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EXAM SAMPLES

MODEL 1:

Question 1: [15 Marks]

I. Choose right answer(s) [10 Marks]
a) Any digital computer can be used for DSP.
1-True 2-False
b) What happens after the signal is passed through the ADC in a DSP?
1- Changed back to analog 2- Stored in a RAM
3- Amplified 4- Attenuated
c) Amongst the signals listed below, select those that are in digital format.
1-An eBook 2-JPEG images
3- Music recorded on a CD 4-Music recorded on a vinyl record.
d) What is the process of examining the frequency content of a signal?
1- Signal decoding 2-Spectrum analysis
3-Signal analysis 4-Data analysis
e) What is the program that is used to speed the spectrum analysis process?
1- DDFT 2-FDFT
3- FGPA 4-FFT
f) What is a filter?
1-Frequency selective circuit 2-Amplitude selective circuit
3-Frequency damping circuit 4-Amplitude damping circuit
g) What type of filter produces a predictable phase shift characteristic in all frequencies?
1-Band pass filters 2-High pass filters
3-Low pass filters 4-All pass filters
h) What is it called when the sampled signal is of lower frequency than the original signal?
1-Aliasing 2-Overshoot
3-Steady state error 4-Damping
i) How is aliasing prevented?
1-Sampling at the same frequency of the original signal
2-Sampling at double the frequency of the original signal
3-Sampling at less than double the frequency of the original signal
4-Sampling at more than double the frequency of the original signal
j) The discrete time function defined as u(n)=n for n=0;=0 for n<0 is an:
1-Unit sample signal 2-Unit step signal
3-Unit ramp signal 4-None of the mentioned
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II. A signal, given by x(t) = 2 cos(3πt), is sampled at a frequency of 20 Hz, starting at time t = 0.
(a) Is the signal sampled frequently enough? (Hint: 'aliasing')
(b) Find the first six samples of the sequence.
(c) Given that the sampled signal is represented by x[n], how could the above signal,
delayed by four sample periods, be represented? [5 Marks]
Ans:

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Question 2: [12 Marks]

I. Noise is considered as an undesired frequency in a signal. Several types of noises can affect
the quality of conversation. Discuss, mention some noise types and applications required
noise reduction. [3 Marks]

Ans:

Noise types:

 White noise: it takes mix of frequencies to make static sound. For example: AC condition
and Fan 2)

 Pink noise: frequency is stable and low wave and deep. For example: Constant raining

Noise Applications:

For Noise reduction, Wiener Filter

For echo cancellation, Least Mean Square (LMS)

Methods used for Noise Reduction:

Wiener filter code:

J = wiener2(I,[m n],noise)

[J,noise_out] = wiener2(I,[m n])

Wiener filter Advantages:

 Many well documented implementations (LMS, RLS, Kalm an filters)

 Are optimal in that they minimize the mean squared estimation error

 Often can be computed in real-time

Wiener filter Disadvantages:

 Provide point estimates

 Mean-squared error is not always relevant

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II. A music recording was noised with broadband background (white) noise. This noise spread
upon the entire spectrum of the music, and is therefore very unpleasant. Suppose a filter
design to reduce noise from the speech, with all discussion needed. [4 Marks]

Ans:
Wiener filter block diagram:

Wiener filter code:

J = wiener2(I,[m n],noise)

[J,noise_out] = wiener2(I,[m n])

Wiener filter Advantages:

 Many well documented implementations (LMS, RLS, Kalm an filters)

 Are optimal in that they minimize the mean squared estimation error

 Often can be computed in real-time

Wiener filter Disadvantages:

 Provide point estimates

 Mean-squared error is not always relevant

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III. De-noising is one of the major applications of the wavelet analysis for one dimensional
signals. Discuss briefly. [2 Marks]
Ans:
• Extracts information from images in different scale and resolutions
• Compress images without losing important information for analyzing data
• Provides good & fast performance in object recognition task
• Used for medical application & Satellite images for EDGE Detection task
• One-dimensional Signal Enhancement, Acoustic, speech, music and electrical transient
signals

IV. De-noise the signal fNoisy using the sym4 and db1 wavelets, with a nine-level wavelet
decomposition. Plot the results & compute SNR. [3 Marks]
Ans:
Signal de-noising needs more levels of decomposition. Here the level of decomposition is
four. In wavelet analysis, the approximate sub band contains the low frequency
components called signal component and detail sub band contains the high frequency
components made of noise.

