Mastering With Acustica 1.0

First 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 by
Acustica; 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
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
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
Mastering with Acustica
An Essential Guide
Table of contents
-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..................................................................................................................................................................0041
-Dynamics Pocessor
-Types of Compressors
-RMS vs Peack
-Compressor Controls
-Release Time
-Side-Chain Control
-Look-Ahead Function
-Parallel Compression
-Multi-Band Compression
-Limiting
VI
-Titanium
-Stereo Processing.......................................................................................................................................................0059
-Sample Rate-Dith Depth Dithering.......................................................................................................................0060
-DDP and Track Sequencing......................................................................................................................................0067
-DC Offset........................................................................................................................................................................0070
-Gain Staging. What is it and is why is it important?..........................................................................................0071
-The Interviews................................................................................................................................................................0074
-Michael Angel
-Ali Zendaki
-Bob Davodian
-Mark Drezzler
-Gabriel Schwarz
-Hubertus Dahlem
-Ken Suen
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)
VII
Chapter 1
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.
Mastering with Acustica / 008

Analog or Digital?
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.

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)

Mastering Tools
Equalizers
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.
Compressors
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).
Limiters
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.

Reverbs
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.
Stereo Processors
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.
Exciters
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.
Monitors and quality headphones
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.

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
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.

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/

Chapter 2
Equalization
“Mastering is 90% EQ” - Craig Anderton - Mastering Engineer - Writer - Educator
E
qualization 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?

Below is a small glossary of the most commonly used terms in the EQ world.
Filter: A circuit which alters a limited 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.

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 with
a 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

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

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 (HDQRTZ 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

Resonances
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!
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)

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.

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 the
final 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.

MAGENTA
Mastering Equalizer

Magenta EQ
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.
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
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
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

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).

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

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:

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
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
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
“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 numerous
that 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 #11 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:

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
6. SRC & Dithering (for the final bounce to 44.1 kHz / 16 bit)
Creative (loudness-oriented) mastering:

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 compressor
4. 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
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 Processing
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 action
2. 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
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 dB
6. 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).

AMBER AC-55
Equalizer

Amber EQ
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?
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

GREEN EQ
Equalizer

Green EQ
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’.
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

Chapter 3
Compression
Dynamics processors
C
ompressors 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

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.
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 method
makes 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
various types of compressors and their uses.
Pure Digital Compressors/Limiters
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).
VCAs (Voltage-Controlled Amplifiers)
Providing high accuracy on the attack and release controls as well as on the gain, this circuit design
makes these compressors extremely versatile.
Vari-Mu Compressors
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).
FETs
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.

Opto Compressors
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.
RMS vs Peak
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 RMS
mode, whereas a FET peak limiter will be suitable for limiting ‘dangerous’ transients.
Compressor controls
Threshold
The threshold control helps define the ‘portion of sound’ we want to process and is the level above
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 compression
of 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.
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.

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

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
2.1 Fast Attack 2.5ms
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 these
settings. 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.

Release Time
Generally, the release time is the time it takes the compressor to return to unity gain (to stop com-
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,
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
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
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 to
listen with a critical ear in order to see if some important detail is missing.

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
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.
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.
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-
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 specific 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.

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.
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 all
kept at the minimum value and we tried to never exceed, in the three bands, a gain reduction of 2 dB.
Our aim was only to ‘glue’ the mix without depriving transients of attack or character (they were only
slightly controlled in the band from 2 kHz upwards and emphasized in the mid band from 400 Hz to 2
kHz to make them slightly more visible). The low range was made more compact by using slower attack
and release times with a deeper threshold to ‘embrace’ a wider portion of signal. Finally, a light stage
of make-up gain was added.

Chapter 4
Stereo Processing
T
his technique is commonly used in mastering to give a track more three-dimensionality and
‘air’, widen the image, tighten it up, or correct phase issues (especially in the low frequency
region of the spectrum).
There are different ways to work on the stereo component of a track (i.e. on all the differences be-
tween the right and left channels of the mix… Remember the Side component?), but it is useless to
use stereo processors on essentially monaural mixes in the hope of adding a little extra ‘size’ to them.
There is simply not enough ‘raw material’ to feed these processors, irrespective of the way they work
(some use the principles of psychoacoustics, others work on phase relationships or the volumes of
the Mid and Side channels and so on…).
The additive effect of stereo expanders is generally desirable and euphonic, but only when the original
stereo image is strong and compact and can be further emphasized without loss of detail and impact.
Some of these processors also provide an additional stage such as, for instance, the harmonic exciter.
The corrective effect of stereo processors (especially multi-band processors) is particularly useful in
mastering. A lot of mixes have phase problems in the low frequencies around 35-65 Hz that become
extremely clear on headphones and cause the mix to be highly problematic on monophonic systems.
Making this frequency interval ‘mono’ (or reducing its width) can be a good solution.
Some mixes are too wide and lose impact. Narrowing the image can help restore a sense of cohesion,
but sometimes changing the arrangement and the position of some elements in the mix gives far
better results.
The best recipe is to experiment and always check the mix by monitoring in mono.

Chapter 5
Sample Rate-Bit Depth

Dithering
S
ample rate and bit depth are two funda-
mental, but often overlooked, concepts
governing the audio world. A third one,
dither, actually follows them, but we will deal with
it later.
The sample rate can be defined as the ‘rate’ at

which a digital system takes a series of snapshots
(samples) of a waveform recorded within a time
frame. Ideally, the more samples you take, the
closer the sound will be to the original. If you consider the film or cartoon frames, designed to create
the illusion of motion in the audience, it is easy to see how frames should be captured fast enough to
create the illusion of smooth motion. Otherwise the result will be approximate.
The higher the sample rate, the more accurately very high frequencies (having very short wavelengths)
can be captured.
Without focusing too much on technicalities, it is important to know that, according to the Nyquist

theorem, a signal is properly sampled without loss of information if the sampling frequency is equal to
or greater than twice the maximum frequency of the signal spectrum.
Undersampling an analog signal produces frequencies that were not within the original signal, which
can be perceived as a sort of metallic sound. This phenomenon is called ‘aliasing’ and must be avoid-
ed or it will have a significant impact on the quality of the sound.
The bit depth determines the range of amplitude values that can be stored for each sample and its
potential dynamic range, from 0 dBFS downwards. In round figures each bit provides a dynamic range
contribution of 6 dB. This means that a 16-bit recording corresponds to a dynamic range of about 96
dB with 65,536 possible levels, and a 24-bit one offers 144 dB with nearly 17 million possible levels
(16,777,216).
Imagine you are walking barefoot on a fine wet sandy beach. You leave your footprint in the sand and
each fine grain, which will probably get trapped in the folds of your skin leaving a mark, can be consid-
ered as a high-resolution sample. What if you were walking on a pebble beach? Aside from the pain,
your footprint would never be like the footprint left in the soft clay sand and would not be as deep
either. What you would see is just a vague approximation of your foot shape!
The bit depth of an audio signal is originally determined by quality at the recording stage. Today
recording at 24 bit is the standard, while the 16 bit is considered as a target value, when no further
processing of the signal is required.
So, when does this value change? Even at the slightest processing, because algorithms apply func-
tions and multiplication to the signal, whose bit depth is immediately extended. Any time a calculation
takes place, even with a variation of half a dB of gain, this change occurs.
At some point, however, after several chained processing stages, the least significant bits have to be
thrown away until only the bits containing useful information remain (as in the case of the final bounce
to 16 bit). This procedure, called ‘truncation’, simply lops off (truncates) these bits of the signal. A major
problem caused by this procedure is the generation of non-linear distortion and harmonics intro-
duced by a ‘truncation error’ correlated with the main signal. Needless to say, this quantisation error
‘sounds’ horrible to the ear.
1 kHz - 24 bit 1 kHz - 16 bit

Let’s suppose that you have a sine wave at 24 bit and reduce the bit depth by simply truncating it:
you will have quite a problematic scenario. The main signal is still there, but the amount of distortion
introduced has literally obstructed its faithfulness to the original signal.
Here is where dither, one of the most widely misunderstood concepts in mastering, enters the scene.
Dither is noise. How can noise help remove the quantisation distortion and enhance the dynamic
range?
We need to stress that noise cannot always be used as dithering noise, because the latter is generated
by considering statistical probabilities which aim to decorrelate the quantisation error from the main
signal through a deterioration of the signal/noise ratio. In doing so, unwanted distortions are removed.
Just try this simple exercise to get a clear idea of how important this step is.
Starting from a 24-bit file, attenuate the signal by 80 dB. The sound will become almost inaudible.
Save two 16-bit versions with and without dither. Reset and normalize them. Now listen: the undith-
ered version will be distorted and damaged, with the signal sometimes disappearing below the 16-bit
quantisation level, where only absolute digital silence reigns. The dithered version will be rich in noise
(of course, we have deteriorated the signal/noise ratio by 80 dB!), but the main signal will be very easy
to discern and will not contain the same distortion components as the first file.
There are several dithering algorithms that work properly and each one of them colours the sound
differently. With time and experience, you should be able to accurately select the type of dithering
to use, depending on what you are trying to achieve. The simplest algorithm is the TPDF (Triangular
Probability Distribution Function). Various algorithms, for instance, use ‘noise-shaping’ techniques
to shift noise into a less audible frequency range, where the human ear is less sensitive, in order to
reduce subjective noise perception. In the figure below, you can see some of the many possibilities a
Mastering Engineer is often faced with. Again, your ears will be the final judge.

Generally, you need to be careful when using dithering algorithms with a strong boost to high fre-
quencies. Although dithering levels are usually very low and hard to hear (and although the addition
of dither does not correspond to an increase of 6 dB of noise each time, since algorithms are not
interrelated), dithering could eventually become audible, particularly in the cymbal or reverb tails or
when instrumentation is less dense.
Can dithering help ‘extend’ the dynamic range? Here again an example serves to illustrate the point.
We started from a remaster at 96 kHz / 24 bit of ‘Thriller’ by Michael Jackson and reduced the gen-
eral gain of the song by 100 dB, a value below a standard 16-bit file capacity (our intention is to see
whether, following conversion to 16 bit, which does not have this dynamic range, we can ‘gain’ a few
extra dBs of signal by using dithering, enabling a ‘deeper listening’ than a 16-bit file would allow). We
exported two 16-bit versions without and with dither (TPDF). We normalized them with a maximum
peak of 0 dBFS.
What we see is that the ‘normalized’ undithered version does not change. Absolute digital silence. The
file is not able to quantize the information.
MJ - Thriller No DITH - 100dB Normalized

The dithered version enables us to include even more digital information. What we can clearly hear,
although at a lower volume than the noise introduced, is Michael’s voice, the kick drum, the guitar and
other elements of the mix. In a few words, we were able to ‘listen deeper than a 16-bit file would allow’.
MJ - Thriller - 100dB TPDF Normalized
So, when should dither be applied?
It is very commonly said that dither should be used only once, before the final bounce to 16 bit. This
is true (and useful) only if you always work in digital, saving at 32-bit floating point all along the way.
However, it is often used more than once along the production process.
Let’s see the most common cases:
In the mixing stage (ITB at 24 bit), dither (24 bit) is usually applied to the aux sends to hardware pro-
cessors if a plug-in has been previously used (as we have seen, plug-ins increase the bit depth of a
digital signal any time a calculation takes place). In case you use a reverb/digital processor, it would
be better to make sure that dither is also applied to its output so as to prevent the signal from being
affected by quantisation error which, although low, would still be there.
During the bounce, you need dither at 16 bit (after the Sample Rate Conversion) for CD production,
streaming, downloading or the iTunes Standard Master (24 bit for Mastered for iTunes™ Encoding).

