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# EE T65 DIGITAL SIGNAL PROCESSING

UNIT I
1. Define DSP.
DSP - Digital Signal Processing.
It is defined as changing or analyzing information which is measured as discrete time sequences.

## 2. List out the basic elements of DSP.

* signal in
* Analog to Digital converter
* Digital Signal processor
* Digital to Analog converter
* signal out

* Veracity
* Simplicity
* Repeatability

## 4. Give the applications of DSP.

* Telecommunication – spread spectrum, data communication
* Biomedical – ECG analysis, Scanners.
* Speech/audio – speech recognition
* Military – SONAR, RADAR

5. Define Signal.
Signal is a physical quantity that varies with respect to time , space or any other independent
variable.
(Or)
It is a mathematical representation of the system
Eg: y(t) = t. and x(t)= sin t.

6. Define system.
A set of components that are connected together to perform the particular task. E.g.; Filters

## 7. What are the major classifications of the signal?

(i) Discrete time signal
(ii) Continuous time signal

## 8. Define discrete time signals and classify them.

Discrete time signals are defined only at discrete times, and for these signals, the independent
variable takes on only a discrete set of values.
Classification of discrete time signal:
1. Periodic and A periodic signal
2. Even and Odd signal

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EE T65 DIGITAL SIGNAL PROCESSING

## 9. Define continuous time signals and classify them.

Continuous time signals are defined for a continuous of values of the Independent variable. In
the case of continuous time signals the independent Variable is continuous.
For example:
(i) A speech signal as a function of time
(ii) Atmospheric pressure as a function of altitude
Classification of continuous time signal:
(i) Periodic and A periodic signal
(ii) Even and Odd signal

## 10. Define discrete time unit step &unit impulse.

Discrete time Unit impulse is
defined as δ[n]={1, n=0; }
{ 0, otherwise.}
Unit impulse is also known as unit sample. Discrete time unit step signal is defined by
U[n]={0,n=0}
{1,n>= 0}

## 11. Define even and odd signal.

A discrete time signal is said to be even when, x[-n]=x[n].
The continuous time signal is said to be even when, x(-t)= x(t)
For example, Cosine wave is an even signal. The discrete time signal is said to be odd when x[-
n]= -x[n]
The continuous time signal is said to be odd when x(-t)= -x(t)
Odd signals are also known as nonsymmetrical signal. Sine wave signal is an odd signal.

## 12. Define Energy and power signal.

A signal is said to be energy signal if it have finite energy and zero power. A signal is said to be
power signal if it have infinite energy and finite power. If the above two conditions are not
satisfied then the signal is said to be Neither energy nor power signal

## 13. 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.

## 14. What is digital signal?

The digital signal is same as discrete signal except that the magnitude of signal is Quantized

## 15. What are the different types of signal representations?

a. Graphical representation
b. Functional representations
c. Tabular representation
d. Sequence representation.

## DEPT. OF EEE 2 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 16. 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 non periodic signals.

## 17. State the classification of discrete time signals.

The types of discrete time signals are
* Energy and power signals
* Periodic and A periodic signals
* Symmetric (Even) and Ant symmetric (Odd) signals

## 18. Define energy and power signal.

If E is finite i.e. 0<E<α , then x (n) is called energy signal.
If P is finite i.e. 0<P<α , then the signal x(n) is called a power signal.

## 19. State sampling theorem.

The sampling frequency must be at least twice the maximum frequency present in the signal.
That is Fs = > 2fm Where, Fs = sampling frequency, Fm = maximum frequency

## 20. Define nyquist rate.

It is the minimumrate at which a signal can be sampled and still reconstructed fromits samples.
Nyquist rate is always equal to 2fm.

## 21. Define aliasing or folding.

The superimposition of high frequency behavior on to the low frequency behavior is referred as
aliasing. This effect is also referred as folding.

## 22. What is the condition for avoid the aliasing effect?

To avoid the aliasing effect the sampling frequency must be twice the maximum frequency
present in the signal.

## 23. What is meant by interpolation?

It is also referred as up sampling. that is , increasing the sampling rate.

## 24. What is an anti-aliasing filter?

A filter that is used to reject high frequency signals before it is sampled to remove the aliasing of
unwanted high frequency signals is called an ant aliasing filter.

## 25. Mention the types of sample/hold?

* zero order hold
* first order hold

## DEPT. OF EEE 3 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 26. What is meant by sampling rate?

Sampling rate = number of samples / second.

## 27. What is meant by step response of the DT system?

The output of the system y(n) is obtained for the unit step input u(n) then it is said to be step
response of the system.

## 28. Define Transfer function of the DT system.

The Transfer function of DT system is defined as the ratio of Z transform of the system output to
the input. That is , H(z)=Y(z)/X(z).

## 29. Define impulse response of a DT system.

The impulse response is the output produced by DT system when unit impulse is applied at the
input. The impulse response is denoted by h(n). The impulse response h(n) is obtained by taking
inverse Z transform from the transfer function H(z).

## 30. Define a causal system.

The causal system generates the output depending upon present &past inputs only. A causal
system is non anticipatory.

## 31. What is meant by linear system?

A linear system should satisfy superposition principle. A linear system should satisfy
F[ax1(n)+bx2(n)] = a y 1(n)+by2(n)
Where, y1(n)=F[x1(n)] & y2(n)=F[x2(n)]

## 32. Define time invariant system.

1 .A system is time invariant if the behavior and characteristics of the system are fixed over time.
2. A system is time invariant if a time shift in the input signal results in an identical time shift in
the output signal.
3. For example, a time invariant system should produce y (t-t0) as the output

## 33. What is a continuous and discrete time signal?

Continuous time signal: A signal x(t) is said to be continuous if it is defined for all time t.
Continuous time signal arise naturally when a physical waveform such as acoustics wave or
light wave is converted into a electrical signal. This is effected by means of transducer.(e.g.)
microphone, photocell.
Discrete time signal: A discrete time signal is defined only at discrete instants of time. The
independent variable has discrete values only, which are uniformly spaced. A discrete time
signal is often derived from the continuous time signal by sampling it at a uniform rate.

## 34. Give the classification of signals?

Continuous-time and discrete time signals, Even and odd signals, Periodic signals and non-
periodic signals Deterministic signal and Random signal Energy and Power signal

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EE T65 DIGITAL SIGNAL PROCESSING

## 35. What are the types of systems?

Continuous time and discrete time systems
Linear and Non-linear systems
Causal and Non-causal systems
Static and Dynamic systems
Time varying and time in-varying systems
Distributive parameters and Lumped parameters systems
Stable and Un-stable systems.

