St. Joseph'S College of Engineering Department of Ece Multiple Choice Questions (MCQ) Subject: Ec 8553-Discrete Time Signal Processing (Unit I)

EC 8553 – DISCRETE TIME SIGNAL PROCESSING Department of ECE UNIT I – Signals & Systems
St. JOSEPH’S COLLEGE OF ENGINEERING

DEPARTMENT OF ECE
MULTIPLE CHOICE QUESTIONS (MCQ)
SUBJECT: EC 8553- DISCRETE TIME SIGNAL PROCESSING ( UNIT I)
1. The formula y(n)=∑x(k)h(n−k) that gives the response y(n) of the LTI system as
the function of the input signal x(n) and the unit sample response h(n) is known
as ______________
a) Convolution sum
b) Convolution product
c) Convolution Difference
d) None of the mentioned
Answer: a
Explanation: The input x(n) is convoluted with the impulse response h(n) to yield the
output y(n). As we are summing the different values, we call it as Convolution sum.
2. What is the order of the four operations that are needed to be done on h(k) in
order to convolute x(k) and h(k)?
Step-1:Folding
Step-2:Multiplication with x(k)
Step-3:Shifting
Step-4:Summation
a) 1-2-3-4
b) 1-2-4-3
c) 2-1-3-4
d) 1-3-2-4
Answer: d
Explanation: First the signal h(k) is folded to get h(-k). Then it is shifted by n to get
h(n-k). Then it is multiplied by x(k) and then summed over -∞ to ∞
3. The impulse response of a LTI system is h(n)={1,1,1}. What is the response of

the signal to the input x(n)={1,2,3}?
a) {1,3,6,3,1}
b) {1,2,3,2,1}
c) {1,3,6,5,3}
d) {1,1,1,0,0}
Answer: c
Explanation: Let y(n)=x(n)*h(n)(‘*’ symbol indicates convolution symbol)
From the formula of convolution we get,
St. Joseph’s College of Engineering Page 1

y(0)=x(0)h(0)=1.1=1
y(1)=x(0)h(1)+x(1)h(0)=1.1+2.1=3
y(2)=x(0)h(2)+x(1)h(1)+x(2)h(0)=1.1+2.1+3.1=6
y(3)=x(1)h(2)+x(2)h(1)=2.1+3.1=5
y(4)=x(2)h(2)=3.1=3
Therefore, y(n)=x(n)*h(n)={1,3,6,5,3}
4. x(n)*(h1(n)*h2(n))=(x(n)*h1(n))*h2(n).
a) True
b) False
Answer: a
Explanation: According to the properties of convolution, Convolution of three signals
obeys Associative property
5. x(n)*[h1(n)+h2(n)]=x(n)*h1(n)+x(n)*h2(n).
a) True
b) False
Answer: a
Explanation: According to the properties of the convolution, convolution exhibits
distributive property
6. The discrete time function defined as x(n)=n for n≥0;x(n)=0 for n<0 is an
_____________
a) Unit sample signal
b) Unit step signal
c) Unit ramp signal
Answer: c
Explanation: When we plot the graph for the given function, we get a straight line
passing through origin with a unit positive slope. So, the function is called a unit
ramp signal
7. Time scaling operation is also known as ___________

a) Down-sampling
b) Up-sampling
c) Sampling

Answer: a
Explanation: If the signal x(n) was originally obtained by sampling a signal xa(t), then
x(n)=xa(nT). Now, y(n)=x(2n)(say)=xa(2nT). Hence the time scaling operation is
equivalent to changing the sampling rate from 1/T to 1/2T, that is to decrease the
rate by a factor of 2. So, time scaling is also called as down-sampling.
8. The function given by the equation x(n)=1, for n=0; x(n)=0, for n≠0 is a
_____________
a) Step function
b) Ramp function
c) Triangular function
d) Impulse function
Answer: d
Explanation: According to the definition of the impulse function, it is defined only at
n=0 and is not defined elsewhere which is as per the signal given.
9. The system described by the input-output equation y(n)=nx(n)+bx3(n) is a

__________
a) Static system
b) Dynamic system
c) Identical system
Answer: a
Explanation: Since the output of the system y(n) depends only on the present value
of the input x(n) but not on the past or the future values of the input, the system is
called as static or memory-less system.
10. Whether the system described by the input-output equations y(n)=x(n)-x(n-1) a

Time-variant system.
a) True
b) False
Answer: b
Explanation: If the input is delayed by k units then the output will be
y(n,k)=x(n-k)-x(n-k-1) If the output is delayed by k units then y(n-k)=x(n-k)-x((n-k)-1)
=>y(n,k)=y(n-k). Hence the system is time-invariant.
11. If the output of the system of the system at any ‘n’ depends only the present or
the past values of the inputs then the system is said to be __________
a) Linear
b) Non-Linear

c) Causal
d) Non-causal
Answer: c
Explanation: A system is said to be causal if the output of the system is defined as
the function shown below y(n)=F[x(n),x(n-1),x(n-2),…] So, according to the
conditions given in the question, the system is a causal system.
12. If a system do not have a bounded output for bounded input, then the system is
said to be __________
a) Causal
b) Non-causal
c) Stable
d) Non-stable
Answer: d
Explanation: An arbitrary relaxed system is said to be BIBO stable if it has a
bounded output for every value in the bounded input. So, the system given in the
question is a Non-stable system.
13. Which of the following is done to convert a continuous time signal into discrete
time signal?
a) Modulation
b) Sampling
c) Differentiation
d) Integration
Answer:b
Explanation: A discrete time signal can be obtained from a continuous time signal by
replacing t by nT, where T is the reciprocal of the sampling rate or time interval
between the adjacent values. This procedure is known as sampling.
14. The even part of a signal x(t) is?
a) x(t)+x(-t)
b) x(t)-x(-t)
c) (1/2)[(x(t)+x(-t)]
d) (1/2)[(x(t)-x(-t)]
Answer: c
Explanation: Let x(t)=xe(t)+xo(t)
=>x(-t)=xe(-t)-xo(-t)
By adding the above two equations, we get
xe(t)=(1/2)*(x(t)+x(-t))

15. All energy signals will have an average power of ___________

a) Infinite
b) Zero
c) Positive
d) Cannot be calculated
Answer:b
Explanation: For any energy signal, the average power P should be equal to 0
16. x(t) or x(n) is defined to be an energy signal, if and only if the total energy content
of the signal is a ___________
a) Finite quantity
b) Infinite
c) Zero
Answer:a
Explanation: The energy signal should have a total energy value that lies between 0
and infinity.
17. Which of the following conditions made digital signal processing more
advantageous over analog signal processing?
a) Flexibility
b) Accuracy
c) Storage
d) All of the mentioned
Answer: d
Explanation: Digital programmable system allows flexibility in reconfiguring the DSP
operations by just changing the program, as the digital signal is in the form of 1 and
0’s it is more accurate and it can be stored in magnetic tapes
18. If x(n)=(0,0,1,2,3,4,0,0) then x(n-2) is?

a) (0,0,2,4,6,8,0,0)
b) (0,0,1,2,3,4,0,0)
c) (1,2,3,4,0,0,0,0)
d) (0,0,0,0,1,2,3,4)
Answer: d
Explanation: The signal x(n) is shifted right by 2

19. What are the important block(s) required to process an input analog signal to get
an output analog signal?
a) A/D converter
b) Digital signal processor
c) D/A converter
Answer: d
Explanation: The input analog signal is converted into digital using A/D converter
and passed through DSP and then converted back to analog using a D/A converter
20. What is the physical device that performs an operation on the signal?
a) Signal source
b) System
c) Medium
Answer: b
Explanation: A system is a physical device which performs the operation on the
signal and modifies the input signal
21. If all the poles of H(z) are inside the unit circle, then the system is said to be
____________
a) Only causal
b) Only BIBO stable
c) BIBO stable and causal
Answer: c
Explanation: If all the poles of H(z) are inside an unit circle, then it follows the
condition that |z|>r < 1, it means that the system is both causal and BIBO stable
22. What are the main characteristics of Anti aliasing filter?

a) Ensures that bandwidth of signal to be sampled is limited to frequency range
b) To limit the additive noise spectrum and other interference, which corrupts the
signal
c) All of the mentioned
Answer:c
Explanation: The anti aliasing filter is an analog filter which has a twofold purpose.
First, it ensures that the bandwidth of the signal to be sampled is limited to the
desired frequency range. Using an antialiasing filter is to limit the additive noise

spectrum and other interference, which often corrupts the desired signal. Usually,
additive noise is wideband and exceeds the bandwidth of the desired signal.
23. What is the configuration of system for digital processing of an analog signal?
a) Analog signal|| Pre-filter -> D/A Converter -> Digital Processor -> A/D
Converter -> Post-filter
b) Analog signal|| Pre-filter -> A/D Converter -> Digital Processor -> D/A
Converter -> Post-filter
c) Analog signal|| Post-filter -> D/A Converter -> Digital Processor -> A/D
Converter -> Pre-filter
Answer:b
Explanation: The anti-aliasing filter is an analog filter which has a twofold
purpose.
Analog signal|| Pre-filter -> A/D Converter -> Digital Processor -> D/A Converter -
> Post-filter
24. The process of converting discrete-time continuous valued signal into discrete-
time discrete valued(digital) signal is known as:
a) Sampling
b) Quantization
c) Coding
Answer: b
Explanation: In this process, the value of each signal sample is represented by
a value selected from a finite set of possible values. Hence this process is
known as ‘quantization’
25. What is the Nyquist rate of the signal

x(t)=3cos(50*pi*t)+10sin(300*pi*t)-cos(100*pi*t)?
a) 50Hz
b) 100Hz
c) 200Hz
d) 300Hz
Answer: d
Explanation: The frequencies present in the given signal are F1=25Hz,
F2=150Hz, F3=50Hz. Thus Fmax=150Hz and from the sampling theorem,
Nyquist rate=2*Fmax
Therefore, Fs=2*150=300Hz.

