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# CS3291 Exam Jan 2006 solutions

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University of Manchester
Department of Computer Science
First Semester Year 3 Examination Paper
CS3291: Digital Signal Processing
Date of Examination: January 2006
Answer THREE questions out of the five given.
Time allowed TWO HOURS
(Each question is marked out of 20). Electronic calculators may be used.
_____________________________________________________________________________
1
(a) Briefly outline four of the main advantages and one disadvantage of digital signal processing
(DSP) as opposed to analogue signal processing.
[5 marks]
(b)

Define each of the following terms as applied to discrete time signal processing systems:
(i) linearity
(ii) time-invariance
(iii) causality
(iv) stability
[4 marks]

## (c) Produce a signal-flow-graph for each of the following difference-equations

(i)
y[n] = x[n] + 0.5 x[n-1] 0.5 x[n-2]
(ii)
y[n] = x[n] + x[n-1] + 0.5 y[n-1]
Determine the impulse-response of each of these difference equations.
[6 marks]
(d)
Calculate, by tabulation or otherwise, the output from difference-equation (ii) when the
input is the impulse-response of difference-equation (i).
[5 marks]
2.

(a) Given the impulse-response {h[n]} of a discrete time LTI system, show that the response to
any other input signal {x[n]} is {y[n]} where:
y[n]

h[m] x[n m]

## for < n <

Hence express the system function H(z) in terms of {h[n]} for values of z with |z| 1.
How is the frequency-response derived from H(z)?
[8 marks]
(b) Explain how the poles and zeros of H(z) affect the stability and the gain-response of the
system. Give H(z) for a DSP system with the following difference equation:
y[n] = x[n] + 1.21x[n-2] - 0.8 y[n-1]
Plot its poles and zeros on the z-plane, determine whether it is causal and stable and sketch its
gain-response.
[9 marks]
(c) If the input signal to a digital filter with frequency-response
H(ej) = (1 + 2cos(2) )e 2j
is {x[n]} with x[n] = 2 cos( 0.5 n) for all n, what is the output signal?

[3 marks]

## CS3291 Exam Jan 2006 solutions

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3.
(a) With the aid of phase-response graphs, explain what is meant by the terms 'linear
phase' and phase delay.
Explain why having linear phase is a desirable property for analogue and digital filters.
Is it true that all linear time-invariant DSP systems have linear phase?
[5 marks]
(b) Use the windowing method with a rectangular window to design a fourth order "low-pass"
FIR digital filter whose cut-off frequency is 2.5 kHz and whose phase-response is linear phase
in the pass-band. The sampling frequency is 30 kHz.
Give the digital filters system function.
Give a signal-flow-graph for the digital filter.
State whether the 4th order FIR digital filter is exactly linear phase or only approximately so.
[8 marks]
(c) Explain how the gain-response of this digital filter could be improved by:
(i)
increasing the order and
(ii)
imposing a non-rectangular window?
How would these improvements affect the phase-response?

[4 marks]

(d) Why is the Remez exchange algorithm generally considered superior to the windowing
method as a design technique for FIR digital filters?
[3 marks]
4
(a) Briefly state the advantages and disadvantages of infinite impulse-response (IIR) digital
filters as compared with finite impulse-response (FIR) types.
[5 marks]
(b) A second order IIR notch digital filter is required to eliminate an unwanted sinusoidal
component of a digitised signal, sampled at 3 kHz, without affecting the magnitudes of other
frequency components too severely. The frequency of the unwanted sinusoid is 250 Hz and the
3 dB band-width of the notch should be approximately 38.2 Hz.
Design the notch filter by pole and zero placement.
Give its transfer function, H(z).
Sketch the gain response of the notch filter.
[9 marks]
(c) Give a direct form II signal flow graph for the notch filter and a program or flow diagram
to indicate how the filter would be implemented on a microprocessor with 16-bit integer
arithmetic only available.
[6 marks]

