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DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING

IT6502 DIGITAL SIGNAL PROCESSING
UNIT-I
SIGNALS AND SYSTEMS
1. Define: Signal and System.
Signal: Anything that carries information can be called a signal. Any physical quantity that varies with
time, space or any other independent variable.
System: System is an interconnection of components. It is a physical device that performs an
operation on an input signal and produces another signal as output.
2. What is signal processing? Give its types.
Signal Processing is any operation that changes the characteristics of a signal. These characteristics
include the amplitude, shape, phase and frequency content of the signal.
Analog signal processing
Digital signal processing
3. Define: Analog signal, Discrete-time signal.
Analog signal: An analog signal is a function having an amplitude varying continuously for all values of
time. Hence, an analog signal is continuous in both time and amplitude.
Ex. Speech signal, ECG signal
Discrete-time signal: A discrete-time is a function defined only at particular time instants. It is discrete
in time but continuous in amplitude.
Ex. Cricket score
4. What is meant by digital signal?
A digital signal is a special form of DT signal which is discrete in both time and amplitude, obtained by
quantizing each value of the DT signal. These signals are called digital because their samples are
represented by numbers or digits.
5. What are the classifications of discrete-time signals?
1. Energy and Power
2. Periodic and aperiodic
3. Even and odd
6. Define: Energy and power signals.
A signal x(n) is called an energy signal if and only if the energy of the signal is finite and P=0
The energy of the DT signal x(n) is defined as

E=

|x (n)|

n=

A signal x(n) is called a power signal if the average power of the signal is finite and E=
The average power of the DT signal x(n) is defined as
N

P= lim
N

2
1
| x( n)|

2 N +1 n =N

7. Define: Periodic and aperiodic.

A DT signal x(n) is said to be periodic with period N if and only if

x(N+n)=x(n)

for all n

If the above condition does not satisfy even for one value of n then the DT signal is aperiodic.
8. Define: Even signal and odd signal.
Even or symmetric signal:
x(n)=x(-n) for all n
Odd or antisymmetric signal:
x(n)=-x(-n) for all n
9. What is Deterministic signal?
A deterministic signal is a signal exhibiting no uncertainty of value at any given instant of time. Its
instantaneous value can be accurately predicted by specifying a formula, algorithm or simply by its
describing statement in words.
Ex. x(t)=Asint
10. What is a random signal?
A random signal is a signal characterized by uncertainty before its actual occurrence.
Ex. Noise
11. What are the different types of signal representation?
1. Graphical representation
2. Functional representation
3. Tabular representation
4. Sequence representation
12. What are the classifications of discrete-time systems?
1. Linear and non-linear
2. Time-variant and time-invariant
3. Causal and non-causal
4. Static and dynamic
5. Stable and unstable
13. Define: Linear and non-linear system.
A system that satisfies the superposition principle is said to be a linear system. Superposition principle
states that the response of the system to a weighted sum of signals should be equal to the
corresponding weighted sum of the outputs of the system to each of the individual input signals. A
system is linear if and only if
T[a1x1(n)+a2x2(n)]=a1T[x1(n)]+a2T[x2(n)]
For any arbitrary constants a1 and a2
14. Define: Time- variant and time-invariant system.
A system is said to be time-invariant or shift invariant if the characteristics of the system do not
change with time. For a time-invariant system if y(n) is the response of the system to the input x(n),
then the response of a system to the input x(n-k) is y(n-k).
If the system is time-variant
y(n,k)y(n-k)
15. Define: Causal and non-causal system.
A system is said to be causal if the output of the system at any time n depends only at present and
past inputs, but does not depend on future inputs.
If the output of the system depends on future inputs, the system is said to be non-causal.
16. Define: Static and dynamic system.
A system is called static or memoryless if its output at any instant n depends on the input samples at
the same time, but not on past or future samples of the input. In any other case, the system is said to
be dynamic.
17. Define: Stable and unstable system.
A system is stable if it produces a bounded output sequence for every bounded input sequence.
If, for some bounded input sequence x(n), the output is unbounded (infinite), the system is classified
as unstable.
18. Define: Sampling theorem.

A band limited continuous time signal, with higher frequency fm Hertz, can be uniquely recovered from
its samples provided the sampling rate

F s 2 f m samples per second.

Where is sampling frequency and fm is maximum frequency component of the signal.

