Creative Effects: The Definitive Guide For Producers

By Eddie Bazil
Ebook Edition
Published 2011 by Samplecraze, [Link]
Copyright Eddie Bazil 2011
Eddie Bazil has asserted his right under the Copyright, Designs and Patents Act
1988 to be identified as the Author of this work.
All rights reserved. No part of this book nor any of its associated tutorial files may be
reproduced, resold, or transmitted in any form or by any means without prior written
permission of the Publisher.
Notice of Liability
The author and publisher have made every effort to ensure the accuracy of the
information herein. However, the information contained in this book is sold without
warranty, either express or implied. Neither the author nor Publishers (Samplecraze),
nor its dealers or distributors will be held liable for any damages to be caused either
directly or indirectly by the instructions contained in this book, or by the software or
hardware products described herein.

Table of Contents

INTRODUCTION

CHAPTER 1 REVERB

CHAPTER 2 MODULATORS

47

CHAPTER 3 DELAY

59

CHAPTER 4 CHORUS

82

CHAPTER 5 - FLANGING AND PHASING

89

CHAPTER 6 DISTORTION

97

CHAPTER 7 FILTERS

126

CHAPTER 8 - PITCH MANIPULATION

154

CHAPTER 9 - CREATIVE EFFECTS

168

CHAPTER 10 - FINAL WORD

210

Introduction
I have never been good at writing songs. I cant string more than two words
together. I have always envied songwriters and their ability to create poetry
through music. I knew from an early age that I would never be a songwriter.
BUT, I had an obsessive fascination with sound and, to my surprise; I found that
I understood sound and its physics much better than words and their
relationships to each other. The motion and effect of a sound was as moving for
me as a whole song. The ability to take a single sound and treat it so it moved
dynamically, much as a song does, and to have it evoke an emotion was exciting
for me.
I decided, from an early age, that I would explore sound and try to acquire the
skills to shape it. This led to countless sleepless nights studying sound and its
physical properties, trying to find a happy medium between data and creativity.
I decided to program as many synthesizers as I could for other people just to
get the experience of manipulating sound. It soon dawned on me that there are
three types of sound design areas: replicating, colouring and warping.
Replicating involves replicating an existing sound like programming a horn sound
on a synthesizer or sampling and designing a piano preset.
Colouring involves using a replicated sound and creating variations of it but in a
more creative manner thus giving rise to a new version of the same sound that
still falls within the parameters of a replicated sound but with a new twist in
representing it. Warping involves creating a completely new sound that doesnt
fall under the replication criteria but can use the replicated sound as a source.
This involves total reshaping into a new texture and one that evokes a specific
emotion. Whereas the replicated sound is about recreating an existing sound,
warping is about twisting it into a new and detached sound.
All forms of sound design start at the waveform stage whether sample based or
pure. But reshaping, colouring, mangling, or warping an existing sound involves
the use of dynamics and effects and how the modulation matrix can best make
use of existing synthetic design tools. Of course, we can mangle a sound within
the modulation matrix using the basic tools that come with the synthesizer but
that can be limited and too specific. It is the area of effects that opens up the
world of sonic mangling and if the effects can then be modulated and routed
within a matrix then a whole new world of sound design opens up and you are
limited only by your imagination. It doesnt end there. Sonic mangling is one
thing but creative production is another ballgame entirely and this is heavily
reliant on effects and dynamics not just for corrective tasks but for creative ones
as well.
An integral part of mixing and production is the area devoted to effects.

Effects are used not only to colour sound, but to create an aural illusion. How
this is achieved is dependent on the effect being used. But before we can delve
into the wondrous world of effects we need to define what an effect is.
What is an effect?
An effect is a process or device that adds to an untreated/dry signal by a user
defined amount, whereas a signal processor treats the whole signal and does
not add to it. In the old days of patching analogue mixers the auxiliaries were
used for effects like reverb etc and the inserts were used for processors such as
compression. The distinction, in terms of processing, is quite obvious. The
device, let us take reverb as an example, adds to the dry signal and outputs the
mix of both dry and wet signals whereas the compressor treats and processes
the entire signal and outputs the result. Even in todays DAWs (digital audio
workstations) this form of patching still exists and is commonly used.
You may think that I am being a little pedantic here but in todays diverse world
of audio technology, terminology and description can often be confusing and
general. Additionally, a number of manufacturers have taken it upon themselves
to rename conventional terminology in favour of sounding hip and now. Sadly,
this makes it a nightmare for tutors to stick to a standard and it can get
extremely confusing trying to reference new terminology against old and to
decipher todays manuals. To add to the headache most manufacturers now
include additional non conventional features into their products to give added
value and not outgun the competition.
Effects are excellent sculpting tools and a number of genres today have made
their mark because of the type of effects used within the genre.
Trance would be a good example of the use of delays and reverbs. Distortion is
prevalent in the Rock genres etc. Effects can be used globally on the whole mix,
or individually on single tracks or events, or as a combination of both.
Effects can be used creatively to evoke an emotion, or, for example, correctively
to encompass space where space is lacking. In the former, a big reverb on
strings can result in the strings sounding huge and warm, or bright and
exploding. In the latter, sensible use of reverb can add space to a certain sound
in a mix that sounds too dry compared to the surrounding instruments that may
have been recorded with space.
Chaining effects can lead to dramatic results. One effect feeding into another
and so on is a great way to enrich a sound and make it evolve. The effects can
run in series whereby one effect feeds or morphs into another or in parallel so
that more than one effect is running at the same time. However, soft and subtle
use of effects can result in track strengthening qualities. Using a chorus on a
bass sound can thicken the bass. Adding a slight amount of delay to vocals can
make the vocals sound fuller and deeper.

It is limitless what can be achieved with effects. You are only limited by your
imagination, and of course, which tools you use. Understanding how best to
utilize an effect is reliant on understanding the mechanics of the effect, what it
does and how it works, and in what quantities to use it for optimum results.
This book has been written to demystify and simplify the obstacles contained
above and to offer the reader a thorough, yet basic, approach to both
understanding and utilising effects. Wherever possible audio and visual examples
have been provided in example and exercise formats. I have found, through
years of teaching, that theory coupled with audio/visual exercises is the best
approach to understanding how a process works. Using different devices with
different parameters and terminology is only best impressed when used in
conjunction with screen captures and before and after audio files.
I sincerely hope that after you have meandered through this book you will have
a good grounding and a basic understanding of how effects work and how to get
the best out of them. But more than anything, I hope you can share some of the
obsessive passion I have for using effects to change a sound and that at some
point this becomes a fun and creative process that will give you a new outlook
on sound.
Using This Book
Throughout the book I will often ask you to listen to specific audio files; these
will be highlighted in red along with a speaker icon:
welcome [Link]
the beat [Link]
All audio files can be found in the associated folder for the chapter you are
reading.
Additionally, have included hi-resolution copies of all screenshots found within
the book, found in the related Images folder for each chapter.
Thank you and I hope you enjoy the book.
Eddie Bazil
[Link]

Chapter1:Reverb

Chapter 1 Reverb
Reverb is the most commonly used and abused effect and yet it is such a simple
and versatile tool.
I think that the problem lies in the fact that reverb seems to be the go to effect
when trying to create the illusion of space and that most users seem to load a
preset from the vsts preset menu and mix to taste not knowing what is actually
happening bar auditioning the resultant output. What makes it even more
confusing is that most vst manufacturers will name their presets based on either
the space that the reverb is trying to emulate or the sound that it is affecting.
Although this is always a good starting point, for most users it can be a
confusing path to go down.
Reverb serves a number of purposes and the two most important ones are that
of colour and space. It can also be used as a corrective tool, for example:
helping to add tails to sounds that have been cut abruptly.
The type of reverb used is as important as how to use it. There are occasions
whereby a certain type of reverb is required on a specific sound or mix because
of its design and build: a plate reverb is a good example.
I will use different types of reverbs for the ensuing exercises and demonstrate
how different they sound and why certain reverbs are better than others at
specific given tasks. But before we even think of having some fun let us get the
mechanics and physics out of the way and understand what reverb actually is
and how it works.
We have been listening to music acoustically, for thousands of years. The natural
acoustical space that the music was played in determined how the music was
perceived. The environment and the materials that made up the surrounding
environment had a huge impact on how the music was heard. We may think
that we are the innovators when it comes to creating the right space for music
to be heard in but the Romans and Greeks had a head start on us and designed
their amphitheatres and arenas to do exactly this. Some of their designs are
truly impressive. Their understanding of space and the materials the space was
constructed from is remarkable even today.
So, how does reverb work?
The listener hears the original sound, plus all the reflected sounds that come
from the original sound reflecting off surfaces within the environment. These
reflections are reflecting at varying distances and times. This is the nature of
how sound moves in a given environment. As a result, the listener hears a
composite of the original audio signal, the first reflections, and the delayed
reflections.

