NOORUL ISLAM COLLEGE OF ENGINEERING

ELECTRONICS AND INSTRUMENTATION DEPARTMENT

SEMESTER-5
DIGITAL SIGNAL PROCESSING(EI335)
UNIT I
1. Define signal?
Any physical quantity that carries information varies with other independent or
dependent variables.
2. What are the main types of signals with respect to time as independent
variable?
Continuous time (analog) signals &discrete time (discrete) signals
3. What is analog signal?

The analog signal is a continuous function of independent variables . The analog

signal is defined for every instant of independent variable and so magnitude of
independent variable is continuous in the specified range. here both the independent
variable and magnitude are continuous.
4. What is discrete signal?
The discrete signal is a function of discrete independent variables .The in
independent variable is divided into uniform interval and is represented by an integer,
The discrete signal is is defined for every integer value of independent variable. here
both the values of signal and independent variable are discrete.
5. What is digital signal?
The digital signal is same as discrete signal except that the magnitude of signal is
quantized.
6.What are the different types of signal representations?
a. Graphical representation
b. Functional representations
c. Tabular representation
d. Sequence representation
7. Define periodic and non periodic discrete time signals?
If the discrete time signal repeated after equal samples of time then it is called
periodic signal. When the discrete time signal x[n] satisfies the condition
x[n+N]=x(n), then it is called periodic signal with fundamental period N samples. if
x(n) ≠ x(n+N) then it is called no periodic signals.
0
8.Define unit sample sequence? the unit sample sequence δ(n) and is defined as

δ(n)= 1 for n=0

0 for n≠0
δ(n)
•1

n
9. .Define unit step sequence
A unit step sequence is denoted as u(n)=1 for n≥0
u(n) 0 other wise

1
…….. n

10.Define unit ramp sequence?

A unit ramp sequence is defined as r(n)= n for n≥0
0 other wise

r(t)

11..Define a system?
A system is a physical device or algorithm that performs an operation on the
signal
12.What is digital signal processing?
The dsp refers processing of signal by digital system.

13. What are the steps involved in digital signal processing?

a. Converting the analog signal to digital signal ,which is performed by A/D
converter
b. Processing the digital signal by digital systems.
c. Converting the digital output signal from the digital system to analog
signal by D/A converter.
14. What are the advantages of DSP?
a. The programme can be modified easily for better Performance.
b. Better accuracy can be achieved by using adaptive algorithm.
c. The digital signal can be easily stored and transported.
d. Digital systems are cheaper than analog equal lent.
15. Give some applications of DSP?
a. Speech processing
b. Communications
 c. Biomedical
16. Write the difference equation governing the Nth order LTI system.

N M
Y(n)=∑ ak y(n-k) +∑ bk x(n-k)
k=1 k=0
a. N is the order of the system
b. ak & bk are constant coefficients
c. y(n)&x(n) are output and input to the system
17. List the various methods of classifying discrete time systems?
a. Static and dynamic systems.
b. Time invariant and time variant
c. Linear and nonlinear
d. Causal and no causal
e. Stable and unstable
f. FIR and IIR systems
g. Recursive and non recursive systems
18. What are static and dynamic systems? Give examples?
A discrete time system is called static(memory less)if it’s output at any instant
n dependent on the input sample at the same time (but does not depend on past or
future samples).If the response depends on past or future samples, then the system
is called dynamic system.
Eg.y(n)=ax(n) static system
Y(n)=ax(n)+bx(n-1)
19. Define time invariant system?
A system is said to be time invariant if it’s input output characteristics does
not change with time. Let H be a system and H{X(n)}=Y(n).now if H{X(n-
k)}=Y(n-k) then the system H is called time invariant.
20. What is linear and nonlinear systems?
If a system satisfies superposition and homogeneity principles then the system
is called linear otherwise it is called nonlinear

If H is a system,X1(n) and X2(n) are inputs a and b are constants then

H{aX1(n)+BX2(n)}=aH{X1(n)}+bH{X2(n)} then His linear.

