m3 Problems
m3 Problems
s 1
1. The system function of a causal LTI system is given by H s . Find output y(t) for the
s 2s 2
2
Since L e a t sin t
; here ω = 1, a = 1
s a 2 2
Since system is causal, yt e t sin t ut
So it is unilateral LT. unilateral or one sided LT is based on only positive time (t > 0) portions of a
signal. By working with causal signals, we remove ambiguity inherent in bilateral transform and thus
do not need to consider ROC in unilateral LT (Lu).
2. Find the transfer function of the following (i) an ideal differentiation, (ii) an ideal integration and (iii)
an ideal delay of T seconds.
dx t
(i) Input – output equation for differentiation is yt
dt
Writing the LT by applying differentiation property in time domain and considering it as ideal, we get
Y(s) = S X(s)
We know that Y(s) = H(s) X(s)
Therefore, H(s) = S
(ii) Input – output equation for integration is yt xd
Assuming zero initial state; yt x d
0
Writing the LT by applying integration property in time domain and considering it as ideal, we get
Ys Xs
1
S
We know that Y(s) = H(s) X(s)
Hs
1
S
(iii) Input – output equation is
Y(t) = x(t – T)
Using time shift property and applying LT on both sides, we get
Ys e sT Xs
We know that Y(s) = H(s) X(s)
Hs e sT
3. Consider a series RL circuit with R = 100Ω and L = 10H and output across R is y(t) with input x(t).
(a) Write differential equation relating input and output, (b) determine impulse response,
(c) determine step response.
Ys
1 10 10
s s 10 s s 10
Using partial fraction expansion, we get
10 A B As 10 Bs
s s 10 s s 10 s s 10
Equating ‘s’ terms and constant terms, we get,
A + B = 0 and 10A = 10
A = 1 and B = -1
10 1 1 1 1
s s 10 s s 10 s s 10
Ys
1 1
s s 10
1 1
yt L1 L1
s s 10
yt ut e 10 t ut 1 e 10 t ut
4. Find transfer function of LTI system described by differential equation
d 2 yt dyt dx t
2
3 2 yt 2 3x t .
dt dt dt
Apply LT on both sides, we get
S2 Ys 3SY s 2Ys 2SX s 3Xs
S 2
3S 2 Ys 2S 3Xs
Hs
Ys 2S 3
2
Xs S 3S 2
Causal and stable systems
5. A system has the transfer function Hs
2 1
. Find the impulse response, (a) assuming that the
s3 s2
system is stable and (b) assuming that the system is causal. Can this system be both stable and
causal?
This system has a pole at s = -3 and at s = 2.
(a) If the system is stable, then the pole at s = - 3 contributes a right-sided term to the impulse
response.
2
i.e., L1 2e
3t
u t
s 3
This is given as
1
L1 e u t
2t
s 2
This is the left sided term which makes the system stable.
This is given as
Thus the required impulse response for the system to be stable is given by ht 2e 3 t ut e 2 t u t
(b) If the system is causal, then both poles must contribute right-sided terms to the impulse
response.
Here the pole at s = - 3 contributes a right-sided term to the impulse response.
2
i.e., L1 2e
3t
u t . This is given as
s 3
Thus the required impulse response for the system to be stable is given by ht 2e 3 t ut e 2 t u t .
Note that this system is not stable, since the term e2tu{t) is not absolutely integrable.
(c) The system cannot be both stable and causal because the pole at s = 2 is in the right half of the s-
plane.
d 2 yt dyt dx t
6. A systems described by differential equations 2 10 yt x t 2 . Find the impulse
dt 2 dt dt
response, assuming that the system is (i) stable and (ii) causal.
Taking LT on both sides and assuming ideal conditions, we get
S2 Y(s) – 2SY(s) + 10Y(s) = X(s) + 2SX(s)
(S2 – 2S + 10)Y(s) = (1 + 2)X(s)
Ys 1 2S
2
Xs S 2S 10
1 2S
Hs
S 2 2S 10
b b 2 4ac
Here pole is in the form of S2 – 2S + 10 which gives root at
2a
2 4 402 6j
S 1 3 j
2 2
2 S 1 3 2 S 1
Hs
3
S 1 3 S 1 3 S 12 32
2 2 2 2
S 1
e t cos 3t u t
L
. This gives left sided term.