Chapter1:Reverb
These signals will eventually lose their energy and dissipate.
Imagine a square room whereby you, the listener, are sitting in the middle of
the room. For now, let us work under the premise that the sound that emanates
from you emanates in all directions as opposed to being directional (which sound
is).
The room is made of brick walls coated with plaster. The walls and ceiling will
have reflective properties. You shout. The shout begins to reflect from the
nearest surface and ensuing reflections come from different angles at different
times from different parts of the room. This makes perfect sense as the further
away a reflective surface is the longer it takes the sound to reach it and reflect.
The trajectory of the reflection depends on the angle the sound reaches the
surface and the angle the surface is at, for example: a sound reaching one of the
corners of the room at a 90 degree angle will reflect at that angle and reflect off
another surface and continue to reflect until it dissipates or loses energy. A good
way to imagine the reverb aspect is to think like this: after you have shouted the
residual sound that remains is the reverb.

You can imagine what this means in rooms that have reflective surfaces,
absorbing surfaces, irregular shapes and so on. High frequencies are more
prone to absorption and rooms with absorbing material (curtains, carpets etc.)
will sound more muffled. Rooms with hard reflective surfaces will sound brighter
and more brittle.
So, reverb is simply a term that defines the reflective properties of a given space
and how those reflections are projected and processed. Today, we emulate the
space of the environment and use this in our music.
Our effects units can not only emulate real spaces but also create spaces that do
not exist naturally in nature, like gated reverbs or reverse reverbs.
Figure 1 is a simple diagram displaying the various features of how reverb
behaves. The terminology used has stayed the same for a long time although
new features and therefore terminology has been introduced in modern day vst
effects.

Chapter1:Reverb
Fig 1

When the sound is triggered there is a pre delay just before the signal reflects
off the first surface. The time taken for the signal to reach and reflect from the
first surface is known as pre delay. In other words, the pre delay controls the
amount of time taken before the reverb sound begins. By adjusting this
parameter you can impress a change in distance. The longer it takes for a sound
to reach a reflecting surface, the further that reflective surface is away from the
sound source. This is the first stage in the reverb process.
This is then followed by the early reflections. The early reflections are the
primary reflections after the pre delay and this is actually quite significant as it
will denote the shape and size of the room before the decay sets in which in
itself further defines the dimensions of the space. We tend to concentrate more
on the pre delay and the early reflections to reference ourselves to our
surroundings/environment than we do to the dissipation process of the ensuing
reflections.
The decay time (also known as reverb time) denotes how long it takes for the
reverb sound to dissipate/lose energy, or die. The decay itself is equally
important when gauging the surface absorption properties of the space. We can
control the texture, length and behaviour of the decay in such a way as to create
a new colour or to expose the surface material.

Chapter1:Reverb
In most reverb units you will have a high frequency roll-off, sometimes
referred to as HF damp. In natural spaces high frequencies dissipate quicker
than low frequencies. By controlling this roll-off we can simulate the frequency
dissipation. However, we can also manipulate this by using traditional filters post
reverb. The depth and detail of control over these features allows us huge
flexibility and scope to create interesting environments and textures/colours.
As the image in Fig 1 shows, there are a number of early reflections spaced out
between each other. This is where diffusion comes into the equation. Diffusion
parameters control the spacing in between the early reflections. The tighter they
are packed together, the thicker the sound, and vice versa. The more diffusion
you apply the thicker the reverb will sound. This can translate across as dark or
confined. If you apply less diffusion, the opposite happens; you space out the
reflections further apart and make for a thinner reverb sound.
Figure 2 shows how sound is reflected in a room.
The direct sound is the sound that comes out of the keyboard and goes directly
into the microphone without reflecting off any surfaces. The black lines represent
the reflections. They are going and coming from all angles and the microphone
records not only the direct sound but all the reflections as well. Of course I have
only drawn a few reflection examples but you can appreciate what happens
when you have countless reflections coming from all angles at different times.
Sound travels at approximately 1130 feet per second which equates to about a
foot per millisecond (ms). Using the example of the room reflections it is easy to
see that some sound waves will travel further than others some will travel
shorter distances and others will bounce around the room. Because the speed of
sound is constant it then follows that the sound waves will all arrive at the
listening or recording position at different times. The bigger the space the longer
it takes for the sound to reflect and arrive at the listener/recording position. This
time factor denotes the size of the space. Add to that the dissipation time, the
time it takes for the sound and reflections to lose energy, and you have further
information about the size and characteristics of the space.

10

Chapter1:Reverb
Fig 2

Working from the image we can now ascertain a few bits of important
information:

The direct sound is the dry sound that comes directly from the sound
source without any colouration whatsoever. The reflections are referred to
as wet. In fact, this word is applied to any effect that is separate from
the dry signal/sound source. This term denotes how much of an effect we
want to apply to the dry sound. I am sure you have come across this on
many effects vsts. The dry/wet knob/fader (also called mix) is used to
mix the dry signal with the wet (effect) signal. In the image the
microphone is picking up both the dry signal and the reflections (wet
signals) and the combination of the two is referred to as the mix. By
using the wet/mix control we can have further control over space and
density.
11

Chapter1:Reverb

Reflections from different angles arrive at different times and this can
further determine the characteristics of the space occupied. It is normal
for higher frequencies to dissipate quicker that lower frequencies in a
given space and this piece of information can go a long way in not only
determining the shape and reflective surfaces of the space but also when
we want to sculpt the reverb for coloured use. It is not uncommon to use
a high-pass filter post reverb to remove unwanted lower frequencies and
vice versa.
In fact, most of todays vsts have some form of EQ/filtering built within
the vst. When dealing with low frequency sounds it can sometimes be a
nightmare taming the reverb as reverb can sound like mush and it is here
where a combination of dry/wet and filtering processes can be a real help.
If reverb is applied incorrectly to low frequency sounds then definition is
compromised.
But this doesnt mean that high frequency sounds dont suffer either.
When using reverb on high frequency sounds the actual reverb effect can
sound considerably more pronounced and it is here where, apart from
using the usual parameter controls, filtering can be your best friend.
Generally, I tend to try to limit reverb use on low end sounds like basses
or kicks and if I have to use reverb then I will almost always filter the
lower frequencies out. And when dealing with high frequency sounds the
HF roll-off is my go to parameter.

The shape of the space is critical when determining the colour and
character of the reverb being applied. In large spaces the echoes can be
further controlled so as to provide a sense of direction and shape. In
smaller spaces this is less pronounced but equally important. When you
shout in an irregular large space you will often hear some of the
reflections as distinct separate sounds emanating from different
directions. This is down to the angles and time taken for the reflections to
arrive.
Caves and mountains are good examples of delayed sound emanating
from different directions. This example may seem a little out there but it
is critical for a producer or sound designer to understand direction and
space. When dealing with sound effects for film this becomes even more
important. However, as an example, it serves us well to understand
distance and position. A number of vsts nowadays exhibit a multitude of
presets that have different shape characteristics with the added
advantage of allowing the user to reshape any space both in terms of
angles and size.

The position of the microphone is crucial when recording a sound. The

image shows the microphone in a central and equidistant position. This
means, if the room is shaped symmetrically as in our example, all the
reflected sounds will arrive equally at the same destination at the pre
determined times and angles. This means there is no bias to either side,
12

Chapter1:Reverb
and the source sound is perceived to be dead centre. If the microphone is
moved a little to either side then the times and angles of the reflections
will also change. This will then denote a change in position. You may be
wondering why this is important when dealing with reverb. Well, it allows
us to understand where a reflective surface is and how we can utilise that
to express our sound. It also serves as a great way to move the
perceived space of a sound simply by panning the reflections.

Early reflections are probably the most important factor when dealing with
a given space as they will be more pronounced than the ensuing
reflections. The early reflections will give us enough information so as to
be able to denote direction (and therefore proximity/distance) and also a
little information about the reflective surfaces. The initial reflection will be
the pre delay and the immediate ensuing reflections will be the early
reflections. A combination of both gives us the necessary information we
require to understand the characteristics of the space. The ensuing and
complex reflections are harder to decipher but no less important than the
pre delay and early reflections. Nowadays, vsts will afford control not only
over the pre delay (standard on almost all units) but also the early
reflections and how they are structured.