21. What is a causal system give an example?

A system is said to be causal, if the output of the system at any time n depends
on present and past inputs ,but does not depend on future inputs.
E.g.(n)=x(n)+x(n-1)
22. Define a stable system?
Any relaxed system is said to be bounded input bounded output stable if and
only if every bounded input yields a bounded output.
∝
∑ h(n)< ∝ where h(n)is impulse response of the system
n=-∝
23. What is LTI system?
 A linear time invariant system is defined that a system obeys both linearity
and time invariant properties.
If a system satisfies superposition and homogeneity principles then the system
is called linear

A system is said to be time invariant if it’s input output characteristics does

not change with time.
24. What are FIR and IIR systems?
FIR (finite impulse response):this type of system has an impulse response
which is zero outside the finite time interval eg. h(n)=0 for n<0 and n>N
IIR (Infinite Impulse Response):An IIR system exhibits an impulse response
of infinite duration.
25. State sampling theorem.
A band limited continuous time signal ,with higher frequency fc Hz can be
uniquely recovered from it’s samples provided that the sampling rate F>2fc
samples per second.
26. Show whether the system is linear?
Y(n)=n x(n)
H{aX1(n)+BX2(n)}=a H{X1(n)}+b H{X2(n)} then H is linear.

a H{X1(n)}+b H{X2(n)}=anx1(n)+bnx2(n) ------------(1)

H{aX1(n)+BX2(n)}= anx1(n)+bnx2(n) --------------(2)

(1)=(2) So the system is linear.

27. Show whether the system is linear?

Y(n)=nx2(n)

Since x2(n) term is present in the system which implies non linearity in to the
system. Therefore the system is nonlinear.
28. Determine if the following system is time invariant or time variant?

Y(n)=x(n)+x(n-1)
If the input is delayed by k units in time we have y(n,k)=H{x(n-k)}=x(n-k)
+x(n-k-1)
If we delay the output by k units then y(n-k)= x(n-k)+x(n-k-1)
So the system is time invariant.
29. Determine if the system described by the following equation is causal or not?
Y(n)=x(n2)
For n = -1
Y(-1)=x(1)
For n = 2 Y(2) = x(4)
 Therefore the output of the system depends on future input and hence the
system is non causal.
30. Define unit sample response of a system and what is it’s significance?
The response of a system denoted as h(n),obtained from a discrete time
system when the input signal is a unit sample sequence is known as unit sample
response.
UNITII
31. Define z transform?
The Z transform of a discrete time signal x(n) is defined as
∞
X(z) = ∑ x(n)z-n
n= -∞
where z is a complex variable. In polar form z=re-jw
32. What is meant by ROC?
The region of convergence (ROC) is defined as the set of all values of z for
which x(z) converges.
33. Explain about the roc of causal and anti-causal infinite sequences?
For causal system the roc is exterior to the circle of radius r.
For anti causal system it is interior to the circle of radius r.
34. Explain about the roc of causal and anti causal finite sequences
For causal system the roc is entire z plane except z=0.
For anti causal system it is entire z plane except z=∞.
35. What are the properties of roc?
a. The roc is a ring or disk in the z plane centered at the origin.
b. The roc cannot contain any pole.
c. The roc must be a connected region
d. The roc of an LTI stable system contains the unit circle.
36. Explain the linearity property of the z transform
If z{x1(n)}=x1(z) and z{x2(n)}=x2(z) then z{ax1(n)+bx2(n)}=ax1(z)+bx2(z)
a&b are constants.

37. State the time shifting property of the z transform

If z{x(n)}=x(z) then z{x(n-k)}=z-kx(z)

38. State the scaling property of the z transform

If z{x(n)}=x(z) then z{anx(n)}=x(a-1z)
39. State the time reversal property of the z transform
If z{x(n)}=x(z) then z{x(-n)}=x(z-1)
40. Explain convolution property of the z transform
If z{x(n)}=x(z) & z{h(n)}=h(z) then z {x(n)*h(n)}=x(z)h(z)
41. Explain the multiplication property of z transform
If z{x(n)}=x(z) & z{h(n)}=h(z) then
z {x(n) h(n)}= 1/2πj ∫c x(γ)h(z/γ) γ-1d γ
42. State Parseval’s relation in z transform
If x1(n) and x2(n) are complex valued sequences then
∞
 ∑x1(n)x2*(n)=1/2 πj ∫c x1(γ)x2*(1/γ*) γ-1d γ
n=-∞

43. State and prove initial value theorem of z transform

If x(n) is causal then x(0)= lt x(z)
z→∞
proof: ∞
X(z)= ∑x(n)z-n ----------------------(1)
n=-∞
in(1) put n=0 →x(n)→x(z)=∞
hence proved
44. State final value theorem of z tramsform
If x(n) is causal z{x(n)}=x(z), where the roc of x(z) includes, but it is not
necessary to confined to │ z │>1 and (z-1)x(z)has no pole on or outside the unit
circle then
x(∞) = lt (z-1) x(z)
z→1
45. Define system function?
The ratio between z transform of out put signal y(z) to z transform of input
signal x(z) is called system function of the particular system
Y(z)
H(z)= ---------
X(Z)
46. What are the conditions of stability of a causal system ?
All the poles of the system are with in the unit circle.
The sum of impulse response for all values of n is bounded
∞
∑ h(n) < ∞
n = -∞
47. Determine z transform and roc of the signal {1,2,3,4}