EC 8553 – DISCRETE TIME SIGNAL PROCESSING DEPARTMENT OF ECE UNIT I – DFT & FFT

DEPARTMENT OF ECE
SUBJECT: EC 8553- DISCRETE TIME SIGNAL PROCESSING ( UNIT I)
1.
The Nth root of unity WN is given as ______________
a) e j2πN
b) e -j2πN
c) e -j2π/N
d) e j2π/N
Answer: c
Explanation: Twiddle factor W N= e –j2π/N = (e-j2π) 1/N = (cos2π – j sin2π)1/N
= (1)1/N = Nth Root of Unity
2. Which of the following is true regarding the number of computations requires to

compute an N-point DFT?
a) N2 complex multiplications and N(N-1) complex additions
b) N2 complex additions and N(N-1) complex multiplications
c) N2 complex multiplications and N(N+1) complex additions
d) N2 complex additions and N(N+1) complex multiplications
Answer: a
Explanation: The formula for calculating N point DFT is given as
X(k)=∑ x(n) e−j2πkn/N
From the formula given at every step of computing we are performing N complex
multiplications and N-1 complex additions. So, in a total to perform N-point DFT we
perform N2 complex multiplications and N(N-1) complex additions
3. What is the DFT of the four point sequence x(n)={0,1,2,3}?

a) {6,-2+2j-2,-2-2j}
b) {6,-2-2j,2,-2+2j}
c) {6,-2-2j,-2,-2+2j}
d) {6,-2+2j,-2,-2-2j}
Answer: d
Explanation: Given x(n)={0,1,2,3}
We know that the 4-point DFT of the above given sequence is given by the
expression
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X(k)=∑x(n)e−j2πkn/N
In this case N=4
=>X(0)=6,X(1)=-2+2j,X(2)=-2,X(3)=-2-2j
4. If W 4100=W x200, then what is the value of x?

a) 2
b) 4
c) 8
d) 16
Answer: c
Explanation: We know that according to the periodicity and symmetry property,
100/4=200/x=>x=8
5. If x(n) and X(k) are an N-point DFT pair, then x(n+N)=x(n).

a) True
b) False
Answer: a
6. If x(n) and X(k) are an N-point DFT pair, then X(k+N)=?

a) X(-k)
b) -X(k)
c) X(k)
Answer: c
6. If x(n) is a real sequence and X(k) is its N-point DFT, then which of the following
is true?
a) X(N-k)=X(-k)
b) X(N-k)=X*(k)
c) X(-k)=X*(k)
Answer: d
7. What is the circular convolution of the sequences x1(n)={2,1,2,1} and

x2(n)={1,2,3,4}?
a) {14,14,16,16}
b) {16,16,14,14}
c) {2,3,6,4}
d) {14,16,14,16}

Answer: d
Explanation: We know that the circular convolution of two sequences is given by the
expression
x(m)= ∑x1(n).x2(m−n)N
For m=0, x2((-n))4={1,4,3,2}
For m=1, x2((1-n))4={2,1,4,3}
For m=2, x2((2-n))4={3,2,1,4}
For m=3, x2((3-n))4={4,3,2,1}
Now we get x(m)={14,16,14,16}
8. If X(k) is the N-point DFT of a sequence x(n), then what is the DFT of x*(n)?
a) X(N-k)
b) X*(k)
c) X*(N-k)
Answer: c
Explanation: According to the complex conjugate property of DFT, we have if X(k) is
the N-point DFT of a sequence x(n), then what is the DFT of x*(n) is X*(N-k)
9. Overlap add and Overlap save are the two methods for linear FIR filtering a long
sequence on a block-by-block basis using DFT.
a) True
b) False
Answer:a
Explanation: In these two methods, the input sequence is segmented into blocks and
each block is processed via DFT and IDFT to produce a block of output data. The
output blocks are fitted together to form an overall output sequence which is identical
to the sequence obtained if the long block had been processed via time domain
convolution. So, Overlap add and Overlap save are the two methods for linear FIR
filtering a long sequence on a block-by-block basis using DFT
10. In Overlap save method of long sequence filtering, what is the length of the input
sequence block?
a) L+M+1
b) L+M
c) L+M-1
Answer:c
Explanation: In this method, each data block consists of the last M-1 data points of the

previous data block followed by L new data points to form a data sequence of length
N=L+M-1.
11. In Overlap save method of long sequence filtering, how many zeros are
appended to the impulse response of the FIR filter?
a) L+M
b) L
c) L+1
d) L-1
Answer:d
Explanation: The impulse of the FIR filter is increased in length by appending L-1 zeros
and an N-point DFT of the sequence is computed once and stored.
12. . In which of the following methods, the input sequence is considered as shown
in the below diagram?
a) Overlap save method

b) Overlap add method
c) Overlap add & save method
Answer:a
Explanation: From the figure given, we can notice that each data block consists of the
last M-1 data points of the previous data block followed by L new data points to form a
data sequence of length N+L+M-1 which is same as in the case of Overlap save
method.
13. In which of the following methods, the output sequence is considered as shown
in the below diagram?

a) Overlap save method

b) Overlap add method
c) Overlap add & save method
Answer: b
Explanation: From the figure given, it is clear that the last M-1 points of the first
sequence and the first M-1 points of the next sequence are added and nothing is
discarded because there is no aliasing in the input sequence. This is same as in the
case of Overlap add method.
14. What is the value of x(n)*h(n), 0≤n≤11 for the sequences x(n)={1,2,0,-3,4,2,-1,1,-
2,3,2,1,-3} and h(n)={1,1,1} if we perform using overlap add fast convolution
technique?
a) {1,3,3,1,1,3,5,2,2,2,3,6}
b) {1,2,0,-3,4,2,-1,1,-2,3,2,1,-3}
c) {1,2,0,3,4,2,1,1,2,3,2,1,3}
d) {1,3,3,-1,1,3,5,2,-2,2,3,6}
Answer: d
15. If the signal to be analyzed is an analog signal, we would pass it through an anti-
aliasing filter with B as the bandwidth of the filtered signal and then the signal is
sampled at a rate __________
a) Fs ≤ 2B
b) Fs ≤ B
c) Fs ≥ 2B
d) Fs = 2B
Answer:c
Explanation: The filtered signal is sampled at a rate of Fs≥ 2B, where B is the
bandwidth of the filtered signal to prevent aliasing.

16. What is the highest frequency that is contained in the sampled signal?
a)2Fs
b)Fs/2
c)Fs
Answer:b
Explanation: We know that, after passing the signal through anti-aliasing filter, the
filtered signal is sampled at a rate of Fs≥ 2B=>B≤ Fs/2.Thus the maximum
frequency of the sampled signal is Fs/2.
17. WNk+N/2=?
a)W Nk
b)-W Nk
c)W N-k
Answer:b
Explanation: According to the symmetry property, we get W Nk+N/2= - WNk
18. Divide-and-conquer approach is based on the decomposition of an N-point DFT

into successively smaller DFTs. This basic approach leads to FFT algorithms.
a)True
b) False
Answer:a
Explanation: The development of computationally efficient algorithms for the DFT
is made possible if we adopt a divide-and-conquer approach. This approach is
based on the decomposition of an N-point DFT into successively smaller DFTs.
This basic approach leads to a family of computationally efficient algorithms known
collectively as FFT algorithms.
19. If we split the N point data sequence into two N/2 point data sequences f 1(n) and
f2(n) corresponding to the even numbered and odd numbered samples of x(n),
then such an FFT algorithm is known as decimation-in-time algorithm.
a)True
b) False
Answer:a
Explanation: Let us consider the computation of the N=2 v point DFT by the divide
and conquer approach. We select M=N/2 and L=2. This selection results in a split
of N point data sequence into two N/2 point data sequences f 1(n) and f2(n)

corresponding to the even numbered and odd numbered samples of x(n),

respectively, that is
f1(n)=x(2n),f2(n)=x(2n+1),n=0,1,2…N/2-1. Thus f1(n) and f2(n) are obtained by
decimating x(n) by a factor of 2, and hence the resulting FFT algorithm is called a
decimation-in-time algorithm.
20. The total number of complex multiplications required to compute N point DFT by
radix-2FFTis?
a)(N/2)log2N
b)Nlog2N
c)(N/2)logN
Answer:a
Explanation: The decimation of the data sequence should be repeated again and
again until the resulting sequences are reduced to one point sequences. For N=2 v,
this decimation can be performed v=log2N times. Thus the total number of complex
multiplications is reduced to (N/2)log2N
21. For a decimation-in-time FFT algorithm, which of the following is true?

