From given equation,
Comparing the above two equations, it can be observed that y 1(n) ≠ y(n − n 0). Thus,
the down-sampling system is not time invariant.
Causal System
A system is said to be causal if its output at time instant n depends only on the present and
past input values, but not on the future input values.
For example, a system defined by
is not causal , as the output at time instant n depends on future values of the input. But,
the system defined by
is causal, since its output at time instant n depends only on the present and past values
of the input.
Stable System
A system is said to be stable if and only if every bounded-input sequence produces a
bounded-output sequence. The input x(n) is bounded if there exists a fixed positive finite
value β x such that
(6.23)
Similarly, the output y(n) is bounded if there exists a fixed positive finite value β y such
that
(6.24)
Solution Assume that the input x(n) is bounded such that |x(n)| ≤ β x < ∞ for all n
Hence, it is linear.
(6.25)
If the input to the system is a unit sample sequence (i.e., impulse input δ(n)), then the
system output is called as impulse response and denoted by h(n). If the input to the system
is a unit step sequence u(n), then the system output is called its step response. In the next
section, we show that a linear time-invariant discrete-time system is characterized by its
impulse response or step response.
(6.26)
(6.28)
Let the response of the system due to input δ(n − k) be h k (n), that is,
Then, the system response y(n) for an arbitrary input x(n) is given by
Since δ(n − k) is a time-shifted version of δ(n), the response h k (n) is the time-shifted
version of the impulse response h(n), since the operator is time invariant. Hence,h k
(n) = h(n ‐ k). Thus,
(6.29)
The above equation for y(n) is commonly called the convolution sum and represented
by
(6.29a)
where the symbol * stands for convolution. The discrete-time convolution operates on
the two sequences x(n) and h(n) to produce the third sequence y(n).
Example 6.10 Determine discrete convolution of the following sequences for large value
of n:
Solution
For large n, tends to zero and hence,
Example 6.11 Determine discrete convolution of the following two finite duration
sequences:
Solution The impulse response h(n) = 0 for n < 0; hence the given system is causal ; and
x(n) = 0 for n < 0, therefore the sequence x(n) is causal sequence:
The following example illustrates the above procedure for computation of linear
convolution.
Example 6.12 Find the convolution of the sequences x(n) = {6, −3} and h (n) = {−3, 6,
3}.
Solution Using Eq. (6.19), the linear convolution of x(n) and h(n) is given by
Thus,
(i) Time reversing or reflecting of the sequence h(k) about k = 0 sample to give h (–k).
(ii) Shifting the sequence h(–k) to the right by n samples to obtain h(n – k).
(iii) Forming the product x(k)h(n – k) sample by sample for the desired value of n.
(iv) Summing the product over the index k in y(n) for the desired value of n.
The length of the convolution sum sequence y(n) is given by n = N 1 + N 2 − 1, where
N 1 is length of the sequence x(n) and N 2 is length of the sequence h(n).
Example 6.13 Compute the convolution of the sequences of Example 6.12 using the
graphical method.
Solution The sequences x(n) and h(n) are shown in Figures 6.7.
(6.31a)
(6.31c)
1) The commutative law shows that the output is the same if we interchange the roles of
the input and the impulse response. This is illustrated in Figure 6.10.
2) To interpret the associative law, we consider a cascade of two systems whose impulse
responses are h 1 (n) and h 2(n). Then y 1(n) = x(n) * h(n) if x(n) is the input to the
system with the impulse response h 1(n). If y 1(n) is now fed as the input to the system
with impulse response h 2(n), then the overall system output is given by
by associative law
(6.32)
This equivalence is shown in Figure 6.11. Hence, if two systems with impulse
responses h 1(n) and h 2(n) are cascaded, then the overall system response is given by
(6.33)
3) We now consider the distributive law given by (6.31c). This can be easily interpreted
as two LTI systems in parallel and that the overall system impulse response h(n) of
the two systems in parallel is given by
(6.34)
Example 6.15 Consider the system shown in Figure 6.13 with h(n) being real. If y
2(n) = y 1(–n), find the overall impulse response h 1(n) that relates y 2(n) to x(n).
Solution
and the overall impulse response h(n) = {1,5,10,11,8,4,1}. Determine the impulse
response h 1(n).
Solution Let the overall impulse response of the cascaded system be h(n). Hence,
Therefore,
Thus, we obtain
Now, h(n) is nonzero in the interval 0 to 6 and h 3(n) is nonzero in the interval 0 to 2:
Thus,
6.4.5 Stability and Causality of LTI Systems in Terms of the
Impulse Response
The output of a LTI system can be expressed as
we have
(6.35)
It is seen from (6.35) that y(n) is bounded if and only if is bounded. Hence,
the necessary and sufficient condition for stability is that
(6.36)
For a causal system, the output at n = n 0 should not depend on the future inputs.
Hence, in the above equation, h(k) = 0 for k < 0.
Thus, it is clear that for causality of a LTI system, its impulse response sequence
(6.37)
Example 6.17 Check for the stability of the systems with the following impulse
responses :
(i) Ideal delay, h(n) = δ(n − n d ); (ii) forward difference , h(n) = δ(n + 1) − δ(n).
(iii) Backward difference, h(n) = δ(n) − δ(n − 1); (iv) h(n) = u(n).
(v) h(n) = a n u(n), where |a| < 1, and (vi) h(n) = a n u(n), where |a| ≥ 1.
Solution Given impulse responses of the systems, stability of each system can be tested
by computing the sum
In case of (i), (ii), and (iii), it is clear that S < ∞. As such, the systems corresponding to
(i), (ii), and (iii) are stable.
For the impulse response given in (iv), the system is unstable since
(iv) ; hence h(n) ≠ 0 for n < 0; so, the system is not causal.
(v) h(n) = u(n + 1) – u(n), and h(n) ≠ 0 for n < 0; so, the system is not causal .
Example 6.19 Check the following systems for stability :
(i) , (ii) h(n) = u(n + 2) − u(n − 5), (iii) h(n) = 5 n u(−n − 3), (iv)
, and (v)
Solution
(ii) h(n) = u(n + 2) – u(n – 5). The system is stable, since S is finite.
(iv)
Summing |h(n)| over all positive n, we see that S tends to infinity . Hence, the system
is not stable.
(v)
|h(n)| is upper bounded by . Thus, Hence the system is stable .
(6.39)
In addition, if x(n) = 0 for n < 0 and b m = 0 for m > N, then Eq. (6.39) becomes
(6.40)
(6.41)
where a m and b m are constants. An Nth-order recursive difference equation has an infinite
impulse response. Hence, an infinite impulse response (IIR) filter is characterized by a
recursive difference equation.
Example 6.20 An initially relaxed LTI system was tested with an input signal x
(n) = u(n) and found to have a response as shown in Table 6.1.
Table 6.1 Response of an LTI system for an input x(n) = u(n)
(i) From Table 6.1, it can be observed that the response y(n) for an input x (n) = u (n) is
given by
For an input x(n) = u(n)−u(n−1), the response of an LTI system is the impulse response
h(n) given by
(6.42)
The solution of the difference equation is the output response y(n). It is the sum of two
components which can be computed independently as
(6.43a)
where y c (n) is called the complementary solution and y p (n) is called the particular
solution.
The complementary solution y c (n) is obtained by setting x(n) = 0 in Eq. (6.42). Thus y
c (n) is the solution of the following homogeneous difference equation:
(6.43b)
where a 0 = 1. To solve the above homogeneous difference equation, let us assume that
(6.43c)
where the subscript c indicates the solution to the homogeneous difference equation.
Substituting y c (n) in Eq. (6.43b), the following equation can be obtained:
(6.44)
(6.45)
The above equation is called the characteristic equation , which consists of N roots
represented by λ 1, λ 2, … … , λ N. If the N roots are distinct, then the complementary
solution can be expressed as
(6.46)
where α 1, α 2, … … , α N are constants which can be obtained from the specified initial
conditions of the discrete-time system. For multiple roots, the complementary solution y c
(n) assumes a different form. In the case when the root λ 1 of the characteristic equation is
repeated m times, but λ 2, … … , λ N are distinct, then the complementary solution y c (n)
assumes the form
(6.47)
In case the characteristic equation consists of complex roots λ 1 , λ 2 = a ± jb, then the
complementary solution results in y c (n) = (a 2 + b 2)n/2 (C 1 cos nq + C 2 sin nq), where
q = tan–1 b/a and C 1 and C 2 are constants.
We now look at the particular solution y p (n) of Eq. (6.42) The particular solution y p
(n) is any solution that satisfies the difference equation for the specific input signal x(n),
for n ≥ 0, i.e.,
(6.48)
The procedure to find the particular solution y p (n) assumes that y p (n) depends on the
form of x(n). Thus, if x(n) is a constant, then y p (n) is implicitly a constant. Similarly, if
x(n) is a sinusoidal sequence, then y p (n) is implicitly a sinusoidal sequence and so on.
In order to find out the overall solution, the complementary and particular solutions
must be added. Hence,
(6.49)
Example 6.21 Determine impulse response for the case of x(n) = δ(n) of a discrete-time
system characterized by the following difference equation:
(6.50)
Solution First, we determine the complementary solution by setting x(n) = 0 and y (n) =
λ n in Eq. (6.50), which gives us
Hence, the zeros of the characteristic polynomial λ 2 + 2λ − 3 are λ 1 = −3 and λ 2 = 1.
