BEng(Hons) Telecommunications

Cohort: BTEL/10B/FT & BTEL/09/FT

Examinations for 2011 - 2012 / Semester 2

Resit Examinations for 2011 - 2012 /

Semester 1
MODULE: DIGITAL SIGNAL PROCESSING

MODULE CODE: SCG 2123

Duration: 3 Hours

Reading Time: 15 minutes

Instructions to Candidates:

1. This question paper contains six (6) questions.

2. All questions carry equal marks.
3. Answer any five (5) questions.
4. Total marks 100.

This question paper contains 6 questions and 9 pages.

Page 1 of 9
SJL 2011-2012 S2
QUESTION 1: (20 MARKS)

(a) An analogue signal x(t) / V is defined mathematically as:

x(t ) 0.6 sin(400t ) 0.8 cos(600t / 4)

(i) Sketch the magnitude spectrum of x(t) in the frequency range [-400,
400] Hz. (2 marks)

(ii) State the minimum sampling frequency for x(t), according to Nyquist
Sampling Theorem. If x(t) is sampled at this minimum sampling
frequency, write down an equation for the sampled signal x(n).
(3 marks)

(iii) x(t) is sampled at a sampling frequency of 500 Hz. In an attempt to

recover the original analogue signal, the sampled signal is passed
through an ideal digital lowpass filter with cutoff frequency radians.
Derive a mathematical equation for the received signal and hence
sketch a graph of the received signal in time domain. Briefly discuss
the difference between the original analogue signal and the received
signal. (7 marks)

(b) Sketch the magnitude spectrum and derive the minimum sampling rate in
Hertz, based on the Nyquist criterion, for each of the following continuous
time signals:
(i) x1 (t ) cos(500t ) (2 marks)
(ii) x2 (t ) 100 sinc(100t ) cos(120t ) (3 marks)
(iii) x3 (t ) 100 sinc(100t ) 200sinc(200t ) (3 marks)

Page 2 of 9
SJL 2011-2012 S2
QUESTION 2: (20 MARKS)

(a) Let x(n) = u(n + 1) u(n 4) + 0.5(n 4). Sketch x(n) and then sketch the
following:
(i) x(2n + 1)
(ii) x(n)u(2 n)
(iii) x(n 1)(n 3)
(iv) 0.5x(n) + 0.5(-1)nx(n)
(v) x(n2). (6 marks)

Note: (n) and u(n) are the impulse and unit step sequences respectively
(i.e. (n) = 1 if n = 0 and (n) = 0 otherwise, and u(n) = 1 if n 0 and u(n) =
0 otherwise.)

(b) A linear time-invariant filter is described by the linear difference equation:

y(n) = x(n) + x(n - 1) + x(n 2) + x(n 3)
(i) Derive a simple expression for H(ej), the frequency response of this
system. (4 marks)

(ii) Sketch the magnitude and phase responses for .

(5 marks)
(iii) Determine the steady state output if the input is:
x(n) = 4 + cos[0.5 (n 1)] 3cos[0.25 n]
(5 marks)

Page 3 of 9
SJL 2011-2012 S2
QUESTION 3: (20 MARKS)

(a) Consider three linear time invariant discrete-time systems whose impulse
responses are given by:
h1[n] [n] 2 [n 1], h2 [n] 2 [n] [n 2],
h3[n] 4 [n] 2 [n 1] [n 3] [n 5]

where [n] is the unit impulse. Determine the impulse response, h[n], of the
system shown in Figure Q3a.

Input
sequence
h1[n] h2[n] Output
sequence

h3[n]

Figure Q3a (6 marks)

(b) Starting from the convolution sum relationship between the input x(n) and
the output y(n) of a discrete-time system,

y ( n) h( k ) x ( n k )
k

and from the definition of H (e j ) as the steady state response to a

sinusoid, derive the Discrete Time Fourier Transform of h(n) as:

H (e j ) h ( n )e
n
jn

Hence prove that Y (e j ) H (e j ) X (e j ) (6 marks)

Page 4 of 9
SJL 2011-2012 S2
(c) Let h(n) represent the impulse response of a five-point running average
digital filter.
1
, 0n4
h( n) 5

0, otherwise

where y ( n) x(k )h(n k ) . Derive expressions for H (e j ) and
k

( ) H (e j ) , and from these roughly sketch H (e j ) and ( ) for

, labelling all the axes appropriately. (8 marks)

