Filters Tutorial
Filters Tutorial
Filters Tutorial
Capacitive Reactance
Capacitive Reactance is the complex impedance of a capacitor who’s value changes with
respect to the applied frequency
In the RC Network tutorial we saw that when a DC voltage is applied to a
capacitor, the capacitor itself draws a charging current from the supply and
charges up to a value equal to the applied voltage.
Likewise, when the supply voltage is reduced the charge stored in the capacitor
also reduces and the capacitor discharges. But in an AC circuit in which the
applied voltage signal is continually changing from a positive to a negative
polarity at a rate determined by the frequency of the supply, as in the case of a
sine wave voltage, for example, the capacitor is either being charged or
discharged on a continuous basis at a rate determined by the supply
frequency.
As the capacitor charges or discharges, a current flows through it which is
restricted by the internal impedance of the capacitor. This internal impedance is
commonly known as Capacitive Reactance and is given the symbol XC in
Ohms.
Unlike resistance which has a fixed value, for example, 100Ω, 1kΩ, 10kΩ etc,
(this is because resistance obeys Ohms Law), Capacitive Reactance varies
with the applied frequency so any variation in supply frequency will have a big
effect on the capacitor’s, “capacitive reactance” value.
As the frequency applied to the capacitor increases, its effect is to decrease its
reactance (measured in ohms). Likewise as the frequency across the capacitor
decreases its reactance value increases. This variation is called the
capacitor’s complex impedance.
Complex impedance exists because the electrons in the form of an electrical
charge on the capacitor plates, appear to pass from one plate to the other more
rapidly with respect to the varying frequency.
As the frequency increases, the capacitor passes more charge across the
plates in a given time resulting in a greater current flow through the capacitor
appearing as if the internal impedance of the capacitor has decreased.
Therefore, a capacitor connected to a circuit that changes over a given range
of frequencies can be said to be “Frequency Dependant”.
Capacitive Reactance has the electrical symbol “XC” and has units measured
in Ohms the same as resistance, ( R ). It is calculated using the following
formula:
Capacitive Reactance
Capacitive Reactance
Formula
Where:
Xc = Capacitive Reactance in Ohms, (Ω)
π (pi) = 3.142 (decimal) or as 22÷7 (fraction)
ƒ = Frequency in Hertz, (Hz)
C = Capacitance in Farads, (F)
By re-arranging the reactance formula above, we can also find at what
frequency a capacitor will have a particular capacitive reactance ( XC ) value.
Or we can find the value of the capacitor in Farads by knowing the applied
frequency and its reactance value at that frequency.
We can see from the above examples that a capacitor when connected to a
variable frequency supply, acts a bit like a “frequency controlled variable
resistor” as its reactance (X) is directly proportional to frequency. At very low
frequencies, such as 1Hz our 220nF capacitor has a high capacitive reactance
value of approx 723.3KΩ (giving the effect of an open circuit).
At very high frequencies such as 1Mhz the capacitor has a low capacitive
reactance value of just 0.72Ω (giving the effect of a short circuit). So at zero
frequency or steady state DC our 220nF capacitor has infinite reactance
looking more like an “open-circuit” between the plates and blocking any flow of
current through it.
We now know that a capacitor’s reactance, Xc (its complex impedance) value
changes with respect to the applied frequency. If we now changed
resistor R2 above for a capacitor, the voltage drop across the two components
would change as the frequency changed because the reactance of the
capacitor affects its impedance.
The impedance of resistor R1 does not change with frequency. Resistors are of
fixed values and are unaffected by frequency change. Then the voltage across
resistor R1 and therefore the output voltage is determined by the capacitive
reactance of the capacitor at a given frequency. This then results in a
frequency-dependent RC voltage divider circuit. With this idea in mind,
passive Low Pass Filters and High Pass Filters can be constructed by
replacing one of the voltage divider resistors with a suitable capacitor as
shown.
The property of Capacitive Reactance, makes the capacitor ideal for use in
AC filter circuits or in DC power supply smoothing circuits to reduce the effects
of any unwanted Ripple Voltage as the capacitor applies an short circuit signal
path to any unwanted frequency signals on the output terminals.
It is important to remember these two conditions and in our next tutorial about
the Passive Low Pass Filter, we will look at the use of Capacitive
Reactance to block any unwanted high frequency signals while allowing only
low frequency signals to pass.
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 generally constructed using
simple RC (Resistor-Capacitor) networks, while higher frequency filters (above
100kHz) are usually made from RLC (Resistor-Inductor-Capacitor)
components.
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 “cut-off” or “corner” 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.
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.
This means that the voltage potential, VC across the capacitor will be much
larger than the voltage drop, VR developed across the resistor. At high
frequencies the reverse is true with VC being small and VR being large due to
the change in the capacitive reactance value.
While the circuit above is that of an RC Low Pass Filter circuit, it can also be
thought of as a frequency dependant 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.
Frequency Response
We can see from the results above, that as the frequency applied to the RC
network increases from 100Hz to 10kHz, the voltage dropped across the
capacitor and therefore the output voltage ( VOUT ) from the circuit decreases
from 9.9v to 0.718v.
By plotting the networks output voltage against different values of input
frequency, the Frequency Response Curve or Bode Plot function of the low
pass filter circuit can be found, as shown below.
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 ( ƒc ). 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 to zero
at a slope of -20dB/ Decade or (-6dB/Octave) “roll-off”. Note that the angle of
the slope, this -20dB/ Decade roll-off will always be the same for any RC
combination.
Any high frequency signals applied to the low pass filter circuit above this cut-
off frequency point will become greatly attenuated, that is they rapidly
decrease. This happens because 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.
Then by carefully selecting the correct resistor-capacitor combination, we can
create a RC circuit that allows a range of frequencies below a certain value to
pass through the circuit unaffected while any frequencies applied to the circuit
above this cut-off point to be attenuated, creating what is commonly called
a Low Pass Filter.
For this type of “Low Pass Filter” circuit, all the frequencies below this cut-
off, ƒc 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 = 4k7Ω. 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 ( Φ ) of the output
signal LAGS behind that of the input and at the -3dB cut-off frequency ( ƒc ) 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:
Then for our simple example of a “Low Pass Filter” circuit above, the cut-off
frequency (ƒc) is given as 720Hz with an output voltage of 70.7% of the input
voltage value and a phase shift angle of -45o.
Second-order Low Pass Filter
Thus far we have seen that simple first-order RC low pass filters can be made
by connecting a single resistor in series with a single capacitor. This single-pole
arrangement gives us a roll-off slope of -20dB/decade attenuation of
frequencies above the cut-off point at ƒ-3dB . However, sometimes in filter circuits
this -20dB/decade (-6dB/octave) angle of the slope may not be enough to
remove an unwanted signal then two stages of filtering can be used as shown.
The above circuit uses two passive first-order low pass filters connected or
“cascaded” together to form a second-order or two-pole filter network.
Therefore we can see that a first-order low pass filter can be converted into a
second-order type by simply adding an additional RC network to it and the
more RC stages we add the higher becomes the order of the filter.
If a number ( n ) of such RC stages are cascaded together, the resulting RC
filter circuit would be known as an “nth-order” filter with a roll-off slope of “n x
-20dB/decade”.
So for example, a second-order filter would have a slope of -40dB/decade (-
12dB/octave), a fourth-order filter would have a slope of -80dB/decade (-
24dB/octave) and so on. This means that, as the order of the filter is increased,
the roll-off slope becomes steeper and the actual stop band response of the
filter approaches its ideal stop band characteristics.
Second-order filters are important and widely used in filter designs because
when combined with first-order filters any higher-order nth-value filters can be
designed using them. For example, a third order low-pass filter is formed by
connecting in series or cascading together a first and a second-order low pass
filter.
But there is a downside too cascading together RC filter stages. Although there
is no limit to the order of the filter that can be formed, as the order increases,
the gain and accuracy of the final filter declines.