Therefore, the complementary solution is of the form
(6.51)
For impulse x(n) = δ(n), x(n) = 0 for n > 0 and x(0) = 1. Substituting these relations in
Eq. (6.50) and assuming that y(−1) = 0 and y(−2) = 0, we get
i.e., y(0) = 1. Similarly y(1) + 2y(0) – 3y(–1) = x(1) = 0 yields y(1) = –2.
Thus, from Eq. (6.51), we get
(6.52)
(6.53)
(6.54)
(6.56)
Determine the response y(n), n ≥ 0 when the system input is x(n) = (–1) n u(n) and the
initial conditions are y(−1) = 1 and y(−2) = −1.
Solution For the given difference equation, the total solution is given by
First, determine the complementary solution by setting x(n) = 0 and y(n) = λ n in Eq.
(6.56); this gives
K(–1) n u(n) – 2K(–1)n–1 u(n – 1) + K(–1)n–2 u(n – 2) = (–1) n u(n) – (–1)n–1 u(n – 1)
For n = 2, the above equation becomes K + 2K + K = 2; .
Using the initial conditions y(–1) = 1 and y(–2) = –1, we get y(0) = 4.
Then, for n = 1, from Eq. (6.56), we get y(1) = 5. Thus, we get from Eq. (6.57)
Solving these two equations, we arrive at α 1 = (7/2) and α 2 = 2. Thus, the response of
the system for the given input is
(6.58)
where b and a are the coefficient vectors of difference equation describing the system, x is
the input data vector, and y is the vector generated assuming zero initial conditions. The
following example illustrates the computation of the impulse and step responses of an LTI
system .
Example 6.24 Determine the impulse and step responses of a discrete-time system
described by the following difference equation:
Solution Program 6.2 is used to compute and plot the impulse and step responses, which
are shown in Figure 6.15a and b, respectively.
Figure 6.15 (a) Impulse response and (b) step response for Example 6.24
(6.59)
(6.60)
Eq. (6.60) implies that the output of an up-sampler can be obtained by inserting (L – 1)
equidistant zero-valued samples between two consecutive samples of the input sequence
x(n), i.e., x e (n) is identical to the sequence obtained from x a (t) with a sampling period
T’ = T/L. For example, if x e (n) = {2,1,4,–2, … .}, then x e (n) = {2,0,0,0,1,0,0,0,4,0,0,0,–
2,0,0,0, …} for L = 4, i.e., L−1 = 3 zero-valued samples are inserted in between the
samples of x(n) to get x e (n).
(6.61)
where y(n) is the system output and ℧(n) is the system input.
Define the following useful set of state variables:
(6.62)
and
(6.63b)
(6.64a)
(6.64b)
(6.65)
(6.66a)
(6.66b)
where
Solution The order of the differential equation is three. We have to choose three state
variables:
Let x 1(n) = y(n − 3), x 2(n) = y(n − 2), x 3(n) = y(n − 1)
Then
(6.67a)
(6.67b)
where
6.8 Problems
(ii)
(iii)
(iv)
(v) x(n) = 6(−1) n
(vi)
(vii)
(viii)
(ix)
(xi)
(xii)
(xiii)
3. Determine if the following discrete-time signals are even, odd, or neither even nor
odd:
(i)
(ii)
(iii)
(iv)
(i) y(n) = 5nx 2(n). (ii) y(n) = x(n)sin2n. (iii) y(n) = e –n x(n + 3)
6. Check for stability and causality of a system for the following impulse responses:
(i) (ii)
7. Determine if the following signals are periodic, and if periodic, find its period:
(ii) sin n (b) e jπn/3 (c)
9. Determine the convolution of the sum of the two sequences x 1(n) and x 2(n), if x
n
1(n) = x 2(n) = c u(n) for all n, where c is a constant.
10. Determine the impulse response (i.e., when x(n) = δ(n) of a discrete-time system
characterized by the following difference equation:
11. A discrete-time system is characterized by the following difference equation:
Determine the step response of the system, i.e., x(n) = u(n), given the initial conditions
y(-1) = 1 and y(-2) = -1.
Determine the response of the system for x(n) = nu(n) and initial conditions y (-1) = 1
and y(-2) = 0.
13. Determine the response of the system described by the following difference
equation:
1. Using the function impz, write a MATLAB program to determine the impulse
response of a discrete-time system represented by
2. Write a MATLAB program to illustrate down-sampling by an integer factor of 4 of a
sum of two sinusoidal sequences, each of length 50, with normalized frequencies of
0.2 Hz and 0.35 Hz.
Further Reading
1. Linden, D.A.A.: Discussion of sampling theorem. Proceedings of the IRE. 47, 1219–1226 (1959)
[Crossref]
2. Proakis, J.G., Manolakis, D.G.: Digital Signal Processing Principles, Algorithms and Applications, 3rd edn. Prentice-
Hall, India (2004)
3. Crochiere, R.E., Rabiner, L.R.: Multirate Digital Signal Processing. Prentice-Hall, Englewood Cliffs (1983)
4. Hsu, H.: Signals and Systems, Schaum’s Outlines, 2nd edn, Mc Graw Hill, New York (2011)
5. Mandal, M., Asif, A.: Continuous and Discrete Time Signals and Systems. Cambridge, UK; New York: Cambridge
University Press, (2007)
© Springer International Publishing AG, part of Springer Nature 2018
K. Deergha Rao, Signals and Systems, https://doi.org/10.1007/978-3-319-68675-2_7
This chapter discusses the transform domain representation of discrete-time sequences by
discrete-time Fourier series (DTFS) and discrete-time Fourier transform (DTFT) in which
a discrete-time sequence is mapped into a continuous function of frequency. We first
obtain the discrete-time Fourier series (DTFS) expansion of a periodic sequence. The
periodic convolution and the properties of DTFS are discussed. The Fourier transform
domain representation of discrete-time sequences are described along with the conditions
for the existence of DTFT and its properties. Later, the frequency response of discrete-
time systems, frequency domain representation of sampling process, and reconstruction of
band-limited signals from its samples are discussed.
(7.1a)
It can easily be verified from Eq. (7.1a) that x(n) = x(n + N ). Also, we know that there
are only N distinct values for ej2πkn/N, corresponding to k = 0, 1, … . N − 1, these being 1,
ej2πn/N, …, ej2πn(N − 1)/N . Hence, we may rewrite (7.1a) as
(7.1b)
It should be noted that the summation can be taken over any N consecutive values of k.
Eq. (7.1b) is called the discrete-time Fourier series (DTFS) of the periodic sequence x(n)
and a k as the Fourier coefficients. We will now obtain the expression for the Fourier
coefficients a k . It can easily be shown that {ej2πkn/N} is an orthogonal sequence satisfying
the relation
(7.2)
Now, multiplying both sides of (7.1b) by e−j2πln/N and summing over n between 0 and
(N − 1), we get
Hence,
(7.3)
It is common to associate the factor (1/N ) with x(n) rather than a k . This can be done
by denoting Na k by X(k); in such a case, we have
(7.4)
(7.5)
It is easily seen that X(k + N ) = X(k), that is, the Fourier coefficient sequence X(k) is
also periodic of period N. Hence, the spectrum of a signal x(n) that is periodic with period
N is also a periodic sequence with the same period. It is also noted that since the Fourier
series of a discrete periodic signal is a finite sequence, the series always converges, and
the Fourier series gives an exact alternate representation of the discrete sequence x(n).
(7.6)
Multiplication x1(n)x2(n)
x*(n) X*(−k)
x*(−n) X*(k)
Symmetry properties
Example 7.1 Determine the Fourier series representation for the following discrete-time
signal :
Solution
X(0) = X(1) = X(2) = X(4) = X(5) = X(6) = X(8) = X(9) = X(10) = X(11) = X(12) = X(14) =
X(3) = , X(7) =
Example 7.2 Discrete-time signal x(n) is periodic of period 8, and x(n) = n for 0 ≤ n ≤ 7.
Solution The sequence is periodic with period N = 8.
X(2) = −4 + 4 j X(6) = −4 − 4 j
(7.7)
(7.8)
(7.9)
Hence, for uniform convergence of X(e jω ), x(n) must be absolutely summable, i.e.,
(7.10)
Then
(7.11)
guaranteeing the existence of X(e jω ), for all values of ω. Consequently, Eq. (7.10) is
only a sufficient condition for the existence of the DTFT, but is not a necessary condition .
(7.12a)
(7.12b)
Linearity If x1(n) and x2(n) are two sequences with Fourier transforms X 1 (e jω ) and
X2(e jω ), then the Fourier transform of a linear combination of x1(n) and x2(n) is given by
(7.13)
(7.14)
Time Shifting If x(n) is a sequence with Fourier transform X(e jω ), then the Fourier
transform of the delayed sequence x(n − k), where k an integer, is given by
(7.15)
(7.16)
Convolution Theorem If x1(n) and x2(n) are two sequences with Fourier transforms
X1(e jω )and X2(e jω ), then the Fourier transform of the convolution of x1(n) and x2(n) is
given by
(7.18)
Hence, convolution of two sequences x1(n) and x2(n) in the time domain is equal to the
product of their frequency spectra. In the above equation, since X1(e jω ) and X2(e jω ) are
periodic in ω with period 2π, the convolution is a periodic convolution.
