Chap 4

Sampling of Continuous-
Time Signals
 Introducti
on
 4.1 Periodic Sampling
 4.2 Frequency-Domain Representation of Sampling
 4.3 Reconstruction of a Bandlimited Signal
from its Samples
 4.4 Discrete-Time Processing of Continuous-
Time Signals
 4.5 Continuous-Time Processing of Discrete-
Time Signals
 4.6 Changing the Sampling Rate Using
Discrete-Time Processing

2
 Block diagram, frequency response, and
impulse response of an ideal bandlimited
signal reconstruction system
Ideal reconstruction
system
Convert xS(t) Ideal
the reconstructio
x[n sequence n filter Hr(j) x r (t)
] to impulse
train Sampling period
T
H r(j) h r(t)
1
T

t
-/T /T 
-3T -T 0 T 3T
Ideal bandlimited
interpolation
Discrete-time processing of continuous-time
signal
 Discrete-Time Processing of
Continuous- Time Signals
2
 C/D converter produces a discrete-time
signal
x [ n ]j  x c 1( n T)  
X(e ) X j )
 T k    ( jc 2 k T
T
 Linear Time-Invariant Discrete-Time
Systems Y(ej) = H(ej)X(ej)
an Yr(j) = Hr(j)H(ejT)X(ejT) with  =
d /T
 D/C converter creates a continuous-time output
signal 
s in [  ( t  n T ) / T ]
y ( t )  n   y [ n ]
r (t  nT ) / T
Y ( j )  H ( j )Y
( e j T )
r r
Example
 Example of Ideal Continuous-Time
Lowpass Filtering Using a Discrete-Time
Lowpass Filter
 Example of Ideal Continuous-Time
Lowpass Filtering Using a Discrete-Time
Lowpass
a) Fourier Filter
transform of a
bandlimited input signal
b) Fourier transform of
the sampled input
c) Discrete-time Fourier
transform
of sequence of samples
and frequency response
of the discrete-time
system
d) Fourier transform of output
of the discrete-time system
e) Fourier transform of output
of
the discrete-time system
and frequency response of
ideal reconstruction filter
f) Fourier transform of output
 Discrete-Time Processing of
Continuous- Time Signals:
 The LTIImpulse
behaviorInvariance
of the system depends
on two factors:
discrete-time system must be linear and
time-invariant
input signal must be bandlimited and the
sampling rate must be high enough so that any
aliased components are removed by the
discrete-time system
 Discrete-time system is said to be an
impulse- h[n] = Thversion
invariant c(nT)
of the
continuous-time
an system
H(ej = Hc(j/T), when
|| < and
d ) =

2018/9/18 DSP 29
 Impulse
Invariance
Figure 4.14 (a) Continuous-time LTI system. (b) Equivalent
system for
bandlimited inputs.
 Impulse
Invariance
H (e j )  H ( j / T ),  
eq. 
c
H c ( j)   /
4.48  T
0,

eq. 4.49  h n   Thc (nT

)

eq.

4.50 
 Impulse
Invariance
From sampling depicted in Eq.
(4.16)
eq. h n   hc (nT 
1    2 k  
H) (e j )  Hc  j 
4.51 T 
  T 
 k
  
j 1  
H (e ) H
eq. 4.52   T c  T 
T
Modifying Eqs (4.51) and (4.53) to
account for j ,  
eq.scale
the 4.53factor of T, we have

eq. h n   Thc (nT
H )(e ) 
j
j , 
c
 T 
4.54  H
 Impulse
Invariance
Example 4.5 A Discrete-time Lowpass Filter
obtained by Impulse Invariance
1,
c

0, 
c  c
H ( j) 


sin(c t)
hc (t)  
t
sin( c nT ) sin(c n)
h  n  Thc (nT )  
T  
 nT n
  c
H (e ) 
j

1, 
0, c   


Continuous-Time Processing of
Discrete- Time Signals
 Continuous-Time Processing of
Discrete- Time Signals
 For ideal D/C : Xc(jW) and Yc(jW) are |  |  /
zero for T
 We can express
 D/Cs in
as[ 
follows.
(t  nT ) / T ]
xc (t )   x [ n ] (t  nT ) / T
n  
s in [  ( t  n T ) / T ]
yc (t )  n
  y[n] (t  nT ) / T
where x[n] = xc(nT) and y[n] = yc(nT) , and frequency-
domain :
Xc ( j)  TX(e ),...........    / T
jT

Y c ( jj )  1H c ( j  ) X c ( j  ) , . . . .  
Y (e  )  Yc ( j  / T ) , . . . . . . . . . .   
 / T T
Overall system behaves as a discrete-time system
H(e j )  Hc (  / T),......   
whose
and jif frequency response of continuous-time
system
H ( j)is  H ( ej T ),........    / T
c
 Changing The Sampling Rate
Using Discrete-Time
Processing
 Sampling Rate Reduction by an Integer
Factor
 Increasing the Sampling Rate by an Integer
Factor
 Changing the Sampling Rate by a
Noninteger Factor

 Sampling Rate Reduction  Increase

period
 Decrease
frequency
 Sampling Rate Reduction by an
Integer Factor
 Sampling Rate Reduction by an Integer Factor
xd[n] = x[nM] = xc(nMT)
system called Sampling Rate Compressor or Compressor
operation called Downsampling
 xd[n] is an exact representation of xc(t) if /(MT) > N
 sampling rate can be reduced by a factor of M without
aliasing if the original sampling rate was at least M times
the Nyquist rate or if the bandwidth of the sequence is
first reduced by a factor of M by discrete-time filtering
 Fourier transform of discrete-time sampled sequence xd[n] is
j 1  
Xd (e ) X (j j
 )c
MT r 2
MT r

MT
where r = i+kM, -infinity < k < infinity and 0 < and = i < and =
M-1 j
Xd (e ) 1 X(eM)
M1 j(/ M2i/

M i )
 0
 Lowpass filter+Compressor =
 General system for sampling rate
reduction by integer factor
M

x[n Lowpass x~[n xd~[n] =

] filter Gain = ] M x~[nM]
1 Cutoff =
/M
Samplin Samplin Sampling
g g period T’ =
period T period T MT
Sampling rate reduction
by 2
Downsampling with aliasing (a to c)
and with prefiltering to avoid
aliasing (d to f).