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# KONGUNADU COLLEGE OF ENGINEERING AND TECHNOLOGY

## NAMAKKAL- TRICHY MAIN ROAD, THOTTIAM.

DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

EC2302 DIGITAL SIGNAL PROCESSING

SEMESTER V

PREPARED BY
R.SHANKAR, AP/ECE

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UNIT I
DISCRETE FOURIER TRANSFORM

1. Define DFT and IDFT (or) whatthe analysis and synthesis equations are of
DFT?

DFT(Analysis Equation)

N-1 nk

X(k)= x(n) W
N
, W
N
= e
-j2/N

n=0

IDFT(Synthesis Equation)

N-1 - nk

x(n)= 1/N X(k) W
N
, W
N
= e
-j2/N

k=0

2. State the properties of DFT.

1) Periodicity
2) Linearity and symmetry
3) Multiplication of two DFTs
4) Circular convolution
5) Time reversal
6) Circular time shift and frequency shift
7) Complex conjugate
8)Circular correlation

3. Define circular convolution.
Let x
1
(n) and x
2
(n) are finite duration sequences both of length N with DFTs X
1
(k) and X
2
(k)
If X
3
(k)=X
1
(k)X
2
(k) then the sequence x
3
(n) can be obtained by circular convolution defined as

N-1

x
3
(n)= x
1
(m)x
2
((n-m))
N
m=0

4. Compute the DFT of x(n) = (n-n0)
Given x(n) = (n-n0) = 1, when n=n0
= 0, otherwise
N-1
X (k) = x(n) e j2kn / N k = 0,1,2,3.N-1
n=0
N-1
X (k) = x(n) e
j2kn0/ N
k = 0,1,2,3.N-1
n=0
= e
j2kn0/ N
5. Find out the DFT of the signal X(n)= (n)
() {

X(n)={1,0,0,0}

() ()

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() ()

() () ()

()

()

X(k)={1,1,1,1}

6. IF N-point sequence x(n) has N- point DFT X(k) then what is the DFT of the following?
N j
N
e n x iv l n x iii n N x ii n x i
ln/ 2
) ( ) ( )) (( ) ( ) ( ) ( ) ( ) (
t

- -

Solution:

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

- -
- -
t
t

7. Calculate the DFT of the sequence x(n) = 16
4
1
=
|
.
|

\
|
forN
n

Solution:

=
1
0
/ 2
) ( ) (
N
n
N kn j
e n x k X
t
K=0, 1 , 2..N-1

8 /
2
16
15
0
8 /
16 / 2
15
0
4
1
1
4
1
1
4
1
4
1
k j
k j
n
n
k j
kn j
n
n
e
e
e
e
t
t
t
t

|
.
|

\
|

=
|
.
|

\
|
=
|
.
|

\
|
=

8. 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
] = ) (
/ 2
k X e
N kn j t

9. 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)
N
N j
l k X e )) (( ]
ln/ 2
=
t

10. Define Twiddle factor (or) Phase factor.
The complex number W
N
is called as Twiddle factor (or) Phase factor. It also represents an Nth
root of unity. It is expressed by W
N
kn
= e
j2kn / N

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11. What are the properties of Twiddle factor (W
N
K
)?
1. Periodicity Property
W
N
K+N
= W
N
K

2. Symmetric Property
W
N
K+N/2
= -W
N
K
12. Find the DFT of the sequence x(n) = { 1,1,0,0 }
Solution:

=
1
0
/ 2
) ( ) (
N
n
N kn j
e n x k X
t
; k = 0,1,2,..N-1.
3
2 / 4
0
3
0
3
/ 2
0
3
0
3
3 / 2
0
( ) 0,1, 2 , 3.
(0) ( ) {1 1 0 0} 2
(1) ( ) {1 0 0} 1
(2) ( ) {1 1 0 0} 0
(3) ( ) {1 0 0} 1
( ) {2,1 , 0, 1 }
j kn
n
n
j n
n
j n
n
j n
n
x n e k
X x n
X x n e j j
X x n e
X x n e j j
X k j j
t
t
t
t

=
=

=
= =
= = + + + =
= = + + =
= = + + =
= = + + + = +
= +

13. Find DFT for {1,0,0,1}.

14. Distinguish between DFT and DTFT.

Sl.No DFT DTFT
1. Obtained by performing sampling
operation in both the time and
frequency domains.
Sampling is performed only in time
domain.
2. Discrete frequency spectrum Continuous function of e

15. Determine the DIFT of a sequence x(n) = a
n
u(n)

X(K) = x(n) e
j2kn/N

The given sequence x(n) = a
n
u(n)
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DTFT{x(n)} = x(n) e
j2kn/N
= (a e
j2k/N
)
n

Where a
n
= 1-a
n
/(1-a)
X(K) = (1 a
N
e
j2k
)/ (1-ae
j2k/N
)
16. Find the Fourier transform of a sequence x(n) = 1 for 2 2 s s n
= 0 otherwise.
Solution:

=

= =
2
2
) ( ) (
n
n j
n
n j
e e n x X
e e
e

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

+ + + + =
e e 2 cos 2 cos 2 1 + + =

17. Define Fourier transform of a discrete time signal.
The Fourier transform of a discrete time signal x(n) is defined as

{ }

= =
n
n j
e n x X n x F
e
e ) ( ) ( ) (
18. Why FT 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.

19. 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 ) 2 0 ( t t t to or to + , and also it is periodic
with periodicity of 2 t .
ii. The FT of analog signals involves integration but FT of discrete time signals involves
summation.
20. Define inverse Fourier transform.
The inverse Fourier transform of X(e) is defined as
{ } e e
t
e
e
t
t
d e X n x X F
n j
}

= = ) (
2
1
) ( ) (
1

21. Give some applications of Fourier transform.
The applications of Fourier transform are
1. The frequency response of LTI system is given by the Fourier transform of the impulse
response of the system.
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.
22. 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(e).

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23. 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 e with a period of 2t .

ii) If h(n) is real then ) (e H is symmetric and ) (e H Z is antisymmetric.

iii) If h(n) is complex then the real part of ) (e H is antisymmteric over the interval t e 2 0 s s .

iv) The frequency response is a continuous function of e .

24. Write short notes on the frequency response of first order system.
A first order system is characterized by the difference equation

) 1 ( ) ( ) ( + = n ay n x n y

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.

25. Write a short note on the frequency response of second order system.
A second order system is characterized by the difference equation

) 1 ( cos ) ( ) 2 ( ) 1 ( cos 2 ) (
0
2
0
+ = n x r n x n y r n y r n y e e
The frequency response of second order system depends on the parameters r and
0
e 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
e . When the value of r is varied from 0 to
1 , the sharpness of resonant peak increases.
26. Define discrete Fourier series.
Consider a sequence x
p
(n) with a period of N samples so that x
p
(n) = x
p
(n/N); Then the discrete
Fourier series of the sequence x
p
(n) is defined as

=
1
0
/ 2
) ( ) (
N
n
N kn j
p p
e n x k X
t

27. How to obtain the output sequence of linear convolution through circularconvolution?
Consider two finite duration sequences x(n) and h(n) of duration L samples and Msamples.
The linear convolution of these two sequences produces an output sequenceof 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 ofsamples in circularconvolution 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.
28. Define circular convolution.
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.
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=
=
=
1
0
2 1 3
2 1 2 1
)) (( ) ( ) (
)} ( ) ( { ) ( ) (
N
m
N
m n x m x n x
n x n x DFT k X k X

29. How the circular convolution is obtained by using Graphical method?
Given two sequences ) (
1
n x and ) (
2
n x , the circular convolution of these two sequences
) ( ) ( ) (
2 1 3
n Nx n x n x = can be obtained by using the following steps.
1. Graph N samples of ) (
1
n x , as equally spaced points around an outer circle in counter
clockwise direction.
2. Start at the same point as ) (
1
n x graph N samples of ) (
2
n x 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.