a)Both input and output are in order
b)Both input and output are shuffled
c)Input is shuffled and output is in order
d) Input is in order and output is shuffled
Answer:c
Explanation: In decimation-in-time FFT algorithm, the input is taken in bit reversal
order and the output is obtained in the order.
22. The following butterfly diagram is used in the computation of __________

a)Decimation-in-timeFFT
b)Decimation-in-frequencyFFT
c)None of the mentioned
Answer:b
Explanation: The above given diagram is the basic butterfly computation in the
decimation-in-frequency FFT algorithm.
23. FFT algorithm is designed to perform complex operations.

a)True
b) False
Answer:a
Explanation: The FFT algorithm is designed to perform complex multiplications
and additions, even though the input data may be real valued. The basic reason
for this is that the phase factors are complex and hence, after the first stage of the
algorithm, all variables are basically complex valued
24. How many complex additions are required to be performed in linear filtering of a
sequenceusingFFTalgorithm?
a)(N/2)logN
b)2Nlog2N
c)(N/2)log2N
d) Nlog2N
Answer:b
Explanation: The number of additions to be performed in FFT are Nlog2N. But in
linear filtering of a sequence, we calculate DFT which requires Nlog 2N complex
additions and IDFT requires Nlog2N complex additions. So, the total number of
complex additions to be performed in linear filtering of a sequence using FFT
algorithm is 2Nlog2N
25. What is the transform that is suitable for evaluating the z-transform of a set of
data on a variety of contours in the z-plane?
a)GoertzelAlgorithm
b)FastFouriertransform
c)Chirp-ztransform

Answer:c
Explanation: Chirp-z transform algorithm is suitable for evaluating the z-transform
of a set of data on a variety of contours in the z-plane. This algorithm is also
formulated as a linear filtering of a set of input data. As a consequence, the FFT
algorithm can be used to compute the Chirp-z transform.
26. How many multiplications are required to calculate X(k) by chirp-z transform if
x(n) is of length N?
a)N-1
b)N
c)N+1
Answer:c
Explanation: We know that yk(n)=W N-kyk(n-1)+x(n).Each iteration requires one
multiplication and two additions. Consequently, for a real input sequence x(n), this
algorithm requires N+1 real multiplications to yield not only X(k) but also, due to
symmetry, the value of X(N-k).
27. What is the total number of quantization errors in the computation of single point
DFT of a sequence of length N?
a)2N
b)4N
c)8N
d)12N
Answer:b
Explanation: Since the computation of single point DFT of a sequence of length N
involves N number of complex multiplications, it contains 4N number of
quantization errors.
28. What is the model that has been adopt for characterizing round of errors in
multiplication?
a)Multiplicative white noise model
b)Subtractive white noise model
c)Additive white noise model
d)None of the mentioned
Answer:c
Explanation: Additive white noise model is the model that we use in the statistical
analysis of round off errors in IIR and FIR filters.
29. The DFT is preferred for
1) Its ability to determine the frequency component of the signal
2) Removal of noise

3) Filter design
4) Quantization of signal
a. 1, 2 and 3 are correct

b. 1 and 2 are correct
c. 1 and 3 are correct
d. All the four are correct
ANSWER: (c) 1 and 3 are correct
30) Frequency selectivity characteristics of DFT refers to

a. Ability to resolve different frequency components from input signal
b. Ability to translate into frequency domain
c. Ability to convert into discrete signal
d. None of the above
ANSWER: (a) Ability to resolve different frequency components from input signal
31)The Cooley–Tukey algorithm of FFT is a

a. Divide and conquer algorithm
b. Divide and rule algorithm
c. Split and rule algorithm
d. Split and combine algorithm
ANSWER: (a) Divide and conquer algorithm
32) FFT may be used to calculate

1) DFT
2) IDFT
3) Direct Z transform
4) In direct Z transform

ANSWER: (b) 1 and 2 are correct
33) DIT algorithm divides the sequence into

a. Positive and negative values
b. Even and odd samples
c. Upper higher and lower spectrum
d. Small and large samples
ANSWER: (b) Even and odd samples

34. In 8- point DFT by radix-2 FFT there are __THREE_ stages of computations with
__FOUR__ butterflies per stage.
N = rm = 2m = 23 ; m- No. of stages of computation, Each stage N/2 butterflies
State whether the following statements are True/False
35.The multiplications of the DFTs of the two sequences is equal to the DFT of the
linear convolution of two sequences - False
DFT supports only Circular Convolution
36.Bit reversing is required for both DIT and DIF algorithms - True
DIT – Bit reversal at I/p

DIF – Bit reversal at O/P
37.The DFT of even sequence is purely imaginary and DFT of odd sequence is purely
real - False
In DFT sequence, real component is even and imaginary component is odd.
38.An LTI is unstable if the impulse response is absolutely summable - False
Condition for Stability - Impulse response should be summable
39.For energy signals, the energy and the average power will be
a) Finite, Zero b) Infinite,1 c) Finite,1 d) Zero, Finite
Ans: a) Finite, Zero

For Power signal, E = Infinite , P = Finite
For Energy signal, E = Finite, P=0
40. The phase factor are multiplied before the add and subtract operations in
a) DIT radix-2 FFT b) DIF radix-2 FFT c) Inverse DFT d) Both a and c
Ans: a) DIT radix-2 FFT
Basic Equation of DIT-FFT is

A= a+ b W Nk

B= a- b W Nk
Basic Equation of DIF-FFT is

A= (a+ b)
B= (a- b) W Nk
41.In a N-point sequence, if N=16, the total number of complex addition and
multiplications using
Radix-2 FFT are,
a) 64 and 80 b) 80 and 64 c) 64 and 32 d) 24 and 12
Ans: c) 64 and 32
Total Number of Computations in FFT
No. of Complex Additions = N log2N
No. of Complex Multiplications = N/2 log2N
42. The complex valued phase factor W N can be represented as,
a) (1)1/2 b) (1)1/N c) (1) N d) (1) -1/N
Ans: b) (1)1/N

Twiddle factor WN= e –j2π/N = (e-j2π) 1/N = (cos2π – j sin2π)1/N = (1)1/N =

Nth Root of Unity
43.Sectioned convolution is performed if one of the sequences is very much larger than
the other in order to overcome,
a) Long delay in getting output b) larger memory space requirement
c) Both a and b d) None of the above
Ans: c) Both a and b

EC 8553- DISCRETE TIME SIGNAL RPOCESSING DEPARTMENT OF ECE UNIT II – IIR FILTER DESIGN

DEPARTMENT OF ECE
SUBJECT: EC 8553- DISCRETE TIME SIGNAL PROCESSING (UNIT II)
1. To implement the linear time invariant recursive system described by the difference
equation y(n)=−∑aky(n−k)+∑bkx(n−k) in Direct form-I, how many number of delay
elements and multipliers are required respectively?
a) M+N+1, M+N
b) M+N-1, M+N
c) M+N, M+N+1
Answer: c
Explanation: From the given equation, there are M+N delays, so it requires M+N number
of delay elements and it has to perform M+N+1 multiplications, so it require that many
number of multipliers.
2. What is the system does the following direct form structure represents?
a) FIR system
b) Purely recursive system
c) General second order system
Answer:b
Explanation: Since the output of the system depends only on the present value of the
input and the past values of the output, the system is a purely recursive system.
3. What is the output of the system represented by the following direct form?

a) y(n)=-a1y(n-1)-a2y(n-2)- b0x(n)-b1x(n-1)-b2x(n-2)
b) y(n)=-a1y(n-1)-a2y(n-2)+b0x(n)
c) y(n)=-a1y(n-1)-a2y(n-2)+ b0x(n)+b1x(n-1)+b2x(n-2)
d) y(n)=a1y(n-1)+a2y(n-2)+ b0x(n)+b1x(n-1)+b2x(n-2)
Answer: c
Explanation: The equation of the difference equation of any system is defined as
y(n)=−∑aky(n−k)+∑bkx(n−k)
In the given diagram, N=M=2
So, substitute the values of the N and M in the above equation.
We get, y(n)=-a1y(n-1)-a2y(n-2)+b0x(n)+b1x(n-1)+b2x(n-2)
4. Which of the following is used in the realization of a system?

a) Delay elements
b) Multipliers
c) Adders
Answer: d
Explanation: From each set of equations, we can construct a block diagram consisting
of an interconnection of delay elements, multipliers and adders.
5. Computational complexity refers to the number of ____________

a) Additions
b) Arithmetic operations
c) Multiplications
Answer: b
Explanation: Computational complexity is one of the factor which is used in the
implementation of the system. It refers to the numbers of Arithmetic operations
(Additions, multiplications and divisions)
6. If M and N are the orders of numerator and denominator of rational system function
respectively, then how many multiplications are required in direct form-I realization of
that IIR filter?
a) M+N-1
b) M+N
c) M+N+1
d) M+N+2
Answer: c
Explanation: From the direct form-I realization of the IIR filter, if M and N are the orders

of numerator and denominator of rational system function respectively, then M+N+1

multiplications are required
respectively, then how many additions are required in direct form-I realization of that
IIR filter?
of numerator and denominator of rational system function respectively, then M+N
additions are required.
respectively, then how many memory locations are required in direct form-I
realization of that IIR filter?
of numerator and denominator of rational system function respectively, then M+N+1
memory locations are required
9. In direct form-I realization, all-pole system is placed before the all-zero system.
a) True
b) False
Answer:b
Explanation: In direct form-I realization, all-zero system is placed before the all-pole
system.
respectively, then how many memory locations are required in direct form-II
realization of that IIR filter?
a) M+N+1
b) M+N
c) Min [M,N]
d) Max [M,N]
Answer:d
Explanation: From the direct form-II realization of the IIR filter, if M and N are the orders
of numerator and denominator of rational system function respectively, then Max[M,N]
memory locations are required.

