make music now Compression masterclass

ADVANCED

EFFECTS

PART 1

NEXT MONTH

REVERB

Compression
For the first in our brand new six-part effects masterclass,
were getting those dynamics under control

On the DVD
TUTORIAL FILES
Weve supplied some files
to go with our walkthroughs
(in the Tutorial Files folder),
but if possible, try to use
your own instead as this will
give you a much better feel
for how compression affects
your style of production

ompression is one of those things

that crops up and time and time
again in cm, and with good
reason. In general terms, its the most
important type of processing that you
can apply to a sound, which also
means that its one of the worst
sounding (and most difficult to undo)

when you get it wrong. In the first in this

six-part series of expert tutorials, were
going to show you how to get the most
out of your compression plug-ins. But
before we go on, lets have a little recap
on the basics
Compression falls into the category
of effects processors that we call

Some different types of vintage compressor

In a horribly clinical world, all compressors would respond in exactly the
Photo-resistive
same way at the same time, but thanks to the subtleties of design, the
fallibility of human beings and the nature of analogue circuitry, every
compressor design has slightly different qualities to the next, and these are
what make the sounds of some so desirable. Now, while not everybody can
afford a genuine vintage analogue compressor (or even an emulation of one),
by understanding the way different types of compressors act, we can coax
much more convincing emulations from even standard plug-ins. Lets have a
look at a couple of types of analogue compressor

1Valves: warmth through imperfection

Valves

Probably the earliest type of

compression circuitry was built
around valves. Valve compression is
famously non-linear. In other words,
the amount of gain reduction applied
when the incoming signal first

064

exceeds the threshold is not

proportional to the amount applied
when the signal far exceeds the
threshold proportionally less is
usually applied the more the signal
exceeds the threshold. Valves also
cant react quickly enough for very
sharp transients, but this is
compensated for by the inherent
limiting of valve circuitry. This limiting
also gives rise to one of the nicest
side effects subtle, warm distortion.
To recreate the effect of valve
compression you can try using the
following chain: compression (with
the attack set above 10ms or so and
any look-ahead function disabled)
followed by some gentle limiting and
a very small amount of overdrive.

Photo-resistive gain control

compressors are as ingenious as
they are slow to respond. They work
by displaying the signal intensity with
a light bulb or LED (the stronger the
signal, the brighter the light) and
using a photo-sensitive receptor to
respond to it and reduce the
volume accordingly.
Because of the non-linear and
slow response of light sources
(particularly the bulbs used in early
devices, for example), the response
of the compressor is much slower
than that of modern plug-ins, and
can easily miss fast transients
altogether, making for some rather
interesting results. So, to get that
vintage photo-compression sound
from your software, all you need to
do is use much slower attack and
release times than you normally
would. And interestingly, because
photo-receptive designs essentially
just reduce the volume, theres no
distortion introduced, so you neednt
worry about adding any

dynamic (effects that control changes in

sound level). When an incoming signal
exceeds a user-specifiable level, the
compressor will automatically reduce
the volume by a user-specified amount.
It will do this at user-specified speed,
and, after the incoming signal drops
below the level again, it'll return the
signal to its original level at another
user-specified speed.

Speed of sound

So far it sounds like compressors are

just glorified volume controls. And in a
sense they are, but with much faster
reactions than any engineer. Imagine if
Clark Kent were a recording engineer by
day, rather than a reporter thats the
kind of reaction speed were talking
about! And even hes only got two
hands so could handle a stereo signal
at most.
Over the years there have been
numerous variations on this theme, but
it's at the core of every compressor you
own, so its vital that youre comfortable
with the concept before you continue.
OK, with the basics out of the way,
lets head over to the next page and
have a look at those controls in some
more detail

PRO TIP

Many producers like to place some gentle

compression on the output of their mixer to help
the different channels gel. If so, you should try
inserting it before setting the levels, as it will affect
the relative balance of the parts when applied.

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Compression masterclass make music

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THE ANATOMY OF A COMPRESSOR

01

THRESHOLD As with most

threshold controls on audio
equipment, this rather
unsexy knob simply tells the compressor
at what point it should start to take
effect. Setting the threshold is a bit of a
black art, and potentially the subject of
its own tutorial check the boxout on
p66 for more)

02

RATIO The most

misunderstood of
compressor controls.
Everybody loves to play around with it,
but few people understand what it
actually does. In reality, its a very simple
device that controls how much volume
attenuation (reduction) is applied to the
signal. So, for example, a ratio of 4:1 will
mean that any signal that exceeds the
threshold by 4dB will only be output at
1dB above the threshold. The greater

SIDECHAIN EQ Some
modern compressors now
offer the ability to exclude
certain frequencies prior to
compression (high
frequencies, for example),
which helps avoid audible
pumping effects.

LOOK-AHEAD Look-ahead is a means of dealing with

very short-lived, fast transients. It does exactly what it
says on the tin ie, it looks at the signal to be played
before it's sent to the gain control stage. In hardware this
is done by delaying the signal by a few milliseconds
before sending it to the gain control stage, but in software
it sometimes involves sneaking a peak at the audio file
before it's played. This digital versatility means you'll
normally only find look-ahead on plug-ins.

STEREO LINK Stereo

linking is vital if youre
applying compression to a
finished mix, as without it
sharp transients in one
stereo channel might
trigger noticeable
compression in that
channel but not the other.

PEAK/RMS Most compressors respond to sharp

transients in much the same way as the human ear,
meaning loud but short transients arent perceived as
being as high volume as more prolonged sounds at
exactly the same level. This is called RMS (Root Mean
Squared dont ask) operation, and allows short
transients through for a more natural sound. It's not so
appropriate for applying pre-AD converters, where even
short peaks above 0dB cause clipping.

the ratio, the more squashed the

dynamics become.

03

ATTACK Attack is a critical

parameter with the power
to completely change the
character of a sound. Very fast and the
compressor will squash your signal;
longer and the crack of any sharp
transients will slip through.

04

RELEASE The release

control simply tells the
compressor how quickly to
bring the signal level back up after
compression has been applied. This can
either be instantaneous or very slow
indeed. The effect the release has varies
wildly depending on what the source
material is. Try manipulating it on a
grouped drum track and then a
complete track or bassline.

05

MAKE-UP GAIN Because,

by nature, compressors
reduce the volume level
overall, its important that the final
output signal is raised, and for this we
use a simple gain control at the output
stage (some designs even include one
at the input stage too).

mix. The knee control compensates for

this by gradually applying compression
to signals approaching the threshold
until the ratio reaches the defined
amount just as the signal hits the
threshold. The resulting compression is
therefore no less determined, but much
more subtle.

06

07

KNEE Another much

misunderstood
compression control is the
knee. Knee is usually measured rather
scientifically from soft to hard. On a
compressor with no knee control (or one
set to hard) the compression doesnt
begin until the signal reaches the exact
threshold. This can result in rather
dramatic and obvious dynamic changes
that sound great on punchy drums, but
like a pile of steaming used grass in a
cow field when applied to a complete

BYPASS (Not shown) One

of the most vital buttons on
your compressor, this
enables you to actually hear what the
compression is doing to the sound. If it
sounds better bypassed then it is! Ignore
this at your peril

08

SIDECHAIN (Not shown) A

compressors action can
often be triggered via a
separate signal (such as a kick drum),
rather than the incoming audio.

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Sticking it in the right place

Get your compression inserted correctly and all kinds of
great things can happen
PRO TIP

Most software compressors come with a variety of

presets and enable you to save your own. These
are usually labelled for different types of
instruments, so try them out and save any
modifications you make as a new preset.

ompression is generally most

efficacious as an insert effect.
That is to say, its placed between
the input to a channel and that
channels fader, so that only the
processed signal is heard. But in these
days of near-limitless plug-ins on every
channel, most stuff tends to end up
inserted, including delays and reverb,

which were traditionally used as send/

return effects. Consequently,
compression in the DAW age is
interacting with our other plug-ins in
new and unexpected ways, meaning
that compressors can now play a much
more creative role in the use of other
effects. You can see what we mean in
the walkthrough below

PRO TIP

Good compression isnt necessarily the loudest, the

most transparent or the fastest its the most
appropriate. Acoustic or orchestral music will need
less compression than bangin hard house, so
check with pro productions in your genre.