## 36. What are even and odd signals?

Even signal: continuous time signal x(t) is said to be even if it satisfies the condition
x(t)=x(-t) for all values of t.
Odd signal: he signal x(t) is said to be odd if it satisfies the condition x(-t)=-x(t) for all t. In
other words even signal is symmetric about the time origin or the vertical axis, but odd signals
are anti-symmetric about the vertical axis.

## 37. What are deterministic and random signals?

Deterministic Signal: deterministic signal is a signal about which there is no certainty with
respect to its value at any time. Accordingly we find that deterministic signals may be modeled
as completely specified functions of time.
Random signal: random signal is a signal about which there is uncertainty before its actual
occurrence. Such signal may be viewed as group of signals with each signal in the ensemble
having different wave forms

## 38. What are energy and power signal?

Energy signal: signal is referred as an energy signal, if and only if the total energy of the
signal satisfies the condition 0<E<∞. The total energy of the continuous time signal x (t) is
given as
E=limT→∞∫x2 (t)dt, integration limit from –T/2 to +T/2
Power signal: signal is said to be powered signal if it satisfies the condition 0<P<∞. The
average power of a continuous time signal is given by
P=limT→∞1/T∫x2(t)dt, integration limit is from-T/2 to +T/2.

## 39. What are the operations performed on a signal?

Operations performed on dependent variables:
Amplitude scaling: y (t) =cx (t), where c is the scaling factor, x(t) is the continuous time
signal.
Multiplication y (t)=x1(t)x2(t)
Differentiation: y (t)=d/dt x(t)
Integration (t) =∫x(t)dt
Operations performed on independent variables
Time shifting Amplitude
scaling Time reversal

## DEPT. OF EEE 5 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 40. What are elementary signals and name them?

The elementary signals serve as a building block for the construction of more complex
signals. They are also important in their own right, in that they may be used to model many
physical signals that occur in nature.There are five elementary signals. They are as follows
 Unit step function
 Unit impulse function
 Ramp function
 Exponential function
 Sinusoidal function

## 41. What are the properties of a system?

Stability: A system is said to be stable if the input x(t) satisfies the condition(t)│≤Mx<∞ and the
output satisfies the condition │y(t)│≤My<∞ for all t.
Memory: A system is said to be memory if the output signal depends on the present and the past
inputs.
Invertibility: A system is said to be invertible if the input of the system can be recovered from
the system output.
Time invariance: A system is said to be time invariant if a time delay or advance of the input
signal leads to an identical time shift in the output signal.
Linearity: A system is said to be linear if it satisfies the super position principle
i.e.) R (ax1(t)+bx2(t))=ax1(t)+bx2(t)

## 42. What is memory system and memory less system?

A system is said to be memory system if its output signal at any time depends on the past
values of the input signal. Circuit with inductors capacitors are examples of memory system.
A system is said to be memory less system if the output at any time depends on the present
values of the input signal. An electronic circuit with resistors is an example for memory less
system.

## 43. What is an invertible system?

A system is said to be invertible system if the input of the system can be recovered from the
system output. The set of operations needed to recover the input as the second system
connected in cascade with the given system such that the output signal of the second system is
equal to the input signal applied to the system.
H-1{y(t)}=H-1{H{x(t)}}.
44. What are time invariant systems?
A system is said to be time invariant system if a time delay or advance of the input signal leads
to an identical shift in the output signal. This implies that a time invariant system responds
identically no matter when the input signal is applied. It also satisfies the condition
R{x(n-k)}=y(n-k).

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EE T65 DIGITAL SIGNAL PROCESSING

45. Is a discrete time signal described by the input output relation y[n]= rnx[n] time
invariant.

## A signal is said to be time invariant if R{x[n-k]}= y[n-k]

R{x[n-k]}=R(x[n]) / x[n]→x[n-k]

## =rnx [n-k] ---------------- (1)

y[n-k]=y[n] / n→n-k
=rn-kx [n-k] -------------------(2)
Equations (1)≠Equation(2)
Hence the signal is time variant.
46. Show that the discrete time system described by the input-output relationship y[n]

=nx[n] is linear?

## For a sys to be linear R{a1x1[n]+b1x2[n]}=a1 y1[n]+b1 y2[n]

L.H.S:R{ a1x1[n]+b1x2[n] }=R{x[n]} /x[n] → a1x1[n]+b1x2[n]
= a1 nx1[n]+b1 nx2[n] -------------------(1)
R.H.S: a1 y1[n]+b1 y2[n]= a1 nx1[n]+b1 nx2[n] --------------------(2)
Equation(1)=Equation(2)
Hence the system is linear
47. What are the steps involved in digital signal processing?

 Converting the analog signal to digital signal, this is performed by A/D converter
 Processing Digital signal by digital system.
 Converting the digital signal to analog signal, this is performed by D/A converter.

## 48. Give some applications of DSP?

 Speech processing – Speech compression & decompression for voice storage system
 Communication – Elimination of noise by filtering and echo cancellation.
 Bio-Medical – Spectrum analysis of ECG, EEG etc.

## 49. Define sampling process.

Sampling is a process of converting Ct signal into Dt signal.

## 50. Mention the types of sampling.

* Up sampling
* Down sampling
PART-B
1.For each of the following systems, determine whether the system is static stable, causal, linear
and time invariant

## DEPT. OF EEE 7 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

a. y(n) = e x(n)
b.y(n) = ax(n) +b
c. y(n) = Σnk=n0 x(k)
d.y(n) = Σn+1k= -∞ x(k)
e. y(n) = n x2(n)
f. y(n) = x(-n+2)
g.y(n) = nx(n)
h.y(n) = x(n) +C
i. y(n) = x(n) – x(n-1)
j. y(n) = x(-n)
k.y(n) = Δ x(n) where Δ x(n) = [x(n+1) – x(n)]
l. y(n) = g(n) x(n)
m. y(n) = x(n2)
n.y(n) = x2(n)
o.y(n) = cos x(n)
p.y(n) = x(n) cos ω0n
2.Compute the linear convolution of h(n) = {1,2,1} and x(n) ={1,-3,0,2,2}
3.Explain the concept of Energy and Power signals and determine whether the following are
energy or power signals
a. x(n) = (1/3)n u(n)
b.x(n) = sin (π / 4)n
4.The unit sample response h(n) of a system is represented by
h (n) = n2u(n+1) – 3 u(n) +2n u(n-1) for -5≤ n ≤5. Plot the unit sample response.
5.State and prove sampling theorem. How do you recover continuous signals from its samples?
Discuss the various parameters involved in sampling and reconstruction.
6.What is the input x(n) that will generate an output sequence
y(n) = {1,5,10,11,8,4,1} for a system with impulse response h(n) = {1,2,1}
7.Check whether the system defined by h(n) = [5 (1/2)n +4(1/3)n] u(n) is stable?
8.Explain the analog to digital conversion process and reconstruction of analog signal from
digital signal.
9.What are the advantages and disadvantages of digital signal processing compared with analog
signal processing?
10. Classify and explain different types of signals.
11. Explain the various elementary discrete time signals.
12. Explain the different types of mathematical operations that can be performed on a discrete
time signal.
13. Explain the different types of representation of discrete time signals.
14. Determine whether the systems having the following impulse responses are causal and stable
a. h(n) = 2n u(-n)
b.h(n) = sin nπ / 2
c. h(n) = sin nπ + δ (n)
d.h(n) = e2n u(n-1)
15. The impulse response of a linear time invariant system is h (n) = {1, 2, 1,-1}. Determine the
response of the system to the input signal x (n) = {1, 2, 3, 1}.

