Elliott Sound Products Equalisers

Equalisers, The Various Types And How They Work

© Rod Elliott (ESP), March 2015
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Articles Index
Main Index

Contents

Introduction
1 - Fixed Equalisers
2 - Tone Controls For Reproduction
3 - Tone Controls For Production
4 - Basic Tone Controls
5 - Graphic Equalisers
6 - Variable Frequency Tone Controls
7 - Parametric Equalisers
8 - Guitar Amp Tone Stacks
9 - Frequency 'Isolators'
10 - 'Tilt' Controls
Conclusions
References

Introduction

Equalisation (EQ) is one of the most contentious areas of hi-fi. For many years, it was expected of any
preamplifier that it would have (at the minimum) bass and treble controls. There were untold variations of
course, but the general scheme that ended up being used by almost all manufacturers was the 'Baxandall'
topology, named after its inventor Peter J Baxandall. This arrangement is used to this day, but for audio
production (as opposed to reproduction) the equalisation available is much more complex and
comprehensive.

The term 'equalisation' probably came from the requirements of various operators (phone, motion picture,
broadcast, etc.) to get their systems back to a flat frequency response - in other words to make it 'equal' to
the intended signal.

In reality, equalisation (or simply 'filtering' as it was known in the early years) has been part of recording
and PA equipment from the beginning of the technology. Western Electric (which eventually became Bell
Labs) described filters (equalisers) for the telephone system to adjust the frequency response and correct
high frequency rolloff in the telephone lines. Early 'tone' controls were in evidence not long after the
advent of AM radio ('wireless' as it was known at the time). These were typically only able to roll off the
high frequencies to make the sound more 'mellow' and reduce extraneous noise.

While audiophiles the world over eschew any form of EQ, at least 99% of the recordings they listen to
have already been processed with individual EQ on each channel, as well as overall EQ, compression,
limiting, and other 'effects' as may be deemed appropriate by the recording and mastering engineers.
However, in this article, I will discuss mainly 'user adjustable' equalisation ('equalization' for North
American readers).

Mixing desks for recording and live production provide extensive EQ, and no-one would be silly enough to
build a mixer without it. Each channel has a comprehensive tone control network, almost always with at
least two bands of parametric equalisation. The term 'parametric' refers to the fact that all the parameters
of the circuit are adjustable - frequency, bandwidth (Q) and boost/ cut.

Daniel Flickinger introduced the first parametric equaliser in early 1971 (US Patent 3752928 A). His
design used opamps to create filter circuits that were not viable with other techniques. Flickinger's patent
("Amplifier system utilizing regenerative and degenerative feedback to shape the frequency response")
shows the circuit topology that was used, and it forms the basis of parametric EQ used to this day.

An earlier form of comprehensive tone control was the graphic equaliser - so-called because the slider
pots described a 'graph' of the final frequency response. To be useful, a graphic EQ system needs a lot of
separate filters. Octave band graphic EQ systems used 10 slide pots, with one for each octave. More
expensive units had 20 sliders (1/2 octave) or 30 sliders (1/3 octave). It was common for these to use
ferrite-cored inductors prior to the development of integrated opamps and the invention of the 'gyrator'
circuit. A gyrator uses an opamp, resistors and a capacitor to simulate an inductor (hence the generic
name 'simulated inductor').

It's often been stated that "tone controls are provided so the user can mess up the sound". In many cases
this is certainly true, but it has to be considered that the end-user is perfectly entitled to mess up the
sound if s/he wants to do so. This article is not about ultimate sound quality, but the various types of
equaliser that are available, and how they work.

It's also worth your while to browse the various circuits from the ESP projects list. There are quite a few
different types of equaliser described, ranging from simple bass and treble controls through to quasi-
parametric designs, graphic equalisers and fixed EQ systems for low frequency response extension for
loudspeakers and subwoofers.

Note that all the circuits shown below rely on a low or very low impedance source. This
can be an opamp (best), transistor emitter follower (ok) or a valve cathode follower
(worst), depending on the other circuitry used. So, although input buffers are not shown
they are essential in all cases. This still applies where the input uses an inverting opamp
stage, because the insertion loss of the circuit depends on a low source impedance.

The circuits below are not for construction (although you can do so if you wish, but don't expect
assistance). Because they are not projects, none has been built as shown, and although all have been
simulated no other tests have been done. Likewise, there's been no attempt to optimise the circuits for
any particular task, so they may not be found suitable as described. I will respond to queries about
projects, but I will not provide assistance to anyone to build any of the circuits shown here.

1 - Fixed Equalisers

The most common fixed EQ circuit is that used for RIAA vinyl phono playback from magnetic pickups.
Although there is vast number of different topologies, the end result is pretty much the same. RIAA
playback EQ provides bass boost and treble cut to match the disc cutting process. This (by design) cuts
the bass response so the grooves aren't so wide as to cut into adjacent grooves, and boosts the treble as
a form of pre-emphasis. Upon playback, the treble cut reduces the disc's surface noise sufficiently to
produce a fairly quiet end result.

Other common fixed equalisers are or were used with recording tape, FM broadcast, long phone lines
used for radio or television distribution and a multitude of other systems. Pre-emphasis (treble boost) and
de-emphasis (complementary treble cut) increase the apparent signal to noise ratio (SNR) and these
have been used for many years. Pre-emphasis is used in FM broadcasts, and the receivers have a
complementary de-emphasis circuit that gives an overall flat response.

Fixed equalisers can also be used to allow a loudspeaker to achieve (or attempt) 'full range' from single
loudspeaker drivers. One of the best known is probably the Bose 901, which uses 9 × 100mm (4") drivers
and has a 'line level' equaliser that supposedly produces flat response (although it also has some tone
control available). Many subwoofers use a fixed equaliser to get as low as possible even in a small
enclosure.

Modern systems using DSP (digital signal processing) may also qualify as 'fixed' EQ, because after the
setup process is complete there is usually no facility to adjust the relative levels. There's also a
movement to apply EQ to 'correct' the speakers for the room, but this is a flawed concept for the most
part, other than for frequencies below ~100Hz or so. In a nutshell, you cannot equalise a room, because
most of the problems are caused by anomalies in time, and you cannot correct time with amplitude.

