Om Sakthi

Adhiparasakthi Engineering College, Melmaruvathur – 603319

Department of Electronics and Communication Engineering
CS2403 Digital Signal Processing
Two Mark Questions and Answers
1. Define Signals and System.
Signal is defined as a function of a time, position, distance, temperature, etc.,
System is defined as integrated unit to perform task. System is also defined as
processing input signal to generate the output sequence.
2. What are the advantages of DSP? (any 4)
 Less sensitive to changes in component values, temperature, ageing, etcà
stable, reliable, flexible, predictable, repeatable
 Coefficient represented by binary words à choose any accuracy by increasing
or decreasing number of bits
 Sharing of digital processor àreduces processing cost, size, weight, and
maintenance per signal
 Achieve linear phase characteristics
 Multi rate processing is possible
 Digital circuits connected in cascade without any loading problem
 Storage of digital data very easy
 For processing low frequency signal (seismic signal), analog circuits requires
inductor and capacitor of very large size, so we prefer digital processor for such
application
3. What are the disadvantages of DSP?
 Needs pre and post processing (ADC & DAC)
 Suffer from frequency limitation
 For perfect reconstruction, we need to sample signal with sampling rate of
larger than twice the higher frequency component
 But higher sampling frequency affects the resolution of ADC and DAC
 That is, resolution decreases with increase in speed
 Analog circuits don’t need much power where digital circuit needs more power
consumption
4. What are the applications of DSP? (any 2)
Telecommunication: Echo cancellation in telephone networks, Adaptive
equalization, ADPCM, transponders, Telephone dialing application, Modems, Line
repeaters, Channel multiplexing, Data communication, Data encryption, Video
conferencing, Cellular phones, FAX
Military: Radar signal processing, Sonar signal processing, Navigation, Secure
communication, Missile guidance
Consumer Electronics: Digital/ Audio TV, Electronic music synthesizer, Electronic
toys, FM stereo applications, Sound recording applications

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 Instrumentation and Control: Spectrum analysis, Position and rate control, Noise
reduction, Data compression, Digital filter, PLL, Function generator, Servo control,
Robot control, Process control
Seismology: Geo physics exploration for oil and gas, Detection of underground
nuclear exploration, Earthquake monitoring
Image processing: Image representation, Image compression, Image restoration,
Image reconstruction, Image analysis and recognition, Pattern recognition, Robotic
vision, Satellite weather map, Animation
Speech processing: Speech analysis methods à automatic speech recognition,
speaker verification, speaker identification, Speech synthesis methods à conversion
of written text text into speech, digital audio and equalization
Medicine: Computerized tomography (CT), X Ray scanning, Magnetic resonant
imaging, Spectrum analysis of ECG and EEG, Patient monitoring, X Ray storage/
enhancement
Signal filtering: Removal of unwanted background noise, Removal of interference,
Separation of frequency bands, Shaping of the signal spectrum
5. What are the properties of CT sinusoid signal?
 X a (t) is periodic; X a ( t )=X a (t+T p ), T p=1/ F , T p is a fundamental period of
sinusoids
 CT sinusoid signal with distinct frequency themselves differ
 Increase in F result in increase in rate of oscillation of the signal
6. What are the properties of DT sinusoid signal?
 X (n) is periodic only if its frequency is rational number; X (n+ N )=X (n), smallest
value of N is a fundamental period
 DT sinusoids whose frequency separated by an integer multiples of 2π are
identical
 The highest rate of oscillation in a discrete time sinusoids is attained when
ω=π (or ω=−π) or equivalently f =½ (or f =−½)
7. Define sampling.
Sampling is defined as process of converting continuous time to discrete time signal
Sampling performed by taking samples of CT signal at definite interval of time
8. Define sampling time and sampling frequency.
Time interval between successive samples is called sampling time
Inverse of sampling period is called sampling frequency F s.
9. What is the relation between continuous time (analog) and discrete time
(digital) signal?
x (n)=x a ( t ) /¿ t=nT ¿
x (n)=x a (nT )
x (n)=x a ( n/ F s )
10. What is the relation between frequency of analog (CT) and digital (DT)
signal?