30. What is zero padding? What are its uses?
Let the sequence x (n) has a length L. If we want to find the N-point DFT(N>L) of the
sequence x(n), we have to add (N-L) zeros to the sequence x(n). This is known as zero padding.
The process of lengthening the sequence by adding zero valued samples is called appending with
Appending zeros to a sequence in order to increase the size or length of the sequence is called zero
The uses of zero padding are

1. We can get better display of the frequency spectrum.
2. With zero padding the DFT can be used in linear filtering.

31. Define sectional convolution.
If the data sequence x(n) is of long duration it is very difficult to obtain the output sequence y(n) due
to limited memory of a digital computer. Therefore, the data sequence is divided up into smaller
sections. These sections are processed separately one at a time and controlled later to get the output.
32. What are the steps involved in circular convolution?
The circular convolution involves 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.
4. Summing the values of product sequence.
33. What are the different methods performing circular convolution?
1. Graphical method
2. Stockhmans method
3. Tabular array method
4. Matrix method.

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34. 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.

{ } 1 , 2 , 3 ) (
1
2
3
1
2
1
1 2 2
2 1 2
2 2 1
) 2 (
) 1 (
) 0 (
) 2 (
) 1 (
) 0 (
) 0 ( ) 1 ( ) 2 (
) 2 ( ) 0 ( ) 1 (
) 1 ( ) 2 ( ) 0 (
=
(
(
(

=
(
(
(

(
(
(

(
(
(

=
(
(
(

(
(
(

n y
y
y
y
x
x
x
h h h
h h h
h h h

35. What are the two methods used for the sectional convolution?
The two methods used for the sectional convolution are
2) Overlap-save method.
In this method the size of the input data block x
i
(n) is L. To each data block we append M-1 zeros
and perform N point circular convolution of x
i
(n) and h(n). Since each data block is terminated
with M-1 zeros the last M-1 points from each output block must be overlapped and added to first
M-1 points of the succeeding blocks. This method is called overlap-add method.
Or

In this method the longer sequence sequences is sectioned into sequence of size equal to smaller
sequences. Then circular convolution of each section of longer sequence and smaller sequence is
performed. The overall output sequence is obtained by combining the output of the sectioned
convolution.
37. What is overlap-save method?
In this method the data sequence is divided into N point sections x
i
(n).Each section contains the
last M-1 data points of the previous section followed by L new data points to form a data sequence of
length N=L+M-1.In circular convolution of x
i
(n) with h(n) the first M-1 points will not agree with the
linear convolution of x
i
(n) and h(n) because of aliasing, the remaining points will agree with linear
convolution. Hence we discard the first (M-1) points of filtered section x
i
(n) N h(n). This process is
repeated for all sections and the filtered sections are abutted together.
Or
In this method the longer sequence sequences is sectioned into sequence of size equal to smaller
sequences. The number of samples that will be obtained in the output of circular convolution of each
section is determined. Then each section of longer sequence is converted to the size of output
sequence using the samples of original longer sequence.

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38. Differentiate (i) overlap-add method (ii) overlap save method.
Sl.No Overlap add method Overlap save method
1. In this method the size of the input
data block is N=L+M-1
In this method the size of the input
data block is L.
2. Each data block consists of the last
M-1 data points of the previous data
followed by the L new data points.
Each data block is L points and we
appended M-1 zeros to compute N-
point DFT.
3. In each output block M-1 points are
corrupted due to aliasing, as circular
convolution is employed.
In this no corruption due to aliasing as
linear convolution is performed using
circular convolution.
4. To form the output sequence the first
M-1 data points are discarded in each
output block and the remaining datas
are fitted together.
To form the output sequence, the last
M-1 points from each output block is
added to the first (M-1) points of the
succeeding block.
39. Why FFT is needed?
The direct evaluation DFT requires N
2
complex multiplications and N
2
Thus for large values of N direct evaluation of the DFT is difficult. By using FFT algorithm the
number of complex computations can be reduced. So we use FFT.\
FFT is needed to compute DFT with reduced number of calculations. DFT is required for spectrum
analysis and filtering operations on the signals u sing digital computers.

40. What is FFT?
FFT is a method for computing the DFT with reduced number of calculations usingsymmetry and
periodicity properties of twiddle factor W
N
k
. The computational efficiency is achieved by
employing divide and conquers approach. This is based on the decomposing of an N-point DFT
into successively smaller DFTs to increase the speed of computation.

41. What is meant by bit reversal and in place commutation as applied to FFT?
"Bit reversal" is just what it sounds like: reversing the bits in a binary word from left to write.
Therefore the MSB's become LSB's and the LSB's become MSB's.The data ordering required by
radix-2 FFT's turns out to be in "bit reversed" order, so bit-reversed indexes are used to combine FFT
stages.
Input sample
index
Binary
Representation
Bit reversed
binary
Bit reversal
sample index
0 000 000 0
1 001 100 4
2 010 010 2
3 011 110 6
4 100 001 1
5 101 101 5
6 110 011 3
7 111 111 7

42. How many multiplications and additions are required to compute N point DFT using redix-2
FFT?
The number of multiplications and additions required to compute N point DFT using radix-2 FFT are
N log
2
N and N/2 log
2
N respectively,.

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43. Calculate the number of multiplications needed in the calculation of DFT and FFT with 64-
Point sequence.
The number of complex multiplications required using direct computation is N
2
= 64
2
= 4096
.
The number of complex multiplications required using FFT is (N/2) log2 N = (64/2) log
2
64 = 192

44. Calculate the number of multiplications needed in the calculation of DFT using FFT with
32- Point sequence.
The number of complex multiplications required using FFT is (N/2) log2N = (32/2) log
2
32 = 80

45. 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 M is an integer, then this algorithm is known as

46. Draw the basic butterfly diagram for radix 2 DIT-FFT and DIF-FFT.
Butterfly Structure for DIT FFT-The DIT structure can be expressed as a butterfly diagram

The DIF structure expressed as a butterfly diagram

47. What is DIT algorithm?
Decimation-In-Time algorithm is used to calculate the DFT of a N point sequence. The idea is to break
the N point sequence into two sequences, the DFTs of which can be combined to give the DFT of the
original N point sequence. This algorithm is called DIT because the sequence x(n) is often spitted into
smaller sub- sequences.

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48. Draw radix 4 butterfly structure for (DIT) FFT algorithm.

