Sampling Theorem and Pulse Amplitude Modulation

(PAM)
Reference
Stremler, Communication Systems, Chapter 3.15, 7.1

I.1

Sampling Theorem
Signals bearing information are either in analog form,
discrete form or digital form.
Sample
and Hold

encoder

10110..

Analog-to-digital converter

Sampling theorem determines the necessary conditions

which allow us to change an analog signal to a discrete one,
or vice versa, without loss of information.
I.2

Sampling Theorem:
A real-valued band-limited signal having no spectral
components above a frequency of B Hz is determined
uniquely by its values at uniform interval spaced no greater
than 1/(2B) second apart.
f(t)

fs(t)

F()

B Freq (Hz)

1
2B

t
T : sampling period (second)
B : signal bandwidth (Hz)

i.e. taking more than or equal to 2B samples in each second

I.3

Example:
To convert a 10kHz sinusoidal signal to digital form, the
minimum sampling frequency is 20kHz.
To convert a voice signal (0-3.3kHz) to digital form, the
minimum sampling frequency is 6.6kHz.
In practice, sampling frequency = 8kHz.
To convert an audio signal (0-20kHz) to digital form, the
minimum sampling frequency is 40kHz.
In practice, sampling frequency for encoding music into CD
is 44.1kHz.
I.4

 This is a sufficient condition such that an analog signal

can be reconstructed completely from a set of uniformly
spaced discrete samples in time.

Proof
Consider a band-limited signal f(t) having no spectral
components above B Hz.
f(t)

F()
-2B 2B

I.5

The signal is sampled using the periodic gate function pT(t).

As pT(t) is a periodic signal, it can be represented by a
Fourier series.
pT (t ) =
Pn =

P e

n =

o = 2 / T
Sa ( x) =

Sa(n / T )

pT (t )

jn o t

T 2T

sin x
x

Pn

2 o
I.6

The sampled signal f s (t ) is

f s (t ) = f (t ) pT (t )

= f (t ) Pn e jn ot
n =

Taking the Fourier transform, we have

Fs ( ) = F f (t ) Pn e jn ot
n =

P F {f (t )e

n =

jn o t

Pn F ( n o )

(Linearity)

Not a function of
(frequency translation property)

n =

f s (t )

Fs ( )

Po F ( )

2 o

P1 F ( o )
I.7

Therefore, the spectral density of the sampled signal f s (t ) is,

within a constant factor, exactly the same as that of f (t ). In
addition, it repeats itself periodically. The spectral density of
the original signal can be retrieved by using a LPF on Fs ( ) .

However, if the sampling period T >1/2B, the replicas of F ( )

will overlap and we cannot retrieve F ( ) from Fs ( ).

I.8

Fs ( )

Po F ( )

Fs ( )

Po F ( )

o = 2 / T > 4B

T = 1/ 2B

T < 1/ 2B
Fs ( )

o = 2 / T

o = 2 / T = 4B

Fs ( )

T > 1/ 2B

o = 2 / T < 4B

I.9

The maximum time interval T of sampling (=1/2B) is called

the Nyquist interval; its reciprocal (2B) is called the Nyquist
sampling frequency.

In practice, oversampling (T < 1/2B) is used.

we cannot build ideal lowpass filter. If the filter
characteristics has a finite slope at the band edges,
frequency components from the spectral replicas may be
transmitted through the filter.

I.10

 A time-limited signal is never strictly band-limited. When

such a signal is sampled, there will be some unavoidable
overlap of spectral components. In reconstruction of the
signal, frequency components originally located above
one-half the sampling frequency will appear below this
point. This is known as aliasing.
Fs ( )

o = 2 / T

I.11

Pulse Amplitude Modulation (PAM)

In pulse amplitude modulation (PAM) the amplitude of a
train of constant-width pulses is varied in proportional to the
sample values of the modulating signal.

f PAM (t )

f(t)

I.12

Generating a PAM signal could be divided into two

processes: sampling and holding
Sampling: Consider a lowpass signal f (t ) that is bandlimited to f m and multiplied by a periodic train of very
narrow rectangular pulses pT (t ). The sampling interval T is
taken as the Nyquist interval 1 / 2 f m seconds.
F()

f(t)

m m

PT ( )

pT (t )

2 / T

I.13

The sampled signal is

f s (t ) = f (t ) pT (t )

= f (t ) Pn e jn ot
n =

where o = 2 / T and Pn = 1 / T

Taking the Fourier transform, we have

Fs ( ) = F { f (t ) pT (t )}

= F f (t ) Pn e jn ot
n =

P F {f (t )e

n =

jn o t

1
F ( n o )
T n =

}
I.14

Fs ( )

fs(t)

2 / T

Holding (lengthening): achieved by applying the sampled

signal to a time-invariant filter with unit impulse response

f (t )

f s (t )

q (t )

f PAM (t )

q (t )

I.15

Q ( )

Q ( ) = Sa ( / 2)

2 /

f PAM (t ) = f s (t ) q(t )

= [ f (t ) pT (t )] q(t )

= f (t ) (t nT ) q (t )
n =

= f (nT ) (t nT ) q (t )
n =

f (nT )[ (t nT ) q (t )]

f PAM (t )

n =

f (nT )q(t nT )

n =

I.16

The spectral density of the PAM signal is

FPAM ( ) = F { f s (t ) q (t )}
= Fs ( )Q( )
=

1
F ( n o )Q( )
T n =

FPAM ( )

2 /

1
F ( )Q( )
T

I.17

The spectrum obtained here is not the same as that

obtained in I.7.
In I.7 the spectrum consists of F ( ) and its replicas at
multiples of the sampling frequency with only a gain
variation of each spectral replica (i.e. P0 F ( ) ).
The present spectrum describes a point-to-point F ( )
multiplication in frequency so that the spectral density
has lost its original shape (i.e. Q ( ) F ( )
). This
distortion is dependent on the pulse shape; at low
frequencies it is not severe if the pulse width is very
narrow.
I.18