Bài giảng Digital Signal Processing - Chapter 9: FIR/IIR Digital Filter Design - Võ Trung Dũng
Filter design task: constructing the transfer function of a filter that meets
prescribed frequency response specifications:
Filter design input: set of desired specifications.
Filter design output:
FIR filters: finite impulse response coefficient vector h = [h0, h1, , hN−1].
IIR filters: numerator and denominator coefficient vectors b = [b0,
b1,., bM], a = [1, a1,., aM]
erval by
Its impulse response
The case k = 0
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Highpass, bandpass, and bandstop filters:
Complementary properties: their impulse responses add up to a unit impulse
δ(k) and their frequency responses add up to unity
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Ideal differentiator filter:
Frequency response:
Impulse response:
Hilbert transformer:
Frequency response:
Impulse response:
Symmetric characteristic:
Symmetric classes: frequency response D(ω) is real and even in ω
Antisymmetric classes: D(ω) which is imaginary and odd in ω
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Rectangular Window: consists of truncating, or rectangularly windowing, the
doublesided d(k) to a finite length
Total number of coefficients: will be odd
FIR impulse response:
Causal FIR impulse response: time-delaying the double-sided sequence d(k),
−M ≤ k ≤ M, by M time units to make it causal
or
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Steps of the rectangular window method:
Pick an odd length N = 2M + 1, and let M = (N − 1)/2.
Calculate the N coefficients d(k), and
Make them causal by the delay.
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Example: Determine the length-11, rectangularly windowed impulse response
that approximates:
(a) an ideal lowpass filter of cutoff frequency ωc = π/4,
(b) the ideal differentiator filter, and
(c) the ideal Hilbert transformer filter
Solution: with N = 11, we have M = (N − 1)/2 = 5.
(a) For the lowpass filter:
(b) For differentiator filter:
(c) For the Hilbert transformer
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
DTFT Fourier series expansion:
Its z-transform:
Final length-N filter:
Frequency response
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Linear phase property - symmetric case: the truncated has the same
symmetry/antisymmetry properties as D(ω). In the symmetric case, will be
real and even in ω
where
Final length-N filter:
its magnitude and phase responses
making the phase response piece-wise linear in ω with 180o jumps at those ω where
changes sign
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Approximation accuracy example: consider the design of an
ideal lowpass filter of cutoff frequency ωc = 0.3π, approximated by a rectangularly
windowed response of length N = 41 and then by another one of length N = 121.
For the case N = 41, we have M = (N − 1)/2 = 20. The designed impulse response
is given by
For the second design has N = 121 and M = 60
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Approximation accuracy example:
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Gibbs phenomenon: The ripples in the frequency response H(ω), observed in
arise from the (integrated) ripples of the rectangular window spectrum W(ω). As N
increases, there are three effects:
For ω’s that lie well within the passband or stopband (i.e., points of
continuity), the ripple size decreases as N increases, resulting in flatter
passband and stopband. For such ω, we have as N→∞.
The transition width decreases with increasing N. Note also that for any N,
the windowed response H(ω) is always equal to 0.5 at the cutoff frequency ω
= ωc. (This is a standard property of Fourier series.)
The largest ripples tend to cluster near the passband-to-stopband
discontinuity (from both sides) and do not get smaller with N. Instead, their
size remains approximately constant, about 8.9 percent, independent of N.
Eventually, as N →∞, these ripples get squeezed onto the discontinuity at ω =
ωc, occupying a set of measure zero.
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Rectangular Window: To eliminate the 8.9% passband and stopband ripples,
h(n) can be multiplied by the rectangular window to taper off gradually at its
endpoints, thus reducing the ripple effect.
Hamming window: There exist dozens of windows. Hamming window is a
popular choice:
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Hamming window: consider the design of a length N = 81 lowpass filter with
cutoff frequency ωc = 0.3π.
Note how the Hamming impulse response tapers off to zero more gradually
DSP-Chapter 9-Dr. Dung Trung Vo
Window Method
Hamming window: consider the design of a length N = 81 lowpass filter with
cutoff frequency ωc = 0.3π.
Hamming impulse response tapers off to zero more gradually. Ripples are virtually
eliminated with maximum overshoot of about 0.2% but with wider transition width.
