A buffer as the first stage of an active crossover/equalizer provides the necessary low source impedance to the following filter

networks. The buffer also provides a high impedance load to the preamplifier output circuit and the option of a highpass filter for dc blocking

Low Pass (First Order)

Let's build a simple filter. A first-order Butterworth low pass filter has the same magnitude and phase response characteristics as a passive RC filter with the exception of the fact that you can have gain in the passband. Take a look at Figure 2.71 and you'll see why this is the case. You'll notice that this circuit is a combination of two things: The first is a passive low pass filter (using capacitor C and resistor R). The second is a non-inverting op amp combination using the op amp and the three resistors.

Figure 2.71: First order low-pass Butterworth filter The characteristics of this filter can be calculated using the following equations. Alternately, if you know the characteristics of the filter you want, you can design the filter using the same equations.

(3.98)

(3.99)

Low Pass Filter Introduction

Basically, an electrical filter is a circuit that can be designed to modify, reshape or reject all unwanted frequencies of an electrical signal and accept or pass only those signals wanted by the circuits designer. In other words they "filter-out" unwanted signals and an ideal filter will separate and pass sinusoidal input signals based upon their frequency. In low frequency applications (up to 100kHz), passive filters are usually made from simple RC (ResistorCapacitor) networks while higher frequency filters (above 100kHz) are usually made from RLC (Resistor-Inductor-Capacitor) components. Passive filters are made up of passive components such as resistors, capacitors and inductors and have no amplifying elements (transistors, opamps, etc) so have no signal gain, therefore their output level is always less than the input. Filters are named according to the frequency of signals they allow to pass through them. There are Low-pass filters that allow only low frequency signals to pass, High-pass filters that allow only high frequency signals to pass through, and Band-pass filters that allow signals falling within a certain frequency range to pass through. Simple First-order passive filters (1st order) can be made by connecting together a single resistor and a single capacitor in series across an input signal, (Vin) with the output of the filter, (Vout) taken from the junction of these two components. Depending on which way around we connect the resistor and the capacitor with regards to the output signal determines the type of filter construction resulting in either a Low Pass Filter or a High Pass Filter. As the function of any filter is to allow signals of a given band of frequencies to pass unaltered while attenuating or weakening all others that are not wanted, we can define the amplitude response characteristics of an ideal filter by using an ideal frequency response curve of the four basic filter types as shown.

Ideal Filter Response Curves

Filters can be divided into two distinct types: active filters and passive filters. Active filters contain amplifying devices to increase signal strength while passive do not contain amplifying devices to strengthen the signal. As there are two passive components within a passive filter design the output signal has a smaller amplitude than its corresponding input signal, therefore passive RC filters attenuate the signal and have a gain of less than one, (unity). A Low Pass Filter can be a combination of capacitance, inductance or resistance intended to produce high attenuation above a specified frequency and little or no attenuation below that frequency. The frequency at which the transition occurs is called the "cutoff" frequency. The simplest low pass filters consist of a resistor and capacitor but more sophisticated low pass filters have a combination of series inductors and parallel capacitors. In this tutorial we will look at the simplest type, a passive two component RC low pass filter.

The Low Pass Filter

A simple passive Low Pass Filter or LPF, can be easily made by connecting together in series a single Resistor with a single Capacitor as shown below. In this type of filter arrangement the input signal (Vin) is applied to the series combination (both the Resistor and Capacitor together) but the output signal (Vout) is taken across the capacitor only. This type of filter is known generally as a "first-order filter" or "one-pole filter", why first-order or single-pole?, because it has only "one" reactive component in the circuit, the capacitor.

Low Pass Filter Circuit

As mentioned previously in the Capacitive Reactance tutorial, the reactance of a capacitor varies inversely with frequency, while the value of the resistor remains constant as the frequency changes. At low frequencies the capacitive reactance, (Xc) of the capacitor will be very large compared to the resistive value of the resistor, R and as a result the voltage across the capacitor, Vc will also be large while the voltage drop across the resistor, Vr will be much lower. At high frequencies the reverse is true with Vc being small and Vr being large. While the circuit above is that of an RC Low Pass Filter circuit, it can also be classed as a frequency variable potential divider circuit similar to the one we looked at in the Resistors tutorial. In that tutorial we used the following equation to calculate the output voltage for two single resistors connected in series.

