An Introduction to Equalization
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An Introduction to Equalization
Page Contents
An Introduction to Equalization
6
Frequency and Level
7
Level and decibels
13
Frequency Response
15
What is equalization for? What are we making ‘equal’ to what?
16
Filters
18
Passive EQ and tone controls
22
Mixing Console EQ
24
EQ IN button
29
Graphic EQ
30
Using Equalization
33
Corrective Equalization
34
Creative Equalization
37
Cut can be better than boost
40
Equalizing the mix
42
Equalization for live sound
44
Appendix - Extreme EQ from hardware equipment
46
Mutronics Mutator
47
Bonus Technique - Turn a low-pass filter into a high-pass filter
49
Conclusion
50
An Introduction to Equalization
Page An Introduction to Equalization
Equalization, or EQ, is one of the most basic yet most important tools
in recording, live sound, and all other activities of sound engineering.
Equalization is used to repair problems, to make individual instruments
and voices sound better, and to help instruments and voices blend together
in the mix. It is also used to improve the mix, and to make tracks on a CD
flow seamlessly from one to another without sudden changes of frequency
balance.
This text will take you through EQ from an understanding of frequency
and level at first principles, all the way to how to shape and control
frequencies in the mix, which is indeed both a skill and an art. When you
understand EQ, you will be able to start to apply it effectively.
Without understanding, you will be flying blind, making random changes
and not really knowing whether you are improving matters or not. But
when you do understand all the principles, all the techniques and all of
the options, you will start to be in control. Over a period of time, you will
master the art of EQ, as well as the science.
An Introduction to Equalization
Page Frequency and Level
One of the most important features of sound is frequency. Imagine one
string of a guitar. When plucked, it vibrates at a certain rate. This is its
frequency. The lowest string of a guitar vibrates approximately 82 times
per second.
We call ‘times per second’, ‘hertz’, named for the German scientist
Heinrich Rudolph Hertz, who discovered radio
waves. The unit is always spelt ‘hertz’ with a lowercase ‘h’. The abbreviation for hertz is Hz, with an
upper-case ‘H’.
The human ear can only hear a certain range
of frequencies. 20 Hz, or twenty vibrations per
second, is about as low as it can go. The human
body can perceive frequencies lower than that, but
it is not hearing them in the true sense, but rather
feeling them in the abdomen.
Heinrich Rudolph Hertz
The upper range of frequencies that we can hear
extends to 20,000 Hz, or 20 kilohertz (20 kHz) – ‘kilo’ is the abbreviation
for one thousand, just as one kilogram equals one thousand grams.
Not everybody can hear all the way up to 20 kHz. The upper frequency
range of the human ear deteriorates with age. So although at age ten you
might be able to hear up to 20 kHz, by age 30 you can probably only
manage 15 kHz or so. By the time you retire you might be down to 10
kHz.
Subjectively, this change isn’t noticeable, and it is perfectly possible to
work as a sound engineer without this high frequency range. It is however
always desirable to have a youngster around to warn of any possible high
frequency noise or interference problem.
In light of the above, in sound engineering we normally quote the
frequency range that we are interested in to be 20 Hz to 20 kHz. Some
rare people can hear beyond that range, but for most purposes 20 Hz to 20
kHz is enough.
An Introduction to Equalization
Page At this point it might be a good idea to put this into perspective in relation
to musical instruments. The piano is a good reference point. The lowest
note on a standard piano is 27.5 Hz, and the highest note is 4186 Hz.
No commonly found instrument goes lower than the piano; the piccolo
extends a little higher.
But if the highest note is 4 kHz or so, does that mean that the rest of the
frequency range up to 20 kHz doesn’t matter? Not so – all musical notes
have many frequency components. Take for example the note A below
Middle C on a piano. This has a frequency of 220 Hz. However, it doesn’t
only contain 220 Hz. It will also contain components at 440 Hz, 660 Hz,
880 Hz, 1100 Hz, 1320 Hz etc. Can you see a pattern?
Any note played by a string or wind instrument obeys what is called
the ‘harmonic series’. It consists of the base note, which is called the
‘fundamental’. This is the pitch we hear and determine the note to
be. It also contains frequencies that are whole-number multiples of the
fundamental.
These are called the ‘harmonics’, or ‘overtones’. These overtones extend
all the way up to 20 kHz and beyond. Even a note from a double bass is
rich in high frequency harmonics. To lose these harmonics would take
all the brightness and presence from music, so the upper frequencies are
important.
High frequencies – which we can abbreviate
‘HF’ (low frequencies are ‘LF’) are particularly
important for metallic percussion instruments
such as cymbals. Cymbals are incredibly rich
in harmonics. They follow a different harmonic
series to string and wind instruments, and they
also have strong random frequency components.
Cymbals and metallic percussion
are rich in high frequencies
can lids banging together.
An Introduction to Equalization
These high frequencies must be captured
and preserved in any recording or amplified
performance. Otherwise a set of expensive,
high-quality cymbals can easily sound like trash
Page Imagine a group of acoustic instruments playing together with no
amplification. As you listen, the various notes and harmonics blend
together in a way that is pleasing to the ear, assuming good players of
course.
When you record or amplify these instruments, ideally all frequencies
between 20 Hz and 20 kHz should be captured. Also, and very
importantly, they should all be captured in the same relative levels as
were created by the instruments. In other words, every frequency should
be treated equally. There’s the ‘equal’ of ‘equalization’, as we shall see
shortly. No groups, or ‘bands’, of frequencies should be either raised or
lowered in level in comparison to the others.
Let’s narrow down to just one piece of equipment in the recording or
amplification chain – the microphone. Let’s imagine a single microphone
pointing at a piano. This microphone should capture all the frequencies
produced by the piano at the same relative levels, including all the
fundamentals and all the harmonics up to 20 kHz. It should not emphasize
or subdue any bands of frequencies. If it can capture all frequencies
equally, we say it has a ‘flat frequency response’.
This is a term that crops up regularly in sound engineering and it is
massively important. Every item of equipment that we used should have
a flat frequency response, meaning that it handles all frequencies equally.
That way, the natural sounds of instruments, including their harmonics,
can be preserved.
We can visualize the concept of a flat frequency response graphically…
Level (dB)
20
An Introduction to Equalization
20,000
Frequency (Hz)
Page What we can see from this is basically that what goes in is what comes
out. If this were the frequency response chart of a microphone, then
that microphone would be able to take in acoustic sound vibrations and
turn them into an electric signal, and the electric signal would be at all
frequencies in exact proportion with the original acoustic signal.
The height of the signal on the y-axis (vertical axis) of the graph is called
its ‘level’. Level is a word that is used all the time in sound engineering
and relates to how loud the sound will be when eventually it is reproduced
by loudspeakers or headphones.