S 12 32
In former case, i.e., u(t) form limit of integral is from 0 to ∞ while in the later case i.e., u(-t) form,
limit is from -∞ to 0. Hence one will give the –ve of the other.
Similarly e t sin 3t u t
L 3
gives right sided term while
S 12 32
e t sin 3t u t
L 3
gives left sided term.
S 12 32
Hence the system is stable when
ht 2e t cos 3t u t e t sin 3t u t
ROC is ReS1 i.e., left sided which includes –jω axis.
(b) Since poles lies in right half of s – plane, the system is causal if the poles contribute right sided
terms to the impulse response.
Here ROC is ReS1 i.e., right sided.
Inverse system
dyt d 2 x t dx t
7. An LTI system described by the differential equation 2 yt 2
2 3x t . Find the
dt dt dt
transfer function of the inverse system. Does a stable and causal inverse system exist?
By taking the Laplace transform of both sides of the given differential equation, obtaining
SY(s) + 2Y(s) = S2X(S) + 2SX(s) – 3X(s)
(S + 2)Y(s) = (S2 + 2S – 3)X(s)
Ys S 2 2S 3
Xs S 2
S 2 2S 3
Hs
S 2
S 2 S 2
H 1 s
1
2 .
Hs S 2S 3 S 3S 1
The inverse system has poles at s = 1 and s = -3. The pole at s = 1 is in the right half of the s-
plane. Therefore, the inverse system represented by H-1(s) cannot be both stable and causal.
s2
8. For a system with transfer function Hs 2 . Find the zero state response, if the input x(t) is
s 4s 3
e t u t .
Solution:
Ys s2
The given transfer function is Hs 2 .
Xs s 4s 3
The input is xt e t ut .
Xs L e t u t
1
s 1
Zero state response is the response when the initial conditions are neglected.
Ys
s2
A
B
C
s 1s 3 A s 3 B s 12 C
s 12 s 3 s 1 s 12 s 3 s 12 s 3
Put s = -1
1 2 0A 1 3 B 0C
1 2B
1
B
2
Put s = -3
3 2 0A 0 B 3 12 C
1 4C
1
C
4
s 2 s 2 4s 3 A s 3 B s 12 C
Compare the coefficients of s on both sides, we get
1 4A
1
A
4
1 1 1
Ys
A B C 4 2 4
s 1 s 12 s 3 s 1 s 12 s 3
Taking inverse Laplace transform on both sides, we get the zero state response
yt e t u t t e t u t e 3t u t
1 1 1
4 2 4
9. Find the impulse and step response of the following systems.
s3
(a) Hs 2 (b) Hs 2
5
s 4s 5 s 6s 8
Solution:
Ys
(a) The given transfer function is Hs
5
2
Xs s 4s 5
For impulse response, x(t) = δ(t)
Xs 1
1
A
2
Put s = -4, we get
4 3 0 4 2 B
1 2B
1
B
2
1 1
1 1 1 1
Ys
A B 2
2
s 2 s 4 s 2 s 4 2 s 2 2 s 4
Taking inverse Laplace transform on both sides, we get the impulse response
yt e 2 t u t e 4 t u t
1 1
2 2
For step response, x(t) = u(t)
Xs
1
s
s 3 1 s3
Therefore, response Ys Hs Xs 2
s 6s 8 s s s 2s 4
Taking partial fractions, we have
Ys
s3 A
B
C
s 2s 4 A s s 4 B s s 2C
s s 2s 4 s s 2 s 4 s s 2s 4
Put s = 0, we get
3 8A 0 0
3
A
8
Put s = -2, we get
1 0 4B 0
1
B
4
Put s = -4, we get
1 0 0 8C
1
C
8
3 1 1
Ys
A B C 8
4 8
s s 2 s 4 s s 2 s 4
Taking inverse Laplace transform on both sides, we get the step response
yt u t e 2 t u t e 4 t u t
3 1 1
8 4 8
10. Consider an LTI system whose response to the input
xt 2e t e 3t ut is
yt 3e 2t 3e 4t ut . Find the system’s impulse response.
