DEPARTMENT OF ELECTRICAL AND ELECTRONICS ENGINEERING

EE6403 DISCRETE TIME SYSTEMS AND SIGNAL PROCESSING

UNIT – I : INTRODUCTION
PART – A (2 Marks)

1. What is meant by signal and signal Processing?

A signal is defined as any physical quantity that varies with time, space, or any other independent variable.

Signal processing is any operation that changes the characteristics of a signal. These characteristics include

the amplitude, shape, phase and frequency content of a signal.

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2. What are the classifications of signals?
There are five methods of classification of signals based on different features:

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a) Based on independent variable.
(i) Continuous time signal
(ii) Discrete time signal
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b) Depending upon the number of independent variable
(i) One dimensional signal
(ii) Two dimensional signal
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(iii) Multi-dimensional signal
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c) Depending upon the certainty by which the signal can be uniquely described as.
(i) Deterministic signal
(ii) Random signal
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d) Based on repetition nature

(i) Periodic signal
(ii) Non-Periodic signal
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e) Based on reflection
(i) Even signal
(ii) Odd signal

3. Define – Discrete System

Discrete time system is defined as a device or algorithm that operates on a discrete time input signal x(n) ,
according to some well-defined rule, to produce another discrete – time signal y(n) called the output signal.

4. What are the classifications of discrete time systems?

 The classifications of discrete time systems are:
i. Static and Dynamic system
ii. Time-variant and time-invariant system
iii. Linear and non-linear system
iv. Stable and Unstable system
v. Causal and non-causal system

5. Differentiate Continuous time from discrete time signal.

Continuous time signal: It is also referred as analog signal i.e., the signal is represented continuously in time.

Discrete time signal :Signals are represented as sequence at discrete time intervals

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6. Define – Digital Signal
A discrete time signal or digital is defined as which discrete valued represented by a finite number of digits is
referred to as a digital signal.

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7. What is deterministic signal? Give example.
A signal that can be uniquely determined by a well-defined process such as a mathematical expression or rule, or
look-up table is called deterministic signal.
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Example: A sinusoidal signal v(t )  Vm sin t
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8. What is random signal?
A signal that is generated in a random fashion and cannot be predicted ahead of time is called a random signal.
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Example : Speech signal , ECG signal and EEG signal.

9. Define – Periodic Signal and Non-periodic Signal

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Periodic signal: A periodic signal is defined as the signal x(n) is periodic with period N if and only if x(n+N) = x(n) for
all n.

Non-periodic signal: A non-periodic signal is defined as if there is no value of N that satisfies the equation x(n+N) 
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x(n).

10. What are the symmetric and antisymmetric signals?

Symmetric signal: A real valued signal x(n) is called symmetric if x(-n) = x(n)

Anti-symmetric signal: A signal x(n) is called anti-symmetric if x(-n) = -x(n).

11. What are energy and power signals? (M-13, M-15)

Energy signal:
 The energy of a discrete time signal x(n) is defined as E  n  x(n)
 2

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

Power signal :

The average power of a discrete time signal x(n) is defined as

N 2
1
P  lim
N  2N  1
 x(n)
n  N

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

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12. What are the different types of signal representation?
The different types of signal representation are:
1. Graphical representation

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2. Functional representation
3. Tabular representation
4. Sequence representation lts
13. What are the different types of operations performed on discrete-time signals?
The different types of operations performed on discrete-time signals are:
i. Delay of a signal
ii. Advance of a signal
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iii. Folding or Reflection of a signal
iv. Time scaling
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v. Amplitude scaling
vi. Addition of signals
vii. Multiplication of signals\
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14. Represent the following duration sequence x(n)={1, 3, –1, –4} as a sum of weighted impulse sequences.

Given:
x(n)={1, 3, -1, -4}
2b


We can write
x(n)  1 x(k ) (n  k )
2

 1 (n  1)  3 (n)  1 (n  1)  4 (n  2)

15. What is a static and dynamic system?

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

16. What is a time-invariant system? (N-12)

A system is called time-invariant if its input-output characteristics do not change with time.
Ex.:y(n) = x(n) + x(n-1)

17. What is a causal system?

A system is said to be causal, if the output of the system at any time depends only on present and past inputs, but

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does not depend on future inputs.
This can be expressed mathematically as,

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y(n) = F[x(n), x(n-1), x(n-2)………]

18. Define – Stable System

Any relaxed system is said to be Bounded Input Bounded Output (BIBO) stable if and only if every bounded input
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yields a bounded output. Mathematically, their exist some finite numbers, Mx and My such that,
x(n)  Mx   and y(n)  My  
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19. What is a linear system?
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A system that satisfies the superposition principle is said to be a linear system. Superposition principle states that ,
the response of the system to a weighted sum of signals be equal to the corresponding weighted sum of the outputs
of the system to each of the individuals input signals.
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20. Define – Unit Sample Response (Impulse Response) of a system

The unit sample response is defined as the output signal designated as h(n), obtained from a discrete time system
when the input signal is a unit sample sequence (unit impulse).
2b

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

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

The necessary and sufficient condition for causality of an LTI system is, its unit sample response h(n) = 0 for
negative values of n i.e.,
 h(n) = 0 for n<0

22. What is the necessary and sufficient condition on the impulse response for stability?
The necessary and sufficient condition guaranteeing the stability of a linear time-invariant system is that its impulse
response is absolutely summable.


 h(k )  


23. What is meant by discrete or linear convolution?

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

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
y ( n)   x ( k ) h( n  k )
k  

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(OR)

y ( n)   h( k ) x ( n  k )
k  
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24. What are the steps involved in the convolution process?
The steps involved in the convolution process are
1. Express both sequences in terms of the index k.
2. Folding: Fold the h(k) about the origin and obtain h(-k).
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3. Time shifting: Shift the h(k) by n units to right if n is positive or left if n is negative to obtain h(n-k).
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4. Multiplication: Multiply x(k) by h(n-k) to obtain w(k)=x(k)h(n-k)

5. Summation: Sum all the values of the product w(k) to obtain the value of output y(n).
6. Increment the index n, shift the sequence h(n-k) to right by one sample and do step4.
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25. What is meant by sampling process?

Sampling is the conversion of a continuous time signal(or analog signal) into a discrete – time signal obtained by
taking samples of the continuous time signal ( or analog signal ) at discrete time instants.
2b

26. State sampling theorem. (A-11, N-11, A-14)

A sampling theorem states that the band limited continuous time signal with highest frequency(band width)
fmhertz , can be uniquely recovered from its samples provided that the sampling rate f sis greater than or equal to 2fm
samples per second.

27. Define – Nyquist Rate (A-12, N-13, A-14, A-15)

The Nyquist rate is defined as the frequency 2fm, which under sampling theorem, must be exceeded by the sampling
frequency.
28. What is meant by quantization process?
The process of converting a discrete time continuous valued signal into discrete time discrete valued signal is called
quantization.

29. What is meant by aliasing effect? (A-11, N-12, N-14)

The superimposition of high frequency component on the low frequency is known as “frequency aliasing” or
“aliasing effect”.

30. How can aliasing be avoided?

To avoid aliasing the sampling frequency must be greater than twice the highest frequency present in the signal.

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31. What is an anti-aliasing filter? (A-16)

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A filter used to reject frequency signals before it is sampled to reduce the aliasing is called an anti-aliasing filter.

32. What is meant by critical sampling?

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If the sampling frequency is exactly equal to the Nyquist rate is known as critical sampling.
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33. What are the steps involved in the A/D conversion?
The steps involved in the A/D conversion are:
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1. Sampling
2. Quantization
3. Coding

34. What is meant by quantization error?

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It is the difference between the quantized value and actual sample value.
eq(n) = xq(n) - x(n)
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35. What is meant by quantization level?

The value that allows in a digital signal is called the quantization level.

36. Define – Resolution or Quantization Step Size

The resolution is defined as the distance between two successive level.

37. What is meant by SQNR?

 The quality of the output of the A/D converter is usually measured by the Signal to Quantization Noise Ratio
(SQNR), which is ratio of the signal power to noise power.
P
SQNR  av
Pq

38. What is the use of a sample and hold circuit?

The sample and hold circuit is used to hold the sample the analog signal and hold the sampled value constant as
long as the A/D converter takes time for accurate conversion.

39. Define – Conversion Time

Conversion time may be defined as the time taken by an ADC for converting a given amplitude, expressed in

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decimal value, of a quantized analog signal applied across its input terminals into corresponding binary – equivalent
value.

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40. Define – Voltage Resolution

The voltage resolution is defined as

Votagereso lution 
VFS
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2n  1

Where, VFS - full scale voltage and n - number of bits.

