TWO MARKS

DIGITAL SIGNAL PROCESSING

SUBJECT CODE: CS2403

SEM/YEAR: V/III

PART A
UNIT I
SIGNALS AND SYSTEMS
1. Define Signal and Signal Processing.
A signal is defined as any physical quantity that varies with time, space, or any
other independent variable.
Signal processing is any operation that changes the characteristics of a signal.
These characteristics include the amplitude, shape, phase and frequency content of a
signal.
2. What are the classifications of signals? (Nov/Dec 2010)
There are five methods of classifying signals based on different features:
(a) Based on independent variable.
(i)

Continuous time signal.

(ii)

Discrete time signal.

(b) Depending upon the number of independent variable.

(i)

One dimensional signal,

(ii)

Two dimensional signal.

(iii)

Multi dimensional signal.

(c) Depending upon the certainty by which the signal can be uniquely described as
(i)

Deterministic signal.

(ii)

Random signal.

(c) Based on repetition nature.

(i)

Periodic signal.

(ii)

Non Periodic signal.

(d) Based on reflection

(i)

Even signal.

(ii)

Odd signal

3. Define discrete system.

A discrete time system is defined as a device or algorithm that operates on a
discrete time input signal x(n) , according to some well defined rule , to produce another
discrete time signal y(n) called the output signal.
4. What are the classifications of discrete time systems?
The classifications of discrete time systems are
1. Static and Dynamic system.
2. Time variant and time invariant system.
3. Linear and non linear system.
4. Stable and Un-stable system.
5. Causal and non-causal system.
6. IIR and FIR system.
5. Differentiate Continuous time and discrete time signal.
Continuous time signal: It is also referred as analog signal i.e., the signal is
represented continuously in time.
Discrete time signal : Signals are represented as sequence at discrete time intervals .
6. Define digital signal.
A discrete time signal or digital is defined as which discrete valued represented by
a finite number of digits is referred to as a digital signal.
7. What is Deterministic signal? Give example.
A signal that can be uniquely determined by a well - defined process such as a
mathematical expression or rule, or look-up table is called a deterministic signal.
Example : A sinusoidal signal v(t ) Vm sin t
8. What is random signal? Give example.

A signal that is generated in a random fashion and cannot be predicted ahead of

time is called a random signal.
Example : Speech signal , ECG signal and EEG signal.
9. Define (a) Periodic signal (b) Non periodic signal.
Periodic signal: A periodic signal is defined as the signal x(n) is periodic with period
N if and only if x(n+N)=x(n) for all n.
Non periodic signal: A non periodic signal is defined as if there is no value of N
that satisfies the equation x(n+N) x(n) .
10. What are the symmetric and ant symmetric signals?
Symmetric signal: A real valued signal x(n) is called symmetric if x(-n) = x(n).
Antisymmetric signal: A signal x(n) is called antisymmetric if x(-n) = -x(n).
11. What are energy and power signals? (April/may 2011, Nov/Dec 2012)
Energy signal:
The energy of a discrete time signal x(n) is defined as

x ( n)

A signal x(n) is called an energy signal if and only if the energy obeys the relation
0 E . For an energy signal P=0.

Power signal:
The average power of a discrete time signal x(n) is defined as
N
1
P lim
x ( n)
N 2 N 1
n N

A signal x(n) is called power signal if and only if the average power P satisfies the
condition 0 P .

12. What are the different types of signal representation?

The different types of signal representation are
(i) Graphical representation

(ii) Functional representation

(iii) Tabular representation

(iv)Sequence representation

13. What are the different types of operations performed on discrete time
signals?
The different types of operations performed on discrete time signals are
(1)Delay of a signal
(2) Advance of a signal
(3)Folding or Reflection of a signal
(4) Time scaling
(5) Amplitude scaling
(6)Addition of signals
(7)Multiplication of signals.
14. Represent the following duration sequence x(n)={1, 3, -1, -4} as a sum of

weighted impulse sequences.

Given
x(n)={1, 3, -1, -4}

We can write
x( n) 1 x(k ) ( n k )
2

1 ( n 1) 3 ( n) 1 ( n 1) 4 ( n 2)

15. What is a static and dynamic system?

A discrete time system is called static or memory less if its output at any
instants n depends on the input samples at the same time , but not an past or future
samples of the input.
Ex., y(n) = ax(n)
Y(n)=ax2(n)
In any other case, the system is said to be dynamic or to have memory.
Ex., y(n) = ax(n-1)+x(n-2)
y(n)=x(n)+x(n-1)

16. What is a linear time invariant system? (Nov/Dec 2010)

A system is called time invariant if its input output characteristics do not
change with time.
Ex., y(n)=x(n)+x(n-1)
17. What is a causal system?
A system is said to be causal if the output of the system at any time n depends
only on present and past inputs, but does not depend on future inputs.
This can be expressed mathematically as,
y(n)=F[x(n), x(n-1), x(n-2)
18. Define a stable system.
Any relaxed system is said to be bounded input-bounded output (BIBO) stable if
and only if every bounded input yields a bounded output. Mathematically, their exist
some finite numbers, Mx and My such that,
x( n) Mx

y ( n) My

and

19. What is a 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 be equal to the corresponding weighted sum of the outputs of the system to
each of the individuals input signals.

20. Define unit sample response (impulse response) of a system and what is its
significance?
The unit sample response is defined as the output signal designated as h(n) ,
obtained from a discrete time system when the input signal is a unit sample
sequence ( unit impulse).

The output y(n) of an LTI system for an input signal x(n) can be obtained by
convolving the impulse response h(n) and the input signal x(n).
y ( n) x (n) h(n)

x ( k ) h( n k )

21. What is the causality condition for an LTI system?

The necessary and sufficient condition for causality of an LTI system is, its unit
sample response h(n)=0 for negative values of n i.e.,
h(n)= 0 for n<0
22. What is the necessary and sufficient condition on the impulse response for
stability?
The necessary and sufficient condition guaranteeing the stability of a linear timeinvariant system is that its impulse response is absolutely summable.

h(k )

23. What is meant by discrete or linear convolution?

The convolution of discrete time signal is known as discrete convolution. Let
x(n) be the input to an LTI system and y(n) be the output of the system. Let h(n) be
the response of the system to an impulse. The output y(n) can be obtained by
convolving the impulse response h(n) and the input signal x(n).
y ( n)

x ( k ) h( n k )

(OR )
y (n)

h(k ) x( n k )

24. What are the steps involved in the convolution process?

The steps involved in the convolution process are
1.

Express both sequence in terms of the index k.

2.

Folding: Fold the h(k) about the origin and obtain h(-k).

3.

Time shifting : Shift the h(k) by n units to right if n is

positive or
left if n is negative to obtain h(n-k).
4.

Multiplication: Multiply x(k) by h(n-k) to obtain

w(k)=x(k)h(n-k)
5.

Summation: Sum all the values of the product w(k) to obtain

the
value of output y(n).
6.

Increment the index n, shift the sequence h(n-k) to right by

one
sample and do step4.
25. What do you meant by sampling process?
Sampling is the conversion of a continuous time signal (or analog signal) into a
discrete time signal obtained by taking samples of the continuous time signal ( or
analog signal ) at discrete time instants.
26. State sampling theorem. (April/May 2011, Nov/Dec 2012)
A sampling theorem states that the band limited continuous time signal with
highest frequency (band width) fm hertz , can be uniquely recovered from its samples
provided that the sampling rate fs is greater than or equal to 2fm samples per second.
27. Define Nyquist rate. (Nov/Dec 2008)
The Nyquist rate is defined as the frequency 2fm , which , under sampling
theorem, must be exceeded by the sampling frequency.
28. What is meant by Quantization process?
The process of converting a discrete time continuous valued signal into
discrete time discrete valued signal is called Quantization.

