Name : ………………………….

Student ID: ……………………….

Signature: …………………………

The University of New South Wales

School of Electrical Engineering and Telecommunications

Supplementary Examination

Summer 2011/2012

ELEC9721: Digital Signal Processing Theory and Applications

Candidates should attempt Questions 1, 2, 3 and either 4 or 5

1. Reading time: 10 minutes

2. Time allowed: 3 hours.
3. The paper contains 5 questions. All candidates should attempt Questions 1, 2
and 3.
4. Candidates should attempt one of either Question 4 or Question 5.
5. The questions are worth 100 marks in total.
6. Print your name, student ID and question number on the front page of each answer
book.
7. This exam contributes 60% of the total course assessment.
8. The questions are NOT of equal value. Part marks are as indicated.
9. The examination paper may NOT be retained by the candidate.
10. This is a closed book exam.
11. The use of UNSW approved calculators is permitted.
12. Transform tables are supplied for your use.
13. This paper contains 6 pages apart from this instruction page.

ANSWERS MUST BE WRITTEN IN INK. EXCEPT WHERE THEY ARE EXPRESSLY

REQUIRED, PENCILS MAY ONLY BE USED FOR DRAWING, SKETCHING OR
GRAPHICAL WORK.
 Information sheet

Table of the z transforms of useful sequences

Basic z-Transforms Sequence z-Transform

1. Unit sample δ [n] 1
δ [n-k] z -k

2. Unit step u[n] z/(z-1)

3. Exponential a u[n]
n
z/(z-a)

z sin θ 0
4. Sinusoidal sin[θ n] u[n]
z 2 − 2 cos θ 0 z + 1
o

z 2 − z cosθ 0
cos[θ n] u[n]
z 2 − 2 cosθ 0 z + 1
o

z
5. Unit ramp nu[n]
( z − 1) 2

−N
−1 −2 −3 −(N−1) 1− z
1+z +z +z + ...........+ z =
1 − z −1

z-Transform Properties Sequence z-Transform

x[n] X(z)
-k
x[n-k] z X(z)
an x[n] X(z/a)
x[-n] X(z-1)
x[n]*y[n] X(z).Y(z)
n x[n] -z (dx(z)/dz)

sin 2θ = 2sin θ cosθ 1 ⎡ jθ − jθ ⎤

cosθ = e +e
⎢
2 ⎣ ⎥⎦
sin3θ = 3sin θ - 4sin 3 θ
1 ⎡ jθ − jθ ⎤
sinθ = e −e
cos2θ = cos 2θ - sin 2θ 2j ⎢⎣ ⎥⎦

1
Fourier Series:
∞
A0 1
x (t ) =
2
+ ∑ An cos(nω t ) + B
n =1
0 n sin(nω 0 t ); ω 0 = 2πf 0 ; f 0 =
T

The fundamental frequency of x(t) is ω0 and T is the period of the signal x(t).

T /2 T /2
2 2
An = x(t ) cos(nω0t ) dt n = 0,1,2.... Bn = x(t ) sin(nω0t ) dt n = 1,2,3...
T ∫
−T / 2
T ∫
−T / 2

Discrete Time Fourier Transform pair (for discrete-time signal)

∞ π
− jnθ 1 jnθ
X (θ ) = ∑ x[n]e
n =−∞
... x[n] =
2π ∫ X (θ )e dθ
−π
θ = digital frequency; − π ≤ θ ≤ π

Impulse Invariant Method:

1 1
→ b>0 T-sampling period
s+b 1 − e − bT z − 1
Bilinear Transformation:
2 1 − z −1
s= T − sampling period
T 1 + z −1

2
Question 1. (38 marks – all students should attempt this question)

A. The signal x[n] = 2cos(0.2πn) + sin(0.7πn) is a sampled signal of x(t) at a sampling

frequency of 1000Hz.

(i) State the sampling theorem. [2 Marks]

(ii) If there is no aliasing during sampling then write down the expression for the
signal x(t).
[2 Marks]
(iii) If there is aliasing during sampling write down an expression for the signal x(t).