Before we continue exploring different types of reverb and how they impart their
own sonic character and behaviour on the audio it would be a good idea to cover
some audio examples. However, it is important to understand how an effect such
as reverb is used in a DAW (digital audio workstation) also known as ITB (in the
box), basically in software.
The usual practice, utilised from the old analogue console days, is to use an
auxiliary send and return to feed varying amounts of the dry signal to the reverb
effect and then output together, both dry and wet. Nowadays, and with the use
of software, this method is still utilised and critical as varying amounts of many
sound sources can be fed to the same reverb effect which allows for less CPU
usage and the ability to use the same reverb and its characteristics on a
multitude of sounds.
But more and more people use a reverb effect as an insert on the channel bus
much like a compressor. The difference here, apart from the multi sound sources
feeding into one central reverb, is that the channel now passes directly through
the reverb and the wet/dry mix has to be used to control the dry and wet
amounts on the same channel via the same unit. This can be constrictive but
does have its uses.
The following examples utilise the insert method whereby the mixture of the dry
and wet signals are controlled directly from the reverbs dry/wet mix control.
Additionally, the examples will be a mixture of space and colour. Hopefully, this
will help you in understanding how each is achieved.

13

Chapter1:Reverb
I will use three examples: one is a dry and slowly spoken voice file, the other is
a faster spoken voice file and the final example uses a dry piano file. This will
allow you to evaluate how the various parameters, discussed earlier, are
represented on different sound sources.
Please bear in mind that whenever you want to apply a time based effect on a
single audio file in an audio editor you will have to add silence after the sound so
as to accommodate the duration of the effect itself.
sound [Link]
experiment with [Link]
piano [Link]
All the audio examples are dry with no effects. The idea is to afford both space
and colour to each using the different parameters. So, we will try a few
examples for the voice files using the different parameters of a reverb unit and
listen to how distance and texture is applied.
The software reverb I am using is the TSAR-1 made by Softube.
Here are some variations for the sound [Link] vocal using standard
parameters:
sound effects [Link]
A more natural sounding reverb with a short 10 ms pre delay, a 1 second decay
time, low diffusion, midway density and a brighter tone which has been achieved
by leaving the HF damp (High Frequency roll-off) untouched and allowing all the
high frequencies to pass through. Of course, these settings are not mandatory in
any shape or form. They are dependent on the sound source, the space it is
meant to reside in and the final application it is required for (speech, music etc).
sound effects no density [Link]
By only adjusting two simple parameters the sound dramatically changes and
sounds far closer and more boomy. By adjusting the density to 0 the sound
becomes clearer and less muddy. Generally, low density settings work quite well
with voices and mid to high range frequency content material. The density
denotes the chickening of the reverb tail and the more density applied the
thicker the reverb tail. Of course there are many instances where a high density
value can be as useful and critical as a low density value. The slightly dark and
boomy quality of this particular audio example is solely down to the HF damping.
I have deliberately removed almost all of the high frequency content of both the

14

Chapter1:Reverb
reverb tail and early reflections and this is what gives the dark and moody effect
to the vocal.
sound effects whisper pre [Link]
This example beautifully displays what happens when the density is set to high,
the HF damp disengaged and, most importantly, the pre delay set at around 50
ms. The whispery or floaty texture is attaining because no HF damping takes
place but it is the pre delay that separates this whisper from the dry sound and
adds a nice little delay to the sound which comes across as light but intimate.
sound effects [Link]
Of course, we can go the other way if we want a specific extreme effect on a
vocal. In this example there is almost zero pre delay, a decay of around 0.15
seconds and HF damping at around 9 kHz.
The next couple of examples relate to a faster delivered vocal line: experiment
with [Link]
experiment [Link]
This effect is achieved by keeping a short decay time of 1 sec and, more
importantly, a pre delay of 0. I have done this because the spacing in between
the words is shorter and the delivery is a little faster. By using these settings I
am allowing the reverb to work itself more evenly sitting right behind and on top
of the vocals. A little trimming off the high end (HF damp at 10 kHz) and a
relatively high density (80%) takes the bright edge off the reverb tail and makes
it sound warmer.
experiment little [Link]
This final example has had a single parameter adjusted, the pre delay. All other
settings are exactly as the experiment [Link] example. However, the pre
delay value of 60 ms gives the sound an echo. I am sure you have heard this
effect before when dealing with public address systems in open or large spaces.
It is a good way to show how the pre delay on its own can dramatically change
the character of a sound.
We will continue with more vocal examples when we come to explore different
types of reverb effects, notably impulse responses (IR) and spring reverbs.
However, let us continue with the piano examples.
piano [Link]
The dry piano file is lifeless and has no movement whatsoever. Pianos are
harmonically rich instruments and really come alive in the environment they are
played in. As this is a synthetic piano sound taken from a Roland XV 5080 sound

15

Chapter1:Reverb
module and stripped of all effects including reverb and chorus it comes across as
really thin and fake. Bearing in mind most piano lines are recorded with multi
mics in a specific room/chamber it will be an interesting challenge to bring life to
this staid piano sound.
piano [Link]
By applying a small pre delay (15 ms), a 50% density value, a reverb
time/decay of around 4 sec and no HF damping the piano comes across as more
lively and richer. It doesnt sound lifeless and up front in the face but has a
smoother and more dynamic quality. To be honest the TSAR-1 has been
designed to be more dynamic and interesting than simple flat lined reverb vsts.
piano rich and [Link]
This piano sounds a lot richer and ambient and you would be surprised to find
out that only two parameters have been adjusted and neither one is decay time
which seems to be what is adding the ambiance. Density has been brought right
down so as to make the reverb sound more spacious. Diffusion has been kept
high, much as before, to allow for a bigger sound. But most important is the pre
delay and that has been increased to 45 ms. The simple combination of
adjusting the density and pre delay have given a more ambient and spacious
effect to the lifeless piano line.
piano backdrop [Link]
This effect is particularly lovely and took little time to create as the algorithmic
stereo reverb vst (TSAR-1), as opposed to an IR, is designed well and aimed for
richness and movement as opposed to a static reverb that simply denotes space
by imprinting the effect over the dry signal. Its a little akin to a minimum phase
designed EQ that imparts colour onto the sound whilst performing the usual
cut/boost of frequencies.
The settings I used to create this texture are as follows: pre delay of 27 ms,
decay of 7.5 secs, density at about 50% and a HF cut at 3 kHz. The idea with
this ambient effect is to allow a soft warmer decay that seems to roll away in the
background whilst adding a little density to afford a thicker texture as opposed
to a wispy one. The longer pre delay allows for the individual piano notes to be
heard in isolation and then affected. The use of a sensible pre delay allows
sounds to be isolated from the effect otherwise it can smear or mush a sound if
the reverb sits directly on top. Because of the size and timing variations applied
the reverb not only sounds warm and ambient but also gives a sense of width
and depth, the effect hangs in the air without exhausting the listener with acres
of high frequency content.
Using reverb on low frequency sounds is always a problem as low frequencies
dont translate across well due to the nature of the way reverb works. Mush,

16

Chapter1:Reverb
wooliness, smearing and such like terms are often used when reverb is applied
to low frequency sounds. This applies not just to percussive sounds like kick
drums but also any sound that contains low energy/frequencies. There are better
ways of dealing with sounds like these and they usually entail using chorus,
delay and so on. We will come onto these later in the book. But for the sake of
maintaining continuity let us wade through some low frequency examples. There
are, of course, ways around the mush problem and I will cover a couple of the
more tried and trusted techniques.
Let us start with a standard synthetic sub bass, a drum beat and a kick drum. I
will also show how to use reverb creatively to attain a new texture and feel for a
sound, particularly the drum beat.
bass [Link]

drum beat1 [Link]

big [Link]

I chose these particular low frequency audio files because they represent a good
selection of frequency ranges, from the low sub bass to the drum beat which
incorporates mid and high range sounds (snare, clap etc).
Let us start with the bass file. The bass is in mono, as most bases are, but I
have created two versions of the render. The first is a mono reverb and the
second is a stereo reverb.

bass horror [Link]

This bass sounds weak and thin and the reason for keeping the reverb in mono
is to avoid too much width caused by the way reverb works. By keeping the
reverb in more the sound is more centrally focused as is the space it occupies
(too much width and the bass will sound unnatural). I have used extremely
frugal settings as anything lengthy in terms of reverb decays and pre delays will
hugely impact the tightness and solidity of the bass. The pre delay is at 0 and
the decay is set to 0.37 secs, density is at 50% so as not to go either way in
terms of extremes and the HF damping is not being used. I have a very small
wet mix as anything more would simply mush the bass. Because there is little
clarity and tightness the bass will not translate across in a mix environment and
it will be even harder to control once other low frequency elements are
introduced.