Cleansym = wdenoise(fNoisy,9, ‘Wavelet’,’sym4’);

Cleandb =wdenoise(fNoisy,9,’Wavelet’,’db1’);
Figure
Plot(cleansym)
title(‘Denoised – sym’)
figure
plot(cleandb)
title (‘Denoised-db’)
snrsym = -20*log10(norm(abs(fClean-Cleansym))/norm(fClean))
snrdb = -20*log10(norm(abs(fClean-cleandb))/norm(fClean))

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Question 3: [18 Marks]

I. The output sequence, y[n], from a recursive filter, is related to its input sequence, x[n]

y[n] = x[n] + 0.5x[n-1] + 0.5y[n-1]. [4 Marks]

Draw the block diagram for this filter, in the z-domain, and find its transfer function. Use the transfer
function to find the filter's response to a unit sample sequence (first 4 terms only).
Ans:

II. Find the transfer function for the processor shown in figure below [3 Marks]

Ans:
Y(z) = X(z) -0.4x[n-1] + y[n-2] + 0.5

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I. How to differentiate IIR and FIR filters? [2 Marks]

Ans:

Finite Impulse Response (FIR) “No Feedback” Infinite Impulse Response (IIR) “Require Feedback”
Implemented non recursively Implemented recursively

Always stable Stability non-guaranteed

Simple to Implement Difficult to Implement

Linear Phase response in pass-band Non-linear phase response in pass-band
More memory & more processing power Less memory & less processing power
Doesn’t have an analog counterpart Analog filters can be converted into IIR Filters

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Long Division Example:

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h
MODEL 2:

Question 1:
I. Select the right answer
1. If x(n) is a discrete-time signal, then the value of x(n) at non integer value of ‘n’ is:
a. Not defined b. Zero
c. Positive d. Negative
2. The discrete time function defined as u(n)=n for n=0; =0 for n
a. Unit sample signal b. Unit step signal

c. Unit ramp signal d. None of the mentioned

3. Which of the following should be done to convert a continuous-time signal to a discrete signal?

a. Sampling b. Differentiating

c. Integrating d. None of the mentioned

4. The process of converting discrete-time continuous valued signal into discrete-time

discretevalued(digital) signal is known as:

a. Sampling b. Quantization

c. Coding d. None of the mentioned

5. What is the z-transform of the following finite duration signal?

a.
b.
c.
d.

6. Incorrect design of IIR filter, may affect filter _ .

a. Analysis b. Efficiency c. Stability d. Restoration
7. A filter has a frequency response, which is similar to that of a high pass filter.
a. IIR b. Running average c. Power-Log d. Negative
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Question 2: [20 Marks]

I. Given The following: [5 Marks]

Plot the following discrete-time signal. Label horizontal & vertical.

Ans:

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III. Produce a signal-flow-graph for each of the following difference-equation [5 marks]

1- y[n] - ½ y[n-1] = 3x[n] - 2x[n-1]

Ans:

2- y[n] - ⅛ y[n-2] = 4x[n-2]

Ans:

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The FFT is a highly elegant and efficient, algorithm, which is still one of the most used algorithms in DSP.
Discuss, & Give MatLab example.
Ans:
Fast Fourier Transform (FFT) is a highly improved implementation of Discrete Fourier Transform (DFT),
which converts discrete signals from the time domain to the frequency domain.
FFT calculations provide information about the frequency content, phase, and other characteristics of the
signal.