-∞
X(z) = ∑ x(n)z-n
n =-∞
3
-n
= ∑x(n)z =x(0)z-0+x(1)z-1+x(2)z-2+x(3)z-3
n=0
= 1z-0+2z-1+3z-2+4z-3
roc is entire z plane except z = 0
48. Determine z transform and roc of the signal {1,2,3,4}

∞
X(z) = ∑ x(n)z-n
n=-∞
 0
X(z)= ∑ x(n)z-n = x(-3)z3+x(-2)z2+x(-1)z1+x(0)
n=-3
= 4+3z1+2z2+1z3
ROC is entire z plane except z=∞

49. Determine z transform and roc of the signal {1,2,3,4}

∞
X(z)= ∑ x(n) z-n
n=-∞
2
X(z)= ∑x(n)z-n = x(-1)z1 + x(0)z0 + x(1)z-1 + x(2)z -2
n=-1
= 1z1+2+3z-1+4z-2
ROC is entire z plane except z=∞,0

50. Find the z transform and roc of anu(n)

∞
X(z) = ∑ x(n)z-n
n=-∞
∞
X(z) = ∑ an z-n =1/(1-az-1) roc│z│>a.
n=0
51. Find the z transform and roc of -anu(-n-1)
∞
X(z)= ∑ x(n)z-n
n=-∞

-1
X(z)= - ∑ an z-n
n= - ∞
∞
= -∑(a-1z)n = 1/(1-az-1) roc│z│<a.
n=1

52. The z-transform of a sequence x(n) is x(z),what is the z transform of nx(n)

If z{x(n)}=x(z) then z{nx(n)}=-zd(x(z))/dz

53. Find the z-transform of (a) A digital impulse (b) A digital step.
(a)Since x(n) is zero except for n = 0, where x(n) is 1, we find x(z) = 1.
(b) Since x(n) is zero except for n≥0, where x(n) is 1, we find

∞ 1
x(z) = ∑ Z-n =
 n=0 1 – z-1

54. What is the relationship between z-transform and DTFT?

The z-transform of x(n) is given by
∞
x(z) = ∑ x(n) Z-n ; where z = rejω ……………….. (1)
n=-∞
Substituting z in x(z) we get,
∞
x(z) = ∑ x(n) r-ne-jωn ………………. (2)
n=-∞
The Fourier transform of x(n) is given by
∞
x(ejω) = ∑ x(n) e-jωn ………………..(3)
n=-∞
Equation (2) and (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(ejω) is equal to H(z) evaluated along the unit circle, or X(ejω) = x(z)│z = ejω
For X(ejω) to exist, the ROC of x(z) must include the unit circle.
55. What are the different methods of evaluating inverse z-transform?
It can be evaluated using several methods.
i. Long division method
ii. Partial fraction expansion method
iii. Residue method
iv. Convolution method
56. Define DFT of a discrete time sequence.
The dft is used to convert a finite discrete time sequence x(n) to an N point
frequency domain sequence x(k).The N point DFT of a finite sequence x(n) of
length L,(L<N) is defined as

N-1
x(k)= ∑ x(n)e-j2πnk/N K=0,1,2,3,…N-1
n=0
57. Define IDTFT
The IDTFT of the sequence of length N is defined as

N-1
X(n)=(1/N ) ∑x(k)ej2πnk/N n=0,1,2,3,…N-1
k=0

58. Define DTFT and IDTFT of a sequence?

The DTFT (Discrete Time Fourier Transform) of a sequence x(n) is
defined as
∞
X(w) = ∑x(n)e-jwn
n = -∞
 π
The IDTFT is defined as x(n)= 1/2 π ∫x(w) ejwn dw
-π
59. What is the drawback in DTFT?
The drawback in discrete time fourier transform is that it is continuous
function of w and cannot be processed by digital systems.
60. What is the relation between DFT and DTFT?
Let x(n) be a sequence. DTFT{x(n)}=x(w) and DFT{X(n)}=x(k).x(k)
is a N point sequence which is obtained by sampling one period of x(w) at N
equal intervals.

