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Without kaiserord, we would need to implement Kaiser's
formula [115,67] for estimating the Kaiser-window
required to achieve the given filter specs:
|
(5.11) |
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 pass-band 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
pass-band width acts as a ``moving average'' lowpass filter on the
side-lobes of the window transform, which brings them down in level.
The kaiserord estimate assumes some of this side-lobe
smoothing is present.
A similar function from [198] for window
design (as opposed to filter design5.7) is
|
(5.12) |
where now
is the desired side-lobe attenuation in dB (as
opposed to stop-band attenuation). A plot showing Kaiser window
side-lobe level for various values of
is given in
Fig.3.28.
Similarly, the filter order
is estimated from stop-band
attenuation
and desired transition width
using the
empirical formula
|
(5.13) |
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
§4.6.
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