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Digital Signal Processing by Dr Muhammad Ashraf, S-V, Test-3, Dated: 8-12-2003, Time allowed: 55min

It is a closed book test. The required formulae are provided with the question sheets. Do all necessary
steps to support your answers. Correct answer without necessary steps will carry only 10% marks.
Attempt any three questions. All questions carry equal marks
Q# 1) The Figure titled “Sampling and signal reconstruction” is shown blow, x-axis is Time(msec) (40)
a.) Calculate frequency of the input signal ’x(t), sampling frequency ‘Fs’, sampling interval ‘T’ and
normalized frequencies ‘f’ and ‘ω’. (2x5 = 10)
b.) How many ‘sinc’ functions are required to reconstruct the signal for an interval of one cycle and
calculate maximum amplitude of each ‘sinc’ function for the above-mentioned interval. (5+10 =15)
c.) The ideal reconstruction formula (ideal interpolation formula) is written below
π (t − nT )
sin( ) c.1) Write a general formula to generate a
n =∞
y (t ) = ∑ x ( nT ) T sinc function and write a formula for
n =−∞ π (t − nT ) sample two taken at t = 0.2 msec by
( )
T placing the numerical values. (10)
c.2) How reconstructed signal ‘Y(t)’ is
generated by all sinc functions (5)

Dr Muhammad Ashraf 8:27 A11/P11 29/11/2009 Page 1 of 5


Digital Signal Processing by Dr Muhammad Ashraf, S-V, Test-3, Dated: 8-12-2003, Time allowed: 55min
Q# 2) A plots of BP FIR digital filter is shown below. M = 5, Fs = 100 samples/sec. The band pass filter is
designed by multiplying the impulse response of the low pass filter with cos(θ), θ=pi/3. (40)
a.) Calculate central, lower and upper cut-off frequencies of the Bpfilter. (3x5=15)
b.) Plot zeroes in the Z-Plane (10)
c.) Calculate h(nBP) of the BP filter for n = –5, 2, 0,1 and 3. (2x5=10)
d.) What is use of the last plot ‘frequency vs. amplitude’. (5)

-M ≤ n ≤ M

1
h( nLP ) =
2M +1
cos(w.n) = cos(2.pi.F.n.T)

Time (Sec) ----


w = pi/3 ==> F=Fs/6

Time (Sec) ----

Time (Sec) ----


1
h( nBP ) = cos( 2.π.F
2 M +1

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Digital Signal Processing by Dr Muhammad Ashraf, S-V, Test-3, Dated: 8-12-2003, Time allowed: 55min
Q# 3) A frequency plots of LP FIR digital filter is shown below. Sampling rate ‘Fs’ = 20 samples/sec. The cut off
frequency ‘fc’ of the low pass filter is 5 Hz. The h(n) is calculated by taking its inverse fourier transform
( sinc function) in time domain. (40)
a.) Calculate time required for the First zero crossing of its inverse fourier transform in time domain. (10)
b.) Calculate the total samples (N = 2M+1) required for the impulse response up to first zero crossing.(10)
c.) Calculate the impulse response h(n) = 2.Fc.sinc(2.Fc.n.T) upto 1st zero crossing. (10)
Fs=20 Samples/sec
1

-10 -5 0 5 10
-Fs/2 -Fc Frequency Hz Fc Fs/2
d.) Repeat parts a,b for Fc = 2Hz. (10)

Q# 4) The equation for the transfer function of an IIR filter written below. (40)
a.) Plot pole-zero in Z-Plane (10)
b.) Write difference equation to calculate y(n). (10)
c.) If Fs = 200Hz, Calculate Amplitude vs frequency for the frequencies F = 0Hz, 25Hz, 50Hz, 75Hz and
100Hz. (15)
d.) Which type (LP. BP or HP) of filter is it. (5)
Y ( z) 0.5( z − 1)( z + 1)
H ( z) = = π π
X ( z ) ( z − 0.5e )( z − 0.5e )
j
2
j
2

Q# 5) The plot of a digital resonator having a complex conjugate pole pair on the unit circle in the Z-Plane and a
2nd order zero at the origin is shown to the right. (40)
Y ( z) Z-Plane
a.) Calculate its Transfer function. H ( z ) = =? (10)
X ( z)
b.) Calculate its difference equation (5)
c.) Calculate its impulse response ‘h(n)’ for 0 ≤ n ≤ 5, by applying
impulse input (X(n) = δ (n)). (10) 45o
d.) The Transfer function of the comb filter H(z) = 1-Z-m assuming m = -45o
20, Write its difference equation. (5)
e.) The combination (cascading) of comb filter and resonator provides
simple recursive filter. Cascade the above mentioned comb filter and
resonator and write the final difference equation of the BP filter. (10)

Q# 6) The plot for numerical integration and numerical


differentiation is shown to the right. For Numerical
a.) Derive the difference equation for For Numerical
differentation
differentiation filter, the value of the numerical differentation
derivative at t = nT and calculate its Z- For Numerical
Transform. (10) For Numerical
integration
b.) Derive the difference equation for integration X(t) integration
filter, the value of the numerical integral at t =
(n+1)T and calculate its Z-Transform. (10)
c.) Convert the analog filter with system function
1
H ( s) = into a digital IIR filter by
( s + 0.1)) + 92
(n-1)T nT (n+1)T
use of (n+2)T t
1.) S= (1-Z )/T
-1
(10)
2.) S= (Z-Z-1)/T (10)

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Digital Signal Processing by Dr Muhammad Ashraf, S-V, Test-3, Dated: 8-12-2003, Time allowed: 55min
Supporting Material to solve the paper.
Fig. for Q-2
H(n) of the Moving Average FIR Filter and its Analysis in Frequency domain

sin π.F .Width


X ( f ) = A.Width .
π.F .Width

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Digital Signal Processing by Dr Muhammad Ashraf, S-V, Test-3, Dated: 8-12-2003, Time allowed: 55min
Fig. for Q-3

Cut off frequency ‘Fc’ of the


ideal low pass filter

=5 samples for first zero


crossing

For first zero crossing -M ≤ n ≤ M


h(n) = 2.Fc.sin(2.n.Fc/Fs)

Dr Muhammad Ashraf 8:27 A11/P11 29/11/2009 Page 5 of 5

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