29. What is aliasing effect? (April/May 2008)

The superimposition of high frequency component on the low frequency is
known as frequency aliasing or aliasing effect.
30. How can aliasing be avoided?
To avoid aliasing the sampling frequency must be greater than twice the
highest frequency present in the signal.
31. What is an anti aliasing filter? (Nov/Dec 2008)
A filter used to reject frequency signals before it is sampled to reduce the
aliasing is called an anti aliasing filter.
32. What is meant by critical sampling?
If the sampling frequency is exactly equal to the Nyquist rate is known as
critical sampling.
33. What are the steps involved in the A/D conversion?
The steps involved in the A/D conversion are
1. Sampling
2. Quantization
3. Coding (Each discrete value is represented by a binary sequence)
34. What is meant by Quantization error?
It is the difference between the quantized value and actual sample value.
eq(n)=xq(n)-x(n)
35. What is a quantization level?
The value of error that allows in a digital signal is called the quantization level.
36. Define resolution or quantization step size. (Nov/Dec 2006)
The resolution is defined as the distance between two successive level.

37. What is SQNR?

The quality of the output of the A/D converter is usually measured by the
signal to quantization noise ratio (SQNR) , which is ratio of the signal power to
noise power.
SQNR

Pav
Pq

38. What is the use of a Sample and Hold circuit?

The sample and hold circuit is used to hold the sample the analog signal
and hold the sampled value constant as long as the A/D converter takes time for
accurate conversion.
39. Define conversion time.
Conversion time may be defined as the time taken by an ADC for converting a
given amplitude , expressed in decimal value , of a quantized analog signal applied
across its input terminals into corresponding binary equivalent value.
40. Define voltage resolution.
The voltage resolution is defined as

Votagereso lution

VFS
2n 1

where
VFS - full scale voltage and

n number of bits.
41. Define percentage resolution.
Percentage resolution is defined as
% resolution

1
100
2n

42. What are the advantages and disadvantages of counter ramp type ADCs?
Advantages of counter ramp type ADCs are
1. The principle is simple and straightforward.
2. It is very easy to construct this ADC.
3. This basic principle is employed in many advanced ADCs.
Disadvantages counter ramp type ADCs are
1. Only increasing voltages can be measured.
2. The system is very slow.
3. This may be mainly used to read DC voltages.
4. Practically the system comes to rest only when Vd > Va . This set an
error in readings.
43. What are the advantages and disadvantages of SAADC?
Advantages of SAADC are
1. It is more accurate than the stair case (counter ramp) ADC.
2. It maintains a high resolution.
3. It is much faster.
4. Its conversion time is much less.
Disadvantages of SAADC are
1. It requires a complex register called the successive approximation
register
2. It is costly, as it contains more components.
44. Write down the formula for conversion time of SAADC?
Conversiontime

n
f

where
n number of bits and
f clock frequency.

10

45. What are the advantages and disadvantages of Flash converter?

Advantages of flash converter are
1. The fastest conversion process ( governed only by propagation delay
of the gates.
2. Highest accuracy
3. Highest resolution possible by increasing the number of comparators.
Disadvantages of flash converter are
1. Very complicated circuitry.
2. Cost is proportional to the number of comparators, which in turn
depends on the resolution required.
46. What are the different types of Analog to Digital converters?
The different types of analog to digital converters are
1. Flash A/D converter.
2. Successive approximation converter.
3. Counting type A/D converter.
4. Over sampling Sigma Delta converter.
47. What are the different types of Digital to Analog converter?
The different types of digital to analog converters are
1. Weighted resistor D/A converter.
2. Resistor ladder D/A converter.
3. Over sampling D/A converter.

48. Define Z transform.

The Z transform of a discrete time signal or sequence is defined as the power
series.
X ( z)

x ( n) z

11

49. What is meant by region of convergence?(April/May 2008)

The region of convergence (ROC) of X(z) is the set of all values of z for
which X(z) attains a finite value.
50. What are the properties of region of convergence?
The properties of region of convergence are
1. The ROC is a ring or disk in the Z plane centered at the origin.
2. The ROC cannot contains any poles.
3. The ROC of an LTI stable system contains the unit circle.
4. The ROC must be a connected region.
51. What are the properties of z- transform? (Nov/Dec 2008)
1. Linearity:

z a1 x1 (n) a 2 x2 (n) a1 X 1 ( z ) a 2 X 2 ( z )

m 1

z
x
(
n

m
)

z
X
(
z
)

x(i ) z m i
2. Shifting: (a)

i 0

(b) z x (n m) z m X ( z )

3. Multiplication: z n m x(n) z

dz

X ( z)

4. Scaling in z- domain: z a n x( n) X a 1 z
5. Time reversal : z x( n) X ( z 1 )
6. Conjugation: z x (n) X ( z )

h(n m)r (m)

7. Convolution: z

H ( z ) R( z )

m0

X ( z)
8. Initial value: z x(0) zLt

(1 z 1 ) X ( z )
9. Final value: z x( ) zLt
1
52. State Parsevals relation in z-transform. (April/May 2011)
Parsevals relation in z-transform state that

12

If x1(n) and x2(n) are complex valued sequences, then

x ( n) x

1
2j

( n)

1 1
v dv

v

(v ) X 2

53. What is the relationship between z-transform and DTFT? (Nov/Dec 2010)
The z-transform of x(n) is given by X ( z )

x ( n) z

.(1)

Where z re j
Substituting z value in eqn (1) we get,

x ( n) r

X (re j )

e jn ..(2)

The Fourier transform of x(n) is given by

X ( e j )

x ( n )e

j n

.(3)

Eqn(2) and Eqn(3) are identical , when r=1. In the z-plane this corresponds to the
locus of points on the unit circle

z 1.

Hence X (e j ) is equal to X(z) evaluated

along unit circle , or

X (e j ) X ( z ) | z e j

For X (e j ) to exist , the ROC of X(z) must include the unit circle.
54. What are the different methods of evaluating inverse z transform?
1. Long division method.
2. Partial fraction method.
3. Residue method.
4. Convolution method.
55. Find the convolution of the following using z- transform.

x(n) 1,1,1 ; h(n) 1, 2,1

Solution:

13

Z x(n) h(n) X ( z ) H ( z )
X ( z )(1 z 1 z 2 )
H ( z ) (1 2 z 1 z 2 )
X ( z ) H ( z ) (1 3 z 1 4 z 2 3 z 3 z 4 )
x(n) h(n) 1,3, 4,3,1
56. Define system function.
Let x(n) and y(n) is the input and output sequences of an LTI system with
impulse response h(n). Then the system function of the LTI system is defined as
the ratio of Y(z) and X(z), i.e.,

H ( z)

Y ( z)
X ( z)

where,
Y(z) is the z transform of the output signal y(n)
X(z) is the z transform of the input signal x(n)

UNIT II

14

FREQUENCY TRANSFORMATIONS
1. Define Fourier transform of a discrete time signal. (April/May 2008)
The Fourier transform of a discrete time signal x(n) is defined as
F x (n) X ( )

x ( n)e

jn

2. Why FFT of a discrete time signal is called signal spectrum?

By taking Fourier transform of a discrete time signal x(n) , it is decomposed
into its frequency components . Hence the Fourier transform is called signal
spectrum.
3. List the difference between Fourier transform of discrete time signal and
analog signal?
i) The FT of analog signal consists of a spectrum with a frequency range
to . But the FT of discrete time signal is unique in the range
to ( or 0 to 2 ) , and also it is periodic with periodicity of 2 .

ii) The FT of analog signals involves integration but FT of discrete time

signals involves summation.
4. Define inverse Fourier transform.
The inverse Fourier transform of X() is defined as
F 1 X ( ) x(n)

1
2

X ( )e

jn

5. Give some applications of Fourier transform. (Nov/Dec 2010)

The application of Fourier transform are
1. The frequency response of LTI system is given by the Fourier transform of
the impulse response of the system.

