MATH 414 Lecture 19

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Linear Time-Invariant Filters

Recall notation fa(t)=f(t−a), which is f shifted to the right by a units.

A filter is called time-invariant if and only if L(fa)=(L(f))a

At the end of last lecture, we discused how if L(t)=f*h for fixed h, then L is linear and time-invariant.

Theorem. L is a linear time-invariant filter if and only if L(f)=f*h, where h is the impulse response function.

Proof. We've already shown that (⇒) is true, this will only show (⇐).

Suppose L is linear and time-invariant. Let's apply L to eiλt:

L(eiλt)=hλ(t)

Applying the filter on a shifted function gives

L(eiλ(t−a))=hλ(t−a)

Observe that eiλ(t−a)=e−iλa⋅eiλt

The first term is just a scalar (with respect to λ, but not with respect to t). Therefore

hλ(t−a)=L(e−iλa⋅eiλt)=e−iλaL(eiλt)=e−iλahλ(t)

holds for all a and for all t; so we can set a=t.

hλ(t−t)=hλ(0)=e−iλthλ(t)hλ(t)=eiλthλ(0)L(eiλt)=eiλthλ(0)

Let's apply L to the inverse fourier transform of a function:

L(f)=L(12π∫−∞∞f^e+iλtdλ)=12π∫−∞∞f^L(eiλt)dλ=12π∫−∞∞f^hλ(0)eiλtdλ

Let's define h^(λ):=12πhλ(0). Then

L(f)=2π∫−∞∞12πf^(λ)h^(λ)eiλtdλ=ℱ−1(f^(λ)h^(λ))2π=f*h

quod erat demonstrandum


Impulse Response

fδ(t)=θ(x+δ)12δ−θ(x−δ)12δ

Then limδ→0+fδ(t) is a spike such that f(t)={∞t=00otherwise

Page Under Construction
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L(fδ(t))=∫−∞∞f(τ)h(τ−t)dτ=…


Constructing Filters

L(f)=f*h(L(f))=2πf^(λ)h^(λ)

Low pass filter
Reduces amplitudes of high magnitude frequencies
High pass filter
Reverse of low pass filter
Notch filter
Removes selection of frequencies. [1]

"Ideal" Low Pass Filter

Take h^(λ)={12π|λ|≤λ00otherwise

Taking the convolution L(f)=f*h, where

h=ℱ−1(h^(λ))=12π∫−∞∞h^(λ)e+iλtdλ=12π∫−λ0λ0eiλtdλ=sin⁡(λ0t)πt

L(f)=∫−∞∞f(τ)sin⁡(λ0(t−τ))π(t−τ)dτ


Put in f(t)=θ(t)−θ(t−t0). Then

L(f)=∫0bsin⁡(λ0t)πt

Perform u-substitution with u=λ0(t−τ)

L(f)=∫λ0tλ0(t−t0)sin⁡uπuλ0−duλ0=∫λ0(t−t0)λ0tsin⁡uπudu=1π(Si(λ0t)−Si(λ0(t−t0)))

See Definition of Si(x) below.

Butterworth Filter

h(t)={ae−αtt≥00t<0

h^(λ)=12π∫−∞∞h(t)e−iλtdt=A2π∫0∞e−αt−iλtdt=A2π(α+iλ)


L^(f)=2π2πAf^(λ)α+iλ=Aa+iλf^(λ)


Now |L^(f)|2=A2α2+λ2|f^(λ)|2. As λ gets big, then the frequencies returned by f^ are reduced. Therefore this is essentially a low-pass filter

The Sine Integral

Si(x):=∫0xsin⁡uudu

Footnotes

  1. ↑ That's what happens to mens' ears when they get married