TE312: Introduction to Digital

Telecommunications
Lecture #2
Sampling Theory
Introduction
Points to be discussed in this lecture:
Instantaneous (Ideal) Sampling
Nyquist Sampling Theorem
Natural Sampling
Flat-Top Sampling
Practical Considerations
Introduction
Reading Assignment
1. Simon Haykin, Digital Communications, J ohn
Wiley & Sons, Inc., 1988, Chapter 4, Sec. 4.1-4.5.
2. Bernard Sklar, Digital Communications:
Fundamentals and Applications, 2
nd
Ed., Prentice
Hall PTR, 2001, Chapter 2, Sec. 2.4.
3. A. Bruce Carlson, Paul B. Crilly, and J anet C.
Rutledge, Communication Systems: An
Introduction to Signals and Noise in Electrical
Communication, 4
th
Ed., McGraw Hill, 2002,
Chapter 6, Sec. 6.1.
Instantaneous (Ideal) Sampling

In most cases, message signals are generated
by analog information sources e.g., audio,
video, etc. and the signals are continuous in
time and amplitude (analog).

Analog message signals are converted to
digital form by an analog-to-digital (A/D)
converter, which involves sampling to produce
a discrete time signal, quantizing to generate a
discrete-amplitude signal and encoding to
produce a sequence of bits.

Instantaneous (Ideal) Sampling

Consider a signal
( )
s
x t obtained by
instantaneous sampling a signal
( )
at a
periodic interval
x t
s
T .
s
T is sampling period and
1
s
s
f
T
= is sampling rate.










( )
x t
t
( )
s
x t
t
s
T
s
T
Instantaneous (Ideal) Sampling

The instantaneous sampled signal
( )
s
x t can be
expressed in time domain by

( ) ( ) ( )
s s s
n
x t x nT t nT

=
=

The can be obtained by taking a product of
( )
and a periodic impulse train
( )
s
x t
x t
( )
s
T
t
(switching function)



Instantaneous (Ideal) Sampling



( ) ( ) ( )
s
s T
x t t x t =
( )
x t









( ) ( ) ( )
( ) ( )

s
s T
s
n
x t x t t
x t t nT

=
=
=

( ) ( )
s
T s
n
t t nT

=
=

Instantaneous (Ideal) Sampling

Multiplication in time domain transforms to
convolution in frequency domain (frequency
convolution property). Thus

( ) ( )
1 1
1 1

s
n
s s
n
s s
X f X f f n
T T
X f n
T T

=

=



=

1.
( )
s
X f is a continuous spectrum.
2.
( )
s
X f is periodic with a period equal to
1
s
T

Instantaneous (Ideal) Sampling

Alternatively,
( )
s
X f can be derived using the
time-shift property of Fourier transform i.e.

( ) ( ) ( )
1
exp 2
s s s
n
s
X f x nT j fnT
T

=
=

This is the complex exponential Fourier series
expansion of the spectrum of
( )
since
s
x t
( )
s
X f
is periodic in f with the coefficients
( )
s
x nT .



Sampling Theorem

Let the spectrum of
( )
x t be strictly band-limited
to Hz B as shown below


( )
| | X f


f



Let the sampling period
1
2
s
T
B
= , which implies
that the sampling rate 2
s
f B = .

B B
Sampling Theorem

In this case,

( ) ( )
2 2
s
n
X f B X f Bn

=
=

f
( )
| |
s
X f
B B
2B
2B 3B






3B


Sampling Theorem

Therefore, the spectrum of
( )
x t ,
( )
X f can be
derived from
( )
s
X f as

( ) ( )
1

2
1
exp
2 2
s
n
X f X f B f B
B
n nf
x j B f B
B B B

=
=

=

It can be concluded that
( )
X f is uniquely
determined by sample values
( )
/2 x n B and so is
( )
. x t

Sampling Theorem

Reconstruction of
( )
x t from
)
can be derived
from the inverse Fourier transform of
(
s
x t
( )
X f .

( ) ( ) ( )
( )
( )
exp 2
1
exp exp 2
2 2
sinc 2
2
B
B
n
n
x t X f j ft df
n nf
x j j ft df
B B B
n
x Bt n
B

=
=

=



=

This is an interpolation formula.

Sampling Theorem

Sampling theorem for band-limited signals of
finite energy states that:

1. A signal of finite energy band-limited to
Hz B is completely described by specifying
the values of the signal at instants of time
separated by 1/2B sec.

