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IFTM UNIVERSITY, MORADABAD

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING


SUBJECT: DIGITAL SIGNAL PROCESSING (EEC-602)
CLASS TEST: 1
Q1. What is ROC? State the properties of ROC in context of Z-transform.
Q2. What is the relationship between Z transform and the Discrete Fourier transform?
Q3. Compute DFT of 4-point sequence x(n) = { 0, 1, 2, 3}
Q4. Compute IDFT for the frequency component X(k) = { 60, 0, -4, 0}
Q5. Perform the circular convolution of the following two sequences.

x1(n) = {2, 1, 2, 1}
x2(n) = { 1, 2, 3, 4}

Solution:Q1
ROC: The z-transform is an infinite power series; it exists only for those values of z for

which the series converges. The region of convergence (ROC) of X (z) is set of all values
of z for which X (z) attains a finite value. The ROC of a finite duration signal is the entire
z-plane, except possibly the point
. These points are excluded because zn
(when n > 0) becomes unbounded for z = and zn (when n > 0) becomes unbounded
for z = 0.

Properties of ROC:
1. If x[n] is of finite duration, then the ROC is the entire z-plane (the z-transform summation converges,
i.e., X(z) exists, for any z) except possibly z =0 and/or
.
2. The ROC of X(z) consists of a ring centered about the origin in the z-plane. The inner boundary can extend
inward to the origin in some cases, and the outer can extend to infinity in other cases.
3. If x[n] is right sided and the circle |z|=r0 is in the ROC, then any finite z for which |z|>r0 is also in the ROC.
4. If x[n] is left sided and the circle |z|=r0 is in the ROC, then any z for which 0<|z|<r0 is also in the ROC.
5. If x[n] is two-sided, then the ROC is the intersection of the two one-sided ROCs corresponding to the two
one-sided parts of x[n]. This intersection can be either a ring or an empty set.
6. If X(z) is rational, then its ROC does not contain any poles (by definition
dose not exist).
The ROC is bounded by the poles or extends to infinity.
7. If X(z) is a rational z-transform of a right sided function x[n], then the ROC is the region outside the outmost
pole. If x[n]=0 for n<0 then the ROC includes
8. If X(z) is a rational z-transform of a left sided function x[n], then the ROC is inside the innermost pole.
If x[n]=0 for

(anti-causal), then the ROC includes z=0.

Solution: Q2
Let us consider a sequence x(n) having z-transform with ROC that includes the

unit circle. If X(z) is sampled at the N equally spaced points on the unit circle. If X(z) is
sampled at N equally spaced points on the unit circle.

We obtain

Expression is (2) identical to the Fourier transform X(w) evaluated at the N. equally
spaced. Frequencies

If the sequence x(n) has a finite duration of length N or less, the sequence can be
recovered from its N-point DFT. Hence its Z-transform is uniquely determined by its Npoint DFI. Consequently, X(z) can be expressed as a function of the DFT {X(k)} as
follows

When evaluated on the unit circle (3) yields the Fourier transform of the finite
duration sequence in terms of its DFT in the form:

This expression for Fourier transform is a polynomial interpolation formula for X(w)
expressed in terms of the, values {x(k)) of the polynomial at a set of equally spaced
discrete frequencies

Solution:Q3
By matrix method:

Solution: Q4

Solution: Q5

Solution by DFT and IDFT method:

IFTM UNIVERSITY, MORADABAD

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING


SUBJECT: DIGITAL SIGNAL PROCESSING (EEC-602)
CLASS TEST: 2
Q1. What are the advantages of FFT algorithm?
Q2. What are the differences and similarities between DIF and DIT
algorithms?
Q3. Compute 4-point DFT for the sequence x(n) = { 14, 16,14, 16} using
DIT & DIF FFT algorithm.
Q4. Explain 8-point DIT FFT algorithm using butterfly diagram.
Q5. Explain 8-point DIF FFT algorithm using butterfly diagram.

Solution: Q1
Advantages of FFT algorithm: Fast fourier transform reduces the computation time.
In DFT computation, number of multiplication is N2 and the number of addition is N(N1). In FFT algorithm, number of multiplication is only N/2(log 2N) . Hence FFT reduces the
number of elements (adder, multiplier Z &delay elements). This is achieved by
effectively utilizing the symmetric and periodicity properties of Fourier transform.

Solution: Q2
Differences:
For DIT the input is bit reversed while the output is in natural order, whereas for DIF the
input is in natural order while the output is bit reversed.
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.

Solution: Q3
Using DIT algorithm:

Using DIF algorithm:

Solution: Q4

Radix-2 DIT Algoritm:

Solution: Q5

In DIF N Point DFT is splitted into N/2 points DFTs. X(k) is splitted with k even and k odd this is
called Decimation in frequency(DIF FFT).
N point DFT is given as
N-1
X(k)

= x (n) WNkn

(1)

n=0

Since the sequence x(n) is splitted N/2 point samples, thus


N/2-1
X(k)

N/2-1

= x (n) WNkn + x (n + N/2) WNk(n+N/2)


m=0

N/2-1

(2)

m=0

N/2-1

X(k) = x (n) WNkn + WNkN/2 x (n + N/2) WNkn


m=0
m=0

N/2-1

N/2-1

X(k) = x (n) WNkn + (-1)k x (n + N/2) WNkn


m=0
m=0
N/2-1
X(k) =
m=0

x (n) + (-1)k x(n + N/2) WNkn

(3)

Let us split X(k) into even and odd numbered samples


N/2-1
X(2k) =
m=0

x (n) + (-1)2k x(n + N/2) WN2kn

(4)

N/2-1
X(2k+1) =
m=0

x (n)+(-1)(2k+1) x(n + N/2)WN(2k+1)n

(5)