19. What is meant by aliasing? How it can be avoided?
Aliasing: When the sampling frequency is less than twice of the highest frequency content of the
signal, then aliasing in frequency domain takes place. In aliasing, the high frequencies of the signal
mix with lower frequencies and create distortion in frequency spectrum.
To avoid aliasing:
i)
Sampling frequency must be higher than twice of highest frequency present in the signal.
ii)
A lowpass filter must be used before sampling to bandlimit the signal to some specific
frequency.
20. What are the different methods of evaluating inverse Z-transform?
1. Long division method
2. Partial fraction expansion method
3. Residue method
4. Convolution method
21. What is ROC? Mention its properties.
ROC: The region of convergence of X(z) is the set of all values of z for which X(z) attains a finite value.
Properties:
i) ROC does not contain any poles
ii)ROC of the causal sequence is of the form |z|>r
iii)
ROC of the left handed sequence is of the form |z|<r
22. Distinguish linear convolution and circular convolution.
Sl.No. Linear convolution
Circular convolution
1. Length of the output sequence N=L+M-1
Length of the output sequence N=max(L,M)
2.
3.

It can be used to find the response of a

linear filter.
Zero padding is not necessary to find the
response of a linear filter.

It cannot be used to find the response of a

linear filter.
Zero padding is necessary to find the
response of a filter.

23. What is the need for zero padding?

When the length of the sequence is to be increased, zeros are inserted as samples. This does not
change meaning.
24. Define: Properties of convolution.
Commutative: x(n)*h(n)=h(n)*x(n)
Associative:
[x(n)*h1(n)]* h2(n)= x(n)*[h1(n)* h2(n)]
Distributive:
x(n)*[h1(n)+ h2(n)]= x(n)*h1(n)+x(n)* h2(n)
25. State linearity property of Z-transform.
If X1(z)=Z{x1(n)} and X2(z)=Z{x2(n)} then,
Z{ax1(n)+bx2(n)}=aX1(z)+bX2(z)
26. State time shifting property of Z-transform.
If X(z)=Z{x(n)} and the initial conditions for x(n) are zeros, then
Z{x(n-m)}=z-m X(z)
27. State time reversal property of Z-transform.
If X(z)=Z{x(n)}, then
Z{x(-n)}=X(z-1) where the ROC is 1/r1<|z|<1/r2
28. State differentiation property of Z-transform.
If X(z)=Z{x(n)}, then

Z {nx (n) }=z

d
X (z )
dz

29. State scaling property of Z-transform.

If X(z)=Z{x(n)}, then
Z{an x(n)}=X(a-1z)
30. State Convolution theorem of Z-transform.
If X(z)=Z{x(n)}, and H(z)=Z{h(n)} then
Z{x(n)*h(n)}=X(z)H(z)

UNIT-II
FREQUENCY TRANSFORMATIONS
1. Define: DTFT pair.

x ( n ) is defined by

The DTFT X(ej)of a sequence

X ( e j )=

x ( n ) e jn

n =

x(n) can be determined from using X(ej) the Fourier integral expressed by

1
x (n)=
X ( e j) e jn d
2

2. Define: DFT and IDFT.

The N-point DFT of the sequence x(n) is expressed by
N1

X ( K )= x ( n ) e j 2 nk / N , K=0,1, N1
n=0

and the corresponding IDFT is given by

x ( n )=

1
N

N 1

X ( K ) e j 2 nk / N , n=0,1, N 1

K =0

3. State the difference between DTFT and DFT.

Sl.No
DTFT
.
1
DTFT is applicable to any arbitrary
sequences
2
Sampling is performed only in time domain
3

Continuous function of

DFT
It can be applied only to finite length sequences
Obtained by performing sampling operation in
both the time and frequency domain.
Discrete frequency spectrum

4. Define: Periodicity property.

x(n+N)=x(n)
for all n
X(N+k)=X(k)
for all k

5. State linearity property of DFT.

If x1 ( n ) DFT X 1 ( k )x 2 ( n ) DFT X 2 ( k ) then ,

a1 x1 ( n ) +a 2 x 2 ( n ) DFT a1 X 1 ( k ) +a2 X 2 ( k )

Here a1 and a2 are constants

6. State time shifting property.

If x ( n ) DFT X ( k ) then ,

j 2 k n0/ N

x ( nn 0 ) DFT e

X (k )

7. State Time reversal property.

If x ( n ) DFT X ( k ) then ,

k
()

x ( (n )) N =x ( Nn ) DFT X

This means when the sequence is circularly folded (reversed), its DFT is also circularly folded
8. State circular convolution property of DFT.

