Signal Processing Toolbox | Help Desk |
firls
Least square linear-phase FIR filter design.
b = firls(n,f,a) b = firls(n,f,a,w) b = firls(n,f,a,'ftype
') b = firls(n,f,a,w,'ftype
')
firls
designs a linear-phase FIR filter that minimizes the weighted, integrated squared error between an ideal piecewise linear function and the magnitude response of the filter over a set of desired frequency bands.
b = firls(n,f,a)
returns row vector b
containing the n+1
coefficients of the order n
FIR filter whose frequency-amplitude characteristics approximately match those given by vectors f
and a
. The output filter coefficients, or "taps," in b
obey the symmetry relationn
odd) and type II (n
even) linear-phase filters. Vectors f
and a
specify the frequency-amplitude characteristics of the filter:
f
is a vector of pairs of frequency points, specified in the range between 0 and 1, where 1 corresponds to half the sampling frequency (the Nyquist frequency). The frequencies must be in increasing order. Duplicate frequency points are allowed and, in fact, can be used to design a filter exactly the same as those returned by the fir1
and fir2
functions with a rectangular or boxcar
window.
a
is a vector containing the desired amplitude at the points specified in f
.
The desired amplitude function at frequencies between pairs of points (f(k), f(k+1)) for k odd is the line segment connecting the points (f(k), a(k)) and (f(k+1), a(k+1)).
The desired amplitude function at frequencies between pairs of points (f(k), f(k+1)) for k even is unspecified. These are transition or "don't care" regions.
f
and a
are the same length. This length must be an even number.
f
and a
vectors in defining a desired amplitude response is
b = firls(n,f,a,w)
uses the weights in vector w
to weight the fit in each frequency band. The length of w
is half the length of f
and a
, so there is exactly one weight per band.
b = firls(n,f,a,'ftype
')
and
b = firls(n,f,a,w,'ftype
')
specify a filter type, where ftype
is
hilbert
for linear-phase filters with odd symmetry (type III and type IV)
The output coefficients in b
obey the relation b(k) = -b(n + 2 - k), k = 1, ..., n + 1. This class of filters includes the Hilbert transformer, which has a desired amplitude of 1 across the entire band.
differentiator
for type III and type IV filters, using a special weighting technique
For nonzero amplitude bands, the integrated squared error has a weight of (1/f)2 so that the error at low frequencies is much smaller than at high frequencies. For FIR differentiators, which have an amplitude characteristic proportional to frequency, the filters minimize the relative integrated squared error (the integral of the square of the ratio of the error to the desired amplitude).
b = firls(255,[0 0.25 0.3 1],[1 1 0 0]);Design a 31 coefficient differentiator:
b = firls(30,[0 0.9],[0 0.9],'differentiator');Design a 24th-order anti-symmetric filter with piecewise linear passbands and plot the desired and actual frequency response:
F = [0 0.3 0.4 0.6 0.7 0.9]; A = [0 1 0 0 0.5 0.5]; b = firls(24,F,A,'hilbert'); for i=1:2:6, plot([F(i) F(i+1)],[A(i) A(i+1)],'- -'), hold on end [H,f] = freqz(b,1,512,2); plot(f,abs(H)), grid on, hold offReference [1] describes the theoretical approach that
![]()
firls
takes. The function solves a system of linear equations involving an inner product matrix of size roughly n/2
using MATLAB's \
operator.
This function designs type I, II, III, and IV linear-phase filters. Type I and II are the defaults for n
even and odd respectively, while the 'hilbert'
and 'differentiator'
flags produce type III (n
even) and IV (n
odd) filters. The various filter types have different symmetries and constraints on their frequency responses (see [2] for details).
|
---|
F must be even length. F and A must be equal lengths. Requires symmetry to be 'hilbert' or 'differentiator'. Requires one weight per band. Frequencies in F must be nondecreasing. Frequencies in F must be in range [0,1].A more serious warning message is
Warning: Matrix is close to singular or badly scaled.This tends to happen when the filter length times the transition width grows large. In this case, the filter coefficients
b
might not represent the desired filter. You can check the filter by looking at its frequency response.
Window-based finite impulse response filter design-- standard response. |
|
Window-based finite impulse response filter design-- arbitrary response. |
|