## CS3291 Exam Jan 2006 solutions

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5
(a) Define the Discrete Fourier Transform (DFT) and explain how it is related to the Discrete
Time Fourier Transform (DTFT).
[4 marks]
(b) Explain why analogue signals are generally low-pass filtered before they are converted to
digital form.
With the aid of simple diagrams, explain how aliasing distortion could arise if such filtering
were not applied.
Explain why increasing the sampling rate simplifies the analogue filters required. [7 marks]
(c) In the absence of an anti-aliasing input filter, what would be the result of sampling an 8 kHz
sine-wave at (i) 10 kHz, (ii) 6 kHz and (iii) 4 kHz
[3 marks]
(d)
Explain the term quantisation noise A DSP system, with a 16-bit uniformly quantising
analogue-to-digital converter and a sampling rate of 20 kHz, is used to process analogue signals
band-limited to the frequency range 0 Hz to 5 kHz. Estimate the maximum achievable signal-toquantisation noise ratio (SQNR) for sinusoidal input signals, and state what assumptions it is
reasonable to make about the statistical and spectral properties of the quantisation noise.
[6 marks]
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## CS3291 Exam Jan 2006 solutions

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Solutions
1. (a)
Advantages of digital as opposed to analogue signal processing include the following
( choose 4 ) :

More and more signals are being transmitted and /or stored in digital form so it makes sense
to process them in digital form also.
DSP systems can be designed and tested in simulation using universally available
computing equipment ( e.g. PCs with sound and vision cards ).
Guaranteed accuracy, as pre-determined by word-length and sampling rate.
Perfect reproducibility. Every copy of a DSP system will perform identically.
The characteristics of the system will not drift with temperature or ageing.
Advantage can be taken of the availability of advanced semiconductor VLSI technology.
DSP systems are flexible in that they can be reprogrammed to modify their operation without
changing the hardware. Products can be distributed / sold and updated via Internet.
Digital VLSI technology is now so powerful that DSP systems can now perform functions
that would be extremely difficult or impossible in analogue form. Two examples of such
functions are :(i) adaptive filtering ( where the parameters of a digital filter are variable and
must be adapted to the characteristics of the input signal) and, (ii) speech recognition which
is again based on information obtained from speech by digital filtering.

## Disadvantages of digital signal processing ( choose one ) :

DSP designs can be expensive especially for high bandwidth signals where fast
analogue/digital conversion is required.
The design of DSP systems can be extremely time-consuming and a highly complex and
specialized activity. There is an acute shortage of electrical engineering graduates with the
knowledge and skill required.
The power requirements for DSP devices can be high, thus making them unsuitable for
battery powered portable devices such as mobile telephones. Fixed point processing devices
( offering integer arithmetic only ) are available which are simpler than floating point
devices and less power consuming. However the ability to program such devices is a
particularly valued and difficult skill.
1(b)
Linearity:Given any two discrete time signals {x 1 [n]} and {x 2 [n]}, if the system's response to {x 1 [n]} is
{y 1 [n]} and its response to {x 2 [n]} is {y 2 [n]} then for any values of the constants k 1 and k 2 ,
its response to k 1{x 1[n]} + k 2{x 2[n]} must be k 1{y1[n]} + k 2 {y 2 [n]} .
To multiply a sequence by k, multiply each element by k.
Time invariance:Given any discrete time signal {x[n]}, if the system's response to {x[n]} is {y[n]}, its
response to {x[n-N]} must be {y[n-N]} for any integer N.
(Delaying the input signal by N samples must delay the output signal by N samples.)
Causality: If the impulse response is {h[n]} then h[n] = 0 for n< 0 if the system is causal.
Stability:

| h[n] |

0 for stability

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(c) SFGs
x[n]

z-1

z-1

-0.5

0.5

y[n]
+

+
{y[n]}

{x[n]}
z

z-1

-1

0.5

## (i) {h[n]} = { , 0, 0, 1, 0.5, -0.5, 0, , 0, }

(ii) {h[n]} = {, 0, , 0, 1, 1.5, 0.75, 0.375, 0.1875, .. }
(d) By tabulation:
n
-1
0
1
2
3
4
5

x[n]
0
1
0.5
-0.5
0
0
etc.