11. Which of the following is true for the given signal flow graph?
a) Two pole system

b) Two zero system
c) Two pole and two zero system
Answer: c
Explanation: The equivalent filter structure of the given signal flow graph in the direct
form-II is given by as
Thus from the above structure, the system has two zeros and two poles
12. If we reverse the directions of all branch transmittances and interchange the input
and output in the flow graph, then the resulting structure is called as
______________
a) Direct form-I
b) Transposed form
c) Direct form-II
Answer: b
Explanation: According to the transposition or flow-graph reversal theorem, if we reverse
the directions of all branch transmittances and interchange the input and output in the
flow graph, then the system remains unchanged. The resulting structure is known as
transposed structure or transposed form
13. The structure shown below is known as ____________
a) Parallel form structure

b) Cascade structure
c) Direct form
Answer:a
Explanation: From the given figure, it consists of a parallel bank of single pole filters and
thus it is called as parallel form structure.
14. In IIR Filter design by the Bilinear Transformation, the Bilinear Transformation is a
mapping from
a) Z-plane to S-plane
b) S-plane to Z-plane
c) S-plane to J-plane
d) J-plane to Z-plane
Answer: b
Explanation: From the equation,
S=2 /T [(1−z-1 / 1+z−1)] it is clear that transformation occurs from s-plane to z-plane
15. In Bilinear Transformation, aliasing of frequency components is been avoided.

a) True
b) False
Answer:a
Explanation: The bilinear transformation is a conformal mapping that transforms the jΩ-
axis into the unit circle in the z-plane only once, thus avoiding the aliasing.
16. In the Bilinear Transformation mapping, which of the following are correct?
a) All points in the LHP of s are mapped inside the unit circle in the z-plane
b) All points in the RHP of s are mapped outside the unit circle in the z-plane
c) All points in the LHP & RHP of s are mapped inside & outside the unit circle in the
z-plane
Answer:c
Explanation: The bilinear transformation is a conformal mapping that transforms the jΩ-
axis into the unit circle in the z-plane and all the points are linked as mentioned above.
17. Which of the following is a frequency domain specification?

a) 0 ≥ 20 log|H(jΩ)|
b) 20 log|H(jΩ)| ≥ KP
c) 20 log|H(jΩ)| ≤ KS
Answer: d
Explanation: We are required to design a low pass Butterworth filter to meet the

following frequency domain specifications.

KP ≤ 20 log|H(jΩ)| ≤ 0
and 20 log|H(jΩ)| ≤ KS
18. What is the lowest order of the Butterworth filter with a pass band gain K P=-1 dB at
ΩP=4 rad/sec and stop band attenuation greater than or equal to 20dB at ΩS = 8
rad/sec?
a) 4
b) 5
c) 6
d) 3
Answer: b
Explanation: KP=-1 dB, ΩP= 4 rad/sec, KS=-20 dB and ΩS= 8 rad/sec; Find A1 &A2
Upon substituting the values in the order equation, we get N=4.289
Rounding off to the next largest integer, we get N=5
19. What is the cutoff frequency of the Butterworth filter with a pass band gain K P=-1 dB
at ΩP=4 rad/sec and stop band attenuation greater than or equal to 20dB at ΩS=8
rad/sec?
a) 3.5787 rad/sec
b) 1.069 rad/sec
c) 6 rad/sec
d) 4.5787 rad/sec
Answer: d
Explanation: We know that KP=-1 dB, ΩP=4 rad/sec and N=5 Upon substituting the
values in the above equation, we get ΩC=4.5787 rad/sec
2
20. If H(s)=1 / s +s+1 represent the transfer function of a low pass filter with a
pass band of 1 rad/sec, then what is the system function of a low pass filter with a
pass band 10 rad/sec?
a) 100/ s2+10s+100
b) s / 2s2+s+1
c) s/ 2s2+10s+100
Answer: a
Explanation: The low pass-to-low pass transformation is s→s/Ωc
Hence the required low pass filter is Ha(s)=H(s)|s→s/10 =100 / s2+10s+100
21. Which of the following defines a chebyshev polynomial of order N, T N(x)?

a)cos(Ncos-1x) for all x

b)cosh(Ncosh-1x) for all x

c)cos(Ncos-1x), |x|<1
cosh(Ncosh-1x), |x|>1
Answer: c
Explanation: In order to understand the frequency-domain behavior of chebyshev
filters, it is utmost important to define a chebyshev polynomial and then its
properties. A chebyshev polynomial of degree N is defined as
TN(x) = cos(Ncos-1x), |x|≤1
cosh(Ncosh-1x), |x|>1.
22. What is the formula for chebyshev polynomial TN(x) in recursive form?
a) 2TN-1(x) – TN-2(x)
b) 2TN-1(x) + TN-2(x)
c) 2xTN-1(x) + TN-2(x)
d) 2xTN-1(x) – TN-2(x)
Answer: d
Explanation: We know that a chebyshev polynomial of degree N is defined as
From the above formula, it is possible to generate chebyshev polynomial using the
following recursive formula TN(x)= 2xTN-1(x)-TN-2(x), N ≥ 2.
23. Chebyshev polynomials of odd orders are _____________

a) Even functions
b) Odd functions
c) Exponential functions
d) Logarithmic functions
Answer:b
Explanation: Chebyshev polynomials of odd orders are odd functions because they
contain only odd powers of x.
24. What is the value of TN(0) for even degree N?

a) -1
b) +1
c) 0
d) ±1
Answer: d
Explanation: We know that a chebyshev polynomial of degree N is defined as

For x=0, we have TN(0)=cos(Ncos-10)=cos(N.π/2)=±1 for N even
25. If NB and NC are the orders of the Butterworth and Chebyshev filters respectively to
meet the same frequency specifications, then which of the following relation is true?
a) NC=NB
b) NC<NB
c) NC>NB
d) Cannot be determined
Answer:b
Explanation: The equi-ripple property of the chebyshev filter yields a narrower transition
band compared with that obtained when the magnitude response is monotone. As a
consequence of this, the order of a chebyshev filter needed to achieve the given
frequency domain specifications is usually lower than that of a Butterworth filter
26. The chebyshev-I filter is equi-ripple in pass band and monotonic in the stop band.
a) True
b) False
Answer: a
Explanation: There are two types of chebyshev filters. The Chebyshev-I filter is equi-
ripple in the pass band and monotonic in the stop band and the chebyshev-II filter is
quite opposite
27. What is the first backward difference of y(n)?

a) [y(n)+y(n-1)]/T
b) [y(n)+y(n+1)]/T
c) [y(n)-y(n+1)]/T
d) [y(n)-y(n-1)]/T
Answer: d
Explanation: A simple approximation to the first order derivative is given by the first
backward difference. The first backward difference is defined by [y(n)-y(n-1)]/T
28. Which of the following is the correct relation between ‘s’ and ‘z’ in approximation of
derivatives ?ransformation
a) z=1/(1+sT)
b) s=1/(1+zT)
c) z=1/(1-sT)
d) none of the mentioned
Answer:c
Explanation: We know that s=(1-z-1)/T=> z=1/(1-sT).

29. What is the center of the circle represented by the image of jΩ axis of the s-domain
in approximation of derivatives transformation?
a) z=0
b) z=0.5
c) z=1
Answer: b
Explanation: Letting s=σ+jΩ in the equation z=1/(1-sT) and by letting σ=0, we get
|z-0.5|=0.5. Thus the image of the jΩ axis of the s-domain is a circle with centre at z=0.5
in z-domain
30. An analog high pass filter can be mapped to a digital high pass filter.
a) True
b) False
Answer:b
Explanation: An analog high pass filter cannot be mapped to a digital high pass filter
because the poles of the digital filter cannot lie in the correct region, which is the left-half
of the z-plane(z < 0) in this case.
31. Which of the following rule is used in the bilinear transformation?

a) Simpson’s rule
b) Backward difference
c) Forward difference
d) Trapezoidal rule
Answer:d
Explanation: Bilinear transformation uses trapezoidal rule for integrating a continuous
time function.
32. In bilinear transformation, the left-half s-plane is mapped to which of the following in
the z-domain?
a) Entirely outside the unit circle |z|=1
b) Partially outside the unit circle |z|=1
c) Partially inside the unit circle |z|=1
d) Entirely inside the unit circle |z|=1
Answer: d
Explanation: In bilinear transformation, the z to s transformation is given by the
expression z = [1+(T/2)s]/[1-(T/2)s]. Thus unlike the backward difference method, the
left-half s-plane is now mapped entirely inside the unit circle, |z|=1, rather than to a part
of it
33. What is the kind of relationship between Ω and ω in BLT?

