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Sampling of signals
Sampling is the conversion of a continuous signal to a discrete signal, by taking
samples from a signal at given moments of times. Sampling is performed by measuring
the value of the continuous signal every Ts seconds, with a sampling frequency of fs=1/Ts.
The sampling frequency the number of samples obtained in one second and it is measured
in hertz or in samples per second. X()s
The ideal sampled signal is:
x ( t ) =
x ( kT ) ( t kT ) .
s
k =
Its spectrum is
1
X ( ) =
Ts
X ( k )
k =
The NyquistShannon Sampling theorem: A finite energy and band limited signal x(t),
with bandwidth B = M is uniquely determined by its samples x ( nTs ) n ] if
s 2M .
where
2
.
T
If we sample the signal with the sampling period Ts we obtain a sampled
signal x ( t ) with the spectrum X ( ) . The original signal is reconstructed from the
s =
spectrum of the sampled signal with lowpass filtering, with the cutoff frequency C , such
that,
M C s M 2M s or
fs 2 fM .
The limit case: The sampling frequency is equal with the bandwidth
s = 2M
and the cutoff frequency is the maximum frequency of the signals spectrum,
C = M
Unfavorable case: The sampling frequency is lower than the bandwidth,
s < 2M
when aliasing appears, the original signal cannot be reconstructed.
The spectrum of the ideal lowpass filter is:
H r ( ) = Ts pC ( ) .
Its impulse response is
hr ( t ) = Ts
sin ( C t )
t
s
2
k =
2C
x ( kTs )
sin (C ( t kTs ) )
C ( t kTs )
sin k
Ts
xr ( t ) = x ( kTs )
t
k =
k
Ts
Problems
1. Consider the system below
x1(t)
( )2
y(t)
x2(t)
x3(t)
The support of the Fourier transform for the signals xi(t), with i=1,2,3, are [-i, i].
a) Find the maximum sampling period T for which the signal y(t) can be reconstructed
from its samples using ideal low pass filtering.
3
y ( t ) = x12 ( t ) x2 ( t ) + x3 ( t ) Y ( ) =
4 2
The support of the Fourier transform X 1 ( ) is:
X 1 ( ) X 1 ( ) X 2 ( ) + X 3 ( )
supp { X 1 ( )} = [ 1 , 1 ]
supp { X 1 ( ) X 1 ( ) X 2 ( )} = [ 21 2 , 21 + 2 ]
supp {Y ( )} = y , y , y = max ( 21 + 2 , 3 )
s 2 y ,
s y
Therefore, the maximum value for the sampling period is:
Ts =
max ( 21 + 2 , 3 )
Ts =
b)
supp { X 1 ( ) X 1 ( )} = [ 21 , 21 ] = ,
2 2
supp { X 1 ( ) X 1 ( ) X 2 ( )} = [ 21 2 , 21 + 2 ] = [ , ]
3
max ( 21 + 2 , 3 ) = max , =
4
supp {Y ( )} = [ , ]
2. Consider a system with the input signal x(t) and the output signal y(t) with the relation
output-input a polynomial:
N
y = P ( x) = ak x k .
k =0
If the signal x(t) is band limited to the interval [-M, M], find the maximum sampling
period for which the signal y(t) can be recovered from its samples using ideal lowpass
filtering.
Solution.
The Fourier transform of the signal y(t) is:
a
a
Y ( ) = a0 + a1 X ( ) + 2 X ( ) X ( ) + ... + N N X ( ) X ( ) ... X ( )
2
( 2 )
N factors
supp (Y ( ) ) = [ N M , N M ]
2 N M => T
.
T
N M
3. Consider the band limited signal z(t), in the interval [1, 2], with the spectrum
represented below:
Z()
1
0
Denote C =
2 + 1
; M =
2 1
2
2
a) Show the relation z (t ) = x1 ( t ) e jC t holds, where x1(t) is a bandlimited signal, in
sin n
j t nT
T
z ( t ) = z ( nT ) e s ( )
t
n =
n
T
Solution.
a) Assume there is a signal x1 ( t ) , a bandlimited signal, in the interval [-M, M], for
which we have
z (t ) = x1 ( t ) e jC t Z ( ) = X 1 ( C )
This implies that the support of Z ( ) is the same as the support of the shifted version of
X 1 ( ) :
supp {Z ( )} = supp { X 1 ( C )} = [1 , 2 ]
supp {Z ( )} = [1 , 2 ]
supp { X 1 ( C )} = [ M + C , M + C ]
sin k
Ts
if C = M = s
x1 ( t ) = x1 ( kTs )
2
t
k =
k
Ts
=> z ( t ) = x1 ( t ) exp ( jC t ) =
=> z ( t ) =
exp ( j t ) x ( kT ) sinc T
C
k =
t
exp ( j ( t kT ) ) z ( kT ) sinc T
k =
k for C = M = s , Ts =
2
M
4. The signal z(t) with the spectrum sketched below is the analytical signal corresponding
to a real signal, x(t).
