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Second Edition


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Foreword

W

elcome to the First Edition of Mastering with Acustica Audio, An Essential Guide.

It took us about three months to get the whole work done. We split this guide up into

two sections: the first section was written by Andrea Zanini and translated into English byAcustica; then we added a second section of interviews with people who have been working with us

through the years and are now part of our “team”.

We hope this guide will give you practical, straight-forward advice on coming to terms with mastering.

“I feel humbled to have been given the opportunity to write this brief guide about mastering, an

enormous subject that would require a hundred books only to start scratching its surface.

Legendary Indian artist Zakir Hussain says that “The best thing you can do, is try to be the best

student you can ever be”. I think this approach applies to any field in music, from studying an in-

strument for the first time to mastering a whole record.

Being given the privilege to “shape” someone else’s music and to listen to it before anyone else, is

something that should be accepted with respect, joy and simplicity.

This is why this guide’s purpose is just to get the reader started on this matter, giving him or her the

basic knowledge upon which to build a creative, critical and personal thought.

Music is no competition. It is a lifelong personal evolution”.

Andrea Zanini

III


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Acknowledgments

I

would like to express my deep gratitude to the Acustica Audio family for their patient guidance, en-

thusiastic encouragement and useful critiques of this work, as well as their advice and assistance in

keeping my progress on schedule. I would also like to deeply thank my friends Earle Holder, for be-ing such a mentor to me, Martin Walker, Eddie Bazil and Simon Barden for their invaluable teachings,

trust and support and Paavo Jummpanen, for his precious tips and knowledge. My grateful thanks are

also extended to Steve Turnidge, a real source of inspiration to many engineers and artists out there,

including me.

Finally, I wish to thank my wife and my parents for their invaluable support throughout this work.

Always Keeping Learning,

Andrea Zanini

IV


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About the Author

A ndrea “Andy” Zanini is a Record Producer & Mastering En-

gineer based in Italy. He started his musical education in

1988, at age two, studying drums (his main instrument) and

piano. He received an intensive education in Indian Classical mu-

sic as a tabla player, production and audio engineering. Founder

of Owl Mastering studio, he is highly regarded for his musicianship,

fresh enthusiasm, and meticulously innovative approach to master-

ing and audio production in general.

V


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Table of contents

Mastering with AcusticaAn Essential Guide

VI

-Introduction...................................................................................................................................................................0007

-Analog or Digital?

-The Mastering Engineer’s Mindset

-Mastering Tools

-Equalization....................................................................................................................................................................0014

-Frequency Ranges

-Resonances

-M/S (Mid-Side) Tecnique

-Stem Mastering

-Magenta

-Amber AC-55

-Green

-Compression................................................................................................................................................................0046

-Dynamics Pocessor

-Types of Compressors

-RMS vs Peack

-Compressor Controls

-Release Time

-Side-Chain Control

-Look-Ahead Function

-Parallel Compression

-Multi-Band Compression

-Limiting


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-Titanium

-Stereo Processing.......................................................................................................................................................0067

-Sample Rate-Dith Depth Dithering.......................................................................................................................0068

-DDP and Track Sequencing......................................................................................................................................0075

-DC Offset.........................................................................................................................................................................0078-Gain Staging. What is it and is why is it important?..........................................................................................0079

-Ear Training....................................................................................................................................................................0082

-The interviews

-Michael Angel

-Ali Zendaki

-Bob Davodian

-Mark Drezzler

-Gabriel Schwarz

-Hubertus Dahlem

-Ken Suen

-Franz Mikorey

Mastering your music is like mastering your life. It’s amazing what happens when you clean up the

noise, maximize your good work, and have your music sparkle and shine as you really want it to.

Steve Turnidge (Mastering Engineer/Author)


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Mastering with Acustica / 008

Introduction

A

lthough it should be called pre-mastering, mastering is often seen by many as a ‘dark art’

or ‘science’ that is difficult to master and is limited to only a select lucky few. Indeed, this

discipline requires specific knowledge and no-compromise equipment in order to achieve a

professional result.

The role of the Mastering Engineer has changed and evolved over the years. They not only need to

have ‘big ears’, but also have the delicate task of enhancing any audio material entering the studio so

that it sounds optimal for the widest number of reproduction systems in the world. This Quality Con-

trol process ensures that specific required technical and artistic standards are achieved.

The tools available to a Mastering Engineer are not so different from those used by a Mixing Engineer:

compressors, equalizers, harmonic exciters, filters, and even reverbs. What distinguishes these de-

vices is their high accuracy and internal architecture, specifically designed and built for this purpose.

Especially when analog, they can be very expensive, or even prohibitive, if we are talking about ‘pro-

fessional’ mastering.

You can spend up to several thousand euros on a stereo tube EQ. A high-quality vintage compressor

could easily run into thirty thousand euros, not to mention the fact that these ‘components’ are worth-

less if they are not interfaced together by a high-level console and esoteric converters.

Then, on top of that, you have to add the wiring costs and the expenses for accurate monitors and

an acoustically treated listening environment, so that the Mastering Engineer’s ears can ‘make’ precise

decisions.

TIP: Read Acustica’s Manifest on the website.


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How come that many professionals in the field seem to regard digital as a second-best option? Much

has been said and written on this subject and a number of myths have developed due to a lack of

information and the production of great works discrediting all those means which, although efficient,

are within everyone’s reach.

Pure digital plug-ins, especially those designed for professional use, are irreplaceable tools in a mas-

tering studio when it comes to high-precision balancing corrections, which would be inconceivable in

the analog world.

“Never turn your back on digital” - Bob Ludwig - Gateway Mastering

Beyond the arguable views and reasons lying behind this debate, there is a kernel of truth to it. The

analog world is characterized by a series of factors (choice and quality of components, circuit archi-

tecture, and so on) and by ‘ imprecisions’ giving the sound a character that can hardly be reproduced

in the digital domain. Whether it is called ‘warmth’, ‘vibe’ or ‘three-dimensionality’, it is not always

easy to describe the euphonic effects that, unlike static digital creations, seem to make the sound

fuller. Tube or tape saturation is the most striking and ‘emulated’ example, although not always with

satisfactory results. Whereas sound is ‘developed’ in the digital domain, it is ‘bent and shaped’ in the

analog domain.

Analog or Digital?


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The Mastering Engineer’s Mindset

A Mastering Engineer is not a wizard or a magician with a one-size-fits-all magic formula. This idea has

been perpetuated in recent years and has led to many false beliefs, fallacies, and legends that can be

easily debunked by simply using well-trained ears.

“Think Holistically” - Bob Katz - Mastering Engineer - Author - Digital Domain

If you want to become good at mastering, you need to learn to listen carefully, with a critical and per-

ceptive ear. You need to learn to ask yourself some questions that can help you in the process, such

as: “What is wrong with the balance of this piece?”, “What can I do to make this music more alive and

vibrant?”, “Sibilants in this voice are terrible, am I able to intervene without compromising the rest of

the instruments or would a remix be better?”, “Does the sound I’m getting fit to the genre?”

With a bit of time and experience, anyone can learn how to master professionally. It’s all about intui-

tion, knowledge and the desire to patiently learn the concepts underpinning this discipline.

First, you become a proficient mix engineer. Then the creativity kicks in and the label of producer

takes over. It is only then that you understand the subtleties of what it takes to achieve the next

stage: Mastering.

Eddie Bazil (Sound Designer/ Author - SampleCraze/Stretch That Note)


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EQs allow to intervene on the frequency content of the audio material. The human ear can hear

sounds whose intensity and dynamics are in the decibel range of about 0 to 120 dB (the threshold

for causing permanent hearing damage), within the ideal 20Hz-20000Hz frequency range. Mastering

EQs usually have an extremely linear and much wider frequency response. Thanks to these devices,

you can ‘balance’ the frequencies so that your track (and tracks between them) sounds in the best

possible way.

Equalizers

Mastering Tools

Compressors control the dynamics of a track. They can be used to smooth out signal peaks and give

the sound more body or reduce the difference in volume between the loudest and softest bits in the

same piece of music. They are perhaps the most ‘characteristic’ tools used in mastering, because they

are often creatively used to give a static and lifeless piece a specific colour. There are several types of

compressors and each can be used for different purposes (see page TBD).