If x1 ( n ) DFT X 1 ( k )x 2 ( n ) DFT X 2 ( k ) then ,

x 1 ( n ) ( N ) x2 ( n ) DFT X 1 ( k ) X 2 ( k )

Here x1(n)(N)x2(n) means circular convolution of x1(n) and x2(n). This property states that multiplication
of two DFTs is equivalent to circular convolution of their sequences in time domain.
9. State circular time shift property of DFT.

If x ( n ) DFT X ( k ) then ,

j 2 kl / N

x ( ( nl ) )N DFT X (k) e

Thus shifting the sequence circularly by l samples is equivalent to multiplying its DFT by e-j2kl/N
10. State circular frequency shift property of DFT.

If x ( n ) DFT X ( k ) then ,

x (n )e

j 2 ln/ N

DFT X ( ( kl ) )N

Thus shifting the frequency components of DFT circularly is equivalent to multiplying the time domain
sequence by ej2ln/N
11. State the property of multiplication of two sequences.

If x1 ( n ) DFT X 1 ( k )x 2 ( n ) DFT X 2 ( k ) then ,

x 1 ( n ) x 2 ( n ) DFT

1
X ( k ) (N) X 2 ( k )
N 1

This means multiplication of two sequences in time domain results in circular convolution of their DFTs
in frequency domain.
12. State the complex conjugate property.
This property states that

If x ( n ) DFT X ( k ) then ,

x ( n ) DFT X ( (k ) )N =X ( Nk)

13. State Parsevals theorem.

Consider the complex valued sequences x(n) and y(n). then

If x ( n ) DFT X ( k ) y ( n ) DFT Y ( k ) then

N 1

x ( n) y ( n) =

n=0

1
N

N 1

X ( k ) Y (k )
k=0

14. What is FFT?

FFT is an algorithm used to compute the DFT that effectively reduces the DFT computation time takes
the advantage of periodicity and symmetry properties of twiddle factor

W N . It is based on the

fundamental principle of decomposing the computation of DFT of a sequence of length N into smaller
DFTs.
It improves the performance by a factor 100 or more over direct evaluation of DFT.
15. What are the advantages of FFT algorithm over direct computation of DFT?
FFT requires less number of multiplications and additions compared to direct computation of DFT.
FFT algorithms can be implemented fast on the DSP processor.
The calculation of DFT and IDFT both are possible by proper combination of FFT algorithms.
16. What is twiddle factor? Write its properties.
Twiddle factor or complex-valued phase factor which is an Nth root of unity expressed by
j2/N

W N =e
K+ N / 2

Symmetry:

WN

Periodicity:

W N =W N

K+ N

=W N
K

17. What is the number of complex multiplications and complex additions required in direct
computation and in FFT algorithm?
Computations
Direct Computation
FFT Algorithm
Complex Multiplications
N2
(N/2)log2N
Complex Additions
N(N-1)
N log2N
18. What is DIT-FFT algorithm?
In DIT the given sequence is decimated into two N/2 point sequences consisting of even numbered
values of x(n) and odd numbered values of x(n).
19. What is DIF-FFT algorithm?
DIF-FFT decomposes the DFT by recursively splitting the sequence elements X(K) in the frequency
domain into sets of smaller and smaller subsequences.
20. What are the differences and similarities between DIT and DIF algorithm?
Sl.No
DIT
DIF

.
1

2
3

In DIT the given sequence is decimated into

two N/2 point sequences consisting of even
numbered values of x(n) and odd numbered
values of x(n).
The input is in bit reversal order while the
output is in natural order.
Phase factor placed before each stage

DIF-FFT
decomposes
the
DFT by
recursively splitting the sequence elements
X(K) in the frequency domain into sets of
smaller and smaller subsequences.
The input is in natural order while the output
is in bit reversal.
Phase factor placed after each stage

21. What are similarities between DIT and DIF algorithm?

Both algorithms require N log2N operations to compute the DFT
Both algorithms can be done in in-place and both need to perform bit-reversal at some place
during the computation.
22. What is meant by in-place computation?
The same memory location is used to store the new values in place of the input values. An
algorithm that uses the same location to store both the input and output sequences is called in-place
computation.
23. What is the number of stages required to compute N-point DFT using radix-2 FFT algorithm.
Number of stages L=log2N
If N=8 then the Number of stages
L= log28 =3
If N=64 then the Number of stages
L= log264 =6
24. Mention the applications of FFT algorithm.
Linear filtering
Digital spectral analysis
Correlation analysis
25. What is meant by bit reversal in FFT algorithm?
Input sample
0
1
2
3
4
5
6
7

Binary
000
001
010
011
100
101
110
111

Bit reversed
000
100
010
110
001
101
011
111

Bit reversal
0
4
2
6
1
5
3
7

26. What is the speed improvement factor in calculating 64-point DFT of a sequence using
direct computation and FFT algorithms?
The number of complex multiplications required using direct computation is N2=642=4096
The number of complex multiplications required using FFT is (N/2)log2N=(64/2) log264= 32*6=192
Speed improvement factor= 4096/192 = 21.33
27. Draw the basic butterfly structure of DIT-FFT algorithm.