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 F
f=
Fs
where,
f is the frequency of digital signal
F is the frequency of analog signal
F s is the sampling frequency
11. Define Aliasing.
Phenomenon of high frequency component getting the identity of low frequency
component during sampling is called Aliasing
12. State Sampling Theorem.
If highest frequency of analog signal is X a ( t ) is F max=B and signal is sampled at
F s >2 F max ≈ 2 B , then X a ( t ) can easily extracted from its sample value using interpolation
function
sin ( 2 π Bt )
g (t)=
2 π Bt
13. Define Deterministic and Random Signal.
Deterministic Signal: Nature and amplitude of signal can be predicted
Random Signal: Nature and amplitude of signal cannot be predicted
14. Define Periodic and Aperiodic Signal.
Periodic Signal: x (n+ N )=x (n) ,−∞< n<∞
Aperiodic Signal: x (n+ N )≠ x (n),−∞< n<∞
15. Define Even and Odd Signal.
Even Signal: x (n)=x (−n)
Odd Signal: x (n)=−x (−n)
16. Define Energy and Power Signal.
Energy signal: finite energy and zero average power
Power signal: infinite energy and finite average power

17. Define Causal and Non Causal Signal.

Causal Signal: Right Sided Sequence, x (n)=0 , n< 0, x ( n ) defined at n ≥ 0
Non Causal Signal: Two Sided Sequence, x ( n ) defined at both n ≥ 0 and n> 0
Anti Causal Signal: Left Sided Sequence, x ( n ) defined at n ≤ 0.
18. Define Energy and Power of the Signal.
Energy of the signal:
N
2
E= lim ∑ |x ( n )|
N → ∞ n=−N

Power of the Signal:

N
1 2
P= lim ∑ |x ( n )|
N → ∞ 2 N +1 n=−N

19. Give the expression for even and odd component of signal.
Even component of signal:

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 1
x even (n)= ( x ( n )+ x (−n ))
2
Odd component of signal:
1
x odd (n)= ( x ( n )−x (−n ) )
2
20. How adder and constant multiplier is represented in block diagram
representation and signal flow graph.
Element Block Diagram Signal flow Graph
Representation
Adder

Constant
Multiplier
21. How unit delay element and unit advance element is represented in block
diagram representation and signal flow graph.

Element Block Diagram Signal flow Graph

Representation
Unit delay
element
Unit advance
element
22. Define Static and Dynamic System.
Static System:
 Output depends on present input not on past or future input
 No memory
Dynamic System:
 Output depends on both present and past input
 Has a memory
23. Define Linear and Non Linear System.
Linear System: Superposition principle holds
H [a∗x (n)+ b∗ y (n)]=a∗H [ x (n)]+b∗H [ y (n)]
Non Linear System: Superposition principle does not holds
24. Define Time Invariant and Time Variant System.
Time Inariant System: Input Output relationship does not vary with time
H [ x (n−k )]= y (n−k )
Time Variant System: Input Output relationship vary with time
25. Define Linear Time Invariant System.
System which satisfy both linearity and time invariant condition called LTI System
H [a∗x (n)+ b∗ y (n)]=a∗H [ x (n)]+b∗H [ y (n)]
H [ x (n−k )]= y (n−k )
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 26. Define Causal and Non Causal System.
Causal System: Output depends only on present and past input and does not
depends on future input
Non Causal System: Output depends on present, past and future input
27. Define Stable and Unstable System.
 System said to be bounded input bounded output (BIBO) stable, if every
bounded input produces bounded output
 Bounded signal has amplitude which remains finite
 BIBO stable system produces bounded output for any bounded input so that it
does not grow unreasonable large
Conditions:
 If system transfer function is a rational fraction, then degree of numerator
must no longer than degree of denominator
 Poles lie in left half plane of S – plane or within unit circle in Z – plane
 No repeated poles lie on the imaginary axis
28. Write the expression for difference equation.
Based on direct solution of input output equation for system
y (n)=F { y ( n−1 ) , y ( n−2 ) , … … … , y ( n−N ) , x ( n ) , x ( n−1 ) , … … … , x ( n−N ) }
Specifically, for an LTI system, input output relationship can be expressed as
M N
y ( n )=∑ bk x ( n−k ) −∑ a k y ( n−k )
k=0 k=1

where {ak } and {bk } are constant parameters that specify the system and are
independent of x (n) and y (n)
This input output relationship is called difference equation
29. Define convolution for two sequence x ( n )and h ( n ) . (or) Define linear
convolution for two sequence x ( n )and h ( n ) . (or) What is the response of the
system with impulse response h ( n ) for a given input x ( n )?
∞
y ( n )= ∑ x ( k ) h ( n−k )
k=−∞

30. State the properties of Convolution.

Commutative Property:
x (n)∗h(n)=h (n)∗x (n)
Associative Property:
[ x(n)∗h(n)]∗z (n)=x (n)∗[h(n)∗z( n)]
Distributive Property:
x (n)∗[h(n)+ z (n)]=[x ( n)∗h(n)]+[x ( n)∗z (n)]
31. What is the condition for stability of an LTI system?
∞

