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

1
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.

1
fs(t) T≤
f(t) F(ω) 2B

t B Freq (Hz) T 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

2
 – 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(ω)

ω
t -2πB 2πB

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 ) = ∑P e
n = −∞
n
jnω o t
ω o = 2π / T

τ sin x
Pn = Sa(nπτ / T ) Sa ( x) =
T x

pT (t ) τ Pn

t ωo ω
T T 2T
2ω o

I.6

3
The sampled signal f s (t ) is
f s (t ) = f (t ) pT (t )
∞
= f (t ) ∑ Pn e jnω ot
n = −∞

Taking the Fourier transform, we have

 ∞

Fs (ω ) = F  f (t ) ∑ Pn e jnω ot 
 n = −∞ 

∑ P F {f (t )e }
∞
jnω o t
= n (Linearity)
n = −∞
∞
Not a function of ω
= ∑ Pn F (ω − nω o )
n = −∞
(frequency translation property)

f s (t ) Fs (ω ) Po F (ω ) P1 F (ω − ω o )
τ
t ωo
2ω o
ω 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 (ω ) .

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

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

I.8

4
 Fs (ω ) Po F (ω ) Fs (ω ) Po F (ω )

ω ω
ω o = 2π / T > 4πB ω o = 2π / T = 4πB

T < 1/ 2B T = 1/ 2B

Fs (ω ) Fs (ω )

ω ω
ω o = 2π / T ω o = 2π / T < 4πB

T > 1/ 2B 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

5
– 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 (ω )

ω
ω o = 2π / 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)

t t

I.12

6
 Generating a PAM signal could be divided into two
processes: sampling and holding

Sampling: Consider a lowpass signal f (t ) that is band-

limited 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(t) F(ω)

− ωm ωm ω
t
pT (t ) PT (ω )

t ω
T 2π / T I.13

The sampled signal is

f s (t ) = f (t ) pT (t )
∞
= f (t ) ∑ Pn e jnω ot where ω o = 2π / T and Pn = 1 / T
n = −∞

Taking the Fourier transform, we have

Fs (ω ) = F { f (t ) pT (t )}
 ∞

= F  f (t ) ∑ Pn e jnω ot 
 n = −∞ 

∑ P F {f (t )e }
∞
jnω o t
= n
n = −∞

1 ∞
= ∑ F (ω − nω o )
T n = −∞
I.14

7
 fs(t) Fs (ω )

t ω
2π / T

– Holding (lengthening): achieved by applying the sampled

signal to a time-invariant filter with unit impulse response

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

τ t
I.15

Q (ω ) = τSa (ωτ / 2)
Q (ω )
ω
2π / τ
f PAM (t ) = f s (t ) ⊗ q(t )
= [ f (t ) pT (t )] ⊗ q(t )
 ∞

=  f (t ) ∑ δ (t − nT ) ⊗ q (t )
 n = −∞  f PAM (t )
 ∞

=  ∑ f (nT )δ (t − nT )  ⊗ q (t )
n = −∞ 
∞ t
= ∑ f (nT )[δ (t − nT ) ⊗ q (t )]
n = −∞
∞
= ∑ f (nT )q(t − nT )
n = −∞
I.16

8
The spectral density of the PAM signal is
FPAM (ω ) = F { f s (t ) ⊗ q (t )}
= Fs (ω )Q(ω )
1 ∞
= ∑ F (ω − nω o )Q(ω )
T n = −∞

FPAM (ω )
ω
2π / τ

1
F (ω )Q(ω )
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 (ω ) and its replicas at
multiples of the sampling frequency with only a gain
variation of each spectral replica (i.e. P0 F (ω ) ).

• The present spectrum describes a point-to-point F (ω )

multiplication in frequency so that the spectral density
has lost its original shape (i.e. Q (ω ) F (ω ) ). This
distortion is dependent on the pulse shape; at low
frequencies it is not severe if the pulse width is very
narrow.
I.18