17

Chapter1:Reverb

bass horror wet [Link]

Although, on first impression, you might feel this audio file sounds good and
spacious it doesnt take long to realise that the reverb movement across the
stereo axis makes the listening experience both unfocused and unnatural.
General rule of thumb is that low frequencies must be kept central in a mix
unless a certain effect is required. The kick and bass elements in a mix are
generally kept central and work as a foundation for the song. Low frequencies
are also extremely obvious once moved across the stereo field. This bass not
only sounds thin and mushy but the movement alone means that very little can
be glued to it in a mix context. It is distracting and will imbalance a mix.

bass horror long [Link]

Finally, I have included a mono render of the reverb but changed a single
parameter. I have raised the decay time to 2.5 secs so that you can hear how
bad the rendered file is even though I have kept it in mono for more focus. Most
of what you hear is low end mush. There is no clarity or definition and the audio
is even weaker than before.
For basses I tend to stick with delay and chorus effects. We will cover these
later. Now we will work with the drum loop and use varying settings to give both
space and colour.

drum beat1 wet no [Link]

Sounds a little wet and washy even though the snap of the snare and the mid of
the kick are still evident. This is more to do with a 0 zero pre delay and no HF
damping. The decay has been kept at a steady 1 sec and the density is midway
(50%). Even though this is bright it does lack some of the definition on both the
kick and snare. If you listen to the dry version you can clearly hear the bit
crushed effect on the snare, but with the treated version that has almost
disappeared along with some of the kicks low energy.
drum beat1 [Link]
This version maintains the same density and decay as the previous version but
the HF damping has been heavily applied (1.5 kHz) and the pre delay has been
taken up to 24 ms to allow the attack transients of both the kick and snare to
cut through a little better. These settings have also afforded the snare a slight
whip effect which is quite nice.
Most of todays vst reverb effects allow us to manipulate the parameters even
further and one of the most important is the control of the early reflection not
only in terms of altering the actual reflection locations but also to mix even

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further between the pre delay and early reflections. This can make for some
serious audio warping plus the fact that you can custom design the space and
reflective surface qualities even further.

drum beat1 [Link]

This comes across as a heavy gated effect and strayed far away from the
original but kept its musicality. A 0.15 sec decay time has kept the reverb decay
very short so it doesnt move and has time to return and slam again. A HF
damping at 6 kHz has let some brightness through but not too much as we want
to keep the aggressive dark flavour intact. Density has been kept extremely low
(10%) so as to give the impression of a larger space and therefore a larger
overall sound. The pre delay is at 20 ms which is just enough to give the sound
a thicker and more flammy feel. The ER mix is at about 59%. This denotes the
ER mix value with the reverb tail. This adds further ERs to the sound.
If the recording is dry you can use this parameter to add ERs. If your recording
was deliberately recorded with ambiance, be it in a room or hall etc, then it will
already have ERs and therefore you might not need to add any further ERs to it.

drum beat1 [Link]

This example displays what can be achieved just by twiddling a few of the
standard parameter knobs and experimenting. For this particular example I used
the Sonitus reverb which, sadly, is not available anymore. However, the
parameters are the same for just about any half decent reverb vst.

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Chapter1:Reverb

The settings I created are pretty standard but what I really pushed to the limit
was the pre delay (250 ms) and the zero diffusion. This gave the sound the
weird flange type of effect with some apparent delay that gave it the off time
feel. I played around a little with the ERs just to tighten up the overall sound.
Never be put off by experimenting with crazy and extreme settings, no matter
how against the rules it is.
The next example is something that can be used to layer either the same beat or
another beat. It is purely creative and just plain fun.
drum beat1 [Link]

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Chapter1:Reverb

I have deliberately kept the mix completely wet, i.e. no dry signal mixed in with
the reverb. Decay time is 0.5 secs and diffusion is at 10% but the real changes
have taken place at the LF and HF damping (low cut and high cut).
A lot of reverb units will allow for LF damping as well although old standards
generally had HF damping. The idea of damping is not just about taking the
edge off some metallic sound reverb tails, or smoothing out or adding to
frequencies, but about simulating the solidity or reflective and absorption
qualities of a space.
Of course, using density and diffusion will get you the results you need but
damping is crucial when it comes to controlling the actual perception of the
space you are trying to emulate. Caverns and open spaces will exhibit different
types of frequency reflections. A big but closed cave will have a completely
different texture to a same sized and shaped space with wooden walls etc. A
stone surface will have different reflective qualities to a soft sanded brick
surface. A bare room of walls, floor and ceiling will exhibit different reflective
qualities to the same room that has carpet, curtains and furniture. It is not

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Chapter1:Reverb
simply about distance and direction of the reflections but about the absorptive
qualities of the reflecting surfaces.
The added advantage of having these filtering options is that we can colour
sound into a new sonic texture. We will end with the kick examples only so that
you can judge what a difference reverb makes to the low frequency content
representation.
big kick1 low [Link]

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Chapter1:Reverb
The TSAR-1 comes back into action for this example. A zero pre delay followed
by 0.58 sec decay with a very small wet/dry mix (14 %) has resulted in a sound
that lacks direct focus and low energy. It is still quite good but the overall sound,
although bigger, is a little weaker and warmer than the cut of the original.

big kick1 [Link]

By changing the following values we can retain a little bit more of the kicks
attack transients and filter out all the higher frequencies to leave a slightly
clearer yet warmer sound:
Pre delay at 45 ms, decay at 0.15 secs, density at 50%, Tone at dark (adjust to
taste) and the HF damping at 300 Hz (0.3 kHz).
Finally, let us have some fun with a creative process using the Sonitus again.
big kick1 [Link]

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Chapter1:Reverb
If you look at the settings I have created you will see that the action takes place
with a completely wet signal being filtered with no diffusion. I will let you check
the input values and feel free to experiment and enjoy yourself. The wide rumble
effect is not a difficult one to create but it is an educational one as it displays
how sound can be manipulated simply by using reflections and filtering as tools.
We will come to using reverbs for creative uses in more detail but now is a good
time to explore some of the different types of reverbs and how to use them. We
will stay with the same audio files as they cover a decent range of frequencies
that we can process.
Plate Reverb
Putting natural reverb aside (mic recording ambiance in a space) most of the
classic tracks of the 1960s onwards used plate and spring reverbs.
A plate reverb was a solid construction of a sheet of metal that was suspended
within an enclosure and held together by springs or clamps and only at the
corners. The idea being that the sheet is allowed to vibrate. A transducer, much
like a speaker driver, was used to direct energy and generate vibrations. These
vibrations were then picked up by mics (microphones), or pickup transducers,
attached to specific parts of the sheet (plate). In the early days a single mic was
used for mono and later 2 mics were used for stereo.
A damping pad would be used to adjust reverb time, the closer the pad the
shorter the reverb time. These reverbs were big (1 meter square and above for
the sheets) and had to be constructed into a solid sound proofed framework. The
reflections travel faster and build up quicker due to the shape and construction
of the sheet and therefore exhibit a denser and brighter sound that has become
synonymous with plates. There is so much energy created by the nature of the
sheet process that the early reflections are so dense that they are almost
inaudible, or rather indistinguishable, and because the decay is long and smooth
(the sheet denotes the time variances here) they are very specific in the type of
colour they impart onto the dry signal. A physical metal sheet will always allow
sound to travel faster than if it were travelling through air and this accounted for
one of its unique qualities: reflections.
The construction of such a device is not easy. Although the actual construction
only requires some elbow grease it is the construction and calibration of the
sheet (thickness, tension mounting etc) that is complex. Taming the sheet is
another issue as metal that vibrates can ring. However, nowadays we are spoilt
for choice with the multitude of reverb plate presets in software reverbs. But, as
in most cases, they are only emulations and only a few truly capture the sound
of this very specific type of reverb.