Solution:
The Laplace transforms of x(t) and y(t) are:
Xs L 2e t e 3t ut
2
1
s 3 2 s 1 s 5
s 1 s 3 s 3s 1 s 3s 1
Ys L 3e 2 t 3e 4 t u t
3
3
s 43 s 23 6
s2 s4 s 2s 4 s 2s 4
Hence, the system response is:
6
Ys s 2s 4 6 s 3s 1
Hs
Xs s5 s 5s 2s 4
s 3s 1
A
B
C
s 2s 4A s 5s 4 B s 5s 2 C
s 5 s 2 s 4 s 5s 2s 4
Put s 2, we get
6 2 3 2 1 2 2 2 4A 2 5 2 4 B 2 5 2 2 C
6 0 6B 0
6
B 1
6
Put s = -5, we get
6 5 3 5 1 5 2 5 4A 5 5 5 4 B 5 5 5 2 C
48 3A
A 16
Put s = -4, we get
6 4 3 4 1 4 2 4 4A 4 5 4 4 B 4 5 4 2 C
6 1 3 1 2 C
18
C 9
2
Hs
16 1 9
s5 s 2 s4
Therefore, the impulse response is:
ht 16 e 5t ut e 2t ut 9e 4t ut
3s 2
11. A system has the transfer function Hs . Find the impulse response assuming that the
s 4s 1
system is causal and stable.
Solution:
Given Hs
3s 2
A
B
s 1 A s 4B
s 4s 1 s 4 s 1 s 4s 1
Put s 1, we get
31 2 1 1 A 1 4 B
5B 5
5
B 1
5
Put s 4, we get
3 4 2 4 1 A 4 4 B
10 5A
10
A 2
5
Hs
A B 2 1
s 4 s 1 s 4 s 1
The poles are s = -4 and s = 1
For an LTI system to be stable, its ROC must contain the imaginary axis of s-plane. So in this case,
there would not be any combination which can be stable as well as causal.
Hence, the impulse response, which is stable but not causal, is:
ht 2e 4t ut e t u t ; ROC; 4 Res 1
Similarly, the impulse response, which is causal but not stable, is:
ht 2e 4t ut e t ut ; ROC; Res 1
We know that
1
e at ut
LT
s a
1
t e at ut
LT
s a 2
X
j 3
A
B
j 2 A j 1 B
j 1 j 2 j 1 j 2 j 1 j 2
Put jω = -2 and equate the numerator of both sides
2 3 0 B
B 1
Put jω = -1 and equate the numerator of both sides
1 3 A 0
A2
X
A B 2 1
j 1 j 2 j 1 j 2
Taking inverse Fourier transform, we get the input
xt 2e t ut e 2t ut
2. Using Fourier transform, find the differential equation description for the system having impulse
response ht 3e 3t 2e 2t ut .
Solution:
Given ht 3e 3t 2e 2t ut
Taking Fourier transform on both sides, we get the frequency response (transfer function)
H 3
1
2
1
j 23 j 3 2
j 3 j 2 j 3 j 2
3 j 6 2 j 6 j
j 3 j 2 j 3 j 2
j
j2 5 j 6
Y j
H
X j2 5 j 6
Nyquist rate f N 2f m
2 170 340 Hz
1 1
Nyquist int erval
f N 2 fm
1
sec .
340
(b) Given x(t) = sinc (100πt) + 3 sinc2 (60πt) = x1(t) + x2(t)
sin 100 t sin m1t
x1 t sin c 100 t
100 t m1t
m1 100 rad / sec
sin 60 t sin 60 t sin 60 t
2
x 2 t 3 sin c 60 t 3
2
3
60 t 60 t 60 t
m 2 60 60 120 rad / sec
m m 2 120 The l arg er of m1 and m 2
Nyquist sampling rate f N 2f m
2 60 120 Hz
1 1
Nyquist int erval
f N 2f m
1
120 sec
1 1
(c) Given xt sinc 100π t sinc 50π t
2 3
1 sin 100 t 1 sin 50 t
x t
2 100 t 3 50 t
i.e.,
1 sin m1 t 1 sin m 2 t
2 m1 t 3 m2 t
m1 100 and m2 50
m m1 100 larger of m1 and m2
m 100
Highest frequency component f m 50 Hz
2 2
Therefore, Nyquist sampling rate f N 2f m 2 50 100 Hz
1 1 1
Nyquist interval 10 ms
f N 2f m 100
MODULE III
Laplace transforms analysis of systems
The transfer function of an LTI system is defined as the Laplace transform of the impulse
response. The output of an LTI system is related to the input in terms of the impulse response via the
convolution
y(t)=h(t)*x(t)
Where h(t) and x(t) can be either causal or non-causal. Hence, if we take the bilateral Laplace
transform of both sides of this equation and use the convolution property, then we have
Y(s) = H(s)X(s)
The Laplace transform of the system output is equal to the product of the transfer function and the
Laplace transform of the input.