Compressors

Limiters can be considered as extreme compressors with very fast attack and release times. Their pur-

pose is to prevent signal levels from exceeding a set threshold. They are usually used at the end of the

mastering chain (before the SRC and dithering) to increase the final volume of a track, adapting it to

the required tastes and standards, without ever exceeding the dangerous threshold level of 0 dBFS,

over which digital distortion will ruin the overall work. One of the features of a well-designed limiter is

that its use may seem transparent to the ear, even when a peak reduction of several decibels occurs.

As a result of the excessive use of these devices, side-effects such as distortion and loss of detail will

be inevitable.

TIP: Use a compressor to add some character or alter the dynamics, a limiter to provide a more transparent result.

Limiters


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Reverbs are more rarely used in mastering, especially convolution reverbs (able to recreate the natural

ambience of real spaces such as halls, small or large rooms, etc.), in order to give some dry and sterile

recordings additional width and depth. These tools are also very useful to soften the sound or ‘restore’

a sense of environmental coherence of all the elements of a track and widen the stereo image subtly.

Reverbs should be used with extreme caution or the ‘cure’ will be worse than the ‘disease’. Moreover,

not all Mastering Engineers agree on their effective use.

Reverbs

Stereo processors intervene on the stereo image of the audio material, by widening and opening it.

Avoid overusing them because significant phase problems and loss of detail may occur. They are the

sworn enemies of mono-compatibility. Middle ways are always to be preferred. Using the ears and

experimenting are normally the winning formula.

Stereo Processors

Exciters are processors using different processes (such as harmonic synthesis, dynamic equalization

and harmonic distortion) to add emphasis to sounds, making them richer and brighter. Originally used

to restore the high-frequency content of tapes without increasing the background noise level (unlike

equalization), today they are used in mastering as a creative effect, in order to increase the sense of

loudness and presence of the sound and add brightness to the highs or body to the lows.

They give excellent results when used in small doses, or they will cause fatigue to the listener who,

instead of enjoying the music, will only have a severe headache.

Exciters

In addition to a listening environment that is free from acoustic phenomena such as resonance, flutter

echoes and standing waves, it is important to monitor the sound on speakers (active or passive) and

quality headphones. Contrary to what many claim (that headphones are to be avoided in mastering),

needless to say that this concept is misleading, if not totally wrong.

Monitors and quality headphones


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Although it is true that sound reproduction by headphones is different from loudspeakers, especially

stereophony (appearing more natural on speakers), it is also true that a lot of important details, such

as clicks, extraneous background noises, some subtle dynamics and even serious phasing issues are

heard more clearly on headphones. Moreover, if the mastering room is not optimized across the entire

frequency spectrum, headphones help ‘bypass’ some deficiencies in the working environment, which

can be ‘mentally compensated for’ when getting back to the monitors. So there is no reason why you

should remove this valuable tool from your workflow. You just need to learn how to listen ‘through’

both systems and ‘take note’ of the information brought to your attention.

Meters are very useful tools because they allow an engineer to both listen and ‘see’ what happens at

the level of the sound. There are several types of meters with different purposes, among which meet-

ing some technical standards required on a worldwide basis. Among the different types of meters,

there are:

VU Meter: The analog (or digital) meter used to measure the ‘volume’ of a track, meaning by this its

RMS (Root Mean Square) value. The higher this value, the higher the sound perceived by the ear, com-

pared to another whose RMS value is lower, with the volume settings being the same in each case. The

VU Meter and Level Meter are equivalent, except that the latter measures peaks and the RMS value

placing them on a digital scale with a maximum level of 0 dBFS.

BBC Meter: Very similar to the VU Meter, but it is mainly used in the broadcast sector.

Spectrum Analyzer: It is a must have in any mastering studio. This meter displays, on two axes, the

frequency content of the track you are burning.

Spectrogram: It displays the frequency content of the audio material over time, so that the energy

content in each frequency range is displayed on a variable color scale. This tool is very useful to bring

to the surface hidden resonances and, in general, areas that could benefit from additional compres-

sion/equalization.

Phase Meter: It provides a measure, on a scale ranging from -1 to 1, of the degree of ‘mono-compati-

bility’ of a mix (where 1 is mono and -1 is completely out of phase). Generally values between 0.5 and

1 represent a good compromise between a good stereo image and acceptable mono-compatibility.

Dynamic Range Meter: It measures the maximum peak and RMS levels, but also the difference be-tween them, the so-called ‘Crest Factor’, not to be confused with the headroom, which refers to the

‘exploitable space’ between the highest peak value at the absolute limit of 0 dBFS. A track may have

an average crest factor of 12 dB and a null residual headroom (maximum peak at 0 dBFS) and still

sound lovely, whereas another track may have a headroom of 10 dB and a crest factor of 3 dB and

have non-existent and muffled dynamics.

Meters


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LUFS Meter: It measures the loudness as perceived by the human ear, according to the ITU-R BS.1770-

3, EBU R128 standards. Many radio and TV stations are adhering to this standard so that the audience

can enjoy the sound without annoying ‘jumps’ in volume. The target level is set to -23 LUFS.

TIP: The procedure of loudness normalization has become automatic for all broadcasters where the standard is

mandatory. As a result, it is totally useless to produce masters with high volumes and squashed transients, because

their volume will be lowered (normalized) to -23 LU. In a few words, a rock mix will sound as ‘loud’ as Brahms’ Ninna

Nanna in order to have a uniform perception of the sound, without the audience having to use the remote control

for frequent volume adjustments.

Focus on the dynamics quality and do not enlist for the Loudness War! If you are curious to know what

the dynamic range of your favourite album is, you can have a look at: http://dr.loudness-war.info/


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Equalization

“Mastering is 90% EQ” - Craig Anderton - Mastering Engineer - Writer - Educator

Equalization is perhaps one of the most difficult aspects to ‘master’ in the world of mastering.

Even the slightest variation has a crucial impact on what the track will eventually sound like. In

addition to one’s personal taste, this makes you stumble around in the endless search for the

perfect sound. Moreover, even the most accurate listening system has its own way to represent the

sound… which the ear has to get used to.

TIP: Both on headphones and monitors, be careful not to go overboard with volume. Besides being dangerous,

your ear fatigue will cause you to make wrong decisions that will mean you having to go back to them at a later

stage.

There are several types of equalizers. In mastering you usually use linear-phase digital EQs and digital

or high-priced analog parametric EQs. Without focusing too much on technicalities whilst still covering

the main practical aspects, you need at least to become familiar with the technical terms and func-

tions associated with equalization.

TIP: Cut/boost levels at different frequencies are usually very subtle in mastering. Provided that the mix spectrum

is well-balanced , even differences of half a decibel could be significant and ‘shift’ the listener’s focus to some ele-

ments over others, as well as alter the track’s character. If a mix is unbalanced, stronger corrections can restore the

order of things (always limiting EQ adjustments to a maximum of +/- 1.5 - 2 dB makes no sense at all). It is clearly

better to go back to the mix and correct the causes of significant imbalances. Mastering can make the difference,

but you cannot always expect miracles, can you?


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Below is a small glossary of the most commonly used terms in the EQ world.

Filter: A circuit which alters a l imited range of frequencies.

Bandwidth: The width of the frequency range altered by a filter.

Peaking Filter: A filter which boosts (or cuts) a specific band of frequencies.

Centre Frequency: In a filter it is the frequency at which a Peaking Filter applies maximum gain (e.g. a

gain of 2 dB in a filter with a centre frequency of 220 Hz will apply a 2 dB boost at this frequency and

a smaller boost at the surrounding frequencies based on the Q value).

Q Value: A measure of the width of the ‘bell’ of a Peak Filter (no wonder it is sometimes called Bell

Filter). High Q figures indicate a narrow bell, whereas low Q figures indicate a wider bell.

High-Pass Filter / HPF: A filter which progressively attenuates frequencies below a certain frequency.

Low-Pass Filter / LPF: A filter which progressively attenuates frequencies above a certain frequency.

Band-Pass Filter: A filter which allows a certain range of frequencies to pass.

N.B. A first-order filter produces a roll-off of 6 dB/oct.; a second-order filter has a 12 dB/oct. slope; a

third-order filter will have a 18 dB/oct. and a fourth-order filter 24 dB/oct. In some digital EQs, you can

also find slopes of 48 dB per octave but such steep slopes may often cause unnatural or undesired

effects (resonances).

(High / Low) Shelving Filter: A filter which boosts or cuts the signal beyond a certain reference fre-

quency.

