Introducao a equalizacao - em ingles

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An Introduction to Equalization

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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



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.


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 lower-case ‘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.

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.

Heinrich Rudolph Hertz


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.

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

can lids banging together.

Cymbals and metallic percussion are rich in high frequencies


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…

Frequency (Hz)

Level (dB)

20 20,000


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…

Frequency (Hz)

Level (dB)

20 20,000

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…


Frequency (Hz)

Level (dB)

20 20,000

or this...

Frequency (Hz)

Level (dB)

20 20,000

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…


Frequency (Hz)

Level (dB)

20 20,000

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).

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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

•

•

•

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decibels to understand EQ. If you can appreciate the above points, then you know much as about decibels as you need.


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.


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 signal back up again periodically.

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

The telephone - what EQ was invented for


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.


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.

Low-pass Frequency (Hz)

Level (dB)

20 20,000

High-pass Frequency (Hz)

Level (dB)

20 20,000

Band-pass Frequency (Hz)

Level (dB)

20 20,000

Band-stop Frequency (Hz)

Level (dB)

20 20,000

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•

•

•

•


Notch Frequency (Hz)

Level (dB)

20 20,000

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.

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

Neve filter section


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…

Frequency (Hz)

Level (dB)

20 20,000

6121824 decibels/octave

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.

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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.


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.

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

Pullet passive equalizer

An Introduction to Equalization Page ��

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.


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


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…

Frequency (Hz)

Level (dB)

20 20,000

Bell

Frequency (Hz)

Level (dB)

20 20,000

Shelf

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


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 low-frequency 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 mid-range, 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.


Here is a graph showing a resonant boost. It could just as well be a cut, the concept works both ways.

Frequency (Hz)

Level (dB)

0 dB

-3 dB

F1 F0 F2

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


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.


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.


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


– 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


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.


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.


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


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


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.


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


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.

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.

Clarinet and trumpet - how to give each its own space in the frequency

spectrum?


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 full-sounding 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


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


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.


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


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.


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 anti-feedback 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


has been correctly set up, then if it sounds good to you, it will sound good anywhere in the house.


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.


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?


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


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 high-pass filter.

You can go this with plug-ins too.


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

Introducao a equalizacao - em ingles

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