15

2. The ratio of the Fourier transform of output to Fourier transform of input

is the transfer function of the system in frequency domain.
3. The response of an LTI system can be easily computed using convolution
property of Fourier transform.
6. What is the frequency response of LTI system?
The Fourier transform of the impulse response h(n) of the system is called
frequency response of the system. It is denoted by H().
7. Write the properties of frequency response of LTI system.
The properties of frequency response of LTI system are
i) The frequency response is periodic function with a period of 2 .
ii) If h(n) is real then

H ( )

is symmetric and H ( ) is

antisymmetric.
iii) If h(n) is complex then the real part of H ( ) is antisymmteric over the
interval 0 2 .
iv)The frequency response is a continuous function of .
8. Write short notes on the frequency response of first order system.
A first order system is characterized by the difference equation
y ( n) x( n) ay (n 1)

The frequency response of first order system depends on the co efficient a in

the difference equation governing the LTI system. When the value of a| is in the
range of 0<a<1, the first order system behave as a low pass filter. When the value
of a is in the range 1<a<0, the first order system behave as a high pass filter.
9. Write a short note on the frequency response of second order system.
A second order system is characterized by the difference equation
y ( n) 2r cos 0 y (n 1) r 2 y (n 2) x( n) r cos 0 x (n 1)

16

The frequency response of second order system depends on the parameters

r and 0 in the difference equation the LTI system. When the value of r is in
the range of 0<r<1, the second order system behave as a resonant filter with
center frequency 0 . When the value of r is varied from 0 to 1 , the sharpness of
resonant peak increases.
10. Define discrete Fourier series.
Consider a sequence xp(n) with a period of N samples so that xp(n)=xp(n/N);
Then the discrete Fourier series of the sequence xp(n) is defined as
N 1

X p (k ) x p (n)e j 2kn / N
n 0

11. What are the two basic differences between the Fourier transform of a
discrete time signal with the Fourier transform of a continuous time
signal?
1. For a continuous signal , the frequency range extends from to .
On the other hand , the frequency range of a discrete time signal extends from
to ( or 0 to 2 ) .

2.The Fourier transform of a continuous signal involves integration ,

whereas , the Fourier transform of a discrete time signal involves summation
process.
12. Find the Fourier transform of a sequence x(n) = 1 for 2 n 2
= 0 otherwise.
Solution:
X ( )

x ( n ) e j n

n 2

e j 2 e j 1 e j e j 2
1 2 cos 2 cos 2

17

j n

13. Define the Discrete Fourier transformation of a given sequence x(n).

The N- point DFT of a sequence x(n) is
N 1

X ( k ) x ( n)e j 2kn / N

k= 0, 1, 2..N-1.

n 0

14. Write the formula for N- point IDFT of a sequence X(k).

The N-point IDFT of a sequence X(k) is
1
N

x ( n)

N 1

X ( k )e

j 2k / N

n= 0, 1 , 2 ..N-1 .

K 0

15. List any four properties of DFT. (Nov/Dec 2008)

(a) Periodicity
If

X(k) is N- point DFT of a finite duration sequence x(n)

then
x(n N ) x (n) for all n
X (k N ) X (k ) for all k .

(b)Linearity
If

X1(k)=DFT[x1(n)] and
X2(k)=DFT[x2(n)]

then
DFT[a1x1(n)+a2x2(n)]=a1X1(k)+a2x2(k)
( c ) Time reversal of a sequence
If DFT {x(n)}=X(k),
then
DFT{x((-n))N}=DFT{x(N-n)}=X((-k))N=X(N-k)
(d)Circular time shifting of a sequence
If DFT {x(n)}=X(k),
then
DFT{x((n-l))N}=X(k) e j 2kl / N

18

13. IF N-point sequence x(n) has N- point DFT X(k) then what is the DFT of the
following?
(i ) x (n)

(iii ) x((n l )) N (iv) x(n)e j 2 ln/ N

(ii ) x ( N n)

Solution:
(i ) DFT {x ( n)} X ( N k )
(ii ) DFT {x ( N n)} X ( k )
(iii ) DFT {x ((n l )) N } X ( k )e j 2kl / N
(iv ) DFT {x ( n)e j 2 ln/ N } X ((k l )) N

14. Calculate the DFT of the sequence x(n) =

forN 16

Solution:
X (k )

N 1

x ( n)e

j 2kn / N

K=0, 1 , 2..N-1

n0

n 0 4
15

e j 2kn / 16

1 j k / 8

e

n 0 4
15

16

1
j 2k
e
4
1
1 e jk / 8
4

18. State the time shifting property of DFT.

The time shifting properties of DFT states that
If

DFT[x(n)]=X(k),

then
DFT[x((n-m))N]= e j 2kn / N X (k )

19. Find the DFT of the sequence x(n)={ 1,1,0,0 }

Solution:

19

N 1

X ( k ) x ( n)e j 2kn / N

k=0,1,2,..N-1.

n 0

x(n)e j 2 kn /4

k 0,1, 2 , 3.

n 0

X (0) x(n) {1 1 0 0} 2
n 0
3

X (1) x( n)e j n /2 {1 j 0 0 } 1 j
n 0
3

X (2) x(n)e j n {1 1 0 0 } 0
n 0
3

X (3) x( n)e j 3 n /2 {1 j 0 0 } 1 j
n 0

X (k ) {2,1 j , 0,1 j }

20. When the DFT X(k) of a sequence x(n) is imaginary?

If the sequence x(n) is real and odd (or) imaginary and even, then X(k) is purely
imaginary.
21. When the DFT X(k) of a sequence x(n) is real?
If the sequence x(n) is real and even (or) imaginary and odd , then X(k) is purely
real.
22. State Circular frequency shifting property of DFT.
The circular frequency shifting property of DFT states that
If

DFT[x(n)]=X(k),

Then
DFT[x(n) e j 2 ln/ N ] X ((k l )) N
23. What is zero padding? What are its uses? (May/June 2009)

20

The process of lengthening the sequence by adding zero valued samples is

called appending with zeros or zero padding.
USES:
(i)

We can get better display of the frequency spectrum.

(ii)

With zero padding, the DFT can be used in linear filtering.

24. What do you understand by periodic convolution?

Let x1 p ( n) and x 2 p ( n) be two periodic sequences each with period N with

DFS x

( n) X

DFS x1 p (n) X 1 p (k )
2p

2p

and

(k )

(1)

If X 3 p (k ) X 1 p (k ) X 2 p (k )

then the periodic sequence x3 p ( n) with Fourier series coefficients X 3 p ( k ) can be

obtained by periodic convolution, defined as
N 1

x3 p (n) x1 p (m) x 2 p (n m) (2)

n 0

The convolution in the form of eqn(2) is known as periodic convolution, as the

sequences in eqn(2) are all periodic with period N, and the summation is over one
period.
25. Define circular convolution. (May/June 2009)
The convolution property of DFT is defined as the multiplication of the DFTs of
the two sequence is equivalent to the DFT of the circular convolution of the two
sequences.
X 1 (k ) X 2 (k ) DFT {x1 (n) x 2 (n)}
N 1

x3 (n) x1 (m) x 2 ((n m)) N

m0

26. How the circular convolution is obtained by using Graphical method?

21

Given two sequences x1 ( n) and x 2 (n) , the circular convolution of these two
sequences x3 (n) x1 ( n) Nx 2 (n) can be obtained by using the following steps.
1. Graph N samples of x1 ( n) , as equally spaced points around an outer circle
in counter clockwise direction.
2. Start at the same point as x1 (n) graph N samples of x 2 (n) as equally
spaced points around an inner circle in clock wise direction.
3. Multiply corresponding samples on the two circles and sum the products to
produce output.
4. Rotate the inner circle one sample at a time in clock wise direction and go to
step3 to obtain the next value of output.
5. Repeat step4 until the inner circle first sample lines up with the first sample of
the exterior circle once again.
27. Distinguish between linear and circular convolution of two sequences.
Sl.NO
Linear Convolution
1.
If x(n) is a sequence of L number of

Circular convolution
If x(n) is a sequence of L number of

samples and h(n) with M number of

samples and h(n) with M number of

samples , after convolution y(n) will

samples , after convolution y(n) will

2.

contain N=L+M-1
Linear convolution can be used to find

contain N=Max(L,M) samples.

Circular convolution cannot be used

3.

the response of a linear filter.