It is obvious therefore that a flat frequency response is desirable. But
real-world conditions often dictate that this ideal cannot be achieved.
For example, the microphone might not be of particularly good quality,
or perhaps it has certain other qualities that outweigh the need for a flat
frequency response, in the mind of the recording engineer. If you can’t
have a flat frequency response, this is the second-best case that you would
hope for…
Level (dB)
20
20,000
Frequency (Hz)
In this example, clearly the frequency response is not flat. We describe
a smoothly descending response as ‘roll off’. A smoothly increasing
response is sometimes called ‘tip up’. Smooth roll off or tip up doesn’t
necessarily sound bad. It doesn’t sound like the original acoustic
performance, but it wouldn’t be offensive to the ear, unless done to
extremes.
Likewise, in real world conditions you might find the frequency response
of a certain item of equipment, or a certain combination of instrument/
room/microphone to be something like this…
An Introduction to Equalization
Page Level (dB)
20
20,000
Frequency (Hz)
20
20,000
Frequency (Hz)
or this...
Level (dB)
Where the response goes up in the middle, we call this a ‘peak’. Where
it goes down, it is a ‘dip’. Since in both cases the response is smoothly
changing, it doesn’t have to sound too bad. If the extent of the peak or dip
is large, then yes it can sound bad. If a peak is narrow, then that can sound
bad too (oddly, a narrow dip doesn’t sound too bad – in fact it may go
unnoticed). But where the response is smooth, the problem can easily be
corrected.
The worst-case scenario is this, where the response is uneven…
An Introduction to Equalization
Page 10
Level (dB)
20
20,000
Frequency (Hz)
Loudspeakers often display this highly irregular response. When it occurs
to a significant extent, it is displeasing to the ear, and is difficult to correct.
Microphones too often display an irregular response to sounds that arrive
at angles other than head-on (such as reverberation).
An Introduction to Equalization
Page 11
Level and decibels
I mentioned level earlier. Level is an important concept and it would be
impossible to fully describe EQ without an understanding of level. An
acoustic sound has a certain loudness, which we can call its level. When
it is translated into an electric signal, clearly that signal doesn’t have a
loudness because electricity is silent. But it still has a level.
When instruments are mixed together in a mixing console, each is set at a
certain level. The overall mix has to be of the correct level to record onto
the eventual delivery medium. See how that word ‘level’ is used all the
time.
Level can be measured in decibels (abbreviated dB). We used decibels
because they can apply to sound and to sound signals traveling or being
stored in any medium. So whether we are talking about an acoustic
sound traveling in air, the signal from a microphone, a recording on a
tape recorder, a vinyl record, a film soundtrack etc. etc., we can always
describe level in terms of decibels.
Without decibels we would continuously have to swap between newtons
per square meter, volts, nanowebers per meter etc. It’s so much easier to
talk in terms of decibels. If a singer is asked to sing 10 decibels louder,
then the signal from the microphone will be 10 dB higher in level; the
recording will be 10 dB hotter (which means the same as ‘higher in
level’), and eventually the sound coming from the loudspeakers will be 10
dB louder too. Decibels make describing level easy.
Working out decibels can be complex, but all we need here are a few
simple points...
• A doubling of level is 6 dB
• A halving of level is –6 dB
• A quadrupling of level is 12 dB
• A quartering of level is –12 dB
This text is not about decibels, otherwise the explanation would be much
longer and much more complex. But you need to have this basic grasp of
An Introduction to Equalization
Page 12
decibels to understand EQ. If you can appreciate the above points, then
you know much as about decibels as you need.
An Introduction to Equalization
Page 13
Frequency Response
So far I have talked about ‘frequency range’. But to say that a certain
item of equipment has a frequency range from 20 Hz to 20 kHz isn’t
precise enough. Maybe it covers that range equally, so that all frequencies
are handled in the same way. Or maybe it only responds just a little at
the extremes of the range. To talk about frequency range is useful to an
extent, but it is not precise. We need to talk of ‘frequency response’.
‘Frequency response’ not only describes frequency range, but it describes
the level of the response too. Ideally the response should be equal for
all frequencies. We would call this a ‘flat’ response, which is good. A
completely flat response is never achievable; there will always be some
deviation, however slight. So for a certain item of equipment, we might
find the frequency response specified as follows…
20 Hz – 20 kHz (+0, -1 dB)
What this means is that the level at 1 kHz (1000 Hz) is taken as a
reference, and the response at other frequencies determined. In this
example, no frequency has a response greater than that at 1 kHz; likewise
the maximum downward deviation is 1 dB – no frequency is more than 1
dB down with respect to 1 kHz.
Let’s try another example…
20 Hz to 20 kHz (+/- 3 dB)
In this case, the response varies quite widely over a six decibel range.
However, at no frequency is the response greater than +3 dB compared to
1 kHz, and at no frequency is the response less than – 3 dB compared to 1
kHz.
Either method of describing frequency response is perfectly good. You
should be absolutely clear though that a frequency response specification
must include both lower and upper frequency limits, and lower and upper
level limits. If not all four items are included, then it is inadequate as a
frequency response specification.
An Introduction to Equalization
Page 14
What is equalization for? What are we making ‘equal’ to what?
Equalization is all about cutting or boosting bands of frequencies with
respect to other bands of frequencies. So what are we making equal to
what? That is a good question.
To answer that question we have to go back into history. The earliest
practise of sound engineering was when it was
discovered that a telegraph cable, used for
sending simple electrical pulses in Morse code,
could be used to transmit speech over long
distances. One of the problems was that level
was lost over long distances. To combat this,
amplifiers were placed along the way to boost the
The telephone - what EQ was
signal back up again periodically.
invented for
However, it was found that certain bands of
frequencies suffered more than others. So the amplifiers were made
frequency selective to bring the response back to flat. This process was
called equalization. So ‘equalization’ means making the output of a
telephone cable equal to the input – equal in terms of frequency response.
For a long time, this was what equalization was used for. Even well
into the era of recording and broadcast, equalization was used to correct
frequency response problems, where and when they occurred. But then
at some point, some bright spark recording or broadcast engineer must
have twiddled an EQ control and thought, “Hey, that sounds nicer!” So
rather than using EQ to correct a problem, it was used to improve the
sound subjectively. So no longer was the output ‘equal’ to the input, it was
enhanced, over and above the input.
Once this idea caught on, there was no limit to how EQ could be used.
Recording engineers in particular used EQ in many and varied creative
ways, particularly during the 1960’s.
Moving on to today, we use EQ in both of these ways. Firstly as a
corrective tool to compensate for frequency response irregularities caused
by inadequate equipment, a less than satisfactory instrument, or poor
acoustics or microphone positioning. When we have done that, we go
An Introduction to Equalization
Page 15
further and use it to enhance the sound to our liking.
