Filter Basics

Filters are devices (or algorithms) which change the spectrum of signals - their most prevalent action on signals is to boost, attenuate or completely block frequencies. In the tutorial about sinusoids, we saw that any signal can be seen a sum of sinusoids, each of which having its own frequency f, amplitude A and Phase '. In general terms, filters modify the amplitudes and phases of incoming sinusoids according to their frequency So, lets call that multiplication factor for the amplitude G (for gain) and lets call that phase shift _ (a Greek lowercase theta). Both, gain and phase-shift depend on the frequency of the incoming sinusoid, so both G and _ can be expressed functions of the frequency f: G = G (f) and _ = _(f)

Ideal Filters
The classical purpose of a filter is to let certain frequencies pass unchanged and to block others - hence the name filter.

Low Pass Filter

An idealized low pass-filter for example would pass all frequencies up to some cutoff-frequency fc and block all frequencies above that cutoff-frequency. The idealized low pass magnitude response GLP (f) would be therefore: GLP (f) =(1 for f < fc 0 for f > fc

LOW PASS

High Pass Filters

As we do not see any reason to intentionally introduce phase shift, we would want the phase response of the ideal filter to be identically zero for all frequencies: _(f) = 0 Ideal high pass filters,

High Passs

Band Pass Filters

Band pass filters should pass everything within some frequency interval between a lower cutoff frequency fl and an upper cutoff frequency fu.

Band Pass

Band Reject Filters

Band reject filters are the opposite of band pass filters- they block everything between fl and fh and let everything outside this interval pass unchanged. For band reject filters with a very narrow rejection interval, one also often uses the term notch-filter

Band Reject

Butterworth

Real Filters
Unfortunately (or maybe not), we do not live in a perfect world and these ideal frequency responses as requested above are not attainable in the real world. Real filters can only approximate these requirements and various filter design techniques exist to obtain different kinds of approximations. As it turns out from the mathematics, the squared magnitude responses of realizable filters are always ratios of even Polynomials (also called rational functions), that is functions of the form: G2 (f) =B2 (f)/A2 (f) Where B (f) and A (f) are both polynomials of their argument f

Passive Filters
Passive implementations of linear filters are based on combinations of resistors (R), inductors (L) and capacitors (C). These types are collectively known as passive filters, because they do not depend upon an external power supply and/or they do not contain active components such as transistors. Inductors block high-frequency signals and conduct low-frequency signals, while capacitors do the reverse. A filter in which the signal passes through an inductor, or in which a capacitor provides a path to ground, presents less attenuation to low-frequency signals than high-frequency signals and is therefore a low-pass filter. If the signal passes through a capacitor, or has a path to ground through an inductor, then the filter presents less attenuation to high-frequency signals than low-frequency signals and therefore is a high-pass filter

Active Filters
Active filters are implemented using a combination of passive and active(amplifying) components, and require an outside power source. Operational amplifiers are frequently used in active filter designs. These can have high Q factor, and can achieve resonance without the use of inductors. However, their upper frequency limit is limited by the bandwidth of the amplifiers used

Digital Filters

A general finite impulse response filter with n stages, each with an independent delay, di and amplification gain, ai. Digital signal processing allows the inexpensive construction of a wide variety of filters. The signal is sampled and an analog-to-digital converter turns the signal into a stream of numbers. A computer program running on a CPU or a specialized DSP(or less often running on a hardware implementation of the algorithm) calculates an output number stream. This output can be converted to a signal by passing it through a digital-to-analog converter. There are problems with noise introduced by the conversions, but these can be controlled and limited for many useful filters. Due to the sampling involved, the input signal must be of limited frequency content or aliasing will occur.

FIR FILTERS
FIR filters are digital filters with finite impulse response. They are also known as non-recursive digital filters as they do not have the feedback (a recursive part of a filter), even though recursive algorithms can be used for FIR filter realization.

Figure 2-1-1. Block diagrams of FIR and IIR filters

FIR filters can be designed using different methods, but most of them are based on ideal filter approximation. The objective is not to achieve ideal characteristics, as it is impossible anyway, but to achieve sufficiently good characteristics of a filter.

The transfer function of FIR filter approaches the ideal as the filter order increases, thus increasing the complexity and amount of time needed for processing input samples of a signal being filtered.

Need of Filters
A distributed filter, or micro-filter, is a small electronic component that fits between your phone line and a regular voice device, such as a phone, a fax, or any device with a regular modem such as a cable box, alarm system or digital TV. When DSL (ADSL) is provided over voice lines, all devices in the house except the DSL modem must be connected through filters.

Micro-filter

In-line filter

Wall mount filter