We also know that the capacitive reactance of a capacitor in an AC circuit is given as:

Opposition to current flow in an AC circuit is called impedance, symbol Z and for a series circuit consisting of a single resistor in series with a single capacitor, the circuit impedance is calculated as:

Then by substituting our equation for impedance above into the resistive potential divider equation gives us:

So, by using the potential divider equation of two resistors in series and substituting for impedance we can calculate the output voltage of an RC Filter for any given frequency.

Example No1
A Low Pass Filter circuit consisting of a resistor of 4k7 in series with a capacitor of 47nF is connected across a 10v sinusoidal supply. Calculate the output voltage (Vout) at a frequency of 100Hz and again at frequency of 10,000Hz or 10kHz.
At a frequency of 100Hz.

At a frequency of 10kHz.

Frequency Response
We can see above, that as the frequency increases from 100Hz to 10kHz, the output voltage (Vout) decreases from 9.9v to 0.718v. By plotting the output voltage against the input frequency, the Frequency Response Curve or Bode Plot function of the low pass filter can be found, as shown below.

Frequency Response of a 1st-order Low Pass Filter

The Bode Plot shows the Frequency Response of the filter to be nearly flat for low frequencies and all of the input signal is passed directly to the output, resulting in a gain of nearly 1, called unity, until it reaches its Cut-off Frequency point ( c ). This is because the reactance of the capacitor is high at low frequencies and blocks any current flow through the capacitor. After this cut-off frequency point the response of the circuit decreases giving a slope of -20dB/ Decade or (-6dB/Octave) "roll-off" as signals above this frequency become greatly attenuated, until at very high frequencies the reactance of the capacitor becomes so low that it gives the effect of a short circuit condition on the output terminals resulting in zero output. For this type of Low Pass Filter circuit, all the frequencies below this cut-off, c point that are unaltered with little or no attenuation and are said to be in the filters Pass band zone. This pass band zone also represents the Bandwidth of the filter. Any signal frequencies above this point cut-off point are generally said to be in the filters Stop band zone and they will be greatly attenuated. This "Cut-off", "Corner" or "Breakpoint" frequency is defined as being the frequency point where the capacitive reactance and resistance are equal, R = Xc = 4k7. When this occurs the output signal is attenuated to 70.7% of the input signal value or -3dB (20 log (Vout/Vin)) of the input. Although R = Xc, the output is not half of the input signal. This is because it is equal to

the vector sum of the two and is therefore 0.707 of the input. As the filter contains a capacitor, the Phase Angle ( ) of the output signal LAGS behind that of the input and at the -3dB cut-off frequency ( c ) and is -45o out of phase. This is due to the time taken to charge the plates of the capacitor as the input voltage changes, resulting in the output voltage (the voltage across the capacitor) "lagging" behind that of the input signal. The higher the input frequency applied to the filter the more the capacitor lags and the circuit becomes more and more "out of phase". The cut-off frequency point and phase shift angle can be found by using the following equation:

Cut-off Frequency and Phase Shift

Then for our simple example of a "Low Pass Filter" circuit above, the cut-off frequency (c) is given as

The frequency response plot from Butterworth's 1930 paper.

The Butterworth filter is a type of signal processing filter designed to have as flat a frequency response as possible in the passband so that it is also termed a maximally flat magnitude filter. It was first described in 1930 by the British engineer Stephen Butterworth in his paper entitled "On the Theory of Filter Amplifiers".[1] Butterworth Filter Design
In the previous sections we looked at simple first-order type low and high pass filters that contain only a single resistor and a single reactive component (a capacitor) within their circuit design. In applications that use filters to shape the frequency spectrum of a signal such as in communications or control systems, the shape or width of the roll-off also called the "transition band", for a simple first-order filter may be too long or wide and so active filters designed with more than one "order" are required. These types of filters are commonly known as "High-order" or "Nth-order" filters.

