VISTA ACADEMY

(SALEM & ERODE)

 1- Digital Signal Processing Questions and Answers – A2D and D2A
Converters

1) Answer: a
Explanation: The process of converting a continuous-time signal into a discrete-
time signal by taking samples of continuous time signal at discrete time instants is
known as „sampling‟.

2) Answer: a
Explanation: The process of converting a continuous-time signal into a discrete-
time signal by taking samples of continuous time signal at discrete time instants is
known as „sampling‟.

3) Answer: d
Explanation: Quantization error is the difference in the signal obtained after
sampling i.e., x(n) and the signal obtained after quantization i.e., xq(n) at any
instant of time.

4) Answer: d
Explanation: The process of joining in terms of steps is known as staircase
approximation, connecting two samples by a straight line is known as Linear
interpolation, connecting three samples by fitting a quadratic curve is called as
Quadratic interpolation.

5) Answer: b
Explanation: Consider an analog signal of frequency „F‟, which when sampled
periodically at a rate Fs=1/T samples per second yields a frequency of
f=F/Fs=>f=F*T.

6) Answer: c
Explanation: From the question F=40Hz, Fs=20Hz
=>f=F/Fs
=>f=40/20
=>f=2Hz
=>x(n)=cos(4*pi*n).

7) Answer: a
Explanation: According to Nyquist rate, to avoid aliasing the sampling frequency
should be equal to twice of the analog frequency.
8) 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.

9) Answer: b
Explanation: From the given analog signal, F1=1000Hz F2=2500Hz and
Fs=5000Hz
=>f1=F1/Fs and f2=F2/Fs
=>f1=0.2 and f2=0.5
=>x(n)= cos(0.4*pi*n)+sin(pi*n).

10) Answer: a
Explanation: If it obeys sampling theorem, then the only error in A/D conversion is
quantization error. So, the error is same for both analog and discrete-time signal.

11) Answer: c
Explanation: The quality is measured by taking the ratio of noises of input signal
and the quantized signal i.e., SQNR and is measured in terms of dB.

12) Answer: b
Explanation: To code a signal with L number of levels, we require a coder with
(log L/log 2) number of bits. So, log16/log2=4 bit coder is required.

2-Digital Signal Processing Questions and Answers – Classification of Signals

1) 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.

2) Answer: a
Explanation: The behavior of the signal is known and can be represented by a saw
tooth wave form. So, the signal is deterministic.

3) 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)).

4) Answer: b
Explanation: Let x(t)=e(jt)
Now, xo(t)=(1/2)*(x(t)-x(-t))
=(1/2)*(e(jt) – e(-jt))
=(1/2)*(cost+jsint-cost+jsint)
=(1/2)*(2jsint)
=j*sint.

5) Answer: d
Explanation: If a signal x(t) is said to be periodic with period T, then x(t+mT)=x(t)
for all t and any integer m.

6) Answer: c
Explanation: Let T be the period of the signal x(t)
=>x(t+T)=x1(t+mT1)+x2(t+nT2)
Thus, we must have
mT1=nT2=T
=>(T1/T2)=(k/m)= a rational number.

7) Answer: a
Explanation: For the sum of x1(t) and x2(t) to be periodic the ratio of their periods
should be a rational number, then the fundamental period is the LCM of T1 and T2.

8) Answer: b
Explanation: For any energy signal, the average power should be equal to 0 i.e.,
P=0.

9) Answer: a
Explanation: The energy signal should have total energy value that lies between 0
and infinity.

10) Answer: b
Explanation: Period of cos2t=(2*pi)/2=pi
Period of sin3t=(2*pi)/3
LCM of pi and (2*pi)/3 is 2*pi.
3-Digital Signal Processing Questions and Answers – Signals, Systems and
Signal Processing

1) Answer: a
Explanation: Speech, EEG and ECG signals are the examples of information-
bearing signals that evolve as functions of a single independent variable, namely,
time.

2) 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.

3) Answer: d
Explanation: First the signal x(t) is shifted by 1 to get x(1+t) and it is reflected to
get x(1-t). So, it exhibits both time shifting and reflecting properties.