∑ |h ( n )|< ∞
n=−∞

32. Difference between IIR and FIR filter.

S. No. IIR Filter FIR Filter

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 1. Impulse response consists of Impulse response consists of
infinite number of samples finite number of samples
2. Requires a infinite memory Requires a memory of length N
3. Described by a difference Described by a difference
equation, equation,
M N N−1
y ( n )=∑ bk x ( n−k ) −∑ a k y ( n−k ) y ( n )= ∑ b k x ( n−k )
k=0 k=1 k=0
33. Define Z transform and inverse Z transform.
Two Sided Z Transform:
∞
X ( z )=Z [ x ( n ) ]= ∑ x ( n ) z−n
n=−∞

One Sided Z Transform:

∞
X ( z )=Z [ x ( n ) ]=∑ x ( n ) z −n
n=0

Inverse Z transform:
1
x ( n )= ∮ X ( z ) z n−1 dz
2 πj
34. What is procedure to perform linear convolution via circular convolution?
 When two N point sequences are circularly convoluted, it produces another N
point sequences. For circular convolution, one of the sequences should be
periodically extended. Also the resultant sequence is periodic with period N.
 The linear convolution of two sequences x1(n) and x2(n) of length N1 and N2
produces an output sequence of length N1+N2-1. To perform linear convolution
via circular convolution both the sequences to be converted to N 1+N2-1 point
sequences by padding with zeros. Then perform circular convolution of N 1+N2-1
point sequences. The resultant sequence will be same as that of linear
convolution of N1 and N2 point sequences.
35. State Linearity property of Z transform.
If Z [ x ( n ) ] =X ( z ) and Z [ y ( n ) ] =Y ( z ), then
Z { ax ( n ) +by ( n ) } =aZ [ x ( n ) ] +bZ [ y ( n ) ] =aX ( z ) + bY ( z )
36. State Convolution property of Z transform.
If Z [ x ( n ) ] =X ( z ) and Z [ y ( n ) ] =Y ( z ), then Z [ x ( n )∗y ( n ) ]= X ( z ) Y ( z )
where,
∞
x ( n )∗y ( n )= ∑ x ( m ) y ( n−m)
m=−∞

37. State initial value and final value theorem of Z transform.

Initial Value Theorem:
If Z [ x ( n ) ] =X ( z ), then
x ( 0 )=lim X ( z )
z →∞

Final Value Theorem:

If Z [ x ( n ) ] =X ( z ), then

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 x ( ∞ )=lim ( 1−z−1 ) X ( z )
z→1
38. State Shifting property of Z transform.
If Z [ x ( n ) ] =X ( z ), then
Z [ x ( n−m ) ] =z−m X ( z )
Z [ x ( n+ m ) ]=z m X ( z )
39. Define ROC.
ROC (Region of Convergence) of X ( z ) is the set of all values of z, for which X ( z ) attains
a finite value.
40. Define auto correlation and cross correlation.
Auto Correlation:
∞
r xx ( m )= ∑ x ( n ) x ( n−m )
n=−∞

Cross Correlation:
∞
r xy ( m) = ∑ x ( n ) y ( n−m )
n=−∞

41. What are the properties of ROC? (All must write)

 ROC of X ( z ) is a ring or disk in Z plane, with center at origin.
 If x ( n ) is finite duration right sided (causal) signal, then the ROC is entire Z
plane except z=0.
 If x ( n ) is finite duration left sided (anti causal) signal, then the ROC is entire Z
plane except z=∞.
 If x ( n ) is finite duration two sided (non causal) signal, then the ROC is entire Z
plane except z=0and z=∞.
 If x ( n ) is infinite duration right sided (causal) signal, then ROC is exterior of the
circle of radius r 1.
 If x ( n ) is infinite duration left sided (anti causal) signal, then ROC is interior of
the circle of radius r 2.
 If x ( n ) is infinite duration both sided (non causal) signal, then ROC is the
region in between two circles of radius r 1and r 2.
 If X ( z ) is rational, then ROC does not include any poles of X ( z ).
 If X ( z ) is rational, and if x ( n ) is right sided (causal), then ROC is exterior of the
circle whose radius corresponds to the pole with largest magnitude.
 If X ( z ) is rational, and if x ( n ) is left sided (anti causal), then ROC is interior of
the circle whose radius corresponds to the pole with smallest magnitude.
 If X ( z ) is rational, and if x ( n ) is two sided (non causal), then ROC is region in
between two circles whose radius corresponds to the pole of causal part with
largest magnitude and pole of anti causal with smallest magnitude.
42. What are the properties of Auto Correlation?

