AT77.

02 Signals, Systems and Stochastic Processes

Asian Institute of Technology
Handout 51
Tue 31 Aug 2010

1.2.12 Sampling of Continuous-Time Signals

If we want to process signals by digital computers for transmission, storage, or analysis
purposes, we must represent an analog signal in a digital form, i.e. as a sequence of
numbers
{v[n]} = {. . . , v[−2], v[−1], v[0], v[1], v[2], . . .}
For convenience, a sequence {v[n]} is normally written as v[n]. The signal v[n] is
referred to as a discrete-time signal whose values are taken from the corresponding analog
signal v(t) by
v[n] = v(nT ), n ∈ Z,
where T is the sampling period while fs = 1/T is the sampling frequency or sampling
rate.
It is convenient to represent the sampling process in two stages, as illustrated in
figure 1.22.

1. Multiplication by a periodic impulse train with period T , i.e.

∞
X
s(t) = δ(t − nT ).
n=−∞

2. Conversion of the impulse train to a sequence, i.e. discrete-time signal.

Fourier Transforms of Sampled Signals

Multiplying v(t) by s(t) gives us the signal
∞
X ∞
X
vs (t) = v(t)s(t) = v(t) δ(t − nT ) = v(nT )δ(t − nT ).
n=−∞ n=−∞

1
P∞
From S(f ) = T k=−∞ δ(f − kfs ) and Vs (f ) = V (f ) ∗ S(f ),

∞
1 X
Vs (f ) = V (f − kfs )
T k=−∞

NOTE: The Fourier transform of the sampled signal vs (t) consists of periodically repeated
copies at V (f ) equally spaced apart by fs , as illustrated in figure 1.23.
1
Course notes were prepared by Dr. R.M.A.P. Rajatheva and revised by Dr. Poompat Saengudomlert.

1
 Conversion from
impulse train
to sequence

0 0

0 1 2 0 1 2
Figure 1.22: Sampling process

Figure 1.23: Fourier transforms of sampled signals.

2
Aliasing and Nyquist rate
Let W be the bandwidth of v(t), i.e. W is the smallest number such that V (f ) = 0 for
|f | > W . From figure 1.23, the following facts are evident.
1. When fs > 2W , the copies of V (f ) do not overlap. Therefore, when they are added
together, there remains a copy of V (f ) around each integer multiple of fs .
2. When fs ≤ 2W , the copies of V (f ) overlap. Therefore, when they are added
together, V (f ) may not be recoverable. This overlapping phenomenon is referred
to as aliasing.
NOTE: The maximum signal frequency W is referred to as the Nyquist frequency. The
frequency 2W that must be exceeded by the sampling frequency is called the Nyquist
rate.
Figure 1.24 illustrates aliasing in the time domain. Note that there are two sinusoidal
signals with frequencies f1 = fs /8 and f2 = 7fs /8. These two signals have exactly the
same samples.

1.5
fs=8,f1=1,f2=7,v1(t)=sin(2π t),v2(t)=-sin(14π t)

0.5

-0.5

-1

-1.5
0 0.2 0.4 0.6 0.8 1
t
Figure 1.24: Simple illustration of aliasing in the time domain.

Discrete-Time Fourier Transforms

The Fourier transform of a discrete-time signal v[n], called the discrete-time Fourier
transform, is defined as
X∞
jΩ
V (e ) = v[n]e−jΩn
n=−∞

where Ω denotes the digital angular frequency expressed in radians. The corresponding
discrete-time inverse Fourier transform is defined as
Z π
1
v[n] = V (ejΩ )ejΩn dΩ
2π −π

3
 A discrete-time Fourier transform is always periodic in Ω with period 2π. To see this,
note that we can write for any nonnegative integer k,
∞
X ∞
X
j(Ω+2πk) jΩ
V (e )= v[n]e −j(Ω+2πk)n
= {z } = V (e ).
v[n]e−jΩn e|−j2πkn
n=−∞ n=−∞ =1