Ys
Hs
X s
That is, the transfer function is the ratio of the Laplace transform of the output signal to the
Laplace transform of the input signal. This definition applies at values of ‘s’ for which X(s) is nonzero.
Relation between transfer function and differential equation
The relationship between the input and output of an Nth order LTI system is described by the
differential equation:
N dk M dk
ak y t b k xt
k 0 dt k k 0 dt k
We know that if xt est , then y t est Hs . Substituting this, we get
d k st d k st
N
ak
k 0 dt k
M
e Hs bk e
k 0 dt k
d k st
Substitute
k
e s k est
dt
N M
a k s k est Hs bk s k est
k 0 k 0
M k st
bk s e
k 0
Hs
N k st
aks e
k 0
H(s) is a ratio of polynomials in ‘s’ and is thus termed a rational transfer function. The
coefficient of sk in the numerator polynomial corresponds to the coefficient bk of the kth derivative of
x(t). The coefficient of sk in the denominator polynomial corresponds to the coefficient ak of the kth
derivative of y(t). Hence, we may obtain the transfer function of an LTI system from the differential-
equation description of the system. Conversely, we may determine the differential-equation
description of a system from its transfer function.
Causality and stability
Causality - The impulse response of a causal system is zero for t < 0. Therefore, if we know that a
system is causal, the impulse response is determined from the transfer function by using the right-
sided inverse Laplace transforms.
1. In terms of ROC
Since h(t) is right sided signal, its ROC will be of the form R e s σ max . Where max represents
maximum value of real part of all poles in H(s). Sketching the ROC, we get
Thus ROC associated with system function for a causal system is a right half plane. For system
with a rational system function, causality of the system is equivalent to ROC being right half plane to
right of rightmost pole.
2. In terms of location of poles and nature of impulse response
Similarly a pole in the right half side of s-plane R e dk 0 corresponds to Hs
1
Then,
s dk
d t
ht e k . Hence it is an increasing exponential impulse response as shown below.
Stability
1. In terms of ROC
If system is stable, then impulse response is absolutely integrable. This implies that the Fourier
transform exists, and thus the ROC includes the jω-axis in the s-plane. Since FT of a signal equals LT
evaluated along jω-axis, an LTI system is stable if and only if ROC of its system function H(s) includes
jω-axis that is R e s 0
1 d t
A pole that is in the left half of the s-plane Hs and ht e k . Hence it contributes a
s dk
right-sided decaying exponential term to the impulse response. Since for t < 0, h(t) tends to infinity and
a stable impulse response cannot contain any increasing exponential terms as an increasing
exponential is not absolutely integrable. This is shown below.
1
A pole of the system transfer function that is in the right half of the s-plane gives Hs and
s dk
d t
ht e k . Hence it contributes a left-sided decaying exponential term to the impulse response, since
for t > 0, h(t) tends to infinity and a stable impulse response cannot contain any increasing exponential
terms as an increasing exponential is not absolutely integrable. Thus we do not draw h(t) for t > 0 as
shown below.
If we take LT of both sides of this equation, we find that inverse system transfer function
H 1 s satisfies
H 1 sHs 1
1
H 1 s
Hs
Inverse system transfer function is therefore inverse of transfer function of original system.
If H(s) is written in pole-zero form as
M
bM
a N k
s c k
1
Hs
N
s d k
k 1
N
s d k
k 1
H 1 s
M
bM
a N k
s c k
1
Zeros of inverse systems are pole of H(s) and poles of inverse system are zeros of H(s). Thus any
system with a rational transfer function has an inverse system.
A stable, causal system must have all of its poles in left half of s-plane. Since poles of inverse
system H 1 s are zeros of H(s), a stable and causal inverse system exists only if all of the zeros of
H(s) are in left half of s-plane. A system whose transfer function H(s) has all of its poles and zeros in
left half of s-plane is said to be minimum phase. A non-minimum phase system cannot have a stable
and causal inverse system as it has zeros in right half of s-plane.
Determining the Frequency Response front Poles and Zeros
The locations of the poles and zeros in the s-plane provide insight into the frequency response of
a system.