a) Many-to-one

b) One-to-many
c) One-to-one
d) Many-to-many
Answer: c
Explanation: The analog frequencies Ω=±∞ are mapped to digital frequencies ω=±π.
The frequency mapping is not aliased; that is, the relationship between Ω and ω is one-
to-one. As a consequence of this, there are no major restrictions on the use of bilinear
transformation
34. Which of the following methods are used to convert analog filter into digital filter?
a) Approximation of Derivatives
b) Bilinear transformation
c) Impulse invariance
Answer: d
Explanation: There are many techniques which are used to convert analog filter into
digital filter of which some of them are Approximation of derivatives, bilinear
transformation, impulse invariance and many other methods
35. If the conversion technique is to be effective, the jΩ axis in the s-plane should map
into the unit circle in the z-plane.
a) True
b) False
Answer:a
Explanation: If the conversion technique is to be effective, the jΩ axis in the s-plane
should map into the unit circle in the z-plane. Thus there will be a direct relationship
between the two frequency variables in the two domains.
36. Physically realizable and stable IIR filters cannot have linear phase.
a) True
b) False
Answer:a
Explanation: If an IIR filter is stable and if it can be physically realizable, then the filter
cannot have linear phase.
37. Which of the following filter transformation is not possible?

a) High pass analog filter to low pass digital filter
b) High pass analog filter to high pass digital filter
c) Low pass analog filter to low pass digital filter

Answer: b
Explanation: We know that only low pass and band pass filters with low resonant
frequencies in the digital can be designed. So, it is not possible to transform a high pass
analog filter into a corresponding high pass digital filter
38. Which of the following mapping is true between s-plane and z-domain?
a) Points in LHP of the s-plane into points inside the circle in z-domain
b) Points in RHP of the s-plane into points outside the circle in z-domain
c) Points on imaginary axis of the s-plane into points onto the circle in z-domain
Answer:d
Explanation: The below diagram explains the given question
39. By impulse invariance method, the IIR filter will have a unit sample response h(n)
that is the sampled version of the analog filter.
a) True
b) False
Answer: a
Explanation: In the impulse invariance method, our objective is to design an IIR filter
having a unit sample response h(n) that is the sampled version of the impulse response
of the analog filter. That is h(n)=h(nT); n=0,1,2…, where T is the sampling interval
40. What is the equation for normalized frequency?

a) F/Fs
b) F.Fs
c) Fs/F
Answer: a
Explanation: In the impulse invariance method, the normalized frequency f is given by
f= F/Fs
41. Aliasing occurs if the sampling rate Fs is more than twice the highest frequency
contained in X(F).
a) True
b) False

Answer:b
Explanation: Aliasing occurs if the sampling rate Fs is less than twice the highest
frequency contained in X(F).
42. Which of the following filters cannot be designed using impulse invariance method?
a) Low pass
b) Band pass
c) Low and band pass
d) High pass
Answer: d
Explanation: It is clear that the impulse invariance method is in -appropriate for
designing high pass filter due to the spectrum aliasing that results from the sampling
process.
43. Which of the following is the correct relation between ω and Ω?

a) Ω=ωT
b) T=Ωω
c) ω=ΩT
Answer: c
Explanation: We know that z=esT, Now substitute s=σ+jΩ and z=r.ejω, that is represent
‘z’ in the polar form. On equating both sides, we get ω=ΩT.
44. In which of the following transformations, poles and zeros of H(s) are mapped
directly into poles and zeros in the z-plane?
a) Impulse invariance
b) Bilinear transformation
c) Approximation of derivatives
d) Matched Z-transform
Answer: d
Explanation: In this method of transforming analog filter into an equivalent digital filter is
to map the poles and zeros of H(s) directly into poles and zeros in the z-plane
45. The poles obtained from matched z-transform are identical to poles obtained from
which of the following transformations?
a) Bilinear transformation
b) Impulse invariance
c) Approximation of derivatives

Answer: b
Explanation: We observe that the poles obtained from the matched z-transform are
identical to the poles obtained with the impulse invariance method.
46. Low pass Butterworth filters are also called as _____________

a) All-zero filter
b) All-pole filter
c) Pole-zero filter
Answer:b
Explanation: Low pass Butterworth filters are also called as all-pole filters because it has
only non-zero poles.
47. Which of the following is true about type-1 chebyshev filter?

a) Equi-ripple behavior in pass band
b) Monotonic characteristic in stop band
c) Equi-ripple behavior in pass band & Monotonic characteristic in stop band
Answer: c
Explanation: Type-1 chebyshev filters are all-pole filters that exhibit equi-ripple behavior
in pass band and a monotonic characteristic in the stop band
48. Which of the following is false about the type-2 chebyshev filters?
a) Monotonic behavior in the pass band
b) Equi-ripple behavior in the stop band
c) Zero behavior
d) Monotonic behavior in the stop band
Answer:d
Explanation: Type-2 chebyshev filters exhibit equi-ripple behavior in stop band and a
monotonic characteristic in the pass band.
49. Bessel filters exhibit a linear phase response over the pass band of the filter.
a) True
b) False
Answer: a
Explanation: An important characteristic of the Bessel filter is the linear phase response
over the pass band of the filter. As a consequence, Bessel filters has a larger transition
bandwidth, but its phase is linear within the pass band
50. What is the Butterworth polynomial of order 3?

a) (s2+s+1)(s-1)
b) (s2-s+1)(s-1)

c) (s2-s+1)(s+1)
d) (s2+s+1)(s+1)
Answer: d
Explanation: Given that the order of the Butterworth low pass filter is 3.
Therefore, for N=3 Butterworth polynomial is given as B3(s)=(s-s0) (s-s1) (s-s2)
=> B3(s)= (s2+s+1)(s+1)
51. Which of the following is a low pass-to-high pass transformation?

a) s → s / Ωc
b) s → Ωc/s
c) s → Ωc.s
Answer:b
Explanation: The low pass-to-high pass transformation is simply achieved by replacing s
by 1/s. If the desired high pass filter has the pass band edge frequency Ω p, then the
transformationis s → Ωc/s.
53) IIR filter design by approximation of derivatives has the limitations

1) Used only for transforming analog high pass filters
2) Used for band pass filters having smaller resonant frequencies
3) Used only for transforming analog low pass filters
4) Used for band pass filters having high resonant frequencies

ANSWER: (c) 2 and 3 are correct
54) The filter that may not be realized by approximation of derivatives

techniques are
1) Band pass filters
2) High pass filters
3) Low pass filters
4) Band reject filters

ANSWER: (b) 2 and 4 are correct

55) In direct form for realization of IIR filters,

1) Denominator coefficients are the multipliers in the feed forward paths
2) Multipliers in the feedback paths are the positives of the denominator
coefficients
3) Numerator coefficients are the multipliers in the feed forward paths
4) Multipliers in the feedback paths are the negatives of the denominator
coefficients

ANSWER:(c) 3 and 4 are correct
56) The cascade realization of IIR systems involves

1) The transfer function broken into product of transfer functions
2) The transfer function divided into addition of transfer functions
3) Factoring the numerator and denominator polynomials
4) Derivatives of the transfer functions

ANSWER:(b) 1 and 3 are correct
N M
59. To implement  ak y(n  k )  bk x(n  k ) using Direct form -I total number of adders,
k 0 k 0
Scalar multipliers and delay elements are
A) M  N , M  N  1, M  N
B) M  N , M  N , M  N
C ) M  N , M  N  1, M  N
D) M  N  1, M  N  1, M  N
Ans: A) M  N , M  N  1, M  N

60. IIR filters are belongs to------------- systems, length of impulse response is-----------
A) Re cursive, Infinite
B) Re cursive, Finite
C ) Non Re cursive, Infinite
D) Non Re cursive, Finite
Ans: A) Re cursive, Infinite
61. Match the following

A) Butterworth 1) Equi ripple in pass band
B)Chebyshev  I 2) Maximally Flat in Passband
C )Chebyshev  II 3) Equi Ripple in passband and stopband
D) Elliptic 4) Equi ripple in stopband
A) A  2, B  1, C  4, D  3
B) A  1, B  2, C  3, D  4
C ) A  2, B  1, C  3, D  4
D) A  4, B  2, C  1, D  3
Ans: A) A  2, B 1, C  4, D  3
62. The poles of Butterworth transfer function lie,

A) Symmetrically on a circle in s-plane
B) Symmetrically on an ellipse in s-plane
C) Anti-symmetrically on a circle in s-plane
D) Anti-symmetrically on an ellipse in s-plane
Ans: A)Symmetrically on a circle in s-plane
63. The poles of Chebyshev transfer function lie,

A) Symmetrically on a circle in s-plane
B) Symmetrically on an ellipse in s-plane
C) Anti-symmetrically on a circle in s-plane
D) Less Symmetrical on an ellipse in s-plane
Ans: D) Less Symmetrical on an ellipse in s-plane

64. Aliasing is a disadvantage in Bilinear Transformation technique

A) True
B) False
C) Meaningless
D) None of the above
Ans: B) False
65. In Bilinear Transform technique and impulse invariant technique the mapping from
analog angular Frequency(Ω) to Digital angular frequency(  ) are
A) One by one, one to two
B) one by two, one to one
C ) Many to one, one to one
D) one to one, Many to one
Ans: D) one to one, Many to one
66. The normalized transfer function of 2rd order lowpass Butterworth filter is
1 1
S  2S  1S  1
A) 2
S  1.414S  1
B) 2
1 1
C) 2
S  S  1S  1 S  1
D) 3
1
Ans: B)