## DEPT. OF EEE 8 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

UNIT II
1. Define z transform?
The Z transform of a discrete time signal x(n) is defined as,

## 2. 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.

## 3. 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.

4. 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=α .

## 5. 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.

## 6. 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), where a&b are constants.

## 7. State the time shifting property of the z transform

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

## 8. State the scaling property of the z transform

If z{x(n)}=X(z) then z{an x(n)}=X(a-1 z)

## 9. State the time reversal property of the z transform

If z{x(n)}=X(z) then z{x(-n)}=X(z-1)

## 10. 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)

## 11. 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.

## DEPT. OF EEE 9 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 12. 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.

## 14. What is the need for Z-transform?

Z-transform is used for analysis the both periodic and a periodic signals.

Z[ δ(n) ] = 1

## 16. Define zeros.

The zeros of the system H(z) are the values of z for which H(z) = 0.

## 17. Define poles.

The poles of the system H(z) are the values of z for which H(z) = α.

## 18. What is the z-transform of A δ (n-m) ?

Z [ A δ (n-m) ] =1.

## 19. State the convolution properties of Z transform?

The convolution property states that the convolution of two sequences in time domain is
equivalent to multiplication of their Z transforms.

20. What are the different methods to calculate the inverse Z transform?
i. Long division method
ii. Partial fraction expansion method
iii. Residue method
iv. Convolution method

## 21. What is Discrete Time Systems?

The function of discrete time systems is to process a given input sequence to generate output
sequence. In practical discrete time systems, all signals are digital signals, and operations on
such signals also lead to digital signals. Such discrete time systems are called digital filter.

## 22. Write the Various classifications of Discrete-Time systems.

 Linear & Non linear system
 Causal & Non Causal system
 Stable & Un stable system
 Static & Dynamic systems

## 23. Define Linear system

A system is said to be linear system if it satisfies Super position principle. Let us consider
x1(n) & x2(n) be the two input sequences & y1(n) & y2(n) are the responses respectively,
T[ax1(n) + bx2(n)] = a y1(n) + by2(n)

## DEPT. OF EEE 10 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 24. Define Static & Dynamic systems

When the output of the system depends only upon the present input sample, then it is called
static system, otherwise if the system depends past values of input then it is called dynamic
system

## 24. Define causal system.

When the output of the system depends only upon the present and past input sample, then it
is called causal system, otherwise if the system depends on future values of input then it is called
non-causal system

## 25. Define Shift-Invariant system.

If y(n) is the response to an input x(n), then the response to an input
X(n) = x(n-n0) then y(n) = y(n-n0)
When the system satisfies above condition then it is said to shift in variant, otherwise it is
variant.

## 26. Define impulse and unit step signal.

Impulse signal (n):
The impulse signal is defined as a signal having unit magnitude at n = 0 and
zero for other values of n.
(n) = 1; n = 0
0; n  0
Unit step signal u(n):
The unit step signal is defined as a signal having unit magnitude for all
values of n  0
u(n) = 1; n  0
0; n  0

## 27. What are FIR and IIR systems?

The impulse response of a system consist of infinite number of samples are called IIR system &
the impulse response of a system consist of finite number of samples are called FIR system.

28. What are the basic elements used to construct the block diagram of discrete time
system?

The basic elements used to construct the block diagram of discrete time Systems are Adder,
Constant multiplier &Unit delay element.

## 29. What is ROC in Z-Transform?

The values of z for which z – transform converges is called region of convergence (ROC). The z-
transform has an infinite power series; hence it is necessary to mention the ROC along with z-
transform.

## DEPT. OF EEE 11 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 30. List any four properties of Z-Transform.

 Linearity
 Time Shifting
 Frequency shift or Frequency translation
 Time reversal

## 31. What are the different methods of evaluating inverse z-transform?

 Partial fraction expansion
 Power series expansion
 Contour integration (Residue method)

## 32. Define sampling theorem.

A continuous time signal can be represented in its samples and recovered back if the
sampling frequency Fs  2B. Here ‘Fs’ is the sampling frequency and ‘B’ is the maximum
frequency present in the signal.

## 33. Check the linearity and stability of g(n),

 since square root is nonlinear, the system is nonlinear.
 As long as x(n) is bounded, its square root is bounded. Hence this system is stable.

## 34. What are the properties of convolution?

1. Commutative property x(n) * h(n) = h(n) * x(n)
2. Associative property [x(n) * h1(n)]*h2(n) = x(n)*[h1(n) * h2(n)]
3. Distributive property x(n) *[ h1(n)+h2(n)] = [x(n)*h1(n)]+[x(n) * h2(n)]

## 35. Define DTFT.

Let us consider the discrete time signal x(n).Its DTFT is denoted as X(w).It is given as
X(w)= x(n)e
-jwn

## 36. State the condition for existence of DTFT?

The conditions are
• If x(n)is absolutely summable then |x(n)|<
• If x(n) is not absolutely summable then it should have finite energy for DTFT to exit.

## 37. What is the DTFT of unit sample?

The DTFT of unit sample is 1 for all values of w.

## 38. List the properties of DTFT.

 Periodicity
 Linearity
 Time shift
 Frequency shift

## DEPT. OF EEE 12 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

 Scaling
 Differentiation in frequency domain
 Time reversal
 Convolution
 Multiplication in time domain
 Parseval’s theorem

## 39. Define DFT.

DFT is defined as X(w)= x(n)e-jwn.
Here x(n) is the discrete time sequence
X(w) is the fourier transform ofx(n).

## 39. Define Zero padding.

The method of appending zero in the given sequence is called as Zero padding.