M
bM
a N k
s c k
1
Hs
N
s d k
k 1
Frequency response is obtained from the transfer function by substituting jω for ‘s’ that is, by
evaluating the transfer function along the jω-axis in the s-plane. This operation assumes that the jω -
axis lies in the ROC. Substituting s = jω for ‘s’, we get
M
bM
a N k
jω c k
1
Hjω
N
jω d k
k 1
If magnitude is at a fixed value of ω, say ω0, we get
M
bM
aN k 1
jω0 c k
Hjω
N
jω0 d k
k 1
This expression involves a ratio of products of terms having the form j0 g , where ‘g’ is either
a pole or a zero. The zero contributions are in the numerator, while the pole contributions are in the
denominator. The factor jω0 g is a complex number that may be represented in the s-plane as a
vector from the point ‘g’ to the point jω0 as shown in figure 1. The length of this vector is jω0 g . By
examining the length of the vector as ω0 changes, we may assess the contribution of each pole or zero
to the overall magnitude response.
Figure 1: The quantity jω0 g shown as a vector from ‘g’ to jω0 in the s-plane.
Figure 2(a) depicts the vector jω g for several different values of ω and figure 2(b) depicts
jω g as a continuous function of frequency. Note that when ω = Im {g}, jω0 g R e g . Hence, if
‘g’ is close to the jω-axis R e g 0 , then jω0 g will become very small for ω = Im {g}. Also, if ‘g’ is
close to the jω-axis, then the most rapid change in jω0 g occurs at frequencies closest to ‘g’.
If ‘g’ represents a zero, then jω0 g contributes to the numerator of |H(jω)|. Hence at
frequencies close to a zero, |H(jω)| tends to decrease. How far |H(jω)| decreases depends on how close
the zero is to the jω-axis. If the zero is on the jω-axis, then |H(ω)| goes to zero at the frequency
corresponding to the zero location.
Figure 2: The function jω0 g corresponds to the lengths of vectors from ‘g’ to the jω-axis in the s-
plane. (a) Vectors from ‘g’ to jω for several frequencies, (b) jω0 g as a function of jω.
At frequencies far from a zero (i.e., when ω g , jω0 g is approximately equal to ω . The
component of the magnitude response due to a zero is shown in figure 3(a). On the other hand, if g
corresponds to a pole, then jω0 g contributes to the denominator of |H(jω)|; and when jω0 g
decreases, |H(jω)| increases. How far |H(jω)| increases depends on how close the pole is to the jω-axis.
A pole that is close to the jω-axis will result in a large peak in |H(jω)|. The component of the magnitude
response associated with a pole is shown in figure 3(b).
Figure 3: Components of the magnitude response, (a) Magnitude response associated with a zero,
(b) Magnitude response associated with a pole.
Hence zeros near the jω-axis tend to pull the response magnitude down, while poles near the jω-
axis tend to push the response magnitude up. Note that a pole cannot lie on the jω-axis, since we have
assumed that the ROC includes the jω-axis.
The phase of H(jω) may be evaluated in terms of the phase associated with each pole and zero and
may be written as:
M N
argjω c k argjω d k
b
argHjω arg M
a N k 1 k 1
In this case, the phase of H(jω) is the sum of the phase angles due to all the zeros, minus the sum of
b
the phase angles due to all the poles. The first term, arg
M a is independent of frequency. The
N
phase associated with each zero and pole is evaluated by considering a term of the form arg{jω0 - g}.
This is the angle of a vector pointing from ‘g’ to jω0 in the s-plane. The angle of the vector is measured
relative to the horizontal line through ‘g’ as shown in figure 4. By examining the phase of this vector as
‘ω’ changes, we may assess the contribution of each pole or zero to the overall phase response.
Figure 4: The quantity jω0 - g shown as a vector from ‘g’ to jω0 in the s-plane. The phase angle of the
vector is ‘ф’, defined with respect to a horizontal line through ‘g’.
Figure 5: The phase angle of jω - g. (a) Vectors from ‘g’ to jω for several different values of ‘ω’. (b) Plot
of arg{jω - g} as a continuous function of ‘ω’.