The complexity or filter type is defined by the filters "order", and which is dependant upon the number of reactive components such as capacitors or inductors within its design. We also know that the rate of roll-off and therefore the width of the transition band, depends upon the order number of the filter and that for a simple first-order filter it has a standard roll-off rate of 20dB/decade or 6dB/octave. Then, for a filter that has an Nth number order, it will have a subsequent roll-off rate of 20N dB/decade or 6N dB/octave. So a first-order filter has a roll-off rate of 20dB/decade (6dB/octave), a second-order filter has a roll-off rate of 40dB/decade (12dB/octave), and a fourth-order filter has a roll-off rate of 80dB/decade (24dB/octave), etc, etc. High-order filters, such as third, fourth, and fifth-order are usually formed by cascading together single first-order and second-order filters. For example, two second-order low pass filters can be cascaded together to produce a fourth-order low pass filter, and so on. Although there is no limit to the order of the filter that can be formed, as the order increases so does its size and cost, also its accuracy declines.

Decades and Octaves

One final comment about Decades and Octaves. On the frequency scale, a Decade is a tenfold increase (multiply by 10) or tenfold decrease (divide by 10). For example, 2 to 20Hz represents one decade, whereas 50 to 5000Hz represents two decades (50 to 500Hz and then 500 to 5000Hz). An Octave is a doubling (multiply by 2) or halving (divide by 2) of the frequency scale. For example, 10 to 20Hz represents one octave, while 2 to 16Hz is three octaves (2 to 4, 4 to 8 and finally 8 to 16Hz) doubling the frequency each time. Either way Logarithmic scales are used extensively in the frequency domain to denote a frequency value when working with amplifiers and filters.

Logarithmic Frequency Scale

Since the frequency determining resistors are all equal, and as are the frequency determining capacitors, the cut-off or corner frequency ( C ) for either a first, second, third or even a fourthorder filter must also be equal and is found by using our now old familiar equation:

As with the first and second-order filters, the third and fourth-order high pass filters are formed by simply interchanging the positions of the frequency determining components (resistors and capacitors) in the equivalent low pass filter. High-order filters can be designed by following the procedures we saw previously in the Low Pass and High Pass filter tutorials. However, the overall gain of high-order filters is fixed because all the frequency determining components are equal.

Filter Approximations
So far we have looked at a low and high pass first-order filter circuits, their resultant frequency and phase responses. An ideal filter would give us specifications of maximum pass band gain and flatness, minimum stop band attenuation and also a very steep pass band to stop band roll-off (the transition band) and it is therefore apparent that a large number of network responses would satisfy these requirements. Not surprisingly then that there are a number of "approximation functions" in linear analogue filter design that use a mathematical approach to best approximate the transfer function we require for the filters design. Such designs are known as Elliptical, Butterworth, Chebyshev, Bessel, Cauer as well as many others. Of these five "classic" linear analogue filter approximation functions only the Butterworth Filter and especially the low pass Butterworth filter design will be considered here as its the most commonly used function.

Low Pass Butterworth Filter Design

The frequency response of the Butterworth Filter approximation function is also often referred to as "maximally flat" (no ripples) response because the pass band is designed to have a frequency response which is as flat as mathematically possible from 0Hz (DC) until the cut-off frequency at -3dB with no ripples. Higher frequencies beyond the cut-off point rolls-off down to zero in the stop band at 20dB/decade or 6dB/octave. This is because it has a "quality factor", "Q" of just 0.707. However, one main disadvantage of the Butterworth filter is that it achieves this pass band flatness at the expense of a wide transition band as the filter changes from the pass band to the stop band. It also has poor phase characteristics as well. The ideal frequency response, referred to as a "brick wall" filter, and the standard Butterworth approximations, for different filter orders are given below.

Ideal Frequency Response for a Butterworth Filter

Note that the higher the Butterworth filter order, the higher the number of cascaded stages there are within the filter design, and the closer the filter becomes to the ideal "brick wall" response. However, in practice this "ideal" frequency response is unattainable as it produces excessive passband ripple.