If v[n] is sampled from v(t) with sampling period T , i.e. v[n] = v(nT ), we can relate
V (ejΩ ) and V (f ) as follows. Recall that the sampled signal can be expressed using the
unit impulse as
∞
X
vs (t) = v(t)s(t) = v(nT )δ(t − nT )
n=−∞

with the corresponding Fourier tranform

∞
1 X
Vs (f ) = V (f − kfs ).
T k=−∞

Alternatively, we can use the Fourier transform pair δ(t − nT ) ↔ e−j2πf nT to write
∞
X
Vs (f ) = v(nT )e−j2πf nT .
n=−∞

By setting
Ω = 2πf T
we can write
∞  
jΩ 1 X Ω
V (e ) = V − kfs
T k=−∞ 2πT

NOTE: V (ejΩ ) is the frequency-scaled version of Vs (f ) with the frequency scaling specified
by Ω = 2πf T . This frequency scaling can be seen as a normalization of the frequency
axis so that the analog frequency ω = 2πfs (in rad/s) in V (f ) is normalized to the digital
frequency Ω = 2π (in rad) for V (ejΩ ).

1.2.13 Reconstruction of a Band-Limited Signal from Its Samples

P
From Vs (f ) = T1 ∞ k=−∞ V (f − kfs ), in the absence of aliasing, i.e. V (f ) = 0 for |f | >
fs /2, we can write
V (f ) = T Vs (f ), |f | ≤ fs /2.
It follows that V (f ) can be recovered from Vs (f ) by passing vs (t) through an ideal lowpass
filter with the following Fourier transform, as illustrated in figure 1.25.

H(f ) = T rect(f /fs ).

Since T rect(f /fs ) ↔ sinc(t/T ), the reconstructed signal can be written as

  ∞
!   ∞  
t X t X t
vs (t) ∗ sinc = v(nT )δ(t − nT ) ∗ sinc = v(nT )sinc −n
T n=−∞
T n=−∞
T

4
 1.2 1.2
T=1
T 1
1

0.8
0.8
V(f)=Trect(fT)

v(t)=sinc(t/T)
0.6
0.6
0.4
0.4
0.2
0.2
0

0 -0.2
-1/2T 1/2T
-0.2 -0.4
T
-1.5 -1 -0.5 0 0.5 1 1.5 -6 -4 -2 0 2 4 6
f t
Figure 1.25: Low pass filter and its Fourier transform.

Conversion from Ideal

sequence to lowpass
impulse train filter
Figure 1.26: Reconstruction process from v[n] to v(t).

If we reconstruct v(t) from the discrete-time signal v[n], then we can write

∞  
X t
v(t) = v[n]sinc −n
n=−∞
T

The overall process of reconstructing v(t) from v[n] is illustrated in figure 1.26.
Figure 1.27 illustrates how the reconstruction of v(t) can be viewed as interpolating
v[n] by the sinc functions. In summary, the overall discussion in this section can be
summarized in the following sampling theorem.

Sampling theorem: Suppose that v(t) is bandlimited such that V (f ) = 0 for |f | > fN .
If v[n] is obtained from sampling v(t) with sampling rate fs = 1/T , i.e. v[n] = v(nT ),
such that fs > 2fN , then v(t) is uniquely defined by its samples v[n] as follows.
∞  
X t
v(t) = v[n]sinc −n
n=−∞
T

5
 1.2
1
0.8
0.6
v[n]

0.4
0.2
0
-0.2
-0.4
-4 -3 -2 -1 0 1 2 3 4 5
n
v(t)=Σn v[n]sinc(t/T-n)

1.2
1
0.8
0.6
0.4
0.2
0
-0.2
-0.4
-4 -3 -2 -1 0 1 2 3 4 5
t
Figure 1.27: Reconstruction by interpolating using sinc pulses.