Zero padding is not necessary to find the

to find the response of a filter.

Zero padding is necessary to find the

response of a linear filter.

response of a filter.

28. What are the steps involved in circular convolution?

The circular convolutions involve basically four steps as the ordinary linear
convolution. These are
1. Folding the sequence
2. Circular time shifting the folded sequence
3. Multiplying the two sequences to obtain the product sequence.
22

4. Summing the values of product sequence.

29. What are the different methods performing circular convolution?
1. Graphical method
2. Stockhmans method
3. Tabular array method
4. Matrix method.
30. Obtain the circular convolution the following sequences
x(n)={1, 2, 1 }; h(n)={ 1, -2, 2 }
Solution:
The circular convolution of the above sequences can be obtained by using matrix
method.
h(0)
h(1)

h( 2)
h(0)

h(2)
1
2

h(1)
2
1

2
2

h(1) x (0)
y (0)
h( 2) x (1) y (1)
y ( 2)
h(0) x (2)
1
3
2 2

1
1

2
1
2
y ( n) 3, 2, 1

31. How will you obtain linear convolution from circular convolution.
Consider two finite duration sequences x9n) and h(n0 of duration L samples and
N samples respectively. The linear convolution of these two sequences produces an
output sequence of duration L+M-1 samples , whereas , the circular convolution of
x(n) and h(n) give N samples where N=Max(L,M) . In order to obtain the number of
samples in circular convolution equal to L+M-1, both x(n) and h(n) must be appended
with appropriate number of zero valued samples. In other words, by increasing the
length of the sequences x(n) and h(n) to L+M-1 points and then circularly convolving
the resulting sequences we obtain the same result as that of linear convolution.
32. What is meant by sectioned convolution?

23

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 up into smaller section. These sections are processed separately
one at a time and combined later to get the output.
33. What are the different methods used for the sectioned convolution?
The two methods used for the sectioned convolution are (i)the overlap-add
method (ii)Over lap-save method.
34. Differentiate (i) overlap-add method (ii) overlap save method.
Sl.No
Overlap add method
1.
In this method the size of the input
2.

3.

4.

Overlap save method

In this method the size of the input data

data block is N=L+M-1

Each data block consists of the last

block is L.
Each data block is L points and we

M-1 data points of the previous data

appended M-1 zeros to compute N-point

followed by the L new data points.

In each output block M-1 points are

DFT.
In this no corruption due to aliasing as

corrupted due to aliasing , as circular

linear convolution is performed using

convolution is employed.
To form the output sequence the first

circular convolution.
To form the output sequence, the last M-

M-1 data points are discarded in each

1 points from each output block is added

output block and the remaining datas

to the first (M-1) points of the

are fitted together.

succeeding block.

35. Distinguish between DFT and DTFT.

Sl.No
1.

2.

DFT
Obtained by performing sampling

DTFT
Sampling is performed only in time

operation in both the time and

domain.

frequency domains.
Discrete frequency spectrum

Continuous function of

24

36. What is FFT?

The term Fast Fourier Transform (FFT) usually refers to a class of algorithms for
efficiently computing the DFT.It makes use of the symmetry and periodicity
properties of twiddle factor W NK to effectively reduce the DFT computation time.
It is based on the fundamental principle of decomposing the computation of DFT of a
sequence of length N into successively smaller discrete Fourier transforms. The FFT
algorithm provides speed increase factors , when compared with direct computation
of the DFT, of approximately 64 and 205 for 256 points and 1024 point transforms
respectively.
37. How many multiplications and additions are required to compute N-point
DFT using radix-2 FFT? (Nov/Dec 2004)
The number of multiplications and additions required to compute N-point DFT
using radix-2 FFT are N log 2 N and

N
log 2 N
2

respectively.

38. How many multiplications and additions are required to compute N-point
DFT directly? (April/May 2008)
The number of multiplications and additions required to compute N-point DFT u
are N ( N 1) and

N2

respectively.

39. What is the speed improvement factor in calculating 64 point DFT of a

sequence using direct computation and FFT algorithms?
(OR)
Calculate the number of multiplications needed in the calculation of DFT and
FFT with 64-point sequence. (May/June 2009)
The number of complex multiplications required using direct computation is
N 2 64 2 4096

The number of complex multiplications required using FFT is

25

N
64
log 2 N
log 2 64 192
2
2

Speed improvement factor=

4096
21.33
192

40. What is meant by radix-2 FFT?

The FFT algorithm is most efficient in calculating N-point DFT. If the number of
output points N can be expressed as a power of 2, that is N 2 m , where is an integer,
then this algorithm is known as radix-2 FFT algorithm.
41. What is decimation in time algorithm?
The computation of 8 point DFT using radix-2 FFT , involves three stages of
computations. Here N=8=23, therefore r=2 and m=3.
The given 8 point sequence is decimated to 2- point sequences . For each 2
point sequence, the 2-popint DFT is computed . From the result of 2 point DFT the
4 point DFT can be computed. From the result of 4-point DFT , the 8 point DFT
can be computed.
42. What is decimation in frequency algorithm?
It is the popular form of the FFT algorithm. In this the output sequence X(k) is
divided into smaller and smaller subsequences, that is why the name decimation in
frequency.
43. What are the difference between and similarities between DIT and DIF
algorithms? (May/June 2006)
Difference between DIT and DIF:
1) In DIT , the input is bit-reversed while the output is in natural order.
For DIF , the reverse is true, i.e., input is normal order, while the
output bit is reversed. However , both DIT and DIF can go from
normal to shuffled data or vice versa.
2) Considering the butterfly diagram, in DIF , the complex multiplication
takes place after the add subtract operation.
26

Similarities between DIT and DIF:

1) Both algorithms require same number of operations to compute DFT.
2) Both algorithms require bit reversal at some place during
computation.
44. What are the applications of FFT algorithms? (Nov/Dec 2006)
The applications of FFT algorithms includes
(i)

Linear filtering

(ii)

Correlation

(iii)

Spectrum analysis

UNIT III
IIR FILTER DESIGN
1. What are the different types of structures for realization of IIR systems?
The different types of structures for realization of IIR system are

27

(i) Direct form I structure

(ii)Direct form II structure

(iii) Cascade form structure

(iv) Parallel form structure

(v) Lattice ladder form structure.

2. Distinguish between recursive realization and non-recursive realization.
For recursive realization the current output y(n) is a function of past outputs, past
and present inputs. This form corresponds to an Infinite Impulse response (IIR)
digital filter.
For non-recursive realizations current output sample y(n) is a function of only
past and present inputs. This form corresponds to an Finite Impulse response (FIR)
digital filter.
3.How many number of additions, multiplications and memory locations are
required to realize a system H(z) having M zeros and N poles in (a) Direct
form I realization (b) Direct form II realization.
(a) The Direct form I realization requires M+N+1 multiplications, M+N
additions and M+N+1 memory locations.
(b) The Direct form II realization requires M+N+1 multiplications, M+N
additions and the maximum of (M,N) memory locations.
4. What is the main advantage of Direct form- II realizations when compared to
Direct form I realization? (Nov/Dec 2008)
In Direct form II realization, the number of memory locations required is less
than that of Direct form I realization.
5. Define signal flow graph.
A signal flow graph is a graphical representation of the relationship between the
variables of a set of linear difference equations.
6. What is transposed theorem and transposed structure?
The transpose of a structure is defined by the following operations.

28

(i)Reverse the directions of all branches in the signal flow graph.

(ii) Interchange the input and outputs.
(iii)Reverse the roles of all nodes in the flow graph.
(iv)Summing points become branching points
(v)Branching points become summing points
According to transposition theorem if we reverse the directions of all
branch transmittance and interchange the input and output in the flow graph, the
system function remains unchanged.
7. What is Canonic form structure?
The Direct form II realization requires minimum number of delays for the
realization of the system. Hence it is called as Canonic form structure.
8. 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.
9. What is the advantage of cascade realization?
Quantization errors can be minimized if we realize an LTI system in cascade
form.
10. What are the different types of filters based on impulse response?
Based on impulse response the filters are of two types
1. IIR filter
2. FIR filter
The IIR filters are of recursive type, whereby the present output sample depends
on the present input, past inputs samples and output samples.
The FIR filters are of non recursive type whereby the present output sample is
depends on the present input sample and previous input samples.