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42. Define – Percentage Resolution

Percentage resolution is defined as
1
%resolution  n 100
2
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41. What are the advantages and disadvantages of counter-ramp type ADCs?
The advantages of counter-ramp type ADCs are:
1. The principle is simple and straightforward.
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2. It is very easy to construct this ADC.

3. This basic principle is employed in many advanced ADCs.
The disadvantages counter-ramp type ADCs are:
1. Only increasing voltages can be measured.
2. The system is very slow.
3. This may be mainly used to read DC voltages.

42. What are the advantages and disadvantages of SAADC?

 The advantages of SAADC are:
1. It is more accurate than the stair case (counter ramp ) ADC.
2. It maintains a high resolution.
3. It is much faster.
4. Its conversion time is much less.

The disadvantages of SAADC are:

1. It requires a complex register called the successive approximation register.
2. It is costly, as it contains more components.

43. Write the formula for conversion time of SAADC?

n
Conversiontime 
f

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where,
n – number of bits and
f – clock frequency

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44. What are the advantages and disadvantages of flash converter?
The advantages of flash converter are:
1. The fastest conversion process (governed only by propagation delay of the gates.
2. Highest accuracy
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3. Highest resolution possible by increasing the number of comparators.
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The disadvantages of flash converter are:
1. Very complicated circuitry
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2. Cost is proportional to the number of comparators, which in turn depends on the resolution required.

45. What are the different types of analog to digital converters?

The different types of analog to digital converters are:
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1. Flash A/D converter

2. Successive approximation converter
3. Counting type A/D converter
4. Over sampling Sigma – Delta converter
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46. What are the different types of digital to analog converter?

The different types of digital to analog converters are
1. Weighted – resistor D/A converter
2. Resistor – ladder D/A converter
3. Over sampling D/A converter

47. Define – Z-transform

 The Z-transform of a discrete time signal or sequence is defined as the power series.

X ( z)   x ( n) z
n  
n

48. What is meant by region of convergence? (M-11,N-11,N-12, M-14)

The region of convergence (ROC) of X(z) is the set of all values of z for which X(z) attains a finite value.

49. What are the properties of region of convergence? (N-12)

The properties of region of convergence are:
1. The ROC is a ring or disk in the Z – plane centered at the origin.
2. The ROC cannot contain any poles.
3. The ROC of an LTI stable system contains the unit circle.

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4. The ROC must be a connected region.

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50. What are the properties of z-transform? (N-12)
1. Linearity: z a1 x1 (n)  a2 x2 (n)  a1 X 1 ( z)  a2 X 2 ( z)
 m 1

2. Shifting: (a) zx(n  m)  z m  X ( z )   x(i) z mi 
 i 0 
3. (b) zx(n  m)  z  m X ( z )
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 
m
 d 
4. Multiplication: z n x(n)    z  X ( z )
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 dz 
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
5. Scaling in z- domain: z a n x(n)  X a 1 z   
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6. Time reversal : zx(n)  X ( z )

1


  
7. Conjugation: z x (n)  X ( z ) 
 n 
8. Convolution: z  h(n  m)r (m)  H ( z ) R( z )
 m 0 
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9. Initial value: zx(0)  Lt X ( z )

z 

10. Final value: zx( )  Lt (1  z 1 ) X ( z )

z 1
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51. State Parseval‟s relation in z-transform.

Parseval‟s relation in z-transform state that
If x1(n) and x2(n) are complex valued sequences, then
1 1

n 
x1 (n) x2 (n)  
2j c
X 1 (v) X 2   v 1dv
v 

52. What is the relationship between z-transform and DTFT? (A-11, A-12)
 
The z-transform of x(n) is given by X ( z )   x ( n) z
n  
n
…….(1)

where, z  re j
Substituting z value in eqn (1) we get,

X (re j )   x(n)r n e  jn …………… …..(2)

The Fourier transform of x(n) is given by


X (e j )   x ( n)e
n  
 jn
…………………… .(3)

Eqn(2) and Eqn(3) are identical , when r=1. In the z-plane this corresponds to the locus of points on the unit
circle z  1 . Hence X (e j ) is equal to X(z) evaluated along unit circle , or
X (e j )  X ( z ) | z e j

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j
For X (e ) to exist, the ROC of X(z) must include the unit circle.

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53. What are the different methods of evaluating inverse z-transform?
The different methods of evaluating inverse z-transform are:
i. Long division method
ii.
iii.
Partial fraction method
Residue method
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iv. Convolution method
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54. Find the convolution of the following using z-transform. (A-12, A-15, A-16)

 
x(n)  1,1,1 ; h(n)  1, 2,1

 

Solution:
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Z  x(n)  h(n)  X ( z ) H ( z )
X ( z ) (1  z 1  z 2 )
H ( z )  (1  2 z 1  z 2 )
2b

X ( z ) H ( z )  (1  3 z 1  4 z 2  3 z 3  z 4 )
x(n)  h(n)   1,3, 4,3,1 

55. Define – System Function

Let x(n) and y(n) be the input and output sequences of an LTI system with impulse response h(n). Then the
system function of the LTI system is defined as the ratio of Y(z) and X(z), i.e.,
 Y ( z)
H ( z) 
X ( z)
where,
Y(z) is the z – transform of the output signal y(n)
X(z) is the z – transform of the input signal x(n)

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PART – B (16 Marks)

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Classification of systems : Linear, causal, stable, dynamic, time variance, recursive

2b

1. Check the linearity, causality, time invariance, stability and dynamicity properties of the following
equations :
(a) y(n) =x(2n )
(b) y(n) = cos ( x(n) )
(c) y(n) = x( n ) cos (x( n ))
(d) y(n) =x(− n + 2 )
(e) y(n) =x(n) +n x (n+1) (N-07)
2. Determine the system stability and causality for each of the impulse response given below :
(a) h(n)=sin (π n / 2)
(b)h(n) = δ (n ) + sin π n (A-11)
(c) h(n) = 2 n u ( − n)
(d) h(n)= e2n u ( n – 1)

3. Determine the linearity and time invariance of the given equations

(a) y ( n ) = n x( n ) ; (b)y(n) = a x(n) + b (A-08)

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4. Explain in detail, recursive and non-recursive discrete time systems.

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Classification of signals : Continuous and Discrete, energy and power, mathematical representation

5. What is meant by energy and power signals? Determine whether the following signals are energy
signal or power signal.
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(a) x1 ( n ) = ( 1 / 2 ) n u(n)
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(b) x2 (n) = sin ( ᴨ n / 6 )
(c) x3 (n) = e j [ ( n ᴨ / 3 ) + ( ᴨ / 6 ) ]
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(d) x4( n ) = e 2n u( n )

6. Explain the following signals.:

(a) Continuous time signal and deterministic signals
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(b) Even and odd signals

(c) Periodic and non-periodic signals
(d) Energy and Power signals (A-11)
2b

7. Check the periodicity of the following signals :

(a) x(n) = ej6πn
(b) x(n) = cos 2πn

8. Perform addition and multiplication of the given discrete time signals:

 x1 (n) = { 2, 2, 1, 2 } and x2 (n) = { − 2, − 1, 3, 2 }

Sampling Techniques, Quantization

9. What is meant by sampling theorem? Explain sampling theorem in detail. (N-12)

10. Describe in detail, the process of sampling and quantization.

Digital Signal Representation

11. Name any four application areas of digital signal processing with suitable example.

12. Explain in detail, A to D conversion with suitable block diagram.

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UNIT – II : FAST FOURIER TRANSFORM

PART – A (2 Marks)
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1. Define – Fourier Transform of a discrete time signal. (N-12)

The Fourier transform of a discrete time signal x(n) is defined as

F x(n)  X ( )   x ( n)e  jn
2b

n  

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

By taking Fourier transform of a discrete time signal x(n) , it is decomposed into its frequency components. Hence
the Fourier transform is called signal spectrum.

3. List out the difference between Fourier transform of discrete time signal and analog signal.
 1. The FT of analog signal consists of a spectrum with a frequency range   to  . But the FT of discrete
time signal is unique in the range   to   ( or 0 to 2 ) , and also it is periodic with periodicity of 2  .
2. The FT of analog signals involves integration but FT of discrete time signals involves summation.

4. Define – Inverse Fourier Transform

The inverse Fourier transform of X() is defined as

F 1 X ( )  x(n) 
1
 X ( )e
jn
d
2 

5. List out some applications of Fourier transform.

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The applications of Fourier transform are
i. The frequency response of LTI system is given by the Fourier transform of the impulse response of the
system.

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ii. The ratio of the Fourier transform of output to Fourier transform of input is the transfer function of the
system in frequency domain.
iii. The response of an LTI system can be easily computed using convolution property of Fourier transform.
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6. What is the frequency response of LTI system?
The Fourier transform of the impulse response h(n) of the system is called frequency response of the system. It is
denoted by H().
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7. Write the properties of frequency response of LTI system.