## 40. Define circularly even sequence.

A Sequence is said to be circularly even if it is symmetric about the point zero on the circle. x(N-
n)=x(n),1<=n<=N-1.

## 41. Define circularly odd sequence.

A Sequence is said to be circularly odd if it is anti-symmetric about point x(0) on the circle

## 42. Define circularly folded sequences.

A circularly folded sequence is represented as x ((-n)) N. It is obtained by plotting x (n) in
clockwise direction along the circle.

## 43. State circular convolution.

This property states that multiplication of two DFT is equal to circular convolution of their
sequence in time domain.

## 44. State parseval’s theorem.

Consider the complex valued sequences x(n) and y(n).If
x(n)y*(n)=1/N X(k)Y*(k)

## 45. Define Z transform.

The Z transform of a discrete time signal x(n) is denoted by X(z) and is given by X(z)= x(n)Z-n.
46. Define N point DFT.
The DFT of discrete sequence x(n) is denoted by X(K). It is given by,
Here k=0,1,2…N-1
Since this summation is taken for N points, it is called as N-point DFT.

## 47. What is DFT of unit impulse δ(n)?

The DFT of unit impulse δ(n) is unity.

## DEPT. OF EEE 13 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 48. List the properties of DFT.

Linearity, P er i o di ci t y , C i r c u l a r s y m m e t r y , s y m m e t r y , T i m e s h i f t , F r e q u e n c y
shift, complex conjugate, convolution, correlation and Parseval’s theorem.

## 49. State Linearity property of DFT.

DFT of linear combination of two or more signals is equal to the sum of linear combination of
DFT of individual signal.

## 50. What is the Periodicity property of DFT?

DFT of a finite length sequence results in a periodic sequence.

PART-B
1. Determine the Z-transform and ROC of

## a. x(n) = rn cos ωn u(n)

b. x(n) = n2an u(n)
c. x(n) = -1/3 (-1/4)n u(n) – 4/3 (2)n u(-n-1)
d. x(n) = an u(n) + bn u(n) + cn u(-n-1) , |a | < | b| < |c|
e. x(n) = cos ωn u(n)
f. x(n) = sin ω0n . u(n)
g. x(n) = an u(n)
h. x(n) = [ 3 (2n) – 4 (3n)] u(n)
2. Find the inverse Z-transform of
a. X(z) = z (z+1) / (z-0.5)3
b. X(z) = 1+3z-1 / 1 + 3z-1 + 2z-2
c. H(z) = 1 / [1 - 3z-1 + 0.5z-2] |z | > 1
2
d. X(z) = [z (z - 4z +5)] / [(z-3) (z-2) ( z-1)] for ROC |2 | < | z| < |3|, |z| > 3, |z|< 1
3. Determine the system function and pole zero pattern for the system described by difference
equation y (n) -0.6 y (n-1) +0.5 y (n-2) = x (n) – 0.7 x (n-2)
4. Determine the pole –zero plot for the system described by the difference equation
y (n) – 3/4 y (n-1) +1/8 y (n-2) = x(n) – x(n-1)
5. Explain the properties of Z-transform.
6. Perform the convolution of the following two sequences using Z-transforms.
x(n) = 0.2n u(n) and h(n) = (0.3)n u(n)
7.A causal LTI system has an impulse response h(n) for which the Z-transform is given by
H(z) = (1+z-1) / [(1 + 1/2z-1) (1 + 1/4z-1). What is the ROC of H (z)? Is the system stable?
8.Find the Z-transform X (z) of an input x (n) that will produce the output
y(n) = -1/3 (-1/4)n u(n) – 4/3 (2)n u(-n-1).Find the impulse response h (n) of the system.
9. Solve the difference equation y(n) -3y(n-1) – 4y(n-2) = 0, n ≥ 0 ,y(-1) = 5
10. Compute the response of the system
y(n) = 0.7 y(n-1)-0.12y(n-2) +x(n-1)+ x (n-2)to the input x(n) = n u(n)
11. What is ROC? Explain with an example.
12. A causal LTI IIR digital filter is characterized by a constant co-efficient difference equation
given by y(n) = x(n-1)-1.2x(n-2)+x(n-3)+1.3 y(n-1) – 1.04 y(n-2)+0.222y(n-3),obtain its transfer
function.

## DEPT. OF EEE 14 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

13. Determine the system function and impulse response of the system described by the
difference equation y(n) = x(n) +2x(n-1)- 4x(n-2) + x(n-3)
14. Solve the difference equation y(n) - 4y(n-1) - +4 y(n-2) = x(n) – x(n-1) with the initial
condition y(-1) = y(-2) = 1
15. Find the impulse response of the system described by the difference equation y(n) = 0.7 y(n-
1) -0.1 y(n-2) +2 x(n) – x(n-2)
16. Determine the z- transform and ROC of the signal x (n) = [3 (2n) – 4 (3n)] u(n).
17. State and prove convolution theorem in z-transform.
18. Given x(n) = δ(n) + 2 δ(n-1) and y(n) = 3 δ(n+1) + δ(n)- δ(n-1). Find x(n) * y(n) and
X(z).Y(z).

Unit – III
1. 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,

## 2. Define Inverse DFT

The Inverse DFT of the sequence of length N is defined as,

## 3. List any four properties of DFT

a. Periodicity
b. Linearity
c. Time reversal
d. Circular time shift e. Duality
f. Circular convolution
g. Symmetry
h. Circular symmetry

## 4. State periodicity property with respect toDFT.

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.

## 5. State periodicity property with respect toDFT.

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.

## 6. 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)

## DEPT. OF EEE 15 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 7. 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.

## 8. What is the need for DFT?

DFT is used for analysis the both periodic and a periodic signals.

## 9. What is zero padding? What are its uses?

Let the sequence x (n) has a length L. If we want to find the N-point DFT (N>L) of the sequence
x(n), we have to add (N-L) zeros to the sequence x(n). This is known as Zero padding.
The uses of zero padding are
1) We can get better display of the frequency spectrum.
2) With zero padding the DFT can be used in linear filtering.

## 10. Why FFT is needed?

The direct evaluation of DFT requires N complex multiplications and N – N complex additions.
Thus for large values of N direct evaluation of the DFT is difficult. By using FFT algorithm the
number of complex computations can be reduced. So we use FFT

## 11. What is FFT?

The Fast Fourier Transform is an algorithm used to compute the DFT. It makes use of the
symmetry and periodicity properties of twiddle factor to effectively reduce the DFT computation
time. It is based on the fundamental principle of decomposing the mutation of DFT of a sequence
of length N into successively smaller DFTs.

12. How many multiplications and additions are required to compute N point DFT using
The number of multiplications and additions required to compute N point DFT Using radix-2
FFT are N log2 N and N/2 log2 N respectively.

