Assignment for B.

Tech ECE
Digital Signal Processing & Applications (EC-302)

Submission date 09/06/2020

Q.1 Determine the IDFT by finding the values of a and b for the sequence

x [ n ] ={21 ,−3+ j5.196 ,a+ j 1.732,−3 ,−3− j 1.732 ,−3− jb }

Q.2 Find the IDFT of the sequence X [ k ]={10 ,−2+ j2 ,−2,−2− j2 } using DIT-FFT and DIF-FFT
algorithms.

Q.3 Draw the cascaded and parallel realization of

3
( 1−z−1 )
H ( z) =
( 1−0.5 z−1 )( 1−1.25 z−1)
Q.4 Find the IDFT of the given below sequence using DIF-FFT algorithm.

X [ k ]={4 ,1− j 2.414 , 0 ,1− j 0.414 , 0 , 1+ j 0.414 , 0 , 1+ j 2.414 }

Q.5 Prove that ℜ [ X ( k ) ] =ℜ [ X (−k ) ] and | X ( k )|=| X (−k )| for a real periodic sequence x (n).

Q.6 (a)Use bilinear transformation to design a first order Butterworth LPF with 3dB cut off
frequency 0f 2 π .

(b) What is the warping effect? Also explain the pre-wrapping.

Q.7 Explain the derivation of optimal solution for a wiener filter.

Q.8 Write short notes of the parametric and non-parametric methods of spectral estimation.

Q.9 (a) Design a high pass filter using Hamming windows with a cut off frequency of 1.2 radians
and M=9.

Q.10 (a) Prove that discrete time harmonics are not always periodic in frequency.

(b) Derive the Expression for impulse invariance technique for obtaining transfer function of
digital filter from analog filter. Derive the necessary equation for relationship between
frequency of analog and digital filter.

Q.11 (a) Discuss the method of determination of Chebyshev polynomials and their
properties.
(b) Find the order and poles of a low pass Butterworth filter that has 3dB bandwidth
of 500Hz and attenuation of 40dB at 1kHz.
Q.12 (a) Realise the following system function using minimum number of multipliers.
1 1 1 1
H ( z ) =1+ z−1+ z −2 + z−3 + z−4 + z −5
3 4 4 3
(b) What are the limitation of optimum filter? Also compare optimum filter versus
adaptive filter.
Q.13 (a) Derive the update equation for least mean square (LMS) algorithm for system
identification.
(b) Write a short notes for the following
a) Kalman filter
b) Adaptive echo cancellation over telephone channels
c) Adaptive line enhancer.
Q.14 (a) The data sequence [0,1,0,1,0,1,0,1] is now assumed to contain a random noise
component such that the new data sequence becomes [0.763,0.1656, 0.424, 1.939, 0.133,
1.881, 0.328, 1.348]. Calculate the power spectrum of this noisy data. Estimate the signal to
noise ratio from the sampled data.

(b) Explain the DSP processor memory architecture.

Q.15 Determine the computational complexity of a single stage interpolator to be designed to

increase the sampling rate from 600Hz to 9kHz. The interpolator to be designed as an equiripple FIR
filter with a passband edge at 200 Hz, a passband ripple of 0.002 and a stopband ripple of 0.004.
Estimate the order of FIR filter. Also develop two stage design of the above interpolator by a factor
of 3 and 5 and compare its computational complexity with that of single stage design.

1
Q.16 (a) convert the analog filter with system function H ( s )= into a digital filter using bilinear
s
transformation. AssumeT=2s. find (i)The difference equation for the digital filter relating the
input x(n) to thr output y(n). (ii) Draw the magnitude and phase characteristics of both analog
and digital filters and compare them.
(b) Compare the IIR and FIR filters, which one is preferred in computer added design and
why?

Q.17 (a) A 126-point DFT X [k ] of a real-valued sequence x[n] has the following DFT
samples; X [0 ]=12.8+ jα , X [13]=−3.7+ j2.2, X [k 1 ]=9.1− j 5.4, X [51]=− j 1.7,
X [ k 2 ]=9.1+ j 5.4 X [63 ]=13+ jβ, X [k 3 ]=γ + j 1.7, X [79]=6.3+ jδ, X [108]=ε + j2.3,
X [k 4 ]=−3.7− j 2.2.Remaining DFT samples are assumed to be zero values.

(i) Determine the values of the indices K 1 , K 2 , K 3and K 4

(ii) Determine the values of α , β , δand ε
(iii)Determine the expression for { x [ n ] } without computing IDFT.
(b) Write short notes about filter bank.

Q.18 (a) Find the number of complex additions and complex multiplications required to find DFT
for 16 point signal. Compare them with number of computations required, If FFT algorithm is
used.

(b) Find 8-point FFT of , x ( n )={ 1,2,2,2,1 } using signal flow graph of Radix-2 Decimation in
frequency FFT.
Q.19 (a) A length 10 sequence x(n) has a real valued 10 point DFT X(k). The first six sample of
x(n) are given by: x(0)=2.5, x(1)=7-i3, x(2)= -3.2+i1.3,x(3)=-2+i5, x(4)=7+I, x(5)=5. Find the
remaining four samples of x (n).
(b) Explain Con Neumann and Harvard architectures and explain why the Von Neumann
architecture is not suitable for DSP operations.
Q.20 (a) Compare the single stage, two stage, three stage and multistage realization of the
decimator with the following specifications. Sampling rate of a signal has to be reduced from
10kHz to 500Hz, the decimation filter H(z) has the pass band edge (F p) to be 150Hz, stop
band edge (Fs) to be 200Hz, pass band ripple (δ p) to be 0.002, stop band ripple (δ s)to be
0.001 and M=20.
(b) Explain the sub-band coding with its applications.
Q.21 (a) Give the expression for power spectrum estimates for AR, MA, and ARMA models?
(b) Why Yule-Walker (AR) model is widely used?
Q.22 (a) What is aliasing and anti-aliasing filter ?
(b) Why is it necessary to have N ≥ L where N: N-point DFT & L: Length of signal x (n)?

Q.23 (a) What are approximation polynomials? Explain the approximation theory for maximally
flat filter by taking proper example.
(b)Calculate the DFT of a sequence x(n) = {1,1-j,5,-1+j, -1}.
z +1
Q.24 (a) Given that H ( z ) = 2 is a causal system, find its (i) Transfer function
z −0.9 z +0.81
representation, (ii) Difference equation representation, (iii) Impulse response representation.

1+ 2 z−1
(b) Determine the causal sequence x(n) for X(z), which is given by X ( z )= .
1−2 z−1+ 4 z−2
Q.25 (a) Realize a direct form I and canonic form for the following linear phase system
h[n]=[1, 2, 3, 3, 2, 1]
(b) A LIT system is defined by following transfer function
1 1
X ( z )=
1− ( 2 ) −1
z +( ) z
4
−2

.
1 1 1
1+ ( ) z −( ) z + ( ) z
−1 −2 −3
8 5 6
Draw the signal flow graph, transposed signal flow graph and transposed direct form-II
structure.
(c) Develop the lattice ladder structure for the filter with difference equation
y(n)+(3/4)y(n-1)+(1/4)y(n-2)=x(n)+2x(n-1)