Cut-off Frequency: The frequency at which a HPF or LPF has attenuated the signal by 3dB.


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Frequency Ranges

What is essential in the equalization process is to learn to precisely identify the frequency ranges

where the different elements of a mix ‘reside’. The older Hi-Fi systems, which you can still find in some

houses, often feature control knobs saying everything and nothing about their use: they are called

Lows / Mids / Highs. Kind of vague, isn’t it? If you want to make your way in this discipline, you defi-

nitely need to start thinking at least in terms of Lows, Low-Mids, Mids, High-Mids, Highs. You will start

from here to increase your ‘array’ of frequency intervals.

TIP: One of the most common mistakes beginners make is thinking that an increase in high frequencies corre-

sponds to an automatically higher ‘definition’ of a mix, only to then find out that their dog is suffering from a terrible

headache.

Let’s start to split our ‘action field’ into small groups:

20 – 40 Hz : This is the lowest frequency region where you can still hear the sub-harmonics of some

instruments such as the bass, kick drums, synthesizers, subkicks, etc. HPFs are very often used to

limit (or filter out) this range of frequencies. Our suggestion is tdecide what to do each time. In vinyl

mastering you try to limit the amount of energy in this frequency interval (as well as high frequencies),

in particular in the SIDE channel (see page TBD), in order to avoid potential vibration of the cartridge

while playing the records. Elliptical EQs, which deliver these frequencies in mono, are therefore used.

40 – 80 Hz : The ‘lows’, as they are often called.

This is perhaps one of the most favourite frequency ranges, as it adds more power and body to the

sound. How many times did we bump into consumer headphones whose strength (or weakness, we

should say) lies in the ‘bass-boost’? The problem is that this frequency range is very difficult to govern

so as to sound good on most listening systems. Cheap speakers rarely extend up to 60 Hz, not to

mention the full scale. Since very common genres of music, e.g. Rap, EDM and Hip-Hop, often have an

extra amount of energy in this range, the aim is to achieve a balance that gives body without muffling

the sound.

80 – 250 Hz : The warmth and body of many instruments reside in this frequency interval. The guitar

and bass strongest frequencies is around 100 Hz.

The lowest frequency on a guitar is around 80 Hz. The vocal warmth, as well as the snare drum body,

often resides between about 200-220 Hz. This frequency range is often reduced to make vocals more

intelligible in some musical genres. Sometimes the solution is to increase the distinction between the

bass and guitar bits by reducing the signal to around 100 Hz, yet with the risk of excessively ‘depriving’

the mix of its basis. In any situation your ears will be the final judge (a practical example of a track witha good measure of energy in this range is ‘Forever in your Eyes’ - Mint Condition).

250 – 500 Hz : Low-Mid frequencies include spaces, mid tones of basses, acoustic and electric gui-

tars, snare drums, and add more tonal consistency to instruments such as the double bass and piano.

Even the drum toms have a considerable content of these frequencies and their excessive energy


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may make the sound muddy. No wonder this range is often slightly attenuated in the tracks of the

drum mics. This helps focus better on cymbals and drum attacks (a good example is Bruce Hornsby’s

famous track ‘The way it is’, which is rich in these frequencies).

500 – 2000 Hz : Two complete octaves whose ‘focus’ is crucial to a successful master. We are talking

about Mid frequencies. In many commercial masters, both on peaks and the average level, you can

hear how these frequencies are (or sound) hollow. Whereas it is true that too many Mids may make the

sound nasal and messy, too few of them will make the mix sound really weak and ‘precarious’. This is

a fulcrum range where a good balance will provide a good starting point for the master to sound good

almost anywhere. One of the most common problems typical of these frequencies are resonances

(see page TBD), which can be partially treated and rectified at the mastering stage, but they should

ideally be identified and fixed during the mixing process (listen to the track ‘You’re the voice’ by David

Foster to have a general idea of how, in a mix, the predominance of frequencies in this range affects

the final result).

2000 – 4000 Hz : The High-Mids are essential to the percussive attack of bass drums, snare drums

and toms. An increase to around 2700 – 3000 Hz (a very sensitive frequency range for the human

ear) may help increase the ‘projection’ of electric guitars or a piano’s high notes, as well as the vocal

strength. Excessive High-Mids will make the mix sound difficult to hear and quite exhausting (e.g.:

‘Swerve City’ - Deftones).

4000 – 6000 Hz : Definition and strength of melodic instruments and vocals. Do not exceed or your

ears will ‘bleed’ due to a cutting and sharp sound.

6000 – 12000 Hz : These frequencies add detail and sparkle but… careful with sibilants! It is not un-

common to hear consonants and sounds like ‘ch’ and ‘s’ (around 6-7 kHz) being really out of control.

Using multiband compressors and dedicated de-essers in mastering can be a solution (not the

best one, though) only if you cannot go back to the mix and deal with these problems. Many Mas-

tering Engineers refuse to use de-essers since they also significantly affect other elements of a mix.

Not only that, but a significant reduction in sibilance would make the singer seem to have… ‘a lisp’ (a

tangible example of the problem arising from sibilance is the track ‘Heatwave’, composed by Dave

Stewart & Barbara Gaskin).

12000 – 20000 Hz : This range imparts a sense of openness and ‘air’ to the sound, in particular on

vocals and cymbals.

In conclusion, it is clear how you cannot act on a single element of a mix without affecting the others.

Mastering is the art of compromise, where each action has multiple outcomes. Some are choices of

production, others are driven by personal taste, some others are caused by misjudgements, but per-fection does not exist!

… “one of the most important aspects of “Mastering” is the correction of the spectral content of an

audio track. The relationship between the frequency range of 20 Hz to 20 kHz should be even and

well balanced. If this initial integral function is ignored you will find that there is a need to test your


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mastered project on a variety of speaker systems.

The genre of the music has no bearing on the final output even if the genre calls for heavier bass, lower

mid range, or sloping highs.

Additionally, it does not make a difference whether your mastering chain is digital or analog. It is very

important that sound is distributed equally throughout the audio spectrum. Balancing the harmonic

content of your track should be a function you become extremely familiar with”. Earle Holder - Mas-

tering Engineer ( HDQTRZ Studios - Public Enemy - Paul B Allen III).

Basic EQ Quick Tips:

Filtering out / attenuating frequencies below 25 – 35 Hz may help clean the mix up of useless noise,

which only sucks out the headroom and is of little help to the musical information, especially in fast

tracks.

In general, it is always better to attenuate frequencies above those of interest rather than

amplify the frequencies you want to hear.

If your mix sounds too clustered, try to attenuate the 100 – 300 Hz range by 1-2 dB.

If your mix sounds too nasal, try to attenuate to around 700 – 1250 Hz.

If your mix sounds too sharp, it is because of the frequencies lying between 5000 – 8000 Hz.

If you want more detail, try with a slight increase (0.5 – 0.8 dB) to around 8000 – 10000 Hz.

If your mix sounds a bit dull, try with a slight increase to 12000 – 15000 Hz.

Use your ears.

Train your ears.

Protect your ears.

Experiment!

TIP: A good tool to keep on hand is this useful interactive chart graphing many instruments’ ranges

across the full audible range.

http://www.independentrecording.net/irn/resources/freqchart/main_display.htm


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Often overlooked and wrongly underrated, resonances are like ‘woodworms’ devouring ‘useful sound’

and taking up precious space. They may lie in any of the above-mentioned ranges, but become par-

ticularly challenging in the mid-range. Resonances have diverse and sometimes unusual profiles, and

it is not always easy to find out what causes them: persistent frequencies ‘ringing’ throughout the track

without ever stopping (some are even off-key, so to speak), ‘puffing’ noises (like when you blow into a

bottle, a glass or your slightly concave palm placed on your mouth), static noises, etc.

An accumulation of resonances caused by different elements of a mix may seriously ‘congest’ one or

more areas of the track spectrum, and not even spasmodic attempts of accurate equalization may

help without distorting the sound and altering the balances. What could be fixed by simply taking rou-

tine measures (such as, for instance, filtering out the useless low frequencies) during the mixing stage

becomes seriously challenging in the mastering process. Not only that, but getting rid of these hidden

enemies can make the difference, at the end of the work, between a good master and a simply great

one. Think about it!

Resonances

M/S (Mid-Side) Technique

What we are going to explain briefly in this short paragraph is an approach used in the mastering pro-

cess, which has made great inroads in recent years: the Mid-Side technique. For those who are not

yet familiar with this approach, we need to introduce the concept starting with some simple formulas

(don’t panic, in practical terms it is easier than it sounds).