QUESTION 4: (20 MARKS)

(a) Consider the following linear difference equation describing the relationship
between the input and output of an IIR digital filter:

y[n] x[n] 0.96 y[n 1] 0.9216 y[n 2]

(i) Derive the z-transfer function of the filter, H(z). (2 marks)

(ii) Estimate the values of all the poles and zeros in the form re j , and
mark all the poles and zeros on a pole/zero plot in the z-plane.
(4 marks)

(iii) Sketch H (e j ) , 0 clearly identifying the resonant frequency of

the circuit, r. What is the resonant frequency in Hz if the sampling

frequency is fs = 3000 Hz ? (4 marks)

Page 5 of 9
SJL 2011-2012 S2
(b) Let the transfer function of a digital filter be :

1
H ( z)
1 0.96 z 0.9216 z 2
1

We would like to modify the above filter to obtain an all-pass filter. The
transfer function of the modified filter is given by:

z 2 H ( z )
G( z )
H ( z 1 )

(i) Determine the corresponding linear difference equation. (2 marks)

(ii) Mark all the poles and zeros of G(z) on a pole/zero plot in the z-plane.
What is the relationship between the obtained poles and zeros?
(4 marks)

(iii) Prove that the filter is an all-pass filter, i.e. the magnitude of the
frequency response is a constant function of the frequency.
(2 marks)

(iv) Let ( ) denote the phase of H (e j ) . Express the phase of G(e j ) as

a function of ( ) . (2 marks)

Page 6 of 9
SJL 2011-2012 S2
QUESTION 5: (20 MARKS)

(a) An FIR filter is characterised by the following frequency response:

sin(5 ) j 5
H (e j ) e
1
sin( )
2
The input x(n) to the filter is a signal with the spectrum in Figure Q5a.

11e j / 4 11e j / 4
-7j 7j

/3 /5 0 /5 /3
Figure Q5a

(i) Determine a formula for the input signal x(n), for n .

(4 marks)
(ii) Using the input signal x(n), determine the output y(n) for n .
(6 marks)

(b) The frequency response of a linear time-invariant filter is given by the

formula:
H (e j ) (1 e j )(1 e j / 4 e j )(1 e j / 4 e j )

(i) Write the difference equation that gives the relation between the input
x(n) and the output y(n) . (5 marks)

(ii) What is the output if the input is x(n) (n) ? (1 mark)

(iii) If the input is of the form x[n] Ae j e jn , for what values of ,

will y(n) 0 for all n? (4 marks)

Page 7 of 9
SJL 2011-2012 S2
QUESTION 6: (20 MARKS)

(a) Consider an ideal digital linear time invariant filter whose frequency
response is
1 if / 2
H I (e j )
0 otherwise

(i) Determine and sketch the impulse response, hI[n], of the ideal filter,
using the inverse Discrete-Time Fourier transform of H I (e j ) .
(4 marks)

(ii) Now, we consider the truncated impulse response given by h[n] =

hI[n] if 2 n 2 and h[n] = 0 otherwise. Determine the frequency
response of the resulting filter by using the Discrete Time Fourier
Transform of h[n]. (3 marks)

(iii) The continuous-time signal x(t) = cos(100 t) is sampled at frequency

fs = 600 Hz and fed to the truncated filter. Using h[n], show that the
steady state output of the filter is given by y[n]= Acos(2 n) where A
is a positive scalar and is a digital frequency (0 0.5) to be
determined. (4 marks)

Page 8 of 9
SJL 2011-2012 S2
(b) We wish to design a digital FIR lowpass filter using the window method.

(i) Describe the four steps of the window method. (2 marks)

(ii) In order to satisfy passband and stopband specifications, the Hanning

window (given below) and a filter length N = 9 are chosen. The cut-off
frequency of the ideal filter is taken to be c 4 . Derive the impulse

response of the causal FIR filter.

(7 marks)

Note that the coefficients of the Hanning window of length N are given by:

1 2n N 1 N 1
w(n) 1 cos , n
2 N 1 2 2

1
X (e
j
Inverse DTFT: x ( n) )e jn d
2

****END OF QUESTION PAPER****

Page 9 of 9
SJL 2011-2012 S2