Fixed EQ is also used in smartphones, tablets and laptops, usually both for the inbuilt microphone and
speakers. The amount and type of EQ depends on the manufacturer, but it's safe to say that it will usually
be done using DSP. Some may allow applications to disable the microphone EQ (and compression) for
wider frequency and dynamic range. Another form of fixed EQ is a notch filter, and these can be
extremely narrow and used to remove an unwanted frequency. An example is the 19kHz notch filter used
in FM receivers to suppress the 19kHz pilot tone that's used for stereo broadcasts. Notch filters can also
be used to remove 50/60Hz hum from a signal without greatly affecting nearby frequencies.
The primary purpose of this article is to describe user adjustable controls, not fixed EQ systems.
Therefore I shall not delve into the realm of fixed equalisers other than in passing.

2 - Tone Controls For Reproduction

The early forms of boost/ cut tone control circuits were passive, and had a significant insertion loss.
Because there was no active circuitry in the circuit itself, in order to be able to boost the bass or treble, the
overall signal was attenuated. Simple filter circuits allowed the end user to independently set the bass
and treble controls to obtain a sound that was pleasing to the listener. Accuracy was never a
consideration, and the setting used was purely subjective.

Probably one of the earliest use of equalisers for audio was to try to get decent (and intelligible) sound
from early movie soundtracks [ 3 ]. It's not known if there were any equalisers used for radio broadcast,
but I'd be surprised if at least some form of (perhaps fixed) filtering wasn't applied to compensate for
deficiencies in the transmitter modulators and other parts of the transmission chain. There was definitely
a requirement to limit the bandwidth, because AM transmission cannot be allowed to be full frequency
range due to the problem of potential adjacent station interference. These don't qualify as tone controls
though, because they had fixed frequency response. The same applies to 'equalisers' used to correct
phone-line transmissions.

The top-cut style of tone control was standard on most mantel radios and even record players up until the
late 1960s. In the valve era, it wasn't possible to include 'proper' tone controls in budget equipment
because valves were expensive, and at least one triode was needed to bring the signal back to normal
level. Although there were many 'high end' hi-fi systems and construction projects published in Wireless
World (UK, now Electronics World), R,TV&H in Australia (Radio, Television & Hobbies) and Practical
Electronics (US) and many other magazines, only the more affluent enthusiasts could afford the off-the-
shelf equipment that had the latest and greatest tone controls (and other specifications to match).

There was a period where the best equipment available was expected to have tone controls. The Quad
22 preamp was an example, and that had quite sophisticated controls, featuring bass and treble as
expected, but also having a switchable low pass filter (5kHz, 7kHz and 10kHz) to help reduce noise from
the signal source. At that time (1950s to 1960s and beyond), nearly all preamps had tone controls, and
many innovative new topologies were developed to provide more control over how the controls
functioned. Some allowed for quite radical amounts of boost and cut. Up to ±20dB wasn't unheard of, but
most were limited to a more sensible ±12dB or so.

When graphic equalisers were first introduced to home hi-fi systems they were usually very basic. Some
had as few as five bands (2 octave range), and although quite limited gave the home listener plenty of
scope to mess up the sound. However, if the end result made the owner happy then that's all that really
matters. With most systems today, the inclusion of DSP (digital signal processing) allows the user to
select any number of 'effects' that can ruin everything with far greater ability than anything that has come
before.

Most simple tone control circuits use the simplest type of filter - resistance/ capacitance (RC) networks
that provide a theoretical maximum slope of 6dB/ octave. Those using capacitors and inductors (real or
simulated) can achieve far greater slopes, but are configured as band-pass or band-stop (depending on
the pot position). Graphic equalisers come in two major formats too, with the most common types
providing a variable Q (bandwidth) depending on the amount of boost and cut. The other type is 'constant
Q', patented by Ken Gundry of Dolby Laboratories and further developed by Rane. These have a (more
or less) constant bandwidth regardless of the amount of boost or cut.

The Langevin Model EQ-251A was the first equaliser to use slide controls, but in this case they were slide
switches, not pots as we expect today. It used only passive sections, and each filter had switchable
frequencies and used a 15-position slide switch to adjust cut or boost. The first true graphic equaliser
was the type 7080 developed by Art Davis's Cinema Engineering. It featured 6 bands with a boost and
cut range of 8dB. It used a slide switch to adjust each band in 1 dB steps. Davis's second graphic
equaliser was the Altec Lansing Model 9062A EQ. In 1967 Davis developed the first 1/3 octave variable
notch filter set, the Altec-Lansing 'Acousta-Voice' system.

3 - Tone Controls For Production

It's important to understand that there is a vast difference between the tone controls that may be used on
a hi-fi or mixing console and those used in guitar or other musical instrument amps. In the hi-fi or mixer
cases, it is essential that a flat response is available, simply by setting the boost/ cut controls to centre.
The circuit then has no effect on the response, so what goes in comes out without change. With musical
instrument amps, the situation is very different. The tone controls work in conjunction with the instrument,
pickups and the loudspeakers, and the overall effect is to provide a wide range of 'tones' through the
speaker that are pleasing to the musician.
For example, a guitar amp is not intended to reproduce sound, it's intended to create (produce) sound.
The amplifier and speaker system form part of the instrument - any one without the other is pretty much
useless. Try playing a well liked recording through a guitar amp - you will never get it to sound right.
Much the same happens if a guitar is played through a hi-fi system. Even if it has tone controls, it will be
difficult or impossible to get 'the sound' that a guitarist is used to hearing, and you'll probably end up with
blown tweeters to add injury to insult (as it were).

Early guitar amplifiers often had no more than a 'top cut' tone control, but users wanted more. The 'tone
stack' as it's generally known now was developed fairly early, but despite much searching I was unable to
find out who designed the first version. The guitar amp style tone stack is only capable of providing bass
and treble boost (which equates to a midrange cut). The midrange control only lifts the average level
across the frequency range, and is deliberately limited so it doesn't render the bass and treble controls
inoperative. In most designs, there is no setting that has a flat frequency response - all you can do is vary
the amount of bass and treble boost. These circuits are always passive, and have an insertion loss of
20dB or more. Insertion loss simply refers to the amount of signal you get at the output vs. the input, with
the controls set to flat or the closest to 'flat' that the circuit can provide.