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  The auto correlation r xx ( m ) is simply a folded version of r xx ( m ) , i.e., r xx ( m )=r xx (−m ).
 The auto correlation sequence satisfies the condition |r xx ( m)|≤ r xx ( 0 ) =E x, where E x
is the energy of x ( n ).
 Using the maximum value of auto correlation sequence, the normalized auto
correlation sequence is defined as
r (m)
ρ xx ( m ) ≤ xx
r xx ( 0 )
43. What are the properties of Cross Correlation?
 The cross correlation r xy ( m) is simply a folded version of r yx ( m ), i.e., r xy ( m) =r yx (−m )
.
 The cross correlation sequence satisfies the condition
|r xy ( m )|≤ √ r xx ( 0 ) r yy ( 0 )= √ E x E y, where E x and E y are energy of x ( n ) and y ( n )
respectively.
 Using the maximum value of cross correlation sequence, the normalized cross
correlation sequence is defined as
r xy ( m )
ρ xy ( m) ≤
√ r xx ( 0 ) r yy ( 0 )
44. Define Circular correlation.
The circular correlation between two sequence x ( n ) and y ( n ) is defined as
N −1
ŕ xy ( m) = ∑ x ( n ) y ¿ ( ( n−m ) )N
n=0
N −1
ŕ xx ( m )= ∑ x ( n ) x¿ ( ( n−m ) ) N
n=0
Unit – II Discrete Fourier Transform
1. What is meant by Twiddle Factor?
W N =e− j 2 π / N
2. How many number of complex multiplication and complex additions are
required to perform N point DFT?
Number of Complex Multiplication = N 2
Number of Complex Addition = N ( N−1 )
3. How many number of complex multiplication and complex additions are
required to perform N point FFT?
N
Number of Complex Multiplication = log 2 N
2
Number of Complex Addition = N log 2 N
4. What are the applications of DFT?
Spectrum Analysis
Correlation Analysis
Linear Filtering
5. What are the applications of FFT?
Spectrum Analysis
Correlation Analysis
Linear Filtering
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 6. Why FFT is needed?
Less Computational Complexity
Less time consumption
Less number of storage due to in place computation
7. What is meant by in place computation?
The outputs performed at each stage are stored in the same memory location where
the input of the stage is stored. This computation is called in place computation.
8. What do you meant by bit reversal in FFT?
When the input to the DIT or DIF algorithm is given in the bit reversed order, then
the output will get as in normal order. When the input to the DIT or DIF algorithm is
given in the normal order, then the output will get as in bit reversed order. The input
to the DIT algorithm to perform 8 point DFT is given in the bit reversed order as
x ( 0 ) , x ( 4 ) , x ( 2 ) , x ( 6 ) , x ( 1 ) , x ( 5 ) , x ( 3 ) , x ( 7 ).
9. State the analysis and synthesis equation for DFT?
N −1
X ( k )= ∑ x ( n ) e− j2 πnk / N , k=0 ,1 , 2 ,… … … , N −1
n=0
N −1
1
x ( n )= ∑ X ( k ) e j 2 πnk / N ,n=0 , 1 ,2 , … … … , N −1
N k =0
10. What is the relation between DTFT and DFT?
X ( k )=[ X ( e jω ) ]ω → 2 πk/ N
11. State any two properties of DFT.
Periodicity property
If X ( k ) is an N point DFT of x ( n ), then
x ( n+ N )=x ( n ) foralln
X ( k + N )=X ( k ) forallk
Linearity property
If X ( k ) and Y ( k ) are the N point DFT of x ( n ) and y ( n ) respectively, and a and b are
arbitrary constants either real or complex valued, then
DFT [ ax ( n )+ by ( n ) ] =aX ( k ) +bY ( k )
Convolution property
If X ( k ) and Y ( k ) are the N point DFT of x ( n ) and y ( n ) respectively, then
DFT [ x ( n )∗y ( n ) ]=X ( k ) Y ( k )
Time reversal property
If X ( k ) is the N point DFT of the sequence x ( n ), then
DFT [ x ( N−n ) ] =X ( N−k )
Circular time shift property
If X ( k ) is the DFT of the sequence x ( n ), then
DFT ¿
Circular frequency shift property
If X ( k ) is the DFT of the sequence x ( n ), then
DFT [ x ( n ) e j 2 πln/ N ]=X ( k −l ,modN )
Complex conjugate property
If DFT [ x ( n ) ] = X ( k ), then