Where the generalised equation representing a "nth" Order Butterworth filter, the frequency response is given as:

Where: n represents the filter order, Omega is equal to 2 and Epsilon is the maximum pass band gain, (Amax). If Amax is defined at a frequency equal to the cut-off -3dB corner point (c), will then be equal to one and therefore 2 will also be one. However, if you now wish to define Amax at a different voltage gain value, for example 1dB, or 1.1220 (1dB = 20logAmax) then the new value of epsilon, is found by:

Where: H0 = the Maximum Pass band Gain, Amax. H1 = the Minimum Pass band Gain.

Transpose the equation to give:

The Frequency Response of a filter can be defined mathematically by its Transfer Function with the standard Voltage Transfer Function H(j) written as:

Where: Vout = the output signal voltage. Vin = the input signal voltage. j = to the square root of -1 (-1) = the radian frequency (2)

Note: (j) can also be written as (s) to denote the S-domain. and the resultant transfer function for a second-order low pass filter is given as:

Normalised Low Pass Butterworth Filter Polynomials

Contents
[hide]

1 Original paper 2 Overview 3 Example 4 Transfer function o 4.1 Normalized Butterworth polynomials o 4.2 Maximal flatness o 4.3 High-frequency roll-off 5 Filter design o 5.1 Cauer topology o 5.2 SallenKey topology o 5.3 Digital implementation 6 Comparison with other linear filters 7 References

[edit] Original paper

Linear analog electronic filters Network synthesis filters[hide]

Butterworth filter Chebyshev filter Elliptic (Cauer) filter Bessel filter Gaussian filter Optimum "L" (Legendre) filter Linkwitz-Riley filter Image impedance filters[show] Simple filters[show]
edit

Butterworth had a reputation for solving "impossible" mathematical problems. At the time filter design was largely by trial and error because of its mathematical complexity. His paper was far ahead of its time: the filter was not in common use for over 30 years after its publication. Butterworth stated that
An ideal electrical filter should not only completely reject the unwanted frequencies but should also have uniform sensitivity for the wanted frequencies.

At the time, filters generated substantial ripple in the passband, and the choice of component values was highly interactive. Butterworth showed that a low pass filter could be designed whose cutoff frequency was normalized to 1 radian per second and whose frequency response (gain) was

where is the angular frequency in radians per second and n is the number of reactive elements (poles) in the filter. If = 1, the amplitude response of this type of filter in the passband is 1/2 0.707, which is half power or 3 dB. Butterworth only dealt with filters with an even number of poles in his paper. He may have been unaware that such filters could be designed with an odd number of poles. He built his higher order filters from 2-pole filters separated by vacuum tube amplifiers. His plot of the frequency response of 2, 4, 6, 8, and 10 pole filters is shown as A, B, C, D, and E in his original graph. Butterworth solved the equations for two- and four-pole filters, showing how the latter could be cascaded when separated by vacuum tube amplifiers and so enabling the construction of higherorder filters despite inductor losses. In 1930 low-loss core materials such as molypermalloy had not been discovered and air-cored audio inductors were rather lossy. Butterworth discovered that it was possible to adjust the component values of the filter to compensate for the winding resistance of the inductors. He used coil forms of 1.25 diameter and 3 length with plug in terminals. Associated capacitors and resistors were contained inside the wound coil form. The coil formed part of the plate load resistor. Two poles were used per vacuum tube and RC coupling was used to the grid of the following tube. Butterworth also showed that his basic low-pass filter could be modified to give low-pass, highpass, band-pass and band-stop functionality.

[edit] Overview

The Bode plot of a first-order Butterworth low-pass filter

The frequency response of the Butterworth filter is maximally flat (has no ripples) in the passband and rolls off towards zero in the stopband.[2] When viewed on a logarithmic Bode plot the response slopes off linearly towards negative infinity. A first-order filter's response rolls off at 6 dB per octave (20 dB per decade) (all first-order lowpass filters have the same normalized frequency response). A second-order filter decreases at 12 dB per octave, a third-order at 18 dB and so on. Butterworth filters have a monotonically changing magnitude function with , unlike other filter types that have non-monotonic ripple in the passband and/or the stopband. Compared with a Chebyshev Type I/Type II filter or an elliptic filter, the Butterworth filter has a slower roll-off, and thus will require a higher order to implement a particular stopband specification, but Butterworth filters have a more linear phase response in the pass-band than Chebyshev Type I/Type II and elliptic filters can achieve.