Windowing Theorem If x(n) and w(n) are two sequences with Fourier transforms X (e
jω ) and W(e jω ), then the Fourier transform of the product of x(n) and w(n) is given by
(7.19)
(7.20a)
is given by
(7.20b)
which is called the cross energy density spectrum of the signals x1(n) and x2(n).
Parseval’s Theorem If x(n) is a sequence with Fourier transform X(e jω ), then the
energy E of x(n) is given by
(7.21)
(7.22)
Thus,
(7.23)
Frequency shifting
Using the definitions of DTFT pair given by (7.7) and (7.8), we may establish the
DTFT pairs for some useful functions. These are given in Table 7.3
Table 7.3 Some useful DTFT pairs
x(n) DTFT
δ(n) 1
(7.24a)
Similarly, the DTFT of the complex conjugate sequence x*(n) can be expressed as
(7.24b)
(7.26)
where
(7.27)
and
(7.28)
(7.29)
where X e (e jω ) and X o (e jω ) are the DTFTs of x e (n) and x o (n), respectively. Using
Eqs. (7.7), (7.25), and (7.27), X e (e jω ) can be expressed as
(7.30)
In a similar way, using Eqs. (7.7), (7.25), and (7.28), Xo(e jω ) can be written as
(7.31)
A complex sequence x(n)can be decomposed into a sum of its real and imaginary parts
as
(7.32)
where
(7.33)
and
(7.34)
(7.35)
(7.36)
The above properties of the DTFT of a complex sequence are summarized in Table
7.4.
Table 7.4 Some properties of DTFT of a complex sequence
Sequence DTFT
x∗(n) X∗(e−jω)
x∗(−n) X∗(e jω )
x R (n) = Re [x(n)]
jx I (n) = j Im [x(n)]
Re[X(e jω )]
j Im [X(e jω )]
(7.37)
The Fourier transform X(e jω ) is a complex function of ω and can be written as the
sum of the real and imaginary parts as
(7.38)
From Eq. (7.37), the real and imaginary parts of X(e jω ) are given by
(7.39)
and
(7.40)
Since cos(−ωn) = cosωn and sin(−ωn) = −sinωn, we can obtain the following relations
from Eqs. (7.39) and (7.40):
(7.41a)
(7.41b)
indicating that the real part of DTFT is an even function of ω, while the imaginary part
is an odd function of ω. Thus,
(7.42)
In polar form, X(e jω ) can be written as
(7.43)
where
(7.44)
and
(7.45)
Using the above relations, it can easily be seen that |X (e jω )| is an even function of ω,
whereas the function θ(ω)is an odd function of ω.
Now, the DTFT of x e (n), the even part of the real sequence x(n) is given by
(7.46)
Thus, the DTFT of even part of a real sequence is the real part of X (e jω ).
Similarly, the DTFT of x o (n), the odd part of the real sequence x(n), is given by
(7.47)
X I (e jω ) = − X I (e−jω)
X(e jω ) = X∗(e−jω)
|X(e jω )| = |X(e−jω)|
∠X(e jω ) = − ∠ X(e−jω)
Example 7.3 A causal LTI system is represented by the following difference equation:
(i) Find the impulse response of the system h(n), as a function of parameter a.
(i) Given
(ii) Now,
Solution Taking Fourier transform on both sides of given difference equation, we get
For m = 2,
Using the differentiation property of DTFT, the Fourier transform of na n u(n) is given
by
Using linearity property of the DTFT , the Fourier transform of x(n) is denoted by
For m = 3,
Using the differentiation and linearity properties of DTFT, the Fourier transform of
x(n) is given by
In general, for m = k, the Fourier transform of x(n) is given by
Example 7.6 Let G1(e jω ) denote the DTFT of the sequence g1(n) shown in Figure
7.1(a). Express the DTFT of the sequence g2(n) in Figure 7.1b in terms of G1(e jω ). Do not
evaluate G1(e jω ).
Example 7.7 Evaluate the inverse DTFT of each of the following DTFTs:
(a) (b)
Solution
(a)
From Table 7.3,
Hence,
(b)
From Example 7.5,
For m = 2,
Then
If the DTFT of x(n) is X(e jω ), calculate the following functions without computing
X(e jω ).
(a) X(e j0 )
From the definition of Fourier transform ,
(b) X(e jπ )
From the definition of Fourier transform,
(c)
From the definition of inverse Fourier transform,
Hence,
(d)
From the definition of Parseval’s theorem ,
Hence,
(e)
From differentiation property and Parseval’s theorem of DTFT,
Example 7.9
(a) The Fourier transforms of the impulse responses , h1(n) and h2 (n), of two LTI
systems are as shown in Figure 7.3. Find the Fourier transform of the impulse
response of the overall system , when they are connected in cascade.
Figure 7.3 (a) Fourier transform of h1(n) (b) Fourier transform of h2 (n)
(b) The Fourier transforms of the impulse responses h1(n) and h2(n) of two LTI systems
are as shown in Figure 7.4. Find the Fourier transform of the overall system, when
they are connected in parallel.
Figure 7.4 (a) Fourier transform of h1(n) (b) Fourier transform of h2(n)
Solution
Then, by the convolution property of the Fourier transform, the Fourier transform of
the impulse response of the cascade system is given by
The Fourier transform of impulse response of the cascade system is shown in Figure
7.5(a).
Figure 7.5 (a) Fourier transform of the impulse response of the cascade system (b) Fourier transform of the impulse
response of the parallel system
Hence, the Fourier transform of impulse response of the cascade system is given by
The Fourier transform of the impulse response of the parallel system is shown in
Figure 7.5(b).
7.3 Frequency Response of Discrete-Time Systems
For an LTI discrete-time system with impulse response h(n) and input sequence x(n), the
output y(n) is the convolution sum of x(n) and h(n) given by
(7.48)
(7.49)
(7.50)
(7.51a)
where
(7.51b)
H(e jω ) is called the frequency response of the LTI system whose impulse response is
h(n), e jωn is an eigenfunction of the system, and the associated eigenvalue is H(e jω ). In
general H(e jω ) is complex and is expressed in terms of real and imaginary parts as
(7.52)
where H R (e jω ) and H I (e jω ) are the real and imaginary parts of H(e jω ), respectively.
Furthermore, due to convolution, the Fourier transforms of the system input and output
are related by
(7.53)
where X(e jω ) and Y(e jω ) are the Fourier transforms of the system input and output,
respectively. Thus,
(7.54)
The frequency response function H(e jω ) is also known as the transfer function of the
system. The frequency response function provides valuable information on the behavior of
LTI systems in the frequency domain. However, it is very difficult to realize a digital
system since it is a complex function of the frequency variable ω. In polar form, the
frequency response can be written as
(7.55a)
where |H(e jω )|, the amplitude response term, and θ(ω), the phase-response term, are
given by
(7.55b)
(7.55c)
(7.56a)
(7.56b)
The above equation can be rewritten as
(7.57a)
It can be clearly seen that the above equation expresses the phase response as a time
delay in seconds which is called as phase delay and is defined by
(7.57b)
(7.57c)
Example 7.10 Determine the magnitude and phase response of a system whose impulse
response is given by
ω 0 π 2π
Figure 7.6 (a) Magnitude and (b) phase responses of h(n) of Example 7.10
Example 7.11 Compute the magnitude and phase responses of the impulse responses
given in Figure 7.7, and comment on the results.
Figure 7.7 (a) Impulse response of h 1 (n) (b) impulse response of h 2 (n)
Solution Since h 1 (n) is an even function of time, it has a real DTFT indicating that the
phase is zero, that is, the phase is a horizontal line; h 2 (n) is the right-shifted version of h 1
(n). Hence, from time shifting property of DTFT, the transform of h 2 (n) is obtained by
multiplying the transform of h 1 (n) by e–j2ω. This changes the slope of the phase linearly
and can be verified as follows:
The frequency response of h1(n) is
The magnitude response of H1(e jω ) is
The magnitude and phase responses of h 1 (n) and h 2 (n) are shown in Figure 7.8(a),
(b), (c), and (d). From the magnitude and phase responses of h 1 (n) and h 2 (n), it is
observed that h 1 (n) has zero phase and h 2 (n) has a linear phase response, whereas both h
1 (n) and h 2 (n) have the same magnitude responses.
Figure 7.8 (a) Magnitude response of h1(n). (b) Phase response of h 1 (n). (c) Magnitude response of h 2 (n). (d) Phase
response of h 2 (n)
If 0 < ω < π, then both cos ω/2 and sin ω/2 are positive, and hence the phase is .
If π < ω < 2π, then cos ω/2 is negative, but sin ω/2 is positive; hence the phase is .
Figure 7.9 (a) Magnitude response of h(n) sequence. (b) Phase response of h(n) sequence
Program 7.1
clear;clc;
w=0:0.05:pi;
h=exp(j*w); %set h=exp(jw)
num=0.5+0*h.^-1+0.5*h.^-2;
den=1;
%Compute the frequency responses
H=num/den;
%Compute and plot the magnitude response
mag=abs(H);
figure(1),plot(w/pi,mag);
ylabel(‘Magnitude’);xlabel(‘\omega/\pi’);
%Compute and plot the phase responses
ph=angle(H)*180/pi;
figure(2),plot(w/pi,ph);
ylabel(‘Phase, degrees’);
xlabel(‘\omega/\pi’)
Example 7.14 Determine the magnitude and phase responses of a system described by
the following difference equation:
Program 7.2
clear;close all;
num=[0.0534 -0.0009 -0.0009 0.0534];% numerator coefficients
den=[1 -2.1291 1.7834 -0.5435];% denominator coefficients
w=0:pi/255:pi;
%Compute the frequency responses
H=freqz(num,den,w);
%Compute and plot the magnitude response
mag=abs(H);
figure(1),plot(w/pi,mag);
ylabel(‘Magnitude’);xlabel(‘\omega/\pi’);
%Compute and plot the phase responses
ph=angle(H)*180/pi;
figure(2),plot(w/pi,ph);
ylabel(‘Phase, degrees’);xlabel(‘\omega/\pi’);
The frequency response shown in Figure 7.10 characterizes a low-pass filter with
nonlinear phase.