X(ω) = X(K)
ω=2πk/N
61. Calculate DFT of the sequence x(n)={1,1,2,2}

N-1
x(k)= ∑ x(n)e-j2πnk/N K=0,1,2,3,…N-1
n=0
3
x(k)= ∑x(n)e-j2πnk/N K=0,1,2,3
n=0

N=4

= x(0)+x(1)e-jkπ/2+x(2)e-jkπ+x(3)e-j3kπ/2

= 1+ e-jkπ/2-2e-jkπ-2e-j3kπ/2 K=0,1,2,3

62. List any four properties of DFT

a. Periodicity
b. Linearity
c. Time reversal
d. Circular time shift
63. State periodicity property with respect to DFT.
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.
64. State periodicity property with respect to DFT.
If x1(k) and x2(k) are N-point DFTs of finite duration sequences x1(n) and
x2(n), then DFT [a x1(n) + b x2(n)] = a x1(k) + b x2(k), a, b are constants.
65. State time reversal property with respect to DFT.
If DFT[x(n) =x(k), then
DFT[x((-n))N] = DFT[x(N-n)] = x((-k))N = x(N-k)
66. State circular time shifting property with respect to DFT.
If DFT[x(n)] = x(k), then DFT [x((n-l))] = x(k) e-j2πkl/N
67. Assume two finite duration sequences x1(n) and x2(n) are linearly combined.
Let x3(n) = a x1(n) + b x2(n). What is the DFT of x3(n)?
Given x3(n) = a x1(n) + b x2(n).
Let DFT[ x1(n)] = x1(k) and DFT[ x2(n)] = x2(k), then
DFT[ x3(n)] = DFT [a x1(n) + b x2(n) ]
= a DFT[ x1(n)] +b DFT[ x2(n)]
= a x1(k) + b x2(k)
68. Compute the DFT of x(n) = δ(n – k1)
Given x(n) = δ(n – k1) = 1, when n = k1
0, otherwise
N-1
x(k)= ∑x(n)e-j2πnk/N K=0,1,2,3,…N-1
n=0
N-1
x(k)= ∑ δ(n – k1)e-j2πnk/N K=0,1,2,3,…N-1
n=0

= e-j2πk1k/N
69. What are the two methods used for sectional convolution?
(a) Overlap and add method
(b) Overlap and save method
70. Define circular convolution.
Let x1(n) and x2(n)are finite duration sequences both of length n with
DFTs x1(k) and x2(k). If x3(k) = x1(k) x2(k), then the sequence x3(k) can be
obtained by circular convolution, defined as
N-1
x(k) = ∑ x1(m) x2((n)) N
n=0
UNITIII
71. Why FFT is needed?
FFT is needed to compute DFT with reduced number of
calculations. DFT is required for spectrum analysis and filtering operations on the
signals using digital computers.
72. 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 N2 = 642 = 4096.
The number of complex multiplications required using FFT
is N log2 N = 64 log264 = 192
2 2

73. What is the main advantage of FFT?

FFT reduces the computation time required to compute
discrete fourier transform.
74. 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 log2N = 32 log232 = 80
2 2
75. What is FFT?
FFT is a method for computing the DFT with reduced number of
calculations using symmetry and periodicity properties of twiddle factor WkN .
The computational efficiency is achieved by decomposing of an N-point DFT into
successively smaller DFTs to increase the speed of computation.
76. How many multiplications and additions are required to compute N-point
DFT using radix-2 FFT?
N log2N multiplications and N log2N additions
2
77. What is meant by radix-2 FFT?
If the number of output points N can be expressed as a
power of 2, i.e., N = 2M Where M is an integer then this algorithm is known as
radix-2 algorithm.
78. What is DIT radix2 algorithm.
The radix 2 DIT FFT is an efficient algorithm for computing DFT.The
idea is to break N point sequence in to two sequences ,the DFT of which can be
combined to give DFT of the original N-point sequence. Initially the N point
sequence is divided in to two N/2 point sequences ,on the basis of odd and even
and the DFTs of them are evaluated and combined to give N-point sequence.
Similarly the N/2 DFT s are divided and expressed in to the combination of N/4
point DFTs. This process is continued until we left with 2-point DFTs
79. What is DIF radix2 algorithm.
The radix 2 DIFFFT is an efficient algorithm for computing DFT in this
the out put sequence x(k) is divided in to smaller and smaller. The idea is to break
N point sequence in to two sequences ,x1(n) and x2(n) consisting of the first N/2
points of x(n)and last N/2 points of x(n) respectively. Then we find N/2 point
sequences f(n) and g(n).f(n)=x1(n)+x2(n)and g(n)= (x1(n)+x2(n))WNn .Similarly
the N/2 DFT s are divided and expressed in to the combination of N/4 point
DFTs. This process is continued until we left with 2-point DFTs

