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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