4) Answer: c
Explanation: Substitute n=0,1,2… in x(2n) and obtain the values from the given
x(n).

5) Answer: d
Explanation: The signal x(n) is shifted right by 2.

6) Answer: a
Explanation: First shift the given signal left by 1 and then time scale the obtained
signal by 3.

7) Answer: b
Explanation: Let the input signal be „t‟. Then the output signal after passing
through the system is y=t2 which is the equation of a parabola. So, the system is
non-linear.

8) 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 D/A converter.

9) Answer: b
Explanation: In the digital processing of the radar signal, the information extracted
from the radar signal, such as the position of the aircraft and its speed, may simply
be printed on a paper. So, there is no need of an D/A converter in this case.
10) Answer: c
Explanation: The multiplicative operation is often encountered in analog
communication, where an audio frequency signal is multiplied by a high frequency
sinusoid known as carrier. The resulting signal is known as “amplitude modulated
wave”.

11) Answer: b
Explanation: A system is a physical device which performs the operation on the
signal and modifies the input signal.

4-Digital Signal Processing Questions and Answers – Z Transform

1) Answer: b
Explanation: The z-transform of a real discrete time sequence x(n) is defined as a
power of „z‟ which is equal to , where „z‟ is a complex
variable.2) Answer: a
Explanation: Since X(z) is a infinite power series, it is defined only at few values
of z. The set of all values of z where X(z) converges to a finite value is called as
Radius of Convergence(ROC).

2) Answer: a
Explanation: Since X(z) is a infinite power series, it is defined only at few values
of z. The set of all values of z where X(z) converges to a finite value is called as
Radius of Convergence(ROC).

3) Answer: d
Explanation: We know that, for a given signal x(n) the z-transform is defined as

Substitute the values of n from -2 to 3 and the corresponding signal values in the
above formula
We get, X(z) = 2z2 + 4z + 5 +7z-1 + z-3.

4 Answer: c
Explanation: We know that, the z-transform of a signal x(n) is
Given x(n)= δ(n-k)=1 at n=k
=> X(z)=z-k
From the above equation, X(z) is defined at all values of z except at z=0 for k>0.
So ROC is defined as Entire z-plane, except at z=0.
5) Answer: a
Explanation: For a given signal x(n), its z-transform

6) Answer: b
Explanation:
Let x(n)= αnu(n)

7) Answer: d
Explanation:

8) Answer: a
Explanation: We know that,
ROC of z-transform of aⁿu(n) is |z|>|a|.
ROC of z-transform of bⁿu(-n-1) is |z|<|b|.
By combining both the ROC's we get the ROC of z-transform of the signal x(n) as
|a|<|z|<|b|.

9) Answer: d
Explanation: Let us an example of anti causal sequence whose z-transform will be
in the form X(z)=1+z+z2 which has a finite value at all values of „z‟ except at
z=∞.So, ROC of an anti-causal sequence is entire z-plane except at z=∞.
10) Answer: c
Explanation: Let us plot the graph of z-transform of any two sided sequence which
looks as follows.

From the above graph, we can state that the ROC of a two sided sequence will be
of the form r2 < |z| < r1.

11) Answer: b
Explanation: The entire timing sequence is divided into two parts n=0 to ∞ and n=-
∞ to 0.
Since the z-transform of the signal given in the questions contains both the parts, it
is called as Bi-lateral z-transform.

12) Answer: c
Explanation: A discrete time LTI is BIBO stable, if and only if its impulse
response h(n) is absolutely summable. That is,

13) Answer: a
Explanation: Since the value of z-transform tends to infinity, the ROC of the z-
transform does not contain poles.
14) Answer: a
Explanation:
Given h(n)= aⁿ(n) (|a|<1)
The z-transform of h(n) is H(z)=z/(z-a),ROC is |z|>|a|
If |a|<1, then the ROC contains the unit circle. So, the system is BIBO stable.

15) Answer: a
Explanation:
Given h(n)= aⁿ(n) (|a|<1)
The z-transform of h(n) is H(z)=z/(z-a),ROC is |z|>|a|
If |a|<1, then the ROC contains the unit circle. So, the system is BIBO stable.