“Mastering is not the icing on the cake, or even the cherry on top of it. Mastering should really be the
final polish on that cherry, so that it helps everything in the cake mix to stand out in perfect detail.
I’ve enjoyed albums for years and then heard a remastered version that is a revelation to listen to,
simply because you can suddenly hear a host of previously submerged details.
Don’t assume that mastering always involves boosting the low and top ends and applying mul-
ti-band compression – just listen, then decide if there’s something that needs enhancing. If you find
yourself piling on more than a few dBs of EQ or compression at the mastering stage there’s some-
thing wrong with the mix, so unless you have no choice, go back to the mix first.
Also, unless your ears have a very good memory, refer every half an hour or so to a selection of
carefully chosen commercial tracks that you admire in a similar genre, to avoid going too far at the
mastering stage – it’s easy to end up with a harsh sound that’s initially pleasing, but which soon
becomes tiring to listen to.”
Martin Walker – Author/Composer (Sound On Sound Magazine / Yew Tree Magic)

Chapter 6
DDP and Track Sequencing
O
ne of the most enjoyable steps in mastering, after tracks have been thoroughly subjected to
the labor limae, is when the end product is produced.
Mastering not only means ‘messing around’ with a lot of awesome tools and their little coloured lights,
but also never losing sight of the overall message of the CD.
Mastering a single track is much easier because you have more freedom. What is more complicated
(but also much more exciting) is mastering a track bearing in mind the album’s genre and background,
the features all tracks have in common, such as the main instruments, voice, stereo image, same per-
ceived volume between the tracks, climax, etc.
Unfortunately, the distribution model adopted in the last years (based on downloading single tracks
‘taken out of context’), with rare exceptions, has come at the expense of creating a ‘Glass Master’ for
the replication process.
This feature, which has actually become a rule in the field of independent record industry, has distort-
ed the idea of the album as a collection of ‘stories, colours and pictures’ with specific start and end
points. Up to a decade ago, a single track acted at least as a ‘lure’ to sell the whole album, but now this
has changed. The single track itself has become the end product!
There are several programs and DAWs that allow to create a Red Book Compliant mastering from
single tracks.
So how to proceed? First, you should clearly recall the track sequence. Then, you will be able to place
the right pauses between the tracks: remember that silence in music is as important as notes.

Now listen with your eyes closed and ask yourself some questions:
•What is the feeling you want to convey? Energy (e.g. with short pauses between each track
not giving respite and following the natural rhythm of the tracks)? Calm and peace (long and medita-
tive fade-outs, long pauses, etc.)?
•Is it a live CD? So you need to find the right points where to ‘separate’ the tracks from one
another as naturally as possible. Generally the audience clapping between two pieces is the point
where a track ends and the next starts, but it could also be a start with a drum solo (on two separate
tracks on the album’s CD version but put together in a file for digital downloading, for instance), the
beginning of a singer’s speech in the case of integral live music and so on.
Once this step is completed, you need to export the DDP image file.
What is it? The DDP (Disc Description Protocol) contains and describes a set of data necessary to cre-
ate an error-free Glass Master for reliable CD duplication and printing.
It is always recommended to send a DDP by e-mail rather than delivering a mastered CD: this could
contain damaged sectors that may compromise not only the duplication, but also the playing of the
CD in some players.
By checking the ‘MD5 Checksum’ (one of the files generated as a result of the export) included in the
DDP, you can easily detect any loss of data during the transfer and try it again. It is based on comparing
the string produced by the file sent with the string produced by the file received.
A DDP file includes:
•Files called DDPID, DDPMS, SUBCODES describing the CD

•The AUDIO.DAT file including the audio files
•The MD5-Checksum.md5 file that we mentioned before
•All metadata composing the CD Text Information (correctly decoded by most players), e.g.:
Composer’s Name, Artist’s Name, Track Titles, ISRC, EAN/UPC codes, etc.
The ISRC (International Standard Recording Code), defined by ISO 3901, is a unique identification
system for sound recordings and consists of a 12-character alphanumeric string (in the form: XX – Y12
– 12 – 12345). The first two characters identify the country code: IT for Italy, GB for the UK, US for the
USA and so on. The next three digits identify the registrant code, the next two numbers are the year
of assignment and the final part is the code assigned to the track. The ISRC is issued by the National
ISRC Agencies. Here is the official list:
http://isrc.ifpi.org/en/contact/national-agency-contacts
Should you worry about using ISRC codes?

If you release a CD on your own independent record label and you expect to earn something from it,
then the answer is yes. Using an ISRC has several pros.
ISRC codes are only required to release, distribute, sell or broadcast your music on a radio station (on
the web or not). When an ISRC is embedded into your track and identifies it as belonging to you, it will
help track your sales and the audience ratings in streaming or on the radio. However, ISRC codes are
only required for the final mastered (stereo, not multitrack) versions of your tracks, which will be sold
and distributed. No code is required for demos and other ‘domestic’ recordings.
You also need to know that any further version of the final track intended for distribution, which has
been rearranged, edited, recorded a cappella, remixed and so on, must have its own code.

Chapter 7
DC Offset
T
he concept of DC offset needs to be men-
tioned in any discussion of mastering.
A waveform with a DC offset has different
amounts of signals in the positive and
negative fields: in a few words, its centre
will never be equal to 0 but will be slightly more or
less than this (positive or negative), as a result of a
component of the recording chain introducing this
problem before the signal is converted from analog
to digital form.
Why could this be a problem? First, a signal with a DC offset will never reach its maximum possible vol-
ume before a digital clip occurs, or even if only normalized. This happens because DC offset uses up
useful headroom for the signal, whose maximum peak is closer to 0 dBFS than in normal conditions.
DC offset may also introduce some slight low volume distortion, which can become audible during
the file processing and cause annoying clicks at the beginning or end of each file (or during editing).
You can use a high-pass filter to get rid of this problem and remove low frequencies below 20/30 Hz.
Moreover, there are a lot of detection and correction software that can be used to pre-emptively solve
this problem. Some plug-ins, including sub-harmonic exciters, may also introduce DC offset. It is very
important to remove it.

Chapter 8
Gain Staging. What is it

and why is it important?
G
ain staging is the process of managing the level in each component of the recording and pro-
cessing chain (from preamp to the master bus sum), in order to optimize the signal-to-noise
ratio without incurring distortion. The aim is to have plenty of headroom that, even in case of
unexpected peaks, provides a good safety margin.
This problem is much more pronounced in the analog domain than in the digital domain, where there
is no presence of background electronic noise and the only impassable limit is 0 dBFS. However, the
best approach is not to make this distinction and just use the analog gain staging in the digital domain.
Indeed, we are working more and more with hybrid systems that somewhere in the chain combine the
use of an analog component (e.g. an EQ or a compressor). Besides, many plug-ins (particularly those
emulating the behaviour and curves of hardware units) are generally inaccurate when they receive
tons of gains and can introduce unpleasant artefacts.
Many home recording users are ‘scared’ of conservative input volumes because they fear they will not
be able to achieve appropriate final levels once the mix is completed. This shows not only a lack of
theory underpinning recording practice, but also some inexperience in the production process and its
steps. How many times do you hear people say, ‘If I set a low volume while recording, how can I get
the same sound as if I were listening to a CD’? But then, if you are reading this guide, you probably
already know the right answer to this.

If a signal at 0 VU on one of the meters appears on an analog console, you should bear in mind that
there is roughly 20 dB of headroom left before incurring distortion. Peak levels may exceed 0 VU
without compromising on sound quality because analog circuitry naturally smooths out these peaks
(an effect and colour looked for by many). Remember that VU meters do not measure a signal’s peak
levels, but its RMS; if you want to keep transients intact, you have to keep peaks at a maximum value
of 0 VU.
If you have a tape recorder with a drum track at 0 VU (RMS), don’t be surprised that peak LEDs are
red and that some degree of compression is already being applied, since this instrument can have a
dynamic range of more than 18-20 dB!
To give you a concrete example, 0 VU corresponds, in the digital domain at 24 bit, to around -18 dBFS.
With current converters and DAW resolution, it is perfectly normal to record, for instance, a snare drum
with very low RMS values and peaks at -14/12 dBFS.
Since a value of -10/12 dBFS on a digital scale corresponds to around the nominal peak value on an
analog console, there is no need to record trying to reach the ceiling of 0 dBFS. [Fig. Reference Levels]
If you are working ITB with third-party samples (often sold normalized), it is good practice to use a gain
plug-in as a first insert in the DAW (simulating the input gain of a channel strip of a mixer) and bring
the level to more conservative values, which are more ‘compatible’, so to speak, with other plug-ins or
external hardware, when used. If it sounds too low, you just need to turn up the loudspeaker volume,
don’t you?
To sum up: If you use the highest peak sound as a reference level and set it to such values as to make
it reach a maximum peak level of -18/-12 dBFS on a digital scale you will not have any problems.
In mastering, gain staging refers to the processing chain as the signal is ‘processed’. However, there is
another scenario where, if the level is too high, some problems may arise, even if you worked in digital
and used a limiter to chop off everything above 0 dBFS: inter-sample peaks.
It may be the case that, while reconstructing a continuous waveform from the digital information, a
converter produces a signal with a higher peak than the value of the samples ‘squashed’ against the
limiter ceiling. The effect this has on the sound you will hear depends on multiple factors, including
the quality of the DAC and the way it is designed but, although sometimes inaudible, many other times
some distortion may be apparent. In any case, you’d better be ‘down-to-earth’!

Chapter 9
The interviews
Michael Angel
CDSoundMaster
What is the true role of a mastering engineer?
Michael Angel: This is a great question! The role of the Mastering Engineer has changed so many
times over the years, but the fundamentals remains the same; the Mastering Engineer is to be the last
human decision maker in the recording process who preserves the best elements of the performance
and its recording, while preventing the technical and sonic issues in translating these to a master and
the subsequent mass-produced copies. In earlier days, the Mastering Engineer had a crucial role in
making sure that the structure of frequencies, balance, volume, and dynamics were under control so
that the needle would not dance out of its groove on a vinyl album. Control of the bass frequencies
was vital, and typically mastering tools were designed to limit extreme changes to low frequencies and
to place these frequencies towards the center of early stereo vinyl recordings.
In the boom of the record industry from the 70’s to the early 80’s, there was a shift in thinking in the
entire structure of the recording industry. What used to be a tightly run ship was being seen more
and more for the vital artistic role of each engineer in the process. The Mastering Engineer was often
previously used for redundant, even menial tasks, transfers, edits, and preparations for masters, with
the hope of inching their way up the ladder to primary Mix Engineer, or Balance Engineer. The amaz-
ingly overlooked final stage was often left to a machine-room mentality. Some of the most talented
and gifted engineers were often sitting side by side in a room of folks with headphones wondering if

