QP For Digital Signal Processing
QP For Digital Signal Processing
Reg. No. :
Fifth/Eighth Semester
(Regulations 2017)
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.
PART B — (5 × 13 = 65 marks)
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)
12. (a) Find the linear convolution of the two signals represented by (13)
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)
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
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
————––––——
3 40495
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.
8. Give the basic structure of Direct form II structure for realizing an IIR filter.
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)
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)
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)
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
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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