5-Digital Signal Processing Questions and Answers – Frequency Analysis of

Continuous Time Signal

1) Answer: a
Explanation: If the given signal is x(t) and F0 is the reciprocal of the time period of
the signal and ck is the Fourier coefficient then the Fourier series representation of
x(t) is given as

2) Answer: c
Explanation: When we apply integration to the definition of Fourier series
representation, we get

3) Answer: d
Explanation: For any signal x(t) to be represented as Fourier series, it should
satisfy the Dirichlet conditions which are x(t) has a finite number of discontinuities
in any period, x(t) has finite number of maxima and minima during any period and
x(t) is absolutely integrable in any period.

4) Answer: b
Explanation: Since we are synthesizing the Fourier series of the signal x(t), we call
it as synthesis equation, where as the equation giving the definition of Fourier
series coefficients is known as analysis equation.

5) Answer: b
Explanation: In general, Fourier coefficients ck are complex valued. Moreover, it
is easily shown that if the periodic signal is real, ck and c-k are complex
conjugates. As a result
ck=|ck|ejθkand ck=|ck|e-jθk
Consequently, we obtain the Fourier series as

6) Answer: a
Explanation: cos(2πkF0 t+θk)= cos2πkF0 t.cosθk-sin2πkF0 t.sinθk
θk is a constant for a given signal.
So, the other form of Fourier series representation of the signal x(t) is

7) Answer: d
Explanation: The average power of a periodic signal x(t) is given as

By interchanging the positions of integral and summation and by applying the

integration, we get

8) Answer: c
Explanation: When we plot a graph of |ck |2 as a function of frequencies kF0,
k=0,±1,±2… the following spectrum is obtained which is known as Power density
spectrum.

9) Answer: a
Explanation: We know that, Fourier series coefficients are complex valued, so we
can represent ck in the following way.
ck=|ck|ejθk
When we plot |ck| as a function of frequency, the spectrum thus obtained is known
as Magnitude voltage spectrum.
10) Answer: b
Explanation: We know that, for an periodic signal, the Fourier series coefficient is

If we consider a signal x(t) as non-periodic, it is true that x(t)=0 for |t|>Tp/2.

Consequently, the limits on the integral in the above equation can be replaced by -
∞ to ∞. Hence,

11) Answer: c
Explanation: Let us consider a signal x(t) whose Fourier transform X(F) is given as

12) Answer: d
Explanation: Let x(t) be any finite energy signal with Fourier transform X(F). Its
energy is

6-Digital Signal Processing Questions and Answers – Properties of Fourier

Transform for Discrete Time Signals

1) Answer: c
Explanation: We know that
2) Answer: a
Explanation: We know that the inverse transform or the synthesis equation of a
signal x(n) is given as

By substituting ejω = cosω + jsinω in the above equation and separating the real
and imaginary parts we get

3) Answer: b
Explanation: If the signal x(n) is real, then xI(n)=0
We know that,

Now substitute xI(n)=0 in the above equation=>xR(n)=x(n)

4) Answer: d
Explanation: We know that, if x(n) is a real sequence

If we combine the above two equations, we get

X*(ω)=X(-ω)

5) Answer: a
Explanation: We know that if x(n) is a real signal, then xI(n)=0 and xR(n)=x(n)
We know that,

6) Answer: b
Explanation: If x(n) is real and odd then, x(n)cosωn is odd and x(n) sinωn is even.
Consequently
XR(ω)=0
7) Answer: c
Explanation: Given, X(ω)= 1/(1-ae-jω ) ,|a|<1
By multiplying both the numerator and denominator of the above equation by the
complex conjugate of the denominator, we obtain
X(ω)= (1-aejω)/((1-ae(-jω) )(1-aejω)) = (1-acosω-jasinω)/(1-2acosω+a2 )
This expression can be subdivided into real and imaginary parts, thus we obtain
XR(ω)= (1-acosω)/(1-2acosω+a2 ).