49. What DIF algorithm?
It is a popular form of the FFT algorithm. In this the output sequence X(k) is divided into smaller and
smaller sub-sequences , that is why the name Decimation In Frequency.

50. What are the advantages of FFT Algorithm over direct computation of DFT?
- Reduces the computation time required by DFT.
- The number of complex multiplications required using direct computation is N
2
.
- The number of complex multiplications required using FFT is (N/2) log2 N
- Speed calculation
51. What are the applications of FFT algorithm?
The applications of FFT algorithm includes
1. Linear filtering
2. Correlation
3. Spectrum analysis
52. Why the computations in FFT algorithm is said to be in place?
Once the butterfly operation is performed on a pair of complex numbers (a,b) to produce (A,B), there
is no need to save the input pair. We can store the result (A,B) in the same locations as (a,b). Since the
same storage locations are used throughout the computation we say that the computations are done in
place.
53. What are the types of convolution?
Convolution is classified into two types, there are
1. Linear Convolution
2. Circular Convolution
3. Sectioned Convolution

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54. Distinguish between linear convolution and circular convolution of two sequences.
No. Linear convolution Circular convolution
1 If x(n) is a sequence of L number of
samples and h(n) with M number of
samples, after convolution y(n) will have
N=L+M-1 samples.
If x(n) is a sequence of L number of samples
and h(n) with M samples, after convolution
y(n) will have N=max(L,M) samples.
2 It can be used to find the response of a
linear filter.
It cannot be used to find the response of a
filter.
3 Zero padding is not necessary to find the
response of a linear filter.
Zero padding is necessary to find the response
of a filter.

55. What are the differences and similarities between DIF and DIT algorithms?
Differences:
1)The input is bit reversed while the output is in natural order for DIT, whereas for DIF the output is
bit reversed while the input is in natural order.
2)The DIF butterfly is slightly different from the DIT butterfly, the difference being that the complex
multiplication takes place after the add-subtract operation in DIF.
Similarities:
Both algorithms require same number of operations to compute the DFT.Both algorithms can
be done in place and both need to perform bit reversal at some place during the computation.

1. The Time domain sequence is decimated. 1. The Frequency domain sequence is
decimated.
2. The input sequences are arranged in bit
reversed Order.
2. The input sequences are arranged in natural
Order.
3. The output sequences are arranged in
natural
Order.
3. The output sequences are arranged in bit
Reversed order.
4. In each stage of computations, The twiddle
factors are multiplied before the add and
subtract operation .
4. In each stage of computations, The twiddle
factors are multiplied after the add and
subtract operation.
5. The value of N should expressed such that
N=2
m
and this algorithm consists of m
stages of computations.
5. The value of N should expressed such that
N=2
m
and this algorithm consists of m stages
of computations.
6. Total number of arithmetic operations
are N log
2
log
2
N complex multiplications.
6. Total number of arithmetic operations
are N log
2
N complex additions and N/2 log
2
N
complex multiplications.
56. What is aliasing?
If we operate the sampler at fx<fm, the frequency components of the frequency spectrum will
overlap with each other i.e., the lower frequency of the second frequency component will overlap with
higher frequency of the first frequency component. This overlapping effect is called as Aliasing effect.
For avoiding overlapping of high and low frequency components, we have to use low-pass filter to cut
the unwanted high frequency components.
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UNIT II
INFINITE IMPULSE RESPONSE DIGITAL FILTERS

1. 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.
2. What is the general form of IIR filter?
The most general form of IIR filter can be written as

=
=

+
=
N
k
k
M
k
k
k
a
z b
z H
1
0
1
) (

3. 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

.... .......... 3 , 2 , 1
1
1
) (
2
1
2
=
(
(

|
|
.
|

\
|
O
O
+
= O N j H
N
c

Where N is the order of the filter and
c
O 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.
4. Compare digital and analog filter.
Digital filter Analog filter
1. Operates on digital samples. 1. Operates on analog samples.
2. It is governed by linear difference
equation.
2. It is governed by linear differential
equation.
3. It consists of adder, multipliers and
delays implemented in digital logic.
3. It consists of electrical components
like resistors, capacitors and inductors.
4. The coefficients are designed to
satisfy the desired frequency
response.
4. The approximation problem is solved
to satisfy the desired frequency
response.

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5. Give any two properties of Butterworth lowpass filters.
- The magnitude response of the Butterworth filter decreases monotonically as the frequency O
increases from 0 to
- The magnitude response of the Butterworth filter closely approximates the ideal response as
the order N increases
- The Butterworth filters are all pole designs
- The poles of the Butterworth filter lies on a circle
- At the cut off frequency
c
O , the magnitude of normalized Butterworth filter I
2
1

6. What is Butterworth approximation?
In Butterworth approximation, the error function is selected such that the magnitude is maximally flat
in the origin (i.e., at O=0) and monotonically decreasing with increasingO.
7. How the poles of Butterworth transfer function are located in s- plane?
The poles of the normalized Butterworth transfer function symmetrically lies on a unit circle in s-
plane with angular spacing of
N
t
.
8. What is Chebyshev approximation?
In Chebyshev approximation, the approximation function is selected such that the error is minimized
over a prescribed band of frequencies.

9. 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.
10. 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.

11. Write the magnitude function of Chebyshev lowpass filter.
The magnitude response of Type -1 lowpass Chebyshev filter is given by
( )
|
|
.
|

\
|
O
O
+
= O
c
N
a
C
H
2 2
1
1
c

Where
c is attenuation constant and

|
|
.
|

\
|
O
O
c
N
C is the Chebyshev polynomial of the first kind of degree N

12. 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.

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13. Compare the Butterworth and Chebyshev Type -1 filter.
Sl.No Butterworth filter Chebyshev filter
1. All pole design All pole design
2. The poles lie on a circle in s-
plane
The poles lie on a ellipse in s-plane
3. The magnitude response is
maximally flat at the origin and
monotonically decreasing
function of O.
The magnitude response is equiripple in
pass band and monotonically decreasing
in the stop band.
4. The normalized magnitude
response has a value of
2
1
at
the cut off frequency
c
O .
The normalized magnitude response has
a value of
2
1
1
c +
at the cut off
frequency
c
O .
5. Only few parameters have to be
calculated to determine the
transfer function.
A large number of parameter has to be
calculated to determine the transfer
function.

14. 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 and (iv)Band reject filter.
15. Distinguish between FIR and IIR filter.
Sl.No FIR filter IIR filter
1. These filters can be easily
designed to have perfectly linear
phase.
These filters do not have linear phase.
2. FIR filters can be realized
recursively and non
recursively.
IIR filters are easily realized recursively.
3. Greater flexibility to control the
shape of their magnitude
response.
Less flexibility, usually limited to
specific kind of filters.
4. Error due to round off noise is
less severe in FIR filters, mainly
because feedback is not used.
The round off noise in IIR filters is more.