Figure 5(a) depicts the phase of this vector for several different frequencies, and figure 5(b)
illustrates the phase as a continuous function of frequency. Since ‘g’ is in the left half of the s -plane,
the phase is - π/2 for ‘ω’ large and negative, increasing to zero when ω = Im{g}, and increasing further
to π/2 for ‘ω’ large and positive. If ‘g’ is in the right half of the s -plane, then the phase begins at -π/2
for ‘ω’ large and negative, decreases to -π when ω = Im{g}, and then decreases to -3π/2 for ‘ω’ large
and positive. If ‘g’ is close to the jω-axis, then the change from --π/2 to π/2 (or -3π/2) occurs rapidly in
the vicinity of ω = lm{g}. If ‘g’ corresponds to a zero, then the phase associated with ‘g’ adds to phase of
H(jω), while if ‘g’ is a pole, phase associated with ‘g’ subtract from phase of H(jω).
System analysis with Fourier transform
Consider an LTI system described by differential equation
N
d k yt M
d k x t
ak dt k
bk dt k
k 0 k 0
When x t t , X(ω) = 1 and Y(ω) = H(ω). The F-1[H(ω)] = h(t) is called the impulse response
of the system.
In general,
H H e jH
Where |H(ω)| is called the magnitude response and ∠H(ω) is the phase response.
The Fourier transform of the impulse response is called the frequency response or transfer
function of the system.
Amplitude and phase spectra
x t C k e jk t and C k T xt e j kt dt
0
1 0
k 0 T0
C k a k jb k (1)
b
Magnitude of Ck is given by C k a 2k b 2k and phase of Ck is given by k tan 1 k .
ak
A plot of C k versus k is called amplitude spectrum of periodic signal x(t) and plot of k versus k
is called phase spectrum of x(t).
Since index ‘k’ assumes only integers, amplitude and phase spectra are not continuous curves
but appear only at discrete values of ‘k’. They are therefore referred to as discrete frequency spectra or
line spectra.
xt e
1 j0 kt
Ck dt [ Ck C k , that is C k C k ]
T0 T0
xt e
1 j0 kt
C k dt
T0 T0
Ck a 2k b 2k C k
C k C k (2)
Since for even signal, x(-t) = x(t) and for odd signal x(-t) = -x(t), equation (2) shows that amplitude
spectra is an even function of ‘k’.
b b
Similarly phase of C k is given by k tan 1 k tan 1 k
k
ak ak
k k k (3)
Equation (3) shows that a phase spectrum is an odd function of k.
The sampling theorem states that A band limited signal x(t) with X 0 for m [that is,
Xf 0 for f f m ] can be represented into and uniquely determined from its samples x(nT) if the
sampling frequency f 2f m , where fm is the highest frequency component present in it. That is, for
signal recovery, the sampling frequency must be at least twice the highest frequency present in the
signal.
This theorem is known as uniform sampling theorem since it pertains to be the specification of
a given signals by its samples at uniform intervals of 1 sec.
2f m
It is also called low pass sampling theorem because it applies to low pass signals, that is, signals
for which X f 0 for all frequencies such that f f m is some finite frequency.
Proof
The sampling operation can be represented as shown in figure. x(t) is a continuous time band
limited signal to be sampled which has no special components above fm cycles per sec. That means
X(ω), the Fourier transform of x(t) is 0 for ω > ωm. δT(t) is an impulse train which samples at a rate of fs
Hz and xs(f) is the sampled signal. T is the sampling period and f s 1 is the sampling frequency.
T
xs(t) is the products of signal x(t) and impulse train δT(t). It is a sequence of impulses located at
regular intervals of T sec and having strength equal to values of x(t) at the corresponding instants.
x s t x t T t where T t t nT
n
1 jn s t 1 jn s t
T t t nT e e
n n T T n
1
x s t x t T t x t e jn s t
T n
Taking Fourier transform on both sides, we have
1 1
Fx s t F x t e jn s t
T n
F x t e jn s t
T n
1 1 2
i.e X s X n s X n
T n T n T
or X s f f s Xf nf s
n
Where X(ω) or X(f) is the spectrum of input signal and Xs(ω) or Xs(f) is the spectrum of the sampled
signal.
Thus, the Fourier transform of the sampled signal is given by an infinite sum of shifted replicas
of the Fourier transform of the original signal.
2
The signal x(t) is band limited to fm. The term X n is the shifting of X(ω) from ω = 0 to
T
2
T T T
n . Hence, Xs(ω) is the sum of shifted replicas of 1 X centering at 2 n , n 0, 1, 2, .