8) Answer: d
Explanation: Given, X(ω)= 1/(1-ae-jω ) ,|a|<1
By multiplying both the numerator and denominator of the above equation by the
complex conjugate of the denominator, we obtain
X(ω)= (1-aejω)/((1-ae(-jω) )(1-aejω)) = (1-acosω-jasinω)/(1-2acosω+a2 )
This expression can be subdivided into real and imaginary parts, thus we obtain
XI(ω)= (-asinω)/(1-2acosω+a2 ).

9) Answer: a
Explanation: For the given X(ω)=1/(1-ae-jω ) ,|a|<1 we obtain
XI(ω)= (-asinω)/(1-2acosω+a2 ) and XR(ω)= (1-acosω)/(1-2acosω+a2 )
We know that |X(ω)|=√(〖X_R (ω)〗2+〖X_I (ω)〗2 )
Thus on calculating, we obtain
|X(ω)|= 1/√(1-2acosω+a2 )

10) Answer: c
Explanation: Clearly, x(n)=x(-n). Thus the signal x(n) is real and even signal. So,
we know that

11) Answer: b
Explanation: First we observe x(n) can be expressed as
x(n)=x1(n)+x2(n)
where x1(n)= an, n>0
=0, elsewhere

x2(n)=a-n, n<0 =0, elsewhere Now applying Fourier transform for the above two
signals, we get X1(ω)= 1/(1-aejω)/((1-ae(-jω) )(1-aejω)) = (1-acosω-jasinω)/(1-
2acosω+a2 )
Now, X(ω)= X1(ω)+ X2(ω)= 1/(1-ae^(-jω) )+(ae^jω)/(1-ae^jω ) = (1-a2)/(1-
2acosω+a2).

12) Answer: d
Explanation: Given

13) Answer: a
Explanation: Given x1(n)=x2(n)={1,1,1}
By calculating the Fourier transform of the above two signals, we get
X1(ω)= X2(ω)=1+ ejω + e -jω = 1+2cosω
From the convolution property of Fourier transform we have,
X(ω)= X1(ω). X2(ω)=(1+2cosω)2=3+4cosω+2cos2ω
By applying the inverse Fourier transform of the above signal, we get
x1(n)*x2(n)={1,2,3,2,1}

14) Answer: b
Explanation: Given x(n)= anu(n), |a|<1
The auto correlation of the above signal is
rxx(l)=1/(1-a2 ) a|l|, -∞< l <∞
According to Wiener-Khintchine Theorem,
Sxx(ω)=F{ rxx(l)}= [1/(1-a2)].F{a|l|} = 1/(1-2acosω+a2 )

7-Digital Signal Processing Questions and Answers – Frequency Domain

Characteristics of LTI System

1) Answer: c
Explanation: If x(n)= Aejωn is the input and h(n) is the response o the system, then
we know that

2) Answer: a
Explanation: An Eigen function of a system is an input signal that produces an
output that differs from the input by a constant multiplicative factor known as
Eigen value of the system.
3) Answer: b
Explanation: First we evaluate the Fourier transform of the impulse response of the
system h(n)

4) Answer: d
Explanation: First we evaluate the Fourier transform of the impulse response of the
system h(n)

5) Answer: a
Explanation: From the definition of H(ω), we have

6) Answer: c
Explanation: If h(n) is the real valued impulse response sequence of an LTI
system, then H(ω) can be represented as HR(ω)+j HI(ω).

=>

7) Answer: b
Explanation: For a three point moving average system, we can define the output of
the system as
8) Answer: a
Explanation: The frequency response of the system is

9) Answer: d
Explanation: Given y(n)=ay(n-1)+bx(n)

10) Answer: b
Explanation: We know that,

Since the parameter „a‟ is positive, the denominator of | H(ω)| becomes minimum
at ω=0. So, | H(ω)| attains its maximum value at ω=0. At this frequency we have,
(|b|)/(1-a) =1 =>b=±(1-a).

11) Answer: c
Explanation: From the given difference equation, we obtain

12) Answer: a
Explanation: If x(n) is the input of an LTI system, then we know that the output of
the system y(n) is y(n)= H(ω)x(n) which means the frequency components are just
amplified but no new frequency components are added.
13) Answer: b
Explanation: The frequency response function of the system is

14) Answer: a
Explanation: Given H(z)=1/(1-0.8z-1)=z/(z-0.8)
Clearly, H(z) has a zero at z=0 and a pole at p=0.8. hence the frequency response
of the system is given as
H(ω)= ejω/(ejω-0.8).