The properties of frequency response of LTI system are
i. The frequency response is periodic function  with a period of 2  .
ii. If h(n) is real then H ( ) is symmetric and H ( ) is antisymmetric.
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iii. If h(n) is complex then the real part of H ( ) is antisymmteric over the interval 0    2 .
iv. The frequency response is a continuous function of  .
2b

8. Write short notes on the frequency response of first order system.

A first order system is characterized by the difference equation
y(n)  x(n)  ay(n  1)
The frequency response of first order system depends on the co efficient “a” in the difference equation governing
the LTI system. When the value of “a|” is in the range of 0<a<1, the first order system behaves as a low pass filter.
When the value of “a” is in the range –1<a<0, the first order system behaves as a high pass filter.
9. Write short notes on the frequency response of second order system.
A second order system is characterized by the difference equation
y(n)  2r cos 0 y(n  1)  r 2 y(n  2)  x(n)  r cos 0 x(n  1)
The frequency response of second order system depends on the parameters “r” and “  0 ” in the difference
equation of the LTI system. When the value of r is in the range of 0<r<1, the second order system behave as a
resonant filter with center frequency  0 . When the value of r is varied from 0 to 1, the sharpness of resonant peak
will increase.

10. Define – Discrete Fourier Series (N-14)

Consider a sequence xp(n) with a period of N samples so that xp(n)=xp(n/N); Then the discrete Fourier series of the
sequence xp(n) is defined as

m
N 1
X p (k )   x p (n)e  j 2kn / N
n 0

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11. What are the two basic differences between the Fourier transform of a discrete time signal with the Fourier
transform of a continuous time signal?
i. For a continuous signal, the frequency range extends from   to  . On the other hand, the frequency
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range of a discrete – time signal extends from   to   ( or 0 to 2 ) .
ii. The Fourier transform of a continuous signal involves integration, whereas, the Fourier transform of a
discrete – time signal involves summation process.
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12. Find the Fourier transform of a sequence x(n) = 1 for  2  n  2 (N-12)
= 0 otherwise.
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Solution:
 2
X ( )   x(n)e  jn 
n  
e
n  2
 jn

 e j 2  e j  1  e  j  e  j 2 1  2 cos   2 cos 2
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13. Define – Discrete Fourier Transformation of a given sequence x(n)

2b

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

N 1
X (k )   x(n)e  j 2kn / N k= 0, 1, 2……..N-1.
n 0

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

The N-point IDFT of a sequence X(k) is
 N 1
1
x ( n) 
N
 X ( k )e
K 0
j 2k / N
n= 0, 1 , 2 …..N-1 .

15. List out any four properties of DFT. (A-14)

1. Periodicity
If X(k) is N- point DFT of a finite duration sequence x(n), then
x(n  N )  x(n) for all n
X (k  N )  X (k ) for all k .

2. Linearity
If X1(k)=DFT[x1(n)] and
X2(k)=DFT[x2(n)],
then

m
DFT[a1x1(n)+a2x2(n)]=a1X1(k)+a2x2(k)

3. Time reversal of a sequence

.co
If DFT {x(n)}=X(k),
then
DFT{x((-n))N}=DFT{x(N-n)}=X((-k))N=X(N-k)

4. Circular time shifting of a sequence

If DFT {x(n)}=X(k),
lts
then
DFT{x((n-l))N}=X(k) e  j 2kl / N
u
res

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

(i) x  (n) (ii ) x  ( N  n) (iii ) x((n  l )) N (iv ) x(n)e j 2 ln/ N

y2

Solution:

(i ) DFT {x  (n)}  X  ( N  k )
(ii ) DFT {x  ( N  n)}  X  (k )
2b

(iii ) DFT {x((n  l )) N }  X (k )e  j 2kl / N

(iv ) DFT {x(n)e j 2 ln/ N }  X ((k  l )) N

n
1
17. Calculate the DFT of the sequence x(n) =   forN  16
4
 N 1
Solution: X (k )   x(n)e j 2kn / N K=0, 1 , 2…..N-1
n 0
n
15
1
    e  j 2kn / 16
n 0  4 
n
15
1 
   e  jk / 8 
n 0  4 
16
1
1    e  j 2k
  4
1
1  e  jk / 8
4

18. State the time shifting property of DFT.

m
The time shifting properties of DFT states that
 j 2kn / N
If DFT[x(n)] = X(k), then DFT[x((n-m))N] = e X (k )

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19. Find the DFT of the sequence x(n)={ 1,1,0,0 } (A-15)
Solution:

N 1
lts
X (k )   x(n)e  j 2kn / N k=0,1,2,…..N-1
n 0
u
3
  x (n)e  j 2 kn / 4 k  0,1, 2 , 3.
res

n 0

3
X (0)   x ( n)  {1  1  0  0}  2
n 0

3
X (1)   x (n)e  j n / 2  {1  j  0  0 } 1  j
y2

n 0
3
X (2)   x( n)e  j n  {1  1  0  0 }  0
n 0
3
2b

X (3)   x (n)e  j 3 n / 2  {1  j  0  0 } 1  j
n 0

X (k )  {2,1  j , 0,1  j }

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

If the sequence x(n) is real and odd (or) imaginary and even, then X(k) is purely imaginary.
21. When is the DFT X(k) of a sequence x(n) is real?
If the sequence x(n) is real and even (or) imaginary and odd , then X(k) is purely real.

22. State the circular frequency shifting property of DFT. (A-14)

The circular frequency shifting property of DFT states that

If DFT[x(n)]=X(k), then DFT[x(n) e j 2 ln/ N ]  X ((k  l )) N

23. What is meant by zero padding? What are its uses?(N-14)

The process of lengthening the sequence by adding zero – valued samples is called appending with zeros or
zero – padding.
Uses:

m
i. We can get “better display” of the frequency spectrum.
ii. With zero padding, the DFT can be used in linear filtering.

.co
24. What is meant by understand by periodic convolution?
Let x1 p (n) and x2 p (n) be two periodic sequences each with period N with
 
DFS x1 p (n)  X 1 p (k ) and
DFS x 2p ( n)   X 2p (k )
lts ……………………………(1)
If X 3 p (k )  X 1 p (k ) X 2 p (k )
then the periodic sequence x3 p (n) with Fourier series coefficients X 3 p (k ) can be obtained by periodic
u
convolution, defined as
res

N 1
x3 p (n)   x1 p (m) x2 p (n  m) ………………………………(2)
n 0

The convolution in the form of eqn(2) is known as periodic convolution, as the sequences in eqn(2) are all
periodic with period N, and the summation is over one period.
y2

25. Define – Circular Convolution

The convolution property of DFT is defined as the multiplication of the DFTs of the two sequence equivalent to the
2b

DFT of the circular convolution of the two sequences.

X 1 (k ) X 2 (k )  DFT {x1 (n)  x 2 (n)}
N 1
x3 (n)   x1 (m) x 2 ((n  m)) N
m 0

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

 Given two sequences x1 (n) and x2 (n) , the circular convolution of these two sequences
x3 (n)  x1 (n) Nx2 (n) can be obtained by using the following steps.
i. Graph N samples of x1 (n) , as equally spaced points around an outer circle in counter clockwise direction.
ii. Start at the same point as x1 (n) graph N samples of x2 (n) as equally spaced points around an inner
circle in clock wise direction.
iii. Multiply corresponding samples on the two circles and sum the products to produce output.
iv. Rotate the inner circle one sample at a time in clock wise direction and go to step 3 to obtain the next value
of output.
v. Repeat step 4 until the inner circle first sample lines up with the first sample of the exterior circle once
again.

27. Distinguish between linear and circular convolution of two sequences.

m
Sl. No. Linear Convolution Circular convolution
If x(n) is a sequence of L number of If x(n) is a sequence of L number of
samples and h(n) with M number of samples and h(n) with M number of

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1.
samples , after convolution y(n) will contain samples , after convolution y(n) will
N=L+M-1 contain N=Max(L,M) samples.
Linear convolution can be used to find the Circular convolution cannot be used to
2.
response of a linear filter. find the response of a filter.

3.
u
response of a linear filter.
lts
Zero padding is not necessary to find the Zero padding is necessary to find the
response of a filter.