29

11. What is the general form of IIR filter?

The most general form of IIR filter can be written as
M

H (z)

b z
k 0

1 ak
k 1

12. Give the magnitude of Butterworth filter. What is the effect of varying order
of N on magnitude and phase response?
The magnitude function of the Butterworth filter is given by
1

H ( j )

1
2

2N

N 1, 2 , 3..............

Where N is the order of the filter and c is the cut off frequency. The magnitude
response of the Butterworth filter closely approximates the ideal response as the order
N increases. The phase response becomes more non-linear as N increases.
13. Give any two properties of Butterworth lowpass filters.
1. The magnitude response of the Butterworth filter decreases monotonically as
the frequency increases from 0 to .
2. The magnitude response of the Butterworth filter closely approximates the
ideal response as the order N increases.
3. The Butterworth filters are all pole designs.
4. The poles of the Butterworth filter lies on a circle.
5. At the cut off frequency c , the magnitude of normalized Butterworth filter
is

1
2

14. What is Butterworth approximation?

30

In Butterworth approximation, the error function is selected such that the

magnitude is maximally flat in the origin (i.e., at =0) and monotonically
decreasing with increasing .
15. How the poles of Butterworth transfer function are located in s- plane?
The poles of the normalized Butterworth transfer function symmetrically lies on
an unit circle in s-plane with angular spacing of

16. What is Chebyshev approximation?

In Chebyshev approximation, the approximation function is selected such that the
error is minimized over a prescribed band of frequencies.
17. What is Type 1 Chebyshev approximation?
In type 1 Chebyshev approximation, the error function is selected such that, the
magnitude response is equiripple in the pass band and monotonic in the stop band.
18. What is Type 2 Chebyshev approximation?
In type 2 Chebyshev approximation, the error function is selected such that, the
magnitude response is monotonic in pass band and equiripple in the stop band. The
Type -2 magnitude response is called inverse Chebyshev response.
19. Write the magnitude function of Chebyshev low pass filter?
The magnitude response of Type -1 low pass Chebyshev filter is given by
H a

1

c

1 2 C N2

where

is attenuation constant and

C N

is the Chebyshev polynomial of the first kind of degree N.

31

20. How the order of the filter affects the frequency response of Chebyshev filter.
From the magnitude response of Type -1 Chebyshev filter it can be observed that
the magnitude response approaches the ideal response as the order of the filter is
increased.
21. How will you determine the order N of Chebyshev filter.
The order N of the Chebyshev filter is given by

cosh 1
N

cosh 1

Where
10

0.1 p

100.1 1
s

22. What are the properties of Chebyshev filter?

1. The magnitude response of the Chebyshev filter exhibits in ripple either in
pass band or in the stop band according to the type.
2. The magnitude response approaches the ideal response as the value of N
increases.
3. The Chebyshev type 1 filter are all pole designs.
4. The poles of Chebyshev filter lies on an ellipse.
5. The normalized magnitude function has a value of

1
1 2

at the cutoff

frequency c .
23. Compare the Butterworth and Chebyshev Type -1 filter. (May/June 2009)
Sl.No
Butterworth filter
1.
All pole design
2.
The poles lie on a circle in s-plane
3.
The magnitude response is
maximally flat at the origin and

Chebyshev filter
All pole design
The poles lie on a ellipse in s-plane
The magnitude response is equiripple in
pass band and monotonically decreasing in

32

4.

monotonically decreasing function

the stop band.

of .
The normalized magnitude

The normalized magnitude response has a

response has a value of

5.

1
at the
2

value of

1
1 2

at the cut off frequency

cut off frequency c .

c .

Only few parameters has to be

A large number of parameter has to be

calculated to determine the transfer

calculated to determine the transfer

function.

function.

24. What are the different types of filters based on the frequency response?
The filters can be classified based on frequency response. They are (i) low
pass filter (ii) high pass filter (iii) Band pass filter (iv) Band reject filter.
25. Distinguish between FIR and IIR filter. (Nov/Dec 2008)
Sl.No
1.

FIR filter
These filters can be easily

IIR filter
These filters do not have linear phase.

designed to have perfectly linear

2.

phase.
FIR filters can be realized

IIR filters are easily realized recursively.

3.

recursively and non recursively.

Greater flexibility to control the

Less flexibility , usually limited to specific

4.

shape of their magnitude response.

Error due to round off noise are

kind of filters.
The round off noise in IIR filters are more.

less severe in FIR filters , mainly

because feedback is not used.
26. What are the design techniques of designing FIR filters?
These are three well-known methods for designing FIR filters with linear phase.
These are (1) Windows method

(2) Frequency sampling method (3) Optimal or

minimax design.
27. What do you understand by linear phase response?

33

For a linear phase filter ( ) . The linear phase filter did not alter the shape of
the original signal. If the phase response of the filter is non linear the output signal
may be distorted one. In many cases a linear phase characteristic is required
throughout the passband of the filter to preserve the shape of a given signal with in
the pass band. IIR filter cannot produce a linear phase. The FIR filter can give linear
phase, when the impulse response of the filter is symmetric about its mid point.
28. For what kind of application , the antisymmetrical impulse response can be
used?
The antisymmetrical impulse response can be used to design Hilbert transformers
and differentiators.
29. For what kind of application, 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 , i.e., lowpass, highpass , bandpass and bandreject.
The symmetric impulse response having even number of samples can be used to
design lowpass and bandpass filter.
30. How can you design digital filters from the 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.
3. Transform the transfer function of the analog prototype into an equivalent
digital filter transfer function.
31. Mention any two procedures for digitizing the transfer function of an analog
filter.
The two important procedures for digitizing the transfer function of an analog
filter are
1. Impulse invariance method.
2. Bilinear transformation method.

34

32. What are the requirements for a digital filter to be stable and causal?
i.

The digital transfer function H(z) should be a rational function of z and the
co-efficient of z should be real.

ii.

The poles should lie inside the unit circle in z-plane.

iii.

The number of zeros should be less than or equal to number of poles.

33. What are the requirements for a analog filter to be stable and causal?
i.

The digital transfer function Ha(s) should be a rational function of s and

the co-efficient of s should be real.

ii.

The poles should lie on the left half of s-plane.

iii.

The number of zeros should be less than or equal to number of poles.

iv.
34. What are the advantages and disadvantages of digital filters?
Advantages:
1. High thermal stability due to absence of resistors , inductors and capacitors.
2. The performance characteristics like accuracy, dynamic range, stability and
tolerance can be enhanced by increasing the length of the registers.
3. The digital filters are programmable.
4. Multiplexing and adaptive filtering are possible.
Disadvantages:
1. The bandwidth of the discrete signal is limited by the sampling frequency.
2. The performance of the digital filter depends on the hardware used to
implement the filter.
35. What is impulse invariant transformation?
The transformation of analog filter to digital filter without modifying the impulse
response of the filter is called impulse invariant transformation (i.e., in this
transformation the impulse response of the digital filter will be sampled version of the
impulse response of the analog filter.)

