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WorksheetsEC 8553 DTSP - MODEL QP
Total questions: 50
Worksheet time: 50mins
What is the circular convolution of the sequences X1(n)={2,1,2,1} and x2(n)=(1,2,3,4)
{14,14,16,16}
{16,16,14,14}
{2,3,6,4}
{14,16,14,16}
A pictorial illustration that shows operations described by the difference equation using interconnections between
adder, multipliers, and delay elements.
Basic elements
Block diagram
All-zero system
All-pole system
Canonical structure is a structure that is implemented using the minimum possible number of delay elements. __________________is a canonical structure.
Direct form I
Direct form II
Both Direct form I & Direct form II
None of these
A structure that is obtained by expressing the system function as a product of second-order sections.
Cascade form
Direct form I
Direct form II
Linear-phase form
Computation of the analog frequency Ω from the digital frequency ω using the frequency warping formula so that
the frequency-distortion in bilinear transformation is compensated
Spectral factorization
Zero-phase filtering
Prewarping
all of the above
The magnitude-squared response of lowpass filter is given by ∣Ha(jΩ)∣2=1+64Ω61 , then the order of filter is
6
4
3
12
To obtain digital filter H(z) by impulse invariance transformation, transform analog poles {pk} into digital poles {epkT} , Where H(z) is,
H(z)=k=1∑N1−epkTZ−1Rk
H(z)=k=1∑N1−epkTZRk
H(z)=k=1∑N1+epkTZ−1Rk
H(z)=k=1∑NepkTZ−1Rk
If the bilinear transformation is used to convert a continuous-time to a discrete-time filter, the frequency transformation may be performed ______________ the bilinear transformation
either before or after
before
after
none of these
Practically realizable IIR filters, that is, causal and stable filters with rational system functions have a ________________ phase response, which complicates filter design using optimization techniques.
nonlinear
Linear
inverse
constant
The impulse invariance and bilinear mappings are two most popular transformations that convert analog into digital filters. The better and more versatile of the two is _______________
bilinear mapping.
impulse invariance mapping
both
Fourier transform
A one-to-one analog to digital filter transformation that maps analog complex frequency s into digital complex frequency z
Butterworth approximation
Cauer filter
Bilinear transformation
Chebyshev approximation
An analog to digital filter transformation that preserves the shape of the analog filter impulse response.
Impulse-invariance transformation
Frequency (band) transformation
Frequency warping
Prewarping
Direct form I structure requires
(M + N) delay elements
(M -N) delay elements
(M + N)/2 delay elements
(M + N+2) delay elements
3-dB Butterworth Lowpass Prototype Transfer Functions, when the order of filter N =1 is
s+11
s−11
s2+s+21
s2+11
3-dB Butterworth Prototype Functions when the order of the filter N=3 is
(s+1)(s2+s+1)1
(s−1)(s2+s+1)1
(s2+s+1)1
(s+1)(s2+2s+1)1
The Bilinear Transformation maps the left half of an s-plane to the__________ unit circle of the z-plane
inside
outside
imaginary axis of
centre of the
If a continuous time signal x(t) with spectrum X(F) is sampled at a rate Fs=1/T samples per second, the spectrum of the sampled signal is _____________
Non periodic repetition
Non periodic non-repetition
Periodic repetition
None of the mentioned
The window function
0.54−0.46 cos (M−12πn) representBlackman
Hamming
Hanning
Bartlett (triangular)
Hanning window function can be expressed as
WHann(n)=0.5(1−cos (M−12πn))
WHann(n)=0.5(1+cos (M−12πn))
WHann(n)=0.5(1−2cos (M−12πn))
WHann(n)=0.5(1+2cos (M−12πn))