they were served a dupe of a dupe of a dupe of a multi-million dollar album. Incorrectly marked or
incorrectly cued masters were pressed with unintentional wide stereo pans that were absolute pro-
cess errors and later attributed to the artistry of the bands and engineers. Some of this changed in
the handing over of the industry from the 50’s-60’s organization of creatives-meet-scientists, into a
messy new aesthetic of “everyone is a producer” idealism.
Because the mix had to survive the process from tape tracks to tape masters to dupe masters, on and
on, it was crucial that the mix account for each stage in the process (often adding vast amounts of high
end gain to account for later high end loss at each generation dupe). The process has always been
about balancing one trick after the next. The engineers of the 50’s were geniuses at mastering speed,
technique, and stream-lined processes and the Mastering Engineer had to make it stick. The 50’s into
the 60’s saw the first truly stunning potential for capturing sound but the process still had a lot of loss
from one stage to the next. Engineers were trained in-house with very precise rules. In some cases,
there were documents about how far microphones had to be placed from instruments and consoles
were marked with maximum fader positions not to be crossed. The “balance” engineer was the top of
the food chain for a long time.
The more complex the mix could become, and the finer the perfection of late 70’s, early 80’s tape
machines had become, the more value was placed on the protection of final masters once again. As
albums, cassettes, and yes 8-tracks made their way to a new frontier, the Compact Disc, the Mastering
Engineer became the new top of the audio food chain in the careful transfer of countless treasures of
1/4” and 1/2” masters into digital. This required restoration in some cases, and revealed hidden weak-
nesses to recordings, splices that could be heard for the first time, and even revealed certain artifacts
in the high-end hardware that had been so well suited to the pleasant masking of such sounds before
digital clarity. Remasters became a critical new path to huge new sales for record labels, featuring the
hard work of Mastering Engineers, further giving a prominence to the important position that had al-
ways been there, but for different critical purposes. Although there have always been rare renegades in
the industry, the financials almost completely eliminated the outside world from the truly professional
sounding mixes and masters, and the Mastering Engineer was even able to present their own inde-
pendent facilities for the sole purpose of serving excellence in their one particular field. The first gen-
eration of truly independent Mastering Engineers had worked hands on with the most famous names
in history at the most prominent labels, and they earned their place at an hourly rate of $300-1500/
hour. As insane as this may sound to many, even a $15,000 mastering job on an album guaranteed
to sell 500,000 copies would come to $.03 per unit. When comparing what the distributor does for
roughly $2-4 per disc, I see quite a value in that 1/33 of a dollar per unit!
As the digital world gave way to the first ‘affordable’ digital audio recording systems on personal com-
puters, some as inexpensive as $4000-$8000 for 8 tracks of digital audio on a 486 with an $800
sequencing program that ran 8 tracks of digital audio (yes, thank your lucky stars you can now own
a $100,000 recording studio for the price of a crappy used $150 laptop), there was a new era of
‘home’ studios beyond the previous demo setups of 4-track cassettes and Fostex 1/4” reel to reels.
The Mastering Engineer was now, for the first time, having to start considering what to do about a new
side to the market- the home-made independent studio recordings. This included a need to apply a
heavy handed approach to sub-par, less professional mixes that required technical notes and redoing

certain elements of the mixes before supplying usable pre-masters. The CD required a glass master
created off of a Sony edited master, replicated in minimum quantities of 1000 units. It was a few years
later that affordable home studio CD Burners existed, with a 2X CD-R running $500 minimum, and
individual blank CD’s costing $12 each. Technically, this opened doors to the first ‘Mastering Engineers’
that had no technical requirements, had not climbed the ladder at a major facility, and whose only
opt-in was that they owned a CD-Rom and could burn a red-book master with their own computer.
I was blessed to arrive just in time to have learned the true traditional craft of Mastering before this
became the norm. I had one foot in the door of the classic way of doing things while watching the price
drop radically to the rest of the music world, which was in itself a blessing to my bands that I recorded,
my budding project studios and growing Record Label and BMI Publishing Company.
The first computer software plug-ins to come along that featured do-it-yourself mastering settings
were an industry joke for several years. Regardless of whether some of them were actually pretty de-
cent in their audio quality, the premise that someone could come to understand the proper way to
use them without having learned the traditional path to an intricate and complex technical career was
ridiculous.
Early online forums were born in the mid 90’s where discussions took place over the new trends in
home-remedy mastering. It was understood in the industry that risking one’s own mastering, espe-
cially without any decent sounding home recording setup remedies as of yet, was considered as risky
as do-it-yourself brain surgery.
As sound cards improved and 16-bit audio was actually a step above 12-bit mini-disc recordings, some
home setups were actually starting to sound competitive with commercial facilities. You may think this
sounds hard to believe, but I actually owned the first pro-sumer 20 Bit audio card before there was
such a thing as a computer that could smoothly operate 24 bit playback live from its pci buss.
Somewhere between Napster and torrents, affordable Pentium computers, and excellent articles
about how to use SM57’s and the world’s first affordable semi-pro mixers led to a new revolution
in home recording. The small commercial facility was finally born, and skilled Mastering Engineers
breathed a sigh of relief that more aspiring home producers were coming into their element as a new
breed of skilled-and-learning Mix Engineers who trusted the expertise of Mastering Engineers to pre-
pare the important translation into final glass masters.
Ah, but panic was about to spring forth at the explosion of the plug-in software industry. It seems that
there was a large profit in explaining to the average beginner recording hobbyist that they had to have
their own mastering tools. For those who were eager to learn the trade, it was a wonderful time for
exchange, learning, tutorials and the birth of the modern recording forums, but it was equally a time
of novice panic and professional audio nightmares, as do-it-yourself audio surgery spawned from a
commercial marketing plan to sell professional audio tools to an eager crowd that was not yet ready to
understand how they worked. “What kind of stuff do I need to master my album with?” “What is Mid-
Side and do I have to have it before my band releases our album?” “Is it ok to use the Eq in Cubase
on the whole mix or do I have to have the T-Racks for it to go on CD?” “Where can I read a red book
and what is an ISRC?” All of a sudden large software developers had spurred more than an interest in

recording music on increasingly powerful computers. People were terribly concerned that they better
learn how to be a Mastering Engineer and quickly, or else!
From the mid 90’s to circa 2005, Mastering had peaked as an exclusive niche that was left to the most
skilled audio engineers who had earned their way to the top. The Mastering Engineer had to also
consider flexible real-world pricing models for the new serious recording musicians who realized that
there was a vital role to be played between the inspiration to record one’s own music and the serious
investment of releasing hard copies that needed to be professionally prepared for replication. Mas-
tering came to include online service, file transfers, and online communication in place of expensive
in-person sessions.
The real transformation to today’s model didn’t fully take effect until the success of iTunes guaranteed
that the new era of recording was yet another drop in quality; the mp3. Now, the Mastering Engineer,
having ‘mastered’ the changes in direction from lacquer to tape reels, then to CD’s and improving
digital, to a now inferior compressed data that had been so improperly researched before taking hold
that literally the bottom dropped out in the quality of preservation of priceless masters. Thousands
of original platinum recordings were ‘ripped’ to 128 kbps mp3’s with little-to-no skill, and without the
enlisting of the Mastering Engineer. Once again, the short-sighted record industry had missed the
mark in assuming that “digital is digital” and people noticed all sorts of artifacts from rumbling to digital
overs, shifting in stereo field to weird warbling artifacts. Re-enters the Mastering Engineer once again
to figure out the minimum necessary quality to reproduce 16 bit, 44.1 kHz lossless PCM Wav files into
mp3 format. Which encoding sounded best? What made for a plausible rendering? Ah, the Mastering
Engineer’s job is never done, rarely understood, always poorly mimicked, and never the same for more
than a decade.
So, we have seen many choices to improve audio go by the wayside. The use of a dynamically superior
AUDIO DVD 24 Bit, 96 kHz format was never established (though many hobbyists record at 96 kHz!)
The first generation HD converters still sound better than the most expensive deeper bit converters of
today, with parts that are no longer possible to source. The average user is listening with earbuds that
cost $.50 to manufacture, or laptop speakers barely capable of ½ the listening spectrum compared
to the average home stereo that easily matched 80% of the master’s actual quality playback. Today’s
Mastering Engineer plays the role of trusting their ear and their nuance in what makes the right master
for the job, but also in educating the potential client to raise the bar on hobbyist-goes-instant online
distribution, while serving files digitally and possibly in multiple formats. It is a balance of excellence in
sound in an industry that is now flooded with amateurs armed with sophisticated, often pirated soft-
ware that easily does the job of the finest analog gear (not all plug-ins are equal in quality, but they
certainly bring more benefits to sound quality than the areas where they are lacking). The professional
Mastering Engineer plays a lonely role in maintaining their value at a fraction of the hourly they need
to earn while competing with those who have a fancy website offering $10/song. I imagine that the
next step for the Mastering Engineer can live another several years, perhaps even a decade, offering
professional advice to those learning to mix and record, and standing out in a pool of audio software
developers selling their own designs that truly make a difference.
What are, in your opinion, the myths/clichés surrounding mastering?

Michael Angel: There are so many wonderful myths, legends, and clichés to mastering. I am so happy
to be given the question! One of my absolute favorite myths has just recently come up again just the
other day. Yes, the Mastering Engineer must be able to identify potential issues in recordings that oth-
ers may not notice, requiring stellar monitoring and sharp, attentive concentration and hearing. But,
the tools of the Mastering Engineer have long been given the same status as the engineer themselves.
True, these are amazing sounding pieces of hardware built with artistry and skill, but the myth of their
absolute precision and perfect redundant rendering is an absolute myth. Well-maintained, high-end
analog gear has abnormalities that have been sonically accepted, but largely unknown by the very
industry that relies too heavily upon them. The finest Eq’s in the world that provide the cleanest sig-
nals for the most surgical of adjustments have massive areas of discord between the marked settings
and the reality of the actual audio path. There are levels that vary by a large amount from the marked
+/- gain, frequencies that truly need to be visually calibrated by an oscilloscope to match the actual
markings.
This leads to another myth of the same topic; detented settings are best for mastering because they
are repeatable. I sit at a different opinion from most other Mastering Engineers because I have learned
what I have from my own work with testing and analysis of the signal. There aren’t many people that
can give as accurate an account of what the actual waveforms of the most sought-after gear actually
look like. I have seen it all, at every level, through every test and using high-end cables and low-end
cables. I have been amazed at the precision of what many would consider mid-grade devices because
they aren’t for the wealthy-only, and I have been blown away at the amount of adjustment required
from some of the most expensive devices that exist. The truth is that the Mastering Engineer is the
most valuable asset to the use of such gear - knowing how to tweak levels until they sound correct,
and for those that truly love preciseness, there are analytic tools on more modern, less classic master-
ing-era devices that let them know for certain that a dB cut with the most narrow slope “Q” is actually
doing exactly that. For all of the other wonderful gear, I am thankful to have a means by which I have
collected a truly priceless catalog of recordings of every setting of every device to be perfectly repli-
cated in digital, never to need the wondering of whether it will behave the same the next time - finally
it is a museum captured in its perfection inside the DAW!
Another myth in mastering is that you can always fix it in the mastering stage. I learned this early in my
friendship with my mastering mentor. It has long been a rumor that the musicians can rely on the Mix-
ing Engineer to make up for their mistakes (yes it is done all the time, but it never accounts for doing
a better job in the actual performance). It is further incorrect that the Mix Engineer can ‘almost’ get it
and that is fine because the Mastering Engineer will fix it. The final stage is the result of those which
come before, and the best master in the world is going to be a representation of just how masterfully
the performance was captured to begin with.
I suppose a cliché to mastering is certainly the loudness wars, so I will save that for that particular
question! But, another cliché to mastering that I have seen come up often times, is “did they actually
do anything?” or “I can’t tell what they did.” When I take on a mastering job, I make notes for the client
in case they want to analyze what has been done step by step. Especially if they have done an amaz-
ing job with the mix already, the Mastering Engineer’s best role is to know when to leave things alone.