8-Digital Signal Processing Questions and Answers – LTI System as

Frequency Selective Filters

1) Answer: b
Explanation: The property of a high pass filter is to pass the signals with high
frequency and stop low frequency signal, which is as shown in the magnitude
frequency response of „b‟.

2) Answer: d
Explanation: In the magnitude response shown in the question, the system is
stopping a particular band of signals. Hence the filter is called as Band stop filter.

3) Answer: a
Explanation: For an ideal filter, in the magnitude response plot at the stop band it
should have a sudden fall which means an ideal filter should have a zero gain at
stop band.
4) Answer: a
Explanation: The time delay taken to reach the output of the system from the input
by a signal component is called as envelope delay or group delay.

5) Answer: b
Explanation: We know that the group delay of the system with phase ϴ(ω) is
defined as
Tg(ω)=(dϴ(ω))/dω
Given the phase is linear=> the group delay of the system is constant.

6) Answer: c
Explanation: Given

Upon solving the above quadratic equation, we get the value of p as 0.32.

7) Answer: d
Explanation: Given

8) Answer: a
Explanation: Clearly, the filter must have poles at P1,2=re±jπ/2 and zeros at z=1 and
z=-1. Consequently the system function is

9) Answer: c
Explanation: The impulse response of a high pass filter is simply obtained from the
impulse response of the low pass filter by changing the signs of the odd numbered
samples in hlp(n). Thus
hhp(n)=(-1)n hlp(n)=(ejπ)n hlp(n)
Thus the frequency response of the high pass filter is obtained as Hlp(ω-π).

10) Answer: b
Explanation: The difference equation for the high pass filter is
y(n)=-0.9y(n-1)+0.1x(n)
and its frequency response is given as
H(ω)= 0.1/(1+0.9e-jω).

11) Answer: a
Explanation: The magnitude response of a band pass filter with two complex poles
located near the unit circle is as shown below.
The filter gas a large magnitude response at the poles and hence it is called as
digital resonator.

12) Answer: d
Explanation: The given figure represents the frequency response characteristic of a
notch filter with nulls at frequencies at ω0 and ω1.

13) Answer: a
Explanation: A comb filter can be viewed as a notch filter in which the nulls occur
periodically across the frequency band, hence the analogy to an ordinary comb that
has periodically spaced teeth.

14 )Answer: c
Explanation: The system with the system function given as H(z)=z -k is a pure
delay system . It has a constant gain for all frequencies and hence called as All pass
filter.

15) Answer: a
Explanation: The given impulse response is h(n)=Asin(n+1)ω0u(n).
According to the above equation, the second order system with complex conjugate
poles on the unit circle is a sinusoid and the system is called a digital sinusoidal
oscillator or a Digital frequency synthesizer.
9-Digital Signal Processing Questions and Answers – Inverse Systems and
Deconvolution

1) Answer: a
Explanation: If we know the output of a system y(n) of a system and if we can
determine the input x(n) of the system uniquely, then the system is said to be
invertible. That is there should be one-to-one correspondence between the input
and output signals.

2) Answer: c
Explanation: . If h(n) is the impulse response of an LTI system T and h1(n) is the
impulse response of the inverse system T-1, then we know that h(n)*h1(n)=δ(n)=>
[h(n)*h1(n)]*x(n)=x(n).

3) Answer: b
Explanation: Given impulse response is h(n)=(1/2)nu(n)
The system function corresponding to h(n) is
H(z)=1/(1-1/2 z-1 ) ROC:|z|>1/2
This system is both stable and causal. Since H(z) is all pole system, its inverse is
FIR and is given by the system function
HI(z)= 1- 1/2 z-1
Hence its impulse response is δ(n)-1/2 δ(n-1).