16. How will you determine the order N of Chebyshev filter.
The order N of the Chebyshev filter is given by
|
|
.
|

\
|
O
O
|
.
|

\
|
=

p
s
N
1
1
cosh
cosh
c

Where
1 10
1 . 0
=
p
o
1 10
1 . 0
=
s
o
c

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17. What are the properties of Chebyshev filter?
The properties of Chebyshev filters are
- The magnitude response of the Chebyshev filter exhibits in ripple either in pass band or in the stop
band according to the type
- The magnitude response approaches the ideal response as the value of N increases
- The Chebyshev type 1 filter are all pole designs
- The poles of Chebyshev filter lies on an ellipse
- The normalized magnitude function has a value of
2
1
1
c +
at the cutoff frequency
c
O
18. How can you design digital filters from the analog filters?
The designs of analog filters to digital filters are
- Map the desired digital filter specifications into those for an equivalent analog filter
- Derive the analog transfer function for the analog prototype
- Transform the transfer function of the analog prototype into an equivalent digital filter transfer
function

19. 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
20. What are the requirements for a digital filter to be stable and causal?
The requirements of digital filter to be stable and causal are

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

21. What are the requirements for aanalog filter to be stable and causal?
The requirements of analog filter to be stable and causal are

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
The advantages of digital filters are
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
The disadvantages of digital filters are
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

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23. 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.)
24. 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.
25. 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 = - p
i
) without
multiplicity is

1
1
1 1

+ z e
to d transforme is
p s
T p
i
i

The impulse invariant transformation used to transform multiple real pole (at s = - p
i
)
is

( )
1 1
1 1
1
1
) 1 (
) 1 ( 1

+ z e dp
d
m
to d transforme is
p s
T p m
i
m m
m
i
i

26. 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 O = e
Where,
O - Analog frequency and
T - Sampling time period
27. What is Bilinear transformation?
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 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

1
1
1
1 2
z
z
T
s
+

=

28. 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,
Digital frequency,
2
tan 2
1
T O
=

e or
Analog frequency
2
tan
2 e
T
= O
where,
O - Analog frequency

T - Sampling time period
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29. What is frequency warping?
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.
30. What is prewarping? Why it is employed?
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.

31. State the structure of IIR filter .
IIR filters are of recursive type whereby the present o/p sample depends on present i/p, past i/p
samples and o/p samples. The design of IIR filter is realizable and stable. The impulse response h(n)
for a realizable filter is
h(n)=0 for n0

32. Compare the impulse invariant and bilinear transformations.

Sl.No Impulse Invariant transformation Bilinear transformation
1. It is many to one mapping It is one to one mapping.
2. The relation between analog and
digital frequency is linear.
The relation between analog and digital
frequency is nonlinear.
3. To prevent the problem of aliasing the
analog filters should be band limited.
There is no problem of aliasing and so
the analog filter need not be band limited.

4.
The magnitude and phase response of
analog filter can be preserved by
choosing low sampling time or high
sampling frequency.
Due to the effect of warping, the phase
response of analog filter cannot be
preserved. But the magnitude response
can be preserved by prewarping.
33. State the advantage of direct form structure over direct form structure.
In direct form structure, the number of memory locations required is less than thatof direct
form structure.
34. What is the mapping procedure between S-plane & Z-plane in the method of mapping
differentials? What are its characteristics?
The mapping procedure between S-plane & Z-plane in the method of mapping of differentials is given
by
H(Z) =H(S)|S=(1-Z
-1
)/T
The above mapping has the following characteristics
- The left half of S-plane maps inside a circle of radius centered at Z= in the Z-plane.
- The right half of S-plane maps into the region outside the circle of radius in the Z-plane.
- The j -axis maps onto the perimeter of the circle of radius in the Z-plane.

35. What is meant by impulse invariant method of designing IIR filter?
In this method of digitizing an analog filter, the impulse response of resulting digitalfilter is a sampled
version of the impulse response of the analog filter.The transfer function of analog filter in partial
fraction form,

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36. Give the bilinear transform equation between S-plane & Z-plane.
S=2/T(1-Z
-1
/1+Z
-1
)
37. What are the different types of structures for realization of IIR systems?
The different types of structures for realization of IIR system are
(i) Direct form I structure (ii) Direct form II structure
(iii) Cascade form structure (iv) Parallel form structure

38. 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 a Finite Impulse response (FIR) digital filter.
39. What are the properties of bilinear transformation?
- The mapping for the bilinear transformation is a one-to-one mapping that is for every point Z,
there is exactly one corresponding point S, and vice-versa.
- The j -axis maps on to the unit circle |z|=1,the left half of the s-plane maps to the interior of the
unit circle |z|=1 and the half of the s-plane maps on to the exterior of the unit circle |z|=1.

40. Write a short note on pre-warping.
The effect of the non-linear compression at high frequencies can be compensated. When the desired
magnitude response is piece-wise constant over frequency, this compression can be compensated by
introducing a suitable pre-scaling, or pre-warping the critical frequencies by using the formula.

- The bilinear transformation provides one-to-one mapping.
- Stable continuous systems can be mapped into realizable, stable digital systems.
- There is no aliasing.
- The mapping is highly non-linear producing frequency, compression at high frequencies.
- Neither the impulse response nor the phase response of the analog filter is preserved in a digital
filter obtained by bilinear transformation.
Quantization errors can be minimized if we realize an LTI system in cascade form.
43. Define signal flow graph.
A signal flow graph is a graphical representation of the relationships between thevariables of a
set of linear difference equations.

44. What is the main advantage of Direct form- II realizations when compared to Direct form I
realization?
In Direct form II realization, the number of memory locations required is less than that of
Direct form I realization.
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45. 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.
46. 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.
Quantization errors can be minimized if we realize an LTI system in cascade form.
48. How can you design a digital filter from analog filter?
Digital filter can de designed from analog filter using the following methods
1. Approximation of derivatives
2. Impulse invariant method (IIM)
3. Bilinear transformation (BLT)

49. What is a disadvantage of BLT method?
The mapping is non-linear and because of this, frequency warping effect takes place.

50. . List the Butterworth polynomial for various orders.
N Denominator polynomial
1 S+1
2 S
2
+1.414s+1
3 (s+1)(s
2
+s+1)
4 (s
2
+.7653s+1)(s
2
+1.847s+1)
5 (s+1)(s
2
+1.6183s+1)(s
2
+1.618s+1)

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21

UNIT III
FINITE IMPULSE RESPONSE DIGITAL FILTERS

1. 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 input samples and output samples.
The 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 different types of filters based on frequency response?
Based on frequency response the filters can be classified as
1. Low pass filter
2. High pass filter
3. Band pass filter
4. Band reject filter
1. FIR filters have exact linear phase.
2. FIR filters are always stable.
3. FIR filters can be realized in both recursive and non recursive structure.
4. Filters with any arbitrary magnitude response can be tackled using FIR sequence.
1.For the same filter specifications the order of FIR filter design
can be as high as 5 to 10 times that in an IIR design.
2. Large storage requirement is requirement
3. Powerful computational facilities required for the
Implementation.

4. What are the design techniques of designing FIR filters?
There are three well known methods for designing FIR filters with linear phase.
They are
(1) Window method
(2) Frequency sampling method
(3) Optimal or minimax design.

5. What is Gibbs phenomenon?
One possible way of finding an FIR filter that approximates H(e
jw
) would be to truncate the
infinite Fourier series at n=(N-1/2).Direct truncation of the series will lead to fixed percentage
overshoots and undershoots before and after an approximated discontinuity in the frequency response.