STEP BY STEP Using compression in an insert chain

First, load a track or loop youve made with a simple

synth or percussive pattern anything will do, just as
long as the sounds are quite distinct (piano-style
envelopes not strings). Solo the synth and insert a
compressor with 3-6ms Attack and 70-100ms Release. >>

Insert a simple delay (preferably a tempo delay, but

make sure it sounds in time!) and a short reverb. Keep
the delay feedback level low at first, then, using the
reverb time and delay feedback controls, fill out the keyboard
pattern. Doesn't sound too great, huh? Have faith >>

Finding the right settings

One of the main uses for compression is
controlling the dynamics of a signal, thus
achieving a louder sound overall. In such
cases, the most critical control will be
threshold ideally, the threshold setting
should be just above the average signal
level. But how do you know where to set
it? Well, in much the same way as one
exaggerates EQ gain and uses a narrow
Q setting when sweeping for the right
frequency to treat with an equaliser,
extreme values are often the order of the
day when searching for the right
compression settings.
If you want to know what the best
threshold setting is for your signal, try
using an extremely short attack and
release with a very high ratio and then
sweeping the threshold until you hear it
starting to kick in from time to time
during louder bursts. From there you
should first reduce the ratio setting to
something more sensible, then begin
increasing the attack and release times
and adjusting the other parameters.
This technique neednt be limited to
controlling peaks. Whatever your use for
066

compression, threshold will be

critical, and careful setting
essential. (Most of the time, when
signals are over-compressed, its
down to the fact that thresholds
have been poorly set.)
A methodical approach
becomes even more vital
when we move into the murky
world of multiband compression
(see p68).

Reduce the reverb and delay feedback settings to their

previous levels and insert another compressor, this time
with more gentle settings, after the reverb and delay. By
playing with the Threshold and Ratio settings, you can
increase the volume of the effects without intensifying them.

Doing it in the
right order
Just as compression can have a huge
impact on other effects in an insert
chain, there are some processes that can
affect a compressors performance, most
notably EQ. Anybody whos been knocked
off their feet by a bass-bin at the Notting
Hill Carnival will tell you it takes more
energy to produce bass sounds than
anything else, so it stands to reason that
by boosting the bass, youre also
substantially boosting the level.
Consequently, any bass gain applied
pre-compression will cause the
compressor to kick in sooner,
specifically when the signal youve
just tried to boost kicks in. At best
this can undo much of our careful
EQing, and at worst itll ruin our
signal completely. So its usually best
to apply EQ after compression!

7Apply compression like you would any other

vital process: be methodical!

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STEP BY STEP Pump it up

Sometimes a good pumping sound is just what the

doctor ordered (check out the strings on Eric Prydz Call
On Me to hear what we mean), but what if you want to
control when the pumps occur? Well, set up three channels
in your softwares mixer and well show you how its done. >>

On channel 3, set up a kick drum on every beat; on

channel 1, any track with a kick on every beat; and on
channel 2, a string pad playing a solid note or chord.
Insert a sidechain-equipped compressor into channels 2 and
3, and disable the audio output of 1. >>

Finally, set the sidechain inputs on both compressors to

receive their input from channel 1. Audition channels 2
and 3 separately and youll hear a pumping effect on
the offbeat. Used overtly, this breathes rhythm into solid
strings; used subtly, it gives life to dull mixes.

STEP BY STEP Keeping it close and clean

Another use for compressors is to flatten the level of

wildly fluctuating signals, and nowhere are these more
common than in real-world recordings. Record
something into a microphone and listen back to it. >>

The volume level of your voice or instrument will almost

certainly be all over the place. Now insert a compressor
with quite a low ratio and moderate Attack and
Release settings. Lower the Threshold slowly and the level
should start to settle. >>

This technique can have many effects slow attack

and high ratios can reduce the sound of unwanted
transients, like a guitar pick, while low threshold and
high ratios on vocals can make them sound more breathy
and intimate. Just don't forget to use a noise gate!

STEP BY STEP Make it bangin

If compressing real world recordings is all about

preserving their natural sound, compressing drums is
the opposite. Drums just sound better processed than
they ever do in real life! Load up an unprocessed acoustic
drum loop and insert a compressor. >>

Play back the loop clean to begin with, and then add
the compression. Have the Attack set to 0ms, initially,
and set the Release to 50ms. Set the Threshold to
about -20dB and try raising the Ratio. As this comes up, you
should notice the weight of the kick increase considerably. >>

After a certain point, the sound will become duller and

flatter, as the ratio starts to take its toll. To counter this,
increase the Attack time slowly to no greater than 8ms.
As this is increased, the attack portion of the drums will
become more pronounced, breathing life back into them.

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make music now Compression masterclass

Mastering (with)
multiband compression

Why have one compressor when you can have, er, loads?! Learn
how multiband compression can help you master your mix

s great as compressors are, they

do have their limitations. Weve
already talked about controlling
the peaks of a final mix with
compression, but we can do much
more than that with a multiband
compressor. This is essentially multiple
compressors in one, each affecting a
different frequency band. Why is this
useful? Well, just remember the two
main uses for compressors: taming
peaks and raising the overall loudness.
If our track happens to be heavy in the
bass section, then any compression
applied will be consistently triggered by
the strongest signal the bass. But the
signal attenuation will be applied to the
entire mix, creating noticeable dips and
pumping in the treble part, resulting in a
dull and unpleasant mix. With this in
mind, a multiband compressor enables
us to compress only the parts of the
mix that need it, so we can increase the
overall level of the bass, but leave the
upper range intact. The advantages of
multiband compression are obvious, but
getting to grips with it can be tricky

the upper band needs processing, dont

un-bypass them straight away
chances are youve just
overcompensated and turned the bass
up too high. Try to fix that first.
Once youve mastered the art of
selective frequency compression, then
you can start applying what youve

DONT RUSH IT, AS EACH BAND

ADDED DOES FAR MORE THAN
DOUBLE THE LEARNING CURVE
learned and refine each band
individually. But dont rush it, as each
band added does far more than double
the learning curve.

Picking your range

The last critical controls to master on a

multiband compressor are the band

Getting started

Our best advice is to start simply.

Multiband compression usually comes
in three bands, so bypass two of them
and concentrate on the area that needs
work (after all, if you find every band
needs heavy compression, perhaps you
should rethink the mix first!). If, for
example, the mids and uppers sound
OK and you apply compression to only
the low band, then suddenly find that
068

crossover points in other words, the

ranges to be processed. While these
will vary to some degree, there are
some common ranges. But these wont
always be applicable, and will depend
on the types of instruments you're using
(low guitar notes may fall into the range
of the low compressor, while high piano
patterns might not), so repeat our more
general advice and start a band at a
time. If you're processing the bass but
you feel too much of the mid is being
affected, sweep down the range until
only the desired range is processed. Its
also worth noting the extreme setting
technique once again at this point. The
easiest way to find out where to set the
band range for compression is to
whack your ratio up, with nice fast
attack and release times. Once youve
found the frequency to be affected, you
can use the same techniques as for
any type of compression.

Easy does it

If we make all this sound simple, thats

because it is. But simple doesnt mean
easy, so just remember to take things
step by step, and youll find that it's
much less impenetrable than you might
have at first thought. cm

Some typical
compression
settings
ACOUSTIC GUITAR
Attack: 1ms
Release: 98ms
Ratio: 6:1
Threshold: -18dB
Soft Knee, RMS
DRUMS
Attack: 0ms
Release: 90ms
Ratio: 4:1
Threshold: -18dB
Soft Knee, Peak
VOCALS
Attack: 1ms
Release: 120ms
Ratio: 6:1
Threshold: -17dB
Soft Knee, RMS
ELECTRIC BASS
Attack: 19.5ms
Release: 130ms
Ratio: 6:1
Threshold: -18dB
Soft Knee, Peak
COMPLETE TRACK
Attack: 26ms
Release: 29ms
Ratio: 3:1
Threshold: -14dB
Soft Knee, Peak
PUMPING
Attack: 6.5ms
Release: 2ms
Ratio: 9:1
Threshold: -35dB
Hard Knee, Peak

Taking it to the limit

No discussion of compression would
be complete without mentioning
limiting, but rather than go over the
same old ground, wed like to dispel an
important myth. Its popularly accepted
wisdom to describe limiting as
compression with an infinite ratio ie,
it attenuates any signal above the
threshold completely. Now, while this is
true in theory, in practice, compressors
usually arent designed to work that
way and will almost always have the

potential to allow some transients

through. Its a bit like bouncers on the
door of a nightclub: a limiter is the
seven-foot Neanderthal blocking the
door, while a compressor is the type of
doorman who wont notice you slip by if
youre small and can move quickly.
Unfortunately, in the digital age,
even small transients can be enough to
cause noticeable signal clipping, so in
practice, don't send a doorman to do a
bouncers work

1Know your limit: limiters are the big, bad

bouncers of the music-making world. Apparently

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make music now FX masterclass