A few designers over the years have used Baxandall (feedback) tone controls in guitar amps (often as
magazine projects), and most qualify as bloody awful at best, unusable at worst. This isn't to say that
they can't be used, but in general guitarists will not be at all happy with the end result. To anyone who
has designed a guitar amp or two (or three, or ...) this comes as no surprise. Music production and
reproduction are very different, and cannot be considered equal in any way. While electric guitar can be
especially hard to get right, bass guitar and acoustic guitar with magnetic or piezo pickups can also be
very demanding.

4 - Basic Tone Controls

The simplest, most basic and least useful tone control simply provides bass and/or treble cut. These are
easily created and were very common in many earlier wireless sets. Bass cut wasn't so common, but
nearly all mantel radios from the 1940s onwards featured a 'tone control', which was nothing more than a
variable treble cut. By varying a pot, the high frequency response could be rolled off to allow the user to
obtain a 'mellow' sound that had a very restricted top-end. Even from an early age, I found that setting
the tone control to the position that gave the most treble (such as it was with an AM mantel radio or
similar) was far more satisfying than the muffled sound that my parents seemed to prefer.

The general principle is shown below. No boost was possible for bass or treble, simply because early
radios and record players barely had enough gain to reach full volume even without any tone control, so
reducing the gain to allow boost for separate bass or treble controls wasn't an option. Gain was
expensive, because it required another valve stage. The important part here is that if you want to be able
to boost bass or treble with a passive network, the entire signal has to be reduced so the filters can be
adjusted to provide an apparent boost. Simple bass and treble cut controls are shown below, as these
are the most basic of all.

Figure 1 - Bass And Treble 'Cut' Controls

These controls have the minimum possible effect on the rest of the signal, so they could be added without
any gain penalty. This meant that an additional valve or transistor wasn't needed, so the cost of including
them wasn't great. A couple of potentiometers, knobs, resistors and capacitors was all that was needed.
With both controls set for maximum cut, the effect was to provide a signal that was all midrange - no bass,
no treble, only the mid frequencies. However, if the two are combined there will be some interaction.

Note that as the controls are adjusted, they can only cut - there is no facility to boost the signal at any
frequency. The treble cut control reduces the level by 6dB/octave from a turnover frequency determined
by the pot position and the bass cut control does the same. Treble control can also use a variable
capacitor, but that was never appropriate because of the physical size of a variable capacitor with enough
capacitance to be useful. It can be done easily with a capacitance multiplier, but these were never used
in the valve era and remained uncommon until opamps became readily available. With the values shown,
the -3dB frequency response with both controls set for maximum cut is from 177Hz to 2kHz. With the
pots set for minimum cut the response is essentially flat from 30Hz to 20kHz. The circuit must be followed
by a high impedance stage and fed from a low impedance.

If you need to apply boost at any frequency, you need to accept a loss that's slightly greater than the
boost allowed or incorporate a gain stage. This can be a valve, transistor, FET or opamp, depending on
the era of the design. Early cut/boost tone controls were passive and could introduce a loss of as much
as 20dB with the pots centred (flat response). This loss had to be made up by adding a gain stage.

The general scheme seen below is often referred to as a 'James' EQ, so called because it was first
published by E.J. James [ 1 ]. You may also see it referred to as a 'passive Baxandall', but that's not
correct. The design published by Peter Baxandall is active, and uses feedback to get symmetrical boost
and cut. The Baxandall tone control requires an inverting amplifier stage with low output impedance to
drive the filter circuits. The James circuit requires a low source impedance and high impedance load, or
performance will suffer.

Figure 2 - Passive Bass And Treble Cut/Boost Controls

There are countless variations on this basic circuit. As shown, it's one of the more common
arrangements and allows a nominal cut and boost of around +18dB and -20dB (it's not perfectly
symmetrical). The bass and treble turnover (±3dB) frequencies are changed by using different capacitor
values. Smaller caps work at higher frequencies. The bass section can use one capacitor (in parallel
with VR1) or two as shown. The treble section may also use two caps as shown, vs. a single cap in
series with the wiper of the treble pot.

There is a slight difference between the circuit variations. Tone control circuits must be driven by a low
output impedance (cathode or emitter follower), and there is some interaction between the controls with
most passive versions. A true flat position is difficult to achieve with the Figure 2 controls, and a
frequency deviation of up to ±2dB is not uncommon. Note that the pots are logarithmic - linear pots do
not work, but log tapers are rarely good enough to ensure front panel calibration for flat response.
Insertion loss is about 20dB. The following stage must have a high impedance input, and direct coupling
to the grid of the following valve (with no additional grid resistor) was not uncommon. A gain of 10 is
needed to restore the level with the controls set for a nominally flat response.

Figure 3 - Restricted Range Passive Bass And Treble Controls

The above shows a very simple EQ circuit that I devised a great many years ago for simple stage mixers
and 'pre-mixers'. The idea was to provide some control, but not so much that it would get inexperienced
users into trouble. The basic scheme is superficially the same as that shown in Figure 2, but the
components are the same value for the 'top' and 'bottom' parts of the circuit (compare this with Figure 2).
The insertion loss is small (6dB with the controls centred), and the maximum boost is limited to a little
under 6dB. There is more cut available, but that only becomes apparent with the control(s) set for
minimum bass or treble cut.

Response of the bass pot is shown in green, and treble in red. The pots are linear, and graphs are shown
at 25% increments. Unlike the version shown in Figure 2, when the pots are centred the response is
completely flat, with almost no deviation at all. There is a small deviation that can be measured, but it's
below audibility (about 0.3dB with a 100k load, or 0.03dB if loaded with 1 megohm).

