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Bilinear Transformation
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When the impulse invariance method is used aliasing is encountered.
To reduce the distortion caused by aliasing one can tighten the specifications on the digital filter. Aliasing occurs because points in the axis separated by 2/T are mapped into the same digital frequency . In the bilinear transformation method, there is a one-to-one correspondence between and . Therefore while transforming the analog filter to a digital filter aliasing is not encountered. Since is limited to [- , ] range but varies from - to , it becomes clear that must be compressed when it is mapped to . In other words bilinear transformation is a non-linear transformation.
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Observation: When <0 , 𝑧 <1. So the left-plane poles of Ha(s) will be mapped into poles within the unit circle in the z-plane. In other words, a stable prototype analog filter will lead to a stable digital filter. When =0, 𝑧 =1. In other words , z lies on the unit circle and can be written as: 𝑧= 𝑒 𝑗𝜔 where the digital frequency is 𝜔=𝜃=2𝑎𝑟𝑐𝑡𝑎𝑛 Ω𝑇 2 Alternatively we can express the analog frequency as: Ω= 2 𝑇 𝑡𝑎𝑛 𝜔 2
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Figure below represents the relationship between and
Figure below represents the relationship between and . Clearly compression occurs in the mapping process.
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Example: Design a digital lowpass filter with the following specifications using the bilinear transformation method and a Butterworth prototype filter.
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Solution: For simplicity we assume T =1 . Therefore Ω=2𝑡𝑎𝑛 𝜔 This means that the digital passband and stopband frequencies below Are mapped into the analog passband and stopband frequencies: Therefore the specifications of the analog filter becomes:
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Next we calculate the Rp and Rs values which denote the passband and
Stopband ripples: 𝑅 𝑝 =20 𝑙𝑜𝑔 =−1 𝑅 𝑠 =20 𝑙𝑜𝑔 =−15 And we know that Ω 𝑝 = and Ω 𝑠 =1.0191 We can then use the MATLAB function buttord: [𝑁,Ω 𝑐 ]=𝑏𝑢𝑡𝑡𝑜𝑟𝑑 Ω 𝑝 , Ω 𝑠 , 𝑅 𝑝 , 𝑅 𝑠 MATLAB will return the order of the Butterworth filter as N and the 3-dB cutoff frequency Ω 𝑐 For this problem N = 6 Ω 𝑐 =
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To complete the design of the Butterworth filter we then use MATLAB function
butter to obtain the zeros, poles and gains of this filter: 𝑍,𝑃,𝐾 =𝑏𝑢𝑡𝑡𝑒𝑟 𝑁, Ω 𝑐 ,′𝑠′ This will return the zeros of Ha(s) in array Z and poles of Ha(s) in array P. The filter gain which is same as Ω 𝑁 𝑐 , is in the variable K. For this example there are no zeros. The gain is and the poles are as follows:
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Since the poles are available we can use them to write the transfer
function of the Butterworth filter as: 𝐻 𝑎 𝑠 = 𝐾 𝑘=1 6 𝑠− 𝑝 𝑘 = 𝑠 𝑠 𝑠 𝑠 𝑠 𝑠 Finally the last step in the design exercise is to map the above transfer function into a Digital filter using the Bilinear Transformation . That is : 𝐻 𝑧 = 𝐻 𝑎 2 1− 𝑧 −1 1+ 𝑧 −1 We can do this by using MATLAB also: [ 𝑍 𝑑 , 𝑃 𝑑 , 𝐾 𝑑 ]=bilinear 𝑍,𝑃,𝐾,1/𝑇 ; where T is the sampling frequency
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For this example : Kd = Zd = [ -1 -1 ] and Pd = [ ]
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Hence the transfer function of the digital filter can be written as:
𝐻 𝑧 = 𝐾 𝑑 𝑘=1 6 1− 𝑧 𝑘 𝑧 −1 𝑘=1 6 1− 𝑝 𝑘 𝑧 −1 = 𝑧 − − 𝑧 − 𝑧 −2 1− 𝑧 − 𝑧 −2 1− 𝑧 − 𝑧 −2
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Example #2: Design a lowpass filter with Cuttoff frequency c = /5 rad/sample Transition bandwidth of = 0.2 radians/sample = 0.1 cycles/sample With a ripple of = 0.1 Use a Butterworth analog filter Set T = 1.
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Map the digital to analog frequencies using
= 2 𝑇 𝑡𝑎𝑛 𝜔 2 𝜔 𝑑1 = 𝜋 5 −0.1𝜋 , 𝜔 𝑑2 = 𝜋 𝜋 Ω 1 = 𝑟𝑎𝑑/𝑠𝑒𝑐 , Ω 2 = 𝑟𝑎𝑑/𝑠𝑒𝑐 We now need to solve for the filter order and 3-dB cutoff frequency 𝑁= 𝑙𝑜𝑔 𝐻 𝑎 Ω 2 −2 −1 / 𝐻 𝑎 Ω 1 −2 −1 𝑙𝑜𝑔 Ω 2 − Ω 1 Ω 𝑐 = Ω 𝐻 𝑎 Ω 2 −2 −1 1 (2𝑁)
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For this example the order N = 3
and Ω 𝑐 = −2 −1 1 (2∗3) = Recall that 𝐻 𝑎 Ω 2 = =0.1 𝐻 𝑎 Ω 1 =1-
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Knowing N and Ω 𝑐 we can write the transfer function of the analog filter below
after its poles are computed: Ha(s) Ha(-s) = 𝑠 𝑗 Ω 𝑐 2𝑁 𝑝 𝑘 = Ω 𝑐 𝑒 𝑗 𝜋 2 + 𝜋 2𝑁 + 2𝜋𝑘 2𝑁 , k =0,1,……, (2N-1) For this example the poles are
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After writing the analog function Ha(s) we can then transform it to discrete-time
filter using MATLAB as before in Ex#1 𝐻 𝑑 𝑧 = Ω 𝑐 𝑧 − ∗ 1− 𝑧 −1 − 𝑝 1 1− 𝑧 −1 − 𝑝 −𝑧 −1 − 𝑝 3
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