80. What are the differences between DIT and DIF algorithms?
For DIT the input is bit reversed and the output is in natural order ,and in
DIF the input is in natural order and output is bit reversed. In butterfly the phase
factor is multiplied before the add and subtract operation but in DIF it is
multiplied after add-subtract operation
81. What is the basic operation of DIT algorithm?
The basic operation DIT algorithm is called butterfly in which two inputs
G(n) and H(n)are combined to give x1(k) and x2(k)
x1(k)= G(n)+WNkH(n)
x2(k)= G(n)-WNkH(n)
k
WN is the twiddle factor

82. What is the basic operation of DIF algorithm?

 The basic operation DIF algorithm is called butterfly in which two inputs
G(n) and H(n)are combined to give x1(k) and x2(k)
x1(k)= G(n)+ H(n)

x2(k)={G(n)- H(n)} WNk

WNk is the twiddle factor

83. Draw the flow-graph of a two-point DFT for a decimation in time
decomposition
The flow-graph of a two-point DFT for decimation in time algorithm is

G(n) x1(k)= G(n)+ H(n)

W20
x2(k)= G(n)- H(n)
H(n)

84. Draw the flow-graph of a two-point DFT for a decimation in frequency

decomposition
The flow graph of a two point DFT for decimation in frequency algorithm
is

G(n) W20 x1(k)= G(n)+ H(n)

x2(k)= G(n)- H(n)

H(n)

85. Draw the basic butterfly diagram for decimation in time algorithm
The flow graph of a two point DFT for a decimation in time algorithm is

G(n) x1(k)= G(n)+WNkH(n)

WNk
x2(k)= G(n)- WNkH(n)
H(n)
 G(n) and H(n) are inputs and x1(k) ,x2(k) are outputs WNk is phase factor

86. Draw the basic butterfly for a decimation in frequency decomposition

The butterfly of a two point DFT for a decimation in frequency algorithm
is

G(n) WNk x1(k)= G(n)+ H(n)

H(n)
x2(k)= {G(n)- H(n)} WNk
G(n)andH(n) are inputs and x1(k) ,x2(k) are outputs WNk is phase factor
87. Arrange the 8 point sequence x(n)={1,2,3,4,-1,-2,-3,-4} inn bit reversed order.
Normal order x(n)={1,2,3,4,-1,-2,-3,-4}
Bit reversal order x(n)={1,-1,3,-3, 2,-2,4,-4}
88. How we can calculate IDFT using FFT algorithm?

-The inverse DFT of N point sequence x(k) is defined as

N-1
X(n)=(1/N ) ∑[x*(k)WNnk ] * n=0,1,2,3,…N-1
k=0
a. Take conjugate of x(k)
b. Compute N point DFT of x*(k) using radix 2 FFT.
c. Take conjugate of output sequence.
d. Divide the output sequence by N.
89. What are the applications of FFT?
1. linear filtering
2. correlation
3. spectrum analysis
90. What are the twiddle factors involved in the first stage of computation in 8
point DIT radix-2, FFT algorithm?
W80, W81, W82, W83
91.What is meant by in place in DIT and DFT algorithm
An algorithm that use the same location to store both the input and output
sequence is called inplace algorithm

UNITIV

92.What is filter?
Filter is a frequency selective device ,which amplify particular range of
frequencies and attenuate particular range of frequencies.

93.What are the types of digital filter according to their impulse response?

IIR(Infinite impulse response )filter

 FIR(Finite Impulse Response)filter.

94. How phase distortion and delay distortion are introduced?

The phase distortion is introduced when the phase characteristics of a filter is
nonlinear with in the desired frequency band.
The delay distortion is introduced when the delay is not constant with in the desired
frequency band.

95.what are FIR filters?

The filter designed by selecting finite number of samples of impulse response (h(n)
obtained from inverse fourier transform of desired frequency response H(w)) are called
FIR filters

96. Write the steps involved in FIR filter design

Choose the desired frequency response Hd(w)
Take the inverse fourier transform and obtain Hd(n)
Convert the infinite duration sequence Hd(n) to h(n)
Take Z transform of h(n) to get H(Z)

97. What are advantages of FIR filter?

Linear phase FIR filter can be easily designed .
Efficient realization of FIR filter exists as both recurrisive and non recursive structures.
FIR filter realized non recursively are stable.
The round off noise can be made small in non recursive realization of FIR filter

98. what are the disadvantages of FIR FILTER

The duration of impulse response should be large to realize sharp cutoff filters.
The non integral delay can lead to problems in some signal processing applications.

99.what is the necessary and sufficient condition for the linear phase characteristic
of a FIR filter?
The phase function should be a linear function of w, which inturn requires
constant group delay and phase delay.