ADVANCED

EFFECTS

PART 2

Reverb

For a quality mix, reverb is essential. Get to grips with this

effect and itll transform your sounds

On the DVD
TUTORIAL FILES
You'll find our reverb loop
in the Tutorial Files folder

hat can we say about reverb

that hasnt already been said?
Well, plenty, hopefully, or weve
all got a very dull four pages ahead of
us! Before we start, though, a brief recap
on the basics
Reverb is the product of sound
interacting with nature. Every time we
make a noise, its transmitted as waves
of vibration through the air. These waves
ricochet off every surface they
encounter, before finding their way to
the listeners ears. Because were so
used to hearing these sonic reflections,
any situation in which theres nothing

for the sound to bounce off sounds

unnatural. With this in mind, successive
generations of music productions have
used a variety of ingenious processes to
recreate these reflections, culminating
(for most of us) in digital reverb units in
the shape of plug-ins.
Theres never been a better time to
explore the possibilities your reverb
plug-ins throw up, which makes it all the

Common types of reverb

more surprising that the vast majority of

us do little more than call up our
favourite reverb and insert it into any
channel we need. If were feeling
adventurous, we might even alter the
reverb time
All of this is a real shame, as almost
every space you encounter will have a
distinct sonic character. Some of these
are suited to vocals, drums, or even
complete tracks. Clever use of reverb
techniques can transform your mix from
a lifeless lump of deadwood to a vibrant
masterpiece. Dont believe us? Well
then, read-on cm

7There are many types of reverb so do play

the field with your plug-ins

SPRING

PLATE

CONVOLUTION

DIGITAL

In the pre-digital age, the only way to

achieve the sound of reverberation
was by mechanical means, and this
generally meant spring or plate reverb.
Spring reverb is a simple concept:
usually found inside analogue guitar
amps, it involves suspending springs
between pairs of transducers with
vibrations being generated at one end
and received at the other. The effect,
while interesting and often useful, is
fairly unrealistic, and the vast amount
of gain required to boost the received
signal makes it prone to extreme
signal noise.

Plate reverb is similar in principle to

spring reverb, in that real vibrations are
induced in a real piece of metal by a
driver and then picked up by
microphones. The difference is in the
quality of results plate reverbs can
sound absolutely lush albeit in a
not-quite-natural way. Originally the
preserve of top-end studios, theyre
surprisingly cheap and easy to build,
although with an average size of 4x6,
they arent exactly practical.

The daddy of modern digital reverbs,

the convolution reverb uses
painstakingly accurate sonic models of
a specific and real acoustic space, so
that any sound to which its applied
appears to be playing in that
environment. For example, in the room
youre currently sitting in, a series of
specialised sounds would be played
and recorded with a variety of mics.
Using clever calculations and
deductions, a sonic blueprint (called
an impulse response, or IR) would be
made so that any sound fed into it
would appear to be interacting with
exactly the same environment, taking
into account every table, chair, window,
sofa even you, if you happened to be
in there when they took the blueprint.

At the risk of oversimplifying things,

lets just say that digital reverb is
usually achieved with a combination of
very fast delay algorithms and filters.
And if youve ever wondered why
reverbs are among the most
processor-intensive effects you can
use, then you must not know that to
create even a modest sounding digital
reverb, said algorithms must churn out
anywhere from 1000 to 3000 delays
every second. And they must impart
clever randomisations of timing
between each one so that the effect
isnt completely unnatural.

060

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now

THE ANATOMY OF A REVERB

01

WET/DRY This specifies

the amount of clean signal
versus effected. If youre
using your reverb as an insert, this will
be used to control the intensity of the
effect, but in a send/return loop, the
normal setting will be 100% wet,
with the send level of each track
determining the intensity and amount of
reverb on each.

02

REVERB TIME Its hard to

say at what point
reverberations actually stop,
so the reverb time (aka decay time) is
accepted as being the time it takes for
the reverberations to decay by 60dB.
Longer equals bigger sounding with
normal rooms in the 1-3 second range,
and huge spaces like cathedrals dishing
out ten seconds or more.

03

EARLY REFLECTIONS
When a sound is produced,
theres a slight delay
(perceptible or not) before it encounters
a reflective surface. Consequently,
smaller rooms will generate early
reflections that are audible before being
lost in the reverberant mush that follows.
The brain makes an educated guess as
to the size of a space based on these
early reflections.

04

PRE-DELAY The amount of

time between the original
sound and the early
reflections. Too short and you lose
definition, too long and theres an
audible gap (such as in a very large
space), generating an echoing effect. As
a rule of thumb, try to achieve the
longest pre-delay time you can without
an audible gap.

05

ROOM SHAPE Many reverb

units include a room shape
option, usually determining
the number of walls, but sometimes also
their placement.

06

STEREO BASE Some

reverb designs enable you
to simulate the placement
of stereo microphones. To accurately
mimic the perception of a real room, try
placing them around the same distance
apart as your ears.

07

INITIAL DELAY This is the

elapsed time between the
original signal and the main
diffused reverb effect. The diffused
reverb should flow on from the early
reflections, and again, a good technique
is usually to have the longest initial
delay without an audible gap
between the early reflections and the
diffuse reverb.

08

CROSSOVER Because
bass reverb can overpower
a track quickly, some
reverbs enable you to set the bass and
mid/treble reverb times independently
the crossover setting in this type of
reverb will determine at what point the
signal is split.

09

LOW RATIO The low ratio

control is essentially a
reverb time control for the
bass end in this case expressed as a
percentage of the main reverb time.

10

LOW LEVEL If your reverb

enables you to split its
action, it will usually have a
low level control so that you can set the
level of the low reverb effect
independently in this case, a setting of
0dB means the levels of both the high
and low bands are equal.

11

DENSITY This controls the

amount of reflection, and
essentially sets the
thickness of the reverb. This should
normally be as high as possible without
colouring the sound.

12

DIFFUSION Diffusion
controls alter the regularity
of the individual reverb
delays (of which weve established there
are thousands per second). High

diffusion values mean more irregularity

and, therefore, greater realism.

13

HIGH CUT High frequencies

tend to get absorbed much
more quickly than low
frequencies, so in nature reverb tends to
lose top end much faster than bottom
(except in stone or other highly reflective
environments). A high-cut (also known as
damping in many units) filter simulates
this effect.

14

SPREAD Natural reverb is

an unnatural and unruly
thing, so it can be very
difficult to detect much in the way of
direction from it. Nevertheless, clever
algorithms that offset the early
reflections and longer reverberations as
they play through the left and right
channels can be used to add stereo
width to even a mono signal. Some
convolution reverbs, such as WizooVerb
W5, can accurately model threedimensional spaces in glorious
5.1 surround!

15

ROOM SIZE Determines the

size of the virtual space
created by your reverb plugin. Obviously.

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make music now FX masterclass

Some tasty tricks with reverberation

STEP BY STEP Natures way

Ambient mics are often used to capture the sound of a

real room when recording drums, but a similar process
can be used with any signal. Start by setting up a
speaker in the space you want to capture or perhaps even
in an adjoining room with the door open. Then place a
microphone in the space to be captured facing away from
the sound source. >>

Make sure the mic is a good distance away from the

speaker and adjust the playback volume until youre
fully aware of the natural reverberations. Be sure to
mute the output of the microphone channel, so as to avoid
any feedback. Then simply play back the source be it a
vocal track, instrumental part or complete mixdown and
record the reverberations. >>

Try recording a few different speaker and mic

placements, then unmute the recordings and play them
quietly behind your original source. For a nice stereo
effect, feed one recording to the left channel and another to
the right. And there you go an easy, effective and natural
reverb effect thats great for any occasion.

STEP BY STEP Reverb 3D

Reverb can be used to add more than just stereo width.

Armed with nothing more than two ears, we can easily
detect direction with sound, so theres no reason why
we shouldnt use stereo reverb to equal effect. Start by
placing our two-bar Reverb Loop (on the DVD) onto a new
track, and a reverb unit (set to 100% wet) on a bus. Send the
output of the track to the reverb bus. >>

Now, use automation to lower the volume of the reverb

bus from a high point of your choice to a low point of
your choice and back up again, with the lowest point
exactly halfway through two bars. Next, automate the pan from
centred to fully left and back to centre over one bar, then fully
right and back to centre over the second bar. >>

If all is well, the reverb should sound like its running in

a large circle in front of you. So far this effect is pretty
severe, so to moderate it slightly, simply ease back the
extreme pan values, and lessen the volume range. To really
add some dynamism, try applying the exact opposite
automation to the audio loop.

Tips on adding reverb to drums

FOR A BIG room sound, try using a hall
setting on your reverb and adding some
pre-delay, but avoid adding too much of
the effect to your kicks and toms, as predelay and long-reverb times will fill your
mix much faster than an orchestra paid
by the note
TOMS GENERALLY DONT NEED much
reverb as their natural envelope has a
pretty extended sustain, but if you do
want to add some presence and a sense
of scale, try using quite short settings
062

and be sure to remove the bass from the

wet signal.

real kick drums, but for electronic kick

drums its best to avoid standard reverb.