But often it has been cliché that expectations are so high, that the master will sound so significantly
different from the mix that the client wonders if something was done at all. My personal philosophy,
though influenced from some different thinking on the topic, has come to a personal sense of artistry;
I care about the consistency of the entire album, but I like to make each track stand out as something
unique in itself and not just consistent with other tracks. I also like to leave a little signature “more” or
“better” than where it began. If it is absolutely perfect from the mix already, I will have run it through
many paces, including loudness options, before choosing to leave it completely alone.
What genre(s) do you mainly focus on? How does that affect your mastering decisions?
Michael Angel: Another great question! I come from a classical training background. My parents were
both professional opera singers. My mother was a touring solo opera singer and the Owner&Founder
of Capitol City Opera Company. I was trained in voice and piano at a very young age and took to falling
in love with drums shortly after. After years of study and playing in concert band, performing all over
the world with the Atlanta Boy Choir and also with the Alumnist Men’s Chorus, receiving a Grammy for
our performance and recording of Britain’s War Requiem, singing and playing percussion under Maes-
tro Fletcher Wolfe and Conductor Robert Shaw, I played in rock bands, ran my first project studios,
recorded blues bands on location, and was a percussion major in college until I eventually got my BA
in English Literature with a triple minor in Art, Music, and Psychology.
My background shaped my love for music and gave me an appreciation for excellence in many differ-
ent genres. Performing in the world’s finest concert halls gave me the fascination of the live experience
but also the impact of sound and ambience on music content. Working with monitors and head-
phones fighting for the perfect analog mix (all tracks by hand no automation - imagine that outside
Cubase!) strengthened my love for the impact that recording has on the song and the artist. I loved
the sound and writing of groups like Queen, who were amazing musicians but also made the studio
an active member of the band. I loved the craft and recording of guitarists, punchy studio drums, and
the rare absolutely skilled mix, but none of it ever wins out over my absolute love for the perfect choral
blending of Maestro Fletcher Wolfe. He is the world’s finest Mix and Mastering Engineer, and he has
no idea what a console or what difference a ribbon speaker would make. What he has done histori-
cally and even is still doing to this day, with the sound of blended voices in choral music, is the most
remarkable musical expression that any human will ever accomplish. I place this at the top including
a love for Rachmaninoff, Beethoven, Mozart, wonderful piano works in general, and the sound of a
brilliant mind controlling a wonderfully trained precision of hand movements and expression.
My love for drums and percussion makes it easy for me to love working with virtually every “pop” gen-
re, and as a multi-instrumentalist, I have just enough skill to understand what the members of a band
hope to see preserved in the final recording.
The most challenging part in this decision-making process is not the genre as much as the content.
I spend 12-15 hours working every single day in a tough economy, so I rarely take on new mastering
projects right now, but even when I was seeking to master full time, I made the decision that I wanted
to be true to myself and to my faith. I am a follower of Christ, and this is the most important element
in anything that I do. We live in a time where this is very unpopular, especially within the confines of

an art form that is based on science. I am certain and steadfast in my belief in one true God, and for
me this means that when I master someone’s music, I want them to feel proud of the seal of my work
added to their hard work. If there is a strong ideological or content clash with whom I choose to be in
my walk with Christ, I would do a disservice to the client and I would feel less of a man in dishonoring
such an awesome God. Beyond that, I can get passionate about anything that sounds good to me in
any musical style. I just love sound!
What is your opinion about the Loudness War? Does “loudness” have an impact on actual sales or
not?
Michael Angel: Yes.
Don’t worry - it would be completely out of character for me to give a one word answer - lol.
Sadly, loudness does impact sales, and it seems that the revelation of improper loudness and how
it harms the content came about from a guitar hero game! As I understand it, there was a release of
a song that was before the final commercial master. Fans were able to compare the decent sonics of
the mix before it was absolutely crushed for mass consumption and they were shocked at the horrible
things that happened to the sound. The truth of the matter is that it does not have to be a quality
trade-off.
There are tricks to the trade and enough excellent mastering tools that a skilled engineer can supply
the Mastering Engineer with a mix that is well-prepared for big, impactful sound without ruining all
that is good and holy about a wide dynamic range.
A short story about dynamics. The movie industry has gone through this process, and because the
production experts in movies are always more adventurous than us purely audio guys, they tend to
reach conclusions with their target audience quicker and with more savvy than the audio industry.
Let’s just be humble enough to admit it.
In the purely analog days, the soundtrack of a movie had the same limitations as our audio mixes.
They were going to lose something at each dupe generation, so levels needed to be a certain percent-
age above the noise floor at each stage, and transfers needed to be made with as few retouches and
retakes as possible, as there was loss to high frequencies and more artifacts the more times tape was
run through playback. Hitting the master too hard and with too little dynamic impact was exhausting
to the listener, but too wide a dynamic range led to some dreadful sounding, hissy parts to some great
movies. Too much of the wrong kind of loud was painful in the theater setting and silent dialogue was
an element killer.
Studies were done and the average and peak listening levels were established by the audience them-
selves. This led to a set of standards for the actual dynamic range of film soundtrack still used today,
or should I say over-used today. The variance in the quality of location and dialogue recording has
just as much to do as volume. Weak recordings at pre-set, pre-configured dynamic settings make it
almost impossible for the home playback of movies to have any logic. We can barely hear the subtle

scenes and turn it up to a point where we hear the lesser quality of our consumer amplifiers. Then,
the huge car wreck or monster appears and we have to jump to turn things back down as the cones
in our speakers beg for mercy. My personal opinion - establishing this dynamic range had a nice logic
but only the most skilled director, producer, and editor gets it right, and it is ok to have some dynamics
without the overkill.
The same is, I believe, true for music. Two things in the past decade have absolutely killed people’s
natural instinct to mix a good song: the loudness wars and -18 dBfs. How many people actually have
any real understanding of either of these? An actual real-world range of 12-14 decibels RMS between
low levels and maximum peak works very well even for classical music, believe it or not, but nothing
can stand in the gap for skilled listening in an accurate monitoring environment. Listen to comparable
recordings that sound good, and listen to them very quietly and very loud. Listen on the worst case
scenario but take it with a grain of salt and compare at the ideal volume on the best speakers. Loud-
ness can be our friend with it is a balanced, excellent sounding record that has its dynamic moments
as is appropriate to the genre. Overboard is horrible and can permanently ruin the mix beyond repair,
but truly there is a narrow range between too dynamic to have maximum impact, and overly treated to
where the life is squeezed out. If good dynamic decisions are made at the mix level, and skill is used to
make things sound balanced and strong, fluid and musical, then it is not very difficult to make musical
decisions that also make things sound a little bigger, fuller, deeper, broader, and more believable, and
yes, commercially viable. More than anything, sales are driven by consumer confidence, marketing
budget, social presence, a good video production, consistent live performance, etc. But, imagine if
someone is the next aspiring pop star and you put their new single in the playlist up against 3 other
currently charting singles, and the stress on natural dynamics won out over the perceived presence
and loudness of the song in comparison. That artist is likely to be instantly judged by most untrained
listeners as being ‘less’ professional and therefore the perception is that it is ‘less’ of a recording. So,
until the commercial side of the industry caves (probably too far in the other direction like cinema) to
a dynamic artistry, loudness is going to be one of the elements of success. If done wrong, it is irrepa-
rable, but if handled with skill, it can enhance everything that has already been done right.
Let’s speak about compression: are there any tricks you want to share?
Michael Angel: Sure! I love sharing thoughts on compression and EQ. When I get serious about making
a full instruction series I hope to make it possible for everyone to try their hand at things they may not
have considered.
I am going to save some more complex compression techniques for more advanced tutorials, but in
general, there are certain secrets that work beautifully in the mix, therefore making it much easier to
attain success at the mastering stage.
First, break your compression thinking into categories. Compressors were created for resolving the
signal to noise ratio of tape machines, so the first place to consider the compressor or limiter is in the
actual tracking process. Lump this in with the list of things people want to learn, and trust me - one
of the reasons that your plug-ins are not getting you the sound of “that” 80’s or 70’s group is partly
due to the compressor choice going TO tape - in this case going to digital capture in a DAW. If you

are just focusing on great sound and you want to make all dynamic decisions after recording, then
keep in mind the performance can still suffer thin source material if too low signal gain is chosen in
most average-to-nice microphone preamps. Too much signal can sound good on its own but can get
messy when tracks layer up. There may be situations where you want to track to the DAW with good
compressors that give a comfortable average signal going in as opposed to an extreme dynamic range.
Next, compression on drums versus other instruments. I consider compression part of the creative
process - it can be transparent to let the instrument come through or it can be a very strong musical
statement. Make sure you know what the purpose of the compressor is. If you have an aggressive song
that needs heavy-hitting drums, I personally like to set up compression on each individual track, on
the buss, and on the master. I have a system for just about every scenario, so I don’t always need to
wait until a full mix is up to make certain choices. I use a relatively slow attack and release on kick and
snare, and often on all of the toms. Each drum and song and performance is unique, but somewhere
between 15MS and 50MS gives a wonderful crack of the snare and kick. The trick to the sound, though,
is the release time. I do not treat this in its own respect, but in its actual length between the attack and
the tempo of the song. So, there is a sweet spot in the right measure of “crack” and “pump” or “breath-
ing” for a song that will come across as aggressive and musical depending on the pace and tempo
of the song. If this changes I will break up, for instance verse and chorus, and treat them differently.
In general, something like 20MS attack 90-150MS release makes for some nice slow drum crack, but
release may need to react in a shorter time for a faster song.
I use less ‘effect’ compression on the buss and go for growing the signal naturally.
I like to follow the compression on the buss with a limiter, which will react very fast.
I normally will use some of my natural tape compression and saturation in the chain to make each
stage sound bigger and more natural.
What about EQ?
Michael Angel: For EQ, this is purely a decision about favorite devices for favorite scenarios. In master-
ing, I will give my most general tips and tricks, because any user can apply these to their own technique
and come out ahead. In general, for mastering specifically, I like to start my session just listening and
making notes. When I first operate on the track, I will go through an extensive signal sweeping process
with a terrible sounding narrow band Eq. I do this for the purpose of exposing the worst mix decisions
whether they are obvious or not. Sometimes I may already hear the issue before needing this, but
commonly I will make a very narrow “Q”, turn up the gain, and start at 20 Hz. As I raise up the spectrum,
I listen where the huge bumps appear. I use a master limiter after the Eq to prevent overs and to keep
from getting angry. It almost never fails that somewhere I will expose some rumble, a sustain in a note
that starts to sound dissonant, a harsh chord or mic option, something edgy in a vocal, and a high end
that could work better for the air in the voice or cymbals on the drums. Then, I decide on whether to
surgically remove these elements first, or whether I know some overlap boost/cut is going to correct
multiple elements at a later stage.
This will come up in a later question, but I created the world’s only corrective filter of its kind in my
Mastering Suite for Nebula Pro, and it is what I use for narrow surgical corrective Eq. Once I listen with

terrible digital Eq for where the problems occur, I use my special formula filters to remove these arti-
facts and you can almost never hear that an Eq was used, but the problem goes away.
Another general trick is the use of wide “Q”. Treating the low, mid, high as 3 sections of the song can
get you a long distance in making nicely balanced, sweetened final masters. If a recording is great
as it is, usually a very small change in one range or the other can make the difference from great to
outstanding. I love classic, vintage inductor-based Eq’s with vacuum tube make-up gain for wide “Q”
and specifically there is no prettier sound that I know of in recording than an NSEQ-2 in Vacuum Tube
Mode at its regular “Q” width for mixing and for mastering.
Good monitors and a pretty neutral room are essential for making conscious decisions while master-
ing, but what about headphones?
Michael Angel: Headphones are a mixed bag. You will hear a lot of people state that it is impossible
to mix or master inside headphones because it lacks the critical element of your ears making a natural
blend of stereo content within a room environment. There is a bit of truth to this, and I would never
recommend someone to isolate an entire album process with a single set of headphones, just like I
pray that the time comes, sometime in the history of studio recording, that facilities will abandon the
“NS” series of near field monitors. I think the end user can attest to the fact that an incredible listening
experience can be had by the owner of an amazing vacuum tube headphone amp or an excellent
solid state high-end headphone amp, and a top notch pair of headphones. This is not to be confused
with an mp3 player and earbuds, which are likely to win you a case of tinnitus in your later years. I per-
sonally have, for necessity in live scenarios, and other times when a monitor had to be replaced, done
some mixes that I am very pleased with, just using headphones. I would attribute this to consistency in
process and the fact that I spend so much time listening critically to sound in general. If I spent weeks
at a time away from musical sound and critical listening, I don’t think I could trust the total isolation of
headphones as much. The other element is that I believe you can actually appreciate precise stereo
placement and distance staging in headphones better than most nearfields and definitely better than
most mains, but you do miss the important way that music develops by allowing the natural wave-
forms to fulfill their maturity. That is to say, 6-15 feet away in a flat, non-reflective room at a triangulat-
ed angle is amazing for every element of mixing. If you want absolute idealism, get a pair of amazing,
well-cared for used ribbon speakers and a pair of $300 subwoofers barely turned up enough to hear
the extra octave. All that you need from these is at least excellent timing down to 40 Hz or so and you
can tell the rest by checking against the mains in headphones (rumble, bad low notes that may need
high pass). I say this because you truly can harm your speakers AND your ears trying to go full volume
listening for the bad stuff that happens in the longest waveforms. These don’t hurt your ears when you
are listening, but they cause the most hearing loss of any other waveform.
One other thought; I find that good studio mixing headphones like the K-240 are excellent for identi-
fying, isolating, and removing clicks, pops, and mouth noises in vocal tracks.
What are your favourite Nebula/Acqua Libraries/Plugins? What other plugins do you think are suita-
ble for mastering?