DSP-Chapter 9-Dr. Dung Trung Vo
Kaiser Window
Kaiser Window: provide better control over the filter design specifications than
the rectangular and Hamming window designs. A flexible set of specifications in
which the designer can arbitrarily specify the amount of passband and stopband
overshoot δpass, δstop, as well as the transition width ∆f
DSP-Chapter 9-Dr. Dung Trung Vo
Kaiser Window
passband/stopband frequencies {fpass, fstop}:
Digital frequencies:
passband and stopband overshoots: are usually expressed in dB
Pass back and forth between the specification sets:
Equal passband and stopband ripples
DSP-Chapter 9-Dr. Dung Trung Vo
Kaiser Window
Kaiser window:
where I0(x) is the modified Bessel function of the first kind and 0th order
Specifications for rectangular, Hamming, and Kaiser windows:
Other FIR Design Methods: Parks-McClellan method based on the so-called
optimum equiripple Chebyshev approximation generally results in shorter filters
DSP-Chapter 9-Dr. Dung Trung Vo
Frequency Sampling Method
Frequency sampling method: For arbitrary frequency responses D(ω) while
The window method is very convenient for designing ideally shaped filters
Impulse response:
where
Frequencies ωi: spanning equally the interval [−π,π], instead of the standard
DFT interval [0, 2π]
Final designed filter: will be the delayed and windowed version of # d(k)
DSP-Chapter 9-Dr. Dung Trung Vo
Fast Fourier transform (FFT)
Homework: 10.1, 10.3-10.4
DSP-Chapter 9-Dr. Dung Trung Vo
IIR Digital Filter Design
Bilinear Transformation: Instead of designing the digital filter directly, the
method maps the digital filter into an equivalent analog filter, which can be
designed by one of the well-developed analog filter design methods, such as
Butterworth, Chebyshev, or elliptic filter designs. The designed analog filter is then
mapped back into the desired digital filter
DSP-Chapter 9-Dr. Dung Trung Vo
IIR Digital Filter Design
Mapping between the s and z planes: The z-plane design of the digital filter is
replaced by an s-plane design of the equivalent analog filter
Mapping between the physical digital frequency and equivalent analog
frequency: due to and , then or
Bilinear transformation: is a particular case of f
Mapping of frequencies (frequency prewarping transformation):
or
DSP-Chapter 9-Dr. Dung Trung Vo
IIR Digital Filter Design
Other versions of the bilinear transformation:
Corresponding frequency maps:
DSP-Chapter 9-Dr. Dung Trung Vo
IIR Digital Filter Design
Overall design method:
Determine magnitude response specifications for the digital filter
Transform the specification by the appropriate prewarping transformation
into the specifications of an equivalent analog filter.
Design the equivalent analog filter using an analog filter design technique,
say Ha(s).
Map back the analog filter using the bilinear transformation into the desired
digital filter H(z)
Or corresponding frequency responses
DSP-Chapter 9-Dr. Dung Trung Vo
IIR Digital Filter Design
Property of the bilinear transformation: it maps the left-hand s-plane into
the inside of the unit circle on the z-plane. Because all analog filter design methods
give rise to stable and causal transfer functions Ha(s), so the digital filter H(z) will
also be stable and causal
From bilinear transformation
DSP-Chapter 9-Dr. Dung Trung Vo
First-Order Lowpass IIR Design
First-Order Lowpass: that has a prescribed cutoff frequency, say fc, and
operates at a given sampling rate fs.
Filter form: the design problem is to determine the filter coefficients {b0, b1, a1}
in terms of the cutoff frequency fc and rate fs
Digital cutoff frequency: to be the frequency at which |H(ω)|2 drops by a
factor of G2c < 1, or a drop in dB
DSP-Chapter 9-Dr. Dung Trung Vo
First-Order Lowpass IIR Design
Cutoff frequency specifications for lowpass digital filter:
Popular choice
DSP-Chapter 9-Dr. Dung Trung Vo
First-Order Lowpass IIR Design
Prewarp the cutoff frequency: to get the cutoff frequency of the equivalent
analog filter:
Design a first-order analog filter: with cutoff frequency is Ωc
DSP-Chapter 9-Dr. Dung Trung Vo
First-Order Lowpass IIR Design
Frequency and magnitude responses:
Cutoff condition:
Lead to
Transform the filter to the z-domain by the bilinear transformation
where
DSP-Chapter 9-Dr. Dung Trung Vo
First-Order Lowpass IIR Design
Frequency and magnitude responses:
Cutoff condition:
Lead to
Transform the filter to the z-domain by the bilinear transformation
where
DSP-Chapter 9-Dr. Dung Trung Vo
First-Order Lowpass IIR Design
Example: Design a lowpass digital filter operating at a rate of 10 kHz, whose 3-
dB frequency is 1 kHz.
Solution: digital cutoff frequency is
Prewarped analog version
Filter parameters: with
Digital filter transfer function
DSP-Chapter 9-Dr. Dung Trung Vo
First-Order Lowpass IIR Design
Example: Design a lowpass digital filter operating at a rate of 10 kHz, whose 3-
dB frequency is 1 kHz.
Solution:
DSP-Chapter 9-Dr. Dung Trung Vo
First-Order Highpass IIR Design
Highpass analog first-order filter form:
Highpass digital filter:
where
Frequency and magnitude responses:
DSP-Chapter 9-Dr. Dung Trung Vo
First-Order Highpass IIR Design
Cutoff condition:
lead to
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