SNARE DRUMS ARE very bright and so

can sound tasty with most types of
reverb it all depends on the effect
youre after. For punch, try a plate style
reverb, as this has a fast, clean envelope.

IF YOU ABSOLUTELY must apply reverb

to your electronic kicks, try playing
around with some gated reverb settings
or reverse reverb. Beware, though: these
are both very bold effects so make sure
the result sounds absolutely mint!

APPLYING REVERB TO kicks can be

troublesome, as the low frequencies
generated can really muddy your mix.
Shorter, ambient settings are better for

TRY TO AVOID long reverb tails on

hi-hats unless youre doing it for specific
effect, as these can sound quite

distracting and unnatural, and will really

muddy up the top-end of your track. For
most purposes, shorter but bright reverbs
add a sense of size and space without
being overly intrusive.
FOR A REALISTIC live sound, you can
still apply different reverb settings to
each drum sound, but try sending the
individually processed sounds to a submix and then applying some subtle
ambience reverb to them as a complete
drum kit.

| COMPUTER MUSIC DECEMBER 2005 TUTORIAL

CMU94.t_fx 062

4/11/05 6:29:33 pm

FX masterclass make music now

Tips for adding

reverb to vocals and guitars
WHEN RECORDING VOCALS, be sure
to add some reverb to the singers
monitoring channel (even if its only a
lower quality one) to ensure smooth
playback. The reverb tail extends after
the singer finishes singing, so many
singers will use this to reference their
pitch accuracy.

5Spring reverb: perfect for guitars, and

on the cm DVD in the shape of SpringVerb

IF YOUR VOCAL line is suffering from

too much sibilance, try adding a
de-esser to the reverb rather than the
original signal. De-essers can sound a
little unnatural sometimes, and this
technique can often provide less
obtrusive but nonetheless highly
effective results.

EARLY REFLECTIONS CAN add a

sense of intimacy to a signal,
particularly with acoustic sounds like
guitars and vocals the bright, quick
reflections sound like a small room or
venue. For this effect try using nothing
but ambience settings initially, and then
add a more conventional reverb setting
slowly afterwards (or not at all).
A GOOD WAY TO stop reverb
dominating your vocal tracks is to use
a ducking gate to lower the reverb

TRY USING DIFFERENT TYPES of

reverb on one track. In days gone by,
reverb was expensive and limited, so
it was placed on busses, but high
quality plug-ins and fast processors
let you use multiple reverbs as
inserts for total control.
REMEMBER THAT IN most cases,
slower, sparser tracks can
accommodate longer reverb times
than quicker and thicker mixes
without making them messy.

IF YOUR VOCAL LINE IS

SUFFERING FROM TOO MUCH
SIBILANCE, TRY A DE-ESSER
USE REVERB TO increase the
separation between vocals. Backing
vocals should be just that, so try adding
plenty of early reflection and longer
reverb tails to push them back in the
mix. Conversely, lead vocals can sound
a little distant with longer settings.

General reverb tips

signal by a few dB when the vocal's

playing. If you dont have a gate with a
ducker, try applying a compressor to
the reverb signal and send the vocal
signal to its sidechain.
ELECTRIC GUITAR PARTS (especially
those of the chord-driven, wall-of-sound
persuasion) are already very full, so be
careful with your reverb lengths. Spring
reverb patches are traditional and
sound the part (bright and clean), but
dont be afraid to try out something a
little different.
GATED AND REVERSE reverb can
sound excellent on guitar parts, as they
fill out the sound without swamping it,
and won't result in the obvious and
clichd sound that they can produce
on drum parts.

1Logics Enverb is just the thing for reverse and gated reverb effects

FOR A REALISTIC live sound, as you

increase the reverb time you should
also reduce the wet
signal using the wet/
dry balance, and as
you shorten the
reverb time you
should increase the
wet signal.
USE REVERB TO
impart a sense of
three dimensional
space rather than
just width. For
example, if a
sound has a large amount of reverb, it
gives a feeling of distance, just as the
sound of somebody singing at the other
end of a room is heavily mingled with
the reverberations.
TRY USING DIFFERENT reverb effects
on the left and right channels. This can
be quite fiddly on individual tracks, but
can be done much more easily for a
global reverb send/receive setup.
IT SOUNDS OBVIOUS, but dont
forget that all the flexibility of
modern digital reverb is wasted if
the source material is a
reverberating mess to begin with.
Were not saying you shouldnt
take advantage of natural reverb
(check our walkthrough), but if
you plan to add any kind of
processed reverberations, the
cleaner and dryer the source
material, the better!
REMEMBER THE BLUES Brothers
scene where Jake says to Elwood, How
often does the train go by?', to which
Elwood replies, So often you wont even
notice'? Well, its true we humans
notice change more than consistency,

so don't be afraid to muck about with

reverb lengths and parameters
throughout a track!
WHEN THICKENING SOUNDS, try
applying chorus or flanging to the reverb
signal. Alternatively, if youre using your
effects in an insert chain, try applying
reverb before any modulation effects
(though for a more natural and subtle
effect, the first technique is probably the
better option).
DONT OVERLOOK THE power of
compression on reverb, particularly
reverb used as a track insert. For a

1Logics Space Designer is one of the most

powerful convolution reverbs on the market. It
deserves to be fed only the cleanest signals!

spectacular pumping reverb effect, try

running your kick drum into the
compressors sidechain.
IF YOU PLACE a gate in the signal path
before a reverb, you can adjust it so that
only the louder signals make it through
and generate reverb. Or by still allowing
a small amount of signal through when
the gate is closed, you generate more
reverb for louder sounds this works
especially well on vocals, but be sure
not to over-compress the source signal
or the effect will be lost.
IF YOURE USING heavily panned
sounds, try applying mono reverbs
panned to the same sides as the
panned sounds to emphasise their
positions in the mix.
TO FILL OUT a sparse mix, try using a
combination of delay and reverb, but
with one panned mostly (or completely)
left and the other right. For even more of
a sense of movement, trying sweeping
these effects so that as the delays
move right, the reverb moves left.

TUTORIAL DECEMBER 2005 COMPUTER MUSIC

CMU94.t_fx 063

| 063

4/11/05 6:29:40 pm

MAKE MUSIC NOW ;MbVhiZgXaVhh

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make music now Modulation FX masterclass

ADVANCED

EFFECTS

PART 4
On the DVD
TUTORIAL FILES
The audio files used in the
walkthrough on p51 can be
found in the Tutorial Files
folder on the DVD

Modulation

Discover what modulations all about and how you can

incorporate its family of effects into your tracks

his month in our Advanced FX

series were looking at the class
of processing tools and
techniques known as modulation
effects. This is a bit of a strange one, as
unlike reverb and delay effects (all the
respective types of which are merely
variations on a theme), modulation
effects are connected in the sense that
they all share a common element

namely, they involve some type of

modulation. In other ways, many
modulation effects can be quite
different. So whats it all about?
Well, modulation, in production
terms, simply means using one

parameter to control another. In

practice, with effects, this almost always
involves using a Low Frequency
Oscillator to exercise said control,
although there are some exceptions.
But never mind that for now: if
modulation effects is simply an
umbrella term for a variety of different
effect types, wed better have a go at
explaining the main ones cm

CHORUS

PHASING AND FLANGING

RING MODULATION

VIBRATO/TREMOLO/PANNER

Legend has it that chorus was invented

by Roland in their Jazz Chorus amplifier.
The amp sent a dry signal to one amp
and speaker, and the same signal was
delayed, modulated and sent to another
amplifier and speaker within the same
cabinet. This is a good explanation of
how chorus works except that instead
of feeding two amps, in a modern plug-in
the two signals are combined. The
second, modulated signal is not only
delayed but also pitchshifted slightly, and
an LFO controls the modulation of this
pitch and thus the character of the
effect. Originally intended to simulate
the sound of more than one instrument
or vocal part, it soon became apparent
that it didnt, and
now chorus is
simply used as a
creative effect.

The terms phasing and flanging have

become confusing over the years, largely
because the effect we now call flanging
was originally called phasing and the
effect we now call phasing is rather
similar to flanging. The concept of
flanging stems from running two synced
tape machines (playing the same signal)
and manually slowing one with your
finger (and then the other). This created a
distinctive whooshing sound. Phasing, on
the other hand, is similar, but without the
fast changes and whooshes. In both
cases, as the two signals move closer
(phasing) and further (flanging) apart,
some interesting frequency cancellations
occur as similar or identical frequencies
cancel each other out or build on each
other. Dynamic variation to these effects
can be produced with an LFO controlling
the timing and intensity of the
modulation, and in the case of flanging,
the effect is intensified by feeding some
of the wet signal back into the equation.