Example 7.15 Determine the magnitude and phase responses of a system described by
the following difference equation:
(a) Find the frequency response H(e jω ) and group delay grd [H(e jω )] of the system.
(b) Determine the difference equation of a new system such that the frequency response
H1(e jω ) of the new system is related to H(e jω ) as H1(e jω ) = H(ej(ω + π)).
Solution (a)
Hence,
Therefore,
As a consequence, the multiplication process gives an impulse train x p (t), which can
be expressed as
(7.59)
Since x a (t) δ(t – nT) = x a (nT) δ(t – nT), the above reduces to
(7.60)
If we now take the Fourier transform of (7.59), and use the multiplication property of
the Fourier transform, we get
(7.61)
where * denotes the convolution in the continuous-time domain and X p (jΩ), X a (jΩ),
and P(jΩ) are the Fourier transforms of x p (t), x a (t), and p(t), respectively. Since p(t) is
periodic with a period T, it can be expressed as a Fourier series
(7.62)
where
Since the convolution of X a (jΩ) with a shifted impulse δ(Ω – kΩ T ) is the shifted
function X a (j(Ω – kΩ T )), the above reduces to
(7.64)
Eq. (7.64) shows that the spectrum of x p (t) consists of an infinite number of shifted
copies of the spectrum of x a (t), and the shifts in frequency are multiples of Ω T ; that is, X
p (jΩ) is a periodic function with a period of Ω T = 2π/T.
(7.65)
(7.66)
Since
and the fact that the DTFT of the sequence x(n) is given by
(7.67)
we obtain
(7.68a)
or equivalently
(7.68b)
(7.69)
(7.70)
(7.71)
where Ω T is called the sampling frequency in radians. Eq. (7.71) is often referred as the
Nyquist condition.
The spectra of the analog signal x a (t) and the impulse train p(t) with a sampling period
T = 2π/Ω T are shown in Figure 7.12(a) and (b), respectively.
Figure 7.12 (a) Spectrum of an analog signal (b) spectrum of the pulse train
Undersampling
If Ω T < 2Ω m , then the signal is undersampled, and the corresponding spectrum X p (jΩ)
is as shown in Figure 7.13. In this figure, the image frequencies centered at Ω T will alias
into the baseband frequencies, and the information of the desired signal is
indistinguishable from its image in the fold-over region.
Figure 7.13 Spectrum of an undersampled signal, showing aliasing (fold-over region). Signals in the fold-over region
are not recoverable
Oversampling
If Ω T > 2Ω m , then the signal is oversampled, and its spectrum is shown in Figure 7.14.
Its spectrum is the same as that of the original analog signal, but repeats itself at every
multiple of Ω T . The higher-order components centered at multiples of Ω T are called
image frequencies.
(7.73)
For a given sequence of samples x(n), we can form an impulse train x p (t) in which
successive impulses are assigned an area equal to the successive sequence values, i.e.,
(7.74)
The n th sample is associated with the impulse at t = nT, where T is the sampling period
associated with the sequence x(n). Therefore, the output x a (t) of the ideal low-pass filter
is given by the convolution of x p (t) with the impulse response h LP (t) of the analog low-
pass filter:
(7.75)
Substituting h LP (t) from Eq.(7.73) in Eq. (7.75) and assuming for simplicity that Ω c
= Ω T /2 = π/T, we get
(7.76)
The above expression indicates that the reconstructed continuous-time signal x a (t) is
obtained by shifting in time the impulse response h LP (t) of the low-pass filter by an
amount nT and scaling it in amplitude by the factor x(n) for all integer values of n in the
range –∞ < n < ∞ and then summing up all the shifted versions.
Example 7.17 Consider the system shown in Figure 7.15(a), where H(e jω ) is an ideal
LTI low-pass filter with cutoff of Π/8 rad/sec, and the spectrum of x a (t) is shown in Figure
7.15(b).
Figure 7.15 (a) Discrete time system (b) spectrum of input x a (t)
(ii)
Figure 7.16
7.6 Problems
1. Obtain the DTFS representation of the periodic sequence shown in Figure P7.1
Figure P7.1 Periodic sequence with period N = 5
4. Let G 1 (e jω ) denote the DTFT of the sequence g 1 (n) shown in Figure P7.2(a).
Express the DTFTs of the remaining sequences in Figure P7.2 in terms of G 1 (e jω ).
Do not evaluate G1(e jω ).
6. A continuous-time signal x a (t) has its spectrum X a (jΩ) as shown in Figure P7.3(a).
The signal x a (t) is input to the system shown in Figure P7.3(b). H(e jω ) in Figure
P7.3(b) is an ideal LTI low-pass filter with a cutoff frequency of (π/2). Sketch the
spectrums of x(n), y(n), and y r (t).
Further Reading
1. Morrison, N.: Introduction to Fourier Analysis. Wiley, New York (1994)
2. Mitra, S.K.: Digital Signal Processing. McGraw-Hill, New York (2006)
3. Oppenheim, A.V., Schafer, W.: Discrete-Time Signal Processing, 2nd edn. Prentice-Hall, Upper Saddle River (1999)
© Springer International Publishing AG, part of Springer Nature 2018
K. Deergha Rao, Signals and Systems, https://doi.org/10.1007/978-3-319-68675-2_8
The DTFT may not exist for all sequences due to the convergence condition, whereas the
z-transform exists for many sequences for which the DTFT does not exist. Also, the z-
transform allows simple algebraic manipulations. As such, the z-transform has become a
powerful tool in the analysis and design of digital systems. This chapter introduces the z-
transform, its properties, the inverse z-transform, and methods for finding it. Also, in this
chapter, the importance of the z-transform in the analysis of LTI systems is established.
Further, one-sided z-transform and the solution of state-space equations of discrete-time
LTI systems are presented. Finally, transformations between continuous-time systems and
discrete-time systems are discussed.
(8.1)
where z is a complex variable. For the existence of the z-transform, Eq. (8.1) should
converge. It is known from complex variables that if is absolutely
convergent, then Eq. (8.1) is convergent. Eq. (8.1) can be rewritten as
(8.2)
(8.3a)
or
(8.3b)
(8.4)
The set of values of z satisfying the above condition is called the region of
convergence (ROC). It is noted that for some sequences r 1 = 0 or r 2 = ∞. In such cases,
the ROC may not include z = 0 or z = ∞, respectively. Also, it is seen that no z-transform
exists if r 1 > r 2.
The complex variable z in polar form may be written as
(8.5)
where r and ω are the magnitude and the angle of z, respectively. Then, Eq. (8.1) can
be rewritten as
(8.6)
When r = 1, that is, when the contour |z| = 1, a unit circle in the z-plane, then Eq. (8.5)
becomes the DTFT of x(n).
Rational z-Transform
In LTI discrete-time systems , we often encounter with a z-transform which is a ratio of
two polynomials in z:
(8.7)
The zeros of the numerator polynomial N(z) are called the zeros of X(z) and those of
the denominator polynomial D(z) as the poles of X(z). The numbers of finite zeros and
poles in Eq. (8.7) are M and N, respectively. For example, the function has a
zero at z = 0 and two poles at z = 1 and z = 2.
(8.8)
It is clear from the above expression that X(z) is convergent for all values of z except
for z = 0, assuming that x(n) is finite. Hence, the ROC is the entire z-plane except for z = 0
and is shown as shaded region in Figure 8.1.
Figure 8.1 ROC of a finite duration causal sequence
Property 3: The ROC for a noncausal finite duration sequence is the entire z -
plane except for z = ∞.
A noncausal finite duration sequence of length N is such that x(n) = 0 for n ≥ 0 and for
n ≤ −N. Hence, X(z) is of the form
(8.9)
It is clear from the above expression that X(z) is convergent for all values of except for
z = ∞, assuming that x(n) is finite. Hence, the ROC is the entire z-plane except for z = ∞
and is shown as shaded region in Figure 8.2.
Property 4: The ROC for a finite duration two-sided sequence is the entire z -
plane except for z = 0 and z = ∞.
A finite duration of length (N 2 + N 1 + 1) is such that x(n) = 0 for n < −N 1 and for
n > N 2, where N 1 and N 2 are positive. Hence, x(z) is of the form
(8.10)
It is seen that the above series is convergent for all values of z except for z = 0 and
z = ∞.
Property 5: The ROC for an infinite duration right-sided sequence is the exterior
of a circle which may or may not include z = ∞.
For such a sequence, x(n) = 0 for n < N. Hence, X(z) is of the form
(8.11)
If N ≥ 0, then the right-sided sequence corresponds to a causal sequence and the above
series converges if Eq (8.3a) is satisfied, that is,
(8.12)
Hence, in this case the ROC is the region exterior to the circle |z| = r 1 or the region
|z| > r 1 including the point at z = ∞. However, if N is a negative integer, say, N = −N 1,
then the series (8.12) will contain a finite number of terms involving positive powers of z.
