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In case you don't have the Matlab Signal Processing Toolbox
Without kaiserord, we would need to implement Kaiser's
formula [102,59] for estimating the Kaiser-window
required to achieve the given filter specs:
![$\displaystyle \beta = \left\{\begin{array}{ll} 0.1102(A-8.7), & A > 50 \\ [5pt]...
...07886(A-21), & 21< A < 50 \\ [5pt] 0, & A < 21, \\ \end{array} \right. \protect$](http://www.dsprelated.com/josimages/sasp/img2503.png) |
(B.3) |
where

is the desired stop-band attenuation
in
dB (typical
values in audio work are

to

). Note that this estimate for

becomes too small when the filter
passband width approaches
zero. In the limit of a zero-width pass-band, the
frequency response
becomes that of the Kaiser window transform itself. A non-zero
passband width acts as a ``moving average''
lowpass filter on the
sidelobes of the window transform, which brings them down in level.
The
kaiserord estimate assumes some of this sidelobe
smoothing is present.
A similar function from [179] for window
design (as opposed to filter design) is
![$\displaystyle \beta = \left\{\begin{array}{ll} 0, & A<13.26 \\ [5pt] 0.76609(A-...
...6< A < 60 \\ [5pt] 0.12438*(A+6.3), & 60<A<120, \\ \end{array} \right. \protect$](http://www.dsprelated.com/josimages/sasp/img2506.png) |
(B.4) |
where now

is the desired
side-lobe attenuation in
dB (as
opposed to stop-band attenuation). A plot showing Kaiser window
sidelobe level for various values of

is given in
Fig.
3.19.
Similarly, the filter order
is estimated from stop-band
attenuation
and desired transition width
using the
empirical formula
where

is in radians between
0 and

.
Without the function fir1, we would have to manually
implement the window method of filter design by (1) constructing the
impulse response of the ideal bandpass filter
(a cosine
modulated sinc function), (2) computing the Kaiser window
using
the estimated length and
from above, then finally (3)
windowing the ideal impulse response with the Kaiser window to obtain
the FIR filter coefficients
. A manual design of
this nature will be illustrated in the Hilbert transform example of
§B.5.
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Comparison to the Optimal Chebyshev FIR Bandpass Filter
written by Julius Orion Smith III
Julius Smith's background is in electrical engineering (BS Rice 1975, PhD Stanford 1983). He is presently Professor of Music and Associate Professor (by courtesy) of Electrical Engineering at
Stanford's Center for Computer Research in Music and Acoustics (CCRMA), teaching courses and pursuing research related to signal processing applied to music and audio systems. See
http://ccrma.stanford.edu/~jos/ for details.