Figure 4 - Restricted Range Passive Control Response

Interestingly, the Figure 3 circuit is almost exactly what you'd expect to see used with an inverting gain
stage in a Baxandall control circuit [ 2 ]. The same values used with an inverting gain stage give perfectly
symmetrical boost and cut, with a maximum of ±15dB with the values shown. This type of control is
shown next, and was very common in home hi-fi systems and mixing consoles. The circuit is seen below,
using the exact same component values as shown in Figure 3, but with the addition of an opamp gain
stage.

Figure 5 - Baxandall Active Tone Control

This type of circuit is possibly the most popular of all time. Some manufacturers have provided switchable
capacitors so the response can be tailored to the user's preferences. There are variations with a
midrange control, which is achieved by adding a third pot that has a cap in parallel (like the bass control)
and another (smaller) cap in series with the wiper (like the treble control). When a midrange control is
included, it's almost always fixed - to make it variable requires switched capacitors.

The generic term for equalisers with the type of response provided by James and Baxandall tone controls
is 'shelving EQ', because the bass or treble is boosted by a set amount, but then returns to being almost
flat above or below a frequency that's determined by the setting of the control pot. You can see this in
Figure 4, the boost and cut level out below 200Hz and 4kHz. Because the Figure 3 circuit is passive and
has no feedback, at maximum cut the bass doesn't level out until about 60Hz, and the treble doesn't really
level out at all. Once feedback is applied, this changes as shown in Figure 6.
 Figure 6 - Baxandall Active Tone Control Response

Colours and pot increments are the same as used for Figure 4. You will notice that boost and cut are now
(almost) perfectly symmetrical. Remember that these plots used the exact same tone filters as shown in
Figure 3, and the only difference is the addition of feedback.

The full performance and symmetry of Baxandall circuits was difficult to realise with valve circuitry,
because getting a very low output impedance from the drive and feedback stages was extremely difficult.
As is common with all valve circuits, the tone control networks were high impedance, using 100k or higher
for pots, and with other components scaled to suit. It became easier when transistors were used, and
was virtually automatic when opamps were used as the source and amplifying devices. Perhaps some of
the nostalgia for valve circuitry was the rather 'sloppy' response obtained due to relatively high
impedances. This can be restored (why?) by adding resistors in series with opamp outputs.

It's to be expected that some people will insist on passive controls, because they imagine that applying
feedback somehow ruins the sound. This is complete nonsense of course, and there seems little point in
using a vastly inferior tone control system that has no real flat setting just to avoid the 'evil' of feedback. If
this approach is taken, only the Figure 3 circuit is really suitable, because a flat setting is possible and
dubious (at best) log pots are not needed.

Figure 7 - Baxandall Active Tone Control With Midrange

In the interests of completeness, the above shows the general arrangement used to add a midrange
control to a Baxandall network. The Q is low (about 0.5) and you can't adjust the frequency, but it does
add some extra functionality that might be useful for a musical instrument amp. While you may see it
added in many circuits on the Net, it's of somewhat dubious value. Because it's not easily adjusted for
frequency and due to the low Q, most users are likely to find it doesn't really do what they need.

5 - Graphic Equalisers

While the basic shelving filters described above are fine for controlling bass and treble, to affect the
midrange or a troublesome frequency anywhere in the audio band isn't possible. In many cases bass and
treble controls don't even work for bass and treble. For example, if you want to get a 'fat' kick drum sound
you might add some bass, but you don't want or need to keep boosting all the way down to a few Hertz.
Look at Figure 6 - if you have 10dB of boost at 70Hz, you have slightly more than that at 40Hz and it's still
there at 20Hz. A peaking filter can be tuned to 70Hz (for example) to give a satisfying 'thump' from the
kick drum, but the level returns to normal (towards 0dB gain) as the frequency increases or decreases.
Graphic equalisers have a series of bandpass filters, with each frequency band controlled by a slide-pot.
Each frequency can be cut or boosted, and uninformed fiddling can cause problems. There was a brief
period where stereo graphic EQ was considered a 'must' for what's probably better known as 'low-end hi-
fi' - comparatively cheap systems that made up for the lack of overall quality by including extras that made
the buyer believe s/he was getting a good deal.

This general form of equaliser was developed in the early 1970s, and inductors were used as part of the
frequency selective networks. Inductors are comparatively large, require many turns of wire and a
magnetic core (steel laminations or ferrite). They are expensive to make, and nearby magnetic fields can
induce hum into the windings.

Graphic EQ was therefore expensive and quite bulky until the invention of the gyrator (a 'simulated'
inductor, using an opamp to invert the action of a capacitor). Although the gyrator was proposed in 1948
(by Bernard Tellegen, a Dutch engineer who also invented the pentode valve), practical realisation wasn't
possible until opamps became readily available. Very basic gyrators can be made using only a transistor,
but their performance is sub-standard. I don't know of anyone who has tried to make a gyrator using
valves because it would not be sensible. The active element of a gyrator is a non-inverting unity gain
buffer, which should have high input impedance and low output impedance.

Gyrators allowed designers to create large numbers of 'inductors' very cheaply compared to true
inductors, and gyrators are unaffected by magnetic fields so induced hum was no longer a major
problem. The general form of a graphic equaliser is shown below, but using inductors for clarity. It
doesn't matter if the inductor is 'real' or simulated, it has exactly the same effect. Note that the value of
the resistor (R2, R3, etc.) is often the winding resistance of the inductor, and/or an external resistor used
to ensure that the series resistance of each tuned circuit is identical. In the following drawing, only the
first 5 octave band filters are included. The remainder follow the standard octave frequencies. Industry
standard frequencies for the three most common equalisers are ...

