Institute of Integrated Information Systems

5. CONTINUOUS & DISCRETE SYSTEMS

• Discrete systems (digitally-implemented systems) have many parallels with
continuous systems.
• Basic notation:

OUTPUT
INPUT ANALOGUE SYSTEM
x (t) y (t)

h (t)

INPUT OUTPUT
DISCRETE SYSTEM
x (nT) or x [n] y (nT) or y [n]

h (nT) or h [n]

• Basic elements:
x1 [n]
(i) Summer ⊕ x1 [n] + x2 [n]
x2 [n]

a
(ii) Multiplier x [n] a x [n]

(iii) Delay x [n] z-1 x [n-1]

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Consider now Sampling & Quantisation and their Effects

5.1 Sampling
Essential in the conversion of analogue signals to digital form and vice-versa.
Consider the following arrangement:

Sampler can be represented as a multiplier. Thus, sampled signal is given by:

∞
x * (t ) = x(t ) f s (t ) = x(t ) ∑δ (t − mT )
m = −∞
input infinite series of
impulsive sampling pulses

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(i) Sampling and the δ-Function

Terminology:

δ ( t ) = impulse at δ (t − τ ) =
time origin impulse at t = τ

τ
The strength (area) of a unit impulse response is unity.

The product of a waveform x(t) and a unit impulse δ(t) can be viewed as a “masking”
process:
∞ ∞
and ∫−∞
δ (t )x(t )dt =x(0) ∫−∞
δ (t − τ )x(t )dt =x(τ )

τ+
since: ∫τ
and
−
δ ( t − τ ) dt = 1 δ (t − τ ) = 0 for all t ≠ τ

Also, from basic Fourier Transform theory:

and δ (t ) ⇔ 1 Aδ ( t ) ⇔ A

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(ii) Fourier Series for Sampling Signal

fs(t) is periodic with period T and can therefore be represented by a Fourier Series:
∞
f s (t ) =
=−∞
∑ δ ( t − mT )
m is an integer. Using complex exponential mform:

∞
f s (t ) = ∑
n =−∞
1
T e jnω s t
n is an integer, where:
ω s = 2π T
∞
1
∴x * (t ) =
T
∑ x(t )e
n = −∞
jnω s t
( 6)
Using standard Fourier result: (shift)
where: f ( t ) e bt ⇔ F ( jω − b )
f ( t ) ⇔ F ( jω )
Taking Fourier Transform of both sides of (6) gives:

∞
1
X * ( jω ) =
T
∑ X[ j ( ω − nω )] s
where: n =−∞

x ( t ) ⇔ X ( jω )

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(iii) Spectrum of Sampled Signal

∞
Recall: 1
X * ( jω ) =
T
∑ X[ j ( ω − nω )]
n =−∞
s

x ( t ) ⇔ X ( jω )
If X(jω) has the form:

The spectrum of x*(t) is:

i.e. X*(jω) is periodic with period ωs in the frequency domain.

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(iii) Spectrum of Sampled Signal (contd.)

From Fourier Theory:
 multiplication   convolution of 
 of waveforms in  ⇔  spectra in the 
 the time domain   frequency domain 

Sampling operation in the time domain described by:

x *( t ) = f s ( t ) x ( t )
and in the frequency domain by:
X *( jω ) = Fs ( jω ) ⊗ X ( jω )
i.e. convolution of sampling signal and input spectra.

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5.2 Sampling (Nyquist) Theorem

Sampled signal spectrum:

(a) If ωs > 2ωc , then spectral components do not overlap and X(jω) can be recovered by
low-pass filtering.
(b) At ωs = 2ωc , LPF needs to be an ideal “brick wall” filter.
(c) If ωs < 2ωc , spectral components overlap and “aliasing” occurs:

Signal cannot then be recovered unambiguously from its samples and sampling
becomes irreversible.

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5.2 Sampling (Nyquist) Theorem (contd.)

In its simplest form, the sampling theorem states that a waveform should be sampled at a
rate which is at least twice its highest significant frequency component if it is to be
recoverable from the samples. This applies to low-pass situations, i.e.

A more general result applies to band-pass signals, i.e.

Here: or: where W is the signal “bandwidth”.

ω s ≥ 2 (ω H − ω L ) ω s ≥ 2W
In practice, sampling rates must be above the minimum to allow for:
(a) non-brick wall spectra and filters;
(b) non-ideal sampling pulses.