28. What are the steps involved in circular convolution?

The circular convolution involves basically four steps as the ordinary linear convolution. They are:
res

i. Folding the sequence

ii. Circular time shifting the folded sequence
iii. Multiplying the two sequences to obtain the product sequence
iv. Summing the values of product sequence
y2

29. What are the different methods performing circular convolution?

The different methods of performing circular convolution are:
2b

i. Graphical method
ii. Stockhman‟s method
iii. Tabular array method
iv. Matrix method

30. Obtain the circular convolution of the following sequences

x(n)={1, 2, 1 }; h(n)={ 1, -2, 2 }
Solution:
 The circular convolution of the above sequences can be obtained by using matrix method.
h(0) h(2) h(1)   x(0)   y (0) 
 h(1) h(0) h(2)  x(1)    y (1) 
    
h(2) h(1) h(0)   x(2)  y (2)
 1 2  2  1   3 
  2 1 2   2   2 
    
 2 2 1  1   1
y (n)  3, 2,  1

31. How is obtain linear convolution obtained from circular convolution?

Consider two finite duration sequences x9n) and h(n0 of duration L samples and N samples respectively. The
linear convolution of these two sequences produces an output sequence of duration L+M-1 samples, whereas, the

m
circular convolution of x(n) and h(n) give N samples where N=Max(L,M) . In order to obtain the number of samples
in circular convolution equal to L+M-1, both x(n) and h(n) must be appended with appropriate number of zero valued
samples. In other words, by increasing the length of the sequences x(n) and h(n) to L+M-1 points and then circularly

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convolving the resulting sequences we obtain the same result as that of linear convolution.

32. What is meant by sectioned convolution?

lts
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 combined later to get the output.
u
33. What are the different methods used for the sectioned convolution?
The two methods used for the sectioned convolution are:
res

i. the overlap-add method

ii. the overlap-save method.

34. Differentiate overlap-add method from overlap – save method.

y2

Sl. No. Overlap – add method Overlap – save method

In this method the size of the input data In this method the size of the input data
1.
2b

block is N=L+M-1 block is L.

Each data block consists of the last M-1 Each data block has L points and we M-1
2. data points of the previous data followed zeros are appeared to compute N-point
by the L new data points. DFT.
In each output block, M-1 points are In this no corruption due to aliasing as
3. corrupted due to aliasing, as circular linear convolution is performed using
convolution is employed. circular convolution.
 To form the output sequence the first M-1 To form the output sequence, the last M-
data points are discarded in each output 1 points from each output block is added
4.
block and the remaining data is fitted to the first (M-1) points of the succeeding
together. block.

35. Distinguish between DFT and DTFT.

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

m
36. What is meant by FFT? (N-12)
The term Fast Fourier Transform (FFT) usually refers to a class of algorithms for efficiently computing the DFT.

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It makes use of the symmetry and periodicity properties of twiddle factor WNK to effectively reduce the DFT
computation time.
It is based on the fundamental principle of decomposing the computation of DFT of a sequence of length N into
successively smaller discrete Fourier transforms. The FFT algorithm provides speed increase factors, when
lts
compared with direct computation of the DFT, of approximately 64 and 205 for 256 points and 1024 – point
transforms respectively.

37. How many multiplications and additions are required to compute N-point DFT using radix-2 FFT? (A-15)
u
The number of multiplications and additions required to compute N-point DFT using radix-2 FFT are
N
N log 2 N and log 2 N respectively.
res

38. How many multiplications and additions are required to compute N-point DFT directly?
The number of multiplications and additions required to compute N-point DFT are N ( N  1) and N2
y2

respectively.
2b

39. What is the speed improvement factor in calculating 64-point DFT of a sequence using direct computation and FFT
algorithms?
(OR)

Calculate the number of multiplications needed in the calculation of DFT and FFT with 64-point sequence.
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 64
log 2 N  log 2 64  192
2 2
4096
Speed improvement factor=  21.33
192

40. What is meant by radix-2 FFT?

The FFT algorithm is most efficient in calculating N-point DFT. If the number of output points N can be
expressed as a power of 2, that is N  2 m , where m is an integer, then this algorithm is known as radix-2 FFT
algorithm.

41. What is a decimation-in-time algorithm?

The computation of 8-point DFT using radix-2 FFT, involves three stages of computations. Here N = 8 = 23,

m
therefore r= 2 and m = 3.
The given 8-point sequence is decimated to 2-point sequences. For each 2-point sequence, the 2-point DFT is
computed. From the result of 2-point DFT the 4-point DFT can be computed. From the result of 4-point DFT, the 8-

.co
point DFT can be computed.

42. What is decimation in frequency algorithm?

subsequences.
u lts
It is the popular form of the FFT algorithm. In this the output sequence X(k) is divided into smaller and smaller

43. What are the differences and similarities between DIT and DIF algorithms? (A-14)
The difference between DIT and DIF are:
res

1. In DIT, the input is bit-reversed while the output is in natural order. For DIF, the reverse is true, i.e., input is
normal order, while the output bit is reversed. However, both DIT and DIF can go from normal to shuffled
data or vice versa.
2. Considering the butterfly diagram in DIF, the complex multiplication takes place after the add-subtract
y2

operation.
The similarities between DIT and DIF are:
1. Both algorithms require the same number of operations to compute DFT.
2. Both algorithms require bit-reversal at some place during computation.
2b

44. What are the applications of FFT algorithms? (A-16)

The applications of FFT algorithms are:
1. Linear filtering
2. Correlation
3. Spectrum analysis
 m
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lts
PART – B (16 Marks)
u
z-transform and its properties
res

1. Determine the z-transform of the signal x(n) = [3(3)n – 4(2)n] u(n). (A-09)

2. Find the z-transform of

y2

(a) x(n) = cos nπ u(n) ; (b)x(n) = anu(n). (A-08), (A-10)

3. Find the z-transform and its ROC of the equation

2b

x ( n ) = ( −1/ 5 ) n u ( n) + 5 ( 1 / 2 ) – n u ( − n – 1 ) (N-11), (A-14)

4. Verify Parseval‟s theorem for the sequence x(n) = [1/3]n u(n).

5. Explain any six properties of z-transform with suitable examples.
6. Find the z-transform of
(a) x ( n) = 2n u( n − 2 ) ; (b) x (n) = n2 u(n)
Inverse z-transform
7. Obtain x(n) for the following :
 (M-11)

8. Using differential property, find the inverse z - transform of

X(z ) = log ( 1 − 0.5 z−1) ; z > 0.5

Difference equation, Solution by z-transform

9. Obtain the system function and impulse response of the following

y (n) – 5y(n –1 ) = x(n) + x(n–1) (M-11)

10. Explain the properties of LTI system with suitable equations. (N-08)

m
Convolution

11. Determine the circular convolution of

.co
x (n) = { 1, 3, 5, 7 } and h ( n ) = { 2, 4, 6, 8 } (N-09)

12. Determine the linear convolution of the following two sequences

x (n) = { 1, 1, -1, -1 } and h(n) = { 1, -1, 2, 1 } (M-12)

lts
UNIT – III : DESIGN OF IIR FILTERS
PART – A (2 Marks)
u
1. What are the different types of structures for realization of IIR systems?
The different types of structures for realization of IIR system are:
res

1. Direct form I structure

2. Direct form II structure
3. Cascade form structure
4. Parallel form structure
5. Lattice – ladder form structure
y2

2. Distinguish between recursive realization and non-recursive realization.

2b

For recursive realization the current output y(n) is a function of past outputs, past and present inputs. This form
corresponds to an Infinite – Impulse response (IIR) digital filter.
For non-recursive realizations current output sample y(n) is a function of only past and present inputs. This form
corresponds to an Finite Impulse response (FIR) digital filter.

3. How many numbers of additions, multiplications and memory locations are required to realize a system H(z)
having M zeros and N poles in (a) Direct form – I realization (b) Direct form – II realization.
 1. The Direct form – I realization requires M+N+1 multiplications, M+N additions and M+N+1 memory
locations.
2. The Direct form – II realization requires M+N+1 multiplications, M+N additions and the maximum of (M,N)
memory locations.

4. What is the main advantage of Direct form- II realizations when compared to Direct form –I realization?
In Direct form – II realization, the number of memory locations required is less than that of Direct form – I realization.

5. Define – Signal Flow Graph

A signal flow graph is defined as a graphical representation of the relationship between the variables of a set of
linear difference equations.

m
6. What is transposed theorem?
The transpose of a structure is defined by the following operations.
1. Reverse the directions of all branches in the signal flow graph

.co
2. Interchange the input and outputs
3. Reverse the roles of all nodes in the flow graph
4. Summing points become branching points
5. Branching points become summing points
lts
According to transposition theorem if we reverse the directions of all branch transmittance and interchange the
input and output in the flow graph, the system function remains unchanged.
u
7. What is canonic form structure?
res

The direct form –II realization requires minimum number of delays for the realization of the system. Hence it is
called as “Canonic form” structure.

8. What is the main disadvantage of direct form realization?

y2

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.
2b

9. What is the advantage of cascade realization?

Quantization errors can be minimized if we realize an LTI system in cascade form.

10. What are the different types of filters based on impulse response?
Based on impulse response, the filters are of two types. They are:
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 depends on the present input
sample and previous input samples.