Z()
1
b) Use the relation between the signals x ( t ) , x ( t ) and z ( t ) , where x ( t ) is the Hilbert
transform of the signal x ( t ) , to find the maximum value of the sampling step T for which
we have:
x (t ) =
n =
x ( t ) =
n =
sin n
T
x ( nT ) cos C ( t nT ) x ( nT ) sin C ( t nT )
t
n
T
sin n
T
x ( nT ) cos C ( t nT ) + x ( nT ) sin C ( t nT )
t
n
T
c) Using the relations from b), systems that recover the original signals x ( t ) and x ( t )
from the sampled signals, xs ( t ) = x ( t ) T ( t ) and xs ( t ) = x ( t ) T ( t ) , respectively, are
represented below. Find the impulse response h1(t) and h2(t) and the frequency responses
H1() and H2() of these systems.
xs(t)
xs(t)
h1(t)
h1(t)
x(t)
xs ( t )
xs ( t )
h2(t)
x ( t )
h2(t)
Solution.
a) x(t) = Re(z(t)), x ( t ) = H ( x ( t ) ) =
1
x ( t ) X ( ) = j sgn X ( )
t
z ( t ) = x ( t ) + jx ( t )
Z ( ) = X ( ) + jX ( ) = X ( ) + j ( j sgn ) X ( ) = X ( ) + sgn X ( )
2 X ( ) ,
Z ( ) =
0,
>0
otherwise
=> < 0 : X ( ) =
Z ( )
2
, > 0 : X ( ) =
Z ( )
2
X()
1/2
-2
-1
n =
x ( nT ) h1 ( t nT )
h1 ( t ) = cos C t sinc
h2 ( t ) = sin C t sinc
sinc 0t
x ( nT )h ( t nT )
2
n =
t H1 ( )
t H 2 ( )
1
Tp ( ) ( C ) + ( + C )
T
2
1
H1 ( ) = T p ( C ) + p ( + C )
T
2 T
H1 ( ) =
Tp ( ) ( C ) ( + C )
T
2
j
1
H 2 ( ) = T p ( C ) p ( + C )
T
2 j T
H 2 ( ) =
H1()
T/2
-C-/T
-C -C+/T
C-/T C
C+/T
jH2()
T/2
-C
0
=> X ( ) H1 ( ) =
C-/T C
C+/T
T
T
X ( ) and jX ( ) jH 2 ( ) = sgn X ( ) jH 2 ( ) = X ( )
2
2
T
=> X ( ) H 2 ( ) = X ( )
2
X ( ) H1 ( ) X ( ) H 2 ( ) = TX ( )
After sampling of the signals x and x we obtain:
xs ( t ) = x ( nT ) ( t nT ) X s ( ) =
1
2
X k
T k =
T
1
2
xs ( t ) = x ( nT ) ( t nT ) X s ( ) = X k
T k =
T
X s ( ) = j sgn X s ( )
T
1
T 1
c) X s ( ) H1 ( ) = X ( ) = X ( )
T
2 2
1
X s ( ) H 2 ( ) = X ( )
2
Therefore, X s ( ) H1 ( ) X s ( ) H 2 ( ) = X ( )
In the time domain:
=>
>
2 1
2
x (t ) =
k =
1
1
j sgn X ( ) => X s ( ) H1 ( ) = X ( )
2
2
1
1
j sgn X ( ) => X s ( ) H 2 ( ) = X ( )
2
2
Therefore, X s ( ) H 2 ( ) + X s ( ) H1 ( ) = X ( )
In the time domain:
x ( t ) =
k =
X()
1
-2
-1
We use the same relations from problem 3. The signal is ideally sampled with the
sampling frequency 2M .
a) Sketch the spectrum of the sampled signal if (4k+3)M = C
b) Sketch the spectrum of the sampled signal if (4k+1)M=C
c) Implement a system that permits the recovery of the original signal x(t) from the
sampled signal in case a) and in case b).
Solution.
a) The sampling pulsation is: s = 2 2M = 4M
The spectrum of the sampled signal:
1
2
X s ( ) = X k
T k =
T
Xs()
1/T
-2
-1
Xs()
1/T
-2 -1
Hr(), case a)
2M/
4M
6M
8M
Hr(), case b)
2M/
4M
6M
8M
6. Consider the system represented below. x(t) is a bandlimited signal at the interval
[-M, M], and p(t) is a periodic signal of period T = M .
x(t)
xs(t)
p(t)
10
X s ( ) =
a X ( n )
n =
How much is s?
b) Assuming the continuous component of x(t) is not null a0 0 , show that x(t) can be
recovered from xs(t) using ideal lowpass filtering. Find the parameters for the lowpass
filter.
c) Implement a system for the reconstruction of the original signal x(t) from the sampled
signal xe(t) for a0=0.
d) For the spectrum of x(t) given below, sketch the magnitude spectrum for the signal
xs(t) in two cases: p(t)=T(t) and p(t)=T(t-).
X()
1
-M
Solution
a) p ( t ) =
P ( ) =
n =
an e
jn
2
t
T
2 a n T
n =
1
X ( ) P ( )
2
2
X s ( ) = an X n
,
T
n
X s ( ) =
2
= 2M
T
2
b) a0 0 => X s ( ) = a0 X ( ) + an X n
n0
s =
H()
1/a0
-M
M
0
11
c) a0 = 0, a1 0 => X s ( ) = a1 X ( ) +
a X n T
n 0,1
H1()
1/a1
xs(t)
x(t)
h1(t)
2/T-M
=M
2M
2/T+M
=3M
exp(-j2t/T)
2
2 ( )
T T
1
1
2
X s ( ) =
X ( ) P ( ) = X k
2
T
T
d) p(t)=T(t); P ( ) =
X()
1/T
0
p(t)=T(t-); P ( ) = e j
X s ( ) =
X s ( ) =
2
2 ( )
T T
1
1
2
X ( ) P ( ) =
X ( ) e j
2
2
T
1
2
X ( ) k
T
T
X s ( ) =
2M
-M
2
jk
k
k
2
T
1
2
X k
T k
T
1
2
X k
T k
T
12
T
2
jk T
e