Michael Angel: I don’t want to come across as if I am using this as an opportunity to only push my own
product, but I admit I consider my own work in every product that I have developed, so I do tend to
use my libraries and plug-ins for my own mixing and mastering work.
I believe that AlexB’s API stereo mastering EQ is an incredible tool - truly for mixing and for mastering.
I am certain that AITB, Tim, Henry, and the others have done an excellent job with EQ’s that I have not
heard, so let me just state that I have had the honor to help in the earliest of stages with the first com-
mercial releases, testing, listening, advising. This has been many years. The most important thing to me
is that when something represents a specific famous piece of hardware, it truly sounds like it. This is
an area where Nebula Pro and Acqua absolutely win hands down over any competitive process. Even
other companies that have used a convolution process have not compared to the quality I have been
able to get in my own work with similar products and I have heard the same thing in testing libraries
from other developers.
I created the Mastering Suite for Nebula Pro a long time ago, but it remains alone in its category, for
me personally, as the world’s finest corrective Eq for any recording that will end up at 44.1 kHz, and I
created a 96K filter that allows its use for higher frequencies as well. I will adapt the narrow bands and
the harmonic enhancer to an Acqua in the near future.
The SongTec, GEQ, and Nice EQ are 3 of the world’s best options for mastering Eq, and since they are
inside Nebula Pro you can repeat their settings over and over without the analog concern for the little
inaccuracies of redundancy that do occur with their analog use. I recommend one or both of the R2R
and Apex for the Studer A-800 at 15 IPS for mixing and 30 IPS for mastering, the Stu A-820 for the
same, and the ATR-102 at 30 IPS for mastering. I recommend Tape Booster Plus for making the signal
of the recording grow before applying too much compression and limiting. It uses natural harmonics
to grow the signal without changing the spectrum or transients.
I recommend contacting the other developers and asking them about the Eq’s that they have worked
with and have created for Nebula Pro, as many of them will have some great options for use in mas-
tering as well! I can go on, but this is a great starting point, and for a few hundred, including the entry
price of the Nebula Pro Plug-in, you can have a literal and honest representation of the real hardware
in the tens of thousands, a promise that many other plug-in companies promise, but very few deliver.
Speaking of other companies, I do like to recommend things that I have been impressed by, so let me
make some honest recommendations that I absolutely do not benefit anything from.
First, please do not just go and buy plug-ins that are supposed to represent hardware JUST because
they look like the same graphics. Large companies have the advertising budget that I WISH I had. They
pay a celebrity handsomely to support a release and give name recognition and it does sell thousands
of downloads, but how many of them are truthful representations of those devices? Well, some of
them are ok.
Personal opinion: I cannot recommend any single plug-in purchase as strongly or sincerely as Nebula
Pro. I hope that the future of Acustica Audio proves consistent as the past decade, as I love the deci-

sions they have made thus far and believe in Giancarlo’s work. I believe its strongest asset is for Eq’s
and complex high-end signal chains like tubes and preamps, but I believe the future will make them
competitive for other areas as well. I have found a great deal of success with the compressors I have
chosen to focus on in my own work, and although I believe the technology has some room to mature
into more complex compression options that might be easier served with algo-based options, you
cannot beat the cost-per-excellence for NebPro based Eq’s.
Waves has made a gazillion programs. I think some of them are very useful and others look pretty and
cost too much. Personally, I think they nailed the sound of the API compressor while only captured
the general usefulness of the Eq’s. They are good, but not as good as Nebula Pro Developers provide
for API Eq’s.
Waves made a nice iteration of the SSL Buss Compressor, but there are some better sounding com-
pressors and the analog still sounds the best of all of them. I am not as impressed by their SSL EQ, and
I once again believe there are good options here in the NebPro world that blow others away.
I have long been very impressed by Voxengo’s Polysquasher. As a mastering grade compressor I
have compared it to analog and software versions of a lot of devices and it is simply an excellent,
well-rounded tool. It is not out to replace a specific device, but is just a great sounding tool that does
not sound like algorithms to me. The same can be said for his Elephant as an option for a mastering
limiter. It is simply a great sounding tool and gives the timing range necessary for a world of loudness
requirements but even with very dynamic uses it is a great tool.
The process that I have developed with my coding partner Michael Heiler is something that took the
same years of research that went into mastering (pun intended) the complex sound of tape machines.
Our process is based upon making complex analog processes come to life without randomization and
without linear algo processes and also without the limitations of schematic-based design. The result is
a massive amount of processing with relatively little cpu impact. Things like tape, compression, tubes,
can be created that complement the natural limitations of Nebula and Acqua, and of course I use my
own designs for the exact purposes I developed them, with Michael’s (the OTHER Michael) amazing
skills.
Har-Bal is an absolutely amazing tool. It cannot replace, in my opinion, the need for a Mastering
Engineer, in the same way that a Ferrari replaces the benefit of a great driver, but - man - they have
done something incredible with this tool and I love its use for diagnostics and I have run into situations
where it was beneficial for fixing complex curves.
I don’t mind recommending UAD to anyone looking for the best tools for pretty much any job. They
have been among the top developers for a long, long time, and I believe they have applied their own
research and have competed even with myself in a smart manner. Again, I wish I had their market pres-
ence and ad buys, but even if I became a millionaire I would still be a fan.
Sadly, my favorite recommendation is something that is no longer valued in the marketplace, but the
LiquidMix was a great tool, and for the money, it was a ridiculous value. I personally found that using a

dsp-powered dynamic convolver still could not match the quality I got with my own work with Nebula
Pro, so that speaks volumes to what Giancarlo has done correctly over the years. But, I did find that if
you have a decent software emulation of any of the LiquidMix compressors, you can match settings
to find the areas where it is the most similar to the original hardware, while other ranges are not as
useful. The LA2A and Vari-Mu inside the LiquidMix are worth the price of finding one used. This tech-
nology was created by Sintefex and license sold to Focusrite, who have a long history of investing in
great individuals and products and then making stupid decisions with them - lol. They took the only
technology, in my opinion, that could come near Nebula’s quality and choked the life out of it. The
cheaper it got, the more they integrated it into a nice, viable option for recording chain and interface
options - perhaps a good market choice for them, but sadly this meant no 64 bit compatible drivers
for LiquidMix - how stupid! - lol.
The Neve 33609, 1176, and LA2A in the UAD(2) are excellent. I personally find there are very few items
left in my massive arsenal that I actually turn to these days outside of NebPro, but when I am working
fast and have very little toe shaping to do, I find that tools like the Waves API and Voxengo’s offerings
are excellent. If I had a lot of money I would support Voxengo and buy more of their plug-ins because
they are unique and sound amazing, but I just do not find that I personally have a need for many things
outside of NebPro. In my opinion, everyone serious about recording needs one great microphone (I
didn’t say expensive, I said great), one great preamp, one transparent set of converters, good drivers,
Reaper and maybe one other DAW, at least 4 Gig Ram, a PC (use Mac for anything besides recording,
please!) a massive hard drive, or two, or five (I currently actively work off of 12 hard drives), one good
pair of monitors, one good set of headphones, talent, NebPro, and a small investment in the right
libraries.
Do you use any analog hardware?
Michael Angel: Yes! I love analog hardware, and although I have learned to deal with the frustration
that goes along with it, I still enjoy moving knobs and tweaking smooth selections and settings. I am
relieved at the impact of digitally recreating my most prized devices has allowed for - reliability, redun-
dancy - not wearing out my more valuable analog assets, and more!
My favorite piece of analog hardware currently is the Source Plus, which sadly is no longer being cre-
ated. I designed it with a very talented hi-fi expert engineer with the purpose of being able to impose
the precise sound of specific vacuum tubes into the buss and mastering chains. The result was a du-
al-tube stage design that allows me to tune in only the amount of each tube chamber that affects the
sound the way I want. I can change the bias to affect the warmth of the sound, adjusting the voltage
of the tube plates, and control the output gain, leaving a wealth of analog options. The purity of the
signal at its nominal levels can be heard in the Vintage Tube Collection for NebPro and VST.
I have used probably 100 different hardware devices, including everything that you have in NebPro
now. I like the flexibility of 500 series items. I have a custom stereo pair of Neve channels that I love for
the original transformers. I have the incredible and vastly under-valued Aphex Studio Dominator - if
you want a crazy deal on a great analog brickwall limiter - get one!

I have API-based mic preamps, a Chameleon Neve-inspired mic pre/eq that is still ridiculously good
compared to most cheapo-Neve designs. The 1084 is still my favorite analog Neve channel/console
and I do not have the budget for a full Neve console or even a full sidecar, the 1084 is now my favorite
inside the DAW for the Neve sound, where my outboard pair of Melbourn Neve pre’s are wonderful
just for shaping and tone.
I have one of the very few $700 MXL V77S classic tube mics (much, much nicer than their other tube
mics, but their other solid state mics are fantastic). I have all of the expected dynamic mics, a lot of
other preamp options, one of the “good” RAMSA consoles that is an absolute steal if you find one.
Obviously, I am a true-blue fan of the Vox AD60VTX. I have a true vacuum tube hybrid Marshall that
sounds gorgeous (none of that digital effect section Marshall stuff that was truly awful). I have a Brian
May guitar and Vox amp that I absolutely love, but you are probably still asking about mastering! :-)
There just isn’t much need for all of the outboard analog mastering equipment I have always used now
that it is inside Nebula Pro, and even mastering compression and limiting are sounding equal inside
the DAW. I enjoy analog for the recording process more than anything, and I still sometimes go in/
out of busses for analog limiting - I mostly like slow, glue-like optical analog compressors and vactrol
compression.
I have bought and sold various versions of my acoustic drum set which is still in flux, but I also have an
electronic pads set hooked to a drum brain, connected to the master control room for MIDI, a wonder-
ful Daking A-Range, an original Trident Series 80 channel + Eq, an original Quad Eight 444x, A Bogen
all-tube rectified mixer with EQ, the best sounding Siemens W295B in the world, and of a room and
of a closet full of tape machines and the tape stock that was used to create the now famous R2R and
Apex collections.
I will be bringing a new line of super high-end analog hardware to our community sometime soon. This
will include line coloration items that every setting is unique and customized - where its analog stands
out the most, and it also includes compression in a classic sense with new features that you would
love to have! These will be wonderful for mixing and mastering, but they simply aren’t finished yet!
How do you normally set up your mastering chain?
Michael Angel: Well, I have more of a process than a chain. I listen to comparable recordings to adjust
my listening and thinking towards the project. I load my terrible narrow boost Eq and limiter to iden-
tify surgical Eq choices that need to be reduced, and then it is a custom process for each song. I load
different Eq’s for decisions I make one at a time. When I feel like the mixes of a full album project are
coming together, I will load a chain that often includes TB+ and a mastering tape option and VTM-M2,
followed by a brickwall limiter. It depends on what other kinds of comparable correction tools are
needed. I fold things down to center mono to check for phase issues and I send things super wide
(increase ‘side’ and reduce ‘mid’) to see if they will benefit from any adjustment in that manner. Rarely
do I do any hard compression or heavy limiting, and if any of that is needed I wait until I am comparing
one track to the next. I serve the client with a list of notations so they get a feel for what was done and
it also lets them ask if they need to understand anything in the process.

