Analog to Digital Conversion

Lecture No. 7

Dr. Aoife Moloney

School of Electronics and Communications

Dublin Institute of Technology
Lecture No. 7: Analog to Digital Conversion

Overview
• Now that we have finished the maths and theory we will
have a look at baseband transmission. The following top-
ics will be covered:
– Analog to digital conversion
– Line coding
– Detection of baseband signals in noise
– Intersymbol interference (ISI)
• The next 2 lectures will look at analog to digital conver-
sion and the following:
February 2005 Slide: 1
Lecture No. 7: Analog to Digital Conversion

– Sampling
– Quantisation
– PCM encoding

February 2005 Slide: 2

Lecture No. 7: Analog to Digital Conversion

Analog to Digital Converter

• Remember the analog to digital converter (ADC) that we
met in the hypothetical transceiver in Lecture 1 (Com-
munications Engineering handout)
• ADCs will generally consist of a sampling circuit, a quan-
tiser and a pulse code modulator
ADC

Sampling

PCM
encoder

Quantisation

February 2005 Slide: 3

Lecture No. 7: Analog to Digital Conversion

Sampling
• Sampling transforms a continuous waveform into a se-
quence of samples, with amplitudes derived from the in-
put waveform. This form of sampling is known as PAM
(pulse amplitude modulation) and is illustrated in the
diagram below.
PAM

February 2005 Slide: 4

Lecture No. 7: Analog to Digital Conversion

• Sampling Theorem: A signal having no spectral com-

ponents above fm Hz can be determined uniquely by val-
ues sampled at uniform intervals of Ts s (fs sampling
rate), where:
1
≥ 2fm or fs ≥ 2fm
Ts
Note: This equation is known as the Nyquist sampling
criterion and fs = 2fm is called the Nyquist rate
• Sampling can be represented as the product of the signal
to be sampled, X(t), with a unit–weight impulse train,

February 2005 Slide: 5

Lecture No. 7: Analog to Digital Conversion

Xδ (t), where:
∞
X
Xδ (t) = δ (t − nTs )
n=−∞

where, Ts is the sampling period of the signal

The sampled signal can be expressed as:

Xs (t) = X(t).Xδ (t)
∞
X
= X (nTs ) δ (t − nTs )
n=−∞

February 2005 Slide: 6

Lecture No. 7: Analog to Digital Conversion

This is illustrated in the diagram below. In the diagram

the weight (area) of each impulse, X(nTs ), is indicated
by the height of the impulse

The sampled signal can be examined in the frequency

February 2005 Slide: 7

Lecture No. 7: Analog to Digital Conversion

domain using the Fourier transform:

Xs (f ) = F T [X(t).Xδ (t)]
= X(f ) ⊗ Xδ (f )
∞
X
= X(f ) ⊗ fs δ (f − nfs )
n=−∞
∞
X
= fs X (f − nfs )
n=−∞

Note: The Fourier transform of an impulse response

February 2005 Slide: 8

Lecture No. 7: Analog to Digital Conversion

train is well known and is given by:

∞ ∞
FT
X X
δ (t − kTs ) ←→ fs δ (f − nfs )
k=−∞ n=−∞

The diagram shows the unsampled signal and its spec-

trum ((a)) and the sampled signal and its spectrum ((b)).

Note: The spectrum of the sampled signal is identical

to the spectrum of the unsampled signal, but repeated
every fs Hz

February 2005 Slide: 9

Lecture No. 7: Analog to Digital Conversion

X(t) W(f)

t -B B f
Ts

(a) Waveform and its spectrum

Low pass
filter TsXs(t)
Xs(t)

t -2fs -fs fs 2fs f

B

(b) Sampled waveform and its spectrum

February 2005 Slide: 10

Lecture No. 7: Analog to Digital Conversion

• Original Signal: As illustrated in the diagram, if fs ≥

2fm the replicated spectra do not overlap. Thus the orig-
inal unsampled signal can be regenerated by filtering the
sampled signal with a low pass filter selecting the base-
band spectrum, as shown in the diagram.
• Aliasing: If fs < 2fm the waveform will be undersam-
pled and the replicated spectra of the sampled signal will
overlap. The spectral overlap is called aliasing. The re-
ceovered signal will be distorted due to the aliasing. As
shown in the diagram.

February 2005 Slide: 11

Lecture No. 7: Analog to Digital Conversion

To avoid aliasing anti–aliasing filters can be introduced

pre or post sampling
– Pre–sampling filters will have cut–off frequencies of fs
– Post–sampling filters will have a cut–off frequency of
fs − fm

February 2005 Slide: 12

Lecture No. 7: Analog to Digital Conversion

W(f)

-B B f

(a) Spectrum of unsampled waveform

Low pass
filter TsXs(t)

-2fs -fs fs 2fs f

B

(b) Spectrum of sampled waveform, fs < 2fm (2B)

February 2005 Slide: 13

Lecture No. 7: Analog to Digital Conversion

Quantisation
• After sampling, the sampled signal is quantised. Each
pulse in the sampled signal is adjusted in amplitude to
coincide with the nearest of a finite set of allowed ampli-
tudes. The figure below shows an analogue signal and its
corresponding quantised signal, where the signals have
been sampled at a sampling rate fs (1/Ts ).

February 2005 Slide: 14

Lecture No. 7: Analog to Digital Conversion

Ts
+7q/2
q
+5q/2
Analogue
+3q/2
signal
+q/2
0
-q/2
Quantised
-3q/2
signal
-5q/2
-7q/2

• The step q between quantisation intervals is called the

quantile interval
• When the quantisation levels are uniformly distributed
over the full range of values taken by the sampled signal,
the quantiser is called uniform or linear
February 2005 Slide: 15
Lecture No. 7: Analog to Digital Conversion

• The difference between the analogue and quantised sig-

nals is random and can be thought of as a noise, known
as quantisation noise. The ratio of the power of this
noise to the peak signal power is known as the signal to
quantisation noise ratio (SNq R).
• Quantisation Noise: Denoting the quantisation error
(i.e. difference between analogue and quantised signals)
as e, then assuming linear quantisation (as shown above)

February 2005 Slide: 16

Lecture No. 7: Analog to Digital Conversion

it follows that the pdf of e, p(e), is uniform and given by:


 1/q, for −q/2 ≤ e ≤ q/2;
p(e) =
 0, elsewhere;

The mean–square quantisation error or noise is therefore:

Zq/2
e2 = e2 p(e)de
−q/2

i.e.
2
q
e2 =
12
February 2005 Slide: 17
Lecture No. 7: Analog to Digital Conversion

• Signal to Quantisation Noise Ratio (SNq R): If L is

the number of quantisation levels the peak analogue (i.e.
unquantised) signal level is Lq/2. The peak power of the
signal (normalised to 1 Ω) is therefore:
 2  2  2 2 
2 Vpp Lq Lq
Vp = = =
2 2 4
The peak signal power to average quantisation noise power,

February 2005 Slide: 18

Lecture No. 7: Analog to Digital Conversion

SNq R is therefore:
 
L2 q 2
4
SNq R =  2  = 3L
q
12

February 2005 Slide: 19

Lecture No. 7: Analog to Digital Conversion

Conclusion
This lecture has looked at the following:
• Analog to Digital converter (ADC)
• Sampling
• Sampling theorem
• Linear quantisation
• Signal to quantisation noise ratio (SNq R)

February 2005 Slide: 20