## 13. 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=2M, where M is an integer, then this algorithm is
known as radix-2 algorithm.

## 14. What is DIT algorithm?

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

## 15. What DIF algorithm?

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

## DEPT. OF EEE 16 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 16. What are the applications of FFT algorithm?

The applications of FFT algorithm includes
1) Linear filtering
2) Correlation
3) Spectrum analysis

## 17. What is DIT radix-2 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.

## 18. Differentiate DTFT and DFT

DTFT output is continuous in time whereas DFT output is Discrete in time.

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

## 20. What is DIF radix-2 algorithm?

1. The radix 2 DIFFFT is an efficient algorithm for computing DFT in this the output sequence
x(k) is divided in to smaller and smaller.
2. 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(nSimilarly).
3. The N/2 DFT s are divided and expressed in to the combination of N/4 point DFT’ s. This
process is continued until we left with 2-point DFT’s.

21. 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.

## 22. What is meant by in place in DIT and DFT algorithm?

An algorithm that uses the same location to store both the input and output sequence is called in-
place algorithm.

## DEPT. OF EEE 17 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 23. Differentiate between DIT and DIF algorithm

DIT – Time is decimated and input is bi reversed format output in natural order
DIF – Frequency is decimated and input is natural order output is bit reversed Format.

## 24. How many stages are there for 8 point DFT?

There are 3 stages are available for 8 Point DFT

25. How many multiplication terms are required for doing DFT by expressional?
Expression Method and FFT method
Expression –N2
FFT - N /2 logN

## 26. What is DFT?

It is a finite duration discrete frequency sequence which is obtained by sampling one period of
Fourier transform. Sampling is done at ‘N ‘equally spaced points over the period extending from
ω = 0 to ω = 2π.

## 27. What is the DFT of unit impulse δ(n) ?

The DFT of unit impulse δ(n) is unity.

28. Why the result of circular and linear convolution is not same ?
Circular convolution contains same number of samples as that of x(n) and h(n), while in linear
convolution, number of samples in the result(N) are, N =L +M -1. Where, L = Number of
samples in x(n). M = Number of samples in h(n). That is why the result of linear and circular
convolution is not same.

29. How to obtain same result from linear and circular convolution?
* Calculate the value of ‘N’, that means number of samples contained in linear convolution.
* By doing zero padding make the length of every sequence equal to number of samples
contained in linear convolution.
* Perform the circular convolution. The result of linear and circular convolution will be same.

30. How will you perform linear convolution from circular convolution?
* Calculate the value of ‘N’, that means number of samples contained in linear convolution.
* By doing zero padding make the length of every sequence equal to number of samples
contained in linear convolution.
* Perform the circular convolution. The result of linear and circular convolution will be same.

31. What methods are used to do linear filtering of long data sequences?
* Overlap save method.
* Overlap add method.

## DEPT. OF EEE 18 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 32. What is the disadvantage of direct computation of DFT?

For the computation of N-point DFT, N2 complex multiplication and N2 – N complex additions
are required. If the value of N is large then the number of computations will go into lakhs. This
proves inefficiency of direct DFT computation.

33. What is the way to reduce number of arithmetic operations during DFT computation?
Numbers of arithmetic operations involved in the computation of DFT are greatly reduced by
using different FFT algorithms as follows,
• Radix-2 FFT algorithm.
- Radix-2 Decimation In Time (DIT) algorithm.
- Radix-2 Decimation In Frequency (DIF) algorithm.
• Radix-4 FFT algorithm.

## 34. What are the properties of twiddle factor?

The two different factors are,
* Twiddle factor is periodic.
* Twiddle factor is symmetric.

## 35. What is up sampling process and what its effect?

Addition of one zero after each sample in x(n) is called up sampling process. Due to this process,
the entire DFT repeats one time.

## 36. How linear filtering is done using FFT?

Correlation is the basic process of doing linear filtering using FFT. The correlation is nothing but
the convolution with one of the sequence, folded. Thus, by folding the sequence h(n), we can
compute the linear filtering (convolution) using FFT.

## 37. What is meant by “in place-computation” in FFT algorithm?

FFT algorithms, for computing the DFT when the size N is a power of 2 and when it is a power
of 4

38. Compare the number of complex multiplications required for direct calculation and
FFT evolution of N – point DFT if N = 1024.
The number of complex multiplications required using direct computation is N2 = 10242
=1048576. The number of complex multiplications required using FFT is (N/2) log2N = (1024/2)
log21024= 5120.

## 39. What is meant by bit reversed indexing?

"Bit reversal" is just what it sounds like: reversing the bits in a binary word from left to write.
Therefore the MSB's become LSB's and the LSB's become MSB's. The data ordering required by
radix-2 FFT's turns out to be in "bit reversed" order, so bit-reversed indexes are used to combine
FFT stages.

40. Determine the number of multiplication required in finding 64-point DFT using Radix-
2 FFT algorithm.

## DEPT. OF EEE 19 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

The number of multiplications required to compute N point DFT Using radix-2 FFT is N
log2 N Here N = 64, Hence Number of multiplication = 64×log2 (64) = 384.

## 41. Differentiate DTFT and DFT

DTFT output is continuous in time whereas DFT output is Discrete in time.
42. What is circular time shift of sequence?
Shifting the sequenceN in time domain by ‘1’ samples is equivalent to multiplying the
sequence in frequency domain by W kl

## 43. What is the disadvantage of direct computation of DFT?

For the computation of N-point DFT, N2 complex multiplications and N[N-1] Complex
additions are required. If the value of N is large than the number of computations will go into
lakhs. This proves inefficiency of direct DFT computation.
44. What is the way to reduce number of arithmetic operations during DFT computation?
Number of arithmetic operations involved in the computation of DFT is greatly reduced by using
different FFT algorithms as follows.
1. Radix-2 FFT algorithms.
-Radix-2 Decimation in Time (DIT) algorithm.
- Radix-2 Decimation in Frequency (DIF) algorithm.
2. Radix-4 FFT algorithm.
45. What is the computational complexity using FFT algorithm?
1. Complex multiplications = N/2 log2N
2. Complex additions = N log2N

## 46. How linear filtering is done using FFT?

Correlation is the basic process of doing linear filtering using FFT. The correlation is nothing
but the convolution with one of the sequence, folded. Thus, by folding the sequence h (n), we
can compute the linear filtering using FFT.

## 47. What is zero padding? What are its uses?

Let the sequence x (n) has a length L. If we want to find the N point DFT (N>L) of the sequence
x(n). This is known as zero padding. The uses of padding a sequence with zeros are
(i)We can get ‘better display’ of the frequency spectrum.
(ii) With zero padding, the DFT can be used in linear filtering.