35

36. What is the main objective of impulse invariant transformation?

The objective of this method is to develop an IIR filter transfer function whose
impulse is the sampled version of the impulse response of the analog filter. Therefore
the frequency response characteristics of the analog filter are preserved.
37. Write the impulse invariant transformation used to transform real poles with
and without multiplicity.
The impulse invariant transformation used to transform real poles (at s = - pi)
without multiplicity is
1
1
is transformed to
piT 1
s pi
1 e z

The impulse invariant transformation used to transform multiple real pole

(at s = - pi) is
1
(1) m 1 d m 1
1
is transformed to
m
m 1
piT 1
(m 1) dpi 1 e z
s pi

38. What is the relation between digital and analog frequency in impulse
invariant transformation.
The relation between analog and digital frequency in impulse invariant
transformation is given by
Digital frequency, T
Where,

- Analog frequency and

T - Sampling time period
39. What is bilinear transformation? (April/May 2008)
The Bilinear transformation is a conformal mapping that transforms the s-plane to
z-plane. In this mapping the imaginary axis of s-plane is mapped into the unit circle in
z-plane, the left half of s-plane is mapped into interior of unit circle in z-plane and the

36

right half of s-plane is mapped into exterior of unit circle in z-plane . The Bilinear
mapping is a one to-one mapping and it is accomplished when
s

2 1 z 1
T 1 z1

40. What is the relation between digital and analog frequency in Bilinear
transformation?
In Bilinear transformation, the digital frequency and analog frequency are related
by the equation,
1
Digital frequency, 2 tan

Analog frequency

T
2

or

tan
T
2

Where,

- Analog frequency
T - Sampling time period

41. What is frequency warping? (Nov/Dec 2008)

In bilinear transformation the relation between analog and digital frequencies is
nonlinear. When the s-plane is mapped into z-plane using bilinear transformation, this
nonlinear relationship introduces distortion in frequency axis, which is called
frequency warping.
42. What is prewarping? Why it is employed? (April/May 2008)
In IIR filter design using bilinear transformation, the conversion of the specified
digital frequencies to analog frequencies is called prewarping.
The prewarping is necessary to eliminate the effect of warping on amplitude
response.
43. Define prewarping in IIR filter.

37

In IIR filter design using bilinear transformation the specified digital frequencies
are converted to analog equivalent frequencies, which are called prewarp frequencies.
Using the prewarp frequencies, the analog filter transfer function is designed and then
it is transformed to digital filter transfer function.
44. Compare the impulse invariant and bilinear transformations.
Sl.No Impulse Invariant transformation
1.
It is many to one mapping

Bilinear transformation
It is one to one mapping.

2.

The relation between analog and

The relation between analog and digital

3.

digital frequency is linear.

To prevent the problem of aliasing

frequency is nonlinear.
There is no problem of aliasing and so

the analog filters should be band

the analog filter need not be band

limited.
The magnitude and phase response of

limited.
Due to the effect of warping, the phase

analog filter can be preserved by

response of analog filter cannot be

choosing low sampling time or high

preserved . But the magnitude response

sampling frequency.

can be preserved by prewarping.

4.

UNIT IV
FIR FILTER DESGIN
1. What is the condition for the impulse response of FIR filter to satisfy for
constant group and phase delay and for only constant group delay?
For linear phase FIR filter to have both constant group delay and constant phase
delay.
( )

For satisfying above condition

h ( n ) h ( N 1 n)

that is the impulse response must be symmetrical about n

If one constant group delay is desired then
( )

For satisfying the above condition

38

N 1
2

h ( n) h( N 1 n)

that is the impulse response must be antisymmetrical about n

N 1
2

2. What are the properties of FIR filter?

1. FIR filter is always stable because all its poles are at the origin.
2. A realizable filter can always be obtained.
3. FIR filter has a linear phase response.
3. What are the steps involved in the FIR filter design?
i)

Choose the desired (ideal) frequency response H d ( ) .

ii)

Take inverse Fourier transform of H d ( ) to get hd (n) .

iii)

Convert the infinite duration hd (n) to a finite duration sequence h(n) .

iv)

Take Z transform of h(n) to get the transfer function H (z ) of the

FIR filter.

4. What is the necessary and sufficient condition for the linear phase
characteristic of a FIR filter? (May/June 2006)
The necessary and sufficient condition for the linear phase characteristic of a FIR
filter is that the phase function should be a linear function of

, which in turn

requires constant phase delay or constant group delay.

5. How the constant group delay and phase delay is achieved in linear phase FIR
filters?
Frequency response of FIR filter with constant group and phase delay
H ( ) H ( ) e j ( )

The following conditions have to be satisfied to achieve constant group and phase
delay.
Phase delay,

N 1
( i.e., phase delay is constant )
2

Group delay,

( i.e., group delay is constant)

2

Impulse response, h(n) = - h( N -1 n ) (i.e., impulse response is anti symmetric )

39

6. What are the possible types of impulse response for linear phase FIR filters.
There are four types of impulse response for linear phase FIR filters
1. Symmetric impulse response when N is odd.
2. Symmetric impulse response when N is even.
3. Antisymmetric impulse response when N is odd.
4. Antisymmetric impulse response when N is even.
7. List the well known design techniques for linear phase FIR filter. (May/June
2006)
There are three well known methods of design techniques for linear phase
FIR filters. They are,
i)

Fourier series method and window method.

ii)

Frequency sampling method.

iii)

Optimal filter design method.

8. Write the two concepts that lead to the Fourier series or window method of
designing FIR filters.
The following two concepts lead to the design of FIR filters by Fourier series
method.
i.

The frequency response of a digital filter is periodic with period equal to

sampling frequency.

ii.

Any periodic function can be expressed as a linear combination of

complex exponentials.

9. Write the procedure for designing FIR filter by Fourier series method.
(April/May 2008)
i)

Choose the desired (ideal) frequency response H d ( ) of the filter.

ii)

Evaluate the Fourier series co-efficient of H d ( ) which gives the

desired impulse response hd (n) .
hd (n)

1
2

( )e jn d

40

Truncate the infinite sequence hd (n) to a finite duration

iii)

sequence h(n) .
Take Z transform of h(n) to get a noncausal filter transfer function

iv)

H (z ) of the FIR filter.

Multiply H (z ) by

v)

N 1

to convert noncausal transfer function to a

realizable causal FIR filter transfer function.

H (z ) z

N 1

h(0)

N 1
2

h( n) z
n 1

10. What are the disadvantages of Fourier series method?

In designing FIR filter using Fourier series method the infinite duration impulse
N 1
. Direct truncation of the series will lead to fixed
2

response is truncated at n=

percentage overshoots and undershoots before and after an approximated

discontinuity in the frequency response.
11. What is Gibbs phenomenon? (Nov/Dec 2004)
The way of finding an FIR filter that approximates H (e j ) would be to
N 1
.The abrupt truncation of the series
2

truncate the infinite Fourier series at n=

will lead to oscillation both in passband and in stopband. This phenomenon is known
as Gibbs phenomenon.
12. Write the procedure for designing FIR filter using windows.
i)

Choose the desired (ideal) frequency response H d ( ) of the filter.

ii)

Evaluate the Fourier series co-efficient of H d ( ) which gives the

desired impulse response hd (n) .
1
hd (n)
2

( )e jn d

41

iii)

Choose a window sequence w(n) and multiply the infinite sequence

hd (n) by w(n)to convert the infinite duration impulse response to

finite duration impulse response h(n) .

h( n) hd ( n) w( n)

iv)

Find the transfer function of the realizable FIR filter .

H (z ) z

N 1

h(0)

N 1
2

h( n) z

n 1

13. What are the desirable characteristics of the window? (May/June 2009)
The desirable characteristics of the window are
1. The central lobe of the frequency response of the window should contain most
of the energy and should be narrow.
2. The highest side lobe level of the frequency response should be small.
3. The side lobe of the frequency response should decrease in energy rapidly as

tends to.
14. What is window and why it is necessary? (April/May 2008)
The way of finding an FIR filter that approximates H (e j ) would be to truncate
N 1
. The abrupt truncation of the series will
2

the infinite Fourier series at n=

lead to oscillation both in passband and in stopband. These oscillations can be

reduced through the use of less abrupt truncation of the Fourier series. This can be
achieved by multiplying the infinite impulse response with a finite weighing w(n) ,
called a window.
15. List characteristics of FIR filter designed using windows.
i). The width of the transition band depends on the type of window.
ii).The width of the transition band can be made narrow by increasing the value
Of N where N is the length of the window sequence.

42

iii).The attenuation in the stop band is fixed for a given window , except in case of
Kaiser window where it is variable.
16. Write the procedure for FIR filter design by frequency sampling method.
1. Choose the desired frequency response H d ( ) .
2. Take N samples of H d ( ) to generate the sequence
3. Take inverse DFT of

~
H ( k ).