28. Draw the basic butterfly structure of DIF-FFT algorithm.

29. What are the steps involved in finding IDFT using FFT algorithm?
1. Take complex conjugate of X(k) that is X*(k)
2. Find DFT of X*(k) that is x*(n)
3. Divide by number of samples N
4. Take complex conjugate of x*(n)
30. What is meant by radix-2 FFT algorithm?
In radix-2 FFT algorithm the decimation arrives finally at 2-point sequence. The 2-point DFT is
calculated by direct computation.

UNIT-III
IIR FILTER DESIGN
1. What is the difference between analog filter and digital filter?
Sl.No
Analog Filter
Digital Filter
.
1
Analog filter processes analog input and A digital filter processes and generates
generates analog outputs.
digital data.
2
Analog filters are constructed from active or It consists of elements like adder, multiplier
passive electronic components.
and delay unit.
3
It is described by differential equation.
It is described by difference equation.
4
The frequency response of an analog filter The frequency response can be changed by
can be modified by changing the changing the filter coefficients.
components.
2. Distinguish FIR and IIR filters.
Sl.No
FIR Filter
.
1
These filters can be easily designed to have
perfectly linear phase.
2
FIR filters can be realized recursively and
non-recursively
3
Greater flexibility to control the shape of their
magnitude response.
4
Errors due to round off noise are less severe
in FIR filters, mainly because feedback is not
used.

IIR Filter
These filters do not have linear phase.
IIR filters are easily realized recursively.
Less flexibility, usually limited to specific kind
of filters.
The round off noise in IIR filters are more.

3. Mention the procedures for digitizing the transfer function of an analog filter.
a) Approximation of derivative
b) Impulse invariant method
c) Bilinear transformation
4. What is approximation of derivative?

This is the method to convert analog filter to digital filter by substituting

1

s=

1z
T

5. What is meant by impulse invariant method of designing IIR filter?

In this method of digitizing an analog filter, the impulse response of resulting digital filter is a sampled
version of the impulse response of the analog filter.
In impulse invariant method, the desired impulse response of the digital filter is obtained uniformly
sampling the impulse response of the equivalent analog filter. It is many-to-one mapping where
many points in s-plane are mapped to a single point in the z-plane.
6. What are the properties of impulse invariant transformation?

1
1

sa 1e aT z1
1
1

aT 1
s+ a 1e z
1eaT ( cos bT ) z1
s+a

( s+ a )2 +b2 12 eaT ( cos bT ) z1 +e2 aT z2

aT
1
e ( sin bT ) z
b

( s+ a )2 +b2 12 eaT ( cos bT ) z1 +e2 aT z2

7. Define bilinear transformation with expressions.

The bilinear transformation is a mapping that transforms the left half of s-plane into the unit circle in
the z-plane only once, thus avoiding aliasing of frequency components. The mapping from the splane to z-plane in bilinear transformation is

s=

2 ( z1 )
T ( z +1 )

All points in the LHP of s-plane are mapped inside the unit circle in the z-plane and all points in the
RHP of s-plane are mapped inside the unit circle in the z-plane
8. Compare bilinear and impulse invariant method.
The bilinear transformation is a conformal mapping that transforms the

j axis into the unit circle

in the z-plane only once, thus avoiding aliasing of frequency components.

In impulse invariant method, the desired impulse response of the digital filter is obtained uniformly
sampling the impulse response of the equivalent analog filter. It is many-to-one mapping where
many points in s-plane are mapped to a single point in the z-plane.
9. What is meant by warping effect and how it is eliminated?
For smaller values of there exist linear relationship between and .
= T
But for larger values of the relationship is non-linear. This effect is known as warping effect. This
effect compresses the magnitude and phase response at high frequencies.

The warping effect can be eliminated by prewarping the analog filter. This can be done by finding
prewarping analog frequencies using the formula

2
tan
T
2

10. What are the advantages and disadvantages of bilinear transformation?

Advantages:
1. The bilinear transformation provides one-to-one mapping.
2. Stable continuous systems can be mapped into realizable, stable digital systems
3. There is no aliasing
Disadvantages:
1. The mapping is highly non-linear producing frequency compression at high frequencies
2. Neither the impulse response nor the phase response of the analog filter is preserved in a
digital filter obtained by bilinear transformation.
11. Mention the properties of Butterworth filter.
i.
The magnitude response of the Butterworth filter decreases monotonically as the
frequency increases from 0 to
ii.
iii.