In this case, the series is not convergent for z = ∞, and hence the ROC is the exterior of the
circle |z| = r 1 but will not include the point at z = ∞.
As an example of an infinite duration causal sequence, consider
(8.13)
Eq. (8.13) holds only if |r 1z −1| < 1. Hence, the ROC is |z| > r 1. The ROC is indicated
by the shaded region shown in Fig. 8.3 and includes the region |z| > r 1. It can be seen that
X(z) has a zero at z = 0 and pole at z = r 1. The zero is denoted by O and the pole by X.
Figure 8.3 ROC of an infinite duration causal sequence
Property 6: The ROC for an infinite duration left-sided sequence is the interior of
a circle which may or may not include z = 0.
For such a sequence, x(n) = 0 for n > N. Hence, X(z) is of the form
(8.14)
If N < 0, then the left-sided sequence corresponds to a noncausal sequence and the
above series converges if Eq. (8.3b) is satisfied, that is,
(8.15)
Hence, in this case, the ROC is the region interior to the circle |z| = r 2 or the region
|z| < r 2 including the point at z = 0.
However, if N is a positive integer, then the series (8.14) will contain a finite number
of terms involving negative powers of z. In this case, the series is not convergent for z = 0,
and hence the ROC is the interior of the circle |z| = r 2 but will not include the point at
z = 0.
As an example of an infinite duration noncausal sequence, consider
(8.16)
Then,
(8.17)
Hence, the ROC is |z| < r 2, that is, the interior of the circle |z| = r 2. The ROC and the
pole and zero of X(z) are shown in Fig. 8.4.
Figure 8.4 ROC of an infinite duration noncausal sequence
(8.18)
and converges in the region r 1 < |z| < r 2, where r 1 and r 2 are given by (8.3a) and
(8.3b), respectively. As mentioned before, the z-transform does not exist if r 1 > r 2.
As an example, consider the sequence
(8.19)
Then,
(8.20)
where the region of convergence is r1< |z| < r2. Thus, the ROC is a ring with a pole on
the interior boundary and a pole on the exterior boundary of the ring, without any pole in
the ROC. There are two zeros, one being located at the origin and the other in the ROC.
The poles and zeros as well as the ROC are shown in Figure 8.5.
Figure 8.5 ROC of an infinite duration two-sided sequence
Example 8.1 Determine the z-transform and the ROC for the following sequence:
Solution respectively.
Linearity If x 1(n) and x 2(n) are two sequences with z-transforms X 1(z) and X 2(z) and
ROCs R 1 and R 2, respectively, then the z-transform of a linear combination of x 1(n) and
x 2(n) is given by
(8.21)
(8.23)
The result concerning the ROC follows directly from the theory of complex variables
concerning the convergence of a sum of two convergent series.
Time Reversal If x(n) is a sequence with z-transform X(z) and ROC R, then the z-
transform of the time reversed sequence x(−n) is given by
(8.24)
(8.25)
Hence,
(8.26)
Since (r 1 < |z| < r 2), we have (1/(r 2) < |z −1| < 1/(r 1)). Thus, the ROC of Z [x(−n)] is
1/R.
Time Shifting If x(n) is a sequence with z-transform X(z) and ROC R, then the z-
transform of the delayed sequence x(n − k), k being an integer, is given by
(8.27)
whose ROC is the same as that of X(z) except for z = 0 if k > 0 and z = ∞ if k < 0
Proof
(8.28)
Substituting m = n − k,
(8.29)
(8.30)
It is seen from Eq. (8.30) that in view of the factor z −k, the ROC of Z [x(n − k)] is the
same as that of X(z) except for z = 0 if k > 0 and z = ∞ if k < 0. It is also observed that in
particular, a unit delay in time translates into the multiplication of the z-transform by z −1.
Scaling in the z-Domain If x(n) is a sequence with z-transform X(z), then Z{a n x (n)} =
X(a −1z) for any constant a, real or complex. Also, the ROC of Z{a n x(n)} is | a|R, i.e., | a|
r 1 < |z| < |a|r 2.
Proof
(8.31)
(8.32)
Since the ROC of X(z) is r 1 < |z| < r 2, the ROC of X(a −1z) is given by r 1 < |a –1z| < r
2, that is,
(8.33)
whose ROC is the same as that of X(z).
Proof From the definition,
(8.34)
(8.35)
(8.36a)
Now, the region of convergence r a < |z| < r b of the sequence nx(n) can be found using
Eqs. (8.3a) and (8.3b).
and
It is to be noted that the ROC of Z[n k x(n)] is also the same as that of X(z).
Convolution of Two Sequences If x 1(n) and x 2(n) are two sequences with z-
transforms X 1(z) and X 2(z), and ROCs R 1 and R 2, respectively, then
(8.37)
(8.38)
(8.39)
(8.40)
(8.41)
Hence,
(8.42)
Since the right side of Eq. (8.42) is a product of the two convergent sequences X 1(z)
and X 2(z) with ROCs R 1 and R 1, it follows from the theory of complex variables that the
product sequence is convergent at least in the region R 1 ∩ R 2. Hence, the ROC of Z[x
1(n) ∗ x 2(n)] is at least R 1 ∩ R 2.
Correlation of Two Sequences If x 1(n) and x 2(n) are two sequences with z-transforms
X 1(z) and X 1(z), and ROCs R 1 and R 2, respectively, then
(8.43)
(8.44)
(8.45)
Since the ROC of X 2(z) is R 2, the ROC of X 2(z −1) is 1/R 2 from the property
concerning time reversal . Also, since the ROC of X 1(z) is R 1, it follows from Eq. (8.45)
that the ROC of is at least R 1 ∩ (1/R 2).
(8.46)
with the ROCs of both X(z) and Z[x *(n)] being the same
Proof The z-transform of x*(n) is given by
(8.47)
(8.48)
In the R.H.S. of the above equation, the term in the brackets is equal to x (z * ).
Therefore, Eq. (8.48) can be written as
(8.49)
It is seen from Eq. (8.49) that the ROC of the z-transform of conjugate sequence is
identical to that of X(z).
Real Part of a Sequence If x(n) is a complex sequence with the z-transform X (z), then
(8.50)
(8.51)
Since the z-transform satisfies the linearity property , we can write Eq. (8.51) as
(8.52)
(8.53)
(8.54)
Thus,
(8.56)
Hence,
(8.57)
Again, since the z-transform satisfies the linearity property , we can write Eq. (8.57) as
(8.58)
Again, it is evident that the ROC of the above is the same as that of X(z). The above
properties of the z-transform are all summarized in Table 8.1.
Table 8.1 Some properties of the z-transform
(8.59)
(8.60)
(8.61)
(8.62)
(8.63)
The ROC is the entire z-plane except for z = 0 if k is positive and for z = ∞ if k is
negative
Example 8.4 Find the z-transform of x(n) = −u(−n − 1)
Solution We know that from Eq. (8.62)
Hence, using the time shifting property
(8.64)
Hence,
(8.65)
we get
Example 8.7 Obtain the z-transform of the following sequence:
Solution
we get
The ROC of X(z) is the same as that of u(n), namely, |z| > 1
Example 8.8 Find the z-transform of x(n) = sin ωn u(n)
Solution
Therefore,
Example 8.10 Find the z-transform of the sequence x(n) = [u (n) − u (n − 5)]
Solution
Example 8.12 Consider a system with input x(n) and output y(n). If its impulse response
h(n) = Ax(L − n), where L is an integer constant, and A is a known constant, find Y(z) in
terms of X(z).
Solution
where
Hence,
A list of some commonly used z-transform pairs are given in Table 8.2
Table 8.2 Some commonly used z-transform pairs
(8.66)
(8.67)
(8.68)
(8.69)
Multiplying the above equation both sides by z n − 1 and integrating both sides on a
closed contour C in the ROC of the z-transform X(z) enclosing the origin, we get
(8.70)
(8.71)
(8.72)
(8.73)
It should be noted that given the ROC and the z-transform X(z), the sequence x(n) is
unique. Table 8.2 can be used in most of the cases for obtaining the inverse transform. We
will consider in Section 8.6 different methods of finding the inverse transform.
(8.74)
where C is a closed contour which encloses the origin and lies in the ROC that is common
to both X 1(v) and .
(8.75)
(8.76)
(8.77)
(8.78)
where C is a contour contained in the ROC common to the ROCs of X 1(v) and .
Hence,
(8.79)
where we have used the result concerning the z-transform of a complex conjugate (see
Table 8.1). That is,
(8.80)
(8.81)
For the energy of real sequences in the z-domain, the above expression becomes
(8.82)
Thus,
(8.83)
(8.84)
(8.85)
We will now consider a few examples of finding the inverse z-transform using the
residue method.
Example 8.14 Assuming the sequence x(n) to be causal, find the inverse z-transform of
Solution Since the sequence is causal, we have to consider the poles of X(z)z n−1 for
only n ≥ 0. For n ≥ 0, the function X(z)z n−1 has only one pole at z = 1 of multiplicity 3.
Thus, the inverse z-transform is given by
It should be mentioned that if x(n) were not causal, then X(z)z n−1 would have had a
multiple pole of order n at the origin, and we would have to find the residue of X(z)z n−1 at
the origin to evaluate x(n) for n < 0.