~
H ( k ).

to get the impulse response h(n).

4. The transfer function H(z) of the filter is obtained by taking Z transform of

impulse response.
17. What is meant by Optimum equiripple design criterion? Why it is followed?
In FIR filter design by Chebyshev approximation technique, the weighted
approximation error between the desired frequency and the actual frequency
response is spread evenly across the passband and stopband.The resulting filter
will have ripples in both the passband and stopband. This concept of design is
called optimum equiripple design criterion.
The optimum equiripple criterion is used to design FIR filter in order to
satisfy the specifications of passband and stopband.
18. Write the expression for frequency response of rectangular window.
The frequency response of rectangular window is given by

WR ( )

N
2

sin
2

sin

19. Write the characteristic features of Rectangular window.

1. The mainlobe width is equal to

4
N

2. The maximum sidelobe magnitude is -13dB.

3. The sidelobe magnitude does not decreases significantly with increasing .
20. List the features of FIR filter designed using rectangular window.
43

v)

The width of the transition region is related to the width of the

mainlobe of window spectrum.

vi)

Gibbs oscillations are noticed in the passband and stopband.

vii)

The attenuation in the stopband is constant and cannot be varied.

21. Give the equation specifying Hanning windows. (Nov/Dec 2010)

The equation for Hanning window is given by
wH n (n) 0.5 0.5 cos

2n
N 1

for

=0

( N 1)
( N 1)
n
2
2

Otherwise.

22. Give the equation specifying Hamming windows. (Nov/Dec 2004)

The equation for Hamming window is given by
wH (n) 0.54 0.46 cos

2n
N 1

=0

for

( N 1)
( N 1)
n
2
2

Otherwise.

23. Give the equation specifying Blackman windows. (Nov/Dec 2010)

The equation for Blackman window is given by

wB (n) 0.42 0.5 cos

2n
4n
0.08 cos
N 1
N 1

for

=0

( N 1)
( N 1)
n
2
2

Otherwise.

24. Give the equation specifying Bartlett windows. (Nov/Dec 2004)

The equation for Bartlett window is given by
wT ( n) 1

2n
N 1

for

=0

( N 1)
( N 1)
n
2
2

Otherwise.

25. Give the equation specifying Kaiser windows.

44

The equation for Bartlett window is given by

2n

N 1
I 0 ( )

wk ( n) I 0

for n

=0
Where

( N 1)
2

Otherwise.

is an independent parameter.
I 0 (x) is the zeroth order Bessel function of the first kind
1 x

k 1 k! 2

I 0 ( x) 1

26. Write the characteristics features of Triangular window.

8
.
N

i)

The mainlobe width is equal to

ii)

The maximum sidelobe magnitude is -25dB.

iii)

The sidelobe magnitude slightly decreases with increasing

27. Why the triangular window is not good a good choice for designing FIR
filters?
In FIR filters designed using triangular window the transition from passband to
stopband is not sharp and the attenuation in stopband is less when compared to filters
designed with rectangular window. For the above two reasons the triangular window
is not a good choice.
28. List the features of hanning window spectrum.
i) The mainlobe width is equal to

8
.
N

ii) The maximum sidelobe magnitude is -31dB.

iii) The sidelobe magnitude slightly decreases with increasing
29. List the features of hamming window spectrum.

45

i) The mainlobe width is equal to

8
.
N

ii) The maximum sidelobe magnitude is -41dB.

iii) The sidelobe magnitude remains constant for increasing .
30. Compare the Rectangular and Hanning window.
Sl.No
Rectangular window
1.
The width of mainlobe in window
spectrum is

Hanning window
The width of mainlobe in window

4
N

spectrum is

8
N

2.

The maximum sidelobe magnitude

The maximum sidelobe magnitude in

3.

in window spectrum is -13dB.

In window spectrum the sidelobe

window spectrum is -31dB.

In window spectrum the sidelobe

magnitude slightly decreases with

magnitude decreases with increasing

increasing
In FIR filter designed using

In FIR filter designed using Hanning

rectangular window the minimum

window the minimum stopband

stopband attenuation is 22dB.

attenuation is 44dB.

4.

31. Compare the Rectangular and Hamming window.

Sl.No
Rectangular window
1.
The width of mainlobe in window
spectrum is

Hamming window
The width of mainlobe in window

4
N

spectrum is

8
N

2.

The maximum sidelobe magnitude

The maximum sidelobe magnitude in

3.

in window spectrum is -13dB.

In window spectrum the sidelobe

window spectrum is -41dB.

In window spectrum the sidelobe

magnitude slightly decreases with

magnitude remains constant.

increasing
In FIR filter designed using

In FIR filter designed using Hamming

rectangular window the minimum

window the minimum stopband

stopband attenuation is 22dB.

attenuation is 51dB.

4.

32. Compare the Rectangular and Hamming window.

46

Sl.No
Hanning window
1.
The width of mainlobe in window
spectrum is

Hamming window
The width of mainlobe in window

8
N

spectrum is

8
N

2.

The maximum sidelobe magnitude

The maximum sidelobe magnitude in

3.

in window spectrum is -31dB.

In window spectrum the sidelobe

window spectrum is -41dB.

In window spectrum the sidelobe

magnitude decreases with

magnitude remains constant. Here the

increasing

increased sidelobe attenuation is achieved

at the expense of constant attenuation at

4.

In FIR filter designed using

high frequencies.
In FIR filter designed using Hamming

Hanning window the minimum

window the minimum stopband

stopband attenuation is 44dB.

attenuation is 51dB.

33. Compare the Hamming and Blackman window.

Sl.No
Hamming window
1.
The width of mainlobe in window
spectrum is

Blackman window
The width of mainlobe in window

8
N

spectrum is

12
N

2.

The maximum sidelobe magnitude

The maximum sidelobe magnitude in

3.

in window spectrum is -41dB.

In window spectrum the sidelobe

window spectrum is -58dB.

In window spectrum the sidelobe

magnitude remains constant with

magnitude decreases rapidly with

increasing
In FIR filter designed using

increasing
In FIR filter designed using Blackman

Hamming window the minimum

window the minimum stopband

stopband attenuation is 51dB.

The higher value of sidelobe

attenuation is 78dB.
The higher value of sidelobe attenuation is

attenuation is achieved at the

achieved at the expense of increased

expense of constant attenuation at

mainlobe width.

4.

5.

high frequencies.

47

34. List the features of Blackman window spectrum.

i) The mainlobe width is equal to

12
.
N

ii) The maximum sidelobe magnitude is -58dB.

iii) The sidelobe magnitude slightly decreases with increasing
iv)

The higher value of sidelobe attenuation is achieved at the expense

of increased mainlobe width.

35. List the features of Kaiser Window spectrum.

i) The width of mainlobe and the peak sidelobe are variable.
ii) The parameter in the Kaiser window function , is an independent
variable that can be varied to control the sidelobe levels with respect to
mainlobe peak.
iii) The width of the mainlobe in the window spectrum ( and so the
transition region in the filter) can be varied by varying the length N of the
window sequence.
36. Compare the Hamming and Kaiser window.
Sl.No
Hamming window
1.
The width of mainlobe in window
spectrum is
2.

3.

4.

Kaiser window
The width of mainlobe in window
spectrum depends on the values of and

8
N

The maximum sidelobe magnitude

N.
The maximum sidelobe magnitude with

in window spectrum is -41dB.

respect to peak of mainlobe is variable

In window spectrum the sidelobe

using the parameter .

In window spectrum the sidelobe

magnitude remains constant with

magnitude decreases with increasing

increasing
In FIR filter designed using

In FIR filter designed using Kaiser

Hamming window the minimum

window the minimum stopband

stopband attenuation is 51dB.

attenuation is variable and depends on the

value of .

48

UNIT V
APPLICATIONS
1. What are the classification digital signal processors?
The digital signal processors are classified as
i.

General purpose digital signal processors.

ii.

