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QP For Digital Signal Processing

The document discusses topics related to digital signal processing including properties of signals and systems, Z-transforms, DFT, FFT, filter design techniques. It contains questions ranging from basic concepts to the applications of DSP concepts. The questions are from different areas of DSP like linear time-invariant systems, Z-transforms, DFT, filter design etc.

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deepika ravi
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0% found this document useful (0 votes)
31 views

QP For Digital Signal Processing

The document discusses topics related to digital signal processing including properties of signals and systems, Z-transforms, DFT, FFT, filter design techniques. It contains questions ranging from basic concepts to the applications of DSP concepts. The questions are from different areas of DSP like linear time-invariant systems, Z-transforms, DFT, filter design etc.

Uploaded by

deepika ravi
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
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Reg. No. :

Question Paper Code : 40495

B.E./B.Tech. DEGREE EXAMINATIONS, NOVEMBER/DECEMBER 2021.

Fifth/Eighth Semester

Electrical and Electronics Engineering

EE 8591 –– DIGITAL SIGNAL PROCESSING

(Common to Electronics and Instrumentation Engineering/Instrumentation and


Control Engineering)

(Regulations 2017)

Time : Three hours Maximum : 100 marks

Answer ALL questions.

PART A — (10 × 2 = 20 marks)

1. What is meant by Nyquist rate? Comment on aliasing effect.

2. When is a system said to be causal and stable?

3. State convolution theorem wrt Z-transform.

4. List any two properties of DTFT, with relevant expression.

5. Compare DFT and FFT based on computational complexity.

6. Draw the basic DIT butterfuly diagram marking input and output with twiddle
factor.

7. Write the windowing function due to Blackmann for FIR filter design.

8. What is meant by prewarping? Give remedial measure to prevent this.

9. Differentiate Von Numann and Harvard architecture

10. Give the different addressing formats of digital signal processors.

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PART B — (5 × 13 = 65 marks)

11. (a) For the system described by y(n ) = x (n )x (n − 2) , check whether it is


static/dynamic, causal/non-casual, linear/non-linear and time
invariant/time variant. (13)

Or
(b) (i) Represent the sequence x (n ) = (3,1,−2,1,4,2,5,1) for
n = −3,−2,−1,0,1,2,3,4 as weighted sum of unit impulses. (6)

(ii) Check whether the signal x (n ) = u(n ) − u(n − 6 ) is an energy signal or


power signal. (7)

12. (a) Find the linear convolution of the two signals represented by (13)

2 for n = −2, 0,1



x (n ) = − 1, for n = −1
0 elsewhere

and h(n ) = δ (n ) − 2δ (n − 1) + 3δ (n − 2) − δ (n − 3) .

Or
(b) Find all possible inverse Z-transforms of
[( 2
)] (
X (z ) = z z − 4 z + 5 / z − 6z + 11z − 6 .
3 2
) (13)

13. (a) (i) Find the frequency response, magnitude response and phase
response of y(n ) = x (n ) + 0.81x (n − 1) + 0.81x (n − 2) − 0.45x (n − 3) . (6)

(ii) Find the 4 point DFT using matrix method if x (n ) = {1,−2,3,2}. (7)

Or
(b) Compute X (k ) of x (n ) = {1,−1,−1,−1,1,1,1,−1} using radix-2 DIT FFT. Also
plot amplitude and frequency spectra. (13)

14. (a) Convert the analog filter with transfer function


H a (S ) = s + 0.1 / (s + 0.1) +9 into digital filter by impulse invariant
2

transformation. (13)

Or
(b) Design a low pass Butterworth digital filter to give response of 3dB or
less for frequencies upto 2kHz and an attenuation of 20 dB or more
beyond 4kHz. Use the bilinear transformation technique and obtain H(z)
of the desired filter. (13)

2 40495

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15. (a) Explain in detail with a neat diagram the architecture of any one of the
latest digital signal processors. (13)

Or
(b) (i) Write a brief technical note on commercial digital signal processors.
(6)
(ii) Describe in detail any four addressing formats of digital signal
processor. (7)

PART C — (1 × 15 = 15 marks)

16. (a) Design an FIR low pass filter satisfying the following specifications :

( α p ≤ 0.1dB, as ≥ 38 dB )

w p = 15 rad / sec, ws = 25 rad / sec, wsf = 80 rad / sec . Use Kaiser window.