Figure shows the plot of X(ω) and Xs(ω) for various values of T . It shows that [figure (b) and (c)] if
, the replicas will not overlap and as a result, the frequency spectrum of TXs(ω) in the frequency
T m
range T , T is identical to X(ω). X(ω) can be recovered from Xs(ω) by passing it through a low pass
filter which has sharp cutoff at T (or with bandwidth B, where m B s m ). If T m [figure
(d)], the successive frequency spectra will overlap and the original signal cannot be recovered from the
sampled signal. Therefore, we can say that for signal recovery,
s m m , i.e, s 2m
or f s f m f m , i.e, f s 2f m
or m , i.e. , f s 2 f m , i.e., f s 2f m
T
1 1 1
or , i.e., T
f s 2f m 2f m
Figure: (a) Frequency spectrum of continuous time signal x(t), (b), (c) and (d) frequency spectrum of
sampled signal xs(t) for (π/T) > ωm, (π/T) = ωm and (π/T) < ωm,
So we can conclude that if the sampling interval T is small 1 , X(ω) can be recovered from
2f m
Xs(ω), but if T becomes larger than 1 , then there is an overlap between successive cycles and X(ω)
2f m
cannot be recovered from Xs(ω). This proves the sampling theorem.
In general there are two basic conditions to be satisfied if x(t) is to be recovered from its
samples.
1. x(t) should be band limited to some frequency ωm.
2. The sampling frequency ωs should be at least twice the band limiting frequency ωm. [i.e., ωs ≥
ωm]
From figure we can observe that:
1. Xs(ω) is a repetitive version of X(ω) with X(ω) repeating itself at regular intervals of ω s, the
sampling frequency.
2. When ωs ≥ 2ωm, [figure (b)], the spectral replicates have a larger separation between them,
known as guard band, which makes the process of filtering much easier and effective. Even a
non-ideal filter which does not have a sharp cutoff can also be used.
3. When ωs = 2ωm, [figure (c)], there is no separation between the replicates, so no guard band
exists, and X(ω) can be obtained from Xs(ω) by using only an ideal low pass filter (LPF) with
sharp cutoff.
4. When ωs < 2ωm, [figure (d)], the low frequency components in Xs(ω) overlap on the high
frequency components of X(ω), there is distortion and X(ω) cannot be recovered from Xs(ω) by
using any filter. This type of distortion is called aliasing. Aliasing can be avoided if
f s 2f m or T 1 .
2f m
Since it is impossible to build filters having an infinite sharpness of cutoff, a guard band
between fm and fs – fm is preferred.
The impulse train at the output of the sampler is processed through an ideal LPF with gain T
and cutoff frequency greater than ωm and less the ωs – ωm. The resulting output signal will exactly
equal x(t).
Nyquist rate of sampling
Nyquist rate of sampling is the theoretical minimum sampling rate at which a signal can be
sampled and still be reconstructed from its samples without any distortion. It is the theoretical
minimum because when the Nyquist rate of the sampling is used, only an ideal LPF can be used to
extract X(ω) from Xs(ω), i.e., to recover x(t) from xs(t). It is always equal to 2fm where fm is the
maximum frequency component present in the signal.
A signal sampled at greater than Nyquist rate is said to be over sampled and a signal sampled at
less than its Nyquist rate is said to be under sampled.
Nyquist interval
Nyquist interval is the time interval between any two adjacent samples when sampling rate is
Nyquist rate.
Nyquist rate, f N 2f m Hz
1 1
Nyquist int erval sec .
f N 2f m
1
the individual terms in equation X s X ns overlap is referred to as aliasing. This
T n
process of spectral overlap is also called frequency folding effect.
Aliasing is defined as the phenomenon in which a high frequency component in the frequency
spectrum of signal takes identity of a lower frequency component in the spectrum of the sampled
signal.
Aliasing can occur if either of the following condition exists:
1. The signal is not band limited to a finite range.
2. The sampling rate id too low.
Theoretically if the signal is not band limited, there is no way of avoiding the aliasing problem
with the basic sampling scheme employed. However, the spectra of most real life signals are such that
they may be assumed to be band limited. A common practice employed in many sampled data systems
is to filter the continuous time signals before sampling to ensure that it does meet the band limited
criterion closely enough for all practical purposes.
To avoid aliasing, it should be ensured that:
1. x(t) is strictly band limited (this can be ensured by using anti-aliasing filter before the
sampler).
2. f s 2f m .