Starting from a L/R track, you can decode the Mid component (which contains the information shared

by the Left and Right channels, namely the mono component) and the Side component (the stereo

component, which contains all the differences between the two channels).

Thus we have two simple formulas:

Mid = Left + Right

Side = Left + ( - Right ) = Left - Right (phase-inverted Right channel)

In order to recreate the original track starting from these two components, we need to use the second

(and the last, I promise!) formulas:

Left = (M+S)/2

Right = (M-S)/2 (phase-inverted Side channel)


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The fraction with the number 2 shows that the volume is halved (equivalent to -6 dB).

If you do not want to use plug-ins automatically decoding and recoding the Mid and Side components

of a track and you want to create a custom M/S routing with sends and groups in a DAW (e.g. for stem

mastering), you need to… use maths.

Having the chance to work on the components separately has several pros, especially that of using all

the instruments available on the two components.

By listening to the Side channel in ‘solo’ mode, we can easily identify phasing issues and redundant

frequencies, as well as balance the environments of a mix and make them more uniform, and correct

resonances in reverbs or panned instruments at the sides of the mix. Filtering out of the Side the

frequencies below 40-50 Hz is already a good move for focusing the power of the lows in the centre

(where we find the kick drum and bass, the very heart and basis of the mix); this little move is also

generally well appreciated by Cutting Engineers for vinyl!

Similarly, we can work in Mono to try to improve the separation of instruments sitting in the ‘centre’ of

the mix and correct any imbalance without affecting the stereo components.

Altering the Mid rather than the Side volume, and vice versa, is another way to widen or narrow the

stereo image of the track, however it is always advisable not to exaggerate. Even a single dB variation

may have striking effects.

TIP:If you want to widen the stereo image of a track, try to apply subtle compression on the Side chan-

nel only. In many cases, just a few dB of compression are enough.


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Stem Mastering

Another ‘modus operandi’ at the mastering stage, which could open up a lot of new possibilities, is

called Stem Mastering. Instead of working on a stereo track, you work on the stereo groups forming this

track (stems) and on their final sum (e.g.: drums, guitars, vocals, bass, percussions, etc.). In addition to

this, you can also add the M/S technique previously described to have almost total control over thefinal outcome.

If you have a problem with drums, such as for instance too sharp cymbals, you can intervene on the

specific stem with a stereo EQ or by simply equalizing the spectrum extremes in M/S.

If guitars have too much energy around 200-250 Hz, you can reduce it without affecting the bass,

snare drum body and vocal warmth.

If the bass has fluctuating dynamics, you can compress it without affecting the kick drum, bass synth

or other instruments sitting in the low-end of the track.

The options are endless.


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MAGENTAMastering Equalizer

M t EQ


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Magenta is a simply unique equalizer and the first Acqua plug-in created with the purpose of provid-

ing those who are involved in mastering, both professionally and personally, with a top quality tool.

Magenta has accurately and rigorously recreated one of the most highly revered and most frequently

used tube Mastering EQs in the world. The sound of this unit is easily identifiable and, in some cases,you just need to turn this EQ on and let the sound pass through it to make it clearer, more compact

and lively.

In case you do not have the sort of money required to buy this hardware unit (about 6000 euros, an

amount that is likely to cause many sleepless nights), never mind. Today, it is much more affordable to

get the same sound.

Magenta EQ

Controls

The interface has been ergonomically designed to be intuitive to operate.

Power: Once on the master bus, you will hardly turn it off.

IN Switches: Used to enable / disable the filter or band they belong to. They are also very useful to

temporarily bypass the EQ effect so as to hear the sound before and after the equalization process.

In / Out gains with I/O meters: Used to balance the input and output signal, they can also be used

‘creatively’ to saturate the tubes, thereby increasing the harmonic distortion.

High Pass & Low Pass Filters: Included in the plug-in Mastering Pack. These filters have a remarkable

musicality and, when used wisely in combination with the other bands, they provide an essential

control on curves.

High Pass: 22Hz; 39Hz; 68Hz; 120Hz; 220Hz

Low Pass: 18kHz; 12kHz; 9kHz; 7.5kHz; 6kHz

Band 1. 28 Hz – 1.5 kHz

Selectable Low Shelf / Bell Filter, CUT/BOOST Gain Range -/+ 0-20dB

Bandwidth: Sets the Q values (maximum on the right and minimum on the left)

Selectable stepped frequencies: 28Hz; 38Hz; 68Hz; 100Hz; 150Hz; 220Hz; 330Hz; 470Hz; 680Hz;

1kHz; 1.5 kHz

Band 2 82 Hz – 3 9 kHz


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Band 2. 82 Hz – 3.9 kHz

Bell Filter, CUT/BOOST Gain Range -/+ 0-20dB


Selectable stepped frequencies: 82Hz; 120Hz; 180Hz; 270Hz; 390Hz; 560Hz; 820Hz; 1.2 kHz; 1.8kHz;

2.7kHz; 3.9kHz

Band 3. 200 Hz – 10 kHz

Bell Filter, CUT/BOOST Gain Range -/+ 0-20dB


Selectable stepped frequencies: 200Hz; 300Hz; 420Hz; 600Hz; 900Hz; 1.5kHz; 2.2kHz; 3.3kHz; 4.7kHz;

6.8kHz; 10kHz

Band 4. 650 Hz – 27 kHz

Selectable High Shelf / Bell Filter, CUT/BOOST Gain Range -/+ 0-20dB


Selectable stepped frequencies: 650Hz; 900Hz; 1.2kHz; 2.2kHz; 3.2kHz; 4.7kHz; 6.5kHz; 10kHz; 12kHz;

18kHz; 27kHz

TIP: When you compare the equalized and non-equalized version of a track, you need to make sure

that the levels perceived are the same! At first, your ear will naturally tend to lean towards the higher

volume version of the track, misleading you.

You need to ‘get to know’ how it works on the signal and how you can use its ‘natural curves’ Clearly


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You need to get to know how it works on the signal and how you can use its natural curves . Clearly

your ear will guide you through this process. Let’s see some examples:

Here is Magenta’s response when, having been turned on, no band or filter is enabled. As you can see,

the sound level tends to be slightly enhanced around 50 Hz.

Keeping this EQ intentionally on without further adjustment can indeed be considered as a ‘choice of

sound’ to be made while mastering your track, in order to exploit Magenta’s natural behaviour and use

its typical tube ‘character’.

By increasing the input volume and balancing it with the output volume, you can exploit the natural

tube saturation and see the harmonics rise and ‘peek out’ (can you see the ripples between 4 and 8

kHz?). You will have a fuller and brighter, in other words more ‘saturated’, sound.

Even when used alone, high-pass filters are extremely useful and musical


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With no other filter on, it helps restore a certain degree of response linearity.

These two settings may be useful to attenuate frequencies at the spectrum lower limit for a more de-

fined and focused sound. For faster tracks, the high-pass filter at 68 Hz could be a good starting point

(note that no curve is too extreme).


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Excellent filters on vocals and backing vocals at the sides of the mix and drum mics (especially whenspecifically suited for recording cymbals).


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The first one is very useful in mastering to obtain the soft sound of a gentle ‘roll-off’ on the very high

frequencies and the second one could find a strategic use during a session of stem mastering on the

stereo groups of electric guitars, helping focus on the high-mids (used in conjunction with the HPF at

120 Hz and a gain of a few dB at 82 Hz with a medium Q value to restore some ‘warmth’ and linearity,while the filter cleans the spectrum of useless energy in the low range).

Magenta also provides an extremely versatile control of the ‘tonal shaping’ of a track.

We can recreate:


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Classic ‘Smiley’ or Hi-Fi curves:

Or we can simply play with rather ‘bizarre’ curves, which is not so typical for a Mastering Engineer! The

ibili i ll i dl


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possibilities are really quite endless.

TIP: If you want to keep the character and body of the tape without losing detail on the highs, you can use Magenta


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to create a small pre-emphasis curve and then use one of the many Tape programs for Nebula to ‘exaggerate the

whole thing’ with some saturation and restore the balance on the highs .

The mastering processing chain


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“Have a light touch and get out of the way” Bob Ludwig – Gateway Mastering

There are many different ways of mastering a track. Some approaches can be considered as standard

practice, whereas others require a necessary debate and careful analysis. Variables are so numerousthat cannot be included in a magic formula always delivering the best achievable outcome.