When identical RC filter stages are cascaded together, the output gain at the
required cut-off frequency ( ƒc ) is reduced (attenuated) by an amount in
relation to the number of filter stages used as the roll-off slope increases. We
can define the amount of attenuation at the selected cut-off frequency using the
following formula.
In reality as the filter stage and therefore its roll-off slope increases, the low
pass filters -3dB corner frequency point and therefore its pass band frequency
changes from its original calculated value above by an amount determined by
the following equation.
where ƒc is the calculated cut-off frequency, n is the filter order and ƒ-3dB is the
new -3dB pass band frequency as a result in the increase of the filters order.
Then the frequency response (bode plot) for a second-order low pass filter
assuming the same -3dB cut-off point would look like:
Applications of passive Low Pass Filters are in audio amplifiers and speaker
systems to direct the lower frequency bass signals to the larger bass speakers
or to reduce any high frequency noise or “hiss” type distortion. When used like
this in audio applications the low pass filter is sometimes called a “high-cut”, or
“treble cut” filter.
If we were to reverse the positions of the resistor and capacitor in the circuit so
that the output voltage is now taken from across the resistor, we would have a
circuit that produces an output frequency response curve similar to that of a
High Pass Filter, and this is discussed in the next tutorial.
Time Constant
Until now we have been interested in the frequency response of a low pass
filter when subjected to sinusoidal waveform. We have also seen that the filters
cut-off frequency ( ƒc ) is the product of the resistance ( R ) and the
capacitance ( C ) in the circuit with respect to some specified frequency point
and that by altering any one of the two components alters this cut-off frequency
point by either increasing it or decreasing it.
We also know that the phase shift of the circuit lags behind that of the input
signal due to the time required to charge and then discharge the capacitor as
the sine wave changes. This combination of R and C produces a charging and
discharging effect on the capacitor known as its Time Constant ( τ ) of the
circuit as seen in the RC Circuit tutorials giving the filter a response in the time
domain.
The time constant, tau ( τ ), is related to the cut-off frequency ƒc as:
or expressed in terms of the cut-off frequency, ƒc as:
The output voltage, VOUT depends upon the time constant and the frequency of
the input signal. With a sinusoidal signal that changes smoothly over time, the
circuit behaves as a simple 1st order low pass filter as we have seen above.
But what if we were to change the input signal to that of a “square wave”
shaped “ON/OFF” type signal that has an almost vertical step input, what would
happen to our filter circuit now. The output response of the circuit would
change dramatically and produce another type of circuit known commonly as
an Integrator.
The RC Integrator
The Integrator is basically a low pass filter circuit operating in the time domain
that converts a square wave “step” response input signal into a triangular
shaped waveform output as the capacitor charges and discharges.
A Triangular waveform consists of alternate but equal, positive and negative
ramps.
As seen below, if the RC time constant is long compared to the time period of
the input waveform the resultant output waveform will be triangular in shape
and the higher the input frequency the lower will be the output amplitude
compared to that of the input.
This then makes this type of circuit ideal for converting one type of electronic
signal to another for use in wave-generating or wave-shaping circuits.
Where as the low pass filter only allowed signals to pass below its cut-off
frequency point, ƒc, the passive high pass filter circuit as its name implies, only
passes signals above the selected cut-off point, ƒc eliminating any low
frequency signals from the waveform. Consider the circuit below.
In this circuit arrangement, the reactance of the capacitor is very high at low
frequencies so the capacitor acts like an open circuit and blocks any input
signals at VIN until the cut-off frequency point ( ƒC ) is reached. Above this cut-off
frequency point the reactance of the capacitor has reduced sufficiently as to
now act more like a short circuit allowing all of the input signal to pass directly
to the output as shown below in the filters response curve.
Frequency Response of a 1st Order High Pass Filter
The Bode Plot or Frequency Response Curve above for a passive high pass
filter is the exact opposite to that of a low pass filter. Here the signal is
attenuated or damped at low frequencies with the output increasing at
+20dB/Decade (6dB/Octave) until the frequency reaches the cut-off point ( ƒc )
where again R = Xc. It has a response curve that extends down from infinity to
the cut-off frequency, where the output voltage amplitude is 1/√2 = 70.7% of
the input signal value or -3dB (20 log (Vout/Vin)) of the input value.
Also we can see that the phase angle ( Φ ) of the output signal LEADS that of
the input and is equal to +45o at frequency ƒc. The frequency response curve
for this filter implies that the filter can pass all signals out to infinity. However in
practice, the filter response does not extend to infinity but is limited by the
electrical characteristics of the components used.
The cut-off frequency point for a first order high pass filter can be found using
the same equation as that of the low pass filter, but the equation for the phase
shift is modified slightly to account for the positive phase angle as shown
below.
The above circuit uses two first-order filters connected or cascaded together to
form a second-order or two-pole high pass network. Then a first-order filter
stage can be converted into a second-order type by simply using an
additional RC network, the same as for the 2nd-order low pass filter. The
resulting second-order high pass filter circuit will have a slope of 40dB/decade
(12dB/octave).
As with the low pass filter, the cut-off frequency, ƒc is determined by both the
resistors and capacitors as follows.
Each cycle of the square wave input waveform produces two spikes at the
output, one positive and one negative and whose amplitude is equal to that of
the input. The rate of decay of the spikes depends upon the time constant,
( RC ) value of both components, ( t = R x C ) and the value of the input
frequency. The output pulses resemble more and more the shape of the input
signal as the frequency increases.
Band Pass Filters can be used to isolate or filter out certain frequencies that lie
within a particular band or range of frequencies. The cut-off frequency or ƒc
point in a simple RC passive filter can be accurately controlled using just a
single resistor in series with a non-polarized capacitor, and depending upon
which way around they are connected, we have seen that either a Low Pass or
a High Pass filter is obtained.
One simple use for these types of passive filters is in audio amplifier
applications or circuits such as in loudspeaker crossover filters or pre-amplifier
tone controls. Sometimes it is necessary to only pass a certain range of
frequencies that do not begin at 0Hz, (DC) or end at some upper high
frequency point but are within a certain range or band of frequencies, either
narrow or wide.
By connecting or “cascading” together a single Low Pass Filter circuit with
a High Pass Filter circuit, we can produce another type of passive RC filter
that passes a selected range or “band” of frequencies that can be either narrow
or wide while attenuating all those outside of this range. This new type of
passive filter arrangement produces a frequency selective filter known
commonly as a Band Pass Filter or BPF for short.
Then clearly, the width of the pass band of the filter can be controlled by the
positioning of the two cut-off frequency points of the two filters.
Then, the values of R1 and C1 required for the high pass stage to give a cut-off
frequency of 1.0kHz are: R1 = 10kΩ and to the nearest preferred value, C1 =
15nF.
Then, the values of R2 and C2 required for the low pass stage to give a cut-off
frequency of 30kHz are, R = 10kΩ and C = 530pF. However, the nearest
preferred value of the calculated capacitor value of 530pF is 560pF, so this is
used instead.
With the values of both the resistances R1 and R2 given as 10kΩ, and the two
values of the capacitors C1 and C2 found for both the high pass and low pass
filters as 15nF and 560pF respectively, then the circuit for our simple
passive Band Pass Filter is given as.
In the RC Passive Filter tutorials, we saw how a basic first-order filter circuits,
such as the low pass and the high pass filters can be made using just a single
resistor in series with a non-polarized capacitor connected across a sinusoidal
input signal.
We also noticed that the main disadvantage of passive filters is that the
amplitude of the output signal is less than that of the input signal, ie, the gain is
never greater than unity and that the load impedance affects the filters
characteristics.
With passive filter circuits containing multiple stages, this loss in signal
amplitude called “Attenuation” can become quiet severe. One way of restoring
or controlling this loss of signal is by using amplification through the use
of Active Filters.