## 48. How can we calculate IDFT using FFT algorithm?

The inverse DFT of an N point sequence X(K) is defined as If we take complex conjugate and
multiply by N, we get The right hand side of the above equation is DFT of the sequence X*(K) and
may be computed using any FFT algorithm. The desired output sequence x (n) can be then
obtained by complex conjugating the DFT of the above equation and dividing by N to give

## 49. What are the applications of FFT algorithms?

1. Linear filtering
2. Correlation
3. Spectrum analysis

## DEPT. OF EEE 20 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 50. What is a decimation-in-frequency algorithm?

In this the output sequence X (K) is divided into two N/2 point sequences and each N/2 point
sequences are in turn divided into two N/4 point sequences.

PART-B
1. Perform circular convolution of the sequence using DFT and IDFT technique
x1(n) = {2, 1,2,1} x2 (n) = {0,1,2,3}
2. Compute the DFT of the sequence x(n) = {1,1,1,1,1,1,0,0}
3. From the first principles obtain the signal flow graph for computing 8 – point DFT using
radix-2 DIF-FFT algorithm. An 8 point sequence is given by x(n)={2,2,2,2,1,1,1,1} compute its
8 point DFT of x(n) by radix-2 DIF-FFT
4. Explain any five properties of DFT.
5. Derive DIF – FFT algorithm. Draw its basic butterfly structure and compute the DFT x(n) = (-
1)n using radix 2 DIF – FFT algorithm.
6. Compute the DFT of the sequence x (n) = 1/3 δ (n) – 1/3 δ (n-1) +1/3 δ (n -2)
7. i) Compute the DFT of the sequence x (n) = (-1)n
ii) What are the differences and similarities between DIT – FFT and DIF – FFT algorithms?
8. Compute 4-point DFT of the sequence x (n) = (0, 1, 2, 3)
9. Explain the procedure for finding IDFT using FFT algorithm
10. Derive the decimation-in-frequency radix-2 FFT algorithm for evaluating DFT of the
discrete-time sequence and draw flow graph for 8-point DFT computation.
11. Explain the calculation of inverse DFT using FFT algorithm.
12. Compute the N point DFT of x{n) = an u(n) for cases |a| < 1 and |a| = 1

Unit – IV
1. What is a digital filter?
A digital filter is a device that eliminates noise and extracts the signal of interest from other
signals.

## 2. Analog filters are composed of which parameters?

* pass band
* stop band
* Cut-off frequency

## 3. Define pass band.

It passes certain range of frequencies. In this, attenuation is zero.

## 4. Define stop band.

It suppresses certain range of frequencies. In this, attenuation is infinity.

## DEPT. OF EEE 21 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 5. What is mean by cut-off frequency?

This is the frequency which separates pass band and stop band.

## 6. What is the difference between analog and digital filters?

Analog filters are designed using analog components (R,L,C)while digital filters are
implemented using difference equation and implemented using software.

## 7. What are the basic types of analog filters?

* Low pass filter - LPF
* High pass filter – HPF
* Band pass filter - BPF
* Band stop filter – BSF

## 8. What is the condition for digital filter to be realize?

The impulse response of filter should be causal, h(n) = 0 for n<0.

## 9. Why ideal frequency selective filters are not realizable?

Ideal frequency selective filters are not realizable because they are non causal. That is, its
impulse response is present for negative values of ‘n’ also.

## 10. For IIR filter realization what is required?

Present, past, future samples of input and past values of output are required.

## 11. Why IIR systems are called recursive systems?

Because the feedback connection is present from output side to input

## 12. Which types of structures are used to realize IIR systems?

* Direct form structure
* Cascade form structure
* Parallel form structure

## 13. Why direct form-II structure is preferred most and why?

The numbers of delay elements are reduced in direct form-II structure compared to direct form-I
structure. That means the memory locations are reduced in direct form-II structure.

14. Why direct form-I and direct form-II are called as direct form structures?
The direct form-I and direct form-II structures are obtained directly from the corresponding
transfer function without any rearrangements. So these structures are called as direct form
structures.

## 15. What is advantage of direct form structure?

Implementation of direct form is very easy.

## DEPT. OF EEE 22 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

Both direct form structures are sensitive to the effects of quantization errors in the coefficients.
So practically not preferred
17. What is the use of transpose operation?
If two digital structures have the same transfer function then they are called as equivalent
structures. By using the transpose operation, we can obtain equivalent structure from a given
realization structure.

## 18. What is transposition or flow graph reversal theorem?

If we reverse the directions of all branch transmittances and interchange input and output in the
flow graph then the system transfer function remains unchanged.

## 19. Howa transposed structure is obtained?

* Reverse all signal flow graph directions.
* Change branching nodes into adders and vice-versa.
* Interchange input and output.

## 20. Why feedback is required in IIR systems?

It is required to generate infinitely long impulse response in IIR systems.

## 24. Write the steps in designing chebyshev filter?

1. Find the order of the filter.
2. Find the value of major and minor axis.
3. Calculate the poles.
4. Find the denominator function using the above poles.
5. The numerator polynomial value depends on the value of n.
If n is odd: put s=0 in the denominator polynomial. If n is even put s=0 and divide it by (1+e2)1/2

## DEPT. OF EEE 23 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 26. What is warping effect or frequency warping?

For smaller values of w there exist linear relationship between w and Ω.but for larger values of w
the relationship is nonlinear. This introduces distortion in the Frequency axis. This effect
compresses the magnitude and phase response. This Effect is called warping effect.

## 27. Write a note on pre warping or pre scaling.

The effect of the nonlinear compression at high frequencies can be compensated. When the
desired magnitude response is piecewise constant over frequency, this Compression can be
compensated by introducing a suitable rescaling or prewar Ping the critical frequencies.

28. Give the bilinear transform equation between s plane and z plane
s=2/T (z-1/z+1)

29. Why impulse invariant method is not preferred in the design of IIR filters other Than
low pass filter?
In this method the mapping from s plane to z plane is many to one. Thus there is an infinite
number of poles that map to the same location in the z plane, producing an aliasing effect. It is
inappropriate in designing high pass filters. Therefore this method is not much preferred.

## 30. What are the properties of chebyshev filter?

1. The magnitude response of the chebyshev filter exhibits ripple either in the stop band or the
pass band.
2. The poles of this filter lies on the ellipse.

## 31. What is a disadvantage of BLT method?

The mapping is non-linear and because of this, frequency warping effect takes place.

