Editing Linear filter (section)

== Mathematics of filter design ==
{{Linear analog electronic filter}}
{{see also|Network synthesis}}
[[LTI system theory]] describes linear ''[[time-invariant]]'' (LTI) filters of all types. LTI filters can be completely described by their [[frequency response]] and [[phase response]], the specification of which uniquely defines their [[impulse response]], and ''vice versa''. From a mathematical viewpoint, continuous-time IIR LTI filters may be described in terms of linear [[differential equation]]s, and their impulse responses considered as [[Green's function]]s of the equation. Continuous-time LTI filters may also be described in terms of the [[Laplace transform]] of their impulse response, which allows all of the characteristics of the filter to be analyzed by considering the pattern of [[zeros and poles]] of their Laplace transform in the [[complex plane]].  Similarly, discrete-time LTI filters may be analyzed via the [[Z-transform]] of their impulse response.

Before the advent of computer filter synthesis tools, graphical tools such as [[Bode plot]]s and [[Nyquist plot]]s were extensively used as design tools. Even today, they are invaluable tools to understanding filter behavior.  Reference books<ref>A. Zverev, ''Handbook of Filter Synthesis'', John Wiley and Sons, 1967, {{ISBN|0-471-98680-1}}</ref> had extensive plots of frequency response, phase response, group delay, and impulse response for various types of filters, of various orders.  They also contained tables of values showing how to implement such filters as RLC ladders - very useful when amplifying elements were expensive compared to passive components.  Such a ladder can also be designed to have minimal sensitivity to component variation a property hard to evaluate without computer tools.

Many different analog filter designs have been developed, each trying to optimise some feature of the system response. For practical filters, a custom design is sometimes desirable, that can offer the best tradeoff between different design criteria, which may include component count and cost, as well as filter response characteristics.

These descriptions refer to the ''mathematical'' properties of the filter (that is, the frequency and phase response). These can be ''implemented'' as analog circuits (for instance, using a [[Sallen Key filter]] topology, a type of [[active filter]]), or as algorithms in [[digital signal processing]] systems.

Digital filters are much more flexible to synthesize and use than analog filters, where the constraints of the design permits their use. Notably, there is no need to consider component tolerances, and very high Q levels may be obtained.

FIR digital filters may be implemented by the direct [[convolution]] of the desired impulse response with the input signal.
They can easily be designed to give a [[matched filter]] for any arbitrary pulse shape.

IIR digital filters are often more difficult to design, due to problems including dynamic range issues, [[quantization noise]] and instability.
Typically digital IIR filters are designed as a series of [[digital biquad filter]]s.

All low-pass second-order continuous-time filters have a [[transfer function]] given by

: <math>H(s)=\frac{K \omega^{2}_{0}}{s^{2}+\frac{\omega_{0}}{Q}s+\omega^{2}_{0}}.</math>

All band-pass second-order continuous-time filters have a transfer function given by

: <math>H(s)=\frac{K \frac{\omega_{0}}{Q}s}{s^{2}+\frac{\omega_{0}}{Q}s+\omega^{2}_{0}}.</math>

where
* ''K'' is the gain (low-pass DC gain, or band-pass mid-band gain) (''K'' is 1 for passive filters)
* ''Q'' is the [[Q factor]]
* <math>\omega_{0}</math> is the center frequency
* <math>s=\sigma+j\omega</math> is the complex frequency