SCHOOL OF COMPUTING & TECHNOLOGY

ELECTRICAL ENGINEERING FIELD

EXAMINATIONS

Module Code: EE3002

Module Title: Digital Signal Processing

Date: January 2008

Time:

INSTRUCTIONS TO CANDIDATES:

Answer FOUR out of SIX questions, all questions carry equal

marks.

Only FOUR questions will be marked. If you attempt more than

FOUR questions please cross out the answers that you do not
wish to be marked, otherwise the FIRST FOUR answers in the
order they appear in your answer book will be marked.
SUBJECT: EE3002 Digital Signal Processing

Q1 Digital Infinite Impulse Response (IIR) filters have an advantage over Finite Impulse
Response (FIR) filters in that they require fewer multiplications in a time domain
realisation. A disadvantage is that they are non-linear phase filters. Two popular IIR
filter are the Butterworth and Chebyshev. Design the digital filters from the
specifications given below, in both cases using the Bilinear z-Transformation (BZT),
by determining their respective difference equations.

a) The differential equation of a Chebyshev low-pass filter is given by the

expression below:

d2y dy
2
+ 1.4526 + 1.5161 y = 1.5161x
dt dt

The equation is normalised to the pass-band, cut-off frequency ωp=1.

The filter is to be implemented digitally for the following specification:

Pass-band 0-2 kHz

Pass-band attenuation (αmax) 0.5 dB
Sampling Rate 60 kHz

(i) Verify the differential equation from the specification

(8 marks)

(ii) Obtain the difference equation for the digital realisation

(5 marks)

b) A low-pass Butterworth filter can be described by

d2y dy
2
+ 2 +y=x
dt dt
where the equation is normalised to ω 0 = 1

i) Obtain the transfer function corresponding to the above differential

equation.
(3 marks)

ii) Given that the filter has the following pass-band requirements:

Pass-band: 0 - 1 kHz
ass-band ripple (αmax): 0.8 dB
Sampling frequency: 60 kHz

Use the bilinear z-transform to obtain the difference equation of a digital

realisation for the Butterworth filter defined above.
(9 marks)

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SUBJECT: EE3002 Digital Signal Processing

Q2. Digital recursive oscillators form an integral part of tone generators and digital
synthesisers. They are used in digital, touch-tone keypads. The tone generators can be
nd
implemented using a pair of programmable 2 order recursive oscillators.
a) A general differential equation describing an analogue recursive filter is given by:

d 2 y  ωo  dy
+  + ω02 y = ω02 x
dt 2  Q  dt
modify this expression in order to obtain the differential equation for an oscillator
and obtain the frequency domain s-plane transfer function of the oscillator.
(4 marks)
b) Using the backward difference approximation, determine the difference equation of
the digital oscillator and its associated z-plane transfer function, which have the
form:
yn = ayn −1 + byn − 2 + cxn

cz 2
H ( z) =
z 2 − az − b
(10 marks)
c) The plots shown in Figure Q2(c) are in the time domain and show the digital
oscillator output for values of the coefficient b in the equations of part Q2(b), given
by b = -1 in the first plot and b = -1.04 in the second plot.
i) Using the transfer function of Q2 (b) explain clearly with complete
mathematical justification the variation between the two plots for a
sampling rate of 50 kHz and a digital frequency of oscillation of 10 kHz.
(8 marks)
ii) Sketch the shape of the digital oscillator output for a value of b = -0.99 and
give a brief justification.
(3 marks)
2

am
1
plit
ud
e 0

-1

-2
0 20 40 60 80 100 120 140 160 180 200
b = -1
Figure Q1(c)
100

am
50
plit
ud
e 0
Figure Q2 (c)
-50

-100
0 20 40 60 80 100 120 140 160 180 200
b = -1.04 ; samples in time

Figure Q2 (c)
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SUBJECT: EE3002 Digital Signal Processing

Q3. The discrete Fourier transform (DFT) is a mathematical tool used in the analysis of
spectral content in various signals and in digital filtering.