## 32. What is filter?

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

33. What are the types of digital filter according to their impulse response?
The two digital filters are
 IIR(Infinite impulse response)filter
 FIR (Finite Impulse Response) filter.

## 34. How phase distortion and delay distortion are introduced?

1. The phase distortion is introduced when the phase characteristics of a filter is Nonlinear with
in the desired frequency band

## DEPT. OF EEE 24 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

2. The delay distortion is introduced when the delay is not constant with in the desired frequency
band

## 35. Define IIR filter.

The filters designed by considering all the infinite samples of impulse response are called IIR
filter.

## 36. What is the limitation of approximation of derivative method?

It is suitable only for designing of low pass and band pass IIR digital filters with relatively small
resonant frequencies.

## 37. What are the reasons to use elliptic filters?

It has smallest transition bandwidth and also it is more efficient.

## 38. What is the use of frequency transformation?

It is used to design other filters like HPF, BPF and band reject filters from LPF.

## 39. What are the properties of FIR filters?

 FIR filter is always stable.
 A realizable filter can always be obtained.
 FIR filter has a linear phase response.

## 40. What is bilinear transformation?

The bilinear transformation method overcomes the effect of aliasing that is caused due to the
analog frequency response containing components at or beyond the nyquist frequency. The
bilinear transform is a method of compressing the infinite, straight analog frequency axis to a
finite one long enough to wrap around the unit circle only once.
S = (2/T)[ (Z-1) / (Z+1)]

## 41. What is the main objective of impulse invariant transformation?

The philosophy of this technique is to transform an analog prototype filter into an IIR discrete
time filter whose impulse response [h(n)] is a sampled version of the analog filter’s impulse
response, multiplied by T. This procedure involves choosing the response of the digital filter as
an equi-spaced sampled version of the analog filter.

## 42. What is meant by frequency warping?

When bilinear transformation is applied, the discrete time frequency is related continuous time
frequency as, Ω = 2tan-1ΩT/2. This equation shows that frequency relationship is highly
nonlinear. It is also called frequency warping. This effect can be nullified by applying pre-
warping. The specifications of equivalent analog filter are obtained by following relationship,
Ω = 2/T tan ω/2. This is called pre-warping relationship.

## 43. Define linear phase shift filter

For a filter to have linear phase the phase response θ(w) α w is the angular frequency.
The linear phase filter does not alter the shape of the signal. The necessary and sufficient
condition for a filter to have linear phase is given by,

## DEPT. OF EEE 25 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 44. Mention the advantage of Bilinear Transformation.

Advantages: 1. Many to one mapping. 2. Linear frequency relationship between analog and its
transformed digital frequency,

45. State the condition for a digital filter to be causal and stable
The response of the causal system to an input does not depend on future values of that input, but
depends only on the present and/or past values of the input.
A filter is said to be stable, bounded-input bounded output stable, if every bounded input
produces a bounded output. A bounded signal has amplitude that remains finite.

46. Mention any two procedures for digitizing the transfer function of an analog filter.
1. Impulse Invariant Technique
2. Bilinear Transform Technique

## 47. Distinguish IIR and FIR filters

FIR IIR
Impulse response is finite Impulse Response is infinite

They have perfect linear phase They do not have perfect linear
phase
Non recursive Recursive
Greater flexibility to control the Less flexibility
shape of magnitude response

## 1. Find the order of the filter.

2. Find the value of major and minor axis. λ
3. Calculate the poles.
4. Find the denominator function using the above poles.
5. The numerator polynomial value depends on the value of n. If n is odd: put s=0 in the
denominator polynomial.
6. If n is even put s=0 and divide it by (1+e2)1/2

## 1. From the given specifications find the order of the Filter

2. Find the transfer function from the value of N
3. Find øc
4. Find the transfer function ha(s) for the above value of øc by su s by that value.

## DEPT. OF EEE 26 C.THIAGARAJAN-AP/ECE

EE T65 DIGITAL SIGNAL PROCESSING

## 50. What are the properties of a system?

The various properties of a systems are Stability, Memory, Invertibility, Time invariance &
Linearity

PART-B
1. With suitable examples, describe the realization of linear phase FIR filters
2. Convert the following analog transfer function H(s) = (s+0.2) / [(s+0.2)2 + 4] into equivalent
digital transfer function H (z) by using impulse invariance method assuming T= 1
sec.
3. Convert the following analog transfer function H(s) = 1 / (s+2) (s+4) into equivalent digital
transfer function H (z) by using bilinear transformation with T = 0.5 sec.
4. Design a high pass filter of length 7 samples with cut off frequency of 2 rad / sec using
Hamming window. Plot its magnitude and phase response.
5. For the constraints
0.8 ≤ │H(ω)│≤ 1.0 , 0 ≤ ω ≤ 0.2π
│ H(ω)│ ≤ 0.2, 0.6 π ≤ ω ≤ π
With T= 1 sec determine the system function H(z) for a Butterworth filter using bilinear
transformation.
6. Discuss about the window functions used in design of FIR filters
7. Design a digital Chebyshev filter satisfying the following constraints with T= 1 sec, using
Bilinear transformation.
0.707 ≤ │H (ω) │≤ 1.0, 0 ≤ ω ≤ π/2
│ H (ω) │ ≤ 0.2, 3π/4 ≤ ω ≤ π
8. Using the bilinear transformation and a low pass analog Butterworth prototype, design a low
pass digital filter operating at a rate of 20 KHz and having pass band extending to a 4 KHz with a
maximum pass band attenuation of 0.5 dB and stop band starting at 5KHzwith a minimum stop
band attenuation of 10 dB.
9. Using the bilinear transformation and a low pass analog Chebyshev type I prototype, design a
low pass digital filter operating at a rate of 20 KHz and having pass band extending to a 4 KHz
with a maximum pass band attenuation of 0.5 dB and stop band starting at 5KHzwith a minimum
stop band attenuation of 10 dB.
10. Design a low pass FIR filter of order 7 with cut off frequency Π/3 rad/sec using Hanning
window
11. Convert the following analog filter into digital using IIM method.
H(S) = S2 /(S2 + 0.3S+0.02)
12. Design and realize a low pass filter using a rectangular window by taking 9 samples of w(n)
and with a cutoff frequency of 1.2 rad/sec.
13. Find the order N and the transfer function of analog Chebychev low pass filter for the
following specification: Pass band ripple 3 dB and pass band cut off frequency 1 KHz, stop band
attenuation of 16 dB at stop band frequency of 2 KHz.

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Unit – V
1. What are all the blocks are used to represent the CT signals by its samples?
* Sampler
* Quantizer

## 2. Define sampling process.

Sampling is a process of converting Ct signal into Dt signal.