[edit] Example

A third-order low-pass filter (Cauer topology). The filter becomes a Butterworth filter with cutoff frequency c=1 when (for example) C2=4/3 farad, R4=1 ohm, L1=3/2 henry and L3=1/2 henry.

A simple example of a Butterworth filter is the third-order low-pass design shown in the figure on the right, with C2 = 4/3 F, R4 = 1 , L1 = 3/2 H, and L3 = 1/2 H. Taking the impedance of the capacitors C to be 1/Cs and the impedance of the inductors L to be Ls, where s = + j is the complex frequency, the circuit equations yield the transfer function for this device:

The magnitude of the frequency response (gain) G() is given by

and the phase is given by

Gain and group delay of the third-order Butterworth filter with c=1

The group delay is defined as the derivative of the phase with respect to angular frequency and is a measure of the distortion in the signal introduced by phase differences for different frequencies. The gain and the delay for this filter are plotted in the graph on the left. It can be seen that there are no ripples in the gain curve in either the passband or the stop band. The log of the absolute value of the transfer function H(s) is plotted in complex frequency space in the second graph on the right. The function is defined by the three poles in the left half of the complex frequency plane.

Log density plot of the transfer function H(s) in complex frequency space for the third-order Butterworth filter with c=1. The three poles lie on a circle of unit radius in the left half-plane.

These are arranged on a circle of radius unity, symmetrical about the real s axis. The gain function will have three more poles on the right half plane to complete the circle.

By replacing each inductor with a capacitor and each capacitor with an inductor, a high-pass Butterworth filter is obtained. A band-pass Butterworth filter is obtained by placing a capacitor in series with each inductor and an inductor in parallel with each capacitor to form resonant circuits. The value of each new component must be selected to resonate with the old component at the frequency of interest. A band-stop Butterworth filter is obtained by placing a capacitor in parallel with each inductor and an inductor in series with each capacitor to form resonant circuits. The value of each new component must be selected to resonate with the old component at the frequency to be rejected.

[edit] Transfer function

Plot of the gain of Butterworth low-pass filters of orders 1 through 5, with cutoff frequency 0 = 1. Note that the slope is 20n dB/decade where n is the filter order.

Like all filters, the typical prototype is the low-pass filter, which can be modified into a highpass filter, or placed in series with others to form band-pass and band-stop filters, and higher order versions of these. The gain G() of an n-order Butterworth low pass filter is given in terms of the transfer function H(s) as

where

n = order of filter

c = cutoff frequency (approximately the -3dB frequency) G0 is the DC gain (gain at zero frequency)

It can be seen that as n approaches infinity, the gain becomes a rectangle function and frequencies below c will be passed with gain G0, while frequencies above c will be suppressed. For smaller values of n, the cutoff will be less sharp. We wish to determine the transfer function H(s) where s = + j (from Laplace transform). Since H(s)H(-s) evaluated at s = j is simply equal to |H(j)|2, it follows that

The poles of this expression occur on a circle of radius c at equally spaced points. The transfer function itself will be specified by just the poles in the negative real half-plane of s. The k-th pole is specified by

and hence;

The transfer function may be written in terms of these poles as

The denominator is a Butterworth polynomial in s.

[edit] Normalized Butterworth polynomials

The Butterworth polynomials may be written in complex form as above, but are usually written with real coefficients by multiplying pole pairs which are complex conjugates, such as s1 and sn. The polynomials are normalized by setting c = 1. The normalized Butterworth polynomials then have the general form

for n even

for n odd

To four decimal places, they are

n 1 2 3 4 Factors of Polynomial Bn(s) (s + 1) s2 + 1.4142s + 1 (s + 1)(s2 + s + 1) (s2 + 0.7654s + 1)(s2 + 1.8478s + 1)

5 (s + 1)(s2 + 0.6180s + 1)(s2 + 1.6180s + 1) 6 (s2 + 0.5176s + 1)(s2 + 1.41