Later in this text I will examine both of these uses of EQ in detail. But
firstly we need to look at some equalizer designs.
An Introduction to Equalization
Page 16
Filters
The filter is the simplest form of equalizer. Some people wouldn’t call
it an equalizer but refer to it specifically as a filter. That isn’t important
however; the function of the filter is very similar, as is the way it is used.
A filter removes bands of frequencies. It never boosts. There are five
principal types of filter…
• Low-pass, where low frequencies are allowed to pass through but
high frequencies are reduced in level (‘attenuated’).
• High-pass, where high frequencies are allowed to pass through but
low frequencies are reduced in level.
• Band-pass, where both low and high frequencies are attenuated; mid
frequencies are allowed through.
• Band-stop, where both low and high frequencies are allowed to
pass, but a region in the mid-band is attenuated.
• Notch filter – a very narrow band-stop filter, taking out a small
range of frequencies.
Level (dB)
Low-pass
20
20,000
Frequency (Hz)
20
20,000
Frequency (Hz)
20
20,000
Frequency (Hz)
20
20,000
Frequency (Hz)
Level (dB)
High-pass
Level (dB)
Band-pass
Level (dB)
Band-stop An Introduction to Equalization
Page 17
Level (dB)
Notch
20
20,000
Frequency (Hz)
Filters almost always have switched controls. They are either in or out
– there is no continuous control to blend the effect of the filter.
So for instance, you might be amplifying a singer on stage, but you can
hear foot noise coming up the microphone stands, which is a common
problem on wooden stages. A quick solution to this would be to switch in
a 100 Hz high-pass filter. Most mixing consoles for public address have
this feature. The low frequency energy coming up the stand is reduced
in level and the problem is solved. Well, maybe not entirely solved but
certainly made better than it was.
A more well-specified filter will offer a number of ‘cut-off frequencies’.
The ‘cut-off frequency’ is the point at which the level is reduced by –3
dB compared to the level in the ‘pass band’. The range of frequencies
beyond the cut-off frequency is known as the ‘stop band’.
Here is an example of a high-pass filter, from
a Neve mixing console. As you can see, it has
switch positions for 50, 80, 160 and 300 Hz.
Simple and effective.
Neve filter section
To summarize filters so far, they have a type,
and they have a cut-off frequency. They also
have another parameter known as ‘slope’.
It might be possible to imagine a filter that
passes everything in the pass band, and stops everything in the stop
band absolutely. This kind of filter has a name – a ‘brickwall filter’. The
problem with the brickwall filter however is a) it is difficult to make with
analog circuits, and b) it doesn’t sound good.
Somehow, the ear can detect the sharp boundary between the presence and
absence of frequencies. Brickwall filters are used in CD players and other
An Introduction to Equalization
Page 18
digital devices, but they are not used in the recording process, live sound
or the operational areas of broadcasting.
Practical filters attenuate frequencies in the stopband, meaning lowering
their level. They don’t completely cut them out. It is the rate of attenuation
that is important, as shown by this diagram…
Level (dB)
24
20
18
12
6
decibels/octave
20,000
Frequency (Hz)
Here you can see the four most commonly used filter slopes – 6 dB/
octave, 12 dB/octave, 18 dB/octave and 24 dB/octave. To explain for
instance 6 dB/octave, it means that beyond the cut-off frequency where
the graph has become a straight descending line, the response drops by six
decibels for every doubling of frequency. Simple as that.
So the greater the slope, the faster the rate at which the level drops. Thus,
a 24 dB/octave filter has a much greater audible effect than a 6 dB/octave
filter. In fact, 6 dB/octave is too gentle for most purposes and it is rarely
found. 24 dB/octave is too harsh and also rare. 12 and 18 dB/octave are
the commonly found values.
You might well ask what happened to the in-between values? What about
15 dB/octave, for example? It turns out that filters with the four values
listed are easy to design and construct. Filters with in-between values are
possible, but much more difficult, and the result is further from the ideal
response. There simply isn’t any point to making filters with in-between
slopes, the standard slopes are quite good enough.
An Introduction to Equalization
Page 19
Just to round off this section, slopes are sometime quoted over a decade
of frequency rather than an octave. An octave is a doubling of frequency,
a decade is a ten-fold increase. So a filter that has a 12 dB/octave
slope could also be described as having a 40 dB/decade response. This
terminology is comparatively rare though.
An Introduction to Equalization
Page 20
Passive EQ and tone controls
Moving on to the simplest kind of EQ, we have the passive EQ circuits
typically found in vintage equipment, retro equipment and guitar
amplifiers. A passive EQ uses resistors, capacitors and sometimes
inductors – all common electronic components – to subtract level from
certain bands of frequencies.
The amount of reduction can be controlled, making this kind of EQ more
versatile than a simple filter (although it doesn’t detract from the value of
having a filter – filters are always useful). Since passive circuitry can only
make the signal smaller, clearly it is necessary to have amplification after
the EQ stage, firstly to bring the signal back up to full level, and secondly
to provide the opportunity of having an EQ boost.
The passive part of the circuit can only cut, so the boost has to be done
by raising the levels of frequencies that were not cut. Complicated, but
understandable if you think about it.
Pullet passive equalizer
The drawback of a passive EQ is that it loses signal level. Therefore the
signal gets closer to the background noise level, and when boosted back
up, the noise also gets boosted. So typically you can expect a passive EQ
to be noisy, although some designs are better than others. The advantage
of passive EQ is that it sounds different to the more modern active EQ.
Although active EQ is better in almost every respect, recording engineers
just like to have something that sounds different – it’s another tool in the
toolbox. There are also advantages when it is necessary to cut or boost a
narrow range of frequencies by a large amount. A well-designed passive
EQ may sound smoother and cleaner.
Passive EQ, apart from the exceptions noted, is now quite rare. Moving
on to the simplest active design (‘active’ in this context means that an
amplifier circuit is itself made frequency-selective, so no level is lost
as in the passive EQ.), we have the tone controls found on hi-fi and
An Introduction to Equalization
Page 21
other consumer audio equipment. Often you will find controls labelled
‘bass’ and ‘treble’. Clearly, the bass control will cut or boost the low
frequencies; the treble control cuts or boosts the high frequencies. There is
one standard circuit that is employed for tone controls, invented by Peter
Baxandall and called the Baxandall tone control.
The Baxandall tone control is a simple and
elegant design, and any hi-fi or domestic
equipment manufacturer would have to be a little
crazy to want to do it any other way. However,
it is really only suitable for modifying the end
product to individual preferences. It is only
capable of a 6 dB/octave slope at most, which,
for pro audio purposes, is a little like trying to cut
with a blunt knife.
I could work up to full-scale EQ gradually, but it’s probably better to
start right at the top with the EQ section from a Solid State Logic mixing
console, which as console EQ goes is about as good as it gets. The
diagram has been edited to show only the features that are relevant to EQ.
