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FIRfilters

FIR filters, or finite impulse response filters, are digital filters whose impulse response h[n] has finite duration. The output y[n] is the convolution of the input x[n] with h[n], expressed as y[n] = sum_{k=0}^{M} h[k] x[n−k], where M is the filter order. Because the impulse response is finite, FIR filters are inherently stable for bounded inputs, and they can be designed to realize a wide range of frequency responses.

A notable property of FIR filters is the possibility of linear phase. If the coefficients are symmetric

Design methods for FIR filters include the window method, the frequency sampling method, and optimal approaches.

In practice, longer filter orders give sharper transitions but higher computational cost. Symmetric coefficients can reduce

Applications of FIR filters are widespread, including anti-aliasing and reconstruction filters in sampling systems, audio processing,

(h[n]
=
h[M−n])
or
anti-symmetric,
the
filter
exhibits
linear
phase,
meaning
it
preserves
the
waveform
shape
within
the
passband
while
applying
the
desired
phase
shift.
The
frequency
response
H(ω)
is
the
discrete-time
Fourier
transform
of
h[n].
The
window
method
starts
from
an
ideal,
infinite
impulse
response
and
truncates
it
with
a
window
function
such
as
Rectangular,
Hamming,
Hann,
Blackman,
or
Kaiser,
trading
transition
width
against
ripple.
The
Parks–McClellan
algorithm
(Remez
exchange)
designs
equiripple
FIRs
that
minimize
the
maximum
error
in
a
specified
passband
and
stopband.
Frequency
sampling
specifies
a
desired
response
at
a
set
of
frequencies
and
derives
h[n]
by
interpolation.
Least-squares
methods
optimize
filter
coefficients
to
best
fit
a
target
response
in
a
defined
sense.
multiplications
by
about
half.
FIR
filtering
can
be
implemented
efficiently
with
direct
convolution
for
small
filters
or
FFT-based
techniques
for
long
filters.
communications,
smoothing,
and
general
digital
signal
processing
tasks
that
require
robust,
stable
filtering
with
controllable
phase
characteristics.