Ring modulators are tricky to understand,

but very simple to use, fortunately. The
basic concept is that two signals are
brought together and then the sum and
difference of the frequencies of both are
generated, while the rest of the signal is
removed. So if you had a 500Hz sine
wave and an 800Hz sine wave, the
output would feature tones at both
1300Hz and 300Hz. Of course, real
sounds are much more complicated
than that, and it would make
your head explode to figure
out on paper what this
actually produces when two
complex signals are
combined. To summarise the
effect, it can sound an awful
lot like the Daleks talking.

If you reduce the delay time in

modulation effects down to 10ms or less,
it's difficult to distinguish between the
original and delayed signal, and so you
can then get on with the business of
creating tremolo and vibrato. Variations
on a theme, the basic difference is that
the latter involves modulating the pitch of
a sound, the former the volume. Things
get a bit more interesting when you offset
the left and right channels of a tremolo
so that as one peaks
the others at its
lowest point, thus
generating either
synced or LFOdriven panning, the
use of which
should be selfexplanatory!

Modulation types

7 Chorus sounds
particularly good on
pads and guitar riffs
048

7 Tremolo is an
endlessly useful
volume-related effect

| COMPUTER MUSIC FEBRUARY 2006 TUTORIAL

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Modulation FX masterclass make music

Q&A focus
now

THE ANATOMY OF MODULATION EFFECTS

Discover the inner workings of some popular modulation effects. The ones
shown are from Cubase, but your software will probably have equivalents
Chorus

MIX
Controls the balance between
the wet and dry signals
DELAY
Sets the delay time of
the repetition(s)

SHAPE
Enables you to choose the
modulating waveform type

STAGES
This adds additional voices
(up to three more), for an
ultra-thick sound

FREQUENCY
This sets the speed of the
modulation effect

Flanger
TEMPO SYNC
Does what it says on the very
handy tin
DEPTH
Sets the depth of the
modulation sweeps

FEEDBACK
Greater feedback settings produce
a distinctive metallic effect

STEREO BASIS
Offers a choice between mono, normal
and enhanced stereo operation

RATE
In sync mode, this determines the note value for
the modulation; otherwise you can set it freely

Ring modulator
OSCILLATOR ENVELOPE AMOUNT
Controls the effect of the envelope
(generated by the incoming signal) on the
oscillator frequency
OSCILLATOR LFO AMOUNT
Controls the effect of the LFO on
the oscillator frequency
OSCILLATOR RANGE
Sets the oscillators
frequency range

OSCILLATOR FREQUENCY
Controls the oscillator frequency to
between plus or minus two octaves,
within the selected range

INVERT STEREO
Inverts the LFO for the right-hand
channel, creating a stereo effect

OSCILLATOR WAVE
Determines the oscillator wave
type (square, sine, etc)

ROLL-OFF
A low pass filter for smoothing
the oscillator waveform

TUTORIAL FEBRUARY 2006 COMPUTER MUSIC

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| 049

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make music now Modulation FX masterclass

Special effects
T
he thing about modulation effects
is that they have almost no
parallels in nature. In fact, the
closest approximation to flanging in the
real world is a low-flying jet aeroplane
skimming a concrete jungle (so much
so that almost every single explanation
of flanging youll ever read uses this
example), and thats about as close to
natural as any of these effects really

come! Consequently, they fall into the

category of effects that we like to call
special. Why? Because, with a very few
exceptions, any time you use them they
will be very apparent and are nearly
always meant to be noticed (there are
very few reasons you might add them to
impart realism and subtlety). As with any
such effect, they can become tiresome
very quickly, so restraint is the name of

the game. We wouldnt recommend you

try all of the forthcoming suggestions
and techniques in one track, but dont
let that put you off trying them all at
some point and coming up with some
of your own. With that in mind, weve
put together a selection of tips to get
you thinking about some creative (or
useful) applications for these potentially
extreme effects

Use them on
There are no hard and fast rules as to what you can and cant use modulation effects on, but
here are some suggestions based on the most popular uses.
CHORUS

PHASING AND FLANGING

RING MODULATION

VIBRATO, TREMOLO

Backing vocals, pads, backing

guitars, strings

Guitars, lead synths, drum

loops, edits

Special vocal effects, drum

loops, edits

Pads, vocals,
percussion, loops

Modulation tips

1Some chorus effects offer multiple delays for an

even thicker sound

MODULATION EFFECTS WILL almost

always take away from the sense of
presence of a sound at the same time
as they give it size and space, so be
careful what you use them on. Used on
lead elements theyll detract from
energy and prominence in most mixes.

TRY PLAYING WITH the modulation

rate on your phaser. Slow speeds create
nice long sweeps, while much faster
speeds can have a great old-school
rotary cabinet effect. You can even
alternate between the two, or have one
panned left and the other right

MODULATION EFFECTS ALL rely on a

very critical wet/dry balance, so its
generally more convenient to use them
as insert effects rather than as part of a
send and receive effects loop. They
should also usually come near the end
of an effects chain (ideally, penultimate).

TO COUNTER THE tendency of chorus

to push sounds into the background, try
applying it only to the reverb send signal
or insert it into the reverb bus itself.
This gives some lush thickness to the
sound, but will allow you to keep the
main sound upfront.

IF YOURE AFTER a classic ensemble

effect, take a leaf from the synth
programmers book and use a
pitchshifter in detune mode to generate
another version of your signal, slightly
detuned from the original. This avoids
the distinctive cycling of the LFO.

A GREAT WAY to get more useable and

musical results from the unpredictable
and usually severe sounding ring
modulator is to mix in some (or loads!)
of the unprocessed main signal. Taken
to the opposite extreme, the ring
modulator output can be used as
nothing more than a backing layer.

IF YOUR PHASER or flanger enables

you to sync its modulation action to
track tempo, be sure to do so, as you
can set up some nice cyclic patterns
(over eight bars, for example), which
make the effect much more useable
and easy to manage when mixing
your track.

THE SOUND OF the electric guitar is,

quite simply, the sound of the effects
being used on it, and some of the most
enduringly popular are chorus, phasing
and flanging. So if youre ever working
with an electric guitar riff, youd be crazy
not to at least try one or all of them.

3If your modulation

effects offer tempo
sync, use it!

050

PERCUSSION LOOPS AND individual

drum hits can sound fantastic when run
through ring modulators. Try combining
snares, hats and cymbals for some very
electronic and bizarre techno
percussion sounds

3Try a straight
pitchshifter to avoid
obvious LFO cycling

| COMPUTER MUSIC FEBRUARY 2006 TUTORIAL

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Modulation FX masterclass make music

Q&A focus
now

MOST MODULATION EFFECTS

(particularly phasers) will create
significant fluctuations in the signal level
of any channel to which theyre applied.
To counter this, add a compressor
immediately after the inserted
modulation effect, thus flattening the
level slightly.
RING MODULATION EFFECTS
REQUIRE simultaneous signals to
generate any noise whatsoever, so if
your ring modulator accepts two
external signals you might want to add
a compressor to each, just to be sure of
constant sound output.
PHASING CAN BE just the ticket for
taking the edge off harsh sounds and
frequencies. Try some subtle phasing on
your lead synths and guitars it can
soften them up and help them fit into a
mix. Its also great for softening
distortion effects.

1Lives phaser puts all the controls you need in

one very clear interface

is disabled. If yours does, try narrowing

the range to around 300Hz-5KHz, as
this will offer the greatest audible effect
without significant volume fluctuation.
ALMOST ANY MODULATION plug-in
can be used as a stereo width effect,
simply by offsetting the modulation of
the left and right channels. As one peak
is reached, the other side is in the
trough just as with tremolo.
DONT FORGET TO automate your
controls. By their nature, modulation
effects are very in-your-face, so listeners
can get tired of them pretty quickly. Turn
this to your advantage and only use
them at certain times or at least
lessen their effect in parts.
IF YOUR FLANGING is more like a
windy fart than an Airbus test-flight, try
whacking the feedback up nice and
high this will intensify the whooshing

SOME PHASERS ENABLE you to set a

frequency floor and ceiling, above and
below which the phasing of frequencies

1When it comes to bundled plug-ins, Logic leads

the pack, and its Phaser is quite excellent

Start by creating one audio channel and one auxiliary channel in

your sequencer. Set the tempo to 128bpm, add the audio file
Synth Riff @ 128bpm (on the DVD) to the audio channel and set
the project to loop over two bars. >>

1For some serious flanging action, make sure you

experiment with the feedback control

IF YOUR FLANGING IS MORE LIKE A

WINDY FART THAN AN AIRBUS TEST
FLIGHT, WHACK UP THE FEEDBACK
FOR CLASSIC CHORUS, flanging and
phasing effects, the best initial wet/dry
balance is 50/50. Using this as your
starting point, try shifting the balance,
and youll find that you usually end up
back on the halfway line

STEP BY STEP Using chorus

to create stereo width

Now pan the audio channel completely left, send the output to
the auxiliary channel and pan the auxiliary channel all the way to
the right. Balance the levels of the left and right channels now,
using the send control on the original channel and the volume on
the auxiliary. >>

effect that we all know and love (blame

Van Halen for that one )
IF YOUR PHASER enables you to set
the delay time manually, you can use
this to shape the sound. Shorter delay
times make for more high frequency
action and a thinner sound. Lengthen
them and the sound is fatter and more
focused on the lower frequencies.
YOU CAN USE chorus and other
modulation effects to generate some
fantastic stereo effects by panning the
dry signal to one side and the
effected signal to the other side. The
resulting effect can be used to give
width to a sound, to help it blend into a
mix, or to add dynamic excitement. To
find out how, follow the walkthrough on
the right cm

Finally, insert the chorus plug-in of your choice into the auxiliary
channel (if it has a wet/dry control, set it to fully wet). You may
need to tweak the left and right balance again. Also, if the effect
is too obvious, try some less extreme panning.

