Chapter 6

THE SAMPLING PROCESS

6.6 Sampling Theorem
6.7 Aliasing
6.8 Interrelations

Copyright
c 2005 Andreas Antoniou
Victoria, BC, Canada
Email: [email protected]

July 14, 2018

Frame # 1 Slide # 1 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Introduction

I In order to process a continuous-time signal using digital

signal processing methodologies, it is first necessary to convert
the continuous-time signal into a discrete-time signal by
applying sampling.

Frame # 2 Slide # 2 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Introduction

I In order to process a continuous-time signal using digital

signal processing methodologies, it is first necessary to convert
the continuous-time signal into a discrete-time signal by
applying sampling.
I Sampling obviously entails discarding part of the
continuous-time signal and the question will immediately arise
as to whether the sampling process will corrupt the signal.

Frame # 2 Slide # 3 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Introduction

I In order to process a continuous-time signal using digital

signal processing methodologies, it is first necessary to convert
the continuous-time signal into a discrete-time signal by
applying sampling.
I Sampling obviously entails discarding part of the
continuous-time signal and the question will immediately arise
as to whether the sampling process will corrupt the signal.
I It turns out that under a certain condition that is part of the
sampling theorem, the information content of the
continuous-time signal can be fully preserved.

Frame # 2 Slide # 4 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem

I The sampling theorem states:

A bandlimited signal x(t) for which
ωs
X (jω) = 0 for |ω| ≥
2
where ωs = 2π/T , can be uniquely determined from its values
x(nT ).

Frame # 3 Slide # 5 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem

I The sampling theorem states:

A bandlimited signal x(t) for which
ωs
X (jω) = 0 for |ω| ≥
2
where ωs = 2π/T , can be uniquely determined from its values
x(nT ).
I Alternatively, in what amounts to the same thing, a
continuous-time signal whose spectrum is zero outside the
baseband (i.e., −ωs /2 to ωs /2) can, in theory, be recovered
completely from an impulse-modulated version of the signal.

Frame # 3 Slide # 6 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

I Consider a two-sided bandlimited signal whose spectrum

satisfies the condition of the sampling theorem.

Frame # 4 Slide # 7 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

I Consider a two-sided bandlimited signal whose spectrum

satisfies the condition of the sampling theorem.
I By virtue of Poisson’s summation formula, i.e.,
∞
1 X
X̂ (jω) = X (jω + jnωs )
T n=−∞

impulse modulation will produce sidebands that are well

separated from one another.

Frame # 4 Slide # 8 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

X( jω)

ωs ωs ω
−
2 2
(a)

^ 1
1 X( jω+ jω )
s
X( jω) T X( jω) 1 X( jω–jω )
s
T T

−ωs ωs ωs ωs ω
−
2 2
(b)

Frame # 5 Slide # 9 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

I Now if we pass the impulse-modulated signal through an ideal

lowpass filter with a frequency response
(
T for ω < ωs /2
H(jω) =
0 otherwise

then frequencies in the sidebands will be rejected and we will

be left with the frequencies in the baseband, which constitute
the original continuous-time signal.

Frame # 6 Slide # 10 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

I Now if we pass the impulse-modulated signal through an ideal

lowpass filter with a frequency response
(
T for ω < ωs /2
H(jω) =
0 otherwise

then frequencies in the sidebands will be rejected and we will

be left with the frequencies in the baseband, which constitute
the original continuous-time signal.
I A baseband gain of T is used to cancel out the scaling
constant 1/T introduced by Poisson’s summation formula.

Frame # 6 Slide # 11 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

ωs ωs ω
−
2 2
(a)
^ 1
1 X( jω+ jω )
s
X( jω) T X( jω) 1 X( jω –jω )
s
T T

−ωs ωs ωs ωs ω
−
2 2
(b)
H( jω)
T

ωs ωs ω
−
2 2
(c)

^
T X( jω)
= X( jω)

ωs ωs ω
−
2 2
(d )

Frame # 7 Slide # 12 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

I What has been done through a graphical illustration can now

be repeated with mathematics.

