INTRODUCTION

Digital filter are linear systems. The impulse response completely

characterizes any linear system. If the impulse response has a finite
number of terms then the filter is known as Finite Impulse Response
(FIR) filters.
The FIR filter generates the output samples by forming a weighted sum of
the input samples and a limited number of previous input samples.






The weights are nothing but the samples of impulse response of the filter.
Thus the output of a FIR filter of length m is given by performing the
convolution of the incoming sequence of samples with the impulse
response of the filter.FIR filters are always stable since there is no
feedback. It is very simple to implement.
The n
th
output will be given by:
y(n) = a
0
x
n
+ a
1
x
n-1
+ + a
m
x
n-m
Note that the above filter will need memory to store m samples and
(m+1) number of co-efficients.

Fig.1 Block Diagram of
FIR Digital Filter
z
m


The frequency response of the above filter is given by:
e
e

Thus the frequency response depends upon the frequency e of the input
signal, the sampling interval and the set of co-efficients a
m
. The
following important points are worth nothing:
1. The frequency response of a digital filter is a periodic function
with a period equal to

.
2. The frequency response given by the function f(je) is the Fourier
Transform of the impulse response (a
0
,,a
m
) of the digital
filter.
The transfer function of the filter in the z-domain is given by:

= a
0
z
m
+ a
1
z
m-1
+ a
2
z
m-2
+ + a
m


FIR Filter Characteristics
The filters are assumed to be characterized by means of finite-
duration impulse responses of duration equal to one power-frequency
cycle of period T and having values +1 or -1 at any instant during that
period


To estimate the magnitude of fundamental and second harmonic
components of the input, four filters are required; two for the fundamental
and two for the second harmonic component. Their responses are S
1
(t),
C
1
(t), S
2
(t) and C
2
(t) which in turn are defined by following equations:
S
1
(t)=










C
1
(t)=

C
1
(t)
T
3T/4
-1
T/4
+1
S
1
(t)
T
-1
+1
T/2
t
Fig 2
T/2
Fig 3

S
2
(t)=

C
2
(t)=

C
2
(t)
-1
+1
7T/8
5T/8
3T/8
T/8
S
2
(t)
-1
+1
3T/4
T
T/2
T
Fig 4
Fig 5
It will be apparent that S
1
(t), C
1
(t) are impulse responses for the sine and
cosine parts of the fundamental component, and S
2
(t), C
2
(t) are impulse
responses for the sine and cosine parts of second-harmonic components.
Assume N is the number of samples per cycle of the current i(t) and
is chosen as a multiple of eight. In this case, the time between successive
samples is t=2/(Ne
o
) and i
k
=i(t
k
) is the k
th
sample at any time t=kt.
The result of time discrete convolution of the samples i
k
with the impulse
responses defined by the previous equations will then be given as:
] [ ) ( 1
2 /
1
2 /
=
+
=
N
k
N k k
i i t S
] ) ( [ ) ( 1
4 / 3 2 /
4 /
1
4 / N k N k
N
k
N k k
i i i i t C
+ +
=
+
+ + =

] [ ) ( 2
4 / 3 2 /
4 /
1
4 / N k N k
N
k
N k k
i i i i t S
+ +
=
+
+ + =

] ) ( ) ( ) ( [ ) ( 2
8 / 7 4 / 3 8 / 5 2 / 8 / 3 4 /
8 /
1
8 / N k N k N k N k N k N k
N
k
N k k
i i i i i i i i t C
+ + + + + +
=
+
+ + + + + =

The criteria used to distinguished between inrush currents and currents due
to internal fault is based on evaluating the ratio c , which is calculated
from the ratio of the larger of the two components of each pair of filter
outputs as described in above equations such that

The values of c for the detection of internal faults are:
0 c for X/R=5
0 c X/R=10
0 c X/R=20
Fig 6
Discrimination
between
inrush and
internal fault
currents
Where X/R is the system reactance-resistance ratio.
PROPERTIES OF FIR FILTER
The Filter coefficient is

Where x(n)-represents the filter input
b
k
- represents the filter co-efficient
y(n)- represents the filter output
N- is the number of filter co-efficient (order of
the filter)
If the signal x[n] is replaced by an impulse o[n] then:

y(n) = b
0
o[0]

+ b
1
o[1] + + b
m
o[-N]

therefore y(n) = b
0
o[n]

+ b
1
o[n-1] + + b
m
o[n-N]

o[n-k] =



Finally b
0
= h[0]
b
1
= h[1]

b
k
= h[k]
The coefficients of a filter are the same as the impulse response samples of
the filter. By taking the z-transform of h[n], H(z):
| | | |

=
=
1
0

N
k
k
k n x b n y
| | | |

=
=
1
0

N
k
k
k n b n y o

Replacing z by e
je
in order to find the frequency response leads to:

Since e
-j2tk
= 1 then:

Therefore Frequency Response of FIR

FIR filters have a periodic frequency response and the period is 2t.
ADVANTAGES
1. FIR filters with exactly linear phase can easily be designed.
2. There exist computationally efficient realizations for implementing
FIR filters.
3. Excellent design methods are available for various kinds of FIR
filters with arbitrary specifications.
4. The output noise due to multiplication round off errors in an FIR
filter is usually very low and the sensitivity to variations in the filter
co-efficient is also low.
DISADVANTAGES
They require, especially in applications demanding narrow transition
bands, considerably more arithmetic operations and hardware
components, such as multipliers, adders and delay elements than do
comparable IIR filters
( ) | |

=
1
0
N
n
n
z n h z H
( ) ( ) | |

=
= =
1
0
N
n
jn j
e z
e n h e H z H
j
e e
e
( ) | |
( )
| |

=
+
=
= =
+
1
0
1
0
2
2
N
n
jn
N
n
jn
e z
e n h e n h z H
e t e
t e
( ) ( )
e t e j k j
e H e H =
+2