[3 Marks]
(iv) Determine the period of the signal x[n] [3 Marks]

B. Find the z-transform of the following sequence.

n
h[n] = (a ) where |a|<1 [5 marks]

C. The transfer function, H(z), of a discrete time system is given by,

H ( z ) = 1 − z −4

a. Sketch the pole-zero diagram of the system [4 Marks]

Hint: 4th order filter should have 4 zeros or poles.
b. Sketch an approximate magnitude response of the system [2 Marks]
c. Determine the frequency response of the system H(θ) and hence sketch the phase
response of the system. [4 Marks]

D. The pole-zero diagram of a discrete-time system is given below.

Sketch an approximate magnitude response of the system [3 marks]

3
E. The transfer function of a discrete time system is given by

1 + z −1
H ( z) =
1 − 0.2 z −1
a. Determine the magnitude response of the filter. [4 Marks]
b. Calculate the DC gain of the filter. [2 Marks]
c. Calculate the -3dB bandwidth of the filter. [4 Marks]

Question 2. (26 marks – all students should attempt this question)

A. The 2 point DFT of x[n] is X[k]. Assume X[0]=3 and X[1]=-1.

(i) Determine x[n]. [2 Marks]
(ii) Determine the sum of squares of the signal samples, x[n], and the sum of squares
of the magnitude of the DFT samples, X[k] and find how these summations are
related to each other. [2 Marks]
(iii) State the name of the Theorem related to the observation in part (ii)
[1 Mark]

B. Describe the overlap-add method of filtering a long sequence using DFT. You may need to use
diagram to explain it clearly. [4 marks]
th
C. Design a 7 order linear phase FIR filter using windowing method with rectangular window for
the magnitude response given below.
⎧ π
⎪1 θ ≤
H d (θ ) = ⎨ 4 [8 marks]
π
⎪0 < θ ≤π
⎩ 4
D. Obtain the transfer function and difference equation for a recursive FIR band pass filter with
simple coefficients for the following specifications.
Center frequency (pass band center frequency) = 3kHz
Sampling frequency = 12kHz [3 Marks]

E. Determine using bilinear transform method the transfer function for analogue filter given below.
The normalized transfer function is
1
H ( s) =
s +1
Assume a sampling frequency of 200Hz and a cut-off frequency of 40Hz. [6 Marks]

4
 Question 3. (22 marks – all students should attempt this question)

A. With diagram explain how adaptive filters can be used to cancel telephone echo. [4 Marks]
B. State the Widrow LMS algorithm (you do not need to derive that equation) [2 Marks]
C. A two-Channel Quadrature-Mirror Filter Bank is given below.

If the analysis filter, H0(z), is given by H0(z)=1+z-1 then determine H1(z), F0(z), F1(z) for
aliased free realization. [6 marks]
D. A system described by the difference equation, y[n]=ay[n-1]+x[n], is implemented with 1.4
fixed point signed representation (first bit is for sign and the other 4 bits are for fraction). If
15
a=0.5 show that limit cycles will occur when the input is given by, x[n] = δ [ n] .
16
[6 Marks]

E. A speech signal sampled at 8kHz is passed through a 4 level Wavelet decomposition.

a. After the decomposition how many bands of signals will you get? [1 Mark]
b. Specify the bandwidth of each of the decomposed bands. [3 Marks]

Question 4. (14 marks – attempt either Question 4 or Question 5)

A. State the biased and unbiased estimates of the autocorrelation function for a signal x[n].
[3 Marks]
B.
(i) Show that the power spectrum estimate using the biased autocorrelation function is
given by
biased 1 2
Pxx (θ ) = X (θ ) ,
M
biased
where Pxx (θ ) is the power spectrum estimate, X(θ) is the Fourier transform of x
[n] and M is the number of samples of signal x[n]. [5 Marks]
biased
(ii) Determine Pxx (θ ) if x[n] = {1 0 2 0 1} [3 Marks]
C. A zero-mean unit variance stationary white noise signal x[n] is applied to a FIR filter with
impulse response {1, -2.75, -0.75}. Determine the power spectrum of the output signal of the
filter. [3 Marks]

5
Question 5. (14 marks – attempt either Question 4 or Question 5)

A. A block diagram of a multi-rate processing system to convert a signal, x[n], originally

sampled at 6 kHz, to a new signal y[n] with a sampling frequency of 4 kHz is given below.

(i) Why is the up-sampling performed before down-sampling? [2 Marks]

(ii) If the magnitude spectrum, |X(θ)|, is as given below, sketch |W(θ)|, |P(θ)| and
|Y(θ)|. [4 Marks]

(iii) If the low pass filter is not used, then sketch |W(θ)| and |Y(θ)| if |X(θ)| is as given
below. [3 Marks]

(iv) Show that performing up-sampling followed by down-sampling is equivalent to

performing down-sampling followed by up-sampling if |X(θ)| is as given in part
(v). [2 Marks]

B. The sampling time of a digital signal, x[n], is T sec. Explain with a block diagram how you
would design a phase shifter that can delay the signal by 0.75T sec. [3 Marks]

End of examination paper