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Chapter1:Reverb
The first audio example features the same piano file we used earlier. I will treat
it with the classic EMT 140 plate reverb although I am using the software
emulation of it.
piano [Link]
The density of the reverb is very obvious with almost no pre delay being evident.
It is smooth and has a nice decay to it but due to its density a little bit of
equalisation would help to sculpt the reverb a little better.
piano long [Link]
This is a standard long plate with the mix set at about 40%. Again, dense and
smooth but no specific clarity.
sound effects flat [Link]
A lovely smooth long decay where the density works in our favour. The
brightness and wispiness of the reverb added to the density affords a rich tone.
sound effects short vocal [Link]
Even though the decay is shorter and the plate itself is smaller the warmth and
smoothness of the reverb is still evident. The brightness comes through as a
wisp of air above the dense decay.
drum beat1 drum [Link]
The drums come through as both bright and dense. This is a classic example of a
drum plate.
Most plate reverbs require additional processing to sculpt the effect according to
the source material. Because a plate is built without any frequency dampening,
this needs to be applied to get the best out of this effect. Eq works a treat to
shape the effect and the final example uses the kick sample and a short drum
plate with additional frequency damping.
big kick1 short drum [Link]
The kick sounds airy and with taming of both low and high frequencies a balance
is found between total mush and total metallic resonances. This, although a little
metallic and airy, is not too bad and goes to show how a plate reverb can affect
various frequencies. As I never use reverbs for kick drum samples I find that the
plate reverb can have some useful tasks to perform in a kit building context.
Let us have a look at another exciting form of reverb: Spring Reverb

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Chapter1:Reverb
Spring Reverb
Not too different to the plate reverb, but instead of a metal sheet a metal spring
coil is used and this is not fixed as rigidly as the plate. A transducer feeds the
signal into the coil and a pickup is used to collect the output as the coil reflects
and vibrates. Due to its nature the spring reverb can be quite metallic and dense
as it behaves similarly to the plate reverb when it comes to the processing of
early reflections. A single transducer and pickup means the spring acts in mono
but running two springs together would create a stereo effect and each side of
the channel could be treated separately thus accounting for some interesting
stereo effects. The spring can also be pushed to create a twang type of effect if
the input is driven heavily and this accounts for one of the characteristics of the
reverb and because of this it is still being used by guitarists. The tension of the
spring/s can be adjusted to create different effects and the springs can also be
shaken or rocked which allows the springs to collide creating huge and
thunderous effects. It was not uncommon to use up to three springs.
The sound of the spring reverb is quite distinct and the following examples
should reflect this colour even when used conservatively. Probably the best
spring reverb emulation plugin I have used that is sensibly priced is the Softube
Spring Reverb. This vst allows for using up to three springs, adjusting the
tensions, the ability to shake the springs and additional control over bass and
treble.

This vst is exquisitely simple. It has a dry/wet mix, a basic bass and treble EQ, a
selector switch that can toggle between 1-3 springs, a tension function for the
springs and the shake function was discussed earlier.

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Chapter1:Reverb
One of the best sound sources to use the spring reverb on is a clean electric
guitar line.
clean tele [Link]

clean tele2 low [Link]

By selecting a very low tension and only using a single spring it is easy to add
life to a dry guitar line. The mix is set to about 20% keeping the mix pretty dry.

clean tele2 [Link]

The mix is at about 70% keeping it very wet. The tension is at maximum and
three springs have been selected with a low (bass) boost and HF (treble)
damping. This has afforded a dark colour to the sound.
Using a slightly different guitar line we can create a sense of richness. I am
alternating between the audio files so as to afford variety and to accustom you
to the way specific reverbs work on different material.

clean tele [Link]

clean tele 1 [Link]

Staying with a low dry/wet mix but with a high tension and 3 springs we can
give the perception of richness without washing the sound in reverb. I have left
the bass EQ as is and boosted the treble slightly so you can hear the reverb a bit
better. You can, of course, adjust to taste and feel free to extreme these
settings as that is the only way you will understand what does what to what.
In the next few of examples I am going to use the shake function on the big
kick1 audio file so you can hear the varying effects when the springs are rocked
against each other. Of course, you can use the shake function on any sound
and it is a great way to create sound effects.
big kick1 [Link]
big kick1 [Link]

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Chapter1:Reverb

big kick1 [Link]

It is fair to say that specifically designed reverb units will colour sound in a very
unique way and it is these qualities that are used when reverb is not required for
the standard transparent space requirements. Plates and springs have very
unique characteristics that package them in the colour and design arsenal of
effects processing. Of course, they have been used extensively in the past as
standard reverb effects within a production and you only have to look back at
songs from the early years prior, to the introduction of digital reverbs, to hear
them in full action. Limiting yourself to standard reverb effects is depriving you
of technology, however old, that can really add new sonic flavours and colour to
sounds and mixes.
Although there are many types of reverb designs in the market the final two I
will concentrate on are reverb chambers and impulse responses.
Reverb Chamber
These are simply rooms that are designed with specific acoustics in mind and, in
some cases; the room dimensions can be altered. The concept is quite simple:
design a room where you have control over the reflections and adjust the rooms
dimensions to attain different reverb values; record within this room by using
multi microphones positioned in different places to attain different textures.
There are still studios that have dedicated chambers that have afforded a
specific colour to a recording and are therefore still in demand by recording
artistes and producers. Some reverb chambers were very specific, an example
being plate reverb chambers that were designed to house large plate reverbs
and the room was used to further shape the reverb information. In fact, it didnt
end there; some rooms were built as trapezoidal with the speakers being placed
against one wall and the microphones were placed against an opposing wall. This
allowed for some very creative reverb responses and the shape was designed to
abate standing waves.
Nowadays we emulate these spaces by recording in rooms with specific
microphones so as to capture the rooms ambiance. In other words, the room
acts as the environment for the natural ambiance/reverb captured by the
microphones. This is a basic and simple version of a designed chamber. I have
worked in studios in the past whereby the chambers had adjustable screens that
acted as portable walls. These screens were designed with control of reflections
in mind and could be moved to create new reflective results thus allowing for an
almost infinite number of room designs and modes.
Of course, not everyone can have a dedicated reverb chamber at their disposal
so multi micing (using more than one microphone) and recording in specific
rooms and locations is the best and most affordable way to capture a rooms

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Chapter1:Reverb
natural response, and this has now become common practise and to some
degree preferred over using conventional reverbs. But this in itself can cause
problems for those on a budget with limited access to different rooms and multi
microphones of different designs and quality. And for this reason alone the best
way to capture a rooms response, or to have access to different reverb types
and designs, is with the use of impulse responses (IR).
Impulse Response
Other relevant terms used to describe IRs are convolution or sampling reverbs.
The idea is to capture the response of an acoustic space and to playback the
response using a convolution playback device, be it software or hardware,
although nowadays the software playback devices are more commonly used. The
basic principle is to record a test signal, a swept sine wave signal is the most
commonly used test tone, and to convert the captured signal into an impulse
response which is then loaded into the playback device and played back as an
effect. You are not limited to just reverb effects but IRs can be created of mic
pres, compressors, amps, tape and so on.
This not only allows you to have access to different types of effects but also the
different characteristics of varying much sought after hardware/software units. I
have IRs of some notable and sought after units that I could not afford to buy or
source. It is an extremely cheap and easy way to access and use just about any
piece of equipment that can be used to act as a medium for the test signal. Play
the signal through a nice boutique piece of gear, record the output and then
convert to an IR to use in a playback device: simple and yet so effective.
I find using IRs are a much better option for creating real spaces than standard
reverb plug-ins although nowadays some reverb plug-ins have made a lot of
headway and are very usable. You can get as creative as you want with IRs. I
have created manic IRs by running a test signal through a hi hat triggered gate,
or through a plastic pipe and so on. If it can act as a medium for a test signal
then it can be recorded and used as an IR.
There are many convolution playback devices available and one or two are free.
There are also a number of sources, some free, for IRs available on the internet.
Basically, this allows you to have access to many IRs of varying sources at no or
very little cost.
I prefer to use IRs for real spaces than standard reverb plug-ins simple because
the real space captures sound more natural than algorithmically generated
responses. There are, of course, certain limitations with using IRs; IRs are static
responses, in other words they are a capture of an instance of a recording and
are therefore simply the equivalent of a single frame of film. Additionally, some
playback devices afford no editing of the IR which limits it to be used as is.

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Finally, your computers cpu overhead is directly related to the length of the IR
as the computer has to calculate the incoming audio in real-time. However,
nowadays we have very powerful computers that can handle many simultaneous
instances of IRs and playback devices have advanced tremendously to include all
sorts of editing features that allow the IRs to be manipulated extensively.
In the next few examples I will use IRs to create both natural spaces and crazy
effects. This will hopefully convince you how useful and creative IRs can be. I
will use standard free IRs I have collected from the internet and some IRs
(Galactic Textures) that I created that are available from my website.
drum beat2 [Link]

I am using Voxengo Pristine Player as the convolution playback device and have
loaded one of my IRs from the Galactic Textures pack. I have made sure that

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the player only plays back the wet signal and does not mix it with the original
dry signal. This will allow you to hear the wet effect and you can mix to taste.
The IR is displayed as a graph and you can see that there are a number of
editing features available to further shape the IR. I have kept everything as is.
The graph beautifully displays the shape of the impulse and this gives the user
an idea as to the type of response and colour imparted onto the audio.
The result is dramatically different to the original drum beat.
drum beat2 [Link]

drum beat2 [Link]

A crazy IR that completely changes the audio and this is evident from the IR
shape.

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drum beat2 [Link]

A gated effect I created as an IR that marries well with this drum beat to give a
manic flavour. Let us have some fun with the vocal line.

experiment with [Link]

experiment voxengo [Link]

By applying a heavy reverb IR with a lot of HF roll-off, low eq boost and a short
pre delay the sound comes across as if the vocal is layered with a low frequency
resonance much akin to a power line sound and yet it maintains the reverb
characteristics.