Frame # 8 Slide # 13 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

I What has been done through a graphical illustration can now

be repeated with mathematics.
I If the impulse-modulated signal is passed through a lowpass
filter with a frequency response H(jω) as defined before, then
the Fourier transform of the output of the filter will be

Y (jω) = H(jω)X̂ (jω)

where (
T for ω < ωs /2
H(jω) =
0 otherwise
and
∞
1 X
X̂ (jω) = X (jω + jnωs )
T n=−∞

Frame # 8 Slide # 14 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

···
Y (jω) = H(jω)X̂ (jω)

I If we apply the inverse Fourier transform, we get

∞
" #
X
−1 −jωnT
y (t) = F H(jω) x(nT )e
n=−∞
∞
X
= x(nT )F −1 [H(jω)e −jωnT ] (A)
n=−∞

Frame # 9 Slide # 15 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

···
Y (jω) = H(jω)X̂ (jω)

I If we apply the inverse Fourier transform, we get

∞
" #
X
−1 −jωnT
y (t) = F H(jω) x(nT )e
n=−∞
∞
X
= x(nT )F −1 [H(jω)e −jωnT ] (A)
n=−∞

I The frequency response of a lowpass filter is actually a

frequency-domain pulse of height T and base ωs , i.e.,
H(jω) = Tpωs (ω) and hence from the table of Fourier transforms,
we have
T sin(ωs t/2)
↔ H(jω) (B)
πt

Frame # 9 Slide # 16 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

··· y (t) =
∞
X
x(nT )F −1 [H(jω)e −jωnT ] (A)
n=−∞

T sin(ωs t/2)
↔ H(jω) (B)
πt
I From the time-shifting theorem of the Fourier transform
T sin[ωs (t − nT )/2]
↔ H(jω)e −jωnT (C)
π(t − nT )

Frame # 10 Slide # 17 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

··· y (t) =
∞
X
x(nT )F −1 [H(jω)e −jωnT ] (A)
n=−∞

T sin(ωs t/2)
↔ H(jω) (B)
πt
I From the time-shifting theorem of the Fourier transform
T sin[ωs (t − nT )/2]
↔ H(jω)e −jωnT (C)
π(t − nT )

I Therefore, from Eqs. (A) and (C), we conclude that

∞
X sin[ωs (t − nT )/2]
y (t) = x(nT )
n=−∞
ωs (t − nT )/2

Frame # 10 Slide # 18 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

The Sampling Theorem Cont’d

··· y (t) =
∞
X
x(nT )F −1 [H(jω)e −jωnT ] (A)
n=−∞

T sin(ωs t/2)
↔ H(jω) (B)
πt
I From the time-shifting theorem of the Fourier transform
T sin[ωs (t − nT )/2]
↔ H(jω)e −jωnT (C)
π(t − nT )

I Therefore, from Eqs. (A) and (C), we conclude that

∞
X sin[ωs (t − nT )/2]
y (t) = x(nT )
n=−∞
ωs (t − nT )/2

I For t = nT , we have y (nT ) = x(nT ) for n = 0, 1, . . . , kT , and for

all other values of t the output of the lowpass filter is an
interpolated version of x(t) according to the sampling theorem.
Frame # 10 Slide # 19 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8
Aliasing

I If the spectrum of the continuous-time signal does not satisfy

the condition imposed by the sampling theorem, i.e., if
ωs
X (jω) 6= 0 for |ω| ≥
2
then sideband frequencies will be aliased into baseband
frequencies.

Frame # 11 Slide # 20 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing

I If the spectrum of the continuous-time signal does not satisfy

the condition imposed by the sampling theorem, i.e., if
ωs
X (jω) 6= 0 for |ω| ≥
2
then sideband frequencies will be aliased into baseband
frequencies.
I As a result, X̂ (jω) will not be equal to X (jω)/T within the
baseband.