First of all, in order to have an excellent master, you have to start from at least a good level mix, leav-

ing little room to chance and having the right ingredients to achieve a – so-to-speak – ‘professional’

sound. Unfortunately mixes are often sent in for mastering with problems that cannot be fixed without

going back to the mix. Let’s see the most common ones:

Too high volume leaving no room for manoeuvre, detrimental compression and clipping/limiting, stat-ic arrangement with poor element separation, out-of-control lows, loud and disconnected vocals,

excessive sibilance, razor-sharp cymbals, unwanted residual resonances, clumsy editing, remaining

noises and clicks, phasing issues, high-volume Charleston, too short fade-ins/outs (cutting off reverb

tails), large imbalances between the right and left channels, monaural mixes (the claim being that an

impeccable stereo image is achieved), etc.

Be careful, because these problems can only partially be fixed in mastering.

TIP: Try to keep the average volume of the mix around -18 dBFS with maximum peaks at -5/-3 dBFS and always

try to deliver the best possible quality work (generally 48 kHz / 24 bit is enough, but if you recorded at 96 kHz keep

this Sample Rate).

Let’s see some examples of mastering chains to draw on for your works.

Ex. 1: Mid-speed modern rock/pop (drums, bass, two guitars, vocals, various effects) received (export-

ed) at 96 kHz / 24 bit.

Minimal mastering:

1. HPF with cut-off frequency of 25/30 Hz and a slope of 12 dB/octave

2. Broadband compressor (with sidechain filter at 150 / 200 Hz) with medium attack and release times,

ratio 1.2:1 and medium threshold. A few dB of gain reduction in the high volume sections will be good.

Compensate with make-up gain

3. EQ

4. Limiter (ceiling at -0.3 dBFS with a target RMS level of about -12/11 dBFS)5. SRC & Dithering (for the final bounce to 44.1kHz / 16 bit)

Standard mastering:

1 HPF with cut-off frequency of 25/30 Hz and a slope of 12 dB/octave


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1. HPF with cut off frequency of 25/30 Hz and a slope of 12 dB/octave

2. Broadband compressor (with sidechain filter at 150 / 200 Hz) with medium attack and release times,

ratio 1.2:1 and medium threshold, Target Gain Reduction: max 2 dB. Compensate with make-up gain

3. EQ

4. Stereo Processing for a wider stereo image

5. Limiter (ceiling at -0.3 dBFS with a target RMS level of about -12/11 dBFS)

6. SRC & Dithering (for the final bounce to 44.1 kHz / 16 bit)

Creative (loudness-oriented) mastering:

1. HPF with cut-off frequency of 25/30 Hz and a slope of 12 dB/octave

2. Multiband compressor (to level out the dynamics of the bass and bass drum, vocals and cymbals)

3. EQ with particular emphasis on the low end (e.g.: +2 dB at 80-100 Hz with a low Q value) to affect

the behaviour of the next compressor4. Broadband compressor (without sidechain filter) with an attack time of about 50 ms and medi-

um-fast release time to cause a ‘pumping’ effect; ratio 1.2:1 and medium threshold, Target Gain Reduc-

tion: max 1.5 dB. Compensate with make-up gain, if necessary

5. EQ to add air and detail after compression (e.g.: -0.5 dB at 300-400 Hz, 0.7-0.9 dB at 2.7/3.3 kHz,

High Shelf at 8-10 kHz, +1.2 dB)

6. Stereo Processing

7. (Mastering colour) Nebula Programs for Saturation / Tape / Console or Preamp

8. Limiter (ceiling at -0.3 dBFS with a target RMS level of about -10/9 dBFS)9. SRC & Dithering (for the final bounce to 44.1 kHz / 16 bit)

Conservative mastering (without compression):

1. HPF at 30 Hz with a slope of 6 dB/octave

2. EQ (e.g.: -1 dB at 45-50 Hz, high-mid Q; +0.8 dB at 70-90 Hz, low Q; -0.4 dB at 450 Hz, low Q; +0.3

dB at 4000 Hz, low Q; High-Shelf at 8000 Hz +0.2 dB; LPF at 18 kHz)

3. Subtle Stereo Processing4. Limiter (ceiling at -0.1 dBFS with a target RMS level of about -14/13 dBFS)


Simple corrective mastering (scenario: too loud bass drum and weak bass, annoying peaks on guitars,

mix normalized to 0 dBFS, impossible remix):

1. General gain – 5/6 dB. Let’s bring the mix back to an optimal level of functioning to get more head-

room and scope for action2. HPF at 40 Hz, slope 12 dB/octave

3. Multiband compressor: Band 1 (50 – 150 Hz): our aim is to pitch down the bass drum and restore the

bass. Side effects: loss of ‘punch’ in the bass drum. Band 2 (500 – 2500 Hz): fast attack times, medi-

um-fast release time, medium-high threshold (we want to intervene on the guitar peaks and leave the

other elements as unchanged as possible); ratio 1.3:1, Target GR: max 1.5 dB

4. EQ



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g

6. Broadband compressor for a general levelling out before using the limiter (medium-slow attack/re-

lease times; ratio 1.1:1 and medium threshold so as to obtain a max of 1-1.5 dB of GR in the high volume

sections of the track)

7. Limiter (ceiling at -0.3 dBFS with a target RMS level of about -12 dBFS)


‘Optimization’ mastering:

1. HPF at 25 Hz, slope 6 or 12 dB/octave

2. LPF at 18 kHz

3. Identification of any residual resonance on the Mid channel by Notch filtering

4. Identification of any residual resonance on the Side channel by Notch filtering

5. Low Shelf at 75 Hz on the Side channel, medium Q, - 4 dB6. Broadband compressor (RMS) with a max Target GR of 1.5 dB. Compensate with make-up gain

7. Second compressor in the signal path (PEAK Compressor). Fast attack and release times with a max

GR of 2-3 dB to obtain a transparent result

8. EQ


10. Limiter (ceiling at -0.2 dBFS with a target RMS level of about -14 dBFS)

11. SRC & Dithering (for the final bounce to 44.1 kHz / 16 bit).


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AMBER AC-55 Equalizer

Amber EQ


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Amber is an extremely versatile EQ used both in the mixing and mastering stages. The option to ena-

ble or disable the preamp section allows to use the colour naturally imparted by this machine to the

sound at will.

Amber interface is very intuitive. At its sides you can find the input and output gain controls to control

the gain staging.

The first band covers the frequencies from 18 to 450 Hz and can be operated in shelving or peak

modes. Likewise, the fourth and last band covers the frequencies from 1.5 kHz to 25 kHz; Gain range:

+/- 24 dB and +/- 20 dB respectively.

It is to be noted how this EQ has a wide range of uses, so you need to become familiar with it in order

to make full use of its features.

Broad Tonal Character

In this figure you can see how Amber is responding when the first and last band are on. The first band

was set to Low Shelving at 50 Hz (-5 dB) and the last one to High Shelving at 20 kHz (about -6 dB).

Looking at the curve in the graph, you can easily see how the gain reduction affects a larger portion of

frequencies because the curve has a broad trend.

The first band affects frequencies up to about 400 Hz, with a significant action even between 100 and

200 Hz. Likewise, the last band affects frequencies below 20 kHz down to 4 kHz.

By enabling Amber’s preamp section, you can immediately perceive a velvety colour that can be used

to add a touch of character. In the figure below you can see how Amber responds with the preamp on.

Pre On

The figure shows how, with the Q and gain values being equal, the response varies across the spec-

trum. Another ‘inaccuracy’ of its ‘analog’ heart?


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Varying Response

Back to the Future!

Here are the actual settings used on Amber for mastering an 80s-style synth wave track in the studio.

As you can see, the corrections are minor, yet the difference, also due to the colour of the preamp

section, is enough to make this well-balanced track absolutely perfect.

80’s Synth Track EQ Example


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GREEN EQEqualizer

Green EQ


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Green is a great mastering EQ providing extremely versatile tonal control. The curves obtained can be

soft or aimed at more specific corrections.

Like Amber, Green also provides the option to enable or disable the preamp section. When on, this

adds some colour and warmth to the sound, although it is not as clear and visible as with Amber

(which may be more suitable for creative mastering).