11. What is the general form of IIR filter?

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

b z k
k

H ( z)  k 0
N
1   ak
k 1

m
12. Write the magnitude of Butterworth filter. What is the effect of varying order of N on magnitude and phase

.co
response?
The magnitude function of the Butterworth filter is given by
1
H ( j)  1
N  1, 2 , 3..............
    2N 2

1    
   c  
lts
where, N is the order of the filter and  c is the cut off frequency. The magnitude response of the Butterworth
u
filter closely approximates the ideal response as the order N increases. The phase response becomes more non-
linear as N increases.
res

13. List out the properties of Butterworth lowpass filters.

1. The magnitude response of the Butterworth filter decreases monotonically as the frequency  increases
from 0 to α.
y2

2. The magnitude response of the Butterworth filter closely approximates the ideal response as the order N
increases.
3. The Butterworth filters are all pole designs.
4. The poles of the Butterworth filter lies on a circle.
2b

1
5. At the cut off frequency  c , the magnitude of normalized Butterworth filter is .
2

14. What is Butterworth approximation?

In Butterworth approximation, the error function is selected in such a way that the magnitude is maximally flat in the
origin (i.e., at  =0) and monotonically decreasing with increasing  .

15. How are the poles of Butterworth transfer function located in s- plane?
 The poles of the normalized Butterworth transfer function symmetrically lies on an unit circle in s-plane with angular

spacing of .
N

16. What is Chebyshev approximation?

In Chebyshev approximation, the approximation function is selected in such a way that the error is minimized over a
prescribed band of frequencies.

17. What is Type –1 Chebyshev approximation?

In type –1 Chebyshev approximation, the error function is selected in such a way that, the magnitude response is
equi-ripple in the pass band and monotonic in the stop band.

18. What is Type-2 Chebyshev approximation?

m
In type-2 Chebyshev approximation, the error function is selected in such a way that, the magnitude response is
monotonic in pass band and equi-ripple in the stop band. The Type -2 magnitude response is called inverse

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Chebyshev response.

19. Write the magnitude function of Chebyshev low pass filter.

The magnitude response of Type -1 lowpassChebyshev filter is given by
H a   
1
lts
 
1   2 C N2  

 c
where,  is attenuation constant and
u
  
C N   is the Chebyshev polynomial of the first kind of degree N.
 c 
res

20. How does the order of the filter affect the frequency response of Chebyshev filter?
From the magnitude response of Type -1 Chebyshev filter it can be observed that the magnitude response
y2

approaches the ideal response as the order of the filter is increased.

2b

21. How is the order N of Chebyshev filter determined?

The order N of the Chebyshev filter is given by

cosh 1  
N  
 
cosh 1  s 
 
 p
0.1 p 0.1
where,   10  1 and   10 s  1
22. What are the properties of Chebyshev filter? (M-13, N-14)
1. The magnitude response of the Chebyshev filter exhibits in ripple either in pass band or in the stop band
according to the type.
2. The magnitude response approaches the ideal response as the value of N increases.
3. The Chebyshev type–1 filters are all pole designs.
4. The poles of Chebyshev filter lies on an ellipse.
1
5. The normalized magnitude function has a value of at the cutoff frequency  c .
1  2

23. Compare Butterworth filter with Chebyshev Type -1 filter. (M-16)

Sl. No. Butterworth filter Chebyshev filter
1 All pole design All pole design

m
2 The poles lie on a circle in s-plane The poles lie on a ellipse in s-plane
The magnitude response is
The magnitude response is equi-ripple in pass
maximally flat at the origin and

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3 band and monotonically decreasing in the stop
monotonically decreasing function of
band.
.
The normalized magnitude response The normalized magnitude response has a
1 1
4 has a value of at the cut off value of at the cut off frequency  c

frequency  c .
2
lts 1  2
.
Only few parameters has to be
A large number of parameter has to be
u
5. calculated to determine the transfer
calculated to determine the transfer function.
function.
res

24. What are the different types of filters based on the frequency response?
The filters can be classified based on frequency response. They are (i) low pass filter (ii)high pass filter (iii)Band
pass filter (iv)Band reject filter.
y2

25. Distinguish between FIR filter and IIR filter.

2b

Sl. No. FIR filter IIR filter

These filters can be easily designed
1. These filters do not have linear phase.
to have perfectly linear phase.
FIR filters can be realized
2. IIR filters are easily realized recursively.
recursively and non – recursively.
Greater flexibility to control the Less flexibility, usually limited to specific kind of
3.
shape of their magnitude response. filters.
Errors due to round-off noise are
4. The round-off noise in IIR filters is more.
less severe in FIR filters, mainly
 because feedback is not used.

26. 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. Windows method
2. Frequency sampling method
3. Optimal or minimax design.

27. What is meant understand by linear phase response? (N-11)

For a linear phase filter  ( )   , the linear phase filter did not alter the shape of the original signal. If the
phase response of the filter is non-linear, the output signal may be a distorted one. In many cases a linear phase
characteristic is required throughout the pass band of the filter to preserve the shape of a given signal within the
pass band. IIR filter cannot produce a linear phase. The FIR filter can give linear phase, when the impulse response

m
of the filter is symmetric about its mid-point.

28. For what kind of application, can the anti-symmetrical impulse response be used?

.co
The anti-symmetrical impulse response can be used to design Hilbert transformers and differentiators.

29. For what kind of application, can the symmetrical impulse response be used?
The impulse response, which is symmetric and having odd number of samples can be used to design all types of
lts
filters, i.e., low pass, high pass, band pass and band reject.
The symmetric impulse response having even number of samples can be used to design low pass and band pass
filter.
u
30. How are digital filters designed from the analog filters?
res

1. Map the desired digital filter specifications into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
3. Transform the transfer function of the analog prototype into an equivalent digital filter transfer function.
y2

31. Write any two procedures for digitizing the transfer function of an analog filter. (M-13)
The two important procedures for digitizing the transfer function of an analog filter are:
1. Impulse invariance method.
2b

2. Bilinear transformation method.

32. What are the requirements for a digital filter to be stable and causal?
1. The digital transfer function H(z) should be a rational function of z and the co-efficients of z should be
real.
2. The poles should lie inside the unit circle in z-plane.
3. The number of zeros should be less than or equal to number of poles.
33. What are the requirements for an analog filter to be stable and causal?
1. The digital transfer function Ha(s) should be a rational function of s and the co-efficient of s should be
real.
2. The poles should lie on the left half of s-plane.
3. The number of zeros should be less than or equal to the number of poles.

34. What are the advantages and disadvantages of digital filters?

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.

m
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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35. What is impulse invariant transformation?
The transformation of analog filter to digital filter without modifying the impulse response of the filter is called
lts
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.)
u
36. What is the main objective of impulse invariant transformation?
The objective of this method is to develop an IIR filter transfer function whose impulse is the sampled version of the
res

impulse response of the analog filter. Therefore the frequency response characteristics of the analog filter are
preserved.

37. Write the impulse invariant transformation used to transform real poles with and without multiplicity.
y2

The impulse invariant transformation used to transform real poles (at s = - pi) without multiplicity is
1 1
is transformed to   piT
s  pi 1 e z 1
2b

The impulse invariant transformation used to transform multiple real pole (at s = - pi) is
1 (1) m1 d m1 1
is transformed to 
s  pi m m 1  piT 1
(m  1) dpi 1  e z

38. What is the relation between digital and analog frequency in impulse invariant transformation?
The relation between analog and digital frequency in impulse invariant transformation is given by
Digital frequency,   T
 where,
 - Analog frequency and
T - Sampling time period

39. What is Bilinear transformation?

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
2 1  z 1
s
T 1  z1

40. What is the relation between digital and analog frequency in Bilinear transformation?

m
In Bilinear transformation, the digital frequency and analog frequency are related by the equation,
T
Digital frequency,   2 tan 1 or
2

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2 
Analog frequency   tan
T 2
where,
 - Analog frequency
T - Sampling time period
lts
41. What is frequency warping? (N-12)
u
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,
res

which is called frequency warping.

42. What is prewarping? Why is it employed? (M-11, M-14)

y2

In IIR filter design using bilinear transformation, the conversion of the specified digital frequencies to analog
frequencies is called prewarping.
Prewarping is necessary to eliminate the effect of warping on amplitude response.
2b

43. Explain the technique of prewarping.

In IIR filter design using bilinear transformation the specified digital frequencies are converted to analog equivalent
frequencies, which are called prewarp frequencies. Using the prewarp frequencies, the analog filter transfer function
is designed and then it is transformed to digital filter transfer function.