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experiment voxengo [Link]

If you look at the display above you will see that the IR has been reversed. I
have also added a lot of pre delay and shortened the IR length. This now sounds
like a reversed reverb sitting with the original. Another way of attaining this
effect, the ghost or poltergeist effect is as follows:
I will use Cubase SX3 to create the ghost effect using reverb.
I have created an audio track and imported the experiment with [Link].
I have applied a hall reverb and hi-passed it at around 200 Hz as I am only
interested in the high end of the reverb effect.
experiment with reverb [Link]

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The next step is to reverse the sample and bounce it to a new track.
experiment with reverb ghost [Link]
Once you have bounced the audio track to a new track you would then need to
reverse it again.

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The file is obviously longer as it now has the reverb imprinted on it.
experiment with reverb ghost [Link]
You are not limited in any shape or form as you can use any type of reverb with
any settings that appeal to you. As this is a creative process the only limits are
those of your imagination.
Once you have the reverb aspect of a sound sorted out there are many things
you can apply to the reverb itself such as gating the reverb (an 80s favourite
snare trick), compressing the reverb to change both timing information and
colour, using filters to shape the texture/colour, automation and so on. However,
for this chapter I am only concerned with using reverb as is so as to change both
the colour and perception of a sound.
The following two examples entail using a standard kick drum sample and
applying reverb to change the sound entirely so it sounds more like a tom burst.
deep kick [Link]

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deep kick [Link]

By using a long pre delay (to keep the attack intact), low diffusion and density
and veering more towards early reflections as opposed to just reverb, the sound
comes out really thick, deep and wide and sounds more like a sound effect than
a kick.
To create a deep yet more distant sound without hearing the attack we can do
the following to the same kick sample.

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deep kick distant [Link]

Fully selected reverb with zero pre delay, high diffusion and sparse density make
this sound darker and more distant. The idea is to try to remove the attack
transients by drowning them in instant reverb and then to shape the decay.
A process I mentioned earlier, gated snare, entails adding a gate after the
reverb. Think of the noise gate like a standard iron gate that you would see in a
field. The gate opens and allows the wet (reverb) signal through and then closes
to cut the reverb out. The most notable example of this technique was

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synonymous with the 80s music but was used as early as the 70s. Basically, the
idea was to control the reverb shape and time by using the noise gate to open
and close when desired. Most of todays vst reverbs will have a gate function
built-in or will have parameters that will emulate the gate effect.
Lets run through a few simple examples using a rim snare sample and Rob
Papens RP Verb, which is a nice and detailed reverb vst, and Softubes Tsar-1
reverb vst with a gate after the reverb.

rim snare [Link]

In Sound Forge I am using the plug-in chainer which is simply a function that
allows effects to be placed in serial (one after the other). I have selected the
Tsar-1 as the reverb vst and followed it with funks Gate. This is the traditional
way of attaining the reverb gate effect and the old school way of showing you
how the process works.
I wont bother with a screen capture of the reverb settings as that really
depends on how you want to shape and apply the reverb but I will include the
gate settings capture so that you can see how the gate is set up to receive and
process the reverb.

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You can see that the chainer has two effects in series (just below untitled) and
that the first effect is the reverb and the second effect is the gate.
I have selected a 0 attack, a short hold and a very short release. I have tamed
the reverb passing through with the HF roll-off. The shape in the display denotes
how the gate opens, holds for a predetermined time and closes. The opening of
the gate is immediate and the shutting is abrupt.
rim snare traditional [Link]
The effect you hear is a standard gated reverb effect and I am sure you have
heard it before. It is actually going through a revival at the moment and more
and more producers are becoming aware of its merits and uses. As I said before,
more and more reverb vsts now come with built-in gates or emulations of the
gate effect, and good examples of these are the RP Verb and the Tsar-1.
Using the same rim snare sample and processing it using the RP Verb we can get
a pretty good gated reverb effect.

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rim snare rp verb [Link]

This reverb vst is more detailed than others we have encountered and offers far
more editing options but the general parameters are the same. The beauty of
this vst is that there is a separate section for shaping the envelope and it is this
section I have played with to get the short and gated reverb effect, but having
additional control over the early and late reflections affords further scope for
sonic mangling. This particular reverb is made by Rob Papen and is one of my go
to reverbs when dealing with ambient and dance genres.
Using the parameters above we can further shape the gated reverb to sound
dark and full.
rim snare rp verb [Link]
By adjusting the low and hi-pass filters, adding diffusion and damping the
shorter reverb length this snare now sounds darker and deeper.
The above are creative uses of IRs.
The following, however, are natural uses of IRs more for space and standard
reverb than mangled colour, and for this I will use the piano sample.
piano [Link]

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Voxengo Pristine Space has some decent editing features and for this example I
have adjusted the IRs length and altered the pre delay to allow the piano note
attacks to come through a little better. I have also mixed the dry with the wet
signal to allow for a more natural effect.
piano voxengo [Link]
The IR is a long and lush response and this marries nicely to the timing of the
notes affording a sweet and lush reverb.

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piano voxengo [Link]

A rich and beautiful IR that exudes a shimmering effect that seems to float
above and behind the dry piano line. By adjusting the pre delay to allow for the
piano notes to come through and hi-passing the reflections so as to give a
lighter and less dense effect the response shines through and makes this piano
come alive.

piano voxengo [Link]

This version is up close and personal but still rich in texture. By using a highpass filter (above blue line) I am able to select only the high frequencies of the
reverb to shine through. This thins out the reverb and stops it from mushing up
the lower frequencies.
Finally, I would like to end with a mix reverb using the TSAR-1 with very
conservative settings to add life to a final mix. Of course, this is not how I would
normally mix reverb within a mix as I tend to set up three different reverbs on
auxiliaries; one for the main global reverb making sure to keep it as transparent
as possible, two will be set up for vocals and will entail room settings depending
on the content and the third reverb will be for colour and will be created to add

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specific sound design colour to certain parts of the song.
However, for this particular example I am only going to use a main reverb to
add some life and space to an existing dry and uninteresting mix.
smile mix [Link]

smile mix wet [Link]

The important parameters here are pre delay and time. The idea is not to wash
the mix in reverb but rather to add a little movement and accentuate the

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dynamics. The ideal situation is to work from auxiliaries with varying amounts
for each sound as opposed to slapping a master reverb across the stereo bus.
However, for the purposes of this tutorial it explains nicely how and what
happens to a mix when specific reverb parameters are played with.
Finally, I would like to touch on the most important, and now very commonly
used, form of reverb: natural ambiance.
Natural Ambiance
Natural ambiance is attained by recording a sound in the environment itself, as
opposed to using artificial reverbs, and using the recorded room sound with the
original dry sound. The most common approach of attaining this is to record the
sound with multi microphones and specifically with overhead microphones placed
at a distance from the dry source.
The overhead microphones capture the ambient sound of the recording along
with the microphone/s that is placed far closer to the source dry sound. This
process is not limited to room sounds but can be used in any environment be it
indoors or outdoors. It can also be used completely as a wet signal, in other
words, recorded with the environments ambiance and used exactly as is with no
layering with the dry sound. The best way to explain this process is with an
example.
I will create a three way recording using a condenser microphone positioned
directly at source (vocal recording) with two condenser microphones (mono)
spaced above the sound and at an equal distance so as to capture the overhead
recording in stereo (or rather, dual mono). The following three audio files
represent the centre and direct recording in mono (voice centre), the overhead
stereo recording with the two microphones (voice overheads) and the final
render of all three files (voice main). The final render includes the mono centre
voice recording mixed with the stereo voice overhead recording.

voice [Link]
Recorded in mono using an AT 4033a with no pop shield. The idea is to record as
naturally as possible for these tests. A pop shield would have helped greatly with
plosives etc but I stayed away from using one so you can use the audio files as
you please. This sounds quite focused and lacks space and width.

voice [Link]
Recorded using Sontronics STC-1s. These are small mono condensers and
spaced equally apart at 6 inches and 6 inches above the source (my voice). Each
recording was a mono audio file later panned when mixed. These sound wider

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and more spacious than the centre recording.

voice [Link]
A combination of the centre recording (AT 4033a mono) and the overheads
(STC-1s dual mono), mixed. This sounds wider, deeper and fuller than the
individual recordings.
Let us end the ambiance recordings with a set of keys being rattled left to right.
keys [Link]
This process is exactly the same as the voice ambient recording and the centre
recording is focused with no movement.
keys [Link]
Recorded with the STC-1s panned left and right at the mix stage. I did this so
that the drastic pans could be heard. Once layered with the centre recording it
will sound fuller and wider.
keys [Link]
Containing the mono centre recording, the individual mono recordings panned
and mixed to stereo and then mixed by adjusting the centre gain to allow for a
wide and dynamic recording.
Of course, the above ambient recordings can be achieved using reverb,
automated panning and so on but to me nothing really compares to natural
ambient recordings of the environment mixed with the direct source.
More and more producers are opting for ambient recording rather than using
reverb simply because recording a sound in its, or a specific, environment
sounds far more pleasing and natural. I regularly layer my direct recordings with
ambient takes of the same sound using overheads and then I mix to taste. In
fact, this is the one of the main reasons why impulse responses were created in
the first place.
The subject of reverb and how to use it is a vast one and it helps to try to
envisage the reverb as something more than simple data. Some people are
gifted with seeing music in colours (Synaesthesia), I, sadly, do not possess this
gift. However, I try to attach images to the sounds or frequencies I hear. I then
try to form an image based on the texture of what I am hearing. In terms of
reverb I use every day images to try to understand what the effect I am hearing
is doing.