Example 8.15 If x(n) is causal, find the inverse z-transform of
Solution Since the sequence is causal, we have to consider the poles of X(z)z n−1 for
only n ≥ 0. Hence , we see that for n ≥ 1, X(z)z n−1 has two simple
poles at 0.8 and −0.4. However for n = 0, we have an additional pole at the origin. Hence,
we evaluate x(0) separately by evaluating the residues of . Thus,
For n > 0,
(8.86)
where the order of the polynomial N 1(z) is less than that of the denominator polynomial.
The partial fraction expansion can be now made on N 1(z)/D(z). The inverse z-transform of
the terms in the sum is obtained from the pair (see Table 8.1) and the time-shift
property (see Table 8.2).
Let X(z) be a proper rational function expressed as
(8.87)
(8.88)
Since X(z) is a proper fraction, so will be [X(z)/z]. If all the poles p i are simple, then,
[X(z)/z] can be expanded in terms of partial fractions as
(8.89)
where
(8.90)
(8.91)
(8.92)
Hence,
(8.93)
Then inverse z-transform is obtained for each of the terms on the right-hand side of
(8.91) by the use of Tables 8.1 and 8.2. We will now illustrate the method by a few
examples.
Example 8.16 Assuming the sequence x(n) to be right-sided, find the inverse z-transform
of the following:
Solution The given function has poles at z = a and z = b. Since X(z) is a right-sided
sequence, the ROC of X(z) is the exterior of a circle around the origin that includes both
the poles. Now X(z)/z can be expressed in partial fraction expansion as
Hence,
We can now find the inverse transform of each term using Table 8.2 as
Example 8.17 Assuming the sequence x(n) to be causal, find the inverse z-transform of
the following:
Each term in the above expansion is a first-order z-transform and can be recognized
easily to evaluate the inverse transform as
Example 8.18 Assuming the sequence x(n) to be causal, determine the inverse z-
transform of the following:
Example 8.19 If x(n) is a right-handed sequence, determine the inverse z-transform for
the function:
Solution
Since the sequence is right-handed and the poles of X(z) are located z = 0, 0.5, and 1,
the ROC of X(z) is |z| > 1. Thus, from Table 8.2, we have
Hence,
Solution We obtain the inverse z-transform by long division of the numerator by the
denominator as follows:
implying that
Example 8.22 Find the inverse z-transform of the following:
Comparing the above two expressions, we get x(n), i.e., the inverse z-transform of
X(z) = log (1 + bz −1) to be
(8.95)
Solution Since the region of convergence is |z| < |b|, the sequence is a left-sided
sequence. We can use the long division to obtain z/(z−b) as a power series in z. However,
we will use the binomial expansion.
Hence,
Example 8.25 Using the z-transform, find the convolution of the sequences:
Solution
Step 1: Determine z-transform of individual signal sequences
and
We now take inverse transform of the above and use Table 8.2 to obtain y(n), which is
the impulse response of the system as
Example 8.27 Determine the response y(n), n ≥ 0 of the system described by the second-
order difference equation
or
Hence,
Solution Taking z-transforms on both sides of the above equation, and using the
factZ[δ(n)] = 1, we get
(8.96)
Taking z-transform on both sides of the above equation and using the convolution
property, we get
(8.97)
indicating the z-transform of the output sequence y(n) is the product of the z-
transforms of the impulse response h(n) and the input sequence x(n). The quantities h(n)
and H(z) are two equivalent descriptions of a system in the time domain and z-domain,
respectively. The transform H(z) is called the transfer function or the system function and
expressed as
(8.98a)
Or equivalently,
(8.98b)
It is also given that the output y(n) = 0 for the input x(n) = (−2) n for all n. Since the
function is an eigenfunction for a discrete-time LTI system, the output to this input is
. From this, it can be inferred that H(−2) = 0. Using this in the above transfer
function, the value of a is calculated to be −1.125
Solution The coefficients of the numerator and denominator polynomial can be written
as
numerator = [0 1 -1];
denominator = [8 -6 1];
The following MATLAB statement yields the poles and zeros and gain of the system:
[z,p,gain] = tf2zp (numerator, denominator)
zeros, z = 1
poles, p = [0.500 0.250] and gain = 0.1250
The MATLAB command z-plane (z, p) plots the poles and zeros as shown in Figure
8.6.
(8.99)
where z i and p i are the zeros and poles of H(z). It should be noted that the zeros are either
real or occur in conjugate pairs. The frequency response of the system can be obtained by
letting z = e jω in the transfer function H(z), that is,
Hence,
(8.100)
The contribution of the zeros and poles to the system frequency response can be
visualized from the above expression.
The magnitude of the frequency response can be expressed by
(8.101)
The zeros contribute to pulling down the magnitude of the frequency response,
whereas the poles contribute to pushing up the magnitude of the frequency response. The
size of decrease or increase in the magnitude response depends on how far the zero or the
pole is from the unit circle. A peak in |H(e jω )| appears at the frequency of a pole very
close to the unit circle.
To illustrate this, consider the following example.
Example 8.31 Consider a system with the transfer function
(8.102)
Figure 8.7(a) indicates that the system has zeros of order 2 at z = −1 and two poles on
the imaginary axis close to the unit circle. In the magnitude response of Figure 8.7(b), a
peak occurs at ω = π/2. This can be attributed to the fact that the frequency of the poles is
π/2. The magnitude response is small at high frequencies due to the zeros.
(8.103)
(8.104)
(8.105)
On the unit circle (i.e., |z| = 1), the above expression becomes
(8.106)
Therefore, for a stable system, the ROC of its transfer function H(z) must include the
unit circle. Thus we have the following theorem.
BIBO Stability Theorem
A discrete LTI system is BIBO stable if and only if the ROC of its system function
includes the unit circle, |z| = 1.
We know from Section 6.4.5 that for a discrete LTI system to be causal h(n) = 0 for
n < 0. Thus, the sequence should be right-sided. We also know from Section 8.2 that the
ROC of a right-sided sequence is the exterior of a circle whose radius is equal to the
magnitude of the pole that is farthest from the origin. At the same time, we also know that
for a right-sided sequence, the ROC may or may not include the point z = ∞. But we know
from Section 8.2 that a causal system cannot have a pole at infinity . Thus, in a causal
system, the ROC should include the point z = ∞. Thus, we may summarize the result for
causality by the following theorem:
Causality Theorem
A discrete LTI system is causal if and only if the ROC of its system function is the exterior
of a circle including z = ∞. An alternate way of stating this result is that a system is causal
if and only if its ROC contains no poles, finite or infinite.
Thus the conditions for stability and causality are quite different. A causal system
could be stable or unstable, just as a noncausal system could be stable or unstable . Also, a
stable system could be causal or noncausal just as an unstable system could be causal or
noncausal. However, we can conclude from the above two theorems that a causal stable
system must have a system function whose ROC is |z| = r, where r < 1. Hence, we can
summarize this result as follows.
Condition for a System to Be Both Causal and Stable
A causal LTI system is BIBO stable if and only if all its poles are within the unit circle.
As a consequence, for a LTI system with a system function H(z) to be stable and
causal , it is necessary that the degree of the numerator polynomial in z not exceed that of
the denominator polynomial. As such, an FIR system is always stable, whereas if an IIR
system is not designed properly, it may be unstable.
Example 8.32 Given the system function
Find the various regions of convergence for H(z), and state whether the system is
stable and/or causal in each of these regions. Also, find the impulse response h(n) in each
case.
Solution The system function can be expressed in partial fraction in the form
The system function has two zeros, viz., , and two poles at Hence,
there are three regions of convergence: (i) , (ii) , and (iii) |z| > 4. Let us
consider each of these regions separately.
(i)
In this region, there are no poles including the origin, but has poles exterior to it.
Hence, the system is noncausal. Also, it is an unstable system, since the ROC does not
include the unit circle. By using Table 8.2, we get
(ii)
This region includes the unit circle and hence the system is stable. However, since the
pole |z| = 4 is exterior to this region, it is noncausal, and the corresponding sequence is
two-sided. Again by using Table 8.2, we have
Example 8.33 The rotational motion of a satellite was described by the difference
equation
The poles of the system are at z = 0.5 ± 0.5j as shown in Figure 8.8
Figure 8.8 Poles of Example 8.33
All poles of the system are inside the unit circle. Hence, the system is stable . It is causal
since the output only depends on the present and past inputs.
Example 8.34 Consider the difference equation
(a) Determine the possible choices for the impulse response of the system. Each choice
should satisfy the difference equation. Specifically indicate which choice
corresponds to a stable system and which choice corresponds to a causal system.
(b) Can you find a choice which implies that the system is both stable and causal? If not,
justify your answer.
Solution
(a) Taking the z-transform on both sides and using the shifting theorem, we get
The system function H(z) has a zero of order 2 at z = 0 and two poles at z = 1/3, 2.
Hence, there are three regions of convergence, and thus, there are three possible choices
for the impulse response of the system. The regions are
(i) , (ii) , and (iii) R 3: |z| > 2.
The region R 1 is devoid of any poles including the origin and hence corresponds to an
anti-causal system , which is not stable since it does not include the unit circle. Region R 2
does include the unit circle and hence corresponds to a stable system; however, it is not
causal in view of the presence of the pole z = 2. Finally, the region R 3 does not have any
poles including at infinity and hence corresponds to a causal system; however, since R 3
does not include the unit circle, the system is not stable .
(b) There is no ROC that would imply that the system is both stable and causal .