Green offers five bands with variable Q values (from 0.4 to 4, including the Shelf setting in the low and

high bands). The gain range rises from -15 dB to +15 dB per band. In the figure below, you can see how

the interaction between the mid-bands can help create more or less complex but precise curves.

Middle Bands

Green and Magenta have one feature in common: they can work in the so-called ‘air band’ (the very

high frequency range from 15 kHz upwards), which is emphasized by many tMastering Engineers to

make the sound ‘crisper’ and provide a sense of ‘air’


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make the sound crisper and provide a sense of air .

Magenta works on 27 kHz, Green on 26 kHz. There is hardly any difference in terms of frequencies,

although the way these two EQs work is different.

Many Mastering Engineers do not agree on the psychoacoustic effect of emphasizing harmonic fre-

quencies out of the audible range (the audible effects are due to the equalization effect on frequen-

cies below 20 kHz), which are often filtered out as a result of the Sample Rate Conversion.

There is little point trying to give ‘definition’ to an old tape recording by emphasizing this range: the

background hiss will only be louder.

In any case, it is better to use a harmonic generator for this purpose. Remember that the largest num-ber of consumer reproduction systems are neither accurate nor consistent with the spectrum limits.

Sometimes it is better to softly attenuate the very high frequencies rather than emphasize them.

In the figure below, Green’s ‘air band’ is shown.

Air Band


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IVORY IAE-2 Equalizer

Ivory EQ


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Ivory is a precise and versatile EQ that helps make your mastering projects smoother and more musi-

cal. On each band you can choose among different Q values and the Shelf with a gain range of -8 dB

to +8 dB on the following frequencies:

Low Frequency 1 (Hz): 19, 22, 26, 31, 37, 43, 51, 60, 71, 84, 98, 114, 134, 158, 185, 218, 258, 305, 364, 435,

540;

Low Frequency 2 (Hz): 21, 24, 29, 34, 41, 48, 57, 67, 79, 92, 108, 126, 148, 173, 203, 240, 280, 332, 400,

477, 572;

High Frequency 1 (Hz): 617, 727, 862, 1k, 1k2, 1k4, 1k7, 2k0, 2k4, 2k8, 3k3, 3k9, 4k6, 5k4, 6k4, 7k6, 9k0, 11k,

13k, 17k, 24k;

High Frequency 2 (Hz): 665, 787, 937, 1k1, 1k3, 1k5, 1k8, 2k2, 2k6, 3k0, 3k6, 4k2, 5k0, 5k9, 7k0, 8k2, 9k7,

12k, 14k, 19k, 27k.

You can use Ivory in many subtle ways, which is very useful when you want to identify annoying fre-

quencies with a reduced margin for error. Ivory’s versatility allows you to “draw” gentle and “tailored”

curves for any kind of sound material.


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In the following example Ivory was used as the main EQ for mastering a pop-rock track.

The original song was already well-balanced (but lacking in character) and the dynamics were far too

broad for this kind of song. The first thing to be improved in the track were mycro-dinamics by means

of two chained compressors: Titanium and Aquamarine.


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Titanium was used to tame extra energy below 120 Hz, which caused some problems in the refrains,

due to the kick drum sharing too many frequencies with the bass in the mix. Frequencies from 100 Hz

to 1200 Hz were slightly adjusted on the peaks.

Aquamarine was used in opto mode to soften the vocals on the track (with a maximum gain reduction

of 1.5 – 2 dB in the loudest sections). It is a Vocal Up mix (the only stored and unchangeable version of

the mix) where vocals were too “exposed” and, although the effect was acceptable on headphones,

on studio monitors it felt like the singer was “distanced” from the rest of the band (this is why it is im-

portant to listen on both systems). Aquamarine’s PRE section was intentionally left disabled, but it was

useful to enable the SCF at 90 Hz to prevent the compressor from running at the lowest frequencies.

The final tonal shaping was left to Ivory that, although used in a conservative way, allowed to achieve

the desired result in a short space of time. Let’s see what choices were made.

Frequencies in the low range were slightly attenuated with a Shelf filter. 1 dB was added at 79 Hz,

which was used as an a posteriori make-up gain for the use of Titanium where, as you can see in the

figure, no make-up gain was applied (a deliberate and intentional choice because Ivory allows you to

choose frequencies with a higher precision). 1 dB was removed at 1200 Hz in order to better expose

the guitar and vocals at around 2300 – 3200 Hz (subtractive approach). A further 2 dB were added at27 kHz in order to “play” with the “air band”, making it sound more open.

For the last creative change, a Tape program of Nebula was used: OTR-15-ATR-0 to add more har-

monic content and enhance the three-dimensional effect due to the dynamic sampling of the tape.


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Compression

Compressors are among the most used (and misused) tools in modern mastering. They are

used for different purposes, from adding colour to altering the dynamics of a track and acting

as a ‘glue’.

Talking about compression in mastering can be extremely easy as well as enormously complex. Leon-

ardo da Vinci is quoted as saying that “simplicity is the ultimate sophistication” and we will try to stick

to this idea.

If I may use an analogy, just as important as the right wine is to a dinner main course, such is the right

compressor to the track it is used on. It is not uncommon to have more than one type of compressor

during the mastering process, with some being extremely sought-after due to their distinctive sound.

In the analog world, there are some revered machines which, although out of production for several

years, still fetch staggering prices on the second-hand market. Just think of the legendary Fairchild

670 compressor, the undisputed king of tube limiting (with 20 vacuum tubes!), which can cost up to

50,000 dollars (slightly more than 40,000 euros). No misprints, unfortunately.

In recent years, the use of compression has laid down the law on the final result of millions of tracks

to such an extent that it has become involved as the ‘accused’ in the case against the Loudness War.

The over-use of compression literally kills the natural dynamics of individual instruments as well as

whole tracks. Killing the sound natural features has several effects and introduces artefacts that make

you lose the musicality of the track you are working on. The stereo image narrows, musical expression

Dynamics processors

is minimized, the balance between the ambience and its elements is so altered that three-dimension-

ality is completely lost. Anyone who really knows how to use a compressor is also able to turn it off at

the right moment.


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You also have to consider that, with the new broadcast and TV audio standards that have been adopt-

ed on a worldwide scale (see, for instance, EBU R128), any audio content is automatically normalized

to be perceived at the same loudness as other contents coming from different sources. This methodmakes it completely useless to try to ‘sound louder than competitors’. In the two figures below, we

show a clear example. Two versions loudness-normalized to -23 LUFS (the reference standard) of one

track by Spandau Ballet and another one by Deftones. We let you draw your own conclusions.

Now, all this having been said, you will study the basic concepts of dynamics processing that will allow

you to make the best possible use of compressors, for both artistic and corrective purposes, without

losing sight of the ultimate goal: to emphasize the musicality of a track.

Types of compressors

Without focusing too much on electronics, whilst still covering the main practical aspects, let’s see the


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g g p p

various types of compressors and their uses.

They use a set of mathematical operations to adjust the input and output levels and each function

generally provides a degree of accuracy and transparency hardly achievable in the analog domain

(especially top-quality digital compressors designed for mastering).

Pure Digital Compressors/Limiters

Providing high accuracy on the attack and release controls as well as on the gain, this circuit design

makes these compressors extremely versatile.

VCAs (Voltage-Controlled Amplifiers)

Their internal architecture is based on tube technology. This type of compressor does not come with

the ratio control, but the level reduction is incremental depending on the input volume. They work

extremely well on percussions, drums and, in general, whenever you want to keep transients intact.

Generally Vari-Mu compressors have higher response speed on the attack and release than optical

compressors (not as high as FETs or VCAs, though).

Vari-Mu Compressors

Capable of very fast attack and release times. Typically they are more commonly used in the mixing

stage, where an extremely creative use may be justified.

FETs

It is interesting to see how they work A light bulb or LED are controlled by the compressor side-chain

Opto Compressors


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It is interesting to see how they work. A light bulb or LED are controlled by the compressor side chain

and a photoresistor in the gain stage responds to their light by varying the gain reduction. Optical

compressors react quite slowly to transients and are often used in mastering to ‘massage’ the track.

This circuit design is also sought-after for the character it imparts to the sound.

A compressor may be able to measure both peak (the highest levels in a very short time frame, usu-

ally transients) and RMS signals. Some compressors offer switchable RMS/Peak operation, others are

designed to work best only in one of the two modes. A tube optical compressor will work well in RMSmode, whereas a FET peak limiter will be suitable for limiting ‘dangerous’ transients.