44. Compare the impulse invariant transformation with bilinear transformation. (N-11)
Sl. No. Impulse Invariant transformation Bilinear transformation
1. It is many-to-one mapping It is one-to-one mapping.
The relation between analog and digital The relation between analog and digital
2.
frequency is linear. frequency is nonlinear.
To prevent the problem of aliasing the There is no problem of aliasing and so the
3.
analog filters should be band limited. analog filter need not be band limited.
The magnitude and phase response of Due to the effect of warping, the phase
analog filter can be preserved by choosing response of analog filter cannot be preserved.
4. low sampling time or high sampling But the magnitude response can be preserved
frequency. by prewarping.

PART – B (16 Marks)

m
DFT and its properties

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1. Explain the following properties of DFT :
(a)Linearity
(b) Symmetry property
(c)Circular convolution
lts (M-11)
2. Find the DFT of the sequence x(n) = {1,1,0,0}. (N-07 / M-11)
3. Determine the 8 point DFT of the sequence (M-11)
u
x( n ) = { 1,1,1,1,1,1,0,0}
res

4. Calculate the IDFT of the sequence X(k) = { 4,0, 0, 0 }.

5. Derive the relation between z-transform and DTFT.
6. Perform the circular convolution of the following two sequences by means of DFT and IDFT.
x1(n) = { 1, 2, 3, 1} and x2(n) = { 4, 3, 2, 2 } (10)(N-07)
y2

Computation of DFT using FFT algorithm – DIT and DIF, Butterfly structure

7. Find the DFT of the sequence x ( n ) = { 1, 2, 3, 4, 4, 3, 2, 1 } using DIT algorithm.

2b

8. Compute 4 point DFT of the sequence x ( n ) = { 0, 1, 2, 3 } using DIT and DIF algorithms.

9. Given x ( n ) = { 0, 1, 2, 3, 4, 5, 6, 7 }, find X(k) using DIT FFT algorithm.

10. Using DIF, radix-2 FFT algorithm, find the 8 point DFT of the given sequence
x(n) = { 0, 1, 2, 3, 4, 5, 6, 7 }.
 11. Using 8 point DIT-FFT, draw the butterfly diagram for the given sequence

x(n) = { 1, 0, 0, 0, 0, 0, 0, 0 }.

FFT using radix-2

12. Derive and draw the radix-2 DIT algorithm for FFT of 8 points. (M-11)

UNIT – IV : FIR FILTER DESGIN

m
PART – A (2 Marks)

1. What is the condition for the impulse response of FIR filter to satisfy constant group and phase delay and only

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constant group delay?
For linear phase FIR filter to have both constant group delay and constant phase delay.
 ( )       
For satisfying the above condition
h(n)  h( N  1  n)
lts N 1
that is, the impulse response must be symmetrical about n 
2
If one constant group delay is desired, then
u
 ( )    
For satisfying the above condition
res

h(n)  h( N  1  n)
N 1
that is, the impulse response must be antisymmetrical about n 
2

2. What are the properties of an FIR filter?

y2

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.
2b

3. What are the steps involved in the FIR filter design?

1. Choose the desired (ideal) frequency response H d ( )
2. Take inverse Fourier transform of H d ( ) to get hd (n)
3. Convert the infinite duration hd (n) to a finite duration sequence h(n)
4. Take Z – transform of h(n) to get the transfer function H (z ) of the FIR filter
4. What is the necessary and sufficient condition for the linear phase characteristic of an FIR filter? (M-11, M-12)
The necessary and sufficient condition for the linear phase characteristic of a FIR filter is that the phase function
should be a linear function of  , which in turn requires constant phase delay or constant group delay.

5. How is the constant group delay and phase delay achieved in linear phase FIR filters?
Frequency response of FIR filters with constant group and phase delay
H ( )   H ( ) e j (  )
The following conditions have to be satisfied to achieve constant group and phase delay:
N 1
Phase delay,   ( i.e., phase delay is constant )
2


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Group delay,    ( i.e., group delay is constant)
2
Impulse response, h(n) = - h( N -1 – n ) (i.e., impulse response is anti symmetric )

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6. What are the possible types of impulse response for linear phase FIR filters?
There are four types of impulse response for linear phase FIR filters. They are:
1. Symmetric impulse response when N is odd.
lts
2. Symmetric impulse response when N is even.
3. Antisymmetric impulse response when N is odd.
4. Antisymmetric impulse response when N is even.
u
7. List out the well-known design techniques for linear phase FIR filter.
res

There are three well known methods of design techniques for linear phase FIR filters. They are,
1. Fourier series method and window method.
2. Frequency sampling method.
3. Optimal filter design method.
y2

8. Write the two concepts that lead to the Fourier series or Window method of designing FIR filters.
The following concepts lead to the design of FIR filters by Fourier series method.
2b

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

9. Write the procedure for designing FIR filter by Fourier series method.
1. Choose the desired (ideal) frequency response H d ( ) of the filter.
2. Evaluate the Fourier series co-efficient of H d ( ) which gives the desired impulse response hd (n) .
 
1
hd (n) 
2 
 H d ( )e jn d

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

4. Take Z – transform of h(n) to get a noncausalfilter transfer function H (z ) of the FIR filter.
 N 1 
 
 2 
5. Multiply H (z ) by z to convert noncausal transfer function to a realizable causal FIR filter transfer
function.
 N 1   N 1

 
 
h(0)  h(n) z n  z  n 
2
H (z )  z  2 
 
n 1

 

10. What are the disadvantages of Fourier series method?

m
In designing FIR filter using Fourier series method the infinite duration impulse response is truncated at n=
 N  1
 . Direct truncation of the series will lead to fixed percentage overshoots and undershoots before and
 2 

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after an approximated discontinuity in the frequency response.

11. What is Gibbs phenomenon? (M-12)

lts j
One possible way of finding an FIR filter that approximates H (e ) would be to truncate the infinite Fourier
 N  1
series at n=   .The abrupt truncation of the series will lead to oscillation both in pass band and in stop
 2 
u
band. This phenomenon is known as Gibbs phenomenon.
res

12. Write the procedure for designing FIR filter using windows.
1. Choose the desired (ideal) frequency response H d ( ) of the filter
2. Evaluate the Fourier series co-efficient of H d ( ) which gives the desired impulse response hd (n)

1
y2

hd (n) 
2 
 H d ( )e jn d

3. Choose a window sequence w(n) and multiply the infinite sequence hd (n) by w(n)to convert the infinite
2b

duration impulse response to finite duration impulse response h(n)

h(n)  hd (n)w(n)

4. Find the transfer function of the realizable FIR filter

 N 1   N 1

 
 
h(0)  h(n) z n  z  n 
2
H (z )  z  2 
 
n 1

 
13. What are the desirable characteristics of the window?
The desirable characteristics of the window are:
1. The central lobe of the frequency response of the window should contain most of the energy and should be
narrow.
2. The highest side lobe level of the frequency response should be small.
3. The side lobe of the frequency response should decrease in energy rapidly as  tends to .

14. What is window? Why is it necessary?

One possible way of finding an FIR filter that approximates H (e j ) would be to truncate the infinite Fourier
 N  1
series at n=   . The abrupt truncation of the series will lead to oscillation both in passband and in stopband.
 2 
These oscillations can be reduced through the use of less abrupt truncation of the Fourier series. This can be
achieved by multiplying the infinite impulse response with a finite weighing w(n) , called a window.

m
15. List out the characteristics of FIR filter designed using windows. (M-11)
1. The width of the transition band depends on the type of window.

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

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

17. What is meant by Optimum equiripple design criterion? Why is it followed?

In FIR filter design by Chebyshev approximation technique, the weighted approximation error between the
y2

desired frequency and the actual frequency response is spread evenly across the passband and stopband.The
resulting filter will have ripples in both the pass band and stop band. This concept of design is called optimum equi-
ripple design criterion.
2b

The optimum equiripple criterion is used to design FIR filter in order to satisfy the specifications of passband
and stopband.

18. Write the expression for frequency response of rectangular window.

The frequency response of rectangular window is given by
N
sin
WR ( )  2

sin
2
19. Write the characteristic features of Rectangular window.
4
1. The mainlobe width is equal to .
N
2. The maximum sidelobe magnitude is -13dB.
3. The sidelobe magnitude does not decreases significantly with increasing.

20. List out the features of FIR filter designed using rectangular window.
1. The width of the transition region is related to the width of the mainlobe of window spectrum.
2. Gibb‟s oscillations are noticed in the passband and stopband.
3. The attenuation in the stopband is constant and cannot be varied.

21. Write the equation specifying Hanning windows.

The equation for Hanning window is given by
2n ( N  1) ( N  1)
wH n (n)  0.5  0.5 cos for   n

m
N 1 2 2

=0 Otherwise.