x[n-1]
0
0
1
0.5
-0.5
0

y[n-1]
0
0
1
2
1
0

y[n]
0
1
2
1
0
0

{h[n]} = { , 0, 1, 2, 1, 0, .., 0 }
By otherwise: For (i) system function is H1(z) = 1 + 0.5 z-1 - 0.5 z-2 = (1 - 0.5z-1)(1+z-1)
For (ii) H2(z) = (1+z-1)/(1-0.5z-1)
Tranfser fn of (i) & (ii) is H1(z)H2(z) = (1 - 0.5z-1)(1+z-1)(1+z-1)/(1-0.5z-1)
= (1+z-1)(1+z-1) = 1 + 2z-1 + z-2
Impulse response of H1(z) H2(z) = {, 0, , 0, 1, 2, 1, 0, , 0, )

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## 2(a) Response to impulse {d[n]} is {h[n]}

Response to {d[n-k]} is {{h[n-k]} by time-invariance
Response to x[k] {d[n-k]} is x[k]{d[n-k]} by linearity, taking x[k] as a constant
Response to

x[k ]{d [n k ]}

x[k ]{h[n k ]}

is

Now

1: n k
0:n k

## x[k ]{h[n k ]} for all n

Now let m = n-k. When k = then m=-. When k = - then m=.It follows that:
y[n]

## (It doesnt matter what order you add things up in.)

If we set x[n] = zn for all n, where z is a complex number with |z| 1 then
y[n]

h[m]x[n m]

h[ m] z n m z n

h[m]z

z n H ( z ) where H ( z )

h[m]z

## H(z) is the system function.

Replacing z by ej gives the frequency-response as a function of relative frequency in
2(b)
Zeros do not affect stability.
Poles must lie inside unit circle for stability.
Gain response determined by the "distance rule":Gain at frequency is:
Product of distances from each zero to the point z = e j on unit circle
G() =
Product of distances from each pole to the point z = e j on unit circle
y[n]=x[n]+1.21x[n-2]-0.8y[n-1]

H ( z)

1 1.21z 2
z 2 1.21 ( z 1.1 j )( z 1.1 j )

z ( z 0.8)
z ( z 0.8)
1 0.8 z 1

## Filter is causal as impulse response will be zero for n<0.

Filter is stable as poles are inside unit circle.
Imag(z)

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Zero at z=1.1j

0.8

Pole

Real(z)

Pole

Zero at z=-1.1j

0
/4
/2
3/4

1.5 * 1.5
0.6 * 1.5
0.1 * 2.1
0.6 * 1.5
1.5 * 1.5

## Prod pole distances

1 * 1.8
1 * 1.5
1 * 0.8
1 * 0.5
1 * 0.2

Gain estimate
1.25
0.6
0.26
1.8
11.25

3 dB points : Assuming negligible changes to pole distances and distance to zero at z = -1.1j,
the gain may be estimated to increase by 3 dB at = /2 0.1 from its value at = /2. This is
because the zero is at a distance 0.1 from the unit circle.
Similarly gain may be estimated to fall by 3dB at = -0.2 radians per sample because the pole
is also at distance 0.2 from the unit circle.
Hence sketch gain response:
G()
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## CS3291 Exam Jan 2006 solutions

2(c)
Response is {2 G() cos(0.5n + ()} with = 0.5
G() = (1+2cos(2)) and ()=-2
Response is 2(1+2cos(1))cos(0.5 n - 1) } = 2(2.08)cos(0.5 n - 1) }
i.e. {4.16 cos(0.5n - 1) }

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## CS3291 Exam Jan 2006 solutions

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3(a)
Expressing the frequency-response H(ej) = G()exp(j()), a digital filter with phase response
() is linear phase if the phase-delay () / is constant for all . A linear phase response
graph is as follows:
()

## Similarly for an analogue filter with replacing ..

Linear phase is a desirable property because then the system delays all frequency components of
a periodic signal (say), expressed as a Fourier series say, by exactly the same amount of time. If
there are no amplitude changes, say because the signal falls within the pass-band of a low-pass
digital filter, the output waveform will not be distorted in shape by phase effects (different
frequency components being delayed by different amounts of time and therefore adding up
differently). All frequency components of a signal will be delayed by the same amount of time,
i.e. the phase delay. Hence phase distortion will not occur.
It is not true to say that all LTI systems have linear phase.
3