S  1.414S  1
2


EC 8553-DISCRETE TIME SIGNAL PROCESSING Department of ECE UNIT III – FIR FILTER DESIGN

DEPARTMENT OF ECE
SUBJECT: EC 8553- DISCRETE TIME SIGNAL PROCESSING ( UNIT III)
1. The system represented by the following direct form structure is:
a) General second order system

b) Purely recursive system
c) Partial recursive system
d) FIR system
Answer: d
Explanation: The output of the system according to the direct form given is
y(n)= b0x(n)+b1x(n-1)+b2x(n-2) Since the output of the system is purely dependent on
the present and past values of the input, the system is called as FIR system.
2. An FIR system is also called as “recursive system”.

a) True
b) False
Answer: b
Explanation: For a system to be recursive, the output of the system must be
dependent only on the past values of the output. For an FIR system the output of the
system must be depending only on the present and past values of the input. So, FIR
system is not an recursive system.
3. The system described by the equation y(n)=ay(n+1)+b x(n) is a recursive system.

a) True
b) False

Answer: b
Explanation: Since the present output depends on the value of the future output, the
system is not called a Recursive system
4. If {x(n)} is the signal to be analyzed, limiting the duration of the sequence to L

samples, in the interval 0 ≤ n ≤ L-1, is equivalent to multiplying {x(n)} by?
a) Kaiser window
b) Hamming window
c) Hanning window
d) Rectangular window
Answer: d
Explanation: The equation of the rectangular window w(n) is given as
w(n)=1, 0≤ n≤ L-1
=0, otherwise
Thus, we can limit the duration of the signal x(n) to L samples by multiplying it with a
rectangular window of length L
5. The characteristic of windowing the signal called “Leakage” is the power that is
leaked out into the entire frequency range.
a) True
b) False
Answer: a
Explanation: We note that the windowed spectrum is not localized to a single
frequency, but instead it is spread out over the whole frequency range. Thus the
power of the original signal sequence x(n) that was concentrated at a single
frequency has been spread by the window into the entire frequency range. We say
that the power has been leaked out into the entire frequency range and this
phenomenon is called as “Leakage”.
6. Which of the following is the advantage of Hanning window over rectangular

window?
a) More side lobes
b) Less side lobes
c) More width of main lobe
Answer:b
Explanation: The Hanning window has less side lobes and the leakage is less in this
windowing technique.

7. Which of the following is the disadvantage of Hanning window over rectangular

window?
a) More side lobes
b) Less side lobes
c) More width of main lobe
Answer:c
Explanation: In the magnitude response of the signal windowed using Hanning
window, the width of the main lobe is more which is the disadvantage of this
technique over rectangular windowing technique.
8. Which of the following is an method for implementing an FIR system?

a) Direct form
b) Cascade form
c) Lattice structure
Answer: d
Explanation: There are several structures for implementing an FIR system,
beginning with the simplest structure, called the direct form. There are several other
methods like cascade form realization, frequency sampling realization and lattice
realization which are used for implementing and FIR system.
9. How many memory locations are used for storage of the output point of a sequence
of length M in direct form realization?
a) M+1
b) M
c) M-1
Answer: c
Explanation: The direct form realization follows immediately from the non-recursive
difference equation given by y(n)=∑bkx(n−k). We observe that this structure
requires M-1 memory locations for storing the M-1 previous inputs.
10. The direct form realization is often called a transversal or tapped-delay-line filter.
a) True
b) False
Answer:a
Explanation: The structure of the direct form realization, resembles a tapped delay
line or a transversal system.

11. The realization of FIR filter by frequency sampling realization can be viewed as
cascade of how many filters?
a) Two
b) Three
c) Four
Answer: a
Explanation: In frequency sampling realization, the system function H(z) is
characterized by the set of frequency samples {H(k+ α)} instead of {h(n)}. We view
this FIR filter realization as a cascade of two filters. One is an all-zero or a comb filter
and the other consists of parallel bank of single pole filters with resonant
frequencies.
12. Which of the following is the application of lattice filter?

a) Digital speech processing
b) Adaptive filter
c) Electroencephalogram
Answer: d
Explanation: Lattice filters are used extensively in digital signal processing and in the
implementation of adaptive filters
13. Which of the following represents the bandwidth of the filter?

a) ωP+ ωS
b) -ωP+ ωS
c) ωP-ωS
Answer: b
Explanation: If ωP and ωS represents the pass band edge ripple and stop band edge
ripple, then the transition width -ωP+ ωS gives the bandwidth of the filter
14. The lower and upper limits on the convolution sum reflect the causality and finite
duration characteristics of the filter.
a) True
b) False
Answer: a
Explanation: We can express the output sequence as the convolution of the unit
sample response h(n) of the system with the input signal. The lower and upper limits
on the convolution sum reflect the causality and finite duration characteristics of the
filter.

15. Which of the following condition should the unit sample response of a FIR filter
satisfy to have a linear phase?
a) h(M-1-n) n=0,1,2…M-1
b) ±h(M-1-n) n=0,1,2…M-1
c) -h(M-1-n) n=0,1,2…M-1
Answer: b
Explanation: An FIR filter has an linear phase if its unit sample response satisfies the
condition h(n)= ±h(M-1-n) n=0,1,2…M-1
16. What is the value of h(M-1/2) if the unit sample response is anti-symmetric?
a) 0
b) 1
c) -1
Answer: a
Explanation: When h(n)=-h(M-1-n), the unit sample response is anti-symmetric. For
M odd, the center point of the anti-symmetric is n=M-1/2. Consequently, h(M-1/2)=0.
17. The anti-symmetric condition is not used in the design of low pass linear phase FIR
filter.
a) True
b) False
Answer: a
Explanation: We know that if h(n)=-h(M-1-n) and M is odd, we get H(0)=0 and
H(π)=0. Consequently, this is not suitable as either a low pass filter or a high pass
filter and when h(n)=-h(M-1-n) and M is even, we know that H(0)=0. Thus it is not
used in the design of a low pass linear phase FIR filter. Thus the anti-symmetric
condition is not used in the design of low pass linear phase FIR filter
18. Which of the following defines the rectangular window function of length M-1?
a) w(n)=1, n=0,1,2...M-1
=0, else where
b) w(n)=1, n=0,1,2...M-1
=-1, else where
c) w(n)=0, n=0,1,2...M-1
=1, else where
Answer: a
Explanation: We know that the rectangular window of length M-1 is defined as

w(n)=1, n=0,1,2…M-1
=0, else where
19. The multiplication of the window function w(n) with h(n) is equivalent to the
multiplication of H(w) and W(w).
a) True
b) False
Answer:b
Explanation: According to the basic formula of convolution, the multiplication of two
signals w(n) and h(n) in time domain is equivalent to the convolution of their
respective Fourier transforms W(w) and H(w).
20. What is the width of the main lobe of the frequency response of a rectangular
window of length M-1?
a) π/M
b) 2π/M
c) 4π/M
d) 8π/M
Answer:c
Explanation: The width of the main lobe width is measured to the first zero of W(ω))
is 4π/M.
21. The width of each side lobes decreases with an increase in M.

a) True
b) False
Answer:a
Explanation: Since the width of the main lobe is inversely proportional to the value of
M, if the value of M increases then the main lobe becomes narrower. In fact, the
width of each side lobes decreases with an increase in M.
22. The width of each side lobes decreases with an decrease in M.

a) True
b) False
Answer: b
M, if the value of M increases then the main lobe becomes narrower. In fact, the
width of each side lobes decreases with an increase in M
23. What is the approximate transition width of main lobe of a Hamming window?
a) 4π/M
b) 8π/M

c) 12π/M
d) 2π/M
Answer: b
Explanation: The transition width of the main lobe in the case of Hamming window is
equal to 8π/M where M is the length of the window
24. What is the peak side lobe (in dB) for a rectangular window?
a) -13
b) -27
c) -32
d) -58
Answer:a
Explanation: The peak side lobe in the case of rectangular window has a value of
-13dB
25. How does the frequency of oscillations in the pass band of a low pass filter varies
with the value of M?
a) Decrease with increase in M
b) Increase with increase in M
c) Remains constant with increase in M
Answer:b
Explanation: The frequency of oscillations in the pass band of a low pass filter
increases with an increase in the value of M, but they do not diminish in amplitude.
26. The oscillatory behavior near the band edge of the low pass filter is known as Gibbs
phenomenon.
a) True
b) False
Answer:a
Explanation: The multiplication of hd(n) with a rectangular window is identical to
truncating the Fourier series representation of the desired filter characteristic H d(ω).
The truncation of Fourier series is known to introduce ripples in the frequency
response characteristic H(ω) due to the non-uniform convergence of the Fourier
series at a discontinuity. The oscillatory behavior near the band edge of the low pass
filter is known as Gibbs phenomenon
27. What is the approximate transition width of main lobe of a Blackman window?
a) 4π/M
b) 8π/M

c) 12π/M
d) 2π/M
Answer: c
Explanation: The transition width of the main lobe in the case of Blackman window is
equal to 12π/M where M is the length of the window.
28. If the value of M increases then the main lobe in the frequency response of the
rectangular window becomes broader.
a) True
b) False
Answer: b
M, if the value of M increases then the main lobe becomes narrower
29. The large side lobes of W(ω) results in which of the following undesirable effects?
a) Circling effects
b) Broadening effects
c) Ringing effects
Answer: c
Explanation: The larger side lobes of W(ω) results in the undesirable ringing effects
in the FIR filter frequency response H(ω), and also in relatively large side lobes in
H(ω)
30. To reduce side lobes, in which region of the filter the frequency specifications have
to be optimized?
a) Stop band
b) Pass band
c) Transition band
Answer: c
Explanation: To reduce the side lobes, it is desirable to optimize the frequency
specification in the transition band of the filter. This optimization can be
accomplished numerically on a digital computer by means of linear programming
techniques
31. In the frequency sampling method for FIR filter design, we specify the desired
frequency response Hd(ω) at a set of equally spaced frequencies.
a) True
b) False