The magnitude response of the Butterworth filter closely approximates the ideal response
as the order N increases.
The poles of the Butterworth filter lie on circle.

12. What are the properties of Chebyshev filter?

i.
The magnitude response of the Chebyshev filter exhibits ripple either in passband or in
stopband according to type.
ii.
The poles of the Chebyshev filter lie on an ellipse
13. Distinguish recursive realization and non-recursive realization.
For recursive realization the present output y(n) is a function of past outputs, past and present
inputs. This form corresponds to an IIR digital filter.
For non-recursive realizations the current output y(n) is a function of only past and present inputs.
This form corresponds to an FIR digital filter.

14. What are the different types of structures for realization of IIR systems?
i.
Direct form-I structure
ii.
Direct form-II structure
iii.
Cascade realization
iv.
Parallel realization
v.
Lattice-ladder structure
vi.
Transposed structure
15. Draw Direct form-I structure for IIR filter.

16. Draw Direct form-II structure for IIR filter.

17. Draw cascade structure for IIR filter.

18. Draw parallel structure for IIR filter.

19. The transfer function of an IIR system has z number of zeros and p number of poles. How
many number of additions, multiplications and memory locations are required to realize the
system in direct form-I and direct form-II?
The realization of IIR system with z zeros and p poles in direct form I and II structure, involves z+p
number of additions and z+p+1 number of multiplications. The direct form I structure requires z+p
memory locations whereas the direct form II structure requires only p number of memory locations.
20. What are the factors that influence the choice of structure for realization of an LTI system?
Computational complexity
Finite word length effects
Parallel processing
Pipelining of computations
21. What is the advantage in cascade and parallel realization of IIR systems?
In digital implementation of LTI system the coefficients of the difference equation governing the
system are quantized. While quantizing the coefficients the value of poles may change. This will end
up in a frequency response different to that of desired frequency response.
These effects can be avoided or minimized, if the LTI system is realized in cascade or parallel
structure.
22. Compare direct form-I and II structures if IIR systems, with M zeros and N poles.
Sl.No.
Direct form-I
Direct form-II
1
Separate delay for input and output
Same delay for input and output
2
M+N+1 multiplications are involved
M+N+1 multiplications are involved
3
M+N additions are involved
M+N additions are involved
4
M+N delays are involved
N delays are involved
5
M+N memory locations are required
N memory locations are required
6
Noncanonical structure
Noncanonical structure
23. What is the main disadvantage of direct-form realization?
The direct-form realization is extremely sensitive to parameter quantization. When the order of the
system N is large, a small change in a filter coefficient due to parameter quantization, results in a
large change in the location of the poles and zeros of the system.
24. How one can design digital filter from analog filters?
1. Map the desired digital filter specifications into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
25. What are the properties that are maintained same in the transfer of analog filter into a
digital filter?

The j 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.
The left half plane of the s-plane should map into the inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
26. Compare between Butterworth and Chebyshev filter.
Sl.No.
Parameter
Butterworth filter
1
Frequency response
Monotonically decreasing
2
3

Order for given set of

specifications
Transition band

Phase response

Higher than Chebyshev

Transition band is broader than
Chebyshev for given order
Fairly linear phase response. It
is better than Chebyshev

Chebyshev filter
Ripples in passband and
monotonic in stopband
Lower than Butterworth
Transition band is narrower
than Butterworth for given order
Relatively nonlinear phase
response. It is inferior to
Butterworth filter

27. What are the limitations of impulse invariant mapping technique?

Frequency mapping is many to one. Therefore aliasing takes place in frequency domain.
Impulse invariant technique is suitable only for low pass and narrow band pass filters.
28. What are the properties of impulse invariant transformation?
Only poles of system function are mapped.
There is aliasing in frequency domain due to mapping.
Frequency relationship =T is linear.
Stable analog filter is converted to stable digital filter.
29. Comment on the passband and stopband characteristics of Butterworth and Chebyshev
filters.
Butterworth filters have monotonically decreasing response. Chebyshev filters have ripples in
passband and monotonic response in stopband.
For the same order, transition band Chebyshev filters is narrow than that in Butterworth filter.
Poles of Butterworth filter lie on the circle, whereas poles of Chebyshev filter lie on the ellipse.
For the same specifications, order of the Chebyshev filter is low compared to Butterworth filter.
30. Why do we go for approximation to design a digital filter?
These are effective filter approximation techniques available in analog domain. Using transformation
methods a stable analog digital filter can be converted to stable digital filters. Hence it becomes
easier to design IIR filters from analog filters. But such effective approximations are not available in
discrete domain.