RMS vs Peak

Compressor controls

Knee

The knee is the point where the ratio changes from ‘unity gain’ to a set ratio. With a hard knee, com-

pression is introduced as soon as the signal exceeds the threshold, whereas with a soft knee, the gain

reduction process starts as the signal approaches the threshold limit, before exceeding it. This makes

the transition softer and the compression effect less perceptible.

Threshold

T e t res o contro e ps e ne t e ‘portion o soun ’ we want to process an is t e eve a ove

which compression starts. If we want to limit signal peaks, we will most likely set a high threshold with

a hard knee. If, however, we want a larger ‘mass’ of sound in the track to be levelled out to our needs,

we should use a lower threshold with a soft knee to soften the compression effect.

TIP: In some cases, two cascade compressors are used in mastering to work on different components (Peak and

RMS) or, in order to reduce the dynamic range, they can be used to split the ‘workload’ so as to make the compres-

sion process as uniform and transparent as possible. Without pumping effects. In general, an overall compressionof 2-3 dB is acceptable. Higher values could alter the original dynamics of the track a bit too much and we need

to decide whether this is what we want. Often in jazz an overall compression of 0.5/1 dB is already considered at

the limit. Compressors should not be used at all in classical music, in particular analog compressors which tend to

colour the sound.

Ratio

There is no threshold without a ratio, which numerically determines the amount of compression ap-


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plied once the audio has crossed the threshold. For example, a ratio of 2:1 means that for every 2 dB

of volume over the threshold, the compressor lets out 1 dB of volume. Compressors start to be called

limiters at ratios of 20:1 all the way up to Infinity:1 (rather extreme). In mastering ratios are kept at very

conservative levels, especially if a chain of multiple compressors is used (it is usual to set a ratio of

between 1.1:1 to 2-2.5:1).

The first important concept you need to understand is that if it is true that high ratio values result in

high compression values, their effect is not necessarily more audible than low ratio values with a lower

threshold, especially if the attack and release times, which we are going to see shortly, are set approx-

imately without considering the speed of the track.

Another thing that is going to sound counter-intuitive is that the higher the ratio, the lower the amount

of compression. How? If, for instance, we have a 16 dB overshoot above the set threshold and increase

the ratio from 1:1 to 2:1, we will have a gain reduction of 8 dB. If, having the same overshoot, we increase

the threshold from 8:1 to 16:1, we will have an additional gain reduction of only 1 dB (compared to 8 dB

in the previous example). Now we see how the difference in the signal reduction is higher in the lower

ratio values than in the higher ones.

By this we are not saying you need a scientific calculator to use a compressor, but with experience

and the right musical sensitivity you will not become a slave to your compressor, as is often the case,

making it your slave instead. Many Mastering Engineers prefer not to rely on numbers, but on their

perception.

Attack Time

The attack time defines the time it takes the compressor to reach the maximum value of gain reduc-

tion, obtained by combining the functions of threshold and ratio, once the set threshold is exceeded.

Returning to the previous example of a 16 dB overshoot with a ratio of 2:1, if we set the attack to, let’s

say, 200 ms, that would be the time it takes the compressor to produce a gain reduction of 8 dB (if the

overshoot signal was constant and static for a period of time sufficient to allow it to reach this value,

which never happens in a song where levels and volumes always change).

In the figures below you can see in detail what has been said so far. We used a standard digital com-

pressor, setting a threshold to -23 dBFS with a ratio of 1:1 and then 2:1. We generated a tone burst

instantly increasing the volume by 16 dB compared to the threshold.


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1.1 (200ms attack)

2.1 (200ms attack)

8.1 (200ms attack)

16.1 (200ms attack)

As you can see, the most critical difference is between 1:1 (i.e. no compression) and 2:1. Compression is

not doubled when going from 8:1 to 16:1. The difference is a mere 1 dB.

Let’s run the experiment again with a faster attack time, 2.5 ms, and the ratio set to 2:1


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We notice that the initial transient has died away and the image is smoothed out. Fast attack times

alter the dynamics of natural transients. In this case, 8 dB of gain reduction were produced within

only 2.5 ms.

The natural attack of instruments such as drums and percussions would be suppressed with thesesettings. Attack times should be quite long if we want to allow transients to pass ‘safely’ and then com-

press the signal as needed. However, very long attack times, like for instance on a snare drum, would

not give the compressor enough time to ‘react’.

TIP: Compressing low frequencies with very fast attack time settings may cause annoying distortion and clicks. This

effect can be deliberately used in mixing (to add more attack to the bass drum, for instance), but not in mastering.

2.1 Fast Attack 2.5ms

Release Time

Generally, the release time is the time it takes the compressor to return to unity gain (to stop com-


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pressing) once the signal falls below the set threshold. If you look at the following pictures, you can see

how the release of a generic compressor behaves once the signal drops back below the threshold. We

used three different release times: 20 ms, 330 ms and 1.5 seconds.

Release 1.5 sec

Release 20ms

Release 330ms

The first thing you notice is how the compressor keeps on working even after the signal has fallen be-

low the threshold, before returning to unity gain. This is why, if we do not want this to happen, we have

to measure out the release time and, accordingly, the threshold, and process only the signal exceeding

the threshold. This feature is clearly emphasized now, because we are using a tone burst; in any case,


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it should always be considered when using a compressor on any type of material, apart from when

we want to deliberately ‘remodel’ the natural envelope of an instrument (as we are going to see later).

As for the attack time, too fast a release setting can also result in unwanted distortion and clicks on low

frequencies, as well as in pumping, especially if the compressor was set to such values as to produce

high levels of gain reduction. A long release time also tends to modify the natural decay of many in-

struments’ sound tail, thereby altering the timbre (which sometimes can be just what we want).

TIP: Both in mixing and mastering, it is important to give a compressor’s release time such values as not to result in a

change to the rhythmic nature of the track. In a few words, we have to ‘train’ the compressor to keep time! ‘Arrhyth-

mic’ release times tend to be clearly perceptible and are not acceptable in the mastering stage, where transparency

should ‘theoretically’ be the rule.

When setting the attack and release times, it is very useful to know roughly how long some frequen-

cies take to complete a frequency cycle. In so doing, even when using a side-chain filter, you will more

precisely identify your scope for action depending on what you want to achieve with your compressor.

Here is a small chart intended as an additional reference tool.

20 Hz – 1 kHz Frequency Cycle Duration Chart

20 Hz : 50 ms 200 Hz : 5 ms

30 Hz : 33 ms 300 Hz : 3.3 ms

40 Hz : 25 ms 400 Hz : 2.5 ms

50 Hz : 20 ms 500 Hz : 2 ms

60 Hz : 17 ms 600 Hz : 1.6 ms

70 Hz : 14 ms 700 Hz : 1.4 ms

80 Hz : 12.5 ms 800 Hz : 1.25 ms

90 Hz : 11 ms 900 Hz : 1.1 ms

100 Hz : 10 ms 1000 Hz : 1 ms

Side-Chain Control

The side-chain allows us to drive the behaviour of the compressor based on an input signal that is

an equalized or filtered version of the main input or any other input signal (more commonly used in


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a equa ed o e ed e s o o e a pu o a y o e pu s g a ( o e co o y used

mixing and for a creative use of the compressor, an aspect that is left out of our analysis).

Compressors, by their nature, respond more to low frequencies than the rest of the spectrum. This

happens for two reasons: low frequencies have much more energy and a longer wavelength than

high frequencies. This makes the signal stay longer above the threshold, once it is exceeded, and the

risk is that the compressor is triggered even when we do not want it. The side-chain allows the input

signal to be filtered so as to prevent the compressor from behaving like this. Note that an equalized

or processed side-chain is always independent of the signal to be compressed. So, even with a high-

pass filter enabled, every audio signal will be compressed according to the side-chain instructions. It

is a common misconception that only the ‘residual’ signal is compressed, leaving the filtered portion

unaltered. We are not talking about a multi-band compressor, where each filtered frequency band is

brought to the ‘attention’ of a compressor operating independently!

De-essers are based on the same principle, but the side-chain is here emphasized in the sibilant

range (6000-9000 Hz) so that the compressor, with fast attack and release settings, kicks in on the

peaks caused by this annoying problem generally localized in this spectrum range.