## the relative cut-off frequency C = (2 / fs ).2500 = /6 radians per sample.

Take phase to be zero initially.
Therefore H(ej) = G()
By the inverse DTFT formula:
h[n]

1
2

G ()e jn d

1
2

/6

e jn d

/6

=
1 1 jn
e

2 jn

/6

/ 6

1
e j / 6 e j / 6
2jn

1
2

/6

- /6

1 d

1
2 j sin( / 6)
2jn

1/6 when n 0

## The impulse response is {h[n]} with

h[n] =
and

(1/(n)) sin(n/6)

h[n] = 0.1667

when

______________________________
n
0
1
2

h[n]
0.1667
0.160
0.138

when

n=0.

n0

when n 0

## CS3291 Exam Jan 2006 solutions

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3
0.106 etc
______________________________
On rectangularly windowing, we obtain the causal finite impulse response:
{h[n]} = { 0, 0.138, 0.16, 0.1667, 0.16 0.138, 0, , 0, }
After delaying by 2 samples to make the impulse response causal,
{h[n]} = { 0, 0, 0.138, 0.16, 0.1667, 0.16, .138, 0, , 0, }
H(z) = 0.138 + 0.16z-1 + 0.1667 z-2 + 0.16 z-3 + 0.138z-4
Signal-flow graph
x[n]

z-1

0.138

z-1

z-1

z-1

0.16

0.16

0.138

y[n
]

Filter is now linear phase with phase delay of 2 sampling intervals (in the pass-band) .
It will have a well defined stop-band decreasing in gain from 0dB at 0 Hz to -6 dB at the cut-off
frequency. The stop-band gain will have ripples (illustration useful).
3(c)Increasing the order of the filter would mean that the phase delay would have to increase
also if the filter remains linear phase. The magnitude response would become closer to the ideal
low-pass response with more stop-band ripples. If the rectangular window is still used, the
highest stop-band ripple would not reduce significantly due to Gibb's Phenomenon.
The use of a Hann or similar raised cosine window would reduce the stop-band ripples at the
expense of a less sharp cut-off rate from pass-band to stop-band.
The phase response is not affected by the imposition of a non-rectangular window.
3 (d) The Remez exchange algorithm gives an 'equi-ripple approximation' to the ideal gain
response required; i.e. equal ripple peaks across pass-band and stop-band.
[1]
With the windowing technique, the peaks of the stop-band ripples are not equal in amplitude
and reduce with increasing frequency. The stop-band approximation gets better with increasing
frequency .
[1]
By making all ripple peaks equal, Remez minimises the difference between the ideal gain
response and the approximation across the whole of the frequency range. It is a 'mini-max'
approximation. Hence the highest stop-band ripple peak will be lower than for the windowing
technique.
[1]

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4. (a)

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## Comparison of IIR and FIR digital filters:

IIR type digital filters have the advantage of being economical in their use of delays, multipliers
[1]
They have the disadvantage of being sensitive to coefficient round-off inaccuracies and the
effects of overflow in fixed point arithmetic. These effects can lead to instability or serious
distortion.
[1]
Also, an IIR filter cannot be exactly linear phase.
[1]
FIR type digital filters may be realised by non-recursive structures which are simpler and
more convenient for programming especially on devices specifically designed for digital
signal processing.
These structures are always stable, and because there is no recursion, round-off and overflow
errors are easily controlled.
A FIR filter can be exactly linear phase.
[1]
The main disadvantage of FIR filters is that large orders can be required to perform fairly simple
[1]
4. (b)

## fs = 3000 Hz. Notch is at 250 Hz

Rel frequency of notch = (2/3000).250 = / 6 radians/sample.

