Content deleted Content added
added Category:Digital signal processing using HotCat |
m →FIR (Finite Impulse Response) Linear filters: Text amended for correct syntax and improved clarity. |
||
(13 intermediate revisions by 8 users not shown) | |||
Line 1:
{{Cleanup rewrite|it is written like a tutorial|FIR transfer function|date=December 2020}}[[Filter (signal processing)#The transfer function|Transfer function filter]] utilizes the transfer function and the [[Convolution theorem]] to produce a filter. In this article, an example of such a filter using finite impulse response is discussed and an application of the filter into real world data is shown.
== FIR (Finite Impulse Response) Linear filters ==
In digital signal processing, an [[Finite impulse response|FIR filter]] is a time-continuous filter that is invariant with time. This means that the filter does not depend on the specific point of time, but rather depends on the time duration. The specification of this filter uses a [[Linear filter#FIR transfer functions|transfer function]]
== Mathematical model ==
Let the output function be <math>y(t)</math> and the input is <math>x(t)</math>. The convolution of the input with a transfer function <math>h(t)</math> provides a filtered output. The mathematical model of this type of filter is:
: <math>y(t) = \int_{0}^{T} x(t-\tau)\, h(\tau)\, d\tau</math>
h(<math>\tau</math>) is a transfer function of an impulse response to the input. The [[Convolution#Visual explanation|convolution]] allows the filter to only be activated when the input recorded a signal at the same time value. This filter returns the input values (x(t)) if k falls into the support region of function h. This is the reason why this filter is called finite response. If k is outside of the support region, the impulse response is zero which makes output zero. The central idea of this h(<math>\tau</math>) function can be thought of as a quotient of two functions
According to Huang (1981)<ref>Huang, T. S. (1981). Topics in applied physics: Two-Dimensional Digital Signal Processing I (3rd ed., Vol. 42, Topics in Applied Physics). Berlin: Springer.</ref> Using this mathematical
# [[#Window design method|Window design method]]
# Frequency Sampling method
Line 28:
:<math>h(t) = \begin{cases} 0, & \forall &-\infty &\le & t &\le 0 \\ e^{-t}, \quad & \forall &0 &\le & t &\le +\infty \end{cases}</math>
[[File:Single sided filter frequency response.jpg|Single sided filter frequency response]]
Line 45:
== FIR Transfer function Linear filter Application ==
Linear filter performs better when it is a double-sided filter. This requires the data to be known in advance which makes it a challenge for these filters to function well in situations where signals cannot be known ahead of time such as radio signal processing. However, this means that linear filters are extremely useful in filtering pre-loaded data. In addition, because of its non-recursive nature which preserves the phase angles of the input, linear filters are usually used in [[
== References ==
|