100. List the well known design technique for linear phase FIR filter design?
Fourier series method and window method
Frequency sampling method.
Optimal filter design method.

101.Define IIR filter?

The filter designed by considering all the infinite samples of impulse response are
called IIR filter.
102 State the properties of Butterworth filter
Magnitude response decreases monotonically as the frequency is increases from o toα
Magnitude response is normally that about w=0 in all the derivative up to the order N are
equal to zero at w=0.the phase response curve approaches –Nπ /2 for large w ,where N is
number of poles on the butterworth circle in the left half s plane.

103. State the properties of chebyshev filter

The magnitude response has ripple in either the passband or stopbandThe poles of
the chebyshev filter to on an ellipse

104.What is the power transfer function of butter worth filter?

|H(jw)|2=1/{1+[w/wc]2N}
where wc is cut off frequency
105.What is the power transfer function of chebyshev filter?
|H(jw)|2=1/{1+ε2CN2[w/wc] }
where wc is cut off frequency
CN (w/wc ) chebyshev polynomial
ε Measure of allowable ripple in pass and or stop band

106.Write down Chebyshev’s polynomial for N=0,1,2

For N=0 CN (w) =1, For N=1 CN (w) =w, For N=2 CN (w) =2w2-1

107.Give the recursive relationship of chebyshev’s polynomial?

CN+1 (w)=2 wCN (w) - CN-1 (w) for N>1
CN (w)=cos(ncos-1(w))

108.How will you obtain digital filter from analog filter?

Response invariance method and Bilinear transform method

109.What are the steps involved in design a digital filter?

Convert the given digital filter’s specification in to analog low pass filter
Find the order by suitable approximation and find transfer function
Demoralize the transfer function and apply frequency transformation formula to get the
desired filter.
Convert the analog transfer function in to digital by Impulse invariant or Bilinear
technique.

110.What is impulse invariant method

The transformation of analog filter to digital filter with out modifying the impulse
response of the filter is called impulse invariant transform.

111. What is aliasing?

The phenomena of high frequency sinusoidal components acquiring the identity of low
frequency sinusoidal components after sampling is called aliasing. It will arise if the
sampling rate does not satisfy the Nyquest rate.
112.Convert the analog filter H(S)=1/(S+1) into Digital one by Impulse invarient
method?

H(S)=1/S+1
1/S+Pi→1/(1-e-PiT z-1)
Let T=1sec
H(z)= 1/(1-e-1 z-1)
=1/(1-0.368 z-1)

113.What are the disadvantages of Impulse invariance method?

This method is unsuccessful for implementing digital filters for which |H(jw)| does not
approaches zero for large values of w (I t is not applicable for high pass, band pass
design)

114.Plot the magnitude response of Butterworth filter

|H(jw)|

1
A1

A2 w
wp ws

114.Give an example of a simple Bilinear transform

S=2/T {z-1/z+1}

115. State the advantages of Bilinear transform

(1)No aliasing
(2)Provides one to one mapping procedure

116.What is Gibbs phenomenon?

In the design of FIR filter the infinite duration impulse response is truncated in to
finite duration impulse response.this abrupt truncation introduces oscillations in pass
band and stop band .This effect is known as Gibbs phenomenon

117. What are the types of filters according to their frequency responses?
Low pass filter
High pass filter
Band pass filter
Band stop filter
UNITV
118. What are the addressing modes available in conventional DSP processor?
Register addressing
Direct addressing
Immediate addressing
Indirect addressing
Post increment/decrement by one
Post increment/decrement by value
119. What are the special DSP indirect addressing modes?
Bit reversal (reverse carry) addressing
Circular addressing
120. What are the classifications in the architecture of DSP?
Integer (fixed point)
Floating point
121. Give examples of fixed point and floating point processors?
Fixed point processors
TMS320C50
ADSP2100
Floating point processors
TMS 320C30
ADSP21020
122. Give some manufacturers of DSP processor
Motorola
Texas instruments
Analog devices
AT&T
123. Compare TMS320C50 and ADSP2100