Is it possible to master at home? What tips would you give to someone who is about to start learning
this discipline?
Michael Angel: There is a value to the concern from years gone by, comparing mastering to do-it-your-
self brain surgery. I really think that anyone who thinks of any of this process, from recording to mixing,
mastering to developing products like Eq libraries and the sort, as being an easy field to master (again,
that pun!), is likely to miss out on the wonderful complexities that exist. My recommendation for an-
yone that is interested is not to under-estimate the complexity of the art and the science. Expect the
unexpected and assume that there is always something in your blind spot that you did not anticipate.
Having said all of this, yes I absolutely believe you can become very skilled mastering at home. To learn
the discipline, do not try to adapt the things you read and the tutorials on YouTube without having a
minimal high quality listening environment. You simply will never get it and you will think you are either
way off course and no good at mastering, or even worse, you will think that you have it all together
when truly you have yet to experience that “aha!” moment that is required to find a true love for ex-
cellent in sound recordings. Until you have at least sat in a high-end hi-fi listening room and heard
your favorite music on the best speakers in a tuned room, you cannot grasp what it means to master a
song. But, the good news is that money is hardly your enemy these days. Seriously, 20 years ago, it was
impossible to consider mastering with a computer, a few software items, and a $500 pair of monitors.
You can almost get by with a good result these days. You truly cannot overdo the choice of monitoring,
and the best studio monitors are still not adequate in every way for the best mastering work, but to
learn the skill and improve your own recordings, a nice silent room with good padding and scientific
decisions in how to treat the room, online tutorials, a lot of good recordings to listen to and practice
with, and a lot of patience… you can learn a lot about mastering that just was not possible in the past.
The best answer of encouragement I can give you is also this; I love sound and recording and that is
what drives me to continue to be interested in music and mastering. I plan to make a lot of information
resources available, both for free and for a full online course. I hope that I can be a living example to
help you get to that goal, and I have no doubt that the tools available inside Nebula Pro are not just
“good enough” but rather are the very best, most ideal tools for the job. All that is missing is your will,
determination, patience, humble devotion to improvements, and a reasonable investment in the right
monitoring environment. With that said, most people are not cut out to be the perfect fit as Mastering
Engineer, but I am certain that if you have the right characteristics, there is no reason you cannot suc-
ceed in the right home environment.
I hope that these thoughts are a blessing and encouragement to some folks out there, and thank you
for the honor of being able to answer these questions.

Ali Zendaki
alizendaki
Ali Zendaki: In my opinion, the role of the mastering engineer is to be the gatekeeper with fresh ears;
to ensure quality, bring a sense of cohesion to the project, and remedy any -minor- mistakes that have
been made in the recording and mixing process.
Ali Zendaki: A myth I hear a lot is that “mastering is easy” and you can “do it at home with a copy of
Ozone 6”. This could not be further from the truth. Mastering takes many years of experience, lots of
trial and error, and an extremely trained and delicate ear. You are responsible for putting the last mark
on the audio before the world hears it, you decide what is a good and bad decision, and taking these
decisions lightly could easily result in a negative impact.
Ali Zendaki: I focus mainly on both EDM and Pop/Urban Pop. These definitely have their own charac-
teristics each. I research the currently charting hits in order to see the ‘mix trends’ and do my best to
conform to industry standards in that specific genre.

not?
Ali Zendaki: It does have a very small impact on sales in my opinion, since this generation has been
groomed to believe louder is indeed better. So when they hear something that maintains a good
amount of dynamics, it make come off as inferior, even if in actuality it is not.
Ali Zendaki: My main tip about compression is: there are no tricks/rules. For many years, I fell in the
trap by reading, researching, and looking for an easy way out. Each project, each song, is its own en-
tity, has its own feel, and requires from-scratch compression. If I did have an actual tip, it’s to be very
careful about compressing while mastering, very small amounts are usually what the doctor ordered,
no more than 1-2 dB of compression, it’s also just as important to make the compression ‘breath’ with
the song, so dynamics and pocket are preserved.
What about EQ?
Ali Zendaki: The same things goes for EQ. For mastering, use EQ for surgical tasks, making a variety of
tracks sound cohesive, or to add polish/finalization. It’s not our job to add our own personal touch,
what you receive already has the fingerprint and DNA that should be preserved after the mastering.
Ali Zendaki: I use headphones to master actually. A lot of people frown on this, but hear me out. Once
you get to a certain level of headphones, most of the common negative remarks regarding head-
phones disappear. I use Sennheiser HD800s, along with a SPL Phonitor (which emulates crossfeed),
and also a Violectric DA Converter, which ensures my digital bits are converted to analog signal with
minimal amounts of artifacts. This combination gives me an amazing sounding ‘room’, with an as-
tronomically high level of detail, and an astronomically small amount of distortion versus traditional
speakers, as ‘rooms’ tend to alter the stereo signal. Another device I use is called a SubPac, it’s a very
accurate bass generator, that goes on your chair. Rather than a subwoofer, which can be very inac-
curate if you don’t shell out a large amount of money for room treatment, the SubPac uses an actual
motor to emulate the bass frequencies, and let you ‘feel’ them, rather than hear, creating a more inti-
mate and accurate environment.
What are your favorite Nebula/Acqua Libraries/Plugins? What other plugins do you think are suitable
for mastering?
Ali Zendaki: I can’t pick a favorite, I love them all! Ok, I lied, my favorite EQ has to be the Amber, and my
favorite compressor is the Titanium 3 Band! Amazing! I use a whole host of other plugins, for mastering
I tend to lean towards plugins by Slate Digital, Eiosis, FabFilter, Massey, Waves, Universal Audio, Tokyo

Dawn Records, and many more, way too many to mention here!
Ali Zendaki: Not anymore. I used to use a summing box, but that soon imposed limitations, which I did
not like. I feel 100% digital, I get complete control of the colorization and distortion I want to impart
on the signal.
Ali Zendaki: To be honest, I just go with my gut; I don’t have a chain or methodology, I just do what I
feel the song needs. If I have to say anything, it’s to be surgical first, then impart processes to make the
signal ‘sound better’.
this discipline?
Ali Zendaki: Absolutely, refer to my spiel on headphones above as to how this is possible. My advice
to anyone mastering at home would be to: learn your room! A/B with monitors, headphones, etc. Find
the weaknesses of the room so you can be sure to compensate if your room is not treated sufficiently.

Bob Davodian
TranscendingMusic
Bob Davodian: For me, mastering is not just about processing audio but also about quality control. It’s
another perspective within a different environment, one assumed to be superior in acoustics and gain
structuring, where a mix can be assessed and the proper steps can be taken to get the best out of it.
Bob Davodian: I think that the biggest misconception surrounding mastering is that it can turn a subpar
mix into gold. You know, mastering isn’t alchemy! It can certainly make things better, but there is in my
opinion a certain overly fervent expectation of what mastering can do for a mix.
Bob Davodian: I mostly focus on metal, rock and pop genres. I have mixed orchestral pieces as well
and hope to continue to do so, as metal and orchestral music are my favorite styles of music. I can’t
fully say how it affects my decisions, I can only say that for whatever is presented in front of me, I try
and bring out the best out of it as much as possible, to the best of my ability. For one metal song, I
may do totally different things versus another metal song. If it varies within the same genre, it certainly

holds true that between genres it could vary even more.
not?
Bob Davodian: I used to have a blog up, but when I changed hosts a couple of years ago the blog
went offline. In this blog I addressed this aspect of loudness in music. My idea spinned off of a polemic
thread at Gearslutz. The thread was called “Vote on the Solutions for Loudness” and although I ap-
plaud the effort, there is no practical means by which one can thwart the loudness war while making
people understand why. I’ve said it time and time again but nobody is understanding the nature of
why the contention for loudness exists. Such a contention that compromises musical fidelity is by all
means rooted in something else other than what it is apparent. It’s not to be taken at face value. Let
me put it this way: any convention or behavior found in cultures is based on a deep psychological
force or impetus. Until that impetus is addressed itself, the convention will continue to exist so long as
it feeds the underlying will to empowerment, no matter the stupidity involved. If doing one thing ei-
ther empowers through belief of something assumed to be true or directly creates a brute force result
that invokes primitive or visceral excitement, no rule or practical solution will stop it. In other words,
devise ways to suppress the loudness wars and you are giving the masses more incentive to defy it.
The only thing that can even touch the onset of a solution for loudness is a profound and psycholog-
ical persuasion. The sense of empowerment used in aiding the detriment of implementing unyielding
loudness, which in short is a type of influence really, can also be used to mold the logic (or whatever
is left of it) into realizing or at least lauding the prospects of perhaps producing not such loud masters
but to show more concern for musicality. First and foremost to achieve this, the most influential and
prominent individuals in both the creative avenues of our industry and the most technically proficient
must set this bar. The individuals that are looked up to in which young minds would follow suite (be-
cause following those at the helm is a sense of empowerment) are the individuals that will create the
model in how we must approach music production. A sense of “Man, that’s just dumb making it sound
that loud and crushed” coming from the creative and technical leaders would create perhaps a scur-
vy of implicit shame for any person who would dare revert back to creating squashed, lifeless, harsh
sounding music. Incidentally, this is the same recursion record labels/A&R/big name producers had
created that started toppling a few pebbles turning it into a loudness avalanche. Why? Again, because
following those at the helm gives a sense of empowerment. Rules, methods of enforcing loudness
control, and the like will only create more defiance. These types of methods are associated with ad-
ministration and bureaucracy: precisely the antipodal world to creativity and art. Not a good way of
getting people on board. I think at this point in time, things implemented for controlling loudness, such
as what we find in iTunes, it may buy some time. As I mentioned above however, I don’t think that will
actually teach people why we shouldn’t aim for such loudness. To be frank, the whole debacle with
Metallica and their 2008 album “Death Magnetic”, now that was a surprisingly good wake-up call for
people and consumers of music. The fact that the Guitar Hero stems, which were not as crushed and
pummeled, were circulating around the web enabled for a direct comparison, in turn shaming the
louder, crushed version which had lost its musicality in a sense and sounded harsher overall. I think
that did the most good for people to understand why it is not such a great idea to crush music for the
sake of loudness.