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5.3 Quantisation
This is also an essential aspect of the process of analogue-to-digital conversion.
Approximates a continuous signal x(t) with a discrete-level signal xQ(t), e.g.

+ ∆V
2

− ∆V
2

− ∆V ∆V
It is seen that: ≤ e( t ) ≤
2 2
 ∆V   ∆V 
The quantiser output level is mi when:  mi −  ≤ x ( t ) ≤  mi + 
 2   2 

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5.3 Quantisation (contd.)

The mean square error voltage associated with level mi is:
mi + ∆V 2
e = ∫ ( x − mi ) 2 p( x ) dx
2
i
mi − ∆V 2

where p(x) is the amplitude PDF of x(t).

If ∆V << [ Total amplitude range of x(t) ] , then: p(x) ≈ p(mi )

p( mi )

mi
(mi − ∆2V ) (mi + ∆2V )
i.e. PDF remains approximately constant over interval ∆V centred on mi. Hence, mean
square error for i th level:
mi + ∆V 2
e = p( mi ) ∫
2
i ( x − mi )2 dx
mi − ∆V 2

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5.3 Quantisation (contd.)

mi + ∆V 2
ei2 = p( mi ) ∫ ( x − mi )2 dx
mi − ∆V
∆V
2

Let (x - mi ) = y ; thus dx = dy and limits of integration become: ±

∆V 2
∆V
 y3  2
ei2 = p( mi ) ∫− ∆V y 2 dy =
2
p( mi )  
2
 3  − ∆V 2
( ∆V )3
= p( mi )
12
Total mean square error for all levels:
1
e2 = ∑
12 i
( ∆V ) 3 p( mi )
1
= ( ∆V ) 2 ∑ [ ∆V p( mi ) ]
12 i

[ ∆V p(mi) ] is approximate area of strip of width ∆V centred on mi .

∴ ∑ [ ∆V p(m ) ]
i
i ≈ total area under PDF curve
=1

Hence: ( ∆V ) 2
e = 2

12
for a “linear” (equi-interval) quantiser.

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5.4 Quantisation Signal-to-Noise Ratios (SNRs)

Assume that the quantiser has q levels, and that peaks of input signal always match the
complete quantiser input range. Hence, if x(t) has peak amplitude ±A, then:
1
A= [ q (∆V ) ]
2
The ratio of:
 Peak input signal level 
 RMS quantisation noise 
 
is taken as a measure of quantisation amplitude SNR. The power SNR will be the square
of this.
Consider different input types:

(a) Sinusoid
For sinusoidal input, output will also be approximately sinusoidal if q>>1.

Mean square output: 2 2

q   1 
≈  (∆V )   
2   2
Hence power SNR: 2
 (∆V )
2
1 q
γ =  (∆V )  ÷
2 2  12
3
= q2
2

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5.4 Quantisation SNRs (contd.)

(b) Uniform Amplitude Distribution (e.g. ramp or triangular waveform)
Mean power of quantiser output, taken over complete output range of q levels, is the
mean of the individual level powers, i.e.
q 2
1  i 
Total mean power =
q
∑
i =1
 2 ( ∆V ) 

( ∆V ) 2 q 2
=
4q i=1
∑i
This is a standard series summation with the value:
q
q ( q + 1) ( 2q + 1)
∑ i2 =
i =1 6
q
q3
If q >> 1 ∑
i =1
i ≈
2

3
( ∆V ) 2 q 3 ( ∆V ) 2 2
∴ Mean Power = = q
4q 3 12

( ∆V ) 2 2 ( ∆V ) 2
Hence: γ = q ÷
12 12
= q2

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5.4 Quantisation SNRs (contd.)

(c) Rectangular Input (Equal mark:space)

q 
Two levels at extremes of quantiser range: ±  ( ∆V )
2 
q2 ( ∆V ) 2
Hence: γ = ( ∆V ) ÷
2
= 3q2
4 12

(d) Gaussian Input (e.g. speech)

Approximation: Peak ≈ 4 x (RMS Value)
∴ Limit quantiser input amplitude to: 4 x (RMS Value). Thus:
1 q
RMS = ( ∆V )
4 2
1 2
Mean power = q ( ∆V ) 2
64
1 2 ( ∆V ) 2
and γ = q ( ∆V ) ÷
2

64 12
3 2
= q
16

γ can therefore be calculated for different input types and different values of q.

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