Answer: a
Explanation: In the frequency sampling method, we specify the frequency response
Hd(ω) at a set of equally spaced frequencies, namely ωk=2πk/M.
32. The major advantage of designing linear phase FIR filter using frequency sampling
method lies in the efficient frequency sampling structure.
a) True
b) False
Answer: a
Explanation: Although the frequency sampling method provides us with another
means for designing linear phase FIR filters, its major advantage lies in the efficient
frequency sampling structure, which is obtained when most of the frequency
samples are zero
33. Which of the following is introduced in the frequency sampling realization of the FIR
filter?
a) Poles are more in number on unit circle
b) Zeros are more in number on the unit circle
c) Poles and zeros at equally spaced points on the unit circle
Answer: c
Explanation: There is a potential problem for frequency sampling realization of the
FIR linear phase filter. The frequency sampling realization of the FIR filter introduces
poles and zeros at equally spaced points on the unit circle
34. Which of the following is the frequency response of an ideal differentiator, H d(ω)?
a) -jω ; -π ≤ ω ≤ π
b) -jω ; 0 ≤ ω ≤ π
c) jω ; 0 ≤ ω ≤ π
d) jω ; -π ≤ ω ≤ π
Answer: d
Explanation: An ideal differentiator is defined as one that has the frequency
response Hd(ω)= jω ; -π ≤ ω ≤ π
35. which of the following unit sample response for ideal differentiator?
a) Symmetric
b) Anti-symmetric
c) Cannot be explained

Answer: b
Explanation: We know that the unit sample response of an ideal differentiator is
given as h(n)=cosπn. So, we can state that the unit sample response of an ideal
differentiator is anti-symmetric because cosπn is also an anti-symmetric function
36. If hd(n) is the unit sample response of an ideal differentiator, then what is the value
of hd(0)?
a) 1
b) -1
c) 0
d) 0.5
Answer: c
Explanation: Since we know that the unit sample response of an ideal differentiator
is anti-symmetric, => hd(0)=0
37. What kind of filter is an ideal Hilbert transformer?

a) Low pass
b) High pass
c) Band pass
d) All pass
Answer:d
Explanation: An ideal Hilbert transformer is a all pass filter.
38. How much phase shift does an Hilbert transformer impart on the input?
a) 45°
b) 90°
c) 135°
d) 180°
Answer:b
Explanation: An ideal Hilbert transformer is a all pass filter that imparts a 90° phase
shift on the signal at its input.
39. Which of the following is the frequency response of the ideal Hilbert transform?
a) -j ;0 ≤ ω ≤ π
j ;-π ≤ ω ≤ 0
b) j ;0 ≤ ω ≤ π
-j ;-π ≤ ω ≤ 0
c)-j;-π≤ω≤π

Answer:a
Explanation: The frequency response of an ideal Hilbert transform is given as
H(ω)=-j;0≤ω≤π
= j ;-π ≤ ω ≤ 0
39. In which of the following fields, Hilbert transformers are frequently used?
a)GenerationofSSBsignals
b)Radarsignalprocessing
c)Speechsignalprocessing
Answer:d
Explanation: Hilbert transforms are frequently used in communication systems and
signal processing, as, for example, in the generation of SSB modulated signals,
radar signal processing and speech signal processing.
40. The unit sample response of Hilbert transform is _______________

a)Zero
b)Symmetric
c)Anti-symmetric
Answer:c
Explanation:The unit sample response of the Hilbert transform
h(n)=-h(-n). Thus the unit sample response of Hilbert transform is anti-symmetric in
nature.
41. Which of the following is true regarding the frequency response of Hilbert transform?
a)Complex
b)Purelyimaginary
c)Purelyreal
d) Zero
Answer:b
Explanation: Our choice of an anti-symmetric unit sample response is consistent with
having a purely imaginary frequency response characteristic
42. Which of the following is the first method proposed for design of FIR filters?
a)Chebyshevapproximation
b)Frequencysamplingmethod
c)Windowingtechnique

Answer:c
Explanation: The design method based on the use of windows to truncate the
impulse response h(n) and obtaining the desired spectral shaping, was the first
method proposed for designing linear phase FIR filters.
43. The lack of precise control of cutoff frequencies is a disadvantage of which of the
followingdesigns?
a)Windowdesign
b)Chebyshevapproximation
c)Frequencysampling
Answer:a
Explanation: The major disadvantage of the window design method is the lack of
precise control of the critical frequencies.
44. In frequency sampling method, transition band is a multiple of which of the following?
a)π/M
b)2π/M
c)π/2M
d) 2πM
Answer:b
Explanation: In the frequency sampling technique, the transition band is a multiple of
2π/M.
45. The frequency sampling design method is attractive when the FIR filter is realized in
the frequency domain by means of the DFT.
a)True
b)False
Answer:a
Explanation: Frequency sampling design method is particularly attractive when the
FIR is realized either in the frequency domain by means of the DFT or in any of the
frequency sampling realizations
46. Which of the following gives the equation for envelope delay?
a) dϴ(ω)/dω
b) ϴ(ω)
c) -dϴ(ω)/dω
d) -ϴ(ω)
Answer:c
Explanation: Instead of dealing with the phase response ϴ(ω), it is more convenient
to deal with the envelope delay as a function of frequency, which is
Tg(ω)= -dϴ(ω)/dω.

47) FIR filters ________

A. are non-recursive
B. do not adopt any feedback
C. are recursive
D. use feedback
a. A & B
b. C & D
c. A & D
d. B & C
ANSWER:(a) A & B

EC 8553- DISCRETE TIME SIGNAL PROCESSING Department of ECE UNIT IV – Finite Word Length Effects

DEPARTMENT OF ECE
SUBJECT: EC 8553- DISCRETE TIME SIGNAL PROCESSING ( UNIT IV)
1. Finite word length effects refer to the quantization effects that are inherent in any
digital implementation of the system, either in hardware or software.
a) True
b) False
Answer: a
Explanation: The parameters of the system must necessarily be represented with
finite precision. The computations that are performed in the process of computing an
output from the system must be rounded off or truncated to fit within the limited
precision constraints of the computer or hardware used in the implementation. Thus,
Finite word length effects refer to the quantization effects that are inherent in any
digital implementation of the system, either in hardware or software
2. Which of the following are called as finite word length effects?

a) Parameters of the system must be represented with finite precision
b) Computations are truncated to fit in the limited precision constraints
c) Whether the computations are performed in fixed point or floating point arithmetic
Answer: d
Explanation: All the three of the considerations given above are called as finite word
length effects
3. If (101.01)2=(x)10, then what is the value of x?

a) 505.05
b) 10.101
c) 101.01
d) 5.25
Answer: d
Explanation: (101.01)2=1*22+0*21+1*20+0*2-1+1*2-2=(5.25)10 =>x=5.25.
4. If the two numbers are to be multiplied, the mantissa are multiplied and the
exponents are added.
a) True
b) False
Answer: a
Explanation: Let us consider two numbers X=M.2E and Y=N.2F. If we multiply both X

and Y, we get X.Y=(M.N).2E+F . Thus if we multiply two numbers, the mantissa are
multiplied and the exponents are added
5. For a twos complement representation, the truncation error is ____________

a) Always positive
b) Always negative
c) Zero
Answer: b
Explanation: For a two’s complement representation, the truncation error is always
negative and falls in the range -(2-b-2-bm) ≤ Et ≤ 0
6. What is the binary equivalent of (-3/8)?

a) (10011) 2
b) (0011) 2
c) (1100) 2
d) (1101) 2
Answer: d
Explanation: The number (-3/8) is stored in the computer as the 2’s complement of
(3/8). We know that the binary equivalent of (3/8)=0011 Thus the twos complement
of 0011=1101
7. Which of the following is the correct representation of a floating point number X?

a) 2E
b) M.2E(1/2<M<1)
c) 2M.2E(1/2<M<1)
Answer: b
Explanation: The binary floating point representation commonly used in practice,
consists of a mantissa M, which is the fractional part of the number and falls in the
range 1/2<M<1, multiplied by the exponential factor 2E, where the exponent E is
either a negative or positive integer. Hence a number X is represented as X=
M.2E(1/2<M<1)
8. The truncation error for the sign magnitude representation is symmetric about zero.
a) True
b) False
Answer: a
Explanation: The truncation error for the sign magnitude representation is symmetric
about zero and falls in the range -(2-b-2-bm) ≤ Et ≤ (2-b-2-bm).