31 63 125 250 500 1k0 2k0 4k0 8k0 16k

Octave Band Frequencies - 10 Band

31 44 63 87 125 175 250 350 500 700 1k0 1k4 2k0 2k8 4k0 5k6 8k0 11k 16k 20k
1/2 Octave Band Frequencies - 20 Band

25 31 40 50 63 80 100 125 160 200 250 315 400 500 630 800 1k0 1k2 1k6 2k0 2k5 3k2 4k0 5k0 6k3 8k0 10k 12k 16k 20k
1/3 Octave Band Frequencies - 30 Band

The frequencies shown above are pretty much agreed upon worldwide, and have been adopted by all
manufacturers making graphic equalisers. The 1/2 octave and 1/3 octave frequencies are often extended
above and below those shown, and may include 20Hz and/or 25Hz, as well as 20kHz. The drawing
below shows ideal values rather than those readily available, purely for convenience. The Q of each filter
is about 2, extreme accuracy is not really possible and fortunately isn't necessary. The circuit below must
be driven from a low impedance. Normally, there would be a unity gain buffer to drive the input. It isn't
shown but must be included unless the previous stage is an opamp or other very low impedance source.

Figure 8 - Graphic Equaliser General Scheme, Using Inductors

Without the frequency selective networks (C1, L1, etc.), the pot sliders simply vary the gain of the circuit
and unity gain is achieved when the slider(s) are centred. When the pot wiper is close to the input (+ve
input of U1), the incoming signal is attenuated (cut), and at the opposite end (shown with a + sign) the
opamp has gain (boost). When each pot connects to a tuned circuit, only the frequencies passed by the
tuned circuit are affected.

The tuned circuit filters have a minimum impedance at a particular frequency as shown, so the pot affects
only those frequencies passed by the filter. A series resonant circuit has a minimum impedance at the
resonant frequency, and this forms the basis of most simple graphic equalisers. The scheme shown
above gives an equaliser whose actual Q (as opposed to the theoretical value) varies with the slider
setting. At low boost or cut settings the bandwidth is much wider than expected.

One thing that is fairly difficult to find explained in simple terms is just how to determine the inductance
and capacitance needed for a specific Q. The values depend on the load (series) resistance, which in the
above circuit is 470 ohms. The impedance (X) of the cap and inductor must be scaled to the load
resistance (RL), and the following formulae apply ...

X = RL × Q
C=1/(2×π×X×f)
L=X/(2×π×f)

The Q (which determines the bandwidth) of each filter depends on the number of sliders used (10, 20 or
30). A 1/3 octave graphic EQ needs higher Q filters than a 1 octave band type. Q is defined as the
centre frequency divided by the bandwidth, and a 1 octave filter requires a Q of 2. A 1/3 octave EQ
system needs filters with a Q of 4.31 (4 is close enough for an equaliser). You may well ask why the Q
isn't constant, and the answer is quite simple.

When the pot is near the centre position, the load on the tuned circuit is no longer 470 ohms, it's 470
ohms plus the equivalent resistance of the pot and the feed resistors (2.7k as shown). As the pot position
varies, so does the Q, and therefore the bandwidth changes as well. This type of circuit cannot provide a
constant loading on the tuned circuit, so cannot provide a constant Q. The effect can be reduced slightly
by using lower value pots, but that increases the noise gain of the opamp, so the circuit will become noisy
even with a low noise opamp. The opamp inputs are not virtual earth types, and both have a relatively
high impedance.

A constant-Q graphic equaliser suitable for subwoofer equalisation is described in Project 75, and this
arrangement was first published by Bob Thurmond [ 5 ] and is shown next. Commercial units were
pioneered by Rane [ 6 ], but using a different circuit.

Figure 9 - Constant Q Graphic Equaliser (One Section Shown)

It's important to understand how this circuit differs from the previous version. The most obvious difference
is that the opamp inputs are both virtual earth (close to zero impedance), and the band pass filters are not
RLC types as shown above. You may see a variety of different active bandpass filters used. Typical
types are multiple feedback, twin or bridged tee or even state-variable. It is possible to use RLC filters
(resistance, inductance, capacitance), and gyrator based filters can be used with some extra circuitry, but
the other filter types remain a simpler and better choice.

You can see that the pots control the output from each band-pass filter (BPF), and the multiple outputs
are summed along with the input signal at the input to U1 (signal cancellation or cut) or U2 (signal
augmentation or boost), depending on the pot position. When the pot is centred, the signal to U1 and U2
is identical, so it cancels and there's no boost or cut for that frequency. The Q remains constant because
it's only the output from the appropriate BPF and the load doesn't change.
 Figure 10 - Variable Q Vs. Constant Q Response

Above you see the response of two equalisers, one configured as a traditional graphic EQ and the other
configured for constant Q. Assuming equal bandwidth for each, both will have the same response at
maximum boost or cut, but the situation is quite different at any setting below maximum. The setting for
constant Q vs. variable Q is shown for a pot setting of 75% (50% boost). Cut response is the same (but
results in a dip of course).

Figure 11 - Gyrator And Band Pass Filter

The general topology of a gyrator and band-pass filter are shown above. The effective inductance of a
gyrator is simply the product of the three components (R1, R2 and C1). When the three are multiplied
together, the answer is the inductance in Henrys. Equivalent winding resistance is the value of R1.
Gyrators are covered in greater detail in the article Active Filters Using Gyrators - Characteristics, and
Examples. The multiple feedback bandpass filter is a simple and fairly straightforward design, although
calculating the values can be very irksome. They are described in detail in Project 63, and there's even a
calculator program available that you can use to work out the component values for you. The multiple
feedback filter is not easily tuned, and when variable frequency is needed the choice is between the
various bandpass filters covered in the section about parametric equalisers.

While a graphic EQ is certainly a very flexible way to control the frequency response, they need a large
number of slide pots and therefore take up a lot of room. This makes their use on mixing consoles (for
example) limited to perhaps a couple of graphic EQs for the main or 'FOH' (front of house) outputs. There
simply isn't enough space on each channel strip to include one for each channel, and simple bass and
treble controls are too limited.

To get the benefits of EQ that can be tailored to the exact needs that doesn't occupy too much space on a
channel strip requires a parametric equaliser, discussed below.