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Chapter1:Reverb
Wispy translucent clouds denote transparent reverb, dark foreboding storm
clouds denote a thick and dark reverb, nebulae in space with lots of colours and
a wide spread denote coloured and uncontrolled reverbs and so on. If you can
attach an image to a sound then you will have an easier task of shaping and
representing that sound.
Reverb is not just about space, so try to experiment as much as you can and
you will be surprised as to what can be achieved!

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Chapter2:Modulators

Chapter 2 Modulators
Now is a good time to touch on the subject of modulators as they become
integral when dealing with specific time based effects.
There are many technical definitions of what a modulator is but the only
definition we are concerned with is in the context of sound design and
production.
A modulator is a device that controls the parameters of another device. The
modulator is called the source and the device being modulated is called the
destination. A good and simple example of a source modulator is the pitch
wheel on a keyboard and in this instance the destination is pitch. In other words,
the pitch wheel when moved alters the pitch of the sound being played.
LFOs are often used as source modulators as they can have varying shapes and
can be both cyclic and random and they can modulate (control) a whole host of
destinations. An LFO is a low frequency oscillator and it oscillates at a far
lower frequency than a traditional oscillator (usually at around 10 Hz). This
means that it is not heard like a traditional oscillator but still cycles like one and
is therefore a great source modulator.
In the next few examples I will use different types of LFOs to modulate different
destinations so that you can see and hear how they behave. I will use a simple
synthesizer vsti, Strobe by FXPansion, as it has a nice display that shows the
shape of the selected LFO waveform and provides simple destinations to play
with.

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Chapter2:Modulators
The LFO section is on the left and a sine wave shape has been selected and
assigned to modulate the pitch of the saw oscillator. The display in the middle
shows the LFO selected waveform shape and labels it as such just below the
display.
strobe lfo sine [Link]

strobe lfo sine pitch [Link]

In this example the LFO rate has been increased to, as the display shows, 1/16
note. You can hear how much faster it is cycling the pitch.
In the next example a saw up LFO shape is used to modulate pitch.

strobe lfo saw up [Link]

In this example a saw up LFO shape controls the pitch and it is obvious to see,
and hear, the effect. The pitch rises on the ramp up side of the saw waveform
and then drops vertically back down to 0 and then rises again and so on.
We are not limited to using only a single destination when using a modulator. In
the next example we will use a square wave LFO shape to modulate both pitch
and filter cut-off.

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Chapter2:Modulators

strobe lfo square pitch [Link]

The shape is a standard square wave, as is seen in the display, and the
destinations are pitch and filter cut-off. This gives it the distinct sound that you
hear in the audio example above. Square waves start at 0 rise to 1, stay there
for the selected period, drop down to 0 again, stay there for the selected period
and then rise again to 1 and so on.
To take this a step further, let us look at what else can be defined with LFOs as
source modulators. If an LFO is used to modulate the amplitude of a sound
(destination) then it is called tremolo.
Using the Ohm Force Ohm Boyz delay plug-in we can create tremolo effects by
simply routing the LFO to modulate the amplitude of the incoming sound.

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Chapter2:Modulators

keys dx1 ohm [Link]

Tap1 (pre delay) Level is modulated using a sine wave shaped LFO with 1/6
timing period. Dont feel overwhelmed with the features of this plug-in as we will
explore it in more detail in the next chapter.
If an LFO is used to modulate pitch then it is called vibrato. Using Camel
Audios Alchemy we can route the LFO to modulate the master tuning.

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Chapter2:Modulators

alchemy slight [Link]

The image above displays the modulation matrix on the left where all the
routings take place and right of that the LFO shape and rate are displayed. By
using the LFO to modulate the master fine tune parameter we are able to create
a vibrato effect. I will now create another example using the same sound and
setup but pushing the LFO rate and depth (right next to the LFO on the left in
the mod matrix) higher so you can hear the warble effect of the pitch being
modulated by the LFO.

alchemy heavy [Link]

By using a higher depth and rate for the LFO we can exaggerate the vibrato
effect.
Source modulators are not restricted to just LFOs as anything can be used to
modulate a destination.
Although we have dealt with LFOs being used to modulate pitch and amplitude
we can use them for creative tasks too. A LFO can be used to modulate pan or

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Chapter2:Modulators
the filter cut-off or just about anything that can be modulated. I will cover these
particular examples as they will become relevant in later chapters.

strobe wave [Link]

A simple sine and square waveform running dry with no effects or modulation.
We will now assign the LFO to modulate the pan.

strobe wave lfo [Link]

A sine wave shape has been selected for the LFO and the destination is the
master pan at the top right hand side of the image. I have highlighted the
range of the pan by having the LFO modulate the pan from extreme left to
right. The images below display the waveform as dry and without modulation
followed by the wet and modulated version.

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Chapter2:Modulators

The second image shows how the sine shaped LFO behaves when modulating
pan and this can be seen by the alternating wave shapes on both channels from
left to right.
Using the same LFO shape we can now modulate the filter cut-off.

strobe wave lfo filter cut [Link]

The shape of the LFO modulating the filter cut-off is clearly defined in the image
above. It is always helpful to both hear and see what a process does.
We can also assign the LFO to modulate the filter resonance.

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Chapter2:Modulators

strobe tone res [Link]

The sine shaped LFO is modulating the filter resonance and the effect is very
distinct and repetitive.
Modulators are not restricted to just LFOs. Almost anything can be used as a
source modulator, within reason of course, and one such favourite is velocity. In
terms of modulation velocity is defined as the harder you hit a note the higher
the modulation value.
An envelope is defined as a shape so in the case of an amplitude envelope or
filter envelope and so on the envelope will have the following parameters: A for
attack, D for the decay of the attack, S for the sustain and R for the release. You
will regularly come across envelopes denoted as ADSR (attack, decay, sustain
and release) and these parameters are used to shape a sound or any
destination.
The next example displays this perfectly. By assigning velocity to control the
filter cut-off we can play at varying velocities and that will modulate the filter
cut-off at different values.

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Chapter2:Modulators

strobe velocity fil cut [Link]

The harder I hit the note the more the filter opens up. The velocity, in this
instance, is controlled by the amplitude envelope (amp env). The diagram
beneath shows the peak values of the velocity when hit.
Lets use another instance of velocity as a source modulator but this time we will
control pan as the destination.

strobe velocity [Link]

On the right hand side of the GUI you can see that the pan has been set to a
maximum range from left to right and the source modulator is velocity (amp
env). This allows for extreme panning the harder and softer I hit the note/key.
The waveform display below the main diagram shows the pan values on the left
and right channels. The right channel displays the higher velocity values which

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Chapter2:Modulators
makes the sound pan to the right. The left channel displays the lower velocity
values which make the sound pan to the left.
In the next example we will use velocity to control pitch.

strobe velocity [Link]

Selecting a range for the pitch of the oscillator and modulating this with velocity
allows us to go through varying pitch values the harder/softer we hit the
note/key.
This, like the examples to follow, is a modulation envelope.
In this example I will use Strobes mod env (modulation envelope) to shape the
filter cut-off and you will see the shape of the modulator in the display of the
vsti and the mod env parameter settings will reflect this shape.

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Chapter2:Modulators

strobe tone mod env cut [Link]

The mod env shape is seen in the first image vsti display and the resultant
waveform (above) shows the behaviour of the filter cut-off being modulated by
the mod env. The attack of the mod env is what creates the swelling filter effect
as it takes time for the filter to open.
The following example highlights what happens when the mod env is assigned to
control filter resonance.
strobe tone [Link]

strobe tone [Link]

The mod env shows that the attack is taken up quite dramatically to allow the
resonance to build up slowly and the sustain has also been taken up quite a bit
to allow the resonance to stay as is before the release is used.
In a later chapter we will use combinations of effects and source and
destinations to create complex effects and textures and with this in mind I would
like to conclude this chapter with a taste of what is to come. However, I will only
really concentrate on the modulation matrix in Rob Papens Albino vsti to
demonstrate the complexity of results using LFOs and velocity to control filter
panning and resonance.
The following preset sound is taken from my Albino sound bank called Dark to
Light and features a good example of LFO filter panning and velocity to
resonance.