As their name implies, Active Filters contain active components such as
operational amplifiers, transistors or FET’s within their circuit design. They draw
their power from an external power source and use it to boost or amplify the
output signal.
Filter amplification can also be used to either shape or alter the frequency
response of the filter circuit by producing a more selective output response,
making the output bandwidth of the filter more narrower or even wider. Then
the main difference between a “passive filter” and an “active filter” is
amplification.
An active filter generally uses an operational amplifier (op-amp) within its
design and in the Operational Amplifier tutorial we saw that an Op-amp has a
high input impedance, a low output impedance and a voltage gain determined
by the resistor network within its feedback loop.
Unlike a passive high pass filter which has in theory an infinite high frequency
response, the maximum frequency response of an active filter is limited to the
Gain/Bandwidth product (or open loop gain) of the operational amplifier being
used. Still, active filters are generally much easier to design than passive filters,
they produce good performance characteristics, very good accuracy with a
steep roll-off and low noise when used with a good circuit design.
The frequency response of the circuit will be the same as that for the passive
RC filter, except that the amplitude of the output is increased by the pass band
gain, AF of the amplifier. For a non-inverting amplifier circuit, the magnitude of
the voltage gain for the filter is given as a function of the feedback resistor ( R2 )
divided by its corresponding input resistor ( R1 ) value and is given as:
Therefore, the gain of an active low pass filter as a function of frequency will
be:
Gain of a first-order low pass filter
Where:
AF = the pass band gain of the filter, (1 + R2/R1)
ƒ = the frequency of the input signal in Hertz, (Hz)
ƒc = the cut-off frequency in Hertz, (Hz)
Thus, the operation of a low pass active filter can be verified from the
frequency gain equation above as:
So for a voltage gain of 10, R1 = 1kΩ and R2 = 9kΩ. However, a 9kΩ resistor
does not exist so the next preferred value of 9k1Ω is used instead. Converting
this voltage gain to an equivalent decibel dB value gives:
The cut-off or corner frequency (ƒc) is given as being 159Hz with an input
impedance of 10kΩ. This cut-off frequency can be found by using the formula:
By rearranging the above standard formula we can find the value of the filter
capacitor C as:
Thus the final low pass filter circuit along with its frequency response is given
below as:
If the external impedance connected to the input of the filter circuit changes,
this impedance change would also affect the corner frequency of the filter
(components connected together in series or parallel). One way of avoiding any
external influence is to place the capacitor in parallel with the feedback
resistor R2 effectively removing it from the input but still maintaining the filters
characteristics.
However, the value of the capacitor will change slightly from
being 100nF to 110nF to take account of the 9k1Ω resistor, but the formula
used to calculate the cut-off corner frequency is the same as that used for the
RC passive low pass filter.
An example of the new Active Low Pass Filter circuit is given as.
Simplified non-inverting amplifier filter circuit
Here due to the position of the capacitor in parallel with the feedback resistor
R2, the low pass corner frequency is set as before but at high frequencies the
reactance of the capacitor dominates shorting out R2 reducing the gain which
bottoms out at unity (0dB) since the gain equation becomes 1 + Xc/R1.
Applications of Active Low Pass Filters are in audio amplifiers, equalizers or
speaker systems to direct the lower frequency bass signals to the larger bass
speakers or to reduce any high frequency noise or “hiss” type distortion. When
used like this in audio applications the active low pass filter is sometimes called
a “Bass Boost” filter.
When cascading together filter circuits to form higher-order filters, the overall
gain of the filter is equal to the product of each stage. For example, the gain of
one stage may be 10 and the gain of the second stage may be 32 and the gain
of a third stage may be 100. Then the overall gain will be 32,000, (10 x 32 x
100) as shown below.
Second-order (two-pole) active filters are important because higher-order filters
can be designed using them. By cascading together first and second-order
filters, filters with an order value, either odd or even up to any value can be
constructed. In the next tutorial about filters, we will see that Active High Pass
Filters, can be constructed by reversing the positions of the resistor and
capacitor in the circuit.
The basic operation of an Active High Pass Filter (HPF) is the same as for its
equivalent RC passive high pass filter circuit, except this time the circuit has an
operational amplifier or included within its design providing amplification and
gain control.
Like the previous active low pass filter circuit, the simplest form of an active
high pass filter is to connect a standard inverting or non-inverting operational
amplifier to the basic RC high pass passive filter circuit as shown.
Then the performance of a “high pass filter” at high frequencies is limited by
this unity gain crossover frequency which determines the overall bandwidth of
the open-loop amplifier. The gain-bandwidth product of the op-amp starts from
around 100kHz for small signal amplifiers up to about 1GHz for high-speed
digital video amplifiers and op-amp based active filters can achieve very good
accuracy and performance provided that low tolerance resistors and capacitors
are used.
Under normal circumstances the maximum pass band required for a closed
loop active high pass or band pass filter is well below that of the maximum
open-loop transition frequency. However, when designing active filter circuits it
is important to choose the correct op-amp for the circuit as the loss of high
frequency signals may result in signal distortion.
This first-order high pass filter, consists simply of a passive filter followed by a
non-inverting amplifier. The frequency response of the circuit is the same as
that of the passive filter, except that the amplitude of the signal is increased by
the gain of the amplifier.
For a non-inverting amplifier circuit, the magnitude of the voltage gain for the
filter is given as a function of the feedback resistor ( R2 ) divided by its
corresponding input resistor ( R1 ) value and is given as:
For a first-order filter the frequency response curve of the filter increases by
20dB/decade or 6dB/octave up to the determined cut-off frequency point which
is always at -3dB below the maximum gain value. As with the previous filter
circuits, the lower cut-off or corner frequency ( ƒc ) can be found by using the
same formula:
The corresponding phase angle or phase shift of the output signal is the same
as that given for the passive RC filter and leads that of the input signal. It is
equal to +45o at the cut-off frequency ƒc value and is given as:
A simple first-order active high pass filter can also be made using an inverting
operational amplifier configuration as well, and an example of this circuit design
is given along with its corresponding frequency response curve. A gain of 40dB
has been assumed for the circuit.
or 16kΩ to the nearest preferred value.
Thus the pass band gain of the filter, AF is therefore given as being: 2.
As the value of resistor, R2 divided by resistor, R1 gives a value of one. Then,
resistor R1 must be equal to resistor R2, since the pass band gain, AF = 2. We
can therefore select a suitable value for the two resistors of say, 10kΩ each for
both feedback resistors.
So for a high pass filter with a cut-off corner frequency of 1kHz, the values
of R and C will be, 10kΩ and 10nF respectively. The values of the two
feedback resistors to produce a pass band gain of two are given
as: R1 = R2 = 10kΩ
The data for the frequency response bode plot can be obtained by substituting
the values obtained above over a frequency range from 100Hz to 100kHz into
the equation for voltage gain:
This then will give us the following table of data.
The frequency response data from the table above can now be plotted as
shown below. In the stop band (from 100Hz to 1kHz), the gain increases at a
rate of 20dB/decade. However, in the pass band after the cut-off frequency,
ƒC = 1kHz, the gain remains constant at 6.02dB. The upper-frequency limit of
the pass band is determined by the open loop bandwidth of the operational
amplifier used as we discussed earlier. Then the bode plot of the filter circuit
will look like this.
Applications of Active High Pass Filters are in audio amplifiers, equalizers or
speaker systems to direct the high frequency signals to the smaller tweeter
speakers or to reduce any low frequency noise or “rumble” type distortion.
When used like this in audio applications the active high pass filter is
sometimes called a “Treble Boost” filter.
Higher-order high pass active filters, such as third, fourth, fifth, etc are formed
simply by cascading together first and second-order filters. For example, a third
order high pass filter is formed by cascading in series first and second order
filters, a fourth-order high pass filter by cascading two second-order filters
together and so on.