Frame # 11 Slide # 21 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing

I If the spectrum of the continuous-time signal does not satisfy

the condition imposed by the sampling theorem, i.e., if
ωs
X (jω) 6= 0 for |ω| ≥
2
then sideband frequencies will be aliased into baseband
frequencies.
I As a result, X̂ (jω) will not be equal to X (jω)/T within the
baseband.
I Under these circumstances, the use of an ideal lowpass filter
will yield a distorted version of x(t) at best.

Frame # 11 Slide # 22 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing Cont’d

I Aliasing can be illustrated by examining an impulse-modulated

signal generated by sampling the continuous-time signal

x(t) = u(t)e −at sin ω0 t

0.2
|X( jω)|
0.1

0
−30 −20 −10 0 10 20 ω 30

(a)

Frame # 12 Slide # 23 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing Cont’d

I Aliasing can be illustrated by examining an impulse-modulated

signal generated by sampling the continuous-time signal

x(t) = u(t)e −at sin ω0 t

I The frequency spectrum of x(t), X (jω), extends over the

infinite range −∞ < ω < ∞.

0.2
|X( jω)|
0.1

0
−30 −20 −10 0 10 20 ω 30

(a)

Frame # 12 Slide # 24 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing Cont’d

I The frequency spectrum of impulse-modulated signal x̂(t) can

be obtained as
∞
1 X
X̂ (jω) = X (jω + jnωs )
T n=−∞

by using Poisson’s summation formula.

Frame # 13 Slide # 25 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing Cont’d

··· ∞
1 X
X̂ (jω) = X (jω + jnωs )
T n=−∞

I The shifted copies of X (jω) or sidebands, namely, . . .,

X (jω − j2ωs ), X (jω − jωs ), X (jω + jωs ), X (jω + j2ωs ), . . . overlap
with the baseband −ωs /2 < ω < ωs /2 and, therefore, the above
sum can be expressed as
1
X̂ (jω) = [X (jω) + E (jω)]
T
where
∞
X
E (jω) = X (jω + jkωs )
k=−∞
k6=0

is the contribution of the sidebands to the baseband.

Frame # 14 Slide # 26 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing Cont’d

I Now if we filter the impulse-modulated signal, x̂(t), using an

ideal lowpass filter with a frequency response
(
T for −ωs /2 < ω < ωs /2
H(jω) =
0 otherwise

we will get a signal y (t) whose frequency spectrum is given by

Y (jω) = H(jω)X̂ (jω)

∞
1 X
= H(jω) · X (jω + jnωs )
T n=−∞
= X (jω) + E (jω)

Frame # 15 Slide # 27 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing Cont’d

···
Y (jω) = X (jω) + E (jω)

I In other words, the output of the filter will be signal x(t) plus
an error
e(t) = F −1 E (jω)
which is commonly referred to as the aliasing error.

Frame # 16 Slide # 28 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing Cont’d

I With a sampling frequency of 12.5 rad/s, |E (jω)|, i.e., the

discrepancy between the solid and dashed curves in the figure is
quite large.
0.2
|X(jω)|
0.1

0
−40 −30 −20 −10 0 10 20 30 40
|XD(ejωT)|
0.2

0.1

0
−40 −30 −20 −10 0 10 20 30 40

0.2
Filtered
|XD(ejωT)|
0.1

0
−40 −30 −20 −10 0 10 20 30 40

Frame # 17 Slide # 29 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing Cont’d

I As the sampling frequency is increased to 25, the sidebands are

spread out and |E (jω)| will be decreased quite a bit as shown.