Approximate transition width of main lobe of Hamming window is
M8π
M+18π
M−18π
M−16π
An FIR filter of length M having the unit sample response h(n)=π (n−2(M−1 ))(sin ω c(n−2(M−1))), n=2(M−1) if M is selected to be odd, the value of h(n) at n=2 (M−1) is
h(2M−1)=πωc
h(2M−1)=2πωc
h(2M−1)=2ωc
h(2M−1)=π2 ωc
Peak sidelobe of -13 dB connected to
Hanning Window
Hamming Window
Rectangular window
Barlett Window
The Fourier Transform of the Rectangular window is
WR(ω )=e−(jω 2(M−1)) (sin(2ω)sin(2ωM))
the Phase response is
θ(ω)={−ω(2(M−1))}, When sin(2ωM)≥0
θ(ω)={−ω(2(M+1))}, When sin(2ωM)≥0
θ(ω)={ω(2(M−1))}, When sin(2ωM)≥0
θ(ω)={ω(2(M+1))}, When sin(2ωM)≥0
The truncation of the Fourier series is known to introduce ripples in the frequency response characteristic H(ω) due to the nonuniform convergence of the Fourier series at a discontinuity. The oscillatory behavior near the band edge of the filter is called the
Gibbs phenomenon
Pass band ripple
Change in Transition band
None of these
Frequency response of Rectangular window is
WR(ω )=sin(ω)sin(ω M)
WR(ω )=(2ω)sin(ω 2M)
WR(ω )=sin(2ω)sin(ω 2M−1)
What is the nyquist rate of the signal x(t)=3cos(50*pi*t)+10sin(300*pi*t)-cos(100*pi*t)?
50Hz
100Hz
200Hz
300Hz
What is the discrete-time signal obtained after sampling the analog signal x(t)=cos(2000*pi*t)+sin(5000*pi*t) at a sampling rate of 5000samples/sec?
cos(2.5*pi*n)+sin(pi*n)
cos(0.4*pi*n)+sin(pi*n)
cos(2000*pi*n)+sin(5000*pi*n)
None of the mentioned
Which bit coder is required to code a signal with 16 levels?
8 bit
4 bit
2 bit
1 bit
What is the dead band of a single pole filter with a pole at 3/4 and represented by 4 bits?
(-1/2,1/2)
(-1/8,1/8)
(-1/4,1/4)
(-1/16,1/16)
In the equation
SQNR = 6.02b + 16.81 – 20log10σxR for R = 6σx the equation becomes?
SQNR = 6.02b-1.25 dB
SQNR = 6.87b-1.55 dB
SQNR = 6.02b+1.25 dB
SQNR = 6.87b+1.25 dB
In the mathematical model for the quantization error eq (n), to carry out the analysis, what are the assumptions made about the statistical properties of eq (n)?
i. The error eq (n) is uniformly distributed over the range — Δ/2 < eq (n) < Δ/2.
ii. The error sequence is a stationary white noise sequence. In other words, the error eq (m) and the error eq (n) for m≠n are uncorrelated.
iii. The error sequence {eq (n)} is uncorrelated with the signal sequence x(n).
iv. The signal sequence x(n) is zero mean and stationary.
i, ii & iii
i, ii, iii, iv
i, iii
ii, iii, iv
The roots of the polynomial H(z) are identical to the roots of the polynomial H(z-1).
True
False
If the unit sample response h(n) of the filter is real, complex valued roots need not occur in complex conjugate pairs.
True
False
Which of the following defines the rectangular window function of length M-1?
w(n)=1, n=0,1,2...M-1
=0, else where
w(n)=1, n=0,1,2...M-1
=-1, else where
w(n)=0, n=0,1,2...M-1
=1, else where
None of the mentioned
In the frequency sampling method for FIR filter design, we specify the desired frequency response Hd(ω) at a set of equally spaced frequencies.
True
False
What is the frequency response of a system with input h(n) and window length of M?
n=0∑M−1h(n)ejωn
n=0∑Mh(n)ejωn
n=0∑Mh(n)e−jωn
n=0∑M−1h(n)e−jωn
Why is it desirable to optimize frequency response in the transition band of the filter?
Increase side lobe
Reduce side lobe
Increase main lobe
None of the mentioned
The frequency sampling design method is attractive when the FIR filter is realized in the frequency domain by means of the DFT.
True
False
Which of the following values can a frequency response take in frequency sampling technique?
Zero
One
Zero or One
None of the mentioned