* Up sampling
* Down sampling

## 4. What is meant by quantizer?

It is a process of converting discrete time continuous amplitude into discrete time discrete
amplitude.

## 5. Define system function?

The ratio between z transform of output signal y(z) to z transform of input signal x(z) is called
system function of the particular system.

## 6. List out the types of quantization process.

Truncation & Rounding

7. Define truncation.
Truncating the sequence by multiplying with window function to get the finite value

## 8. What is meant by limit cycle oscillations?

In fixed point addition, overflow occurs due to excess of results bit, which are stored at the
registers. Due to this overflow, oscillation will occur in the system. Thus oscillation is called as
an overflow limit cycle oscillation.

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23. How would you relate the steady-state noise power due to quantization and the b bits
representing the binary sequence?

## Steady state noise power

Where b is the number of bits excluding sign bit.

24. What are the two kinds of limit cycle behavior in DSP?

## 1. Zero input limit cycle oscillations

2. Overflow limit cycle oscillations

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The autocorrelation of a sequence is the correlation of a sequence with its shifted version, and
this indicates how fast the signal changes.

## 26. What do finite word length effects mean?

The effects due to finite precision representation of numbers in a digital system are called finite
word length effects.

27. List some of the finite word length effects in digital filters.

##  Errors due to quantization of input data.

 Errors due to quantization of filter co-efficient
 Errors due to rounding the product in multiplications
 Limit cycles due to product quantization and overflow in addition.

## a. Sign magnitude format

b. One’s Complement format
c. Two’s Complement format.
In all the three formats, the positive number is same but they differ only in representing
negative numbers.

## 29. Explain the floating-point representation of binary number.

The floating-point number will have a mantissa part. In a given word size the bits allotted for
mantissa and exponent are fixed. The mantissa is used to represent a binary fraction number and
the exponent is a positive or negative binary integer. The value of the exponent can be
adjusted to move the position of binary point in mantissa. Hence this representation is called
floating point.

## 30. What are the types of arithmetic used in digital computers?

The floating point arithmetic and two’s complement arithmetic are the two types of
arithmetic employed in digital systems.

31. What are the two types of quantization employed in digital system?
The two types of quantization in digital system are Truncation and Rounding.

## 32. What is truncation?

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

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## 33. What is rounding?

Rounding is the process of reducing the size of a binary number to finite word sizes of b-bits
such that, the rounded b-bit number is closest to the original unquantized number.

## 34. Explain the process of upward rounding?

In upward rounding of a number of b-bits, first the number is truncated to b-bits by
retaining the most significant b-bits. If the bit next to the least significant bit that is
retained is zero, then zero is added to the least significant bit of the truncated number. If the
bit next to the least significant bit that is retained is one then one is added to the least
significant bit of the truncated number.

## 35. How the digital filter is affected by quantization of filter coefficients?

The quantization of the filter coefficients will modify the value of poles & zeros and so
the location of poles and zeros will be shifted from the desired location. This will create
deviations in the frequency response of the system. Hence the resultant filter will have a
frequency response different from that of the filter with unquantized coefficients.

## 36. How the sensitivity of frequency response to quantization of filter coefficients is

minimized?
The sensitivity of the filter frequency response to quantization of the filter coefficients is
minimized by realizing the filter having a large number of poles and zeros as an
interconnection of second order sections. Hence the filter can be realized in cascade or
parallel form, in which the basic buildings blocks are first order and second order sections.

## 37. What is meant by product quantization error?

In digital computations, the output of multipliers i.e., the product are quantized to finite
word length in order to store them in registers and to be used in subsequent calculations.
The error due to the quantization of the output of multiplier is referred to as product
quantization error.

## 38. Why rounding is preferred for quantizing the product?

In digital system rounding due to the following desirable characteristic of rounding
performs the product quantization
1. The rounding error is independent of the type of arithmetic
2. The mean value of rounding error signal is zero.
3. The variance of the rounding error signal is least.

## 39. Define noise transfer function (NTF)?

The Noise Transfer Function is defined as the transfer function from the noise source to
the filter output. The NTF depends on the structure of the digital networks.

## 40. What are limit cycles?

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

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## 41. What are the two types of limit cycles?

The two types of limit cycles are zero input limit cycles and overflow limit cycles.

## 42. What is zero input limit cycles?

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
cycles, it will continue to remain in limit cycles even when the input is made zero. Hence
these limit cycles are also called zero input limit cycles.

## 43. How overflow limit cycles can be eliminated?

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

## 44. What is the drawback in saturation arithmetic?

The saturation arithmetic introduces nonlinearity in the adder which creates signal
distortion.

## 45. What are the errors generated by A/D process?

The A/D process generates two types of errors. They are quantization error and
saturation error. The quantization error is due to representation of the sampled signal by
a fixed number of digital levels. The saturation errors occur when the analog signal
exceeds the dynamic range of A/D converter.

## 46. What is quantization step size?

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

## 47. Why errors are created in A/D process?

In A/D process the analog signals are sampled and converted to binary. The
sampled analog signal will have infinite precision. In binary representation of b- bits we
have different values with finite precision. The binary values are called quantization
levels. Hence the samples of analog are quantized in order to fit into any one of the
quantized levels. This quantization process introduces errors in the signal.

## 48. What is saturation arithmetic?

In saturation arithmetic when the result of an arithmetic operation 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 r4esult is set to minimum possible value.

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## 49. What is overflow limit cycle?

In fixed point addition the overflow occurs when the sum exceeds the finite word
length of the register used to store the sum. The overflow in addition may lead to
oscillations in the output which is called overflow limit cycles.

## 50. How overflow limit cycles can be eliminated?

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

## 51. What is the drawback in saturation arithmetic?

The saturation arithmetic introduces nonlinearity in the adder which creates signal
distortion.

PART-B
1.Explain the quantization effects in design of digital filters.
2. Illustrate the impact of quantization of filter coefficients on the poles and zeros with an
example
3.Obtain the cascade and parallel realization of system described by difference equation
y(n) = -0.1 y(n-1) + 0.2 y(n-2) + 3x(n) +3.6 x(n-1) + 0.6 x(n-2)
4.Draw the structure for IIR filter in direct form – I and II for the following transfer function H
(z) = (2 + 3 z-1) (4+ 2 z-1 +3 z-2) / (1+0.6 z-1) (1+ z-1+0.5 z-2)
5.Obtain the direct form – I, direct form – II, cascade and parallel form of realization for the
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)
6.Write short note on (a) Truncation and rounding. (b) Coefficient Quantization.
7.Explain about zero input limit cycle oscillations
8. Explain the structure of IIR system