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 DFT [ x ¿ ( n ) ]=X ¿ (−k ,modN )= X ¿ ( N−k )
N −1 N −1 ¿
1 1
¿
IDFT [ X ( k ) ] =
N
∑X
k=0
¿
(k ) e j 2 πkn / N
=
[N
∑ X (k ) e
k=0
j 2 πk ( N −n) / N
]
Circular convolution property
If DFT [ x 1 ( n ) ]= X 1 ( k )and DFT [ x 2 ( n ) ]= X 2 ( k ), then
DFT [ x ( n ) ] =DFT [ x 1 ( n ) ⨂ x 2 ( n ) ]=X 1 ( k ) X 2 ( k )
Circular correlation property
If DFT [ x 1 ( n ) ]= X 1 ( k )and DFT [ x 2 ( n ) ]= X 2 ( k ), then
1
DFT [ r x 12 ( l ) ]= R x 12 ( k ) =X 1 ( k ) X ¿2 ( k )
N
where,
N −1
r x12 ( l )= ∑ x1 ( n ) x ¿2 ( n−l, modN )
n=0

Multiplication of two sequences property

If DFT [ x 1 ( n ) ]= X 1 ( k )and DFT [ x 2 ( n ) ]= X 2 ( k ), then
1
DFT [ x ( n ) ] =DFT [ x 1 ( n ) x 2 ( n ) ]= [ X 1 ( k ) ⨂ X 2 ( k ) ]
N

Parseval’s property
For complex valued sequence x (n)and y ( n ), if DFT [ x ( n ) ] = X ( k ) and DFT [ y ( k ) ] =Y ( k ),
N−1 N −1

∑ x ( n ) y ( n )= N1 ∑ X ( k ) Y ¿ ( k )
¿

n =0 k=0

If x ( n )= y ( n ), then
N−1 N−1

∑ |x ( n )| = N1 ∑ |X ( k )|
2 2

n =0 k=0
12. Compare DIT and DIF Algorithm
S. No. DIT Algorithm DIF Algorithm
Frequency Domain Sequence is
Time Domain Sequence is decimated
1 decimated into odd and even
into odd and even numbered values
numbered values
For N point DFT, DIT algorithm For N point DFT, DIT algorithm
2 performs from 2 point DFT to N point performs from N point DFT to 2
DFT. point DFT.
3 It is suitable for performing IDFT It is suitable for performing DFT
13. Compare DFT and FFT
S. No. DFT FFT
Number of Complex Multiplication
Number of Complex Multiplication =
1 N
N2 = log 2 N
2
Number of Complex Addition =
2 Number of Complex Addition = N ( N−1 ) N log 2 N
14. Compare Overlap add and Overlap save method
S. No. Overlap Add Method Overlap Save Method
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 Linear convolution is performed using Circular convolution is performed
1
circular convolution
2 No corruption due to aliasing Corruption due to aliasing
15. Define sectioned convolution.
If the data sequence x (n) is of long duration, it is very difficult to obtain the output
sequence y (n) due to limited memory of a digital computer. Therefore, the data
sequence is divided into smaller sections. These sections are processed separately
one at a time and combined later to get the output.

16. What is the relation between DFT and Z transform?

X ( k )=[ X ( z ) ]z=e
j2 πk/N

17. List the uses of FFT in linear filtering.

Overlap Add Method: Linear convolution is performed using circular convolution.
Overlap Save Method: Circular convolution is performed.
18. Draw the basic butterfly diagram for DIT and DIF algorithm.
DIT Algorithm

DIF Algorithm

Unit – III IIR Filter

1. When the analog system is to be stable?
The analog system is to be stable if all the poles of the analog system lies on the left
half plane of the s plane
2. When the digital system is said to be stable?
The digital system is said to be stable if all the poles of the digital system lies inside
the unit circle of the z plane
3. What are the conditions of transforming the analog systems to digital systems?
The poles on the left half of the s plane should map into the poles lies inside the
unit circle of the z plane
The poles on the jΩ axis should map into the poles lies on the boundary of the unit
circle of the z plane
The poles on the right half of the s plane should map into the poles lies outside the
unit circle of the z plane
4. What are the conditions of transforming the stable analog systems to stable
digital systems?