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6. What are Gibbs oscillations?
Oscillatory behavior observed when a square wave is reconstructed from finite
number of harmonics.
The unit cell of the square wave is given by

Its Fourier series representation is

7. Distinguish between FIR filters and IIR filters.

FIR filter IIR 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.
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 is more.

8. List the steps involved in the design of FIR filters using windows.
1.For the desired frequency response H
d
(w), find the impulse response h
d
(n) using Equation

h
d
(n)=1/2 H
d
(w)e
jwn
dw
-
2.Multiply the infinite impulse response with a chosen window sequence w(n) of
length N to obtain filter coefficients h(n),i.e.,

h(n)= h
d
(n)w(n) for |n|(N-1)/2
= 0 otherwise

3.Find the transfer function of the realizable filter
(N-1)/2
H(z)=z
-(N-1)/2
[h(0)+h(n)(z
n
+z
-n
)]
n=0

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9. What are the desirable characteristics of the window function?
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 lobes of the frequency response should decrease in energy
rapidly as tends to .

10. What are the advantages of Kaiser window?
o It provides flexibility for the designer to select the side lobe level and N
o It has the attractive property that the side lobe level can be varied continuously from the low
value in the Blackman window to the high value in the rectangular window
11. Give the equations specifying the following windows.
a. Rectangular window
b. Hamming window
c. Hanning window
d. Bartlett window
e. Kaiser window
a. Rectangular window:
The equation for Rectangular window is given by
W(n)= 1 0 n M-1
0 otherwise
b. Hamming window:
The equation for Hamming window is given by
W
H
(n)= 0.54-0.46 cos 2n/M-1 0 n M-1
0 otherwise
c. Hanning window:
The equation for Hanning window is given by
W
Hn
(n)= 0.5[1- cos 2n/M-1 ] 0 n M-1
0 otherwise
d. Bartlett window:
The equation for Bartlett window is given by
W
T
(n)= 1-2|n-(M-1)/2| 0 n M-1
M-1
0 otherwise
e. Kaiser window:
The equation for Kaiser window is given by
W
k
(n)= I
o
[1-( 2n/N-1)
2
] for |n| N-1
I
o
() 2
0 otherwise
where is an independent parameter.

12. What is the necessary and sufficient condition for linear phase characteristic in FIR
filter?
The necessary and sufficient condition for linear phase characteristic in FIR filter is, the
impulse response h(n) of the system should have the symmetry property i.e.,
H(n) = h(N-1-n)
DSP TWO MARKS Q&A 2013

24

where N is the duration of the sequence.
13. What is the principle of designing FIR filter using frequency sampling method?
In frequency sampling method the desired magnitude response is sampled and a linear
phase response is specified .The samples of desired frequency response are identified as DFT
coefficients. The filter coefficients are then determined as the IDFT of this set of samples.
14. 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.

15. Draw the direct form realization of FIR system.

16. Draw the direct form realization of a linear Phase FIR system for N even.

17. Draw the direct form realization of a linear Phase FIR system for N odd

1. z
-
1
1
1
1
1
1
1

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18. When cascade form realization is preferred in FIR filters?
The cascade form realization is preferred when complex zeros with absolute magnitude is less
than one.
19. Draw the M stage lattice filter.

20. 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.
t e t oe e u s s = ) (
For satisfying above condition
) 1 ( ) ( n N h n h =
that is the impulse response must be symmetrical about
2
1
=
N
n
If one constant group delay is desired then
oe | e u = ) (
For satisfying the above condition
) 1 ( ) ( n N h n h =
that is the impulse response must be antisymmetrical about
2
1
=
N
n

21. What are the properties of FIR filter?
The properties of FIR filters are
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.

22. How the constant group delay and phase delay is achieved in linear phase FIR filters.
Frequency response of FIR filters with constant group and phase delay

) (
) ( ) (
oe |
e e

=
j
e H H
The following conditions have to be satisfied to achieve constant group and phase
delay.
Phase delay,
2
1
=
N
o (i.e., phase delay is constant)
Group delay,
2
t
| = (i.e., group delay is constant)
Impulse response, h(n) = - h( N -1 n ) (i.e., impulse response is anti symmetric)

DSP TWO MARKS Q&A 2013

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23. 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.
24. 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.
The frequency response of a digital filter is periodic with period equal to sampling frequency
Any periodic function can be expressed as a linear combination of complex exponentials

25. Write the procedure for designing FIR filter by Fourier series method.
The procedure for designing FIR filter by Fourier series method is
i) Choose the desired (ideal) frequency response ) (e
d
H of the filter
i) Evaluate the Fourier series co-efficient of ) (e
d
H which gives the desired impulse response
) (n h
d

}

=
t
t
e
e e
t
d e H n h
n j
d d
) (
2
1
) (
ii) Truncate the infinite sequence ) (n h
d
to a finite duration sequence ) (n h
iii) Take Z transform of ) (n h to get a noncausal filter transfer function ) (z H of the FIR filter

iv) Multiply ) (z H by
|
.
|

\
|

2
1 N
z to convert noncausal transfer function to a realizable causal FIR
filter transfer function
= ) (z H ( )
(
(
(

+ +

|
.
|

\
|
2
1
1
2
1
) ( ) 0 (
N
n
n n
N
z z n h h z

26. 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=
(

2
1 N
. Direct truncation of the series will lead to fixed percentage overshoots and
undershoots before and after an approximated discontinuity in the frequency response.

27. Write the procedure for designing FIR filter using windows.
The procedure for designing FIR filter using windows are
i) Choose the desired (ideal) frequency response ) (e
d
H of the filter
ii) Evaluate the Fourier series co-efficient of ) (e
d
H which gives the desired impulse response ) (n h
d

}

=
t
t
e
e e
t
d e H n h
n j
d d
) (
2
1
) (
DSP TWO MARKS Q&A 2013

27

iii) Choose a window sequence w(n) and multiply the infinite sequence ) (n h
d
by w(n)to convert the
infinite duration impulse response to finite duration impulse response ) (n h
) ( ) ( ) ( n w n h n h
d
=

iv) Find the transfer function of the realizable FIR filter
= ) (z H ( )
(
(
(

+ +

|
.
|

\
|
2
1
1
2
1
) ( ) 0 (
N
n
n n
N
z z n h h z

28. What is window and why it is necessary?
One possible way of finding an FIR filter that approximates ) (
e j
e H would be to truncate the infinite
Fourier series at n=
(

2
1 N
. The abrupt truncation of the series will 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 ) (n w , called a window.
29. List characteristics of FIR filter designed using windows.
Thecharacteristics of FIR filter designed using windows are
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
iii) The attenuation in the stop band is fixed for a given window, except in case of
Kaiser window where it is variable

30. Write the procedure for FIR filter design by frequency sampling method.
The procedures for FIR filter design by frequency sampling method are
1. Choose the desired frequency response ) (e
d
H
2. Take N samples of ) (e
d
H to generate the sequence ) (
~
k H
3. Take inverse DFT of ). (
~
k H 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

31. 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.