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ADVANCED

EFFECTS

PART 5

Gating
You may think you know all there is to know about gating,
but theres more to this effect than meets the ear

On the DVD
TUTORIAL FILES
The loops used in the
walkthroughs can be found
in the Tutorial files folder

5 Using a gate gives you more flexibility than a

noise reduction plug-in such as this

060

o far in this series weve focused

on processing effects that you
know you may not fully
understand, but that you use regularly
nonetheless. This month were doing
the opposite were covering a type of
processor you probably think you fully
understand, but that you dont often use
very much. Thats right, were talking
about the humble gate.
Like compressors, gates fall into the
category of dynamics processors, as
theyre triggered by the strength of a
signal and are used to affect that
signals volume. At its most basic level,
a gate reads an incoming signal and,
unless that signal exceeds a certain
volume, stops it from passing through. In
the dark ages of analogue, when just
about every stage of the recording
process added hum and hiss, and
everything was recorded to either
analogue or digital tape, the only real
way to stop all this noise drowning out
the sound of your music was to use a
gate. This would prevent unwanted
noise when no notes or other desired
sounds were playing, reducing the
cumulative noise level. And because
you cant normally hear hiss and hum
when there are other sounds going on,
this was usually enough to get rid of all
the perceived noise.
OK, we know what youre thinking
why should you care about any of this?

The near-universal adoption of DAWs

means that any analogue interference is
a thing of the past, and when you do
need to get rid of unwanted background
noise, there are any number of
intelligent noise-removal plug-ins that
can learn the sound of said noise and
remove it from the signal. Even if you
dont have one of those, computerbased editing enables you to silence all
of the sections in between the words,
notes or other sounds that you actually
want on your tracks.
Well, let us just set you straight right
now theres a whole world of gating
possibility out there still, both creative
and practical. First, the practical

Gateway to heaven

Yes, its true that digital editing enables

you to strip out unwanted audio (such
as hi-hat spill in your snare drum
channel), but this isn't the simplest
process in the world. Some sequencers
will do it for you, but you still have to set
the threshold, just as you would with a
gate. And as this is an offline,
destructive process it can be a bit hit
and miss, and often impossible to undo.
So why not just use a gate? You can
even employ more sophisticated gates,
which give you a greater degree of
control and enable you to reduce the
amount of reduction rather than just
having the gate entirely open or closed.

They also allow for more creative

envelope shaping, all of which makes
for subtle and organic noise reduction.
Because an open gate spits out the
same signal it receives, theres no risk
of damaging the desired sound, which
can happen with noise-reduction plugins. So,
not only
can gates
(whether
digital or
analogue)
be quicker
to set up
than many
of the
more
1 We love logic's Noise Gate plug-in
advanced
DSP processes, but they can actually do
a better (and safer) job.
And then theres the creative aspect.
Just as compressors can be used to
enhance different signals and solve
various problems, so too can gates be
used to alter the characteristics of a
sound and even to selectively play
patterns by utilising a sidechain input.
But before we have a look at these
techniques, we first need to familiarise
ourselves with the front-panel GUI of a
modern gate plug-in, and what better
than Logics trusty Noise Gate, which
offers up as many parameters as youre
ever likely to find on one cm

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Q&A focus
now

THE ANATOMY OF A GATE

01

THRESHOLD It helps if you

think of the noise gate as
being naturally closed (on a
metaphorical spring). It takes a certain
amount of noise pressure to push the
gate open, and the threshold is where
you set the springs strength. Set it too
low (ie, a low level of sound) and you
risk letting all of the sound push
through, but set it too high and you
could lose some of the sound, as the
some of its not strong enough to get
through. The result will be snippets of
sound and loud clicks and pops as the
gate repeatedly opens and slams shut.

02

REDUCTION This is where

you set the amount of gain
reduction applied to any
signal that falls below the threshold (or
how closed the gate really is, if you
prefer our metaphor). When set to full,
the reduction ensures total silence
whenever the signal drops below the
threshold. You might think full reduction
sounds just the ticket, but in most
practical cases its a little too extreme
and results in a very noticeable and
unnatural opening and closing sound
(although it can be useful for creating
effects). Another useful side-effect of
having, for example, only -30dB
reduction rather than the full cut, is that
the reaction time of the gate will be
faster particularly on analogue or
realistic analogue modelling gates. Why?
Well, quite simply, it takes time for the

gate to open, and the further it has to

open, the longer it will take, so
sometimes you can actually achieve
punchier attacks when going from a
quiet signal to a loud one than from
silence to loud.

03

HYSTERESIS When
signals hover around the
threshold you can
sometimes experience chattering (little
snippets of sound slipping through as
the gate opens and closes very quickly),
and this is particularly true with very fast
attack and release times. Hysteresis
is a process whereby the threshold for
the gate closing again is a few dBs
lower than the threshold for the gate
opening, so that once open, the gate
gives fluctuating signals the benefit of
the doubt, thus vastly reducing the
chattering effect.

04

LOOK-AHEAD The
look-ahead function on
dynamic plug-ins is one of
the coolest things about working with
computers, as it enables the plug-in to
have a little look at whats coming next
and react just before it happens
incredibly handy for preserving sensitive
transients. With this control, you can set
how far ahead the plug-in will look.

05

ATTACK As with
compressors, the attack
defines the speed at which

the gate will open. Depending on what

you hope to achieve, you can have it
vary fast for things like punchy drums, or
slower for instruments with slower
attack envelopes, such as bowed
instruments or even voices, where
abrupt bursts of sound would sound
incongruous and unnatural.

06

HOLD This is another

solution for unwanted
chattering. As you might
have guessed, the hold function is the
amount of time that elapses after the
signal drops beneath the threshold
before the gate closes.

07

RELEASE The release

control determines the
speed at which the gate will
close after the signal drops below the
threshold and the hold period has
elapsed. As with compression, its very
important that you set the attack and
release correctly for the purpose in
hand. For transparent practical uses you
want settings that complement the
attack and release portion of your
signals basic envelope, while more
extreme settings are useful for the
creative side of things.

08

SIDECHAIN As with all

sidechains, this enables
you to use a signal other
than the original source to trigger the
effect of the gate, but it also allows you

to use an adapted version of the original

source signal, while the untreated
version is actually processed by the
gate. In essence, the incoming signal is
split into two identical versions one
which will be processed, and another
that will trigger the processing, and
which can have other things done to it
first. This leads us to the high- and lowcut filters

09

HIGH-CUT If youre using

the gate on a snare drum
mic, for example, there
might well be enough noise-spill from
the hi-hats to open the gate at
unwanted times, or at least make
setting the threshold and envelope
rather tricky. But by rolling off some of
the top-end of the sidechain signal, you
can alleviate this problem whilst leaving
the actual signal being processed free
from EQing.

10

LOW-CUT This employs

exactly the same principle
as the high-cut function, but
is useful for rolling off the bass to stop
things like kick drum spill accidentally
opening up the gate.

11

MONITOR The monitor

function routes the sidechain
signal to the audio outputs
so you can, for example, actually hear
all of the high- and low-cut tweaks that
youre making.

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Putting gates into action

STEP BY STEP Giving it the chop

One of the coolest gating tricks is rhythmically chopping

a signal. Play some long power chords on a full-fat synth
patch, holding each chord for two beats or more. If you
arent musical, just start with C, E, G and move your fingers to
the white notes above and/or below each time. >>

Insert a gate with a sidechain input into the channel

youve just created (set it to full gain reduction). Now
create a new channel offering a loud and constant
synth patch with instant attack and release, then mute the
output and send the signal to your gates sidechain input. >>

Use the second sound to create a rhythmic pattern

the gate will open each time you press a key and stay
open for as long as you hold it. When used on full
tracks and instruments with normally smooth envelopes
(violins, voices, etc), this is creates a great, unrealistic effect.