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 The poles on the left half of the s plane should map into the poles lies inside the
unit circle of the z plane
The poles on the jΩ axis should map into the poles lies on the boundary of the unit
circle of the z plane
5. What is the disadvantage of Impulse invariant technique?
It is only applicable to design a filter to allow the low frequencies, i.e., Low Pass Filter
and limited case of Band Pass Filter. It is not applicable to design a high pass filter
and band reject filter.
6. What is the advantage of Bilinear Transformation?
It is applicable to design any filter such as Low Pass Filter, High Pass Filter, Band
Pass Filter, and Band Reject Filter.
7. What is the disadvantage of Bilinear Transformation?
The relationship between analog and digital frequency is non-linear for higher
frequencies. Due to the non-linear relationship between analog and digital frequency,
frequency compression or frequency warping may occur.
8. Define IIR Filter.
If the impulse response sequence of the filter has infinite number of non-zero term,
then the filter is called IIR filter. Example: h ( n ) =n , 0 ≤n ≤ ∞
9. What are the Properties of Butterworth Filter?
The response of the Butterworth filter is monotonic both in pass band and stop band
The response of the Butterworth filter goes to ideal response by increasing the order
of the filter
10. What are the Properties of Chebyshev Type 1 Filter?
The response of the Chebyshev Type 1 filter is monotonic in pass band and have
ripples in stop band
The response of the Chebyshev Type 1 filter goes to ideal response by increasing the
order of the filter
11. What are the Properties of Chebyshev Type 2 Filter?
The response of the Chebyshev Type 2 filter is monotonic in stop band and have
ripples in pass band
The response of the Chebyshev Type 2 filter goes to ideal response by increasing the
order of the filter
12. What are the different design techniques available for IIR filter design?
Approximation of derivative
Impulse invariant transformation
Bilinear transformation
13. List out the mapping formulas for the impulse invariant technique.
1 1
→
s−a 1−eaT z−1
1 (−1 )m−1 d m−1 1
( s+ a )
m
→
( m−1 ) ! d s[m−1
]
−sT −1
1−e z s →a
s+ a 1−e−aT cos bT z−1
→
( s+ a )2 +b2 1−2 e−aT cos bT z−1 +e−2 aT z−2
b e−aT sin bT z−1
→
( s+ a )2 +b2 1−2 e−aT cos bT z−1 +e−2 aT z−2
14. List the properties of Chebyshev polynomial.
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 The Nth order chebyshev polynomial has a following properties:
|T N ( x )|≤1for all |x|≤1
|T N ( 1 )|=1for all N
All the roots of the polynomial T N ( x ) occur in the interval −1 ≤ x ≤1
15. What is Butterworth approximation?
In Butterworth approximation, the error function is selected such that the
magnitude is maximally flat in the origin (i.e., at Ω=0) and monotonically decreasing
with increasing Ω.
16. Define Bilinear Transformation with its expression.
The bilinear transformation is a conformal mappingthat transforms the jΩ-axis into
the unit circle in the z-plane only once, thusavoiding aliasing of frequency
components.The mapping from the s-plane to the z-plane is
2 1−z−1
s→
T 1+ z −1
This is called the bilinear transformation.
17. What are the disadvantages of digital filter?
Speed limitation
Finite word length effects
Long design and development times
18. Define filter. What is the objective of filter?
A filter is essentially a network that selectively changes the wave shapes of a signal
in a desired manner.
The objective of filtering is to improve the quality of a signal or to extract information
from signals.
19. What are the advantages of digital filters?
Linear phase response
Performance does not vary with environmental changes
Frequency response can be adjusted
Several input signals can be filtered by one digital filter
Digital filters can be used at very low frequencies.
Unit IV FIR Filter
1. Define FIR Filter
The FIR filter has a finite number of non – zero terms, i.e., its impulse response
sequence of finite sequence.
FIR filters are usually implemented using structures with no feedback.
Example:
h ( n ) = 2 ,|n|≤ 4
{ 0 ,ot h erwise
2. What are the advantage of FIR filter over IIR filter
They can have an exact linear phase.
They are always stable.

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 The design methods are generally linear.
They can be realized efficiently in hardware.
The filter start up transients has finite duration.
3. Define Phase Delay and Group Delay.
The phase and group delays of the filter are given by
−∅ ( ω )
τ p=
ω
−d ∅ ( ω )
τ g=
dω
The group delay is defined as the delayed response of the filter as a function of ω to a
signal.
4. Define Linear Phase FIR filter.
Linear phase filters are filters in which phase delay and group delay are constants,
i.e., independent of frequency.
Linear phase filters also called constant time delay filters.
5. What is the condition for FIR filter to be linear phase?
Symmetric:h ( n ) =h ( M −1−n )
Antisymmetric:h ( n ) =−h ( M −1−n )
6. State the effects of windowing.
A major effect of windowing is that the discontinuities in H ( e j ω ) are converted into
transition bands between values on either side of the discontinuity. The width of
these transition bands depends on the width of the main lobe of W ( e j ω ).
The secondary effect of windowing is that the ripples from the side lobes W ( e j ω )
produces approximation errors for all ω.
7. What are the characteristics of window function?
Small width of main lobe containing as much of the total energy
In side lobes energy is decreasing rapidly as ω tends to π
8. Write down the window function of hamming window.
Causal Hamming Window Function:
2π n
{
w H ( n )=
0.54−0.46 cos
M −1
,0 ≤ n< M −1
0 , otherwise
Non Causal Hamming Window Function:
2π n M −1
{
w H ( n )=
0.54 +0.46 cos
M −1
,|n|≤
0 , otherwise
2