UNIT-IV
FIR FILTER DESIGN

1. What are the different types of filters based on impulse response?

IIR filters are of recursive type, whereby the present output sample depends on the present input,
past input samples and past output samples.
FIR filters are of non-recursive type whereby the present output sample depends on the present
input sample and previous input samples.
2. What are the advantages and disadvantages of FIR filters?
Advantages:
FIR filters have exact linear phase and are always stable.
FIR filters can be realized in both recursive and non recursive structure.
They can be realized efficiently in hardware.
Disadvantages:
The implementation of narrow transition band FIR filters are very costly, as it requires
considerably more arithmetic operations and hardware components such as multipliers,
adders and delay elements.
Memory requirements and execution time are very high.
3. What are the applications of FIR filters?
Wideband differentiator
Hilbert transformer
4. What are the techniques of designing FIR filter?
Fourier series method
Windowing technique
Frequency sampling technique
5. What condition on the FIR sequence h(n) are to be imposed in order that this filter can be
called a linear phase filter?
Symmetric condition:
h(n)=h(N-1-n)
Antisymmetric condition:
h(n)=-h(N-1-n)
6. What do you understand by linear phase response in filters?
For a linear phase filter () is directly proportional to frequency , the linear phase filter
does not alter the shape of the original signal. If the phase response of the filter is nonlinear the
output signal may be distorted one. In many cases a linear phase characteristic is required
throughout the passband of the filter preserve the shape of a given signal within the passband. FIR
filter can give linear phase, when the impulse response of the filter is symmetric about its mid-point.
7. What are the disadvantages of Fourier series method?
In designing FIR filter using Fourier series method the infinite duration impulse response is truncated
at

n=(

N 1
) . Direct truncation of the series will lead to fixed percentage overshoots and
2

undershoots before and after an approximated discontinuity in the frequency response.

8. Define phase delay and group delay.
Phase delay:

p=

( )

Group delay: The group delay is defined as the delayed response of the filter as a function of
frequency to a signal.

g=

d()
d

9. What is the reason that FIR filter is always stable?

FIR filter is always stable because all its poles are at the origin.
10. What are the characteristic features (or) properties of FIR filters?
FIR filter is always stable
A realizable filter can always be obtained
FIR filter has a linear phase response
11. What is Gibbs phenomenon? (Or) What are Gibbs oscillations?
The finite duration causal filer can be obtained by truncating the infinite duration impulse
response and delaying the resulting finite duration impulse response. This modification does not
affect the amplitude response of the filter; however, the abrupt truncation of the Fourier series results
in oscillations in the pass band and stop band. This effect is known as Gibbs phenomenon.
These undesirable oscillations can be reduced by multiplying the desired impulse response
coefficients by an appropriate window function.
12. What is the principle of designing FIR filter using frequency sampling method?
In this method, a set of samples is determined from the desired frequency response and are
identified as DFT coefficients. The IDFT of this set of samples then gives the filter coefficients.
13. For what type of filters frequency sampling method is suitable?
Frequency sampling method is attractive for narrow band frequency selective filters where only a few
of the samples of the frequency response are non-zero.
14. Draw direct form realization of FIR structure.

15. Draw the direct form realization of a linear phase FIR system.

16. Compare the direct form and linear phase structures of an Nth order FIR system.
Sl.No.
Direct form-I
Direct form-II
1
Impulse response need not be symmetric
Impulse response should be symmetric
2
N multiplications are involved
N/2 or (N+1)/2 multiplications are involved
3
N-1 additions and delays are involved
N-1 additions and delays are involved
4
N-1 memory locations are required
N-1 memory locations are required

17. State the condition for a digital filter to be causal and stable.
A digital filter is causal if its impulse response h(n)=0 for n<0
A digital filter is stable if its impulse response is absolutely summable i.e.,

|h(n)|<

n=

18. When a cascade form realization is preferred in FIR filters?

The cascade form realization is preferred when complex zeros with absolute magnitude are
less than one.
19. Suppose the axis of symmetry of impulse response h(n) lies half way between two samples,
for what kind of applications this type of impulse response is used.
If the axis of symmetry lies midway between two samples, such type of sequences can be used to
design Hilbert transformers and differentiators.
20. For what kind of applications, the antisymmetrical impulse response can be used?
The antisymmetrical impulse response can be used to design Hilbert transformers and differentiators.
21. For what kind of applications, the symmetrical impulse response can be used?
The impulse response, which is symmetric having odd number of samples can be used to design all
types of filters that is low pass, high pass, band pass and bandstop filters.
22. Draw the cascade form structure for FIR filter.