TMS320C50 is made by Texas instruments

ADSP2100 is made by Analog devices
Both are having 16 bit data word size
Both are fixed point processors
The addressing modes are similar except modulo addressing is not there in ADSP2100
124. Give some types of instructions in ADSP2100
ALU based
Data memory write
ALU/MAC with data memory writes or read
Conditional ALU/MAC
DO until
Shift immediate………..etc
125. How many types of instructions in ADSP2100 available
There are 31 types of instructions available in ADSP2100 family
126. What are the types of rounding?
Round to nearest (unbiased rounding)
Round towards zero (truncation)
127. What are the data formats of ADSP2100 processor?
32 bit single precision IEEE floating point
40 bit extended single precision 32 bit fixed point with 80 bit accumulation
128. What are the functions of programme sequencer in ADSP2100 processor?
Selective cache for three bus performance
Zero overhead DOLOOP control
Stack control and nesting
129...What is MAC?
MAC is Multiplier/Accumulator like ALU connects directly to the PMD, DMD, and R
busses
130. Give some software tools for ADSP family devices?
ADSP2101 EZ-Lab kit which includes EZ-Lab demonstration board, an ADSP2101-5
family assembler, linker, and other softwares.
PC-based software simulator
131. Give some software tools for TMS family processor
TMS320 programmers interface(C/Assembly source debugging, standard evaluation
module(EVM),Analog interface board(AIB2)emulator(XDS s –Extended development
system and TMS 320 XDS/22 upgrade package
132.What are the features of TEXAS Instruments family of DSP devices
High degree of parallelism Specialized instruction sets provides speed and flexibility
CMOS family capable of executing up to 50 million floating point operations per second

Part B

UNIT I
1).Define the following signals
Unit sample sequence
Unit step sequence
Unit ramp sequence
Exponential sequence
When you answer this question keep in mind that sequence stands for discrete time
signals. Definition of each sequences with proper drawings (Refer class notes)

2)For each of the following system determine if the following system is (1)
Linear(2)Time invariant
(i)Y(n)=nX(n)
(ii)Y(n)=nX2(n)
First check for linearity using superposition and homogeneity principles(2marks)
Check for time invariant (2marks)
Ans i)Linear but time variant
ii)Nonlinear and time variant {Refer classnotes}
3)Clearly define the following with suitable examples(1)Linear system(2)time
invarient system(3)Causal system
Refer class notes
4)Discuss about different types of systems with examples

Refer class notes& salivahanan book pp17-20

5)Consider a causal and stable LTI system whose input x(n) and output y(n)are
related through the second order differential equation
Y(n)-1/6Y(n-1)-1/6Y(n-2)=X(n).Determine the impulse response h(n) of the system
Find the roots of auxiliary equation m1,m2
Write the complementary solution A1(m1)n +A2(m2)n
Apply the initial conditions and find the values of A1,A2 (Refer classnotes)
UNITII
1.Discuss the properties of z transform
Statement and proof are important
Ref assignment-I ,Class notes & salivahanan book pp203-213
2.Discuss the properties of DFT
Statement and proof are important
Ref assignment-I, Class notes & salivahanan book pp308-311

3.Discuss the properties of DTFT

Statement and proof are important
Ref assignment-I, Class notes
4.Determine the output response Y(n) if H(n)={1,1,1},X(n)={1,2,3,1} by using linear
convolution and circular convolution
By linear convolution H(n)*X(n)=Y(n)
By circular convolution H(n) consists of 3 samples and X(n) consists of 4 samples There
fore padding with 3zeros in H(n)and two zeros in X(n) to make each in to 6 point
sequence now H(n)=(1,1,1,0,0,0) and X(n)=(1,2,3,1,0,0,) perform circular convolution of
padded sequence of X(n) and H(n) implies Y(n)
{Refer classnotes)
5.Find the output of the filter whose impulse response h(n)={1,1,1} and the input
signal x(n) is {3,-1,0,1,3,2,0,1,2,1} Using overlap save and overlap add method
Overlap and add
Discard X(n) in to three sample sequences
Perform linear convolution of each pieses with h(n)
Write in the order of overlapping =2
Add the overlapped samples
Resultant sequence consists of 12 samples
Overlap and save method
Discard X(n) in to Fivesample sequence by overlapping two samples
Padd two zeros with h(n)
Perform circular convolution of each pieses with h(n)
 Write in the order of overlapping =2
discard the overlapped samples
Resultant sequence consists of 12 samples

Refer classnotes

6.Consider a stable linear time invariant system,

1
H(z)=
(1-1/2 Z-1)(1+1/3 Z-1)
Find it’s impulse response using convolution

Take H(z)=X1(Z)xX2(Z)

X1(n)=(1/2)n U(n) X2(n)=(-1/3)n U(n)

X1(n)* X2(n)=H(n) (Convolution property)
(Refer classnotes}

7.Find the causal signal x(n) whose z transform is given by

X(z)=1/ (1-1.5 Z-1+0.5 Z-2)
Since x(n) is causal we seek a power series expansion in negative powers of Z
{refer class notes}
Ans {1,3/2,7/4.15/8……}

UNITIII

1Derive a radix-4 decimation in time FFT for N=4v, and draw the corresponding flow
graph for v=4
refer classnotes and salivahanan book pp360-364