Let’s speak about EQ/compression: are there any tricks you want to share?
Bob Davodian: I can honestly say I have no tricks per say. The methods I use for the actual compres-
sion and EQ’ing of audio are things you can read or find online. My routing and signal manipulation
may be somewhat unique, things I’ve worked on and built upon for years working at my craft but those
things I mainly do if I mix music. Overall for mastering, I just listen. I know it’s cliché and a boring answer,
but I honestly just use my ears and then tune things to how I want them to sound. In fact, that’s my
trick: I practice sweeping EQ’s and setting compressors in extreme ways to understand what NOT to
do in a real situation. I train just as an athlete would in the gym, except this is for my ear-brain.
Bob Davodian: Headphones, I think, are a good tool for “zooming in” on blemishes. For editing, they
can be great to ensure that one hasn’t left in any bad fades, blips or glitches that may otherwise go
unnoticed listening on monitors.
for mastering?
Bob Davodian: I really like the AlexB consoles, I use CLC and VBC most. I also really like Tim Pethrick’s
libraries, he keeps pushing the boundries. My favorite Acqua’s are probably Silk, Titanium, Green EQ,
Amber EQ, and the D361A. I do think all of these are great for mastering, but mostly the Titanium and
the two EQ’s are great for broad strokes and sweetening in the mastering stage.
Bob Davodian: I used to. Everything from multi-track cassette tapes to some reel to reel tape, not
much at all and consoles. But around 2009 or so I consciously decided to go all in the box, as they
say. I don’t see audio as “analog sound” vs “digital sound”. I see it as “good sound” vs “bad sound”.
There are certain effects you can induce in an audio signal and you can essentially reach that effect
through either means, analog or digital. Getting there and knowing how to get there may vary between
the two different means, but the result is the result. That’s all that matters. Nobody says, “Wow! That
song sounds so analog or digital!” They just hear and feel it sounding good or not.
Bob Davodian: How I set it up is normally as a template. With my tools ready to go. But I don’t neces-
sarily use everything there. They are in bypass until I feel I need to use them. As far as order of process-
ing, I don’t have a given order except for a fundamental process: technical EQ, sweetening EQ, and
peak/ISP limiting. In between those anything goes. What actually goes is what the song and situation
call for. That’s what dictates it.

this discipline?
Bob Davodian: I think you can get some results “at home”. But what does it mean to say “at home”?
It’s all about how you read into those words. If you don’t have the experience, knowledge, and time
behind the craft, like anything else it wouldn’t necessarily be possible to do it or at least not on a con-
sistent basis – which is the most vital thing. Also, to go ahead and simply say “I will master my own
mix”, is sort of defeating the purpose of mastering, right? Like above, I mentioned quality control and
having another perspective within a different environment, one assumed to be superior in acoustics
and gain structuring. Well, of course you can’t have those things if you decide to DIY the mastering
process. For anybody who wants to get into the discipline of mastering, I suggest to put time into it.
When I was learning, at first in the 90’s there was really no Internet like we have now. Even later on, at
the start of the millennium, there wasn’t as much information and options online as there are now, so
I went to school for it. Now, forget it. There are simply no excuses. Between the knowledge online, the
option for schooling, and the numerous hands-on workshop opportunities by seasoned veterans out
there, I’d say you would be hard-pressed not to be able to get the information you need. But any way,
discipline yourself to absorb and learn all of this information, then practice and practice and practice.
Listen to your favorite commercial tracks and at first maybe try to emulate what you hear. Make mis-
takes! Just keep doing it. That is truly the only way to get good at it.

Mark Drezzler
Drezz
Drezz: To present the music in the best possible light and make it sound good on all different playback
systems.
Drezz: Clichés? Myths? … That mastering is some kind of ‘dark art’ and something mysterious. It’s not,
the processes are well established and there are many great tools available to achieve that aim. Of
course, experience and a track record is important and you should choose a mastering engineer who
is compatible with the musical genre or style in question and whose mastering you like. Listening to
the work of various engineers is key to finding the right one for you, also looking out for the mastering
engineers of albums you like or people who are enthusiastic about your material.
What genre(s) do you mainly focus on?
Drezz: Electronica, Psy-Trance, Indie, Reggae, Dub, Ska, Funk, Jazz, Folk.

not?
Drezz: Tricky question! It’s a question of perception and completely subjective. For some genres loud-
ness is unavoidable, dance music for instance is very much a loudness/competitive genre. Artists will
generally want it as loud as the rest of the genre, for example, if your tunes are going out on beatport
or similar and will be in playlists with many artists, then the artist in question will not want their tune
to be the quiet track in the playlist, so loudness is a consideration, but of course it’s a skillful engineer
who will be able to make it sound loud whilst maintaining as much dynamic as possible. For other
genres it can be a little different and perhaps the mastering engineer has a little more flexibility in re-
gards to how the material is to be mastered. For real instruments and more natural sounding music like
jazz or acoustic genres, there is a little more scope for retaining dynamics and being a bit more tasteful.
If the music is particularly dynamic to start with, featuring intricate and quiet passages alternatively
with louder crescendos, then perhaps holding off on the heavy compression or limiting may be prefer-
able. Mastering for vinyl of course will require another set of rules to be applied, for instance retaining
a good dynamic and sorting out the bass and making everything below a certain frequency Mono, so
as to ensure that the master cut plays correctly and doesn’t jump or distort or display unwanted ar-
tifacts of any kind. I am not a fan of balls-to-the-wall slammed and full on maximized loudness!!! It’s
much better and musical to have the dynamic in the music. It’s a little sad that we are in this position
now, and I must admit I prefer a lot of older records which have the dynamic range and sound sweeter
for it, but hopefully we will end the war at some point and we can just concentrate on it sounding as
good and dynamic as possible. It’s a very thorny and tricky issue…
Drezz: It’s good to know intimately the different types on offer and their strengths and weaknesses,
for example the difference between opto, vari mu, vca, fet and so on. Working with a limited arsenal is
a good idea, so just using a few specific tools will help one become more familiar with the particular
nuances of those particular units and ‘learn’ their secrets. Certain things respond to certain tasks, for
instance the vari mu design has always been traditionally employed very successfully as a ‘glue’ on
entire mixes, but there really are no hard and fast rules here. If it sounds good it is good, but arming
oneself with good knowledge and practice, practice, practice is a good habit to get into. Compression
can seem daunting to the beginner, but these devices are awesome tools for shaping the sound and
it’s not just about dynamics either as certain units are just as relevant for their unique specific tone and
you may want to use a unit not so much for its dynamic effect but perhaps just for its tone.
What about EQ?
Drezz: Again, so many EQs to choose from, but finding a handful of quality units is a good thing. Dif-
ferent beasts for different tasks. Clean. Coloured. Crisp. Grainy… It’s good to have a palette of choices
here and again it depends on the material one is mastering, some things will want a bit of tube color
and warmth to really make them sing, whereas other times a clinical and surgical approach is called
for. Do not be afraid to cut! It’s not always about boosting and sometimes you may have to turn the
bass down (laughs)!

Drezz: Headphones are an indispensable tool for the mixing and mastering process in my opinion.
Not everyone will agree. I know mastering engineers who don’t ever use them, but I think if you are
so inclined, they are a very useful addition to your tool kit. There may be sounds revealed to you on
headphones that you just don’t hear to the same degree on speakers. Again it’s good to know the
market and the choices available. There are many good ones to choose from. Try different ones if
possible, and find out online at forums and talks with others, certain phones will be suitable for dif-
ferent genres perhaps but people are usually willing to share their thoughts and experiences on these
matters. Speakers are also subjective. One person’s preference would be different to another, it’s all
about preference. Knowing your speakers or phones is very important. You have to rack up many
hours listening to a lot of different music on your system. Listen constantly, at a sensible volume with
lots of different material. Listen to your favorites. Listen to new commercial releases. Listen as much
as you can to get used to them. This is key. Obviously the room will need to be treated. This is a com-
plex subject which I cannot delve too deeply into here, but there is so much available info out there.
Minimizing the anomalies and frequency peaks and troughs is very important. Bass traps, diffusers,
broadband absorbers… there are many ways to tame the room. Building your own treatment can be
good for some, but it is important to get the right advice before embarking on such an undertaking,
but luckily there are some great people on the forums who are more than willing to give the benefits
of their advice. Check out what others have done!
for mastering?
Drezz: The new Acqua plugs are absolutely wonderful tools for mastering. Never before have ITB tools
been so good for this task. The Green EQ is a truly splendid EQ, clean, transparent, very lush and 3d,
this is the one I’d reach for most when mastering, but the Magenta and Amber are also fine tools for
the job. Again, it all depends on the source material. I find the Green a good all-rounder, it fits the bill
most times for the power and flexibility it offers. The Magenta, I feel, suits certain material more than
others. It’s a colored, warm beast, sometimes the material will call for that and sometimes not. The
Amber for me falls in between these two, it has some color, but is also useful for precise boosts or
cuts. They all sound great to me. As I said before, an unprecedented quality is now available to us and I
would not hesitate one moment in recommending any of them for mastering or mixing purposes. They
are an authentic replacement for their hardware counterparts, and used with experience, patience
and skill they can produce results that rival their hardware inspirations. The Titanium is a fantastic tool.
The multi-band is so versatile, but requires some learning if one is unfamiliar with multi-band com-
pressors, you can alter the tonal character of the track quite dramatically with this tool, it’s almost like
an EQ as well as a compressor, but caution is recommended, it’s quite easy to overcook!!! Patience
and caution required, but the rewards are there for the determined engineer. The new Aquamarine is
just awesome. This plug has some serious mojo, and the amount of attention to detail and flexibility
it offers is, for in the box, unparalleled and staggering. A future classic and nothing else quite like it out
there, the tone and characteristics of the compression are simple great. Again, it’s a beast and requires
some tweaking and learning, it’s sensitive, and very, very powerful, but so good (you may tell I really

like this plug and consider it a very special tool)! There are so many great libraries to choose form,
where is one to start? Tim P’s offerings are meticulous and very impressive, everything he’s done has
a real quality to it and attention to detail, the U76 and Silk EQ are standouts for me. The Cupwise 660
compressor is a great sounding vari mu type which just sounds lush to my ears. The Alex B consoles
are also very good. The new Prime bundle is an absolutely awesome suite of plugs, very suitable for
mastering purposes, and just sound great… warm, a lovely vintage character, but extremely high qual-
ity… a great sound.
Drezz: I am lucky in the fact that I have some nice vintage pieces of hardware in my studio. Sometimes
I will run something through a vintage console to get a bit of ‘hair’ or saturation/harmonic distortion.
I like to go through the transformers of the desk, which gives a very pleasant sound. Of course there
are transformer libraries for Nebula too, and sometimes I will use these, but occasionally it’s nice to go
through the desk, even if it’s going back in the box for additional processing, just for another layer of
color or some change in the tone. I also sometimes record stuff down to tape. I have Otari and Studer
machines and they sound very nice too. Of course there are the R2R and Apex collections by CD-
SoundMaster to emulate the tape, and very good they are too, but for me, being a very experienced
tape fanatic, there’s nothing still quite like going to tape. It’s my hope that one of the future develop-
ments of Nebula/Acqua is that the tape sound is finally nailed as this would be truly revolutionary…
we shall see!!!
this discipline?
Drezz: Start small, start slow, get to know the tools. You will soon start to find which tools work for which
tasks. Read, talk to others, don’t feel afraid to ask potentially embarrassing questions, everybody has
to start someplace, and most people who really know their stuff are willing to give advice. A mastering
chain is a very personal thing and comes about after experience. Personally I usually have something
like this, but it does change subtlety from time to time and job to job. Console>EQ>Comp>Tape>Lim-
iter… Sometimes the EQ after the comp, sometimes before. I don’t impose certain rules, it’s specific to
the raw material.

Gabriel Schwarz
Gemini Audio
Gabriel Schwarz: Nowadays the role of the mastering engineer is to make a finished mix sound as good
as it possibly can, while optimizing it for the most commonly used playback systems. In some cases
a mix gets mastered for more particular uses like, for example, the iTunes Store. Often it is expected
of the mastering engineer to make the mix as loud as possible. This seems especially important for
certain music genres, while in other genres it is secondary or even counterproductive.
Gabriel Schwarz: The biggest myth is that the mastering engineer can make crap to gold. This is not
the case. The mix has to sound as good as it can without processing the stereo buss except maybe a
bit of compression.
Gabriel Schwarz: I do all kinds of genres and the decisions vary greatly. Each genre has its own overall
sound which makes a record of that particular genre sound authentic.

not?
Gabriel Schwarz: Loudness is part of many genres and they probably would not be what they are
without it. But if genres which aren’t expected to be very loud are too loud, it can sound very bad.
Generally music has to be produced and mixed in a certain way in order to become very loud at the
end without sounding bad. I think if it sounds great, it is not too loud. But every mix has its own op-
timal level of loudness which should not be exceeded. I do not think loudness is a primary factor for
sales. I think it is more a question of “is it loud enough, or too loud for this specific genre, does it sound
authentic?”
Gabriel Schwarz: Parallel compression can be a great way to make the mix fuller and louder without
killing its transients. Also M/S compression can do great things to the spacial perception. For example,
it can be used to tighten up the lead vocals, kick etc. in the middle without affecting the sides.
What about EQ?
Gabriel Schwarz: Sometimes applying a high pass filter a little higher as you would normally do while
boosting the frequency right above your high pass.
Gabriel Schwarz: If it’s very good headphones, they can sometimes be helpful for spacial decisions.
Also since many people listen to music via headphones, it might be good to check the mastering
through some headphones as well. Most mastering engineers however limit themselves to one to
three different speakers because they’re experienced enough to know that what they do will translate
well on all playback systems.
What are your favourite Nebula/Acqua Libraries/Plugins? What other plugins do you think are suit-
able for mastering?
Gabriel Schwarz:
AlexB Vintage Master EQ
AlexB German Mastering Console
CDSoundMaster Mastering Suite
Gemini Audio NVE PRE 73
Gemini Audio KultComp Tube
Gemini Audio G2500
I used different tape programs from different developers.