An Introduction to Equalization
Page 22
Mixing Console EQ
Here we can see four separate bands of EQ. The topmost band deals with
high frequencies, the middle two work on mid-range frequencies, the
lowest band is for low frequencies. Obvious really. But let’s look at each
band in detail…
The high frequency band has two rotary
controls. The ‘kHz’ control, incorrectly labelled
‘KHz’ by SSL’s graphics person, controls the
frequency at which the high frequency section
starts to take effect. Below this frequency,
not much will change. Above this frequency,
changes will be audible. As you can see, the
range of frequencies extends from 16 kHz all
the way down to 1.5 kHz.
Most people would say that 1.5 kHz is a
distinctly mid-range frequency rather than a
high frequency. However, extending the range
this low offers more flexibility in control, which
is always a good thing. Important note - the
control here labelled ‘kHz’ is more commonly
known as ‘frequency’.
One thing that confuses newcomers to EQ is
that it is possible to turn this control all the way
from one end stop to the other without hearing
any change in sound quality. This will happen
if the ‘dB’ control is set to its center position.
In the center position, the dB control does
absolutely nothing – it neither cuts nor boosts.
Therefore the position of the kHz control
doesn’t matter – nothing is being changed.
So you have to set a certain amount of cut or boost for the kHz control to
become relevant. Important note – the control here labelled ‘dB’ is more
correctly known as ‘gain’, but also commonly known as ‘level’. (‘Gain’
is the property of a circuit that changes the level of a signal. Gain can be
An Introduction to Equalization
Page 23
either positive or negative).
All equalisers offer a certain range of cut or boost on their gain controls.
Typically it would be around +/-15 dB. +/-18 dB is better, +/-12 dB isn’t
as good. But they are the commonly found limits.
There is one more control here – a button marked ‘BELL’. This button is
more properly called the bell/shelf button. It affects the way the equalizer
processes frequencies well above the frequency to which the kHz control
is set. Here are the options…
Bell
Level (dB)
20,000
Frequency (Hz)
20
Shelf
Level (dB)
20
20,000
Frequency (Hz)
As you can see, the bell setting boosts a certain range of frequencies, but
at the extreme high frequency end of the graph, the boost returns back
down to zero. In comparison, the shelf setting boosts frequencies all the
way up to the limit of the audible range. Whether you choose bell or
shelf is entirely down to subjective perception. There is no right or wrong
An Introduction to Equalization
Page 24
choice in any particular situation – it is entirely up to you to decide what
sounds best.
I’ll skip over the two mid-frequency sections for the moment. The lowfrequency EQ section, as you can see, is very similar in layout to the
HF section. There is a dB (gain) control and a Hz (frequency) control in
exactly the same manner. They just work on low frequencies rather than
high. There is also a bell/shelf button that does exactly the same as the HF
bell/shelf but extending towards the lowest frequencies of the range.
The range of the frequency control is from 30 Hz to 450 Hz. This is a
wider range than you would probably ever need. If anything below 30
Hz needs controlling, it probably just needs filtering out, and of course
you have a filter for that. 450 Hz, is far from being anything you could
describe as a low frequency – it is well into midrange. However, having
that extra scope is good because you never know when it might come in
useful.
Going back to the mid-range controls, the two sections are identical in
everything except frequency. The upper section deals with high midrange, the lower section with low mid-range. On some consoles, the
two mid-range bands are entirely identical and cover the same ranges of
frequencies.
At this point I will assume that you know what the kHz (frequency) and
dB (gain) do, it’s just the same as for the other bands. But there is an extra
control – the Q control. What is that for?
To explain the Q control, I have to start by explaining Q. The concept of
Q dates back to the early days of radio when engineers were struggling
to achieve a good ‘quality’ of resonance, which apparently is where the
Q comes from. A circuit that would resonate well (to ‘resonate’ means to
vibrate or oscillate readily at a certain frequency, given an energy input)
would form the basis of a good transmitter or a good receiver.
We use exactly the same concept in sound engineering today, but at audio
frequencies rather than the much higher radio frequencies.
An Introduction to Equalization
Page 25
Here is a graph showing a resonant boost. It could just as well be a cut,
the concept works both ways.
Level (dB)
0 dB
-3 dB
F1
F0
F2
Frequency (Hz)
What we can see here is that the bell of the curve can be wide or narrow.
If it is wide, we say it has a low Q; if it is narrow we say it has a high Q.
You can calculate Q by taking the two frequencies either side of the peak
of the resonance, subtracting the lower from the higher, and then dividing
the result into the center resonant frequency.
Q = f0 / (f2 – f1)
Since the top and bottom of the dividing line are both measured in hertz,
the units cancel out so that Q has no units. Q is a simple ratio.
Going back to the equalizer, we can see that Q is adjustable to give a wide
or a narrow bandwidth to the curve of the EQ. If Q is set low, then a broad
range of frequencies will be affected. If Q is set high then only a narrow
range is changed. Sometimes it is difficult to hear the effect of changing
the Q. To get a feel for what Q can do, set a large boost at a mid-range
frequency so you can easily hear what the EQ is doing, then sweep the Q
up and down and listen for what it does. When you have a feeling for what
Q sounds like, you will be able to use it effectively. In general, a high Q is
used where there is a small band of frequencies causing a problem – like
an unpleasant resonance in a snare drum that needs filtering out. A low Q
is more useful for musically-inspired changes, just to make things sound
the way you want. Often, you would set a low or high Q first, before
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adjusting the other two controls of the section. This type of EQ section
with controls for frequency, gain and Q is often known as a ‘parametric
EQ’.
On some equalizers, the Q control is labeled ‘bandwidth’. It does exactly
the same job, but where a Q control would be calibrated from ‘low’ to
‘high’, or from ‘0.5’ to ‘7’, a bandwidth control would be calibrated from
‘wide’ to ‘narrow’ or from ‘one-third octave’ to ‘two octaves’.
Just remember that wide bandwidth = low Q, narrow bandwidth = high Q.
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EQ IN button
One more very important control on the EQ section as a whole is the ‘EQ
IN’ button, which I would prefer to call the EQ In/Out button. This simply
switches the EQ in or out of circuit. There are two reasons why this is
necessary. Firstly, the EQ circuit is complex and to a small but possibly
audible extent degrades the signal. So if you don’t need EQ, it is better to
switch it out. The degradation is small though and few people would be
likely to hear any difference in the context of an entire mix.
The other reason is much more important – so that you can easily hear the
difference between EQ-in and EQ-out. You need to be sure that you are
improving the signal! EQ is a powerful tool and it is perfectly possible
that you are making it worse. With the EQ In/Out button, you can easily
tell.