Therefore, there is no choice for h(n ) which make the system both stable and causal.
(i) Determine the transfer function and discuss the stability of the system.
(ii) Determine the impulse response h(n) and show that it behaves according to the
conclusion drawn from (i).
(iii) Determine the response when x(n) = 10 for n ≥ 0. Assume that the system is initially
relaxed.
Solution
(i) Taking the z-transforms on both sides of the given equation, we get
Hence,
The pole is at z = −1, that is, on the unit circle. So the system is marginally stable or
oscillatory.
(iii) Since
Thus,
or
Therefore,
(8.107)
(8.108)
(8.109)
The pole-zero plot of the above transfer functions are shown in Figure 8.9 (a), (b), and
(c), respectively. The transfer function H 1(z) has a zero at z = −0.4 and a pole at z = −0.3,
and they are both inside the unit circle. Hence, H 1(z) is a minimum-phase function. The
transfer function H 2(z) has a pole inside the unit circle, at z = −0.5, and a zero at z = −2.5,
outside the unit circle. Thus, H 2(z) is a maximum-phase function . The transfer function H
3(z) has two poles, one at z = −0.3 and the other at z = −0.5, and two zeros one at z = −0.4,
inside the unit circle, and the other at z = −2.5, outside the unit circle. Hence, H 3(z) is a
mixed-phase function.
Figure 8.9 Pole-zero plot of (a) a minimum-phase function, (b) a maximum-phase function, and (c) a mixed-phase
function
(8.110)
(8.111)
The recovery system is both stable and causal, since all the poles of the system H I (z)
are inside the unit circle.
(8.112)
(8.113)
or
(8.114)
Hence,
(8.115)
and
(8.116)
Therefore,
(8.117)
Thus,
(8.118)
(8.119)
Thus,
(8.120)
(8.121)
We know from Parseval’s relation that the output energy of a LTI system is given by
(8.122)
(8.123)
Hence,
(8.124)
Thus, the output energy is equal to the input energy for an all-pass system . Hence, an
all-pass system is that it is a lossless system.
Example 8.37 A discrete-time system with poles at z = −0.6 and z = −0.7 and zeros at
z = −1/0.6 and z = −1/0.7 is shown in Figure 8.10. Demonstrate algebraically that
magnitude response is constant.
(8.125)
where H 1(z) is a minimum-phase function as its N poles and (n−m) zeros are inside the
unit circle. Eq. (8.125) can be equivalently expressed as
(8.126)
In the above equation, the factor H 1(z)(1−z −1a 1)(1−z −1a 2)⋯(1−z −1a m ) is also a
minimum-phase function , since |a 1|, |a 2|, …, |a m | are less than 1, the zeros are inside the
unit circle, and the factor is all-pass. Thus, any transfer function
H(z) can be written as
(8.127)
H min(z) has all the poles and zeros of H(z) that are inside the unit circle in addition to
the zeros that are conjugate reciprocals of the zeros of H(z) that are outside the unit circle,
while H ap(z) is an all-pass function that has all the zeros of H(z) that lie outside the unit
circle along with poles to cancel the conjugate reciprocals of the zeros of H (z) that lie
outside the unit circle, which are now contained as zeros in H min(z).
Example 8.38 A signal x(n) is transmitted across a distorting digital channel
characterized by the following system function:
Consider the compensating system shown in Figure 8.11. Find H 1C(z) such that the
overall system function G 1(z) is an all-pass system.
Solution
Then,
is an all-pass system .
(8.128)
which depends only on x(n) for (n ≥ 0). It should be mentioned that the two-sided z-
transform is not useful in the evaluation of the output of a non-relaxed system. The one-
sided transform can be used to solve for systems with nonzero initial conditions or for
solving difference equations with nonzero initial conditions . The following special
properties of X + (z) should be noted.
1. The one-sided transform X + (z) of x(n) is identical to the two-sided transform X(z) of
the sequence x(n)u(n). Also, since x(n)u(n) is always causal, its ROC and hence that
of X + (z) are always the exterior of a circle. Hence, it is not necessary to indicate the
ROC of a one-sided z-transform.
2. X + (z) is unique for a causal signal, since such a signal is zero for n < 0.
3. Almost all the properties of the two-sided transform are applicable to the one-sided
transform, one major exception being the shifting property.
then
(8.129)
However, if x(n) is a causal sequence, then the result is the same as in the case of the
two-sided transform and
(8.130)
Proof By definition,
which proves (8.129). If the sequence x(n) is causal, then the second term on the right
side of the above equation is zero, and hence we get the result (8.130).
Shifting Theorem for X + (z) When the Sequence is Advanced by k
If
then
(8.131)
Proof By definition
(8.132)
The above limit exists only if the ROC of (z−1) X(z) exists.
Proof Since the sequence x(n) is causal, we can write its z-transform as follow
(8.133)
Also
(8.134)
(8.135)
Thus,
Thus,
or
Hence,
It should be noted that the limit exists only if the function (z−1) X(z) has an ROC that
includes the unit circle; otherwise, system would not be stable and the limn→∞x(n) would
not be finite.
Example 8.39 Find the final value of x(n) if its z-transform X(z) is given by
Solution The final value or steady value of x(n) is given by
The result can be directly verified by taking the inverse transform of the given X (z)
Example 8.40 The following facts are given about a discrete-time signal x(n) with X(z)
as its z-transform:
(v) .
Determine X(z) and specify its region of convergence.
Solution It is given that x(n) is real and that X(z) has exactly two poles with one of the
pole at . Since, x [n] is real, the two poles must be complex conjugates of each
other. Thus, the other pole of the system is at . Also, it is given that X(z) has two
zeros at the origin. Therefore, X(z) will have the following form:
for some constant K to be determined. Finally, . Substituting this
in the above equation, we get K = 2. Thus,
Since x[n] is right-sided, its ROC is (note that both poles have magnitude ).
Hence,
Taking the inverse transform, we get
(8.136a)
(8.136b)
(8.137a)
(8.137b)
where
(8.139)
(8.140)
(8.141)
(8.142)
Thus,
(8.143)
(8.144)
Example 8.42 Consider an initially relaxed discrete time system with the following state-
space representation. Find y(n).
Solution
Hence,
8.10 Transformations Between Continuous-Time Systems and
Discrete-Time Systems
The transformation of continuous-time system to discrete-time system arises in various
situations. In this section, two techniques, namely,
(8.145)
Step 3: Sample the impulse response of the analog filter with a sampling period T.
Then, the sampled impulse response h(n) can be expressed as
(8.147)
Step 4: Apply the z-transform on the sampled impulse response obtained in Step 3, to
form the transfer function of the digital filter, i.e., H(z) = Z[h(n)]. Thus, the transfer
function H(z) for the impulse invariance method is given by
(8.148)
This impulse invariant method can be extended for the case when the poles are not
simple.
Example 8.43 Consider a continuous system with the transfer function:
(8.149)
From Eq. (8.149), it is evident that the frequency response of the digital filter is not
identical to that of the analog filter due to aliasing in the sampling process. If the analog
filter is band-limited with
(8.150)
(8.151)
In the above expression, if T is small, the gain of the filter becomes very large. This
can be avoided by introducing a multiplication factor T in the impulse invariant
transformation . In such a case, the transformation would be
(8.152)
(8.154)
Hence, the impulse invariance method is appropriate only for band-limited filters, i.e.,
low-pass and band-pass filters, but not suitable for high-pass or band-stop filters where
additional band limiting is required to avoid aliasing. Thus, there is a need for another
mapping method such as bilinear transformation technique which avoids aliasing.
(8.155)
Then a transfer function H a (s) in the analog domain is transformed in the digital
domain as
(8.156)
(8.157)
We now study the mapping properties of the bilinear transformation. Consider a point
s = −σ + jΩ in the left half of the s-plane. Then, from Eq. (8.157),
(8.158)
Hence, the left half of the s-plane maps into the interior of the unit circle in the z-plane
(see Figure 8.12). Similarly, it can be shown that the right-half of the s-plane maps into the
exterior of the unit circle in the z-plane. For a point z on the unit circle, z = e jω , we have
from Eq. (8.155)
(8.159)
Figure 8.12 Mapping of the s-plane into the z-plane by the bilinear transformation
Thus,
(8.160)
or
(8.161)
showing that the positive and negative imaginary axes of the s-plane are mapped,
respectively, into the upper and lower halves of the unit circle in the z-plane. We thus see
that the bilinear transformation avoids the problem of aliasing encountered in the impulse
invariant method, since it maps the entire imaginary axis in the s-plane onto the unit circle
in the in the z-plane. Further, in view of the mapping, this transformation converts a stable
analog filter into a stable digital filter.
Example 8.44 Design a low-pass digital filter with 3 dB cutoff frequency at 50 Hz and
attenuation of at least 10 dB for frequency larger than 100 Hz. Assume a suitable sampling
frequency .
Solution Assume the sampling frequency as 500 Hz. Then,
Hence, the order of the Butterworth filter is 2. The normalized low-pass Butterworth
filter for N = 2 is given by
The digital transfer function H(z) of the desired filter is now obtained by using the
bilinear transformation (8.157) in the above expression:
8.11 Problems
2. Find the z-transform and ROC of the sequence x(n) tabulated below.
n −2 −1 0 1 2 3 4
x(n) 1 2 3 4 5 6 7
6. Find the z-transform of the following discrete-time signals , and find the ROC for
each.