32. Write the expression for frequency response of rectangular window.
The frequency response of rectangular window is given by
2
sin
2
sin
) (
e
e
e
N
R
W =
DSP TWO MARKS Q&A 2013

28

33. Write the characteristic features of rectangular window.
The characteristic features of rectangular window are
The main lobe width is equal to
N
t 4

The maximum sidelobe magnitude is -13dB
The sidelobe magnitude does not decreases significantly with increasing e

34. List the features of FIR filter designed using rectangular window.
The features of FIR filter designed using rectangular window are
i) The width of the transition region is related to the width of the mainlobe of window spectrum
ii) Gibbs oscillations are noticed in the passband and stop band
iii) The attenuation in the stopband is constant and cannot be varied
35. Write the equation specifying Hanning windows.
The equation for Hanning window is given by

1
2
cos 5 . 0 5 . 0 ) (

+ =
N
n
n w
n H
t
for
2
) 1 (
2
) 1 (
s s

N
n
N

= 0 Otherwise.

36. Write the equation specifying Hamming windows.
The equation for Hamming window is given by

1
2
cos 46 . 0 54 . 0 ) (

+ =
N
n
n w
H
t
for
2
) 1 (
2
) 1 (
s s

N
n
N

= 0 Otherwise.
37. Write the equation specifying Blackman windows.
The equation for Blackman window is given by
1
4
cos 08 . 0
1
2
cos 5 . 0 42 . 0 ) (

+ =
N
n
N
n
n w
B
t t
for
2
) 1 (
2
) 1 (
s s

N
n
N

= 0 Otherwise.
38. Write the equation specifying Kaiser windows.
The equation for Kaiserwindow is given by

(
(
(
(
(

|
.
|

\
|

=
) (
1
2
1
) (
0
0
o
o
I
N
n
I n w
k
for
2
) 1 (
s
N
n
= 0 Otherwise.
Where
o is an independent parameter.

0
I (x) is the zeroth order Bessel function of the first kind

2
1
0
2 !
1
1 ) ( I

= (
(

|
.
|

\
|
+ =
k
k
x
k
x
DSP TWO MARKS Q&A 2013

29

39. Compare the Rectangular and Hanning window.

Sl.No Rectangular window Hanning window
1. The width of mainlobe in window
spectrum is
N
t 4

The width of mainlobe in window spectrum is
N
t 8

2. The maximum sidelobe magnitude in
window spectrum is -13dB.
The maximum sidelobe magnitude in window
spectrum is -31dB.
3. In window spectrum the sidelobe
magnitude slightly decreases with
increasing e
In window spectrum the sidelobe magnitude
decreases with increasing e
4. In FIR filter designed using rectangular
window the minimum stopband
attenuation is 22dB.
In FIR filter designed using Hanning window
the minimum stopband attenuation is 44dB.

40. Compare the Rectangular and Hamming window.
Sl.No Rectangular window Hamming window
1. The width of mainlobe in window
spectrum is
N
t 4

The width of mainlobe in window spectrum is
N
t 8

2. The maximum sidelobe magnitude in
window spectrum is -13dB
The maximum sidelobe magnitude in window
spectrum is -41dB
3. In window spectrum the sidelobe
magnitude slightly decreases with
increasing e
In window spectrum the sidelobe magnitude
remains constant
4. In FIR filter designed using rectangular
window the minimum stopband
attenuation is 22dB.
In FIR filter designed using Hamming window
the minimum stopband attenuation is 51dB

41. Compare the Hanning and Hamming window.
Sl.No Hanning window Hamming window
1. The width of mainlobe in window
spectrum is
N
t 8

The width of mainlobe in window spectrum is
N
t 8

2. The maximum sidelobe
magnitude in window spectrum is
-31dB
The maximum sidelobe magnitude in window
spectrum is -41dB
3. In window spectrum the sidelobe
magnitude decreases with
increasing e
In window spectrum the sidelobe magnitude
remains constant. Here the increased sidelobe
attenuation is achieved at the expense of constant
attenuation at high frequencies
4. In FIR filter designed using
Hanning window the minimum
stopband attenuation is 44dB
In FIR filter designed using Hamming window the
minimum stopband attenuation is 51dB

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42. Compare the Hamming and Blackman window.
Sl.
No
Hamming window Blackman window
1. The width of mainlobe in window
spectrum is
N
t 8

The width of mainlobe in window spectrum is
N
t 12

2. The maximum sidelobe magnitude in
window spectrum is -41dB
The maximum sidelobe magnitude in window
spectrum is -58dB
3. In window spectrum the sidelobe
magnitude remains constant with
increasing e
In window spectrum the sidelobe magnitude
decreases rapidly with increasing e
4. In FIR filter designed using Hamming
window the minimum stopband
attenuation is 51dB
In FIR filter designed using Blackman window
the minimum stopband attenuation is 78dB
5. The higher value of sidelobe attenuation
is achieved at the expense of constant
attenuation at high frequencies
The higher value of sidelobe attenuation is
achieved at the expense of increased mainlobe
width
43. Write the features of Blackman window spectrum.
The features of Blackman window spectrum are
i) The main lobe width is equal to
N
t 12
.
ii) The maximum sidelobe magnitude is -58dB
iii) The sidelobe magnitude slightly decreases with increasing e
i) The higher value of sidelobe attenuation is achieved at the expense of increased
mainlobe width
44. Write the features of Kaiser Window spectrum.
The features of Kaiser window spectrum are
i) The width of mainlobe and the peak sidelobe are variable
ii)The parameter o 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
45. Compare the Hamming and Kaiser window.
Sl.No Hamming window Kaiser window
1. The width of mainlobe in window spectrum
is
N
t 8

The width of mainlobe in window spectrum
depends on the values of o and N
2. The maximum sidelobe magnitude in
window spectrum is -41dB
The maximum sidelobe magnitude with
respect to peak of mainlobe is variable using
the parameter o
3. In window spectrum the sidelobe
magnitude remains constant with
increasing e
In window spectrum the sidelobe magnitude
decreases with increasing e
4. In FIR filter designed using Hamming
window the minimum stopband attenuation
is 51dB
In FIR filter designed using Kaiser window
the minimum stopband attenuation is
variable and depends on the value of o

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46. What is transposition theorem & transposed structure?
The transpose of a structure is defined by the following operations.
- Reverse the directions of all branches in the signal flow graph
- Interchange the input and outputs.
- Reverse the roles of all nodes in the flow graph.
- Summing points become branching points.
- 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 flowgraph, the system function remains unchanged.

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UNIT IV
FINITE WORD LENGTH EFFECTS
1. What is meant by Finite word length effect?
The fundamental operations in digital filters are multiplication and addition. When theseoperations are
performed in a digital system the input data as well as the product and sum have to be represented in
finite word length, which depends on the size of the register used to store the data. In digital
computation the input and output data are quantized by rounding and truncation toconvert them to a
finite word size. This creates an error in the output or creates oscillations in the output. These effects
due to finite precision representation of the number in a digital system are called Finite Word Length.

2. List some of the finite word length effects in digital filters.
1. Errors due to the quantization of input data.
2. Errors due to the quantization of filter co-efficients.
3. Errors due to the quantization of rounding the product in multiplications.
4. Errors due to over flow in addition.
5. Limit cycles.