6 - Variable Frequency Tone Controls

Simple bass and treble controls can benefit from having adjustable frequencies. It's no longer possible to
use the Baxandall topology, so it's done using various other techniques. The easiest is to use the same
basic arrangement as used in common graphic equalisers. There have been many schemes used, but
most use a variable frequency high and low pass filter in a feedback network. A few (including some that
I designed) use an opamp to create a variable capacitance (a capacitance multiplier), and others have
used a variety of circuits. It would be silly to try to include them all, so only two variants are shown.
The first is fairly conventional, and there are quite a few references to very similar circuits on the Net. The
circuit consists of two inverting gain stages and two unity gain buffers. The latter isolate the boost and cut
controls from the frequency networks, and are essential to prevent unwanted interactions. VR1 changes
the bass frequency from 200Hz to 740Hz at the ±3dB point. VR2 does the same for the treble, from
460Hz to 1.4kHz, again at the ±3dB frequencies with full boost or cut.

Figure 12 - Variable Frequency Bass & Treble Circuit

The above circuit works well and is not critical, and component values can be changed to increase the
frequency range or provide more (or less) boost and cut. As a general purpose tone control, it's far more
flexible than the standard Baxandall circuit, but gives almost identical results for any given frequency
setting. A better solution uses a variable gyrator for the low frequencies and a variable capacitance
multiplier for high frequencies. This has the advantage that the bass control can be switched from
shelving to peaking, and additional sections can also be added.

Figure 13 - Variable Frequency Bass & Treble Circuit #2

The gain and boost/cut circuitry is identical to that used for a conventional graphic EQ, and is easily
expanded as described in the Project 28 page. As a parametric equaliser it's not wonderful, but it's still
surprisingly effective. If you only want variable frequency bass and treble controls it's better than the
circuit shown in Figure 12 because you get the option of shelving or peaking for the bass control. As
noted above this can be especially useful for percussion (kick drum, toms and kettle drums for example).
The circuit uses three unity gain buffers and one gain stage. The input buffer is not needed if the source
has a low output impedance, such as from another opamp in the circuit.

Be aware that the variable capacitance multiplier (U3) can be temperamental. On occasion, it may not
settle properly to normal quiescent conditions (output at zero DC voltage), and it might need to be
powered off and on again before it settles down. I've been unable to replicate this on the workbench, so it
seems that the circuit knows when test equipment is nearby . Mostly it works perfectly - I have one in
an equaliser I use for my workshop system that's never missed a beat in over 20 years.
In shelving mode, the circuit works almost identically to that shown in Figure 12. The range of each
frequency control can be changed by using a higher (or lower) value pot, and the frequencies are
changed by replacing C1, C2 and/or C3 with values that provide the desired ranges. For the peaking filter
section (C1 in circuit), the ratio of C1 and C2 determines the resonant circuit Q (C2 determines the
inductance of the gyrator). Normally there is an optimum ratio (typically around 10:1) for C1 and C2, but
because the inductance is variable vis VR3, the optimum ratio can't be maintained.

There is one thing that the Figure 13 circuit does that is not especially desirable, When in peaking mode,
the Q changes depending on the setting of VR3. At very low frequencies the Q is higher than at higher
frequencies. This variable Q is either a benefit or a curse, depending on what you want to do. With the
values shown, the Q ranges from 9.5 to 2.0 (at maximum boost or cut, and at 35Hz and 150Hz
respectively). At settings below the maximum cut or boost the Q is reduced. It's normal for this type of
equaliser, and if you need a circuit that has consistent Q you need a proper parametric EQ as described
next.

7 - Parametric Equalisers

The most flexible EQ that occupies the least space is a parametric equaliser. Provided the bass can be
switched from shelving to peaking mode (and many can), you can insert a peak or dip anywhere you like
to get the sound you want. Parametric EQ ranges from simple fixed bandwidth types (such as the one
shown in Project 28) through to fully variable 'true' parametric equalisers based on state-variable filters.
Simple versions like the P28 circuit provide no control over the bandwidth (Q), but are nonetheless very
flexible and can perform most 'sound-shaping' EQ tasks very well. Variable Q is needed if you happen to
have a requirement to notch out a particular troublesome frequency. High Q (narrow bandwidth) peaks
are rarely needed and if used can create problems with the final mix.

It's almost unheard of to use parametric EQ for a home system. These equalisers are not easy to drive,
and should only be used by those who understand how they work and what they do. If adjusted
incorrectly it's quite possible for an inexperienced user to not just mess up the sound, but it may be
possible to kill tweeters if a substantial boost is applied close to the crossover frequency, so the tweeter
receives too much energy at its lowest recommended frequency. Home systems aren't just operated by
adults, and kids like to experiment! Hi-fi manufacturers assume (not unwisely) that the average user
would be confused by all the options provided, and most 'high-end' equipment offers no form of tone
control at all.

As with graphic equalisers, a parametric EQ can be configured for variable or constant Q. Each requires
a different approach to the circuit. There are countless variations for parametric equalisers, but the best
all-round filter network is the state-variable topology. This is a relatively complex circuit, but has the
advantage of being easily adjusted both for frequency and Q. Demands on the opamps are fairly modest
and comparatively cheap opamps can perform well.

Figure 14 - Wien Bridge Based Parametric EQ

A simpler version uses a Wien bridge as the variable frequency element. These really qualify as 'quasi
parametric' EQ, because the Q is fairly low (around 1.3) and can't be changed. However, they are well
behaved and easily tuned. A variable frequency stage needs only one opamp in its simplest form, and
the tuning network is completely passive. It might seem unlikely that this would be useful, but the Q is
actually much greater than a 3-band Baxandall tone control (which only manages a Q of about 0.5), and it
can be tuned with a dual-gang pot. This network is not suitable where high levels of boost or cut are
needed, as the circuit will oscillate if the gain is too high (set by R6). Without R6 the maximum boost and
cut is 9dB and with the values given the maximum is 12dB. The performance can be improved a little by
adding a buffer between the pot and the Wien bridge network, but in general the benefit does not
outweigh the added expense. There's a lot more info on this topology in Project 150. The Wien bridge
network consists of VR2 (A & B), R2, R3 and C1, C2.

Most 'true' parametric equalisers use a state-variable filter (see State-Variable Filters for a detailed
analysis). Although comparatively complex, the state variable filter gives independent control of Q and
frequency. There are many variations on the scheme, but the end result is fairly similar. In the following
drawing, the control section is identical to that shown in Figure 14, and the filter is simply changed from a
Wien bridge to a state-variable.