2.Compute the IDFT of the sequence

X(K)={28,-4+j9.656,-4+4j,-4+j1.656,-4,-4-j1.656,-4-4j,-4-j9.656} using DIT
algorithm
Find X*(k)
Apply in algorithm
Multiply1/N
Get the answer X(n)={0,1,2,3,4,5,6,7} Refer notes

3.By deriving necessary equation and drawing the relevant flow graph explain the
radix-2 DIF FFT algorithm
Ref class notes & Salivahanan book pp334-340

4) Using DIF FFT algorithm Compute the 8 point DFT of the

X(n)= {1,2,3,4,4,3,2,1}.Draw the flow graph and show the intermediate results
Ans
Given N=8
X(k)={20,-5.828-j2.414,0,-0.172-j0.414,0,-0.172+j0.414,0,-5.828+j2.414}
{Refer classnotes}
5.By deriving necessary equation and drawing the relevant flow graph explain the
radix-2 DIT FFT algorithm
Refer classnotes &salivahanan book pp 320-324

UNITIV

1. A digital band pass filter is required to meet the following

specification.

Monotic pass band : ≤ 1 db

Pass band edge : 4 Khz & 6 Khz
Stop band attenuation : ≥ 40 db(monotonic)
Stop band edge : 3 Khz & 7 Khz
Sample frequency : 24 Khz
Use Bilinear transformation technique
Monotonic passband and stopband (butterworth)
Convert f to ω
Convert bandpass specification in to lowpass specification(Ωp=1 & find Ωs min)
Convert di gital ω in to analog frequency by bilinear transform(ΩT/2=tan(ω /2)
Convert the db into A1,A2
Find the order of lowpass filter
Find the butterworth transferfunction
Apply frequency transformation(L.P toB.P)
Apply Bilinear transformation(H(s)toH(z)){s=Z-1/Z+1}
Refer classnotes
2.Design a digital butterworth filter that satisfies the following constraint using
bilinear transform

0.9 ≤ |H(ejw) | ≤1 0 ≤ ω ≤ π/2

|H(ejw) | ≤0.2 3π/4 ≤ ω ≤ π

The given specification is for low pass filter and select Butterworth approximation
A1=0.9,A2=0.2, ωp= π/2, ωs=π/2
 Convert digital ω in to analog frequency by bilinear transform(Ω=2/Ttan(ω /2)
Find the order of lowpass filter
Find the butterworth transferfunction .Find Ωc
Apply frequency transformation(L.P toL.P ie S=S/ Ω c)
Apply Bilinear transformation(H(s)toH(z)){s=Z-1/Z+1}
Refer classnotes

3. A LPF is designedwith the following frequency response

Hd(ejw) = e-2jw | ω |≤π/4
=0 π/4 ≤ | ω |≤π
using rectangular window
From gn data (N-1)/2=2
N=5
Find Hd(n) by inverse fourier
H(n)= Hd(n)xW(n)
W(n)={1 0 ≤ n≤ 4
0 otherwise}
Take Z transform and find H(z)
4.Explain impulse invariance and bilinear transformation method of designing IIR
digital filter
refer class notes
5.Design and realize a digital LPF using Bilinear transformation method to satisfy the
following characteristics
(i)Monotonic stop and passband
(ii)-3.01 db cutoff frequency at 0.5 π
(iii)Magnitude down atleast 15db at 0.75 π
Refer classnotes
6.Explain the needs for the use of window sequences in the design of FIR filter.Discuss
the desirable properties of a window sequence.Describe the window sequences
generally used and compare their properties.

Refer classnotes and Nagoorkani book pp281-294

(7) Show how an FIR filter gives linear phase characteristic

Refer classnotes and Nagoorkani book pp261-264

UNITV

1.With functional block diagram explain the architecture of ADSP2181 processor.

Refer classnotes and Xerox material

2 With functional block diagram explain the architecture of TMS320C50 DSP

processor.
Refer classnotes and Xerox material
3.Compare the architecture of ADSP 2181 family of DSP processor
with that of conventional Microprocessor in terms f performance.
Compare the architecture of ADSP 2181 family of DSP processor
with 8085 Microprocessor and explain about their functional blocks
Refer classnotes and Xerox material

4. Draw and explain how the architecture of a general purpose single

chipmicrocontroller architecture with an ALU and multiplier
resulting in a 32 bit internal Harvard architecture of a
TMS320C50 processor.
Compare the architecture of TMS320C50 family of DSP processor
with 8051 microcontroller and explain about their functional blocks
Refer classnotes and Xerox material