Look-Ahead Function

Although many compressors are fast enough to respond to sudden volume changes, some transients

in the signal are so fast that trying to compress them can cause very unmusical effects. Imagine there

is a fly zipping past your dish and you clap your hands in an attempt to catch it, but it’s too late. Now,

what if you were able to ‘predict the future’ and know up front where the fly is going to land? By in-

troducing a few millisecond delay (usually 2 to 5 ms) between the signal sent to the compressor side-chain and the actual signal to be processed, every transient will be detected and processed without

risk of artefacts. The look-ahead option indeed allows to use less extreme attack times, with fewer

unwanted effects. In the digital domain, auto delay compensation can make up for this delay.

Parallel Compression

This is a common trick used in mixing on vocals, drums, and rhythm section, but it appears to have

found more fertile ground in mastering. The theory behind this technique is very simple: instead of


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compressing the highest peaks of the signal to add body and sustain to the sound, yet often losing the

attack, a copy of the signal is compressed and then blended back in with the main signal. This allows

to leave the original dynamics unchanged, while adding volume to subtle details that are often left

overlooked.

When should we use a compressor in mastering? Here are some scenarios:

• To make dynamics more homogeneous

• To attenuate the peaks and increase the general RMS of the track

• To ‘create’ a more compact and powerful sound

• To exploit the euphonic characteristics (e.g.: tube compressors)

• (On some stems) To ‘remodel’ the envelopes

• To create a sense of movement in static genres (digital electronic or synth music).

TIP: Never use a compressor only because you have it! And do not compress the left and right channels separately

in order to avoid an image-shifting in the stereo image of the sound.

Multi-Band Compression

In this type of compressors, the input signal is divided by dedicated filters into separate frequency

bands which are then compressed individually. Each compressor operates completely independent-

ly. The bands are then mixed and sent to the output gain control. This type of processor tends to

‘re-equalize’ the sound much more than broadband compressors and should be used carefully so as

not to drastically change the balance of the mix.

Limiting

Limiters are used in mastering for two specific functions: to prevent a signal from going above a set

level (the ‘ceiling’) and to lift the final level of a mix. Although well-designed limiters are very trans-

parent-sounding, it is always important not to be misled by the perception of a higher volume and tolisten with a critical ear in order to see if some important detail is missing.


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TITANIUM Mastering Multi-Band Compressor

The classic sound of analog multi-band compression

Mastering Engineers may not always agree on the use of multi-band compression in mastering, be-

cause these compressors not only affect the dynamics, but also have a strong impact on the mix


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timbre. However, multi-band compression can be applied to specific genres, such as pop, rock, EDM

and hip-hop for artistic effect, especially in the low frequency range, to knit the bass and kick drum all

together or give a static mix more character and life.

Titanium is an excellent creative and corrective tool. You can freely choose whether you want to use

the natural response of its compression or to keep the dynamics under control as transparently as

possible.

Let’s become familiar with the interface.

As you can see, you can choose among three different configurations.

The first one, TITANIUM 1B (One Band), is not a multi-band device, but a stereo version of Titanium.

The second one, TITANIUM 2B (Two Bands), splits the spectrum into two regions. There is a control

knob called X-Over High, which is a crossover filter for high frequencies. Available values are: 1.2 kHz /

2 kHz / 3.2 kHz / 4 kHz / 4.8 kHz and 6 kHz.

At 3.2 kHz, for instance, the High Band will operate on the frequency region above 3200 Hz, the Mid

Band on the frequencies below.

The ON switches on the bands are very useful because, when off, they disable the reference band in

order to better hear what the complementary band is working on.

Main controls (on each band in the two- and three-band compressor):

Threshold (0/-48): This knob sets the level above which the compressor starts acting.

Ratio: Available values range from a minimum of 1.5:1 to a maximum of 10:1.

Attack: This knob sets the attack times. No values, you go by ear! From Fast to Slow.


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Drum Stem

Release: This switch sets the release times. Again, no values, from Fast to Slow.

Gain (0-25 dB): This control compensates for the Gain Reduction carried out by the compressor. Each

band has its set of meters, which are very useful.

Meters I / O / R: They allow to monitor the Input, Output and Reduction levels.

The third configuration is TITANIUM 3B (Three Bands). Now you can apply three different compres-

sions to three different frequency spectrum areas.

The main controls are the same, with two additional knobs: a second crossover filter and a frequency

multiplier (f x 4) which, once enabled, allows the second crossover to have the bracketed frequencies

as reference values.

The second crossover allows to separate the mid-frequency band from the low-frequency band; se-

lectable frequencies (in Hz) are: 60 (240) / 100 (400) / 150 (600) / 200 (800) / 250 (1kHz) / 300 (1.2

kHz).

Here are a few ideas on how to use Titanium in a hypothetical mastering session starting from specific

stems. The following settings have been determined based on the audio files we have been working

on to give you these suggestions. The options are endless, but it is easy and quick to see how many

different paths can be taken!

3B on Triplet-Feel Drum Stem.

This drum stem was essentially static and we had some annoying resonance from the kick drum in the

low frequency range (the kick drum tail was too long and the volume too loud). Although pleasant, the

timbre needed more control and power. The Low Band (from about 200 Hz downwards) was set to

such values as to emphasize the kick drum attack (residing around 70-75 Hz) and ‘squash’ the tail for a

stronger impact. The Mid Band (ranging from about 200 Hz to 4 kHz) steps in slightly and, with mod-


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erate attack and release times, aims to ‘massage’ the kick drum percussive mid-range, making it softer.

The band from 4 kHz upwards was configured to emphasize the snare drum ‘snap’ and the kick drum

beating, with a deliberately long and out-of-time release compared to the drummer’s triplet groove;

the use of the compressor is more evident and especially audible on cymbal tails (creative use).

TITANIUM 1B, followed by Amber Mastering EQ (see page TBD), on a rhythmic jazz vocal group.

Jazz Rhythmic Vocals

We used the Titanium Stereo Compressor to emphasize the labial and dental consonants in some

syllables, in particular ‘Paa’, ‘Baa’ and ‘Daat’, which needed more character.

Corrections with Amber:

• 1 dB at 170 Hz to give a deeper sense of warmth. Low-mid Q

• - 1 dB at 2300 Hz to soften the sound. Low-mid Q

• + 1 dB at 5000 Hz where vocals have gained more detail. Low-mid Q

• - 10 dB at 50Hz (although Amber has a much wider response – see specic section) to at-

tenuate the low-frequency region of the spectrum, where other elements of the mix reside.

• The preamp section was left on to add colour and three-dimensionality.

TITANIUM 2B on Electric Guitar Stereo Groups.


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Stereo Guitar Riff

Titanium works greatly on electric guitar groups, both to control the attack and add more ‘grit’, without

necessarily resorting to a static EQ or effectively compress the signal. These settings have turned a

rather muddy group of three Fender Stratocasters into a roaring lion (we operated Titanium from 2 kHz

downwards, trying to emphasize the content of the complementary band, kind of a reverse approach).

TITANIUM 2B Conga Stem.

Conga Stem

Congas tend to be rich not only in harmonic content, but also in the frequency range 125-400 Hz.

To work on this group, we chose to use Titanium mostly below 1200 Hz: a relatively high ratio, fast

attack and slow release times. The conga body was extremely robust and the dynamics a bit variable

(the beats were not clearly audible in the mix and blended in too much with the drum sound). The

compressor was set from 1200 Hz upwards so as to emphasize the strokes made with the palm of the

hand. The result is a fuller, clearer and more defined sound.

Note: Indian percussion instruments produce a similar but more ‘sustained’ sound. This is the case of

tabla drums, where transients should be clear, clean and well-defined because they correspond to

specific beats in the traditional forms of Indian music, such as the insanely fast ‘tabla relas’, where a

wide variety of stroke patterns are used. If you happened to work on a track with this instrument, which

is more and more used in jazz fusion, Titanium would serve as an excellent tool to model the sound

even better than an Eq!

TITANIUM 3B on the Stereo Mix.


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This is the most common way to use a multi-band compressor. Some Mastering Engineers use it be-

fore applying a broadband compressor or an EQ preceding another compressor, others do not use it

at all so as not to affect the original dynamics of a track of even half a dB. Many jazz artists do not like

the effect of multi-band compression on their music. It is up to those who employ this tool to find a

suitable use for it.

The settings for this example are much more conservative than the previous cases. Ratios were allkept at the minimum value and we tried to never exceed, in the three bands, a gain reduction of 2 dB.

mastering with acustica 1.4

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