4) Answer: c
Explanation: The system function of given system is H(z)= 1- 1/2 z-1
The inverse of the system has a system function as H(z)= 1/(1-1/2 z-1 )
Thus it has a zero at origin and a pole at z=1/2.So, two possible cases are |z|>1/2
and |z|<1/2
So, h(n)= (1/2)nu(n) for causal and stable(|z|>1/2)
and h(n)= -(1/2)nu(-n-1) for anti causal and unstable for |z|<1/2.

5) Answer: d
Explanation: Given h(n)= δ(n)-aδ(n-1)
Since h(0)=1, h(1)=-a and h(n)=0 for n≥a, we have
hI(0)=1/h(0)=1.
and
hI(n)=-ahI(n-1) for n≥1
Consequently, hI(1)=a, hI(2)=a2,….hI(n)=an
Which corresponds to a causal IIR system as expected.
6) Answer: a
Explanation: Given H(z)=6+z-1-z-2
By factoring the system function we find the zeros for the system.
The zeros of the given system are at z=-1/2,1/3
So, the system is minimum phase.

7) Answer: b
Explanation: Given H(z)= 1-z-1-z-2
By factoring the system function we find the zeros for the system.
The zeros of the given system are at z=-2,3
So, the system is maximum phase.

8) Answer: c
Explanation: Given H(z)= 1-5/2z-1-3/2z-2
By factoring the system function we find the zeros for the system.
The zeros of the given system are at z=-1/2, 3
So, the system is mixed phase.

9) Answer: a
Explanation: For an IIR filter whose system function is defined as
H(z)=(B(z))/(A(z)) to be said a minimum phase,
then both the poles and zeros of the system should fall inside the unit circle.

10) Answer: d
Explanation: For an IIR filter whose system function is defined as
H(z)=(B(z))/(A(z)) to be said a mixed phase and if the system is stable and causal,
then the poles are inside the unit circle and some, but not all of the zeros are
outside the unit circle.

11) Answer: b
Explanation: The system function is easily determined by taking the z-transforms
of x(n) and y(n). Thus we have
H(z)=(Y(z))/(X(z)) = (1+0.7z-1)/(1-0.7z-1+0.1z-2 ) = (1+0.7z-1)/((1-0.2z-1)(1-0.5z-1))
Upon applying partial fractions and applying the inverse z-transform, we get
[4(0.5)n-3(0.2)n]u(n).
10-Digital Signal Processing Questions and Answers – Frequency Domain
Sampling DFT

1) Answer: a
Explanation: If x(n) is a finite duration sequence of length L, then the Fourier
transform of x(n) is given as

If we sample X(ω) at equally spaced frequencies ω=2πk/N, k=0,1,2…N-1 where

N>L, the resultant

2) Answer: d
Explanation: If X(k) discrete Fourier transform of x(n), then the inverse discrete
Fourier transform of X(k) is given as

3) Answer: b
Explanation: The Fourier transform of this sequence is

If N=L, then X(k)= L for k=0

=0 for k=1,2….L-1

4) Answer: c
Explanation: We know that the Discrete Fourier transform of a signal x(n) is given
as
Thus we get Nth rot of unity WN= e-j2π/N

5) Answer: a
Explanation: The formula for calculating N point DFT is given as

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.

6) Answer: b
Explanation: If XN represents the N point DFT of the sequence xN in the matrix
form, then we know that

7) Answer: c
Explanation: The first step is to determine the matrix W4. By exploiting the
periodicity property of W4 and the symmetry property
WNk+N/2= -WNk
The matrix W4 may be expressed as

8) Answer: a
Explanation: The Fourier series coefficients are given by the expression

9) Answer: d
Answer: Given x(n)={0,1,2,3}
We know that the 4-point DFT of the above given sequence is given by the
expression

In this case N=4

=>X(0)=6,X(1)=-2+2j,X(2)=-2,X(3)=-2-2j.

10) Answer: c
Explanation: We know that according to the periodicity and symmetry property,
100/4=200/x=>x=8.
11-Digital Signal Processing Questions and Answers – Properties of DFT

1) Answer: a
Explanation: We know that the expression for an DFT is given as

Therefore, we got x(n)=x(n+N)

2)Answer:c
Explanation: We know that

Therefore, we have X(k)=X(k+N)

3) Answer: b
Explanation: We know that, the DFT of a signal x(n) is given by the expression

=>X(k)= aX1(k)+bX2(k).