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Graphic EQ
The EQ sections provided in mixing consoles are flexible, easy to use
and powerful. Plug-ins on computer-based recording systems mimic the
features of analog console EQ. But there are other styles of equalizer, one
popular type being the graphic EQ. Here we can see the Klark Teknik
DN360 two-channel graphic EQ.
The Klark Teknik DN360 has thirty bands of EQ per channel, each band
covering one third of an octave. So this would be called a third-octave
graphic equalizer. Each band has a cut or boost up to +/-12 dB. There are
graphic equalizers that work in whole octaves with fewer bands, but they
are not nearly as effective or precise.
The idea is that you can set any frequency response curve you like, and
not necessarily symmetrical in the mid-range as it is with a conventional
resonant EQ. And when you have set the graphic EQ the way you like it,
the positions of the knobs show a graph of the frequency response! In fact,
the supposed graph is merely an approximation because the bands overlap
and interact with each other. However, even an approximation is useful
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– you can glance quickly at a graphic equalizer and see what kind of curve
it is set to.
You don’t have to know much about a graphic equalizer to operate one.
It is simple and intuitive. However there are some things you could know
to make your understanding better. Firstly, when you raise one slider, you
are not only affecting the frequencies between that and the two adjacent
sliders.
The Q of each section is low and a wide range of frequencies is altered.
It has been attempted to make graphic equalizers with high-Q circuits,
but they just don’t sound as good. So expect the bands to interact. The
other thing that you might care to know is that graphics come in two types
– variable-Q and constant-Q. With a variable-Q circuit, the Q of a section
gets higher as you apply more boost or cut.
With a constant-Q graphic, the Q always stays the same. Opinions are
divided on which sounds the best. I prefer variable-Q, but only by a small
margin. If I only had a constant-Q graphic to hand, I wouldn’t hesitate to
use it.
Something else you need to know about graphics is that they really screw
up the signal. They do nasty things to a signal’s phase. I need to explain
‘phase’. The quick explanation is that an instrument emits a sound,
and every frequency component of that sound reaches your ears at the
same time. The speed of sound does not vary with frequency. Transform
that sound into an electric signal and things change. Common circuit
components delay certain bands of frequencies with respect to others.
This applies particularly to EQ. When you EQ a signal to change the level
at certain frequencies, you also change the timing of those frequencies.
In practice, it is not possible to design a circuit that doesn’t mess up the
phase of a signal. But it turns out that if the phase changes smoothly
through the frequency band, the ear doesn’t notice. Many equalizers are
designed to be ‘minimum phase’, which means that they change the phase
to as small an extent as theoretically possible.
Unfortunately, the graphic equalizer is anything but minimum phase.
Having said that, the audible differences are slight and generally obscured
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by the loudspeaker which, phase-wise, is the worst offender of all. So it’s
not a big thing, but worth knowing anyway.
The principal application of the graphic equalizer is in live sound. PA
systems are prone to howlround when the sound from the speaker enters
the microphone and is re-amplified, resulting in a loud tone filling the
auditorium.
So the sound engineer has to find the frequency at which howlround is
most likely to occur, which is different for every set-up and every venue,
and reduce the gain at that frequency. The graphic equalizer is simply the
most convenient tool for doing this, hence graphics are in almost universal
use in this application. There is no howlround problem in studios, at
least not if you’re doing things correctly, so graphic equalizers are less
commonly found.
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Using Equalization
It is important to understand what equalization does and how it works,
which has been covered and you now have a good preparation for what
follows – using EQ.
I have said already that the use of EQ is a skill and an art. You may have
seen elsewhere instructions to cut or boost certain frequencies for certain
instruments to achieve ‘attack’, ‘clarity’, ‘air’ or some such quality. By
all means, read everything you can. But there are no rules of EQ. No-one
can tell you exactly what frequencies to cut or boost, and by how much,
because they don’t know what sound it is you are dealing with.
Every instrument is different, every player is different, every acoustic
space is different, every model of microphone is different. No, there is
little value in set instructions. What you need are tools that will allow you
to assess a sound and decide what needs to be done to it, within its own
frame of reference, and also precisely the way you want to hear it.
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Corrective Equalization
The first use of equalization is to correct something that doesn’t sound
right. Let’s consider single instruments to start off with. It is perfectly
possible that an acoustic guitar, even an expensive one, has an irritating
resonance where the instrument itself boosts a band of frequencies, and it
doesn’t sound good.
The first action to take is to experiment with microphone position and
selection (position is nearly always more important than which mic you
choose). The way a professional recording engineer would do this is to
listen from the control room while his or her assistant moves the mic to
various positions.
When a rough position is found, the engineer will give precise instruction
on the exact placement down to the last centimeter. This in a way is a kind
of acoustic EQ; finding the spot where the balance of frequencies just
happens to be optimum. Finding the best microphone position first is a
necessary step before EQ.
Another example would be a drum, say a snare drum. Drums often have
annoying resonances that would benefit from being removed. But the first
step isn’t EQ; the first step is to tune the drum. Drum tuning is outside of
the scope of this text, but the quick solution is to find a drummer who is
experienced in recording to show you how.
Once the drum is tuned it will sound a whole lot better. It may need
damping to reduce the resonance, and of course care and attention should
be given to mic positioning. All of that comes before EQ.
Do you get the picture? Acoustic sounds, and electric sounds from guitar
amplifiers, should be optimized at source. Microphone positions too
should be optimized. Only after that, if there is still a remaining problem,
should you start with corrective EQ.
Even with the best care and attention to the above, you might still end up
with an acoustic guitar that has a cheap-sounding resonance, indicating
a certain band of frequencies that is naturally strong in that particular
instrument. Now is the time to correct that with EQ, before it goes onto
the recording. If it is indeed a resonance that is causing the problem, then
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you need to attack the band of frequencies that the guitar is boosting
acoustically with EQ that does the inverse. Here is what you do…
Step one is to set the gain of the EQ to a significant boost – a halfway
boost of eight or nine decibels is usually enough. If there is a Q
control, set it to around 3, or to a moderately high value if there are no
calibrations.
Now, as the instrument is playing, sweep the frequency control from
low to high and back down again, slowly and repeatedly. The band of
frequencies that was causing the unpleasantness will now be doubly
boosted and you will hear very clearly where the problem lies. This is an
easy way of identifying troublesome frequencies – the ear hears a boost
much more readily than it does a cut.
Now that you have identified the problem frequencies, simple change
the boost to cut. Fine tune the gain and Q controls so that you shape the
correction to the shape of the problem. Now your guitar will sound much
better. It won’t sound better than a better guitar would have done – EQ is
a powerful tool but it can’t work miracles.
This technique for corrective EQ in the midrange always works. As
you gain experience you will be able to set the correct frequencies to
cut directly, without the intermediate boost phase. But it will still be
something that you experiment with on particularly troublesome sound
sources.