(i)
(ii)
(iii)
10. Determine the z-transform and the ROC for the following time signals. Sketch the
ROC, poles, and zeros in the z-plane.
(i)
(ii)
11. Find the inverse z-transform of the following, using partial fraction expansions:
(i)
(ii)
(iii)
(iv)
12. Find the inverse z-transform of the following using the partial fraction expansion .
(i)
(ii)
(iii)
13. Determine all the possible signals that can have the following z-transform:
14. Find the stability of the system with the following transfer function :
Specify the ROC of H(z ) and determine h(n) for the following conditions:
17. In the system shown in Figure P8.1, if S 1 is a causal LTI system with system
function,
Determine the system function for a system S 2 so that the overall system is an
all-pass system.
Determine the response when x(n) = u(n ). Assume that the system is initially
relaxed.
20. Using the one-sided z-transform, solve the following difference equation:
Determine the transfer function and pole-zero pattern for the discrete-time
system by using the impulse invariance technique.
22. Design a low-pass Butterworth filter using Bilinear transformation for the following
specifications:
Passband edge frequency: 1000 Hz
Stopband edge frequency: 3000 Hz
Passband ripple: 2 dB
Stopband ripple: 20 dB
Sampling frequency : 8000 Hz
23. Consider an initially relaxed discrete time with the following state-space
representation. Find y(n).
8.12 MATLAB Exercises
1. Write a MATLAB program using the command residuez to find the inverse of the
following by partial fraction expansion:
2. Write a MATLAB program using the command impz to find the inverse of the
following by power series expansion:
3. Write a MATLAB program using the command z-plane to obtain a pole-zero plot for
the following system:
4. Write a MATLAB program using the command freqz to obtain magnitude and phase
responses of the following system:
Further Reading
1. Lyons, R.G.: Understanding Digital Signal Processing. Addison-Wesley, Reading (1997)
2. Oppenheim, A.V., Schafer, R.W.: Discrete-Time Signal Processing, 2nd edn. Prentice-Hall, Upper Saddle River
(1999)
3. Mitra, S.K.: Digital Signal Processing. McGraw-Hill, New York (2006)
4. Hsu, H.: Signals and Systems, Schaum’s Outlines, 2nd edn. McGraw-Hill, New York (2011)
B
Band-pass filter
Band-stop filter
Basic continuous-time signals
complex exponential function
ramp function
real exponential function
rectangular pulse function
signum function
sinc function
unit impulse function
unit step function
Basic sequences
arbitrary
exponential and sinusoidal
unit sample
unit step
Bessel filter
BIBO stability theorem
Bilateral Laplace transform
Bilinear transformation
Block diagram representation
described by differential equations
Butterworth analog low-pass filter
C
Cauchy’s residue theorem
Causal and stable conditions
Causality
Causality for LTI systems
Causality theorem
Characteristic equation
Chebyshev analog low-pass filter
Classification of signals
analog and digital signals
causal, non-causal and anti-causal signals
continuous time and discrete time signals
deterministic and random signals
energy and power signals
even and odd signals
periodic and aperiodic signals
Commutative property
Complex Exponential Fourier Series
Computation of convolution integral using MATLAB
Computation of convolution sum using MATLAB
Computation of linear convolution
graphical method
matrix method
Conjugate of complex sequence
Continuous Fourier Transform
convergence of Fourier transform
Continuous Fourier transform properties
convolution property
differentiation in frequency
differentiation in time
duality
frequency shifting
integration
linearity
modulation
Parseval’s theorem
symmetry properties
time and frequency scaling
time reversal
time shifting
Continuous time signal
complex exponential Fourier series
convergence of Fourier series
properties of Fourier series
trigonometric Fourier series
symmetry conditions
Continuous-time systems
causal system
invertible system
linear systems
memory and memoryless system
stable system
time–invariant system
Convergence of the DTFT
Convolution integral
associative property
commutative property
distributive property
graphical convolution
Convolution of two sequences
Convolution sum
Convolution theorem
Correlation
Correlation of discrete-time signals
Correlation of two sequences
Correlation theorem
D
Direct form I
Direct form II
Discrete-time Fourier series (DTFS)
Fourier coefficients
multiplication
periodic convolution
symmetry properties
Discrete-time Fourier transform (DTFT)
linearity
Discrete time LTI systems in z-domain
Discrete-time signal
Discrete-time signals classification
energy and power signals
finite and infinite length
periodic and aperiodic
right-sided and left-sided
symmetric and anti-symmetric
Discrete-time system characterization
non-recursive difference equation
recursive difference equation
Discrete-time systems classification
causal
linear
stable
time-invariant
Discrete transformation
Distributive property
Down-sampler
E
Elementary operations on signals
Elliptic analog low-pass filter
Energy and power signals
Examples of real world signals and systems
audio recording system
global positioning system
heart monitoring system
human visual system
location-based mobile emergency services system
magnetic resonance imaging
F
Filtering
Final value theorem
Fourier transform
Fourier transform of discrete-time signals
convergence of the DTFT
properties of DTFT
for a complex sequence
for a real sequence
theorems on DTFT
convolution theorem
correlation theorem
differentiation in frequency
frequency shifting
linearity
Parseval’s theorem
time reversal
time shifting
windowing theorem
Frequency division multiplexing (FDM)
Frequency response computation using MATLAB
Frequency response from poles and zeros
Frequency response of continuous time systems
Frequency response of discrete-time systems
computation using MATLAB
Frequency shifting
G
Generation of continuous-time signals using MATLAB
Graphical convolution
H
Half-wave symmetry
High-pass filter
I
Imaginary part of a sequence
Impulse and step responses
Impulse and step responses computation using MATLAB
Impulse response
Impulse step responses
Initial Value Theorem
Input-output relationship
Interconnected systems
Inverse discrete Fourier transform
Inverse Fourier transform
Inverse Laplace transform
partial fraction expansion
partial fraction expansion using MATLAB
partial fraction expansion with multiple poles
partial fraction expansion with simple poles
Inverse system
Inverse z -transform
Cauchy’s residue theorem
modulation theorem
Parseval’s relation
partial fraction expansion
partial fraction expansion using MATLAB
power series expansion
power series expansion using MATLAB
L
Laplace transform
block diagram representation
definition of
existence of
inter connection of systems
properties
convolution in the frequency domain
convolution in the time domain
differentiation in the s-domain
differentiation in the time domain
division by t
final value theorem
initial value theorem
integration
linearity
properties of even and odd functions
shifting in the s-domain
time scaling
time shifting
region of convergence
finite duration signal
left sided signal
right sided signal
strips parallel to the jΩ axis
two sided signal
region of convergence (ROC)
relationship to Fourier transform
representation of Laplace transform in the s-plane
system function
block diagram representation
interconnection of systems
table of properties
transfer Function
unilateral Laplace transform
differentiation property
Linear constant coefficient difference equations
Linear constant-coefficient differential equations
Linearity
Linearity property of the Laplace transform
Low-pass to band-pass
Low-pass filter
LTI discrete-time systems
LTI systems with and without memory
M
Matrix method
Maximum-phase systems
Minimum-phase decomposition
Minimum-phase systems
Mixed-phase systems
Modulation
Modulation and demodulation
Modulation property
Modulation theorem
Multiplexing and demultiplexing
N
Nonperiodic signals
Non-recursive difference equation
Notch filter
O
One-sided z-transform
properties
shifting theorem
final value theorem
solution of linear difference equations
with initial conditions
P
Parseval’s relation
Parseval’s theorem
Partial fraction expansion
Partial fraction expansion using MATLAB
Particular solution
Periodic convolution
Phase and group delays
Poles and zeros
Pole-zero pairing
Pole-zero placement
Power series expansion using MATLAB
Properties of the convolution integral
Properties of the convolution sum
Properties of the impulse function
sampling property
scaling property
shifting property
Proposition
Q
Quantization and coding
Quantization error
R
Rational transfer function
Rational z -transform
Real part of a sequence
Reconstruction of a band-limited signal from its samples
Recursive difference equation
Region of convergence (ROC)
Representation of signals in terms of impulses
S
Sampling
continuous time signals
discrete time signals
frequency domain
Nyquist rate
Sampling frequency
Sampling in frequency-domain
aliasing
over-sampling
sampling theorem
under sampling
Sampling period
Sampling theorem
Scalar multiplication
S-domain
Single-side-band (SSB) AM
Singularity functions
Solution of difference equations
characteristic equation
complementary solution
using MATLAB
particular solution
Solution of linear differential equations using MATLAB
Stability and causality
Stability and causality of LTI systems in terms of the impulse response
Stability for LTI systems
Step and impulse responses
Step response
Systems
all-pass
causal
inverse
maximum-phase
minimum-phase
mixed-phase
stable
T
Theorems on DTFT
Time-invariant
Time reversal
Time scaling
Time shifting
Transfer function
Transformation
U
Under sampling
Unilateral Laplace transform
differentiation property
Unit doublet
Unit impulse response
Unit ramp
Unit sample sequence
Unit step response
Unit step sequence
Unstable system
Up-sampling
W
Windowing theorem
Z
Zero locations
Zero-order hold
Z-transform
definition of
properties of
region of convergence (ROC)
Z-transform properties
conjugate of a complex sequence
convolution of two sequences
correlation of two sequences
differentiation in the z-domain
imaginary part of a sequence
linearity
real part of a sequence
scaling in the z-domain
time reversal
time shifting
Z-transforms of commonly-used sequences
unit step sequence