0.2

0.1

0
−40 −30 −20 −10 0 10 20 30 40

0.2
|X(jω+jωs)/T|

0.1

0
−40 −30 −20 −10 0 10 20 30 40

0.2

0.1

0
−40 −30 −20 −10 0 10 20 30 40

Frame # 18 Slide # 30 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Aliasing Cont’d

I A further increase to 40 rad/s will render |E (jω)| for all practical

purposes negligible as can be seen.

0.2

0.1

0
−40 −30 −20 −10 0 10 20 30 40

0.2

0.1

0
−40 −30 −20 −10 0 10 20 30 40

0.2

0.1

0
−40 −30 −20 −10 0 10 20 30 40

Frame # 19 Slide # 31 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Summary of Interrelations

I Impulse-modulated signal:
∞
X
x̂(t) = x(nT )δ(t − nT ) (6.42c)
n=−∞

Frame # 20 Slide # 32 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Summary of Interrelations

I Impulse-modulated signal:
∞
X
x̂(t) = x(nT )δ(t − nT ) (6.42c)
n=−∞

I Spectrum of impulse-modulated signal or discrete-time signal

in terms of the spectrum of the original continuous-time
signal:
∞
1 X
X̂ (jω) = XD (e jωT ) = X (jω + jnωs ) (6.45a)
T n=−∞

where
∞
X
XD (e jωT ) = x(nT )e −jωnT
n=−∞

Frame # 20 Slide # 33 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Summary of Interrelations Cont’d

I Spectrum of impulse-modulated signal (or discrete-time

signal) in terms of the spectrum of the original
continuous-time signal for a right-sided signal:
∞
x(0+) 1 X
X̂ (jω) = XD (e jωT ) = + X (jω+jnωs ) (6.45b)
2 T n=−∞

Frame # 21 Slide # 34 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Summary of Interrelations Cont’d

I Spectrum of impulse-modulated signal (or discrete-time

signal) in terms of the spectrum of the original
continuous-time signal for a right-sided signal:
∞
x(0+) 1 X
X̂ (jω) = XD (e jωT ) = + X (jω+jnωs ) (6.45b)
2 T n=−∞

I Laplace transform of impulse-modulated signal in terms of the

Laplace transform of the original continuous-time signal for a
right-sided signal:
∞
x(0+) 1 X
X̂ (s) = XD (z) = + X (s + jnωs ) (6.46a)
2 T n=−∞

where z = e sT .

Frame # 21 Slide # 35 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Summary of Interrelations Cont’d

I Recovery of a continuous-time signal by lowpass filtering an

impulse-modulated signal – frequency domain:

Y (jω) = H(jω)X̂ (jω) (6.48)

where (
T for |ω| < ωs /2
H(jω) =
0 for |ω| ≥ ωs /2

Frame # 22 Slide # 36 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Summary of Interrelations Cont’d

I Recovery of a continuous-time signal by lowpass filtering an

impulse-modulated signal – frequency domain:

Y (jω) = H(jω)X̂ (jω) (6.48)

where (
T for |ω| < ωs /2
H(jω) =
0 for |ω| ≥ ωs /2
I Recovery of a continuous-time signal by lowpass filtering an
impulse-modulated signal – time-domain:
∞
X sin[ωs (t − nT )/2]
y (t) = x(nT ) (6.51)
n=−∞
ωs (t − nT )/2

Frame # 22 Slide # 37 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

Graphical Representation of Interrelations

X(s)

L s → jω
L −1 jω → s

x(t) X( jω)

F −1

Eq. (6.42d) Eq. (6.45a)

Eq. (6.51) Eq. (6.48)
or (6.42e) or (6.45b)

F
ˆ
x(t) ˆ jω )
X(

F −1

Replace
Replace 1
impulses by z → e jωT jω → ln z
numbers by T
numbers
impulses

Z
x(nT ) XD(z)

Z −1

Frame # 23 Slide # 38 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8

 This slide concludes the presentation.
Thank you for your attention.

Frame # 24 Slide # 39 A. Antoniou Digital Signal Processing – Secs. 6.6-6.8