9. Write down the window function of hanning window.

Causal Hanning Window Function:
2π n
{
w Hann ( n )=
0.5−0.5cos
M −1
, 0≤ n< M −1
0 ,otherwise
Non Causal Hanning Window Function:
CS2403 Digital Signal Processing: Two MarksPage 14
 2π n M −1
w H ( n )=
{
0.5+ 0.5cos
M −1
,|n|≤
0 , otherwise
2

10. Write down the window function of Blackmann window.

Causal Blackmann Window Function:
2πn 4πn
wB (n)=
{
0.42−0.5 cos
M −1
+0.08 cos
0 , otherwise
M −1
,0 ≤ n< M −1

Non Causal Blackmann Window Function:

2π n 4πn M −1
wB (n)=
{
0.42+0.5 cos
M −1
+0.08 cos
0 , otherwise
M −1
,|n|≤
2

11. Compare the frequency domain characteristics of hamming, hanning and

blackmann window.
Approximate Minimum Stop
Peak of first side
Type of Window Transition Width of band attenuation
lobe (dB)
Main Lobe (dB)
Rectangular 4 π /M −22 −13
Hanning 8 π/M −44 −31
Hamming 8 π/M −51 −41
Blackmann 12 π /M −78 −58
12. State Gibb’s Phenomenon.
The phenomenon of abrupt truncation in Fourier Series which causes oscillations in
pass band and stop band is called Gibb’s Phenomenon.
13. Define Fixed point representation.
It is a generalization of the familiar decimal representation of a number as a string of
digits with a decimal points. In this notation, the digits to the left of the decimal
point represent the integer part of the number and the digit to the right of the
decimal point represents the fractional part of the number.
14. Define floating point representation.
A floating point representation can be employed as a means for covering a large
dynamic range. 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
1
range ≤ M <1, multiplied by the exponential factor 2 E, where the exponent E is either
2
a positive or negative integer. Hence a number X is represented by X =M . 2 E
15. What are the advantages of floating point representation?
It is used to represent larger range of numbers and it gives accurate value.
16. What is truncation?
Truncation is definedasthe removal of excessive bits. This leads to the reduction in
the magnitude of the number.
17. What is rounding?

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 Rounding is defined as changing a fractional value to the nearest integer. This leads
to the decrease or increase in the magnitude of the number.
18. What is the range of truncation error for sign magnitude representation
in infinite and finite precision?
Infinite precision: −2−B ≤ ε T ≤2−B
Finite precision: −( 2−B−2−L ) ≤ ε T ≤ ( 2−B−2−L )
where B is the number of bits reduced after quantization and L is the length of the
original number
19. What is the range of truncation error for one’s complement
representation in infinite and finite precision?
Infinite precision: 0 ≤ ε T ≤ 2−B
Finite precision: 0 ≤ ε T ≤ ( 2− B−2−L )
where B is the number of bits reduced after quantization and L is the length of the
original number
20. What is the range of truncation error for two’s complement
representation in infinite and finite precision?
Infinite precision: −2−B ≤ ε T ≤0
Finite precision: −( 2−B−2−L ) ≤ ε T ≤0
where B is the number of bits reduced after quantization and L is the length of the
original number
21. Define quantization error.
The difference between the quantized signal amplitude x Q ( n ) and the actual signal
amplitude x ( n ) is called quantization error e ( n ). That is e ( n )=x Q ( n )−x ( n )
22. Define dynamic range of the quantizer.
DR=−10 log Pe (n )=−20 log R+6 B +10.8
23. What is the output noise power from the digital system H ( z ) ?
2 σ 2e
σe =
0
2π j
∮ H ( z ) H ( z−1 ) z−1 dz
where the closed contour of integration is around the unit circle |z|=1.
24. Define product quantization.
The error due to the quantization of the output of multiplier is referred to as product
quantization error.
25. Define noise transfer function.
The noise transfer function (NTF) is defined as transfer function from the noise
source to the filter output. NTF is the transfer function obtained by treating the noise
source as actual input.
26. Define limit cycle oscillation.
In recursive systems, when the input is zero or some nonzero constant value, the
nonlinearities due to finite precision arithmetic operators may cause periodic
oscillations in the output.During periodic oscillations, the output y ( n ) of a system will