2. A signal of finite energy band-limited to
Hz B may be completely recovered from its
samples taken at the rate 2B per second,
which is called the Nyquist rate.
Sampling Theorem

Reconstruction of
( )
x t from
)
is implemented
by means of a low-pass filter (LPF) called
reconstruction filter.
(
s
x t

For realizable reconstruction LPF, the
sampling rate
s
f must be larger than . 2B

If 2
s
f B < , frequency-shifted versions of
( )
X f
overlap. This results in aliasing effect and the
original signal
( )
cannot be recovered from
( )

x t
s
x t

Sampling Theorem
( )
| |































f
( )
| |
s
X f
B B
2B
2
s
f B < 2
s
f B >
s
X f
B B
2B
2
s
f B < 2
s
f B >
f
Natural Sampling

A natural sampled signal
( )
s
x t can be
expressed in time domain by (from practical
switching circuits)

( ) ( ) ( )
( )
1 0
0 otherwise
s s
n
x t x t h t nT
t
h t

=
=

The switching signal
) ( ) (
s
n
p t h t nT

=
=

is a
periodic rectangular pulse train.

Natural Sampling








( )
x t
( ) ( ) ( )
s
x t p t x t =












( ) ( )
s
n
p t h t nT

=
=

Natural Sampling




























( )
x t
t
t
( )
s t
t
( )
p t
Natural Sampling

Complex exponential Fourier series of the
switching signal
( )
p t is expresses as:

( ) ( ) ( ) ( )
exp 2 , = sinc exp
n s n s s s
n
p t c j nf t c f n f j nf t

=
=

Using the Fourier series expression for
( )
p t , it
follows that

( ) ( ) ( )
exp 2
s n s
n
x t x t c j nf t

=
=




Natural Sampling

From the frequency-shifting property, Fourier
transform of
( )
s
x t becomes

( ) ( )
s n s
n
X f c X f nf

=
=

Note:
(
1.
)
consists of an infinite number of copies
of
( )
shifted every
s
X f
X f Hz
s
f .
2. The copy is scaled by .
th
n
n
c



Natural Sampling








f
B B 3B 3B
( )
| |
s
X f
Reconstruction of
( )
x t from
)
is implemented
by means of a low-pass filter (LPF) called
reconstruction filter if 2
(
s
x t
s
f B .

Note that .
0
1.0 c =
Flat-Top Sampling

Flat-top sampling of a signal
( )
x t is obtained by
instantaneous sampling at every sampling
period
s
T and holding the sample value for
duration of (e.g. sample-and-hold circuit) sec. T

The flat-top sampled signal is defined by
( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( )



s s s
n
s s
n
s
n
x t x nT g t nT
g t x nT t nT
g t x t t nT

=
=
=
=

Flat-Top Sampling

( )
s
x nT : n-th sample of the message signal
( )
x t .
s
T : sampling period
1/ 2
s
T B: sampling rate


( )
1, 0
0, elsewhere
t T
g t

The Fourier transform of
( )
s
x t is given by
( ) ( ) ( )
s s s
n
X f f G f X f nf

=
=

( ) ( ) ( )
sinc exp G f T fT j fT =
Flat-Top Sampling
































( )
( ) ( ) ( )
s
s
T
x t
t x t g t
=



( )
g t
( )
x t
( ) ( )
s
T s
n
t t nT

=
=

Flat-Top Sampling










( )
X f
( )
s
x t
t






















f
( )
s
X f
f
1/T 1/T
Flat-Top Sampling

Recovery of
( )
x t is accomplished by the
reconstruction LPF.

The reconstructed signal (LPF output) is
processed by an equalizer to minimize the
aperture effect caused by
( )
G f . The frequency
response of the equalizer is given by
( )
eq
H f

( )
( )
sinc
s
eq
T
H f
T Tf
=


Practical Considerations
Practical signals are time-limited which implies
that they are not band-limited.

To avoid aliasing, an anti-aliasing (prefilter)
low-pass filter processes a signal with cut-off
frequency equal to half the Nyquist rate.

Realizable filters require a nonzero transition
bandwidth which implies 2
s
f B > (i.e sampling
rate
s
f must be much larger than baseband
signal bandwidth B.)

Practical Considerations

To minimize the sampling rate, which implies
lower transmission rates and less storage
memory, small transition bandwidth of filters is
desired.

A good engineering balance is to allow a
transition bandwidth 20% of the baseband
signal bandwidth such that 2.2
s
f B .

44,100 samples/s is used for a high quality
compact disc (CD) digital audio system for a
music signal with bandwidth of 20 kHz.
Practical Considerations

A sampling rate of 8000 samples/s is used for
digital telephone systems for telephone quality
speech with bandwidth of 4 kHz.