9. What is the 2’s complement of (1100)2?

a) (0100)2
b) (0011)2
c) (0111)2
d) (1100)2
Answer:a
Explanation: The ones complement of (1100)2 is (0011)2. Thus the two complement
of this number is obtained as (0011)2+(0001)2=(0100)2.
10. If the input analog signal is within the range of the quantizer, the quantization error
eq (n) is bounded in magnitude i.e., |eq (n)| < Δ/2 and the resulting error is called?
a) Granular noise
b) Overload noise
c) Particulate noise
d) Heavy noise
Answer:a
Explanation: In the statistical approach, we assume that the quantization error is
random in nature. We model this error as noise that is added to the original
(unquantized) signal. If the input analog signal is within the range of the quantizer,
the quantization error eq (n) is bounded in magnitude
i.e., |eq (n)| < Δ/2 and the resulting error is called Granular noise.
11. If the input analog signal falls outside the range of the quantizer (clipping), e q (n)
becomes unbounded and results in _____________
a) Granular noise
b) Overload noise
c) Particulate noise
d) Heavy noise
Answer: b
Explanation: In the statistical approach, we assume that the quantization error is
random in nature. We model this error as noise that is added to the original
(unquantized) signal. If the input analog signal falls outside the range of the
quantizer (clipping), eq (n) becomes unbounded and results in overload noise
12. If ak is the filter coefficient and āk represents the quantized coefficient with Δa k as the
quantization error, then which of the following equation is true?
a) āk = ak.Δak
b) āk = ak/Δak
c) āk = ak + Δak

Answer: c
Explanation: The quantized coefficient āk can be related to the un-quantized
coefficient ak by the relation
āk = ak + Δak , Where Δak represents the quantization error
13. If pk is the set of poles of H(z), then what is Δpk that is the error resulting from the
quantization of filter coefficients?
a) Pre-turbation
b) Perturbation
c) Turbation
Answer: b
14. In recursive systems, which of the following is caused because of the nonlinearities
due to the finite-precision arithmetic operations?
a) Periodic oscillations in the input
b) Non-Periodic oscillations in the input
c) Non-Periodic oscillations in the output
d) Periodic oscillations in the output
Answer: d
Explanation: In the recursive systems, the nonlinearities due to the finite-precision
arithmetic operations often cause periodic oscillations to occur in the output even
when the input sequence is zero or some non zero constant value
15. The oscillations in the output of the recursive system are called as ‘limit cycles’.
a) True
b) False
Answer:a
Explanation: In the recursive systems, the nonlinearities due to the finite-precision
arithmetic operations often cause periodic oscillations to occur in the output even
when the input sequence is zero or some non zero constant value. The oscillations
thus produced in the output are known as ‘limit cycles’.
16. Limit cycles in the recursive are directly attributable to which of the following?
a) Round-off errors in multiplication
b) Overflow errors in addition
c) Both of the mentioned
Answer: c
Explanation: The oscillations in the output of the recursive system are called as limit
cycles and are directly attributable to round-off errors in multiplication and overflow
errors in addition

17. What is the range of values called as to which the amplitudes of the output during a
limit cycle are confined to?
a) Stop band
b) Pass band
c) Live band
d) Dead band
Answer:d
Explanation: The amplitudes of the output during a limit circle are confined to a
range of values that is called the ‘dead band’ of the filter.
18. Zero input limit cycles occur from non-zero initial conditions with the input x(n)=0.
a)True
b) False
Answer:a
Explanation: When the input sequence x(n) to the filter becomes zero, the output of
the filter then, after a number of iterations, enters into the limit cycle. The output
remains in the limit cycle until another input of sufficient size is applied that drives
the system out of the limit cycle. Similarly, zero input limit cycles occur from non-zero
initial conditions with the input x(n)=0.
19. What is the dead band of a single pole filter with a pole at 1/2 and represented by 5
bits?
a)(-1/2,1/2)
b)(-1/4,1/4)
c)(-1/8,1/8)
d) (-1/16,1/16)
Answer:d
ie., Dead band is (-1/16,1/16)
20. An effective remedy for curing the problem of overflow oscillations is to modify the
addercharacteristic.
a)True
b) False
Answer:a
Explanation: An effective remedy for curing the problem of overflow oscillations is to
modify the adder characteristic, so that it performs saturation arithmetic. Thus when
an overflow is sensed, the output of the adder will be the full scale value of ±1.

21.
Ans: d
22.
Ans: c
23.
Ans: b
24.
Ans: c
25.
Ans: c

26.
Ans: d
27.
Ans: b
28.
Ans: b
29.
Ans: c

30.
1. Quantization error
2. Saturation error
31.
Negative
32.
Cascade Structure
33.
TRUE/FALSE
- TRUE

EC 8553- DISCRETE TIME SIGNAL PROCESSING Department of ECE UNIT V – Digital Signal Processors

DEPARTMENT OF ECE
SUBJECT: EC 8553- DISCRETE TIME SIGNAL PROCESSING ( UNIT V)
1.
ANS: d) VLIW architecture
2.
ANS: c) Executing different phases of two or more instructions in parallel
3.
ANS: a) 224k-words
4.
ANS: c) 16-bit, 32-bit, 32-bit

5.
ANS: d) It is used by the processor exclusively for address computation
6.
ANS: b) 2
7.
ANS: c) 4,6
8.
ANS: a) Execute the next instruction n-1 times
9.
ANS: c) 192k-words

10.
ANS: a) 16-bit, 40-bit, 40-bit
11.
ANS: b) One pair of program bus and three pair of data buses
12.
ANS: c) Guard bits
13.
ANS: a) to extract the exponent from floating point data

14.
ANS: d) to insert wait states in external bus cycle
15.
ANS: Depth/Level
16.
ANS: MAC
17.
ANS: Page-o
18. TRUE/FALSE
ANS: True
19) Which units are generally involved in Multiply and Accumulate

(MAC)?
a. Adder
b. Multiplier
c. Accumulator
d. All of the above
ANSWER: (d) All of the above

20) In DSP processors, which among the following maintains the track of
addresses of input data as well as the coefficients stored in data and
program memories?
a. Data Address Generators (DAGs)
b. Program sequences
c. Barrel Shifter
d. MAC
ANSWER: (a) Data Address Generators (DAGs)



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EC 8553 – DISCRETE TIME SIGNAL PROCESSING Department of ECE UNIT I – Signals & Systems

St. JOSEPH’S COLLEGE OF ENGINEERING

3. The impulse response of a LTI system is h(n)={1,1,1}. What is the response of

St. Joseph’s College of Engineering Page 1

7. Time scaling operation is also known as ___________

St. Joseph’s College of Engineering Page 2

9. The system described by the input-output equation y(n)=nx(n)+bx3(n) is a

10. Whether the system described by the input-output equations y(n)=x(n)-x(n-1) a

St. Joseph’s College of Engineering Page 3

St. Joseph’s College of Engineering Page 4

15. All energy signals will have an average power of ___________

18. If x(n)=(0,0,1,2,3,4,0,0) then x(n-2) is?

St. Joseph’s College of Engineering Page 5

22. What are the main characteristics of Anti aliasing filter?

St. Joseph’s College of Engineering Page 6

25. What is the Nyquist rate of the signal

St. Joseph’s College of Engineering Page 7

St. JOSEPH’S COLLEGE OF ENGINEERING

SUBJECT: EC 8553- DISCRETE TIME SIGNAL PROCESSING ( UNIT I)

2. Which of the following is true regarding the number of computations requires to

3. What is the DFT of the four point sequence x(n)={0,1,2,3}?

St. JOSEPH’S COLLEGE OF ENGINEERING Page 1

4. If W 4100=W x200, then what is the value of x?

5. If x(n) and X(k) are an N-point DFT pair, then x(n+N)=x(n).

6. If x(n) and X(k) are an N-point DFT pair, then X(k+N)=?

7. What is the circular convolution of the sequences x1(n)={2,1,2,1} and

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St. JOSEPH’S COLLEGE OF ENGINEERING Page 3

a) Overlap save method

St. JOSEPH’S COLLEGE OF ENGINEERING Page 4

a) Overlap save method

St. JOSEPH’S COLLEGE OF ENGINEERING Page 5

18. Divide-and-conquer approach is based on the decomposition of an N-point DFT

St. JOSEPH’S COLLEGE OF ENGINEERING Page 6

corresponding to the even numbered and odd numbered samples of x(n),

21. For a decimation-in-time FFT algorithm, which of the following is true?

St. JOSEPH’S COLLEGE OF ENGINEERING Page 7

23. FFT algorithm is designed to perform complex operations.

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a. 1, 2 and 3 are correct

ANSWER: (c) 1 and 3 are correct

30) Frequency selectivity characteristics of DFT refers to

31)The Cooley–Tukey algorithm of FFT is a

ANSWER: (a) Divide and conquer algorithm

32) FFT may be used to calculate

a. 1, 2 and 3 are correct

ANSWER: (b) 1 and 2 are correct

33) DIT algorithm divides the sequence into

ANSWER: (b) Even and odd samples

St. JOSEPH’S COLLEGE OF ENGINEERING Page 10

N = rm = 2m = 23 ; m- No. of stages of computation, Each stage N/2 butterflies

State whether the following statements are True/False

DFT supports only Circular Convolution

DIT – Bit reversal at I/p

In DFT sequence, real component is even and imaginary component is odd.

38.An LTI is unstable if the impulse response is absolutely summable - False

Condition for Stability - Impulse response should be summable