3. What are the types of number representations in digital system?
There are two types of number representation used in digital systems.
1. Fixed point representation of binary numbers.
2. Floating point representation of binary numbers.

4. Explain the fixed point representation of binary numbers.
In fixed point representation of binary numbers in a given word size, the bits allotted for integer part
and fraction part of the numbers are fixed. Therefore the position of binarypoint is fixed. The most
significant bit is used to represent the sign of the number. Since thenumber of digits is fixed it is
impossible to represent too large and too small numbers byfixed point representation.

5. What are the different formats of fixed point representation?
In fixed point representation there are three different formats for representing the binary numbers.
1. Sign-Magnitude format.
2. Ones Complement format.
3. Twos Complement format
In all the above three formats the positive number is same but they differ only in representing the
negative numbers.

6. Explain the floating point representation of binary numbers.
The floating point representation is employed to represent larger range of numbers in a given binary
word size. The floating point number represented asFloating point number Nf = M * 2E
Where,M is called mantissa and it will be in binary fraction format. The value of M will be inthe range
of 0.5 M 1.
E is called exponent and it is either positive or negative integer.
In floating point representation both mantissa and exponent uses one bit for representing sign. A one
in the left most bit position represents negative, a sign and zero inthe left most bit position represents
positive sign.
Mantissa Exponent
Sign bit Sign bit

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7. Compare the fixed point and floating point number representation.

8. What are the two types of quantization employed in digital system?
There are two types of quantization employed in digital systems
1. Truncation.
2. Rounding.

9. What are the advantages of floating pint representation?
1. Large dynamic range
2. Overflow is unlikely.

10. What is truncation?
The truncation is the process of reducing the size of binary number by discarding all bits less
significant than the least bit that is retained. In the truncation of a binary number to be b bits allthe less
significant bits beyond bth bit are discarded.

11. What is rounding?
Rounding is the processes of reducing the size of a binary number to finite word size of b-bits such
that, the rounded b-bit number is closest to the original unquantized number.

12. What is quantization?
Quantization is the process of converting the discrete time continuous signal into a digital signal.

13. What is quantization noise (or) quantization error?
For most of an engineering application the input signal is continuous in time or analog waveform. This
signal is converted into digital by using analog to digital converter. The errorcan be introduced in
representing the conversion of analog to digital form is called quantization noise (or) quantization
error.

14. What is meant by Quantization step size?
In digital system, the numbers are represented in binary with b-bit binary we can generate the 2b
different binary codes. Any range of analog value to be represented in binary should bedivided into 2b
levels with equal increment.
The 2b levels are called quantization levels and the increment in each level is called quantization step
size.
Quantization Step Size q = R / 2b
Where,
R is the range of analog signal.
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
4.Round off error occur only for
Round off errors can occur with both
5.Overflow occur in addition Overflow does not arise
6.Used in small computers Used in larger, general purpose
computers
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15. What are the three quantization errors due to finite word length registers in digital filter?
1. Input quantization error.
2. Coefficient quantization error.
3. Product quantization error.

16. What do you understand by input quantization error?
In digital signal processing, the continuous time input signals are converted into digital using analog to
digital converter. The representation of continuous signal amplitude by a fixeddigit produces an error,
which is known as input quantization error.

17. What is meant by Coefficient quantization error?
In the realization of FIR and IIR filters the location of poles and zeros of digital filters directly
depends on the value of coefficients. The quantization of the filter coefficient will modifythe value of
poles and zeros and so the location of the poles and zeros will be shifted from thedesired location. This
will create deviations in the frequency response of the system. So that the system fail to met the actual
frequency response specifications.If the poles of the desired filter are close to the unit circle, then
those of the filter withquantized coefficient may lie just outside the unit circle.

18. What is meant by product quantization error?
In digital computations, the output of multipliers i.e. the products are quantized to finite word length in
order to store them in registers and to be used in subsequent calculations. Theerror due to the
quantization of the output of multiplier is referred to as product quantizationerror.

19. Define Noise Transfer Function (NTF).
The Noise Transfer Function (NTF) is defined as the transfer function from the noise sourceto the
filter output. The NTF depends on the structure of the digital network.

20. What is limit cycles?
In recursive systems when the input is zero or nonzero constant value, the nonlinearities dueto finite
precision arithmetic operations may causes periodic oscillations in the output theseoscillations are
called limit cycles.

21. What are the types of limit cycle? (Or) what are the two kinds of limit cycle behavior in
DSP?
There are two types of limit cycle,
1. Zero input limit cycle.
2. Overflow limit cycle.

22. 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 willcontinue to remain
in limit cycle even when the input is made zero. Hence these limit cycles arealso called zero input
limit cycles.

23. 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.

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In a limit cycle the amplitude of the output are confined to a range of values, which is calleddead band
of the filter.
Dead band = 2-b / 1-|a|
25. Why scaling is need in the digital filter implementation? (Or) what is the need for scaling in
the digital filter implementation?
To prevent overflow, the signal level at certain points in the digital filter must be scaled sothat no
26. How overflow limit cycles can be eliminated? (Or) what are the methods used to prevent
overflow?
The overflow limit cycles can be eliminated either by using saturation arithmetic or byscaling the

27. What is meant by saturation arithmetic?
When the sum of two fixed point numbers exceeds the dynamic range, overflow is occurs. Which
causes the output of adder to oscillate between maximum amplitude limits, when anoverflow 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.

28. What is the drawback in saturation arithmetic?
The saturation arithmetic introduces nonlinearity in the adder which creates signal distortion.

29. What is meant by sign magnitude representation?
For sign magnitude representation the leading binary digit is used to represent the sign.If it is
equal to 1 the number is negative, otherwise it is positive.

30. What is meant by 1s complement form?
In 1,s complement form the positive number is represented as in the sign magnitude form. To
obtain the negative of the positive number, complement all the bits of the positive number.

31. What is meant by 2s complement form?
In 2s complement form the positive number is represented as in the sign magnitude form. To
obtain the negative of the positive number, complement all the bits of the positive number and add 1 to
the LSB.
.
32. Define truncation error for sign magnitude representation and for 2s complement
Representation
Truncation is a process of discarding all bits less significant than least significant bit that is
retained For truncation in floating point system the effect is seen only in mantissa.if the mantissa is
truncated to b bits ,then the error satisfies
0 > -2.2
-b
for x >0 and
0 < -2.2
-b
for x <0

33. What are the different quantization methods?
Amplitude quantization
Vector quantization
Scalar quantization
34. List the advantages of floating point arithmetic.
- Large dynamic range
- Occurrence of overflow is very rare
- Higher accuracy
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35. Give the expression for signal to quantization noise ratio and calculate the improvement with
an increase of 2 bits to the existing bit.
SNR
A / D
= 16.81+6.02b-20log
10
(R
FS
/
x
) dB.
With b = 2 bits increase, the signal to noise ratio will increase by 6.02
X 2 = 12dB.
36. What is truncation error?
Truncation is an approximation scheme wherein the rounded number or digits after the pre-

37. Give the rounding errors for fixed and floating point arithmetic.
A number x represented by b bits which results in b
R
after being Rounded off. The quantized
error
R
due to rounding is given by

R
=Q
R
(x)-x
where Q
R
(x) = quantized number(rounding error)
The rounding error is independent of the types of fixed point arithmetic, since it involves the
magnitude of the number. The rounding error is symmetric about zero and falls in the range.
-((2
-bT
-2
-b
)/2)
R
((2
-bT
-2
-b
)/2)

R
may be +ve or ve and depends on the value of x.
The error
R
incurred due to rounding off floating point number is in the range
-2
E
.2
-bR/2
)
R
2
E
.2
-bR/2

38. How the multiplication and addition are carried out in floating point arithmetic?
In floating point arithmetic, multiplications are carried out as follows
Let f
1
= M
1
x 2
c1
and f
2
= M
2
x 2
c2
. Then f
3
= f
1
x f
2
= (M
1
xM
2
)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.
39. 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.