Then an Active High Pass Filter with an even order number will consist of
only second-order filters, while an odd order number will start with a first-order
filter at the beginning as shown.
Although there is no limit to the order of a filter that can be formed, as the order
of the filter increases so to does its size. Also, its accuracy declines, that is the
difference between the actual stop band response and the theoretical stop
band response also increases.
If the frequency determining resistors are all equal, R1 = R2 = R3 etc, and the
frequency determining capacitors are all equal, C1 = C2 = C3 etc, then the cut-
off frequency for any order of filter will be exactly the same. However, the
overall gain of the higher-order filter is fixed because all the frequency
determining components are equal.
In the next tutorial about filters, we will see that Active Band Pass Filters, can
be constructed by cascading together a high pass and a low pass filter.
For a low pass filter this pass band starts from 0Hz or DC and continues up to
the specified cut-off frequency point at -3dB down from the maximum pass
band gain. Equally, for a high pass filter the pass band starts from this -3dB
cut-off frequency and continues up to infinity or the maximum open loop gain
for an active filter.
However, the Active Band Pass Filter is slightly different in that it is a
frequency selective filter circuit used in electronic systems to separate a signal
at one particular frequency, or a range of signals that lie within a certain “band”
of frequencies from signals at all other frequencies. This band or range of
frequencies is set between two cut-off or corner frequency points labelled the
“lower frequency” ( ƒL ) and the “higher frequency” ( ƒH ) while attenuating any
signals outside of these two points.
Simple Active Band Pass Filter can be easily made by cascading together a
single Low Pass Filter with a single High Pass Filter as shown.
The cut-off or corner frequency of the low pass filter (LPF) is higher than the
cut-off frequency of the high pass filter (HPF) and the difference between the
frequencies at the -3dB point will determine the “bandwidth” of the band pass
filter while attenuating any signals outside of these points. One way of making
a very simple Active Band Pass Filter is to connect the basic passive high
and low pass filters we look at previously to an amplifying op-amp circuit as
shown.
Active Band Pass Filter Circuit
This cascading together of the individual low and high pass passive filters
produces a low “Q-factor” type filter circuit which has a wide pass band. The
first stage of the filter will be the high pass stage that uses the capacitor to
block any DC biasing from the source. This design has the advantage of
producing a relatively flat asymmetrical pass band frequency response with
one half representing the low pass response and the other half representing
high pass response as shown.
The higher corner point ( ƒH ) as well as the lower corner frequency cut-off point
( ƒL ) are calculated the same as before in the standard first-order low and high
pass filter circuits. Obviously, a reasonable separation is required between the
two cut-off points to prevent any interaction between the low pass and high
pass stages. The amplifier also provides isolation between the two stages and
defines the overall voltage gain of the circuit.
The bandwidth of the filter is therefore the difference between these upper and
lower -3dB points. For example, suppose we have a band pass filter whose
-3dB cut-off points are set at 200Hz and 600Hz. Then the bandwidth of the filter
would be given as: Bandwidth (BW) = 600 – 200 = 400Hz.
The normalised frequency response and phase shift for an active band pass
filter will be as follows.
This type of band pass filter is designed to have a much narrower pass band.
The centre frequency and bandwidth of the filter is related to the values of R1,
R2, C1 and C2. The output of the filter is again taken from the output of the op-
amp.
We can see then that the relationship between resistors, R1 and R2 determines
the band pass “Q-factor” and the frequency at which the maximum amplitude
occurs, the gain of the circuit will be equal to -2Q2. Then as the gain increases
so to does the selectivity. In other words, high gain – high selectivity.
The closest standard value is 10nF.
Where:
ƒr is the resonant or Center Frequency
ƒL is the lower -3dB cut-off frequency point
ƒH is the upper -3db cut-off frequency point
and in our simple example in the text above of a filters lower and upper -3dB
cut-off points being at 200Hz and 600Hz respectively, then the resonant center
frequency of the active band pass filter would be:
As the quality factor of an active band pass filter (Second-order System) relates
to the “sharpness” of the filters response around its centre resonant frequency
( ƒr ) it can also be thought of as the “Damping Factor” or “Damping
Coefficient” because the more damping the filter has the flatter is its response
and likewise, the less damping the filter has the sharper is its response. The
damping ratio is given the Greek symbol of Xi, ( ξ ) where:
The “Q” of a band pass filter is the ratio of the Resonant Frequency, ( ƒr ) to
the Bandwidth, ( BW ) between the upper and lower -3dB frequencies and is
given as:
Then for our simple example above the quality factor “Q” of the band pass filter
is given as:
346Hz / 400Hz = 0.865. Note that Q is a ratio and has no units.
When analysing active filters, generally a normalised circuit is considered
which produces an “ideal” frequency response having a rectangular shape, and
a transition between the pass band and the stop band that has an abrupt or
very steep roll-off slope. However, these ideal responses are not possible in
the real world so we use approximations to give us the best frequency
response possible for the type of filter we are trying to design.
Probably the best known filter approximation for doing this is the Butterworth or
maximally-flat response filter. In the next tutorial we will look at higher order
filters and use Butterworth approximations to produce filters that have a
frequency response which is as flat as mathematically possible in the pass
band and a smooth transition or roll-off rate.
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.
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 filter 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.
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.
In practice however, Butterworth’s 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:
H0 = the Maximum Pass band Gain,
Amax.
H1 = the Minimum Pass band Gain.
The Frequency Response of a filter can be defined mathematically by
its Transfer Function with the standard Voltage Transfer
Function H(jω) written as:
Where:
Vout = the output signal
voltage.
Vin = the input signal voltage.
j = to the square root of -1
(√-1)
ω = the radian frequency
(2πƒ)
Note: ( jω ) 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:
2 (1+1.414s+s2)
3 (1+s)(1+s+s2)
4 (1+0.765s+s2)(1+1.848s+s2)
5 (1+s)(1+0.618s+s2)(1+1.618s+s2)
6 (1+0.518s+s2)(1+1.414s+s2)(1+1.932s+s2)
7 (1+s)(1+0.445s+s2)(1+1.247s+s2)(1+1.802s+s2)
8 (1+0.390s+s2)(1+1.111s+s2)(1+1.663s+s2)(1+1.962s+s2)
9 (1+s)(1+0.347s+s2)(1+s+s2)(1+1.532s+s2)(1+1.879s+s2)
10 (1+0.313s+s2)(1+0.908s+s2)(1+1.414s+s2)(1+1.782s+s2)(1+1.975s+s2)
Secondly, the minimum stop band gain Amin = -20dB which is equal to a gain
of 10 (-20dB = 20*log(A)) at a stop band frequency (ωs) of 800 rads/s or
127.3Hz.
Substituting the values into the general equation for a Butterworth filters
frequency response gives us the following:
Since n must always be an integer ( whole number ) then the next highest
value to 2.42 is n = 3, therefore a “a third-order filter is required” and to
produce a third-order Butterworth filter, a second-order filter stage cascaded
together with a first-order filter stage is required.
From the normalised low pass Butterworth Polynomials table above, the
coefficient for a third-order filter is given as (1+s)(1+s+s2) and this gives us a
gain of 3-A = 1, or A = 2. As A = 1 + (Rf/R1), choosing a value for both the
feedback resistor Rf and resistor R1 gives us values
of 1kΩ and 1kΩ respectively as: ( 1kΩ/1kΩ ) + 1 = 2.
We know that the cut-off corner frequency, the -3dB point (ωo) can be found
using the formula 1/CR, but we need to find ωo from the pass band
frequency ωp then,
So, the cut-off corner frequency is given as 284 rads/s or 45.2Hz, (284/2π) and
using the familiar formula 1/CR we can find the values of the resistors and
capacitors for our third-order circuit.
Note that the nearest preferred value to 0.352uF would be 0.36uF, or 360nF.