There are also commonly-found problems in the bass and high frequency
ends of the sound spectrum. One such is low frequency noise. LF noise
can be caused by vibrations from foot falls coming up the mic stand. This
can be minimized by using an elastic cradle to hold the mic rather than the
regular clip. However, such cradles are expensive and not available to fit
every microphone. So it will not always be possible to use one.
Another source of LF noise is ventilation and air conditioning. It is
not possible to have a soundproofed studio without ventilation, and
air conditioning is a valuable extra. Noise from these sources must be
managed effectively. Even in rooms that do not have ventilation provided
by a fan, natural air currents exist that can cause noise at extremely low
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frequencies. Listen out for low frequency noise, and if it exists, simply
use a filter to magic it away. Do not however do this for bass instruments.
Whatever you take away can never be put back, and equalization of bass
instruments should always be left until the mix.
Sometimes instruments themselves are excessively bass heavy. The
problem can be too much bass at very low frequencies – below 40 Hz or
so. Although most people, when asked to express an opinion, will say that
they like lots of bass, what they really mean is that like to hear frequencies
of 80 to 100 Hz or so pumping out at a high level. High levels of very
low frequencies are more likely to make you feel sick. Although these
very low frequencies may not be desirable, and you might consider that
corrective action is necessary, it is usually best to play safe and not make
any change that will affect the recording.
The same applies to fizzy high frequencies. Although a harsh and brittle
top end might seem to be an undesirable feature that needs correction, you
will find it difficult to get the top end brightness back if you take too much
away at this stage.
In summary, corrective EQ is very appropriate to the mid range of
frequencies, also very appropriate to low frequency noise. But the low
frequency content of bass instruments, and the high frequency content
of instruments and voices should be left intact until mixing. It is very
difficult to put back frequencies that you have taken away.
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Creative Equalization
Prime time for creative equalization is in the mix, where all the sounds
you have recorded come together and need to blend well. But there
are several alternative approaches, all of which can work well, given
attention, thought and care. Let’s start with the scenario of a live recording
of a jazz band. How should you approach that EQ-wise?
The thing about recording a band live as they play, rather than doing it
instrument by instrument, is that you have the opportunity to hear what
the band really sounds like. And that sound will become a benchmark for
your mix. If your mix doesn’t achieve the same level of quality as the live
sound, then you haven’t done your job properly. However you will score
points massively if your mix sounds even better than the live sound.
In this situation, the best approach is to start mixing (the band have all
gone home now) with the EQ sections all either switched out, or set to
flat (all EQ gains at their center positions). Balance the instruments on
the faders and panpots and get as good an overall sound as you can. Work
hard at this stage and don’t be satisfied easily. Try different options, often
the first balance you arrive at isn’t necessarily the best. Explore the mix,
play with it, get to know it.
An hour doing this is an hour well spent. When you have become
thoroughly familiar with your source material you can start to think about
EQ. As you listen to your best faders-and-panpots mix, you will find that
some instruments are not being heard properly, yet raising the fader makes
them too loud. Conversely, other instruments stick out like a sore thumb,
but lowering the fader makes them go away. You just can’t find the right
fader positions, or the right fader positions have to be tuned to within a
couple of millimeters. You need EQ!
What happens in a band is that several instruments or groups of
instruments will try to compete for the same frequency space, in their
fundamentals but also in their harmonics. And whichever instrument
happens to be louder at any particular time will mask other instruments
competing for the same frequency space. So in this case, let’s say that you
are having difficulty hearing the trumpets and clarinets distinctly when
they are playing together. Set an EQ boost for the trumpet channel and
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sweep the frequency control until they stand out more prominently. Do the
same for the clarinets. If you find that the same center frequency works
equally for both, skew one channel upwards in frequency and the other
down. Now you have differentiated these instruments sufficiently for them
not to mask each other.
Clarinet and trumpet - how to give
each its own space in the frequency
spectrum?
As a finesse, if you have two mid-range
EQ sections per channel, or have enough
computer processing power to run
additional pug-ins, whatever frequencies
you boosted on one channel, cut on the
other, and vice versa. So not only are you
making the trumpets more prominent at
their key frequencies, you are scooping
out a ‘hole’ in the same frequencies on the
clarinet track. This technique is sometimes
known as ‘complementary EQ’. It is a
powerful tool.
When you are mixing a band like this, when you have either heard them
play live in the studio, or it is a conventional line up and you know what
it should sound like, always apply EQ in context. This means that you do
not solo any channel while you EQ, apply EQ while all of the instruments
are audible. In this way, you can see what effect the EQ has with reference
to the entire mix.
Many recordings are not made with conventional band instruments, or
with the musicians not playing simultaneously. In cases like this, there is
no reference point. You don’t know what it ‘should’ sound like. Rather
than ‘live up to’ a standard, it is your responsibility to create that standard.
This is more difficult, but it offers more creative opportunities too.
In this case I will assume that you applied corrective EQ during the
recording process, so all the instruments and voices sound fully adequate
at least. You could try a faders-and-panpots mix, as in the whole-band
example above, but the result will probably be something of a jumble.
Since the instruments were recorded separately, there wasn’t much
information to go on as to how they should blend.
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In this situation, one very effective approach is to start from a ‘foundation
mix’ – the very fewest instruments that can stand on their own and support
the rest of the track. Very likely this will be the drums, bass and one ‘pad’
instrument – guitar or keyboard perhaps. If you can get this ‘rhythm
section’ blending well, then everything else will hook in easily with that.
As before, you can do a faders-and-panpots mix of the foundation
instruments. Set the EQ of each so that the sound is full and rich – it could
be a track in its own right but for the lack of vocal and color. This can be
done by EQing in context, and of course applying complementary EQ
– particularly in frequency areas where the kick drum and bass instrument
clash. When you have all of this sounding really good, you can start
adding the other components. The vocal will be next.
What you will typically find when you add the vocal to an already fullsounding track is that the vocal doesn’t have a space to fit into. Once
again, complementary EQ will come to our assistance. Unlike other
instruments, the human voice is pretty consistent in the frequency bands
in which it is strong. This stems from human evolution – we needed to
communicate effectively so the ear has evolved to be very sensitive at
frequencies where speech is also strong – the range round 3 kHz.
Notice that we are talking harmonics here, not fundamentals. But this is
the range that allows us to differentiate between the consonants, vowels
and phonemes of speech, in both the male and female voice. So if you
apply an EQ boost to a vocal at around 3 kHz, it will suddenly sound very
much more present and stand out wonderfully. Of course the next step is
to apply complementary EQ to the other instruments to make a ‘hole’ for
the vocal to ‘sit’ in.
As you start to add other instruments you will find that they need to be
‘thinned’. Your foundation tracks are already fat – or ‘phat’! – because
you spent time optimizing them. The vocal is complementary EQ’d to
perfection. So there is no room in the frequency space for anything else!