Figure 15 - State-Variable Parametric EQ

VR3 controls the filter Q without affecting the gain, and VR2(A & B) controls the frequency. With the
values given, the frequency range is exactly the same as the EQ in Figure 14, because the values that
determine the frequency are the same. The Q can be varied between 5.3 down to 0.5 which gives a very
wide control range. Note that VR1 (cut/ boost) operates opposite to the way it does with the Wien bridge
circuit. As shown, boost and cut are limited to 9.5dB, but this can be extended by adding a resistor from
the inverting input of U1 to earth. If a 2.7k resistor is added, boost and cut are increased to 12dB.

Parametric Equalisers come in multiple types, and usually include variable frequency bass and treble
controls, along with one, two or sometimes three bands of true parametric. Frequency ranges usually
overlap, and care is needed to ensure that boost isn't used with two sections tuned to the same
frequency. There is always a chance that the equaliser will clip, or the output at one frequency will be so
high as to place horn compression drivers (in particular) at risk of damage.

When very high Q is used, it's generally only needed to cut a troublesome frequency. High Q boost is
rarely needed other than for a special effect. Because the parametric EQ is so flexible, it takes some time
to get used to using it properly. Most DAW (digital audio workstation) software includes digital parametric
EQ, and there are some on-line tutorials [ 7 ] that explain how the EQ should be used. One of the general
tenets of parametric EQ is to "cut narrow, boost wide", referring to the Q or bandwidth of the filter(s). A
high Q notch can be very useful, but boost should normally be low Q and kept to the minimum whenever
possible. A high-Q boost will almost certainly cause feedback in a live sound system, and can easily
damage high frequency drivers.

8 - Guitar Amp Tone Stacks

The 'tone stack' as it is commonly known is only suitable for guitar or other musical instrument amps. It's
very difficult to know where it came from (opinions abound, but proof is hard to come by), but tone stacks
are used by most guitar amp manufacturers almost exclusively. The arrangement is quite different from a
traditional passive control network, and the control pots are wired in series to form a 'stack' (hence the
name). They are very economical, and use the minimum possible number of parts, but the controls are
usually highly interactive and there is almost always a significant midrange 'scoop' (essentially a broad
notch).

Since electric guitars in particular usually have a quite prominent midrange with little bass or extended
treble, the midrange scoop makes up for that by boosting the bass and treble and suppressing the
midrange. Varying the bass and treble controls shifts the notch or 'scoop' centre frequency and its depth.
Where a 'midrange' control is included, the closest to flat response is obtained with bass and treble at
zero, and midrange at maximum. A true flat response is usually impossible though. The controls are
used to get a guitar sound that suits the player, and the tone controls (as well as the speaker, cabinet and
power amp) are used to create sound. The amp has to be considered as part of the instrument, as most
guitarists will choose an amp based on the overall sound they can get from the pairing of guitar and
amplifier, and linked to their playing style.

There are very wide differences between tone stacks, not only between different manufacturers but even
between different models from the same maker. Most are high impedance and are designed for use with
valve stages. For best performance they should be driven from a cathode follower, but in some cases
even that is abandoned. While guitarists will think that the tone stack in their favourite amp is a work of
art, they are really very basic and usually don't work well with sources other than guitar. From a
manufacturing perspective, these are the cheapest possible options for tone control, but it just so happens
that the characteristics are pretty close to ideal for the purpose. Only a few designers have strayed, and
those who were silly enough to try using Baxandall (active) tone controls have never been well received.
An Australian magazine once published a guitar amp using an active tone control, and it didn't go down at
all well with most experienced guitar players.

Figure 16 - Tone Stacks, Fender & Marshall

The two tone stacks shown above are typical only. There are many variations between different models,
but all have fairly similar overall characteristics. There is no true flat setting, and the midrange control
only reduces the depth of the notch, the frequency of which varies with control settings. The controls are
interactive, so changing the treble will change the notch frequency and affects the bass to some degree.
The bass control has less interaction with treble and mid, and the mid control is fairly subtle.

Insertion loss with all controls centred is much greater with the Fender style (average about 15dB) than
the Marshall (about 8dB), and the Fender circuit has more boost for both bass and treble. The notch
(which varies from around 300Hz to 1kHz for both) is deeper in the Fender circuit. There is no doubt that
the two circuits will sound quite different, but that doesn't mean that a good guitar sound can't be obtained
from the two. Many guitarists have a preference, but that's often because a particular amp brand is
preferred. There are many other guitar amps, and they nearly all use variations of the two circuits shown.
It would be silly for me to even try to show all the different circuits because there are so many.

Figure 17 - Tone Stack Response, Fender & Marshall

The two response graphs shown above are with the controls set at 50%. Because there's often a mixture
of linear and log pots, this doesn't relate directly to any knob setting. The midrange scoop is clearly seen
in both traces, and this is one of the main features of tone stacks in general. I don't know of any stack
that has eliminated the midrange scoop. Only the frequency and depth change.
These controls are easily modified by changing cap values. There is no design process involved, it's
purely a case of trial and error, and ultimately it's all about getting the desired sound. What the controls
actually do to the response is secondary to what it sounds like. If it does what the player needs then it's
good, if not ...

9 - Frequency 'Isolators'

This type of equaliser is almost only ever used by DJs, and it's quite common in DJ mixers. You will rarely
see it elsewhere, but if you were to build a 4-way active system based on Project 125 (a 4-way active
crossover) you get this ability free. A frequency isolator is usually simply a 3-way crossover network with
its outputs summed to return to a flat response. Project 153 describes a 3-band, 12dB/octave, variable
frequency isolator, and if you want to see the full version please refer to the project article. The version
shown below has fixed frequencies, and although this may seem quite limiting it's often as much as you
are likely to need. The term itself is something of a misnomer, in that you can't really isolate the
frequency bands because they have a finite rolloff. You can use 24dB/octave filters (as found in Project
125), but that's generally not necessary to get the effects needed.