So for our 3rd-order Butterworth Low Pass Filter with a corner frequency of
45.2Hz, C = 360nF and R = 10kΩ
Second Order Filters which are also referred to as VCVS filters, because the
op-amp is used as a Voltage Controlled Voltage Source amplifier, are another
important type of active filter design because along with the active first order
RC filters we looked at previously, higher order filter circuits can be designed
using them.
In this analogue filters section tutorials we have looked at both passive and
active filter designs and have seen that first order filters can be easily
converted into second order filters simply by using an additional RC network
within the input or feedback path. Then we can define second order filters as
simply being: “two 1st-order filters cascaded together with amplification”.
Most designs of second order filters are generally named after their inventor
with the most common filter types
being: Butterworth, Chebyshev, Bessel and Sallen-Key. All these types of filter
designs are available as either: low pass filter, high pass filter, band pass filter
and band stop (notch) filter configurations, and being second order filters, all
have a 40-dB-per-decade roll-off.
The Sallen-Key filter design is one of the most widely known and popular 2nd
order filter designs, requiring only a single operational amplifier for the gain
control and four passive RC components to accomplish the tuning.
Most active filters consist of only op-amps, resistors, and capacitors with the
cut-off point being achieved by the use of feedback eliminating the need for
inductors as used in passive 1st-order filter circuits.
Second order (two-pole) active filters whether low pass or high pass, are
important in Electronics because we can use them to design much higher order
filters with very steep roll-off’s and by cascading together first and second order
filters, analogue filters with an nth order value, either odd or even can be
constructed up to any value, within reason.
Second Order Low Pass Filter
Second order low pass filters are easy to design and are used extensively in
many applications. The basic configuration for a Sallen-Key second order (two-
pole) low pass filter is given as:
This second order low pass filter circuit has two RC networks, R1 – C1 and R2
– C2 which give the filter its frequency response properties. The filter design is
based around a non-inverting op-amp configuration so the filters gain, A will
always be greater than 1. Also the op-amp has a high input impedance which
means that it can be easily cascaded with other active filter circuits to give
more complex filter designs.
The normalised frequency response of the second order low pass filter is fixed
by the RC network and is generally identical to that of the first order type. The
main difference between a 1st and 2nd order low pass filter is that the stop
band roll-off will be twice the 1st order filters at 40dB/decade (12dB/octave) as
the operating frequency increases above the cut-off frequency ƒc, point as
shown.
The frequency response bode plot above, is basically the same as that for a
1st-order filter. The difference this time is the steepness of the roll-off which is
-40dB/decade in the stop band. However, second order filters can exhibit a
variety of responses depending upon the circuits voltage magnification
factor, Q at the the cut-off frequency point.
In active second order filters, the damping factor, ζ (zeta), which is the inverse
of Q is normally used. Both Q and ζ are independently determined by the gain
of the amplifier, A so as Q decreases the damping factor increases. In simple
terms, a low pass filter will always be low pass in its nature but can exhibit a
resonant peak in the vicinity of the cut-off frequency, that is the gain can
increases rapidly due to resonance effects of the amplifiers gain.
Then Q, the quality factor, represents the “peakiness” of this resonance peak,
that is its height and narrowness around the cut-off frequency point, ƒC. But a
filters gain also determines the amount of its feedback and therefore has a
significant effect on the frequency response of the filter.
Generally to maintain stability, an active filters gain must not be more than 3
and is best expressed as:
Then we can see that the filters gain, A for a non-inverting amplifier
configuration must lie somewhere between 1 and 3 (the damping
factor, ζ between zero an 2). Therefore, higher values of Q, or lower values
of ζ gives a greater peak to the response and a faster initial roll-off rate as
shown.
The amplitude response of the second order low pass filter varies for different
values of damping factor, ζ. When ζ = 1.0 or more (2 is the maximum) the filter
becomes what is called “overdamped” with the frequency response showing a
long flat curve. When ζ = 0, the filters output peaks sharply at the cut-off point
resembling a sharp point at which the filter is said to be “underdamped”.
Then somewhere in between, ζ = 0 and ζ = 2.0, there must be a point where
the frequency response is of the correct value, and there is. This is when the
filter is “critically damped” and occurs when ζ = 0.7071.
One final note, when the amount of feedback reaches 4 or more, the filter
begins to oscillate by itself at the cut-off frequency point due to the resonance
effects, changing the filter into an oscillator. This effect is called self oscillation.
Then for a low pass second order filter, both Q and ζ play a critical role.
We can see from the normalised frequency response curves above for a 1st
order filter (red line) that the pass band gain stays flat and level (called
maximally flat) until the frequency response reaches the cut-off frequency point
when: ƒ = ƒr and the gain of the filter reduces past its corner frequency at 1/√2,
or 0.7071 of its maximum value. This point is generally referred to as the
filters -3dB point and for a first order low pass filter the damping factor will be
equal to one, ( ζ = 1 ).
However, this -3dB cut-off point will be at a different frequency position for
second order filters because of the steeper -40dB/decade roll-off rate (blue
line). In other words, the corner frequency, ƒr changes position as the order of
the filter increases. Then to bring the second order filters -3dB point back to the
same position as the 1st order filter’s, we need to add a small amount of gain to
the filter.
So for a Butterworth second order low pass filter design the amount of gain
would be: 1.586, for a Bessel second order filter design: 1.268, and for a
Chebyshev low pass design: 1.234.
We know from above that the gain of a non-inverting op-amp is given as:
Therefore the final circuit for the second order low pass filter is given as:
We can see that the peakiness of the frequency response curve is quite sharp
at the cut-off frequency due to the high quality factor value, Q = 5. At this point
the gain of the filter is given as: Q × A = 14, or about +23dB, a big difference
from the calculated value of 2.8, (+8.9dB).
But many books, like the one on the right, tell us that the gain of the filter at the
normalised cut-off frequency point, etc, etc, should be at the -3dB point. By
lowering the value of Q significantly down to a value of 0.7071, results in a gain
of, A = 1.586 and a frequency response which is maximally flat in the passband
having an attenuation of -3dB at the cut-off point the same as for a second
order butterworth filter response.
So far we have seen that second order filters can have their cut-off frequency
point set at any desired value but can be varied away from this desired value
by the damping factor, ζ. Active filter designs enable the order of the filter to
range up to any value, within reason, by cascading together filter sections.
In practice when designing nth-order low pass filters it is desirable to set the cut-
off frequency for the first-order section (if the order of the filter is odd), and set
the damping factor and corresponding gain for each of the second order
sections so that the correct overall response is obtained. To make the design of
low pass filters easier to achieve, values of ζ, Q and A are available in
tabulated form for active filters as we will see in the Butterworth Filter tutorial.
Let’s look at another example.
Choosing a suitable value of 1kΩ for the filters resistors, then the resulting
capacitor values are calculated as:
The voltage gain for a non-inverting op-amp circuit was given previously as:
Therefore the second order low pass filter circuit which has a Q of 1, and a
corner frequency of 79.5Hz is given as:
Since second order high pass and low pass filters are the same circuits except
that the positions of the resistors and capacitors are interchanged, the design
and frequency scaling procedures for the high pass filter are exactly the same
as those for the previous low pass filter. Then the bode plot for a 2nd order
high pass filter is therefore given as:
As with the previous low pass filter, the steepness of the roll-off in the stop
band is -40dB/decade.
In the above two circuits, the value of the op-amp voltage gain, ( Av ) is set by
the amplifiers feedback network. This only sets the gain for frequencies well
within the pass band of the filter. We can choose to amplify the output and set
this gain value by whatever amount is suitable for our purpose and define this
gain as a constant, K.
2nd order Sallen-Key filters are also referred to as positive feedback filters
since the output feeds back into the positive terminal of the op-amp. This type
of active filter design is popular because it requires only a single op-amp, thus
making it relatively inexpensive.
State Variable Filter
State variable filters use three (or more) operational amplifier circuits (the
active element) cascaded together to produce the individual filter outputs but if
required an additional summing amplifier can also be added to produce a
fourth Notch filter output response as well.
