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Digital filter (IV) -- converting analog filter into digital filter
2022-07-28 17:58:00 【A bone loving cat】
Analog filter design
lead
Digital filter ( One )–IIR And FIR The basic structure and MATLAB Realization
Digital filter ( Two )– Minimum phase delay system and all pass system
Digital filter ( 3、 ... and )– Analog filter design
1. Mapping method
The purpose of mapping is to convert from analog filter to digital filter , This process is from the known analog filter system function H a ( s ) H_a(s) Ha(s) System functions mapped to digital filters H ( z ) H(z) H(z), Therefore, from analog filter to digital filter is from s Convert plane to z Plane . This mapping needs to meet two requirements :
- H ( z ) H(z) H(z) The frequency response of should be able to imitate H a ( s ) H_a(s) Ha(s) Frequency response of ,s The imaginary axis of the plane is mapped to z The unit circle of the plane .
- Causal stable H a ( s ) H_a(s) Ha(s) Can be mapped to causal instability H ( z ) H(z) H(z), namely s The left half plane of the plane R e [ s ] < 0 Re[s]<0 Re[s]<0 To map to z Inside the plane unit circle ∣ z ∣ < 1 |z|<1 ∣z∣<1
There are two general transformation methods : Impulse response invariance method and bilinear transformation method
2. Impulse response invariant method
2.1 Transformation steps
Impulse response invariable method is to make the unit impulse response of digital filter h ( n ) h(n) h(n) Simulate the impulse response of analog filter h a ( t ) h_a(t) ha(t), in other words , We will h a ( t ) h_a(t) ha(t) Sampling at equal intervals , bring h ( n ) h(n) h(n) Exactly equal to h a ( t ) h_a(t) ha(t) Of T Interval sampling value , namely :
h ( n ) = h a ( t ) ∣ t = n T h(n)=h_a(t)|_{t=nT} h(n)=ha(t)∣t=nT
Assume h ( n ) < − > H ( z ) , h a ( t ) < − > H a ( s ) h(n) <->H(z), h_a(t)<->H_a(s) h(n)<−>H(z),ha(t)<−>Ha(s), The process of digitizing the analog filter is :
H a ( s ) − > h a ( t ) − > h ( n ) − > H ( z ) H_a(s)->h_a(t)->h(n)->H(z) Ha(s)−>ha(t)−>h(n)−>H(z)
This process is also called time domain sampling 、 The process of periodic extension in frequency domain .
The steps of designing digital low-pass filter using impulse response invariance method are as follows :
- First step
According to the given index of digital low-pass filter w p w_p wp, w s t w_{st} wst, δ p \delta_p δp, δ s \delta_s δs; - The second step
Choose the right one T value , Solve the simulation index Ω p = w p T \Omega_p=\frac{w_p}{T} Ωp=Twp, Ω s t = w s t T \Omega_{st}=\frac{w_{st}}{T} Ωst=Twst - The third step
According to the index w p w_p wp, w s t w_{st} wst, δ p \delta_p δp, δ s \delta_s δs, Design analog filter , And the system function H a ( s ) H_a(s) Ha(s) - Step four
Set the system function H a ( s ) H_a(s) Ha(s) Carry out partial fractional expansion , Unfold into ( Look up the table for Factorization )
H a ( s ) = ∑ k = 1 N A k s − s k H_a(s)=\sum_{k=1}^N \frac{A_k}{s-s_k} Ha(s)=k=1∑Ns−skAk - Step five
According to the impulse response unchanged method , The system function of the digital filter is
H ( z ) = ∑ k = 1 N T A k 1 − e s k T z − 1 H(z)=\sum_{k=1}^N \frac{TA_k}{1-e^{s_kTz^{-1}}} H(z)=k=1∑N1−eskTz−1TAk
Sampling interval T The value of does not affect the design of the filter , For the sake of calculation , Usually take 1 Mostly .
Here are two examples to illustrate :
- example 1


- example 1


2.2 Advantages and disadvantages of impulse response unchanged method
Advantages of impulse response unchanged method :
- The time domain approximation of impulse response invariable method is good ;
- Analog frequency Ω \Omega Ω And digital frequency w w w There is a linear mapping relationship between w = Ω T w=\Omega T w=ΩT
Disadvantages of impulse response unchanged method :
- The filter designed by impulse response invariable method will have the aliasing effect of frequency response
- Impulse response invariant method Only applicable to band limited analog filters ( For example, the attenuation characteristics are very good Low pass or bandpass filter ), And the faster the high-frequency attenuation , The smaller the aliasing effect ; And for qualcomm 、 Band stop filter , It is inconvenient to use this method for design .
3. bilinear transformation
3.1 Transformation steps
Bilinear transformation method first adopts the method of nonlinear frequency compression , Compress the frequency azimuth on this frequency axis to [ − π T , π T ] [-\frac{\pi}{T},\frac{\pi}{T}] [−Tπ,Tπ], Through z = e s T z=e^{sT} z=esT take s The plane is mapped to z Plane , such s Flat and z The plane establishes a one-to-one single value relationship , The multi value transformation is eliminated , Thus, the phenomenon of spectrum aliasing is eliminated , As shown in the figure below :
The expression of bilinear transformation relation is :
s = 2 T 1 − z − 1 1 + z − 1 s=\frac{2}{T} \frac{1-z^{-1}}{1+z^{-1}} s=T21+z−11−z−1
The steps of designing analog low-pass filter by bilinear transformation method are :
- First step
According to the given index of digital low-pass filter w p w_p wp, w s t w_{st} wst, δ p \delta_p δp, δ s \delta_s δs; - The second step
Through pre distortion , Determine simulation indicators : Ω = 2 T t a n ( w 2 ) \Omega=\frac{2}{T}tan(\frac{w}{2}) Ω=T2tan(2w)(T The value of is generally 2) - The third step
According to the index w p w_p wp, w s t w_{st} wst, δ p \delta_p δp, δ s \delta_s δs, Design analog filter , And get the system function H a ( s ) H_a(s) Ha(s) - Step four
According to bilinear transformation , The system function of the digital filter is
H ( z ) = H ( s ) ∣ s = 2 T 1 − z − 1 1 + z − 1 H(z)=H(s)|_{s=\frac{2}{T} \frac{1-z^{-1}}{1+z^{-1}}} H(z)=H(s)∣s=T21+z−11−z−1
It is worth noting that , During the transformation , Sampling interval T Usually take 2, Will be offset by calculation , Therefore, its value does not affect the design .
Now let's experience the design of digital filters by bilinear transformation through two examples :
- example 1


- example 2


3.2 Advantages and disadvantages of bilinear transformation
advantage :
It eliminates the aliasing effect of impulse response invariable method , Various types of filters can be designed .
shortcoming :
There are serious nonlinear frequency transformations :
Ω = 2 T t a n ( w 2 ) \Omega=\frac{2}{T}tan(\frac{w}{2}) Ω=T2tan(2w)
For filters with piecewise constants , After bilinear transformation , The filter with piecewise constant amplitude frequency characteristic is still obtained , But the position of the critical frequency point of each segment edge will produce distortion , This frequency distortion , It can be corrected by pre distortion of frequency .
Pre distortion refers to the distortion of the critical analog frequency realization , After transformation, it can be mapped to the required digital frequency ; The expression of pre distortion is :
Ω p = 2 T t a n ( w p 2 ) \Omega_p=\frac{2}{T}tan(\frac{w_p}{2}) Ωp=T2tan(2wp)
When designing digital filter with bilinear transformation method, pre distortion operation must be carried out .
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