23. What is canonic structure?

If the number if delays in the structure is equal to order of the difference equation or order of the
transfer function, then it is called canonic form realization.
24. What us windowing and why it is necessary?
Unit sample response of the desired filter is obtained from frequency response H d(). This unit sample
response is normally infinite in length. Hence it is truncated to some finite length. This truncation
creates oscillations in passband and stopband of the filter. This problem can be avoided with
windowing. The desired unit sample response is multiplied with suitable window. The length of the
window can be selected to desired value. Due to windowing, the unit sample response of the filter is
reshaped such that ringing (oscillations) are reduced.
25. What are the attractive aspects of frequency sampling design?
In frequency sampling method, the desired magnitude response is sampled. These samples of
frequency response are DFT coefficients. These frequency samples can be set as per the
requirement. When the number of samples are limited, then desired frequency response is
implemented efficiently. Narrowband frequency selective filters can be implemented better with the
help of frequency sampling.

26. Write the design steps involved in FIR filter design.

From the given frequency response calculate required order of the filter.
From the order and desired frequency response calculate desired unit sample response hd(n)
From the attenuation characteristics select suitable window function w(n)
Calculate h(n)=hd(n)w(n)
27. Compare FIR filters and IIR filters with regard to: a) Stability b) Complexity
Sl.No.
Parameter
FIR filter
IIR filter
1
Stability
Inherenty, stable
Can be unstable
2
Complexity
Simple
complex

28. List the features of rectangular window.

The main-lobe width is 4/N
The maximum side-lobe magnitude is -13dB
The side-lobe magnitude slightly decreases with increasing
The minimum stopband attenuation is 22dB
29. List the features of Hamming window.
The main-lobe width is 8/N
The maximum side-lobe magnitude in window spectrum is -41dB
The side-lobe magnitude remains constant
The minimum stopband attenuation is 51dB
30. List the features of Hanning window.
The main-lobe width is 8/N
The maximum side-lobe magnitude in window spectrum is -31dB
The side-lobe magnitude decreases with increasing
The minimum stopband attenuation is 44dB
UNIT-V
FINITE WORD LENGTH EFFECTS IN DIGITAL FILTERS
1. What is meant by finite wordlength effects in digital filters?
The digital implementation of the filter has finite accuracy. When numbers are represented in
digital form, errors are introduced due to their finite accuracy. These errors generate finite precision
effects or finite wordlength effects.
2. List some of the finite word length effects.
i.
Quantization effects in analog-to-digital conversion
ii.
Product quantization and coefficient quantization errors in digital filters.
iii.
Limit cycles in IIR filters
iv.
Finite word length effects in FFT
3. What is product round-off noise?
When the digital filters are implemented using fixed point arithmetic, the results of product or
multiplication operations are quantized to fit into the finite word length. This quantization uses
rounding operation. Hence errors generated in such operation are called product round-off errors.
4. What are the two types of quantization methods?
Truncation: Process of discarding all bits less significant than least significant bit that is retained.
Rounding: Rounding of a number of b bits is accomplished by choosing the rounded result as the b bit
number closest to the original number unrounded.