Or
(b) The desired frequency response of a low pass filter is
1; ≤w ≤ π2
( )
−π
2
H d e jw = 
0;
π
2 ≤w≤π

Determine hd (n ) for M=7 using rectangular window.

————––––——

3 40495

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*X10402* Reg. No. :

Question Paper Code : X10402


B.E./B.Tech. Degree Examinations, november/december 2020
Fifth Semester
Electrical and Electronics Engineering
Ee 8591 – digital signal processing
(Common to Electronics and Instrumentation Engineering)
(Regulations 2017)

Time : Three Hours Maximum : 100 Marks

Answer all questions

Part – A (10×2=20 Marks)

1. Show that δ(n) = u(n) – u(n – 1).

 n
1 + 3 , − 3 ≤ n ≤ −1

2. Given that x(n ) =  1 , 0 ≤ n ≤ 3 , sketch the signal x( – n + 4).
 0 , elsewhere


3. State and prove the time shifting property of Z-transform.

4. State and prove parseval’s theorem in the context of DTFT.

5. Discuss the circular convolution property.

6. Draw the basic butterfly structure of a radix-4 DIT algorithm.

7. Differentiate IIR and FIR filters.

8. Give the basic structure of Direct form II structure for realizing an IIR filter.

9. Give the data formats of any one DSP processor.

10. How is pipelining effected in a DSP processor ?

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X10402 -2- *X10402*

Part – B (5×13=65 Marks)

11. a) Determine whether the following systems are static, linear, time invariant,
causal and stable with proper justifications. (4+4+5)
i) y(n) = x(n) + nx(n + 1)
ii) y(n) = x( – n)
iii) y(n) = sign (x(n))
(OR)
b) i) Determine the zero-input response of the difference equation given by the
following :
x(n) – 3y(n – 1) – 4y (n – 2) = 0 (6)
ii) Determine the impulse response of the following causal system. (7)
y(n) – 3y(n – 1) – 4y(n – 2) = x(n) + 2x (n – 1)

12. a) i) Determine the Z-transform and sketch the ROC of the following signal by
applying the appropriate property of the Z-transform wherever necessary. (7)

x(n) = n2u(n)
ii) Determine the inverse z-transform of (6)
1 + 2z −1
x(z ) =
1 − 2z −1 + z −2
If x(n) is causal, x(n) is anti-causal.
(OR)
b) i) Determine the magnitude and phase spectra for the following signal by
computing its Fourier transform. (7)
x(n) = u(n) – u(n – 6)
ii) Consider the following signal, determine its power density spectrum and
evaluate the power of the signal. (6)
nπ nπ 1 3nπ
x(n ) = 2 + 2 cos + cos + cos
4 2 2 4

13. a) i) Discuss the savings in time for a radix-2 DIT algorithm to compute
FFT. (4)
ii) Determine the eight point FFT using DIT algorithm. (9)
x(n) = {1, 1, 1, 1, 1, 1, 0, 0}
(OR)
b) Derive the butterfly structure for a radix-2 DIF algorithm that is used to
compute FFT. Explain with an example. (13)

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*X10402* -3- X10402

14. a) A digital low-pass filter is required to meet the following specifications : (13)
Pass band ripple : ≤ 1 dB
Pass band edge : 4 kHz
Stop band attenuation : ≥ 40 dB
Stop band edge : 6 kHz
Sample rate : 24 kHz
The filter is to be designed using bilinear transformation on an analog system
function. Use Butterworth approximation.

(OR)

b) Determine the coefficients h(n) of a linear-phase FIR filter of length M = 15,


which has a symmetric unit sample response and a frequency response that
satisfies the following condition : (13)

 1, k = 0, 1, 2, 3
2πk 
H( k ) = 0.4, k=4
15 
 0, k = 5, 6, 7

15. a) Discuss the architecture of any one DSP processor and explain its features. (13)

(OR)

b) Discuss the addressing modes supported by a DSP processor and explain how
each is used for various DSP operations. (13)

Part – C (1×15=15 Marks)

16. a) i) Obtain the cascade and parallel structures for the following system and
realize it using Direct form II. (8)
1
y(n ) = y(n − 1) − y(n − 2) + x(n ) − x(n − 1) + x(n − 2)
2

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X10402 -4- *X10402*

ii) Determine the coefficients a1, a2, c1, c0 in terms of b1 and b2 so that the two
systems in the given figure below are equivalent. (7)

(OR)
b) Discuss about implementation of FFT with any suitable digital signal
processor. Also write a ‘C’ program to implement the FFT with the same
processor.


–––––––––––––

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