STEP BY STEP Kicking it

Start by importing the two audio files Hip-kick loop and

Hop-hat loop into your sequencer on two different
audio channels. The tempo for them is 102, and you
should get them to loop over the first bar. Now play them
back together, so you get a feel for the sound of the kick. >>

Now create a new instrument channel and load up a

soft synth that offers pure sine waves (were using
Logics EXS24 default No instrument setting). Draw in a
long note covering the whole bar (try somewhere around C1
or lower), and set the Attack to zero on your sine wave. >>

Insert a gate on your sine wave channel and send the

kick drum signal to the gates sidechain. Set any Attack
and Hold controls to zero and hit play. Your bass note
should now be reinforcing your kick. Tweak your gates
Release to get rid of any clicking at the end of drum hits.

STEP BY STEP Roughing things up

Tips on adding reverb to drums

Depending on the gate youre using, you can get some

quite useable and musical distortion-style effects from
the normally undesirable chattering phenomenon
caused by audio levels hovering around the threshold point.
Start by loading the loop Talk talk all day long. >>

062

Now insert a gate that enables you to alter attack,

release and hold times. Were using the one in Ableton
Live, set to Flip mode (this reverses the effect of the
threshold so that signals below it pass through while signals
above it are blocked). >>

Set the Attack and Release times to quite fast and

make sure that the gate is closing fully. Move the
Threshold level until you hear the chattering effect.
Now, by slowly adjusting the Hold time, you can tweak the
chattering until it sounds more like distortion nice!

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Gating tips

Apply a gate to an unruly

bassline and feed your kick
drum channel into the gates
sidechain input. This will
ensure your bass notes and kicks
trigger simultaneously for a punchy,
tight, clubby sound. And with slower
attack times, you can delay the sound
of the bass, for a realistic live sound
(people generally play a little behind
the drums).

The signal path of your effects

is vital, and gates should
normally go upfront. For
example, if you place EQ
before a gate, any subsequent EQ
changes will alter the level and disrupt
the gates activation. And as a
compressor dynamically alters the
signal, it's bound to complicate the
setup of your gate settings.

Despite what we've just said,

have a go at applying a gate
after any heavily
delayed sound with a fast
attack and use it to accentuate each of
the individual hits. With the gain
reduction setting fixed higher than
usual, you can create a really cool
volume pumping effect each time a
delay sounds. Just use the attack and
release controls to hone the pumping,
and adapt to taste

Just as with compressors, if

youre gating a stereo signal,
be sure that stereo link is
switched on. This ensures
that any dynamic changes applied to
one side of the stereo split will be
applied to the other vital if your
source material has a left or right bias.

Gating can have an extremely

strong (and useful) effect on a
signal, but also a destructive
one, so if you do think you
need to apply it to a multitrack
recording session (to eliminate
unwanted hiss or hum, for example) its
usually better to get the recording down
dry first and then apply gating at the
mixing stage, where the effects can
always be discarded.

You neednt just sidechain

elements of songs why not
whole tracks? Sure, you can
always chop up sections of a
track, but nothing beats bashing out a
pattern on your MIDI keyboard and have
it open the gate applied to your whole
track. Just ask Mylo!

You can fatten up kicks by

adding a sine wave or
booming 808 kick sound, but
the envelopes are usually just
messy, so use a gate on each
additional sine wave or low kick sample
to be layered, and trigger their
sidechains with the original kick. Just
make sure you have fast attack times in
place so that the kick layers punch in
nice and quickly.

For an authentic 80s gated

reverb sound, record some
drums in a concrete stairwell
or put them through a phat
reverb patch. Then apply a gate with a
high threshold and fast attack and
release times. You can then extend the
hold parameter, allowing through a
burst of big sound.

7Gates can be very powerful and destructive, so

place them first in the signal path

If youre layering a bunch of performances,

bring them together and add punch by
applying a gate to each, and then supplying their
sidechains with the audio from the best take.

10

You can create a more punchy sound on some percussion

and bass notes by placing the threshold slightly higher than
it need be to remove noise and setting your attack time as
fast as it goes. The gate will now not open until the high
threshold is reached, and when it is, the sound will explode through the gate.

1Begin by using fast attack and release settings, then tweak the threshold

3Even this basic MDA

gate can add punch
to your percussion

11

Start out with fast

attack and release
settings so that its
clear what's
happening to the signal. Then adjust the

threshold until you find the approximate

level ie, the point at which the sound
you want comes through, but the sound
you dont doesnt. From there you can
fine tune the other settings.

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make music now EQ FX masterclass

ADVANCED

EFFECTS

PART 6

Equalisation
In this, the last of our Advanced Effects tutorials, we
take a look at the king of processors EQ

o far during the course of this

in-depth FX series weve looked
at a variety of wild and wonderful
processing techniques, designed to
shape, shift and generally mangle or
spangle your sounds. And for this, the
last in the series, we've been saving the
best, the sexiest and the sassiest
thats right, its EQ!
Now, before were deafened by a
chorus of disparaging coughs and sniffs,
let us assure you that EQ really is the
daddy of effects. More than any other

process, it can make or break your mix,

even if only by omission!
The key to great production is simple
take a good element and make it as
good as it can be. This means choosing
the right sounds for the right job. If you
try to force a sound, it just wont sound
nice. A great mix is about a great
balance of great elements. If you
compromise this philosophy at any
stage it will show, believe us. But
because of its overt immediacy and
power, EQ is usually the first recourse of

7Old analogue
desks often only
had one mid-band
with a frequency
sweep, like this one
from GarageBand

066

the lazy producer trying to force a

sound. Why? Simple if you boost EQ,
you boost the level of a signal, and
louder almost invariably sounds
better at first. Whack the bass up to
11 it sounds phat. Turn up the treble
it stands out a mile. And we humans
are a fickle bunch.
Its for all these reasons that some
engineers will tell you that you should
never accentuate frequencies with EQ,
only attenuate (reduce) them. But
engineers are superstitious and prone
to routine and habit, so its best to add
at least a grain of salt to every wisdom
nugget they pass your way. Before we
go any further, though, lets have a brief
recap on what EQ is all about.
EQ is essentially a cross between a
filter and a level control. It affects only
specific frequencies and is used to
either boost or reduce the level of said
frequencies. In times gone by, most
budget EQ was very simple, and limited
to three bands at least two of which
were fixed-range. These days we have
the facility to apply highly precise EQ to
every channel, which makes it even
more critical that we understand what
were doing (if this were a martial arts
tutorial wed remind you that, With great
power comes great responsibility), so
later on in this tutorial well be looking
at the various different types of EQ you
might encounter. cm

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Q&A focus
now

THE ANATOMY OF AN EQ PLUG-IN

01
02

EQ GRAPH A visual representation of

the frequencies affected by EQ
SHELVING These EQs affect only
frequencies above or below the
cutoff frequency

03
04

PARAMETRIC/BAND-PASS This EQ
type only affects frequencies within the
specified range
FREQUENCY This sets either
the cutoff or centre frequency to
be affected

05
06

GAIN CONTROLS The boost or gain

applied to either the shelving or
parametric EQ

Q This specifies the width (parametric)

or angle (shelving) of frequencies
affected either side of the
centre/cutoff frequency

Parametric EQ
Its strange how quickly we humans
adapt and forget the way things were,
which is why theres not a computer
musician who could function without
fully parametric EQ, despite the fact that
most of us had never even used one
prior to 1999.
So what is it? Well, a fully parametric
EQ strip will have three controls. The
first is Frequency. All EQ is based
around a curve, centred on a certain
frequency, and the Frequency control
determines this frequency. The next
control is Gain (well, its probably Q,
actually, but well get to that in a
minute). The Gain control specifies how
much the aforementioned frequency is

made louder or quieter. So if you set

the Frequency control to 3kHz and
apply 3dB of gain, then any
frequencies at 3kHz will be boosted by
exactly 3dB.
The next control and heres where
it gets interesting is Q, and the Q
control determines the amount of gain
or attenuation applied to either side (in
case youre interested, industry folklore
has it that Q stands for quality). To
understand the Q control, imagine a
strip of rubber, pulled straight and
attached at either end, with a row of
pen marks at regular intervals (Blue
Peter fans can actually try this out using
a rubber band, a biro, two nails and a

pair of scissors). Each of the

marks represents a specific
frequency, and if you push any
part of the rubber band up it
represents gain to those
frequencies, whilst pushing it
down represents attenuation. The
first thing you notice is that all of
the notches either side of the
central notch are pushed up or
down too, and this is what the Q control
is used for shaping the curve either
side of the centre frequency.
The Q is assigned a value, and this
is derived from dividing the centre
frequency value by the bandwidth of the
affected signal. Of course, the frequency