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 oscillate between a finite positive and negative value for increasing n or the output
will become constant for increasing n.Such oscillations are called limit cycle
oscillation.
27. Define Zero input limit cycle oscillation.
In recursive systems, if a system output enters a limit cycle, it will continue to
remain in limit cycle even when the input is made zero.Hence these limit cycles are
called zero input limit cycle.
28. Define dead band.
In limit cycle, the amplitudes of the output are confined to the range of values, which
is called dead band of the filter.
29. What is the dead band for a first order system y ( n )=ay ( n−1 ) + x ( n )?
2−B
Dead band=±
1−|a|
30. What is the dead band for a second order system
y ( n )=a1 y ( n−1 )+ a2 y ( n−2 ) + x ( n )?
2−B
Dead band=±
1−|a 2|
31. What is overflow limit cycle oscillation?
The oscillation occurs due to the truncation of output of the adder or multiplier is
called overflow limit cycle oscillations.
32. What are the two types of quantization employed in digital systems?
Truncation error, Rounding error
33. What do you understand by input quantization error?
The truncation due to the output of the Analog and Digital Converter is called input
quantization error.
34. State the methods used to prevent overflow.
Saturation arithmetic, Scaling
35. What is meant by saturation arithmetic?
In saturation arithmetic, if the output exceeds the maximum value then the output
is set to maximum value and if the output goes below the minimum value then the
output is set to minimum value.
36. What is the disadvantage of saturation arithmetic?
In saturation arithmetic, if the output exceeds the maximum value then the output
is set to maximum value and if the output goes below the minimum value then the
output is set to minimum value.
37. How the scaling used to prevent overflow?
Scaling can be done by scale the input at certain points in the digital filter to prevent
overflow.
Unit – V DSP Applications
1. What is decimation?
Decimation is the process of reducing sampling rate of a signal.
Also called sampling rate compression
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 2. What is anti aliasing filter?
Anti aliasing filter is a low pass filter used in decimator to avoid the aliasing effect.
H ( e jω )=1 ,|ω|≤ π / M
3. What is interpolation?
Interpolation is the process of increasing sampling rate of a signal.
Also called sampling rate expansion
4. What is anti imaging filter?
Anti imaging filter is a low pass filter used in interpolator to avoid the imaging effect.
H ( e jω )=1 ,|ω|≤ π / L
5. What is multi rate signal processing?
It is required in digital systems where more than one sampling rate is required.
6. What is meant by down sampling and up sampling?
Down sampling or Decimation is the process of reducing sampling rate of a signal.
Up sampling or Interpolation is the process of increasing sampling rate of a signal.
7. What are the applications of multi rate processing?
Sub band coding
Voice privacy using analog phone lines
Signal compression by subsampling
Used in ADC, DAC, etc
8. What are the advantages of multi rate processing?
Less computational requirements
Less storage for filter coefficients
Less finite arithmetic effect
Low filter order required in multi rate applications
Less sensitivity to filter coefficient lengths
9. What is meant by sampling rate conversion?
It is the process of converting the sequence x ( n ) which is got from continuous time
signal x ( t ) with a period T , to another sequence y ( k ) obtained from samplingx ( t ) with a
period T '.
10. What are the applications of Adaptive filter?
Noise cancellation
Line enhancing
Frequency tracking
Channel equalizations
11. Write down the expressions of spectrum of decimator?
1
Y ( e jω )= X ( e j ω / M )
M
12. Write down the expression of spectrum of interpolator?
Y ( e jω )= X ( e jωL )
13. What is the purpose of adaptive equalizer?
The purpose of adaptive equalizer is for compensating the channel distortion so that
the detected signal will be reliable.
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 14. Why adaptive filter is necessary?
A filter will be optimal if it is designed with some knowledge about the input data. If
this information is not known, then adaptive filter is used.
15. Why FIR filters are used in adaptive filtering?
FIR filters are mostly used in adaptive filtering because of its simplicity and it has
only adjustable zeros.
16. Define adaptive filtering.
Adaptive filters are used in various areas where the statistical knowledge of the
signals to be filtered are not known apriori or the signals may be time variant.
17. Define polyphase decomposition.
Polyphase decomposition results in reduction of computational complexity in FIR
filetr realization.
18. What is the reason to use the multi stage implementation of sampling
rate converter?
Reduced computation
Less storage space
Simple filter design
Less finite word length effect

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