State variable filters are second-order RC active filters consisting of two
identical op-amp integrators with each one acting as a first-order, single-pole
low pass filter, a summing amplifier around which we can set the filters gain
and its damping feedback network. The output signals from all three op-amp
stages are fed back to the input allowing us to define the state of the circuit.
One of the main advantages of a state variable filter design is that all three of
the filters main parameters, Gain (A), corner frequency, ƒC and the filters Q can
be adjusted or set independently without affecting the filters performance.
In fact if designed correctly, the -3dB corner frequency, ( ƒc ) point for both the
low pass amplitude response and the high pass amplitude response should be
identical to the center frequency point of the band pass stage. That is ƒLP(-
3dB) equals ƒHP(-3dB) which equals ƒBP(center). Also the damping factor, ( ζ ) for the band
pass filter response should be equal to 1/Q as Q will be set at -3dB, (0.7071).
Although the filter provides low pass (LP), high pass (HP) and band pass (BP)
outputs the main application of this type of filter circuit is as a state variable
band pass filter design with the center frequency set by the two RC integers.
While we have seen before that a band pass filters characteristics can be
obtained by simply cascading together a low pass filter with a high pass filter,
state variable band pass filters have the advantage that they can be tuned to
be highly selective (high Q) offering high gains at the center frequency point.
There are several state variable filter designs available all based on the
standard filter design with both inverting and non-inverting variations available.
However, the basic filter design will be the same for both variations as shown in
the following block diagram representation.
Then we can see from the basic block diagram above that the state variable
filter has three possible outputs, VHP, VBP and VLP with one each from the three
op-amps. A notch filter response can also be realized by the addition of a fourth
op-amp.
With a constant input voltage, VIN the output from the summing amplifier
produces a high pass response which also becomes the input of the first RC
Integrator. The output from this integrator produces a band pass response
which becomes the input of the second RC Integrator producing a low pass
response at its output. As a result, separate transfer functions for each
individual output with respect to the input voltage can be found.
The basic non-inverting state variable filter design is therefore given as:
and the amplitude response of the three outputs from the state variable filter
will look like:
One of the main design elements of a state variable filter is its use of two op-
amp integrators. As we saw in the Integrator tutorial, op-amp integrators use a
frequency dependant impedance in the form of a capacitor within their
feedback loop. As a capacitor is used the output voltage is proportional to the
integral of the input voltage as shown.
To simplify the math’s a little, this can also be re-written in the frequency
domain as:
Then by rearranging this formula we can find the transfer function of the
inverting integrator, A2
Exactly the same assumption can be made as above to find the transfer
function for the other op-amp integrator, A3
Note that each integrator stage provides an inverted output but the summed
output will be positive since they are inverting integrators. If exactly the same
values for R and C are used so that the two circuits have the same integrator
time constant, the two amplifier circuits can be regarded with one single
integrator circuit having a corner frequency, ƒC.
As well as the two integrator circuits, the filter also has a differential summing
amplifier providing a weighted summation of its inputs. The advantage here is
that the inputs to the summing amplifier, A1 combines oscillatory feedback,
damping and input signals to the filter as all three outputs are fed back to the
summing inputs.
and
As the differential inputs, +V and -V of an operational amplifier are the same,
that is: +V – -V, we can rearrange the two expressions above to find the
transfer function for the output of A1, the high pass output.
We can choose a value for either the resistor, or the capacitor to find the value
of the other. If we assume a suitable value of 10nF for the capacitor then the
value of the resistor will be:
From the normalised transfer function above, the DC passband gain is defined
as Ao and from the equivalent state variable filter transfer function this equates
to:
Therefore the DC voltage gain of the filter is calculated at 1.9, which basically
equates to R2/R3. Also the maximum gain of the filter at ƒC can be calculated
as: Ao x Q as follows.
We can now plot the individual output response curves for the state variable
filter circuit over a range of frequencies from 1Hz to 1MHz onto a Bode Plot as
shown.
Then we can see from the filters response curves above, that the DC gain of
the filter circuit is at 5.57dB which equates to an open loop voltage gain, Ao or
1.9 as calculated above. The response also shows that the output curves
peaks at a maximum voltage gain of 25.6dB at the corner frequency due to the
value of Q. As Q also relates the band pass filters center frequency to its
bandwidth, the bandwidth of the filter will therefore be: ƒo/10 = 100Hz.
We have seen in this state variable filter tutorial that instead of an active filter
producing one type of frequency response, we can use multiple-feedback
techniques to produce all three filter responses, Low Pass, High
Pass and Band Pass simultaneously from the same single active filter design.
But as well as the three basic filter responses, we can add an additional op-
amp circuit onto the basic state variable filter design above to produce a fourth
output response resembling that of a standard Notch Filter.
Here to keep things simple we have assumed that the two input
resistors, R5 and R6 as well as the feedback resistor, R7 all have the same
value of 10kΩ the same as for R3 and R4. This therefore gives the notch filter a
gain of 1, unity.
The output response of the notch filter and bandpass filter are related with the
center frequency of the bandpass response being equal to the point of zero
response of the notch filter, and in this example will be 1kHz.
Also the bandwidth of the notch is determined by the circuits Q, exactly the
same as for the pass band response. The downward peak is therefore equal to
the center frequency divided by the -3dB bandwidth, that is the frequency
difference between the -3dB points either side of the notch. Note that the
quality factor Q has nothing to do with the actual depth of the notch.
This basic notch filter (band-stop) design has only two inputs applied to its
summing amplifier, the low pass output, VLP and the high pass output, VHP.
However, there are two more signals available for us to use from the basic
state variable filter circuit, the band pass output, VBP and the input signal
itself, VIN.
If one of these two signals is also used as an input to the notch filter summing
amplifier along with the low pass and high pass signals, then the depth of the
notch can be controlled.
Depending upon how you wanted to control the output from the notch filter
section would depend upon which one of the two available signals you would
use. If it was required that the output notch changes from a negative response
to a positive response at the undamped natural frequency ƒo then the band
pass output signal VBP would be used.
Likewise, if it was required that the output notch only varies in its downward
negative depth, then the input signal, VIN would be used. If either one of these
two additional signals was connected to the op-amp summing amplifier through
a variable resistor then the depth and direction of the notch could be fully
controlled. Consider the modified notch filter circuit below.
We can see from the amplitude and phase curves above for the band pass
circuit, that the quantities ƒL, ƒH and ƒC are the same as those used to describe
the behaviour of the band-pass filter. This is because the band stop filter is
simply an inverted or complimented form of the standard band-pass filter. In
fact the definitions used for bandwidth, pass band, stop band and center
frequency are the same as before, and we can use the same formulas to
calculate bandwidth, BW, center frequency, ƒC, and quality factor, Q.
The ideal band stop filter would have infinite attenuation in its stop band and
zero attenuation in either pass band. The transition between the two pass
bands and the stop band would be vertical (brick wall). There are several ways
we can design a “Band Stop Filter”, and they all accomplish the same purpose.
Generally band-pass filters are constructed by combining a low pass filter
(LPF) in series with a high pass filter (HPF). Band stop filters are created by
combining together the low pass and high pass filter sections in a “parallel”
type configuration as shown.
The summing of the high pass and low pass filters means that their frequency
responses do not overlap, unlike the band-pass filter. This is due to the fact
that their start and ending frequencies are at different frequency points. For
example, suppose we have a first-order low-pass filter with a cut-off
frequency, ƒL of 200Hz connected in parallel with a first-order high-pass filter
with a cut-off frequency, ƒH of 800Hz. As the two filters are effectively
connected in parallel, the input signal is applied to both filters simultaneously
as shown above.