## 3dB bandwidth = 38.2 Hz

= 38.2 / 3000 x 2 = 0.08 radians/sample
Therefore 3 dB points are at / 6 0.04 radians/sample.
Poles must be placed at 0.04 from the unit circle
Distance from zeros on unit circle = 0.04
Place zeros z1 and z2 at exp( j /4).
Place poles p1 and p2 at 0.96 exp( j /4)
(z - e j / 4) (z - e - j / 4)

H(z) =

(z - 0.96 e j / 4) (z - 0.96 e - j / 4)
z2 - 2 cos (/4) z

z2
H(z) =

1.414 z

z2

1.92 z +

0.922

1.414 z - 1

+ z-2

## CS3291 Exam Jan 2006 solutions

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H(z) =

12

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1.92 z - 1 +

0.922 z - 2

4(b) continued
Difference equation is:
y[n] = x[n] - 1.414 x[n-1] + x[n-2] + 1.92 y[n-1] - 0.922 y[n-2]
Sketch gain response.
G()

0.7

dB

-3 dB

/6

/6-0.04

/6+0.04

W
x[n]

y[n]
z-1

1.92

W1
z-1

-0.922
-

W2

-1.414

## CS3291 Exam Jan 2006 solutions

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4(c) continued
% Direct Form II in fixed point arithmetic & shifting.
K=1024;
A0=K; A1=round(-1.414*K); A2=K;
B1=round(-1.92*K); B2=round(0.922*K);
W1 = 0; W2 = 0;
%For delay boxes
while 1
Input X ;
%Input a sample
W =K*X - B1*W1 - B2*W2;
% Recursive part
W =round( W / K);
% By arith shift
Y = W*A0+W1*A1+W2*A2; % Non-rec. part
W2 = W1;
W1 = W;
%For next time
Y = round(Y/K);
%By arith shift
Output Y;
end;
%Back for next sample

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## 5.(a) Considering first the DTFT formula:

X(ej) =

x n e

- jn

where = / f s T radians/sample

This transforms a (possibly complex) discrete time signal {x[n]} of infinite duration to the
relative frequency () domain.
Defining: X(e j k ) X k , the DFT transforms a finite (possibly complex valued)
sequence {x[n]}0,N-1 to the finite complex valued sequence {X[k]}0,N-1.
The DFT formula is:N 1

## X k x n e j k n where k 2k / N for k = 0, 1, 2, ....., N - 1

n 0

For each k = 0,1, 2, , N-1, X[k] is a sample of the spectrum X(e j) at =2k/N. In this case,
X(ej) is the spectrum (DTFT) of an infinite discrete time signal {x[n]} comprising {x[n]} 0,N-1
padded out to infinity (in both directions) with zeros.
Therefore is in the range 0 to 2 is and X(ej) is uniformly sampled over this range.

X(e jT ) =

1
T

X a ( j ( n 0 ))

with 0 2 / T

## If xa( t ) is band-limited between -/T and +/T radians/sec ( fs/2 Hz ), then

Xa( j ) =0 for
/T.
It follows that :
X( ejT ) = ( 1/T ) Xa( j ) for -/T < < /T
This is because Xa( j( - 2/T ) ), Xa( j( + 2/T ) ) and Xa( j ) do not overlap.
Where Xa(j) is not band-limited to the frequency range -/T to /T, overlap occurs.
If now we take Xs( ejT ) to represent Xa( j )/T for -/T < < /T, it will be distorted.
This is aliasing distortion.
To avoid aliasing distortion, low-pass filter xa( t ) to band-limit the signal to fS/2 Hz
before sampling at fs Hz. It then satisfies Nyquist sampling criterion .
5 (c)
(i) We obtain an aliased sine wave of frequency 5 -3 kHz = 2 kHz
(ii) we obtain aliased sine wave of frequency 2 kHz
(iii) A constant (dc) signal seen. No sine wave at all.

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## 5 (d) Quantisation noise power : 2/12 where is quantisation step.

Sinusoidal signal power = A2 / 2 where A is the maximum possible signal amplitude.
16- bit ADC, therefore 216 quantisation levels.
A = 2 15
Signal-to-quantisation noise ratio (SQNR) = (A2/2) / (2 / 2)
= 2 29 2 / (2 /12)
= 2 31 x 3 = 25.166 x 10 6
In dB SQNR = 10 log10(2 31 x 3) = 97.7 dB ( = 6 x 16 + 1.7)
The quantisation noise spectrum may be assumed white in the frequency range 0 to fs / 2 Hz.
In the time-domain, the quantisation error samples may be assumed random and statistically
uniformly distributed between -/2 and /2.

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