Gabriel Schwarz: Yes, I use a SPL Kultube and sometimes a Bricasti M7.
Gabriel Schwarz:
High Pass Filter
Preamps / Tape stuff
EQs and Compressors
Brick-wall limiter
this discipline?
Gabriel Schwarz: Yes, but it is definitely harder and you need tools which can compete with the gear
which mastering engineers are using. It’s very important to get to know your speakers and room, and
to listen to your music on many different systems, so you know how what you are doing will sound
everywhere else.

Hubertus Dahlem
hubi123

Hubertus Dahlem: Polishing a hopefully good mix. He needs to understand where the mixing engineer
wants to go sonically. If it is a great mix he needs to polish it, if it is not the best mix he needs to make
it better or tell the mix engineer to rework the bits that lack in quality.
What are, in your opinion, the myths/clichés surrounding mastering ?
Hubertus Dahlem: Give the mastering a shit mix and he will make it a great one!
Hubertus Dahlem: Since I’m a Producer/Writer I work in all genres. In terms of decisions I may refer-
ence a track from a certain genre I’m working for in order to know where I want to go.
not?
Hubertus Dahlem: I think loudness is more a product of the record companies! If I have a new artist

and I introduce him to an A&R I definitely make sure that I’m in the same ballpark in terms of loudness
like my competitors. Loudness tricks your brain and I don’t want someone to think another track is
better just because it is louder. It should definitely never be „Metallica Death Magnetic“ loud.
Hubertus Dahlem: It took me a very long time to actually understand compression. For me it is one
of the most complicated subjects in mixing/mastering. What helped me a lot is to adjust the makeup
gain of the compressor you are working with per ear. Just bypass and engage and see if it has the same
volume. Some compressor have presets where once engaged it automatically adds 2 dB of gain. An
unexperienced mixer would be amazed but in reality it’s just the loudness that tricks your ears and
brain. Regarding compression on the Stereo Bus, play the loudest part of the song (the hook) and I
would max do 1-2 dB gain reduction.
What about EQ?
Hubertus Dahlem: Use Acqua EQ’s!! Since I’m part of Nebulas Beta Team my mind set in terms of EQ
has changed a lot. Back in the day I may have used a preset here and there and then tweak it. With
Acqua plugins you just use your ears. The plugs sound super close to the hardware, you use them like
hardware, there are no presets! I love all of them but my favourite is the EQP1, Blue and Green. My
friend Zino owns the Avalon 2077 and we tested it against the Acqua Amber EQ and it was pretty hard
to distinguish between the real one vs the plugin.
Hubertus Dahlem: At the end of last year my friend Zino told me that he bought a pair of headphones
for 1700€. I told him he’s crazy and couldn’t believe that any pair of headphones is worth that price,
especially because people say you can’t mix on headphones. I met him in Berlin and heard them. It
was mind blowing!! I bought myself a pair of Audeze LCD-X and didn’t look back since then. I’m not
kidding, but since using them I became a 20% better mixer. I now mix 70% of the time on the Audezes
and do the rest on my NS10’s.
What are your favourite Nebula/Acqua Libraries/Plugins? What other plugins do you think are suit-
able for mastering?
Hubertus Dahlem: As with EQ and compression, Nebula is unbeatable with reverb!

Reverb: AITB EAR is just beautiful rooms and halls.
EMT 140 from VNXT is from another world.
EQ: EQP1, BLUE, GREEN

Compression: Titanium, Aquamarine
Distortion: Vcult
I also love all the Henry Olonga’s and many more.

Hubertus Dahlem: Yes, I have a lunchbox with Elyisia Xfilter and Xpressor.
Hubertus Dahlem: It still depends on the track, but in general I have a Nebula console emulation ->
tape emulation -> Aquamarine -> Hardware EQ/COMP -> Titanium -> Limiter.
this discipline?
Hubertus Dahlem: Since I’m not a Mastering Engineer, it’s hard for me to answer that. But I think for
a mixing engineer it’s normal today to do a “pre-mastered” mix. What I mean by that is that having
a good working combination on the Stereobuss is definitely going to benefit your mix. Don’t put too
much on your master bus during the mix! It might make you believe that your mix is great where in
reality, after turning off your master bus, it falls apart!

Ken Suen
Himhui
Ken Suen: In my mind, the aim of a mastering engineer is to polish and finalize the sound with an
appropriate process in the conclusive stage. A mastering engineer has to maintain those songs at a
similar level and tone or adjust a definitive version of an individual song. The original character of the
mix should be kept as much as possible, unless there are unacceptable mistakes or producers have
special requests (I know, producers often are the “mission impossible” kind of guys). Honestly, it’s
better to do the mix again if there are big problems, it makes no sense to have a huge change after
mastering. Every previous process affects the upcoming processes. As we know, nothing is perfect, we
can only push it towards what we think is “perfect” in that moment. The mastering engineer has the
responsibility to make sure that the final product is compatible with different audio systems, though it
will perform differently amongst audio systems in cars, pc speakers, HIFI speakers, headphones, etc.
Ken Suen: Like many mastering engineers advised, get some well mastered CD’s playing before start-
ing your own session. It’s especially good for beginners, home-studio users or even professional
mastering engineers. Of course, practicing cannot be avoided. As we all know there is no shortcut,
thousands hours of previous mixing experience supports one’s mastering skill. Some problems may

be fixed while mixing and some can only be fixed while mastering. No two songs are identical, we will
face new challenges every time we have a new task. We’d better regard any gears as tools. Study what
is the good and bad of each gear on hand, we always have to make a comparatively good decision
to finish our job. Don’t expect that magic machines in the world might heal all the bad scores within
a single process. A good starting point is to write down every problem we detect, fix it step by step
until we hit the target. Remember to take a break after an hour or so of listening, refreshing our ears is
a smart move. Arguably, compressors and equalizers are the main tools in a mastering chain. We can
consider M/S processing in case we want to improve the sense of space in a dense mix. Again, don’t
over process with this plugins to avoid phase problems. Another practical plugin called monofilter4
from Nugen Audio is great to make certain frequency ranges from stereo to mono.
Ken Suen: I like all kinds of music with real instruments. Generally, I listen to the song and get the first
impression which is most likely what other people do. I then take notes and inspect the frequency
response over the song to see whether it matches the problem that I heard. There is some good soft-
ware to do cleaning job such as Izotope RX or even some free stuff. I would also like to recommend a
software called HAR-BAL which helps to understand different styles of music with different spectrum
phenomenon, though there is a learning curve to start. We might not finish our job using just that,
however it is capable of providing us with an overall picture of different styles of music. Our knowl-
edge assists us to determine the character the song should have. The pictures below demonstrate the
spectrum of two different kinds of music, the first one is a song of Rock’n’Roll, while the second one
is classical music.

We easily distinguish the different shapes between these 2 songs. We can go deeper to investigate
what instruments affect this behavior. What a gorgeous idea to analyze songs like this!
not?
Ken Suen: Personally I don’t like to push loudness to the limit. I like to preserve dynamics and space for
all instruments to live happily together. Once you squash sounds to its limit, it ruins the sound which
finally appears harsh and distorted. As a matter of fact, we can turn up the volume knob and thus we
are still able to perceive the relaxation in a higher sound pressure environment. The Loudness War has
been continuing for more than 10 years. We listened to a lot of beautiful music destroyed by this prac-
tice. I am happy to see that the Loudness War is gradually, although slowly, fading away. Occasionally
though, we will still be asked to make a super loud product.
Ken Suen: Compressors and EQ’s are daily tools in our life. We can’t help touching them once we see
them. Just like a cat never lets go of the fish! Compressor is the tool to control volume/dynamics. We
must have the basic knowledge about the attack times, release times, threshold and ratio. Once we
understand those concepts, we take all of these parameters as a starting point. Each compressor, es-
pecially hardware units, has its own response and character, we just use our ears to adjust the sonic

results. If we want to get more control in a stereo track, the “LINK” mode will be more appropriate, Dual
Mono compression reacts more aggressively, and sometimes unpredictably. Obviously, either choice
is just a matter of taste.
What about EQ?
Ken Suen: EQ may sweeten, tighten and clean the sound. One very important rule, don’t abuse it!
Apply it when it is necessary. Let the sound perform naturally unless you have a special purpose. We
can make it bright but not harsh, we can make it warm but not blurry or muddy. The best way is always
to use our ears to decide what sound we need. If it sounds right, it is right!
Ken Suen: Headphones help us zoom on small details like clicks and pops in the track. As I mentioned,
because many people listen to music via earphones or headphones, we should always reference our
work on a good pair. We should carefully choose our headphones. I like the SONY MDR-V6’s very
much for this purpose.
for mastering?
Ken Suen: My favorite mastering plugs are Magenta EQ\Green EQ, Titanium Multiband Compressor
and Nebula tape programs such as R2R from CDSoundMaster, sometimes I will add German Console
(from AlexB) and Vintage Tape machine (VTM-M2) in the chain. I also like the Dynamic Spectrum Map-
per (DSM from Paul Frindle).
Ken Suen: Yes, I still keep some hardware like M7, Elysia Empressor, TFPRO P38ex Compressor and P9
Equalizer. I use it occasionally when I feel that it is suitable.
Ken Suen: I don’t have any special setup for the mastering chain. I like tweaking sound from the begin-
ning so that I can feel and hear each variation during the process.
Is it possible to master at home? What tips what you give to someone who is about to start learning
this discipline?
Ken Suen: I understand the perfect solution is hiring a room with good acoustic treatment, good far
field/mid field/near field monitor and all top notch machines. As a bedroom producer, I realize that
there are tight budget projects, sometimes we have to struggle in a difficult environment. Yes, we still
have to invest in a pair of workable speakers and some gear. We must check frequency interactions

and response between the room and the speakers in order to get a flat-frequency-response monitor-
ing position. There is some hardware or software products that work well, for example, the ARC2. Fully
use what you have, even a pair of PC speakers may do the job. I find that it’s good to check the reverb
tails and vocals with most PC speakers. If you intend to master your own mixes, I do not recommend
doing it in the same day. The best approach is to let the music rest for a little while.

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Steve Turnidge (Mastering Engineer/Author)

TIP: Read Acustica’s Manifest on the website.

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

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

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

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Monitors and quality headphones

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Filter: A circuit which alters a limited range of frequencies.

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

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

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

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

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Basic EQ Quick Tips:

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M/S (Mid-Side) Technique

Thus we have two simple formulas:

Mid = Left + Right

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

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

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The options are endless.

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

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

Band 1. 28 Hz – 1.5 kHz

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

Band 3. 200 Hz – 10 kHz

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

Band 4. 650 Hz – 27 kHz

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

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

Mastering with Acustica / 0033

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

Creative (loudness-oriented) mastering:

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

Conservative mastering (without compression):

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

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

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