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22. Write the equation specifying Hamming windows. (M-14)
The equation for Hamming window is given by
2n ( N  1) ( N  1)
wH (n)  0.54  0.46 cos for   n
N 1
lts 2 2

=0 Otherwise.
u
23. Write the equation specifying Blackman windows.
res

The equation for Blackman window is given by

2n 4n ( N  1) ( N  1)
wB (n)  0.42  0.5 cos  0.08 cos for   n
N 1 N 1 2 2

=0 Otherwise.
y2

24. Write the equation specifying Bartlett windows.

The equation for Bartlettwindow is given by
2b

2n ( N  1) ( N  1)
wT (n)  1  for   n
N 1 2 2

=0 Otherwise.

25. Write the equation specifying Kaiser windows.

The equation for Bartlettwindow is given by
   2n  
 1   
  N  1  ( N  1)
wk (n)  I 0 for n 
 I 0 ( )  2
 
 

=0 Otherwise.
where,  is an independent parameter.
I 0 (x) is the zeroth order Bessel function of the first kind
2
  1  x k 
I 0 ( x)  1      
 k!  2  
k 1 

m
26. Write the characteristics features of Triangular window.
The characteristics features of Triangular window are:
8

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1. The mainlobe width is equal to .
N
2. The maximum sidelobe magnitude is -25dB.
3. The sidelobe magnitude slightly decreases with increasing  .
lts
27. Why is the triangular window not good a good choice for designing FIR filters?
In FIR filters designed using triangular window the transition from passband to stopband is not sharp and the
attenuation in stopband is less when compared to filters designed with rectangular window. For the above two
u
reasons the triangular window is not a good choice.
res

28. List out the features of hanning window spectrum.

8
1. The mainlobe width is equal to .
N
y2

2. The maximum sidelobe magnitude is -31dB.

3. The sidelobe magnitude slightly decreases with increasing  .
2b

29. List out the features of hamming window spectrum.

8
1. The mainlobe width is equal to .
N
2. The maximum sidelobe magnitude is -41dB.
3. The sidelobe magnitude remains constant for increasing  .

30. Compare Rectangular window with Hanning window.

 Sl. No. Rectangular window Hanning window
The width of mainlobe in window The width of mainlobe in window spectrum is
1 4 8
spectrum is
N N
The maximum sidelobe magnitude in The maximum sidelobe magnitude in window
2
window spectrum is -13dB. spectrum is -31dB.
In window spectrum the sidelobe
In window spectrum the sidelobe magnitude
3 magnitude slightly decreases with
decreases with increasing 
increasing 
In FIR filter designed using rectangular
In FIR filter designed using Hanning window the
4 window the minimum stop band
minimum stop band attenuation is 44dB.
attenuation is 22dB.

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31. Compare Rectangular window with Hamming window.

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Sl. No. Rectangular window Hamming window
The width of mainlobe in window The width of mainlobe in window spectrum is
1 4 8
spectrum is
N N

2
window spectrum is -13dB.
lts
The maximum sidelobe magnitude in The maximum sidelobe magnitude in window
spectrum is -41dB.
In window spectrum the sidelobe
In window spectrum the sidelobe magnitude
3 magnitude slightly decreases with
u
remains constant.
increasing 
In FIR filter designed using rectangular
res

In FIR filter designed using Hamming window

4 window the minimum stopband
the minimum stopband attenuation is 51dB.
attenuation is 22dB.

32. Compare Hanning window with Hamming window.

y2

Sl. No. Hanning window Hamming window

The width of mainlobe in window
8
2b

1 8 The width of mainlobe in window spectrum is

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

Sl. No. Hamming window Blackman window

The width of mainlobe in window The width of mainlobe in window spectrum is
1 8 12
spectrum is
N N
The maximum sidelobe magnitude in The maximum sidelobe magnitude in window
2
window spectrum is -41dB. spectrum is -58dB.
In window spectrum the sidelobe
In window spectrum the sidelobe magnitude
3 magnitude remains constant with
decreases rapidly with increasing 
increasing 

m
In FIR filter designed using Hamming
In FIR filter designed using Blackman window
4 window the minimum stop band
the minimum stopband attenuation is 78dB.

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attenuation is 51dB.
The higher value of sidelobe attenuation The higher value of sidelobe attenuation is
5 is achieved at the expense of constant achieved at the expense of increased mainlobe
attenuation at high frequencies. width.
lts
34. List out the features of Blackman window spectrum.
The features of Blackman window spectrum are:
12
u
1. The mainlobe width is equal to .
N
2. The maximum side lobe magnitude is -58dB.
res

3. The sidelobe magnitude slightly decreases with increasing  .

4. The higher value of sidelobe attenuation is achieved at the expense of increased mainlobe width.
y2

35. List out the features of Kaiser window spectrum.

1. The width of mainlobe and the peak sidelobe are variable.
2. The parameter  in the Kaiser window function, is an independent variable that can be varied to control the
sidelobe levels with respect to mainlobe peak.
2b

3. The width of the mainlobe in the window spectrum (and so the transition region in the filter) can be varied
by varying the length N of the window sequence.

36. Compare Hamming window with Kaiser window.

Sl. No. Hamming window Kaiser window

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

m
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u lts
res
y2
2b

PART – B (16 Marks)

 FIR and IIR filter realization – Parallel and cascade forms
1. Obtain a direct form I realization for the system described by the difference equation

y (n) = 0.5 y(n−1) − 0.25 y(n−2) + x(n) + 0.4 x(n−1). (M-11)

2. Obtain the direct form II and parallel form realization for the following system.

y (n) = − 0.1 y(n – 1 ) + 0.2 y(n – 2) + 3 x(n) + 3.6 x(n – 1 ) + 0.6 x( n – 2). (N-12)

3. Draw the cascade realization structure of the following impulse response

H(z) = (1 − z−1 ) (1 + 2 z−1 ) (1 + 3z−3 )

(1 + z−1 + 2z−2 )(1 − z−1 + 3z−2 ) (1 + 2z−1 − 4z−2 )

m
FIR design : Windowing Techniques

4. The desired response of a low pass filter is

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Hd(e jω) = e-j3ω , − 3π/4 < ω < 3π/4
0 , 3π/4 < ω ≤ π lts (M-09)

Determine the frequency response of the filter for M=7 using Hamming window.

5. Using Blackman window for N=11, design an Ideal Hilbert transform having frequency response of
u
Hd(e jω) = j, π ≤ω<0
res

0, 0≤ω≤ π

6. Using a rectangular window technique, design a lowpass filter with passband gain of unity, cutoff frequency of
y2

1000 Hz and working at a sampling frequency of 5 Hz. The length of the impulse response is 7.
2b

7. Determine the frequency response of FIR filter defined by

y (n) = 0.25 x(n) + x(n – 1) + 0.25 x(n – 2 ).

Calculate the phase delay and group delay.
8. List the design techniques of FIR filters and explain any one technique in detail.

IIR design : Analog filter design – Butterworth and Chebyshev approximations

 9. For the following specifications, design a Butterworth filter using the impulse variance method.

(N-07)

10. Design a Butterworth filter with 2 dB passband attenuation at a frequency of 20 rad / sec and 10 dB
stopband attenuation at 30 rad/sec.

11. Design a Chebyshev filter using bilinear transformation for the following specifications

m
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Digital design using impulse invariant method

12. Explain impulse invariant method of designing IIR Filter. (N-12)

u lts
res
y2
2b

UNIT – V : PROGRAMMABLE DSP CHIPS AND QUANTIZATION EFFECTS

PART – A (2 Marks)

1. What are the classification digital signal processors?

The digital signal processors are classified as
1. General purpose digital signal processors.
 2. Special purpose digital signal processors.

2. Write some examples for fixed point DSPs.

Some examples for fixed point DSPs are:
1. TMS320C50
2. TM 320C54
3. TM 320C55
4. ADSP-219x
5. ADSP-219xx

3. Write some examples for floating point DSPs.

Some examples for floating point DSPs are:

m
1. TMS320C3x
2. TMS320C67x

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3. ADSP-21xxx

4. What are the factors that influence selection of DSPs? (M-11, N-11, M-15, M-16)
1. Architectural features
2. Execution speed
3. Type of arithmetic
lts
4. Word length
u
5. What are the applications of PDSPs? (M-15)
The applications of PDSPs are:
res

1. Digital cell phones

2. Automated inspection
3. Voicemail
4. Motor control
5. Video conferencing
y2

6. Noise cancellation
7. Medical imaging
8. Speech synthesis
2b

9. Satellite communication, etc.

6. What are the advantages and disadvantages of VLIW architecture? (M-16)

The advantages of VLIW architecture are:
1. Increased performance
2. Better compiler targets
3. Potentially scalable
4. Potentially easier to program
5. Can add more execution units, allow more instruction to be packed into the VLIW instruction.
 The disadvantages of VLIW architecture are:
1. New kind of programmer/compiler complexity
2. Program must keep track of instruction scheduling
3. Increased memory use
4. High power consumption
5. Misleading MIPS ratings

7. What is meant by pipelining? (N-11, N-12)

Pipelining a processor means breaking down its instruction into a series of discrete pipeline stages which can be
completed in sequence by specialized hardware.

m
8. What is pipeline depth?
The number of pipeline stages is referred to as the pipeline depth.