Special purpose digital signal processors.

2. Give some examples for fixed point DSPs.

(a) TMS320C50
(b) TM 320C54
(c) TM 320C55
(d) ADSP-219x
(e) ADSP-219xx
3. Give some examples for floating point DSPs.
a) TMS320C3x
b) TMS320C67x
c) ADSP-21xxx.
4. What are the factors that influence selection of DSPs?
1. Architectural features
2. Execution speed
3. Type of arithmetic
4. Word length
5. What are the applications of PDSPs?

49

Digital cell phones, automated inspection, voicemail, motor control, video

conferencing, Noise cancellation, Medical imaging, speech synthesis, satellite
communication etc.
6. What are the advantages and disadvantages of VLIW architecture?
Advantages:
1. Increased performance
2. Better compiler targets
3. Potentially scalable
4. Potentially easier to program
5. Can add more execution units, allow more instruction to be
packed into the VLIW instruction.
Disadvantages:
1. New kind of programmer/compiler complexity
2. Program must keep track of instruction scheduling
3. Increased memory use
4. High power consumption
5. Misleading MIPS ratings
7. What is pipelining?
Pipelining a processor means breaking down its instruction into a
series of discrete pipeline stages which can be completed in sequence by
specialized hardware.
8. What is pipeline depth?
The number of pipeline stages is referred to as the pipeline depth.
9. What is the pipeline depth of TMS320C50, TM 320C54x?
TMS320C50 4
TM 320C54x 6

50

10. What are the different stages in pipelining?

i. The Fetch phase
ii. The Decode phase
iii. Memory read phase
iv. The Execute phase
11. What are the different buses of TM 320C5x and their functions?
The C5x architecture has four buses
1. Program bus (PB)
2. Program address bus (PAB)
3. Data read bus (DB)
4. Data read address bus (DAB)
The program bus carriers the instruction code and immediate operands from
program memory to the CPU.
The program address bus provides address to program memory space for both
read and write.
The data read bus interconnects various elements of the CPU to data memory
spaces.
The data read address bus provides the address to access the data memory spaces.
12. List the various registers used with ARAU.
Eight auxiliary registers (AR0 AR7)
Auxiliary register pointer (ARP)
13. What are the elements that the control processing unit of c5X consists of
1.Central arithmetic logic unit (CALU)
2.Parallel logic unit (PLU)
3.Auxiliary register arithmetic unit (ARAU)
4.Memory mapped registers
5.Program controller

51

14. What is the function of parallel logic unit?

The parallel logic unit is second logic units, which execute logic operations on
data without affecting the contents of accumulator.
15. List the on chip peripherals in c5x.
The on-chip peripherals interfaces connected to the c5x CPU include
(i)

Clock generator

(ii)

Hardware timer

(iii)

Software programmable wait state generators

(iv)

General purpose I/O pins

(v)

Parallel I/O ports

(vi)

Serial port interface

(vii)

Buffered serial port

(viii)

Time-divisions multiplexed (TDM) serial port

(ix)

Host port interface

(x)

User unmaskable interrupts

16. What are the general purpose I/O pins?

Branch control input

BIO

External flag (XF)

17. What are the arithmetic instructions of C5x?
ADD, ADDB, ADDC, SUB, SUBB, MPY, MPYU
18. What are the logical instructions of c5x?
AND, ANDB, OR, ORB, XOR, XORB
19. What are the shift instructions?
ROR, ROL, ROLB, RORB, BSAR

52

20. What are load/store instructions?

LACB, LACC, LACL, LAMM, LAR, SACB, SACH, SACL, SAR
21. What are the different types of arithmetic in digital systems?
There are three types of arithmetic used in digital systems
1.Fixed point arithmetic
2.Floating point arithmetic
3.Block Floating arithmetic
22. What do you understand by a fixed-point number?
In fixed-point arithmetic the positions of the binary point is fixed. The bits to the
right represent the fractional part of the number and those to the left represent the
integer part. For example, the binary number 01.1100 has the value 1.75 in decimal.
23. What is meant by block floating point representation? What are its
advantages?
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
with in the block uses 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.
24. What are the advantages of floating point arithmetic?
1.Larger dynamic range
2.Overflow in floating point representation is unlikely.
25. Compare the fixed point and floating point arithmetic. (Nov/Dec 2006)
Fixed Point Arithmetic
1.Fast Operation
2.Relatively economical

Floating Point Arithmetic

Slow Operation
More expensive because of costlier

3.Small dynamic range

4.Round off error occur only for

hardware
Increased dynamic range
Round off errors can occur with both

additions
5.Overflow occur in addition

additions and multiplication

Overflow does not arise
53

6.Used in small computers

Used in larger, general purpose

computers
26. What are the three quantization errors due to finite word length registers in
digital filters?
1.Input quantization error
2.Coefficient quantization error
3.Product quantization error
27. How the multiplication and addition are carried out in floating point
arithmetic?
In floating point arithmetic, multiplications are carried out as follows
Let f1=M1 x 2c1 and f2 = M2 x 2c2. Then f3=f1x f2 = (M1xM2)2(c1+c2)
That is, mantissas are multiplied using fixed point arithmetic and the exponents
are added.
The sum of two floating point numbers is carried out by shifting the bits of the
mantissa of the smaller number to the right until the exponents of the two number are
equal and then adding the mantissas.
28. Brief on coefficient inaccuracy. (or)
What is coefficient quantization error? What is its effect?
The filter coefficients are computed to infinite precision in theory. But, in digital
computation the filter coefficients are represented in binary and are stored in registers.
If a b bit register is used, the filter coefficients must be rounded or truncated to b
bits, which produces an error.
Due to quantization of coefficients, the frequency response of the filter may differ
appreciably from the desired response and some times the filter may actually fail to
meet the desired response and some times the filter may actually fail to meet the
desired specifications. If the poles of desired filter are close to the unit circle, then
those of the filter with quantized coefficients may lie just outside the unit circle,
leading to unstability

54

29. What is product quantization error (or) what is product round off error in
DSP? (Nov/Dec 2010)
Product quantization errors arise at the output of a multiplier. Multiplication of a b
bit data with a b bit coefficient results a product having 2b bits. Since a b bit register
is used, the multiplier output must be rounded or truncated to b bits, which produces
an error. This error is known as product quantization error.
30 What do you understand by input quantization error?
In DSP, the continuous time input signals are converted into digital using a b bit
ADC. The representation of continuous signal amplitude by a fixed digit produce an
error, which is known as input quantization error.
31. What is truncation? (Nov/Dec 2006)
Truncation is process of reducing the size of binary number by discarding all bits
less significant than the least significant bit that is retained. (In the truncation of a
binary number to b bits all the less significant bits beyond bth bit are discarded)
32. What is rounding?
Round is the process of reducing the size of a binary number to finite word sizes
of b bits such that, the rounded b-bit number is closest to the original unquantized
number.
33. What is Quantization step size? (Nov/Dec 2006)
In digital systems, the numbers are represented in binary. With b-bit binary we
can generate 2b different binary codes. Any range of analog value to be represented in
binary should be divided into 2b levels with equal increment . The 2b levels are called
Quantization levels and the increment in each level is called Quantization step size . It
R is the range of analog signal then,
Quantization step size, q = R/2b.

55

34. What are limit cycle oscillations? (Nov/Dec 2008)

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 cycle.
35. What is zero input limit cycle?
In recursive system, the product Quantization may create periodic oscillations in
the output. These oscillations are called limit cycles. If the 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 also called zero input limit cycles.
36. What is dead band?
In a limit cycle the amplitudes of the output are confined to a range of values,
which is called dead band of the filter.
37. 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.
38. What is overflow limit cycle?
In fixed point addition the overflow occurs when the sum exceeds the finite word
length of the register used to store the sum. The overflow in addition may lead to
oscillations in the output which is called overflow limit cycle.
39. How overflow limit cycle can be eliminated? (May/June 2009)
The overflow limit cycles can be eliminated either by using saturation arithmetic
or by scaling the input signal to the adder.

56

57