5. What are the errors can be occurred due to quantization?

1. Input quantization error
2. Product quantization error
3. Coefficient quantization error
6. What is quantization error?
The input x(n) is obtained by sampling the analog input signal. Since the quantizer takes on
only fixed (discrete) values of x(n), error is introduced. The actual input can be denoted by xq(n).
Hence quantization error is given as,
e(n)=xq(n)-x(n)
Here e(n) is the quantization error introduced during analog to digital conversion process due
to finite wordlength of the quantizer.
7. What is input quantization error?
Any analog signal needs to be converted to digital form for processing in DSP system. The analog
signal has infinite precision. But it is represented by finite number of amplitude levels (finite number of
digits) in A/D conversion process. This introduces an error, which is called as input quantization error.
8. What is product quantization error?
When the two b bit numbers are multiplied, the result is 2b bits. But the output register of multiplier
or the result bus has the wordlength of b bits. Hence it is necessary to truncate or roundoff the 2b
bits result to b bits. This introduces an error, which is called as product quantization error.
9. What is coefficient quantization error?
The design calculations of digital filters are done with infinite precision. But the filter coefficients are
quantized according to finite wordlength of the registers. This quantization shifts the filter coefficients
(and hence the poles) from their actual designed values. This error is called coefficient quantization
error. It makes the filter unstable some times.
10. What are the different types of arithmetic in digital systems?
1. Fixed point representation
2. Floating point representation
3. Block floating point representation
11. What do you understand by fixed-point number?
In fixed-point arithmetic the position of the binary point is fixed. The bit to the right represent the
fractional part og the number and those to the left represent the integer part.
12. What are the different types in fixed-point number representation?
1) Sign-magnitude
2) 1s complement
3) 2s complement
13. What is meant by floating point arithmetic?
In floating point representation a positive number is represented as F=2C M
Where M called Mantissa is a function such that 0.5M<1
C is the exponent can be either positive or negative
14. What is meant by block floating point arithmetic?
In block floating point arithmetic the set of signals to be handled is divided into blocks. Each block has
the same value for the exponent. The arithmetic operations within the block use fixed point arithmetic
and only one exponent per block is stored thus saving memory. This representation of numbers is
most suitable in certain FFT flow graphs and in digital audio applications.

15. What are the advantages of floating point arithmetic?

1) Larger dynamic range
2) Overflow in floating point representation is unlikely
16. Differentiate between fixed point and floating point arithmetic.
Sl.No
Fixed point arithmetic
Floating point arithmetic
.
1
Fast operation
Slow operation
2
Relatively economical
More expensive because of costlier hardware
3
Small dynamic range
Increased dynamic range
Roundoff errors occur only for addition Roundoff errors can occur with both addition and
4
multiplication
5
Overflow occurs in addition
Overflow does not arise
6
Used in small computers
Used in larger, general purpose computers

17. What are limit cycles?

In recursive systems when the input is zero or some nonzero constant value, the nonlinearities due to
finite precision arithmetic operations may cause periodic oscillations in the output. These oscillations
are called limit cycles.
18. What are two types of limit cycles?
1. Zero input limit cycles
2. Overflow limit cycles
19. What is zero input limit cycle oscillations?
In the recursive systems, the finite-precision arithmetic operations cause periodic oscillations in the
output. These oscillations are called limit cycle oscillations. These oscillations continue to remain even
when the input is made zero. Then they are called zero input limit cycle oscillations.
20. What is Overflow limit cycle oscillations?
Limit cycle oscillations causing by rounding the result of multiplication, there are several types of limit
cycle oscillations caused by addition, which makes the filter output oscillate between maximum and
minimum amplitudes. Such limit cycles have been referred to as overflow oscillations.
21. How overflow limit cycles can be eliminated?
i) Saturation arithmetic
ii) Scaling
22. What is meant by saturation arithmetic?
One way to avoid the overflow is to modify the adder characteristics so that it performs saturation
arithmetic. Thus when an overflow is sensed, the sum of the adder is set equal to the maximum value.
But saturation arithmetic causes undesirable signal distortion due to non-linearity in the adder.
23. What is the need for signal scaling?
Scaling is required in filter implementation to prevent overflow of values in digital hardware. The digital
hardware has finite number of bits. Hence they can handle only limited range of values. If any
parameter becomes large during computation, it is to be scaled to prevent overflow.
24. Define: Dead Band.
The limit cycles occur as a result of the quantization effects in multiplications. The amplitudes of the
output during a limit cycle are confined to a range of values that is called the dead band of the filter.

25. Why rounding is preferred to truncation in realizing digital filter?

i.
The quantization error due to rounding is independent of the type arithmetic.
ii.
The mean of rounding error is zero.
iii.
The variance of the rounding error signal is low.
26. Why the limit cycle problem does not exist when FIR digital filter is realized in direct form
or cascade form?
In the case of FIR filters, there are no limit cycle oscillations if the filter is realized in direct form or
cascade form since these structures have no feedback.
27. How the system output can be brought out of limit cycle?
The system output can be brought out of limit cycle by applying an input of large magnitude, which is
sufficient to drive the system out of limit cycle.
28. What us the drawback of saturation arithmetic?
The saturation arithmetic introduces nonlinearity in the adder which creates signal distortion.
29. Define: Noise transfer function (NTF).
The noise transfer function is defined as the transfer function from the noise source to the filter output.
The NTF depends on the structure of the digital network.
30. What are the assumptions made regarding the statistical independence of the various
noise sources in the digital filter?
i.
Any two different samples from the same noise source are uncorrelated.
ii.
Any two different noise source, when considered as random processes are uncorrelated.
iii.
Each noise source is uncorrelated with the input sequence.