All of the input frequencies below 200Hz would be passed unattenuated to the
output by the low-pass filter. Likewise, all input frequencies above 800Hz would
be passed unattenuated to the output by the high-pass filter. However, and
input signal frequencies in-between these two frequency cut-off points of
200Hz and 800Hz, that is ƒL to ƒH would be rejected by either filter forming a
notch in the filters output response.
In other words a signal with a frequency of 200Hz or less and 800Hz and
above would pass unaffected but a signal frequency of say 500Hz would be
rejected as it is too high to be passed by the low-pass filter and too low to be
passed by the high-pass filter. We can show the effect of this frequency
characteristic below.
The use of operational amplifiers within the band stop filter design also allows
us to introduce voltage gain into the basic filter circuit. The two non-inverting
voltage followers can easily be converted into a basic non-inverting amplifier
with a gain of Av = 1 + Rƒ/Rin by the addition of input and feedback resistors,
as seen in our non-inverting op-amp tutorial.
Also if we require a band stop filter to have its -3dB cut-off points at say, 1kHz
and 10kHz and a stop band gain of -10dB in between, we can easily design a
low-pass filter and a high-pass filter with these requirements and simply
cascade them together to form our wide-band band-pass filter design.
Now we understand the principle behind a Band Stop Filter, let us design one
using the previous cut-off frequency values.
The upper and lower cut-off frequency points for a band stop filter can be found
using the same formula as that for both the low and high pass filters as shown.
Assuming a capacitor, C value for both filter sections of 0.1uF, the values of the
two frequency determining resistors, RL and RH are calculated as follows.
Now that we know the component values for the two filter stages, we can
combine them into a single voltage adder circuit to complete our filter design.
The magnitude and polarity of the adders output will be at any given time, the
algebraic sum of its two inputs.
If we make the op-amps feedback resistor and its two input resistors the same
values, say 10kΩ, then the inverting summing circuit will provide a
mathematically correct sum of the two input signals with zero voltage gain.
Then the final circuit for our band stop (band-reject) filter example will be:
We have seen above that simple band stop filters can be made using first or
second order low and high pass filters along with a non-inverting summing op-
amp circuit to reject a wide band of frequencies. But we can also design and
construct band stop filters to produce a much narrower frequency response to
eliminate specific frequencies by increasing the selectivity of the filter. This type
of filter design is called a “Notch Filter”.
Notch Filters
Notch filters are a highly selective, high-Q, form of the band stop filter which
can be used to reject a single or very small band of frequencies rather than a
whole bandwidth of different frequencies. For example, it may be necessary to
reject or attenuate a specific frequency generating electrical noise (such as
mains hum) which has been induced into a circuit from inductive loads such as
motors or ballast lighting, or the removal of harmonics, etc.
But as well as filtering, variable notch filters are also used by musicians in
sound equipment such as graphic equalizers, synthesizers and electronic
crossovers to deal with narrow peaks in the acoustic response of the music.
Then we can see that notch filters are widely used in much the same way as
low-pass and high-pass filters.
Notch filters by design have a very narrow and very deep stop band around
their center frequency with the width of the notch being described by its
selectivity Q in exactly the same way as resonance frequency peaks in RLC
circuits.
The most common notch filter design is the twin-T notch filter network. In its
basic form, the twin-T, also called a parallel-tee, configuration consists of two
RC branches in the form of two tee sections, that use three resistors and three
capacitors with opposite and opposing R and C elements in the tee part of its
design as shown, creating a deeper notch.
Here the output from the twin-T notch filter section is isolated from the voltage
divider by a single non-inverting op-amp buffer. The output from the voltage
divider is fed back to “ground” point of R and 2C. The amount of signal
feedback, known as the feedback fraction k, is set by the resistor ratio and is
given as:
The basic circuit shown consists of two resistors in series connected across an
input voltage, VIN.
Ohm’s Law tells us that the voltage dropped across a resistor is the sum of the
current flowing through it multiplied by its resistive value, V = I*R, so if the two
resistors are equal, then the voltage dropped across both
resistors, R1 and R2 will also be equal and is split equally between them.
The voltage developed or dropped across resistor R2 represents the output
voltage, VOUT and is given by the ratio of the two resistors and the input voltage.
Thus the transfer function for this simple voltage divider network is given as:
Resistive Voltage Divider Transfer Function
But what would happen to the output voltage, VOUT if we changed the input
voltage to an AC supply or signal, and varied its frequency range. Well actually
nothing, as resistors are generally not affected by changes in frequency
(wirewounds excluded) so their frequency response is zero, allowing AC,
Irms2*R voltages to be developed or dropped across the resistors just the same
as it would be for steady state DC voltages.
RC Voltage Divider
Thus when a steady state DC supply is connected to VIN, the capacitor will be
fully charged after 5 time constants (5T = 5RC) and in which time it draws no
current from the supply. Therefore there is no current flowing through the
resistor, R and no voltage drop developed across it, so no output voltage. In
other words, capacitors block steady state DC voltages once charged.
If we now change the input supply to an AC sinusoidal voltage, the
characteristics of this simple RC circuit completely changes as the DC or
constant part of the signal is blocked. So now we are analysing the RC circuit
in the frequency domain, that is the part of the signal that depends on time.
In an AC circuit, a capacitor has the property of capacitive reactance, XC but
we can still analyse the RC circuit in the same way as we did with the resistor
only circuits, the difference is that the impedance of the capacitor now depends
on frequency.
For AC circuits and signals, capacitive reactance (XC), is the opposition to
alternating current flow through a capacitor measured in Ohm’s. Capacitive
reactance is frequency dependant, that is at low frequencies (ƒ ≅ 0) the
capacitor behaves like an open circuit and blocks them
At very high frequencies (ƒ ≅ ∞) the capacitor behaves like a short circuit and
pass the signals directly to the output as VOUT = VIN. However, somewhere in
between these two frequency extremes the capacitor has an impedance given
by XC. So our voltage divider transfer function from above becomes:
RC Filter Circuit
The graph shows the frequency response of this simple 1st-order RC circuit. At
low frequencies the voltage gain is extremely low, as the input signal is being
block by the reactance of the capacitor. At high frequencies the voltage gain is
high (unity) as the reactance is causes the capacitor to effectively become a
short-circuit to these high frequencies, so VOUT = VIN
However, there becomes a frequency point where the reactance of the
capacitor is equal to the resistance of the resistor, that is: XC = R and this is
called the “critical frequency” point, or more commonly called the cut-off
frequency, or corner frequency ƒC.
As the cut-off frequency occurs when XC = R the standard equation used to
calculate this critical frequency point is given as:
RC Filter Circuit
One of the main disadvantages of an RC filter is that the output amplitude will
always be less than the input so it can never be greater than unity. Also the
external loading of the output by more RC stages or circuits will have an affect
on the filters characteristics. One way to overcome this problem is to convert
the passive RC filter into an “Active RC Filter” by adding an operational
amplifier to the basic RC configuration.
By adding an operational amplifier, the basic RC filter can be designed to
provide a required amount of voltage gain at its output, thus changing the filter
from an attenuator to an amplifier. Also due to the high input impedance and
low output impedance of an operational amplifier prevents external loading of
the filter allowing it to be easily adjusted over a wide frequency range without
altering the designed frequency response.
Consider the simple active RC high-pass filter below.
The calculated value of C is 63.65nF, so the nearest preferred value used is
62nF.
The gain of the high pass filter in the passband region is to be +9dB which
equates to a voltage gain, AV of 2.83. Assume an arbitrary value for feedback
resistor, R1 of 15kΩ, this gives a value for resistor R1 of:
Again the calculated value of R2 is 8197Ω. The nearest preferred value would
be 8200Ω or 8.2kΩ. This then gives us the final circuit for our active high-pass
filter example of:
We have seen that a simple first-order high-pass filters can be made using a
single resistor and capacitor producing a cut-off frequency, ƒC point where the
output amplitude is –3dB down from the input amplitude. By adding a second
RC filter stage to the first, we can convert the circuit into a second-order high-
pass filter.