Well, yes there is, but you can’t add more ‘phat’ tracks to an already ‘phat’
mix. You need to ‘thin’ the new instruments so they will fit in.
Thinning can be accomplished by cutting low frequencies and often
cutting high frequencies too. If an instrument isn’t thin enough at this
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stage, you can apply a boost where it is harmonically strong. If the worst
comes to the very worst, you can apply a complementary EQ to your
foundation track to make a space for the new instrument to fit in. Oddly
enough, in a world of ‘phat’, there is an amazing power in thinning things
down.
It can’t be emphasized too strongly that there is only a limited audio
spectrum and everything has to fit into that. If every sound is rich in
a wide range of frequencies, they will clash and mask each other. So
you have to make your instruments complementary to each other in
their frequency characteristics, and thin them down where necessary.
Ultimately, your finished mix will sound so much bigger for that.
Cut can be better than boost
The brain doesn’t react symmetrically to boost and cut. It pays far more
attention to frequencies that are boosted than to frequencies that are cut
down in level. But EQ cut can be a very effective tool in sound shaping
and blending. Let’s take the case of a vocal again. As I said, adding a lift
around 3 kHz will give it much more presence. But you might still find
that it can’t find its place in the mix – the fader is either too low or too
high and you just can’t seem to find the right spot.
What is happening here is that the lower range of the vocal is clashing
with the rest of the mix. When the vocal is loud enough to be clearly
audible, the lower range is too loud and is sticking out. So the answer is
not only to boost around 3 kHz, but also to cut in the sub-1 kHz region.
I often find myself cutting around 800 Hz, but by all means experiment
between about 300 Hz and 1 kHz.
Yes, between these frequencies you are cutting down fundamentals, but
the brain has a mysterious way of reconstructing missing fundamentals
from the harmonics that it hears. Just listen carefully to what you are
doing and all will be well.
Another key place to cut is in bass instruments, kick drum and bass guitar
or synth. It is quite common for the kick drum to be over-rich in energy
below 40 Hz or so. Subjectively this isn’t pleasant and from a technical
point of view eats up loudspeaker ‘excursion’ (the distance the cone can
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move). So excessive low frequency energy not only sounds bad, it makes
the loudspeakers distort before the music is really loud. And this applies
to anyone’s listening system, not just your monitors. So by reducing the
level below 40 Hz – 50 Hz or so, you can set the kick drum fader higher
so that you get more level around 80 – 100 Hz, which subjectively gives
exactly the ‘kick’ we need. (As an aside, you might consider a boost in
the high hundreds of hertz for attack, and a further boost around 5 kHz for
crispness and click.)
I’m sure you get the idea. But you can take cut a stage further and use cut
instead of boost. Think of it like this – whenever you use boost, you are
making certain frequencies louder than others. You can do the same by
cutting the frequencies you don’t want to be quite as loud, and then raising
the fader. It is a subtle but useful difference. It is perfectly possible to
construct an entire mix using cut only. I’m not saying you would always
want to, but my strong feeling is that when you scan the EQ controls
across the board, most should be set to cut rather than boost.
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Equalizing the mix
So far I have talked about equalising individual instruments and voices.
But it is also important to equalize the stereo mix. There are some
situations where you probably would not want to do this – let’s say you
have recorded an orchestra playing live in a hall with good acoustics,
with very accurate microphones. Why would you ever want to equalize
something that is already so obviously perfect? Actually you might – if
the label on which the recording is released has a ‘house sound’, they
might EQ it so that it sounds subjectively comparable to their other
releases.
For popular music however, there is no
point of reference other than the many
classic recordings that have been made
over the years, and of course recordings
that have sold well recently. If you
aim to get your balance of frequencies
comparable with music in a similar genre
that is commercially successful, you won’t
be making a mistake.
This isn’t a lesson on mastering, which is
a massive subject in itself. But you should
aim to get a mix that sounds good on a
variety of playback systems, on your MP3 player, car stereo, portable
system and full-on hi-fi. And if you apply only EQ to your mix and no
other process, a pro mastering engineer can work his magic unrestricted.
If you compress a mix and get it wrong, that cannot be undone.
The best way to EQ a mix is to set up a track that you like in a similar
genre. Aim to give your mix a similar frequency balance. The EQ that
you use here is totally dependent on the EQ you applied to the individual
channels during mixing, so everyone’s situation is different. But I find
myself usually boosting the low end gently, the high end also gently, and
taking a low-Q scoop out of a band somewhere between 100 and 1000 Hz.
This is a time for very attentive and detailed comparative listening with
your reference track. Take your track and the reference track to various
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playback systems. Typically you will find that the reference track travels
better than yours – that’s because it was worked on by a seasoned
mastering engineer who equalizes mixes every day of his life. In time
however you will learn to recognize the differences between mixes
produced as the fruit of massive experience, and your mixes. Gradually
your mixes will acquire the polished professional sound too.
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Equalization for live sound
Equalizing for live sound is very much like EQing for recording, but there
are certain extremely important differences.
The first is that a sound check for a live show often has to be completed
in as few as three songs, sometimes even just one song. This means that
you have to set the EQ for the entire show from very scant information,
perhaps with a band you have never seen before.
In fact, the live sound engineer isn’t EQing songs as much as EQing the
band, and then perhaps fine-tuning as the show progresses.
A live sound engineer will quickly find preferences for EQ that he or
she will apply from show to show, for the drums, bass guitar, guitar
and vocals. Keyboards are too varied to be
able to have much in the way of expectations.
Adjustments will be made during the sound
check to those basic settings.
The second difference between live sound and
recording is that the live sound engineer is
always battling against howlround (feedback).
Setting too much of an EQ boost on a vocal mic
is almost bound to create howlround, so EQ
has to be done with this in mind. As described earlier, a graphic EQ will
be used to equalize the system as a whole, so the engineer is in a sense
fighting against this.
One point that is important about setting the graphic is that if, in the
interests of combating feedback, too much level is taken out in the vocal
range, then the vocals will simply be quieter, so the engineer will have to
push the fader higher on the console, thus counteracting the initial antifeedback EQ. Keep this thought in mind when setting the graphic.
The third difference between live sound and recording is actually a bonus
- you get to hear your sound the way the audience hears it! You don’t have
to worry about people listening on different playback systems - you’re all
in the same auditorium. OK, you are in the best position, but if the system
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has been correctly set up, then if it sounds good to you, it will sound good
anywhere in the house.
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Appendix - Extreme EQ from hardware equipment
What goes through a mixing console designer’s mind when he starts work
designing the EQ section? Is he thinking, “How can I give the engineer
more power and control?”, or is it more along the lines of, “I’d better not
give the engineer too much power and control - what might he do with
it?”. The more consoles I listen to, the more I am inclined to think the
latter.