- r = imread(‘lenna.gif’);
- figure,imshow(r)
- F = fft2(r);
- S = fftshift(log(1+abs(F)));
- figure,imshow(S, [])
- img = imread('cameraman.tif');
- f = fft2(double(img));
- figure ;
- imshow(img);
- title ('Original Image')
- figure ;
- imshow(abs(fftshift(f)), [24 10000])
- title ('Magnitude Image') –
figure ;
- imshow(angle(fftshift(f)), [-pi pi])
- title ('Phase Image')

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Discuss in short sentences

Signal Synthesis is a type of signal processing to put on information out to the world.
Ans:
Signal Synthesis is used for modifying and analyzing information with computers to represent,
transform, and manipulate signals with the information they contain.

Discuss how are wavelets useful for signal processing?

Ans:
• Extracts information from images in different scale and resolutions
• Compress images without losing important information for analyzing data
• Provides good & fast performance in object recognition task
• Used for medical application & Satellite images for EDGE Detection task
• One-dimensional Signal Enhancement, Acoustic, speech, music and electrical
transient signals

Aliasing Occur when under-sampling signal frequency. Discuss

Ans:
Aliasing occurs when analogue input voltage is not sampled frequently enough where high
frequency input signals will appear as low frequency signals at the output.

1. Which of the following are the disadvantages of FIR filter?

a) High storage space
b) More delay
c) Non Linear
d) All the above
2. Identify the type of response of FIR filter below.
a) Symmetric Odd
b) Symmetric Even
c) Asymmetric Odd
d) Asymmetric Even

3. Which of the following are the advantages of FIR filter?

a) Linear phase
b) Easy in implementation
c) Represents values in fractional arithmetic
d) All of the above

4. Identify the type of response of FIR filter below.

a) Symmetric Odd
b) Symmetric Even
c) Asymmetric Odd
d) Asymmetric Even

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5. Design of filter depends on which of the following factors of FIR filter?

a) Length
b) Coefficient
c) Size
d) Both a and b

6. The output obtained from FIR filter is _____ form?

a) Linear
b) Nonlinear
c) Abrupt
d) Both b and c

7. FIR filter is also called ____?

a) Recursive filter
b) Non-recursive
c) Higher resistance
d) Lower resistance

8. Which of the following is the impulse response of FIR filter?

a) Infinite
b) Finite
c) Zero
d) Negative

9. Which system requires infinite memory?

a) FIR filters
b) IIR filters
c) Both a and b
d) None of the above

10. The _________ filters use feedback

a) IIR
b) FIR
c) Both a and
d) None of the above

11. The _________ filters are always bounded and hence they are inherently stable.
On the other hand, the _________ filter may become unstable if not designed properly
a) FIR , IIR
b) IIR , FIR
c) Stopband , passband
d) None of the above

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12. The similarity between the Fourier transform and the z transform is that
a) Both convert frequency spectrum domain to discrete time domain
b) Both convert discrete time domain to frequency spectrum domain
c) Both convert analog signal to digital signal
d) Both convert digital signal to analog signal

13. Discrete cosine transforms (DCTs) express a function or a signal in terms of

a) Sum of cosine functions oscillating at different frequencies
b) Sum of cosine functions oscillating at same frequencies
c) Sum of cosine functions at different sampling intervals
d) Sum of cosine functions oscillating at same sampling intervals

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MatLab Code Using array to plot LTI system operation

MATLAB using Array:

A= [ 1 2 3 4 5 ]
X= [ 0 0 0 0 0].
while 1
input (“x=”);
x(1) = x;
y= A(1)*5
for k = 2:5
y = y+ A(s) + x (k)
end.
disp(('y = ' num2str (y)]);
for k=5 : - 1:2
X (K) = x ( k-1);
end;
end;

MatLab Code to plot a signal:

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Signal operations using MatLab:

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