(a) Use the diagram shown in Figure Q3 to perform the frequency domain circular
convolution in the of the two sequences

x(n) = [1 0 0 1]
h(n) = [1 3 2 1]

(8 marks)
x(n)

y(n)

h(n)

Figure Q3

(8 marks)

(b) A signal processing system operating at 100 MHz, is based on a floating-point

processor which can perform a multiply-and-add in one clock cycle. The system is
programmed to analyse cardiac data in the following manner:

(1) First the cardiac data is pre-filtered with a 32-tap, zero-phase, finite impulse
response (FIR) filter to remove the dc, then

(2) The now pre-filtered data is convoluted in the frequency domain using the
scheme of part Q3(a) in order to remove the respiratory cycle from the
cardiac signal, using a 1024-length FFT.

Estimate:

i) the total computation time for the above data analysis scheme described by
(1) and (2).
(7 marks)

ii) the percentage reduction in computation time if the FIR filter is replaced by
th
an equivalent 4 order IIR filter.
(10 marks)

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SUBJECT: EE3002 Digital Signal Processing

Q4. Convolution is arguably the most fundamental operation in digital signal

processing since it appears in almost all applications.

a) i) Explain why the frequency spectrum of a sampled signal, as

represented mathematically by the convolution summation, is
periodic with a period equal to the sampling frequency.
(6 marks)

ii) Explain the difference between circular and linear convolution

and hence indicate the type of signal for which each operation
would be most suited.
(3 marks)

iii) Given the two sequences: -

h[n] = [1, 3, 2, 1]
x[n] = [-2, 2, -1, 1]

calculate: -

the circular convolution of the two sequences

(3 marks)

and the linear convolution of the two sequences.

(3 marks)

b) i) Explain the difference between convolution and correlation and

hence show how correlation can be used in a radar system to
determine the range of a target.
(6 marks)

ii) Describe the relationship between the correlation function and

the covariance function and hence show that no information is
lost when the covariance function is used (a mathematically
rigorous answer is NOT required).
(4 marks)

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SUBJECT: EE3002 Digital Signal Processing

Q5. Digital signal processors (DSPs) can be described as microprocessors whose

architecture has been optimised for discrete time signal processing. With
respect to DSP architecture: -

a) Define the meaning of the terms dynamic range,

resolution and word length as applied to a digital
signal processor.
(5 marks)

b) The result register in a DSP is always larger than twice

the size of the input registers. Explain why this feature
is necessary.
(4 marks)

c) Explain the meaning of the term “Saturation” as

applied to the result register of the multiplier in a DSP
and show, by example, why such an instruction may
be necessary.
(5 marks)

d) With reference to circular buffers: -

i) Explain, with the aid of a diagram, the concept of

a circular buffer.
(5 marks)

ii) How is it implemented in a DSP?

(3 marks)

iii) Explain why such a hardware facility is useful in

the implementation of digital filters.
(3 marks)

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SUBJECT: EE3002 Digital Signal Processing

Q6. a) Describe the process of converting a continuous-time signal into a

discrete-time signal (sampling) and hence derive the expression for the
convolution summation.

(5 marks)

b) Explain the condition under which “aliasing” occurs and show by

example the effect that it has on a sampled signal.

(5 marks)

c) A discrete-time system produces the output sequence [1 2 3 1 2 ] in

response to a discrete impulse at the input.

i) Derive the discrete-time equation and the transfer

function of the system and draw the corresponding
signal flow graph (in any realisation of your choice).

(6 marks)

ii) If the system is driven by a ramp input of 4 samples duration

and a maximum value of 3, calculate the output sequence and
sketch the output waveform.

(6 marks)

d) State the conditions that must be met in order for a discrete time
system to be stable and hence show why a non-recursive is
unconditionally stable.
(3 marks)

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