Equation (4) and (5) are thus simplified as


g1(n) =
g2(n) =

x (n) + x(n + N/2)


x (n) - x(n + N/2) WNn

Fig 1 shows Butterfly computation in DIF FFT.


a

A= a + b

W Nr
B= (a b)WNr
Fig 1. BUTTERFLY COMPUTATION

Fig 2 shows signal flow graph and stages for computation of radix-2 DIF FFT algorithm of N=4
x(0)

x(1)

x(2)
x(3)

w40

w41

X(0)
w40

X(2)
X(1)

w4 0

X(3)

Fig 2. SIGNAL FLOW GRAPH FOR RADIX- DIF FFT N=4


Fig 3 shows signal flow graph and stages for computation of radix-2 DIF FFT algorithm of N=8
x(0)

A1

A2

X(0)

x(1)

B1

B2

w8 0

X(4)

x(2)

C1

w8 0

C2

w80

X(2)

x(3)

D1

w8 2

D2

X(6)
X(1)

x(4)

w80

E1

E2

x(5)

w8 1

F1

F2

w80

X(5)

x(6)

w82

G1

G2

w80

X(3)

x(7)

w8 3

H1

w8 0
w82

H2

Fig 3. SIGNAL FLOW GRAPH FOR RADIX- DIF FFT N=8

X(7)

IFTM UNIVERSITY, MORADABAD


DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING
SUBJECT: DIGITAL SIGNAL PROCESSING (EEC-602)
CLASS TEST: 3
Q1.What is the difference between Direct form-I & Direct form II realization of IIR filter?
Q2.Write short note on cascade realizations of IIR filter.
Q3.What are the basic building blocks for realizing a digital filter?
Q4. What is the importance of Windowing?
Q5. Compare Bilinear Transformation and Impulse Invariant Transformation.

Solution: Q1
SL
No.
1.

2.

3.

DIRECT FORM-I

DIRECT FORM-II

Direct form I realization of H(z) can be


obtained by cascading the realization of
H1(z) which is all zero system first and
then H2(z) which is all pole system.
There are M+N-1 unit delay blocks. One
unit delay block requires one memory
location. Hence direct form structure
requires M+N-1 memory locations.

Direct form realization of H(z) can be


obtained by cascading the realization of
H1(z) which is all pole system and H2(z)
which is all zero system.
Direct Form II structure has reduced
memory requirement compared to Direct
form I structure. Hence it is called canonic
form.

Direct Form I realization requires M+N+1


number of multiplications and M+N
number of additions and M+N+1 number
of memory locations.

The direct form II requires same number of


multiplications(M+N+1)
and
additions
(M+N) as that of direct form I.

Solution: Q2
CASCADE FORM STRUCTURE FOR IIR SYSTEMS:
In cascade form, stages are cascaded (connected) in series. The output of one system is input to
another. Thus total K number of stages are cascaded. The total system function 'H' is given by
H= H1(z) . H2(z). Hk(z)
H= Y1(z)/X1(z). Y2(z)/X2(z). Yk(z)/Xk(z)

(1)
(2)

k
H(z)= Hk(z)

(3)

k=1
x(n)=x1(n)

H1(z)

y1(n)=x2(n)

H2(z)

y2(n)=x3(n)

Hk(z)

yk(n)=y(n)

FIG - CASCADE FORM REALIZATION OF IIR SYSTEM


Each H1(z), H2(z) etc is a second order section and it is realized by the direct form as shown in
below figure.
System function for IIR systems
M
N
k
H(z) = bk z / 1+ ak zk
K=0
k=1

(1)

Expanding the above terms we have


H(z)= H1(z) . H2(z). Hk(z)
where HK(z) = bk0 + bk1 z-1 + bk2 z-2 / 1 + ak1 z-1 + ak2 z-2

(2)

Thus Direct form of second order IIR system is shown as

X(n)

Y(n)

+
Z-1

-ak1

bk1

+
Z-1

-ak2

bk2

FIG - DIRECT FORM REALIZATION OF IIR SECOND ORDER SYSTEM (CASCADE)

Solution: Q3
The basic building Blocks for realizing the digital filters are Adder, Multiplier and delay or
memory elements

A y[n]
y[n] x[n]
x[n]
Multiplier
w[n]
Adder x[n]
x[n]
x[n]
y[n]
x[n]
Unit delay Pick-off node
z 1

Solution: Q4
Ans: 1. The infinite

duration impulse response can be converted to a finite duration


impulse response by trucating the infinite series at
But this results in undesirable
oscillations in the pass-band and step-band of the digital filter. This is due to slow
convergence of the Fourier series near the point of discontinuity. These undesirable
oscillations can be reduced by using a set of time limited weighing functions z e referred
as windowing function.
2. The windowing function consists of main lobe which contains most of the energy of
window function and side lobes which decay rapidly
3 A major effect of windowing is that the discontinuities is
transition bands between values on either side of the discontinuity

are converted into

4 Window function have side lobes that decrease in energy rapidly as tends to

Solution: Q5
Bilinear Transformation

Impulse Invariant Transformation

1. It is one to one mapping.

1. It is many to one mapping.

2. The relation between analog and


digital frequency is nonlinear, ie
=2/T tan( /2).

2.The relation between analog and


digital frequency is linear, i.e. =T or
=/T.

3. Due to nonlinear relation between


and distortion occurs in frequency
domain of digital filter.

3. The aliasing error occur due to


sampling thus this method is suitable for
design of only band limitied filters such
Low pass and Band pass.

4. Due to the warping effect both


amplitude and phase response of
analog filter are affected but the

4. The frequency response of analog can


be preserved by selecting low sampling

magnitude
response
may
be
preserved by applying pre- warping
procedure.

time or high sanpling frequency.