1Cubase's built-in parametric EQ is a

good model to get started with

width is hard to define, so it's taken to

mean the point on either side of the
curve where the signal level drops by
3dB. High values equate to narrow
frequency ranges, for precise cutting or
boosting, while low values equal wide
curves, for broad boost or cut.
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make music now EQ FX masterclass

GRAPHIC EQ
The graphic equaliser probably seems
like an historical curiosity to most
producers under the age of 25. The
only place you seem to see them
these days is on your iTunes controller
or parents (or grandparents) living
room hi-fi. But in the days before
DAWs, the pro-quality graphic EQ was
a highly valued and much sought-after
final link in the mixing chain.
To understand its primary purpose,
you first need to picture your mix in
terms of frequencies. If you have
access to a spectrum analyser, run
one of your finished tracks through it
now. Then, run one of your favourite
mixes through. See the difference? In
case you dont have a spectrum
analyser, check out the screengrab at
the bottom right of this boxout. See
the curve and shape? Thats a
graphical representation of a mix, and
though you should always trust your
ears first, its handy to get an objective
look, especially if your ears arent very
experienced yet.
A good mix should have a
reasonably smooth frequency
response (though not necessarily flat
see boxout on p69). There might be a
bias to the bass and/or treble,

Shelving and
other sonic DIY
Weve already established that EQ
is essentially a type of filtering, and
this is why on a number of EQ
plug-ins youll see similar controls
as you would on a variable-type
filter. Most notably, these have to do
with shelving, cut and band-pass
settings. The first of these is found
at the top- or bottom-end of the
frequency range, and is used for
applying broad boost to all the
frequencies above (if its at the treble
end) or below (at the bass end),
ideally with the ability to select the
cutoff frequency. The second is like a
basic low- or high-cut filter, cutting
off everything below a certain point
(although be careful, as some, such
as that in Cubase SX 3, actually have
a small slope, rather than an exact
cut). Finally, the band-pass settings
are those, such as with fully
parametric EQ, where you can sweep
the affected frequency range up or
down, affecting only those
frequencies within the range. All of
these types can be useful for
different purposes the trick is
thinking logically about which one is
best for the job.

068

depending on your genre, but

generally speaking, you want to avoid
too many narrow peaks.
And thats what a graphic EQ is for
shaping your mix subtly. And it really
does work. You should be aiming for a
very decent EQ though 30-band
used to be the industry standard.
Essentially, this means that the entire
audible frequency range (roughly 20Hz
to 20kHz) has been divided up into
useful frequency segments, and with a
30-band EQ each band is spaced
roughly a third of an octave apart from
the next. Of course, the nature of
graphic EQ is that the preset Q curve
of each band will also affect any
surrounding bands (the effect of one
slider will usually be felt on the range
of the two or three above and below
as well), so having boosted or cut one,
youll need to be more subtle with
those on either side, otherwise the
effect on the crossover frequencies
between them will be exaggerated.
It goes without saying that the
benefits of graphic EQ are limited
largely to mix shaping, and any
corrective or creative EQ applied to
specific sounds will benefit from the
precision of a parametric EQ instead.

7Graphic equalisers are

often relegated to consumer
hi-fi equipment and software
like iTunes

3This is the modern equivalent

of the graphic EQ a pro
quality 10-band parametric EQ
plug-in for mastering

7Spectral analysis lets you see

exactly what's going on with
your mix

Its just a phase

One of the interesting things about EQ
units and plug-ins is how different they
can sound, despite the fact that they all
have the same theoretical effect. And
one of the main causes of this is now
thought to be due to the phase
relationship of different frequencies.
Because frequencies travel at varying
speeds through analogue circuitry (even
if its only that in your speaker), the
exact relationship between the time
that high and low frequencies hit our
ears can be affected, making the
sounds appear duller. Indeed, many
enhancer plug-ins rely on this very
principle, and offset higher or lower
frequencies slightly, making the sound
ultimately brighter. By the same token,
different EQs can affect the phase
relationship in different ways, which
probably explains why some EQs can
make sounds shine through with
minimal boost, while others require
much more radical gain settings and
yet still dont offer the same clarity and
punch. For this reason its important to
experiment with as many EQs as you
can, and then settle on the one that
gives the best results and requires
lower settings.

1Some enhancer plug-ins offset the high and low frequencies

to compensate for the phase distortion in your speakers

1These two Logic EQs might look identical, but one of them prevents any phase distortion, for a
clearer and more natural sound (albeit at the cost of CPU cycles and latency!)

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EQ FX masterclass make music

Q&A focus
now

EQ tips

7Human speech is centered

around 1-3kHz, which is why
phones only reproduce
frequencies from 300Hz to
3kHz, as shown here

IF YOU FIND YOURSELF

applying excessive EQ on a
regular basis, there are two
possibilities youre picking
the wrong sounds to begin with or
youre the next big thing. If you think its
the latter, wait a year, and if you havent
had three acclaimed singles and a hit
album then refer to the first point.

ANY FREQUENCIES BELOW

30Hz are usually inaudible
sonic pollutant, soaking up
precious headroom and
generally muddying up your mix. Its
therefore always a good idea to cut
everything below around 30Hz
this way your mix will be louder and
your bass will sound much punchier
and heavier.

1Applying boost to signals will usually

only end up cluttering your mix

seem inherently less bright. You can use

this effect and EQ your mixes to give
certain elements a greater sense of
distance by applying high-shelving cut.

EQ BOOST AND CUT both

affect the volume of a signal,
and obviously the greater the
cut or boost, the more the
volume is affected, so whenever you

JUDICIOUS USE OF HIGH AND LOW

SHELVING OR CUT CAN HELP GIVE
EVERYTHING THE SPACE IT NEEDS

THE KEY TO GOOD

parametric EQing is finding
the right frequency, and the
easiest way to do that is to
apply excessive gain and set your Q to
a reasonably narrow frequency. Now,
when you sweep the frequency range,
youll be able to hear when you hit the
target frequency.

AIR SOAKS UP high

frequencies more than low
frequencies, and low
frequencies travel slightly
faster than high ones, so distant sounds

apply EQ, be prepared to tweak your

level settings too.

IF YOURE USING mastering

EQ (either graphic or
parametric) on a complete
mix, then you should almost
always place it before any limiting youre
applying, as otherwise youll risk losing
some volume or pushing the signal
into clipping.

ANALYSE ALL OF YOUR

main sounds (such as vocals
and guitars) together and in
isolation and see if theyre

fighting for the same space in the mix.

For example, vocals tend to be centred
around 1-3kHz, so try a little cut in that
range on your guitars and hear the two
separate beautifully.

ONE OF THE BEST WAYS to

get parts such as backing
vocals or layered guitar lines
to gel together is to run
them off to their own bus and apply any
EQ tweaks to the bus, rather than the
original parts.

MOST SOUNDS ARE

CENTRED around a certain
part of the frequency
spectrum but also have stuff
going on in other sections everything
else is a potential source of sonic
pollution. Judicious use of high and low
shelving or cut can help give everything
the space it needs.

11

EAR FATIGUE is
much more of a
problem with high
frequencies, so
always apply treble boost in small steps,
and when A/B-ing the sound with and
without treble, give yourself at least a
minute to readjust to the dry version
and allow objective assessment.

12

SPEND SOME TIME

learning the different
frequency ranges for
common instruments.
Use your parametric EQ with plenty of
boost and a medium Q setting and
sweep the frequency range up and
down while you listen to some tracks.
Make notes about which frequency
range each instrument is centred on.

10

GENERALLY
SPEAKING, human
hearing is less
attuned to EQ cut
than it is to EQ boost, so if you want to
achieve a more natural sound from
your processing, try to avoid any
excessive EQ boost. Instead,
make sure you opt for cuts in
competing frequencies.

1A spectrum analyser can help you learn the

frequency ranges of different instruments

Keeping it flat

7Applying heavy EQ gain

will make the signal
significantly louder, so
reduce the channel
volume afterwards

While we might all scrimp and save to

buy speakers and amps boasting
perfectly flat frequency responses,
without any bias to top, middle or
bottom, the truth is that our hearing
response is anything but flat. And the
plot thickens as we reduce or raise the
volume of what were listening to. For
example, at low levels our hearing is
more attuned to middle frequencies,
but as we raise volume, the mid-range
response falls off, while the upper and
lower frequencies become more

noticeable, resulting in the smile curve

on a frequency response graph of our
hearing. We can use this fact to make
our mixes sound apparently louder,
simply by emulating this type of curve
in our mixdowns, and two of the best
tools for this job are the graphic EQ
and spectrum analyser. Of course, the
extent to which you do this depends
largely on the type of music youre
making. Electronic styles can obviously
benefit more from the smile technique
than classical recordings can!

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