OK, it is important for an EQ section to be musical, and to allow very fine
differentiation in settings for when just subtle changes are required. But
what do you do when you want to rip a sound apart, tear out its entrails
and shove the bleeding mass in the face of the listener? If you’ll excuse
my metaphor, you could turn to an outboard EQ but I think you will find it
still a little too polite.
In all probability these days an outboard EQ will just be part of a mixing
console channel taken out and put in a rack mounting box, although there
are exceptions. For Extreme EQ, we have to look outside the cosy world
of what we consider to be audio equipment into what is known to the
trade as the MI - musical instrument - market. Here we will find hardware
and software that will go far beyond the capabilities of most of the EQ
units we would normally consider.
Of course, conventional EQ units can do lots of things that MI systems
cannot, but we already have lots of fine control, subtlety and musicality
- we need raw power! I have chosen examples of filters which I can
guarantee will amaze you if you have never heard anything but a
conventional EQ before from a traditional piece of hardware with knobs
and switches, and of course you can achieve similar effects with computer
plug-ins.
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Mutronics Mutator
The Mutator is basically a two channel low-pass filter, with LFOs to
modulate the cut-off frequency and envelope followers to allow the
envelope characteristics of one sound to be superimposed upon another.
The filter really is the essence of the Mutator. Basically all it is is the
filter circuitry of a traditional analog synthesizer brought up to modern
standards of noise and distortion performance.
As simple as that, and in fact you could think of Mutator as an analog
synth without oscillators - just plug in your own sound source. It is a
low-pass filter meaning, as you already know, that high frequencies are
attenuated, in this case with a slope of 24 dB/octave. In a 24 dB/octave
filter, above the cut-off frequency, as the frequency doubles the output
voltage is reduced to a sixteenth. This is the first and major difference
between this and standard EQ.
With conventional equalisers the slope will be a mere 12 dB/octave or 18
dB/octave which reduces the levels of higher frequencies but still leaves
them audible. A slope of 24 dB/octave chops them off with an axe. The
result is that you can input a signal with a fizzy irritating high end and
reduce the cut-off frequency to leave only the useful components. With a
24 dB/octave filter the result can still be sharp and incisive, whereas with
a 12 dB/octave or 18 dB/octave filter by the time you have eliminated the
fizziness the sound will just be dull.
Another difference between the Mutator’s filter and the filter you would
find on a conventional EQ is that where the conventional designer would
only allow you to filter frequencies down to, say, 2 kHz with a low-pass
filter, the Mutator goes all the way down to subjectively nothing at all the cut-off frequency is so low that the only signal left is a vague rumbling
in the distance. Don’t conventional EQ designers trust us?
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Although a simple 24 dB/octave filter is a powerful tool, analog
synthesizers commonly have a resonance control too, and so does
Mutator. The resonance control sets a certain amount of boost just below
the cut-off frequency. This is in fact more of a synthesis tool and even
for Extreme EQ you would only need to use a fraction of the boost that is
available. But who knows what people might choose to do with it given
the chance. Conventional EQs never have a resonance control like this,
and the Q control of a parametric just isn’t the same thing.
At this point you might be thinking that this is all very well, but why does
the Mutator only have a low-pass filter; why doesn’t it have high-pass
and band-pass too? The answer to this is that low-pass is the function you
will probably need most; band-pass is available (hopefully with variable
Q) on your console already, and you already know how to flip a low-pass
filter into high-pass mode without too much difficulty. Don’t you? (answer
supplied later...)
Although the LFO and envelope follower functions of the Mutator are
not strictly necessary for Extreme EQ, they are still useful to the sound
engineer (and even more so to the creative musician). Once you have
found a useful low-pass filter setting it is a worthwhile bonus to set a
small amount of LFO modulation so that the cut-off frequency isn’t static
but changes over time. This makes the sound just that little bit more
interesting than it would otherwise have been. LFO modulation can be
applied to the level too if you wish.
My feeling on the envelope follower is that it is best used with an external
trigger that matches the rhythm of the music, a drum track for example.
You could then apply the envelope of the drums to a pad or continuously
sounding instrument in a similar way to triggering a noise gate from an
external source, except that Mutator is more versatile. Those who wish to
take Extreme EQ to the ultimate will probably also take advantage of the
optional MIDI input which can control a number of Mutator’s parameters,
including the ability to allow the cut-off frequency to follow MIDI note
number.
Wishes
Basically, I feel that EQ as commonly found in audio equipment is just
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too tame. High end mixing consoles do offer very musical and versatile
EQ sections, but mid range consoles don’t really have the ability to
change a sound, only to modify it. As a starting point I’m not looking
for much, just a 24 dB/octave variable frequency low-pass filter with
resonance control added to the conventional EQ on a mixing console, and
available on outboard EQs and multi-effects units.
These filters are more commonly found as plug-ins, but hardware filters
are desirable too. We live in a new era of creativity and we need not only
new tools, but well-known, tried and tested facilities presented in an
easy-to-use form. It’s nice to have these things available as outboard and
software plug-ins but I would like particularly to see the development
of a mixing console specifically designed not just for Extreme EQ but
for Extreme Creativity in the recording studio where, in addition to
conventional mixing, a selection of simple but powerful tools could be
right there ready to be used in new imaginative ways.
Bonus Technique - Turn a low-pass filter into a high-pass filter
As mentioned earilier in this appendix it is possible to turn a low-pass
filter, such as that of the Mutator, into a high-pass quite easily.
You can do this too with a conventional EQ section, hardware or plug-in.
Connect the signal to be filtered directly to one channel of the console
(hardware or virtual) and parallel it across to another. Mix both together at
the same level with the phase of one channel (either one) inverted.
With a neutral setting on the filter no signal should be heard since the
two identical but opposite phase signals cancel each other out. But if
you reduce the HF EQ on one channel so that fewer high frequencies are
present, they will not cancel.
The result, allowing for a little unpredictability due to the phase
characteristics of the unit as a whole, is a variable cut-off frequency highpass filter.
You can go this with plug-ins too.
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Conclusion
Thank you for reading An Introduction to Equalization. However, there is
something else you have to do now...
You have to put these techniques into practice. And you have to listen.
This e-Learning Module has introduced you to the foundation knowledge
of equalization. However, the only way you will become skilled in this
difficult art is to work and listen, work and listen...
You can learn equalization with both hardware and software EQ’s.
Hardware is easier - turn a knob and hear the sound. Plug-ins are often
more flexible and versatile, but you don’t have such hands-on control.
And you can’t help but be distracted by looking at the monitor.
Either way however, the more you practise equalization, and the more
intensely you listen to the results you are getting, the faster you will
progress to being master of the EQ arts.
Good luck!
David Mellor, Audio Masterclass
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