Second-order RC Filter
The simplest second order RC filter consists of two RC sections cascade
together as shown. However, for this basic configuration to operate correctly,
input and output impedances of the the two RC stages should not affect each
others operation, that is they should be non-interacting.
High Pass Filter Circuit
Therefore for the unity-gain buffer configuration, the voltage gain (AV) of the
filter circuit is equal to 0.5, or -6dB (over damped) at the cut-off frequency point,
and we would expect to see this because its a second-order filter response, as
0.7071*0.7071 = 0.5. That is -3dB*-3dB = -6dB.
However, as the value of Q determines the response characteristics of the
filter, the proper selection of the operational amplifiers two feedback
resistors, R1 and R2, allows us to select the required passband gain A for the
chosen magnification factor, Q.
Note that for a Sallen-key filter topology, selecting the value of A to be very
close to the maximum value of 3, will result in high Q values. A high Q will
make the filter design sensitive to tolerance variations in the values of feedback
resistors R1 and R2. For example, setting the voltage gain to 2.9 (A = 2.9) will
result in the value of Q being 10 (1/(3-2.9)), thus the filter becomes extremely
sensitive around ƒC.
Then we can see that the lower the value of Q the more stable will be the
Sallen and Key filter design. While high values of Q can make the design
unstable, with very high gains producing a negative Q would lead to
oscillations.
Sallen and Key Filter Example No2
Design a second-order high-pass Sallen and Key Filter circuit with the following
characteristics: ƒC = 200Hz, and Q = 3
To simplify the math’s a little, we will assume that the two series
capacitors CA and CB are equal (CA = CB = C) and also the two
resistors RA and RB are equal (RA = RB = R).
The calculated value of R is 7957Ω, so the nearest preferred value used is
8kΩ.
For Q = 3, the gain is calculated as:
The calculated value of R2 is 5998Ω, so the nearest preferred value used
6000Ω or 6kΩ. This then gives us the final circuit for our Sallen and Key high-
pass filter example of:
Decibels
Decibels
The decibel is the base-10 logarithm ratio used to express an increase or decrease in
power, voltage, or current in a circuit
When designing or working with amplifier and filter circuits, some of the
numbers used in the calculations can be very large or very small. For example,
if we cascade two amplifier stages together with power or voltage gains of say
20 and 36, respectively, then the total gain would be 720 (20*36). Likewise if
we cascaded together to first-order RC filter circuits with attenuations of 0.7071
each, the the total attenuation would be 0.5 (0.7071*0.7071). Remembering of
course that if a circuits output is positive, then it produces amplification or gain,
and if its output is negative, then it produces attenuation or loss.
When analysing circuits in the frequency domain, it is more convenient to
compare the amplitude ratio of the output to input values on a logarithmic scale
rather than on a linear scale. So if we use the logarithmic ratio of two
quantities, P1 and P2 we end up with a new quantity or level which can be
presented using Decibels.
Unlike voltage or current which is measured in volts and amperes respectively,
the decibel, or simple dB for short, is just a ratio of two values, well actually
the ratio of one value against another known or fixed value, so therefore the
decible is a dimensionless quantity, but does have the “Bel” as its units after
the telephone inventor, Alexander Graham Bell.
The ratio of any two values, where one is fixed or known and of the same
qunatity or units, whether power, voltage or current, can be represented using
decibels (dB) where “deci” means one tenth (1/10th) of a Bel. Clearly then
there are 10 decibels (10dB) per Bel or 1 Bel = 10 decibels.
The decibel is commonly used to show the ratio of power change (increasing or
decreasing) and is defined as the value which is ten times the Base-10
logarithm of two power levels. For example, 1 watt to 10 watts is the same
power ratio as 10 watts to 100 watts, that is 10:1, so while there is a large
difference in the number of watts, 9 compared to 90, the decibel ratio would be
exactly the same. Hopefully then we can see that the decibel (dB) is a ratio
used for comparing and calculating levels of power change and not the power
itself.
Thus, if we have two quantities of power, P1 and P2, the ratio of these two
values is represented by the equation:
dB = 10log10[P2/P1]
Where, P1 represents the input power and P2 represents the output power,
(POUT/PIN).
As the decibel represents the Base-10 logarithm change of two power levels,
we can expand this equation further to show by how much change one decibel
(1dB) really is.
dB = 10log10[P2/P1]
We can express the power gain of the amplifier in units of decibels regardless
of its input or output values, as an amplifier delivering 40 watts output for
40mW input will also have a power gain of 30 dB, and so on.
We could also, if we so wished, convert this amplifiers decibels value back into
a linear value by first converting from decibels (dB) to a Bel remembering that a
decibel is 1/10th of a Bel. For example:
A 100 watt audio amplifier has a power gain ratio of 30dB. What will be its
maximum input value.
So the result is 100mW as declared in example No1.
One of the advantages of using the base 10 logarithm ratio of two powers is
that when dealing with multiple amplifier, filter or attenuator stages cascaded
together, we can simply add or subtract their decibel values instead of
multiplying or dividing their linear values. In other words, a circuits overall gain
(+dB), or attenuation (-dB) is the sum of the individual gains and attenuations
for all stages connected between the input and output.
For example, if a single stage amplfier has a power gain of 20dB and it
supplies a passive resistive network that has an attenuation of 2, before the
signal is amplified again using a second amplifier stage with a gain of 200.
Then the total power gain of the circuit between the input and output in decibels
would be:
For the passive circuit, an attenuation of 2 is the same as saying the circuit has
a positive gain of 1/2 = 0.5, thus the power gain of the passive section is:
dB Gain = 10log10[0.5] = -3dB (note a negative value)
The second stage amplifier has a gain of 200, thus the power gain of this
section is:
dB Gain = 10log10[200] = +23dB
Then the overall gain of the circuit will be:
20 - 3 + 23 = +40dB
We can double check our answer of 40dB by multiplying the individual gains of
each stage in the usual way as follows:
A power gain of 20dB in decibels is equal to a gain of 100, as 10(20/10) = 100. So:
100 x 0.5 x 200 = 10,000 (or 10,000 times greater)
Converting this back to a decibel value gives:
dB Gain = 10log10[10,000] = 40dB
Then clearly we can see that a gain of 10,000 is equal to a power gain ratio of
+40dB as shown above and that we can use the decibel value to express large
ratios of powers with much smaller numbers as 40dB is a power ratio of
10,000, whereas -40dB is a power ratio of 0.0001. So using decibels makes the
maths a little easier.
that is 20log(voltage gain), and for the current gain would be:
Thus the only difference between defining the power, voltage, and current
decibel (dB) calculations is the constant of 10 and 20, and that for the dB ratio
to be correct in all instancies the two quantities must both have the same units,
either watts, milli-watts, volts, milli-volts, amperes or milli-amperes, or any other
unit.
0dB 1 1
6dB 4 2
20dB 100 10
We can see from the above decibel table that at 0dB the ratio gain for power,
voltage and current is equal to “1” (unity). This means that the circuit (or
system) produces no gain or loss between the input and output signals. So
zero dB corresponds to a unity gain i.e. A = 1 and not zero gain.
We can also see that at +3dB the output of the circuit (or system) has doubled
its input value, meaning a positive dB gain (amplification) so A > 1. Likewise, at
-3dB the output the circuit is at half its input value, meaning a negative dB gain
(attenuation) so A < 1. This -3dB value is commonly called the “half-power”
point and defines the corner frequency in filter networks.
It is all well and good tabulating the power gains against decibels in a reference
table, but when dealing with amplifier and filters, Electrical Engineers prefer to
use Bode Plots, charts or graphs as a visual display of the circuits (or systems)
frequency response characteristics. Then using the data values in the table
above we can create the following “decibel” Bode plot showing the various
positions of the power points.
Then we can clearly see that the power curve is not linear but follows the
logarithmic ratio of 1.259.