4) Answer: b
Explanation: We know that, the DFT of a signal x(n) is given by the expression

=>X(k)= aX1(k)+bX2(k).

5) Answer: d
Explanation: We know that

Therefore,
X(N-k)=X*(k)=X(-k)

6) Answer: b
Explanation: Given x(n) is real and even, that is x(n)=x(N-n)
We know that XI(k)=0. Hence the DFT reduces to
7) Answer: a
Explanation: If x(n) is real and odd, that is x(n)=-x(N-n), then XR(k)=0. Hence
X(k) is purely imaginary and odd. Since XR(k) reduces to zero, the IDFT reduces
to

8) Answer: c
Explanation: If x1(n),x2(n) and x3(m) are three sequences each of length N whose
DFTs are given as X1(k),X2(k) and X3(k) respectively and X3(k)=X1(k).X2(k),
then according to the multiplication property of DFT we have x3(m) is the circular
convolution of x1(n) and x2(n).

9) Answer: d
Explanation: We know that the circular convolution of two sequences is given by
the expression

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}.

10) Answer: b
Explanation: Given x1(n)={2,1,2,1}=>X1(k)=[6,0,2,0] Given
x2(n)={1,2,3,4}=>X2(k)=[10,-2+j2,-2,-2-j2] when we multiply both DFTs we
obtain the product
X(k)=X1(k).X2(k)=[60,0,-4,0] By applying the IDFT to the above sequence, we
get
x(n)={14,16,14,16}.

11) Answer: a
Explanation: According to the circular time shift property of a sequence, If X(k) is
the N-point DFT of a sequence x(n), then the N-pint DFT of x((n-l))N is X(k)e-
j2πkl/N
.

12) Answer: c

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).
12-Digital Signal Processing Questions and Answers – Efficient Computation
of DFT FFT Algorithms 1

1) Answer: a
Explanation: The formula for calculating N point DFT is given as

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.

2) Answer: d
Explanation: The formula for calculating N point DFT is given as

From the formula given at every step of computing we are performing N complex
multiplications and N-1 complex additions. So, it requires 4N real multiplications
and 4N-2 real additions for any value of „k‟ to compute DFT of the sequence.

3) Answer: b
Explanation: According to the symmetry property, we get WNk+N/2=-WNk.

4) Answer: c
Explanation: For a complex valued sequence x(n) of N points, the DFT may be
expressed as

5) Answer: d
Explanation: The expression for XR(k) is given as

So, from the equation we can tell that the computation of XR(k) requires 2N2
evaluations of trigonometric functions, 4N2 real multiplications and 4N(N-1) real
additions.

6) Answer: a
Explanation: T he 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.
7) Answer: b
Explanation: If we consider the mapping n=Ml+m, then it leads to an arrangement
in which the first row consists of the first M elements of x(n), the second row
consists of the next M elements of x(n), and so on.

8) Answer: a
Explanation: We know that if N=LM, then WNmqL= WN/Lmq= WMmq.

9) Answer: d
Explanation: The expression for N point DFT is given as

The first step involves L DFTs, each of M points. Hence this step requires LM2
complex multiplications, second require LM and finally third requires ML2. So,
Total complex multiplications= N(L+M+1).

10) Answer: b
Explanation: The expression for N point DFT is given as

The first step involves L DFTs, each of M points. Hence this step requires LM(M-
1) complex additions, second step do not require any additions and finally third
step requires ML(L-1) complex additions. So, Total number of complex additions=
N(L+M-2).

11) Answer: c
Explanation: According to one of the algorithm describing the divide and conquer
method, if we store the signal in column wise, then compute the M-point DFT of
each row and multiply the resulting array by the phase factors WNlq and then
compute the L-point DFT of each column and read the result row wise.

12) Answer: a
Explanation: According to the second algorithm of divide and conquer approach, if
the input signal is stored in row wise, then the result must be read column wise.

13) Answer: d
Explanation: According to the second algorithm of divide and conquer approach, if
the input signal is stored in row wise, then we calculate the L point DFT of each
column and we multiply the resulting array by the factor WNpm.