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UNIT V

MULTIRATE SIGNAL PROCESSING

1. What is multirate signal processing?
The theory of processing signals at different sampling rates is called multirate signal
processing

2. Define down sampling
Down sampling a sequencex(n) by a factor M is the process of picking every Mth sample and

3. What is meant by up sampling?
Up sampling by factor L is the process of inserting L-1 zeros between two consecutive samples
4. Define the basic operations in multirate signal processing.
The basic operations in multirate signal processing are
(i)Decimation
(ii)Interpolation
Decimation is a process of reducing the sampling rate by a factor D, i.e., down-sampling.
Interpolation is a process of increasing the sampling rate by a factor I, i.e., up-sampling.

5. What is meant by decimation?
The process of decreasing the sampling rate by an integer factor D is called as decimation.
down-sampling
After downsampling, keeping every Dth sample in x(n) and removing D-1 in between samples
y(n) =x(nD)

6. What is meant by interpolation?
The process of increasing the sampling rate by an integer factor I is called as decimation.
up-sampling
After upsampling, inserting I-1 zeros between consecutive samples
y(n) =x(n/I)

7. Define sub band coding of speech.
Sub band coding of speech is a method by which the speech signal is subdivided into several
frequency bands and each band is digitally encode separately. In the case of speech signal
processing, most of its energy is contained in the low frequencies and hence can be coded with
more bits then high frequencies.

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8. What is the effect of quantization on pole locations?
N
D(z) = (1-p
k
z
-1
)
k=1
pk is the error or perturbation resulting from the quantization of the filter coefficients

9. What is an anti-imaging filter?
The image signal is due to the aliasing effect. In caseof decimation by M, there will be M-1
additional images of the input spectrum. Thus, the input spectrum X() is band limited to
the low pass frequency response. An anti-aliasing filter eliminates the spectrum of X() in
the range (/D .
The anti-aliasing filter is LPF whose frequency response H
LPF
() is given by
H
LPF
() = 1, || /M
= 0, otherwise.
D Decimator

10. What is a decimator? If the input to the decimator is x(n)={1,2,-1,4,0,5,3,2}, What is the
output?
Decimation is a process of reducing the sampling rate by a factor D, I.e., down-sampling.
x(n)={1,2,-1,4,0,5,3,2}
D=2
Output y(n) = {1,-1,0,3}

11. If the spectrum of a sequence x(n) is X(e
(jw)
), then what is the spectrum of a signal down
sampled by factor 2

Y(exp(jw))= 0.5[X(exp(jw/2))+X(exp(j(w/2-pi)))

12. What is the need for anti-aliasing filter prior to down sampling?

The spectra obtained after down sampling a signal by a factor M is the sum of all the uniformly
shifted and stretched version of original spectrum scaled by a factor 1/M, then down sampling will
cause aliasing. In order to avoid aliasing the signal x(n) is to be band limited to plus or minus pi/M .
This can be done by filtering the signal x(n) with a low pass filter with a cutoff frequency of pi/M.
This filter is known as antialiasing filter.

13. Define sampling Rate conversion.
Sample rate conversion is the process of changing the sampling rate of a discrete-time signal to
obtain a new discrete-time representation of the underlying continuous-time signal. When applied to
an image, this process is sometimes called image scaling.

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14. Mention two applications of multirate signal processing.
Up-sampling, i.e., increasing the sampling frequency, before D/A conversion in order to relax the
requirements of the analog low pass antialiasing filter. This technique is used in audio CD, where
the sampling frequency 44.1 kHz is increased fourfold to 176.4 kHz before D/A conversion.
Various systems in digital audio signal processing often operate at different sampling rates. The
connection of such systems requires a conversion of sampling rate.
- Design of Phase shifters
- Interfacing digital systems with different sampling rates
- Implementation of Narrowband Low pass filters
- Implementation of Digital Filter Banks
- Sub band coding
- Transmultiplexers

15. What are signal rate systems?
The systems that use single sampling rate from A/D converter to D/A converter are known as
single rate systems.

16. What are multi rate systems?
The discrete time systems that process data at more than one sampling rate are known as
multirate systems.

17. Where is multi rate digital signal processing required?
Multi rate digital signal processing is required in digital systems where more than one
sampling rate is required. For example in digital audio, the different sampling rates used are 32 kHz
for broadcasting,44.1 kHz for compact disc and 48 kHz for audio tape.

18. What are the advantages of multi rate signal processing?
- Computational requirements are less.
- Storage for filter coefficients is less
- Finite arithmetic effects are less
- Filter order required in multirate applications is low
- Sensitivity to filter coefficient lengths is less
19. Name the areas in which multi rate signal processing is used.

- Speech and audio processing system
- Antenna system
- Communication systems
20. State sampling theorem.
Sampling theorem states that a band limited signal x(t) having a finite energy, which has no
spectral components higher than f
h
hertz can completely be described and reconstructed from its
samples taken at a rate of >2f
h
samples per second. The sampling rate of 2f
h
samples per second is
called the nyquist rate and its reciprocal f
h
is called the nyquist period.

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21. What do you mean by aliasing?
The over lapping of the spectra at the output of the down sampler due to the lack of band limiting
of the signal fed to the down sampler is called aliasing.

22. What do you mean by imaging?
The phenomenon of getting image spectra in the output of up sampler in addition to the scaled
input spectra is called imaging.

23. What do you mean by image spectra?
Insection of I-1 zeros between successive values of input signal x(n) results in a signal whose
spectrum X(e
jwI
) is an i-fold periodic repetition of the input signal spectrum X(e
jwi
) .these
additional spectra are called image spectra.

24. What do you mean by anti imaging filter?
The low pass filter which is used after the up sampler to remove the image spectra is called the anti
imaging filter.

25. When can a cascade of a factor of D down sampler and a factor of I up sampler
interchangeable with no change in the input and output relation?
A cascade of a factor of D down sampler and a factor of I up sampler interchangeable with no
change in the input and output relation if and only if I and D are co-prime.

26. How many types of filter banks are there? What are they?
There are two types of filter banks .they are analysis filter bank and synthesis filter bank.

27. What is the relation between D-channel synthesis filter bank and D-channel analysis filter
bank?
The relation between D-channel synthesis filter bank and D-channel analysis filter bank is that
they are dual of each other

28. When do you go for multistage implementation for sampling rate conversion?

For performing sampling rate conversion we go in for multistage implementation when either
D>>1 and/or I>>1.