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9. What is the pipeline depth of TMS320C50, TM 320C54x?
TMS320C50 – 4
TM 320C54x – 6
lts
10. What are the different stages in pipelining? (N-14)
u
The different stages in pipelining are:
1. the Fetch phase
res

2. the Decode phase

3. Memory read phase
4. the Execute phase
y2

11. Write the different buses of TM 320C5x and their functions. (M-14)
The „C5x architecture has four buses. They are:
1. Program bus (PB)
2b

2. Program address bus (PAB)

3. Data read bus (DB)
4. Data read address bus (DAB)
The program bus carriers the instruction code and immediate operands from program memory to the CPU.
The program address bus provides address to program memory space for both read and write.
The data read bus interconnects various elements of the CPU to data memory spaces.
The data read address bus provides the address to access the data memory spaces.
12. List out the various registers used with ARAU. (M-14, N-14)
1. Eight auxiliary registers (AR0-AR7)
2. Auxiliary register pointer (ARP)

13. What are the elements that the control processing unit of „C5X consist of?
1. Central arithmetic logic unit (CALU)
2. Parallel logic unit (PLU)
3. Auxiliary register arithmetic unit (ARAU)
4. Memory mapped registers
5. Program controller

14. What is the function of parallel logic unit? (N-12)

The function of the parallel logic unit is to execute logic operations on data without affecting the contents of

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accumulator.

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15. List out the on chip peripherals in „C5x.
The on-chip peripherals interfaces connected to the „C5x CPU include
1. Clock generator
2. Hardware timer
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3. Software programmable wait state generators
4. General purpose I/O pins
5. Parallel I/O ports
6. Serial port interface
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7. Buffered serial port
8. Time-divisions multiplexed (TDM) serial port
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9. Host port interface

10. User unmaskable interrupts

16. What are the arithmetic instructions of „C5x?

The arithmetic instructions of „C5x are:
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1. ADD
2. ADDB
3. ADDC
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4. SUB
5. SUBB
6. MPY
7. MPYU

17. What are the logical instructions of „C5x?

The logical instructions of „C5x are:

1. AND
 2. ANDB
3. OR
4. ORB
5. XOR
6. XORB

18. What are the shift instructions?

The shift instructions are:
1. ROR
2. ROL
3. ROLB
4. RORB
5. BSAR

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19. What are load/store instructions?
The load/store instructions are:

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1. LACB
2. LACC
3. LACL
4. LAMM
5. LAR
6. SACB
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7. SACH
8. SACL
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9. SAR
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20. What are the different types of arithmetic in digital systems?

There are three types of arithmetic used in digital systems. They are:
1. Fixed point arithmetic
2. Floating point arithmetic
3. Block Floating arithmetic
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21. What is meant by a fixed-point number?

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In fixed-point arithmetic the positions of the binary point is fixed. The bits to the right represent the fractional part of
the number and those to the left represent the integer part. For example, the binary number 01.1100 has the value
1.75 in decimal.

22. What is meant by block floating point representation? What are its advantages?
In block floating point arithmetic the set of signals to be handled is divided into blocks. Each block has the same
value for the exponent. The arithmetic operations within the block uses fixed point arithmetic and only one exponent
 per block is stored, thus saving memory. This representation of numbers is most suitable in certain FFT flow graphs
and in digital audio applications.

23. What are the advantages of floating point arithmetic?

The advantages of floating point arithmetic are:
1. Larger dynamic range
2. Overflow in floating point representation is unlikely

24. Compare fixed point arithmetic with floating point arithmetic. (M-15)

Fixed Point Arithmetic Floating Point Arithmetic

Fast Operation Slow Operation
Relatively economical More expensive because of costlier hardware

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Small dynamic range Increased dynamic range
Round off errors can occur with both additions
Round off error occur only for additions

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and multiplication
Overflow occur in addition Overflow does not arise
Used in small computers Used in larger, general purpose computers
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25. What are the three quantization errors due to finite word length registers in digital filters?
The three quantization errors due to finite word length registers in digital filters are:
1. Input quantization error
2. Coefficient quantization error
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3. Product quantization error
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26. How are the multiplication and addition carried out in floating point arithmetic?

In floating point arithmetic, multiplications are carried out as follows:

Let f1=M1 x 2c1 and f2 = M2 x 2c2, then f3=f1x f2 = (M1xM2)2(c1+c2)
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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 numbers are equal and then by adding the mantissas.
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27. Write short notes on coefficient inaccuracy.

(OR)
What is coefficient quantization error? What is it‟s effect?
The filter coefficients are computed to infinite precision in theory. But, in digital computation the filter coefficients are
represented in binary and are stored in registers.
If a b bit register is used, the filter coefficients must be rounded or truncated to b bits, which produces an error.
Due to quantization of coefficients, the frequency response of the filter may differ appreciably from the desired
response and some times the filter may actually fail to meet the desired response and the desired specifications. If
 the poles of desired filter are close to the unit circle, then those of the filter with quantized coefficients may lie just
outside the unit circle, leading to unstability.

28. What is product quantization error?

Product quantization errors arise at the output of a multiplier. Multiplication of a b bit data with a b bit coefficient
results in a product having 2b bits. Since a b bit register is used, the multiplier output must be rounded or truncated
to b bits, which produces an error. This error is known as product quantization error.

29. What is meant by input quantization error?

In DSP, the continuous time input signals are converted into digital using a b bit ADC. The representation of
continuous signal amplitude by a fixed digit produces an error is known as input quantization error.

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30. What is meant by truncation?
Truncation is process of reducing the size of binary number by discarding all bits less significant than the least

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significant bit that is retained. (In the truncation of a binary number to b bits all the less significant bits beyond b th bit
are discarded)

31. What is meant by rounding?

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Round is the process 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 un-quantized number.

32. What is Quantization step size?

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In digital systems, the numbers are represented in binary. With b-bit binary we can generate 2b different binary
codes. Any range of analog value to be represented in binary should be divided into 2 b levels with equal increment.
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The 2b levels are called Quantization levels and the increment in each level is called Quantization step size. If
R is the range of analog signal then,
Quantization step size, q = R/2b.
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33. What is called limit cycle?

In recursive systems when the input is zero or some nonzero constant value, the nonlinearities due to finite
precision arithmetic operations may cause periodic oscillations in the output. These oscillations are called limit
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cycle.

34. What is zero input limit cycle?

In recursive system, the product Quantization may create periodic oscillations in the output. These oscillations are
called limit cycles. If the system output enters a limit cycle, it will continue to remain in limit cycle even when the
input is made zero. Hence these limit cycles are also called zero input limit cycles.

35. What is meant by dead band?

 In a limit cycle the amplitudes of the output are confined to a range of values, which is called dead band of the filter.

36. How can the system output 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.

37. What is meant by 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.\

38. How can overflow limit cycle be eliminated?

The overflow limit cycles can be eliminated either by using saturation arithmetic or by scaling the input signal to the

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adder.

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39. What is saturation arithmetic?
In saturation arithmetic when the result of an arithmetic operations exceeds the dynamic range of number system,
then the result is set to maximum or minimum possible value. If the upper limit is exceeded then the result is set to
maximum possible value. If the lower limit is exceeded then the result is set to minimum possible value.
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PART – B (16 Marks)

Introduction – Architecture
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1. Explain the addressing formats in the DSP processors. (M-11)

2. Explain pipelining in DSP. (M-12)

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3. Draw the block diagram of Harvard architecture and explain the various blocks.

4. Explain the logical instructions of TMS320C5X processor.

5. Explain the functions of multiply and accumulate unit.
6. Write a note on arithmetic instructions of TMS320C5X processor.

Addressing Formats
7. Write short notes on : (N-12)

(a) Memory mapped register addressing

(b) Circular addressing mode

(c) Auxillary registers

Functional modes

8. Explain the functional modes present in the DSP processor. (M-12)

Introduction to commercial processors

9. What are the different buses of TMS320C54X processor and mention their functions. (M-08)

10. What are the different addressing modes of TMS320C5Xprocessor? (M-11)

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11. Explain the following techniques in relation to DSP processor

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SIMD and VLIW
In each case, clearly point out the advantages and the disadvantages of the technique in signal
processing. lts (N-12)

12. Explain the architecture of TMS320C54X with a neat diagram. (N-07/M-11/ N-12)
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