To be able to get a flat response without having to bypass the equaliser, the filters must use a Linkwitz-
Riley alignment. If Butterworth filters were to be used, there would be +3dB peaks at each crossover
frequency (410Hz and 4.1kHz) when the pots are all set the same.

Figure 18 - Typical Frequency Isolator

The circuit is simply a 3-way crossover, with the outputs summed. When all pots are set to the same
level, the summed output is flat, and the pots let the user turn the level of any band up or down. As
shown the frequencies are 410Hz and 4.1kHz, but they are easily changed by changing cap values. The
multiple feedback filter (U2) used in the midrange circuit reduces the opamp count. Because it's an
inverting stage it means that a separate inverter isn't needed for the midrange. It also is a nuisance
because the caps are not the same as those used in the high pass filter (U1) and changing frequencies is
more difficult. The alternative is to use a Sallen-Key filter like all the others, followed by an inverter.
 Figure 19 - Frequency Isolator Response

With the circuit shown, the gain is 0dB with the pots all set for 50%, and the summed response is flat to
better than 0.1dB. The summed response is shown above with bass and treble at 50% and midrange at
zero. This is just an example using fixed frequencies, but of course there are many other possibilities.
This type of equaliser is not intended to correct the frequency response, they are used by DJs as an
effect.

10 - 'Tilt' Controls

Finally, there's one last tone control arrangement that was popular for perhaps 5 minutes or so, sometime
in the 1970s. It was used in at least one Quad preamp as well as a couple of others, but it died out fairly
quickly because it's not really very useful. The effect was to literally tilt the frequency response, so if the
bass is boosted, the treble is simultaneously reduced and vice versa. I'm not entirely sure why anyone
thought this was a good idea, but it's part of tone control history, so is included. There are many possible
tweaks that can shift the centre frequency or provide asymmetrical response, but these are generally as
useless as the circuit itself.

Figure 20 - 'Tilt' Tone Control

The circuit is straightforward, and uses a frequency selective network wired in reverse phase for high and
low frequencies. When one end of the spectrum is boosted, the other end is cut. When the pot is
centred, response is flat. The following response graph shows the response at 25% intervals of the pot.
Despite not being very useful overall, there are quite a few different versions on the Net. All behave
more-or-less equally, but the Quad version was limited to ±3dB unlike most you will see (including the one
shown). To reduce the range, resistors are used in series with each end of the pot (VR1).
 Figure 21 - 'Tilt' Tone Control Response

The circuit would be more useful (or maybe less useless) if the range was restricted to perhaps 6dB of
maximum boost or cut, but the same thing can be done with more conventional tone controls, and that
allows bass and treble to be boosted (or cut) by different amounts to balance the overall sound. As noted,
only a few manufacturers decided to use this type of EQ, and it was short lived - presumably because the
buying public didn't like it. I expect it seemed like a good idea at the time, but it's really a rather pointless
waste of parts. As you may have gathered, I don't recommend it.

Conclusions

It's fair to say that with the ready availability of opamps, tone controls with greater flexibility and more
usable features became possible than were ever available before. When a modern design is set for flat
response, there is virtually no change to the signal at all, other than a truly tiny amount of added noise
and distortion which is inevitable with any active circuit. Earlier designs could also be fairly flexible, but at
the expense of many components (including inductors), and frequencies that could only be switched
rather than continuously variable.

When it comes to wide range, flexible EQ, opamp circuits simply cannot be beaten by any earlier
technology, despite any contrary claims you might hear. Using DSP is the next level, but there are still
many people who prefer to keep signals in the analogue domain if possible. Controlling a complex filter
using a touch-screen may be 'high tech', but it's often very hard to beat the feel of knobs on high quality
pots. Rotary encoders can be used with digital systems, but you usually lose the ability to see the
settings by looking at the knob pointers.

Analogue circuits have another major benefit - they can be built by anyone who can use Veroboard and a
soldering iron, or mount parts on a PCB. This isn't even an option for most digital systems unless the
person building the circuit can not only solder surface mount parts, but also knows how to program a
DSP. There's another disadvantage to the digital approach, and that's IC continuity. Many modern digital
ICs (DSPs, ADCs, DACs, etc.) have a short production life, so if the IC fails after a few years it may be
impossible to replace. In contrast, opamps have been with us for many years, and there's no indication
that any of the popular devices will disappear. Even if an opamp does become unavailable, you can be
sure that a suitable replacement with equal or better performance can be found easily.

Whether you like the idea of EQ or not, it's inevitable that it has been used during the production of the
original recording. There may be a very small number of tracks that have been created as direct to tape
or hard disk without any processing, but they will be few and far between. If such material is not a genre
you even like, then there's no point at all. In general, EQ will hopefully be applied only where necessary,
and preferably with as little change as possible. However, many producers will abuse your senses and
the recording by manipulating frequency response so that even more compression can be added without
turning the music to mush. Regretfully, this seems to be a popular pastime .

Phase response wasn't even mentioned in any of the descriptions, because it's extremely variable. All
equalisers cause phase shift, and the change of phase is much more rapid with a high Q circuit. We can
hear the frequency response variation caused by any equaliser, but the phase shift is not audible. There
is any number of people who claim that phase is audible, but the claim belies the fact that most
programme material has had at least some equalisation, and therefore has phase that varies from the
original either for particular instruments and/or for the complete mix. No double-blind test has ever shown
that phase shift is audible, provided it's static. Varying phase shift is used to create vibrato (cyclically
varying pitch) which is audible, and is used as an 'effect' with many electric musical instruments.

References
 1. Simple Tone Control Circuit - E.J. James, Wireless World, February 1949
2. Negative Feedback Tone Control - P.J. Baxandall, Wireless World, October 1952
3. Operator Adjustable Equalizers: An Overview - Dennis Bohn, Rane Corp.
4. Duncan's Amp Pages - Tone Stack Calculator
5. G.R. Thurmond, "New Devices for Equalization," 52nd Convention of the AES, (Abstracts), vol. 23,
p. 827 (Dec. 1975) preprint 1076
6. Constant-Q Graphic Equalizers - Dennis Bohn, Rane Corp.
7. How to Use a Parametric Equalizer

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