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Chapter2:Modulators

albino fil pans and res [Link]

The LFOs, bar LFO4, are routed to control the two filter pans which gives the
subtle and smooth movement across the stereo field. Velocity is assigned to
modulate both filter2 resonance and the main amplitude. You can tell by
listening to the audio that every time I hit the chord harder I get a different
resonance value based on how hard I hit it. This allows for a really sweet and
rich dynamic content.
Hopefully, this short chapter explains the use of a modulator to control specific
destinations and demonstrates how important a source modulator is in
controlling the overall sound it is designated to control. I have kept this simple
as we will not need to cover a vast array of source modulators because the
ensuing effects chapters predominantly use LFOs as sources to modulate specific
time based destinations. All this will become more apparent as we journey
through more examples.

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Chapter3:Delay

Chapter 3 Delay
Delay is actually very simple to understand. The process entails taking an audio
signal, holding it in temporary memory, and then playing it back after a certain
time.

The earliest form of delay involved using tape loops that would record and play
back the signal. A signal was fed through the tape via the record head and a
series of playback heads provided the delays. The length of tape, tape speed and
switching the head positions determined the delay time. Additionally, the
delayed signal could then be fed back into the record heads to create further
delays and decays. Complex delay times could be formed by adjusting the
distances between the playback heads.
Tape delays have been around for a long time and are still used today. Famous
makes included the Watkins Copicat, Roland Space Echo (RE201), Echoplex and
so on.
The beauty of analogue delays, as opposed to their digital counterparts, is that
the delays come across as more distant and spatial as they are less distinct and
coloured. Digital delays are far more precise and clinical but some have specific
parameters that emulate tape delay.
Control over the delays affords numerous different effects. Echo, which is a very
short delay of less than one second, is a common effect but doubling, flanging,
phasing, chorus and dub delay, are also very common. The dub delay effect is
achieved by feeding the output of the delay back into the input and if enough of
the output is fed back into the delay unit the delay will start to self oscillate thus
creating the famous dub delay. This is called feedback.

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Chapter3:Delay
Modulation, a varying of the delay time over a particular range is another key
parameter of time based delays. By assigning minimum and maximum values,
the delay time will sweep between the two and create a moving delay as the
times vary over the set range.
You can also tap a delay time into some of the more modern delay units. This
can create some unique time delays and allows the user more control over the
time. Different amounts of time delays give different effects.
Today, we are afforded delay units with vast amounts of control from the pre
delay all the way to panning and retiming of the delays along with EQ and filter
sections.
However, the use of delay doesnt end with simple repetitions of the dry signal.
Delays have been the preferred choice for creating space and width in a mix.
Stereo width can be achieved by simply delaying one channel of the stereo file.
Vocals can be both spread and thickened by using delays. Drum beats can be
livened up and shaped to produce a shuffle effect by simple use of delays and a
lot of todays delay units, software and hardware, allow for tap tempos, a
function that allows the user to tap their own delay times into the unit.
For me, delays are one of the most, if not the most, potent of all the effects
available as so many other effects can be achieved simply by altering the base
parameters of the delay unit. As with reverb timing is everything.
Let us start with some very simple examples and a basic principle in timing.
Creating Stereo Width
To create stereo width using either a mono file or a stereo file that has the same
information on both channels involves changing the timing information of one of
the channels. By having a tiny offset on one channel the brain and ears perceive
the sound as wide. This happens because the brain has to decipher the timing
variances of both channels, in other words, one channel is heard fractionally
before the other and therefore the brain conceives and treats both channels as
one source. If one channel is offset against the other by too high a value then
the brain treats the overall sound as two signals. This all comes down to timing
and choosing the right offset value.
The first example entails using:
experiment with [Link]
This file is actually a mono recording presented as two mono channels summing
as a stereo file. Basically, the same information is repeated on both channels
and therefore a stereo output is created but is not what we call a true stereo file,
meaning that it has not been recorded in stereo using two mics and both
channels carry identical information.

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As you can see, both channels carry identical information. In terms of listening
to this file you would hear no difference between the mono and stereo versions.
I have converted this mono file into stereo because I need both channels active
so I can alter the timing information of one channel and play the two together.

I am using the insert silence function in Sound Forge to insert 10 ms of silence

before one channel of audio. You can use any function that allows one channel to
be offset against another. I am using a simple method of moving one channel
further up the timeline against the other. You can achieve the same result using
delay, moving a track in your DAW against the original dry mono track, dragging
one channel away from another and so on.
Here is the final render with one channel offset by 10 ms.

experiment with reverb [Link]

The lower (right) channel has been offset and this can clearly be seen with the
image above. The distinction between the two channels is now obvious and the
vocal line comes out wider.

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This process leads us nicely into the next effect: doubling. This process is a nice
way to thicken sounds and basically you have to apply enough offset, or delay,
to hear the sound as a doubled effect.

experiment with reverb [Link]

By applying an offset of 30 ms, as opposed to 10 ms in the example earlier, the
vocal channels are now separated enough so that you can hear two extremely
close vocal lines. It is not enough to hear the vocal lines as two separate and
distinct channels but just enough so the listener hears a thicker and doubled
sound. This method is the old school standard method of applying stereo width
without using M/S (middle and side) etc. However, we can do exactly the same
with a delay effect.

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Chapter3:Delay
Using the Ohm Force Ohm Boyz delay effect plug-in I am able to create a delay
on the second Tap whilst leaving the first Tap at 0. This is the same as offsetting
one of the stereo channels even though the Tap pre delays are not actual
channels. Tap 1 refers to the first pre delay and Tap 2 to the second pre delay,
also known as repeats. By having two varying Tap times there is an audible
difference between the two pre delays, and this acts as a stereo spread. I have
also panned each Tap to opposing sides so as to create a wider stereo image.
experiment with reverb ohm boys [Link]
Using only pre delays we are able to have the timing variances. This can also be
done with two channels of reverbs at varying pre delays or by using a standard
stereo delay effect which is covered in the next example.

In the above example I am only using one Tap (pre delay) but two delay lines
(left and right stereo). By varying the delay on line 2 I am able to create the
same channel offset which gives us the thick doubled sound but I have kept
away from extreme timing variances and pans.
experiment with ohm [Link]

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Chapter3:Delay
Ohm Forces Ohm Boyz affords a lot of extra features and detailed control over
all aspects of the delayed signal including some very usable filters and effects.
We can use delays to create some interesting rhythms with drum beats. By
varying the delay times, feedback and syncing the pre delays to a tempo we can
generate standard or crazy effects. Some delay units have LFOs (low frequency
oscillators) that can be assigned to control the movement of the delays and
pans.
Ohm Boyz is one such effect incorporating both LFOs as source modulators and
additional effects like distortion. An LFO is simply an oscillator that runs below
the hearing range (in the old analogue days they ran at around 10 Hz) and is
used to modulate a destination by oscillating the destination parameter around a
central value. I will run through a few examples of this and more using the audio
file:
drum beat4 [Link]
This drum beat runs at 120 bpm and this is an important piece of information to
have when using delays as delays can be crafter around the tempo of a piece.
When in arrange mode in your DAW you can sync the delay unit to tempo and
have the host control the sync values. The following is a standard time delay
using 120 bpm as the master tempo.

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Chapter3:Delay

drum beat4 3 qtr [Link]

This is a simple sync delay to tempo with no pans or crazy filter effects. It is a
nice way to liven up drum beats. By keeping the levels of the delays very low in
the mix of a song (whereby they are not heard) the listener feels a sense of
presence and movement.
In the next example we are going to get a lot more creative with the use of the
LFO, filter and pans.

drum beat4 lfo [Link]

If you look carefully at the image above you will see (main edit) that delay line 1
is assigned to the filter resonance and that the LFO is used to pan the delay as a
sine wave. The sine wave shape is the source for the pans and the shape itself
controls both the pan movement and filter frequency (in this instance it is a lowpass filter).
Its not too difficult to see where you can go with so many parameters and
functions at your disposal. The next example is a great way to create a
completely new sonic texture just by routing sources to control frequencies and
pans and so on.

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Chapter3:Delay

drum beat4 [Link]

The distortion has come into play here but very slightly. The real meat of this
effect is in the handling of the LFO and filter cut-off.
The next example, using the same delay unit, is a purely sound design effect as
opposed to a mix effect and entails the use of the LFO modulating the filter cutoff using a noise wave form and another LFO controlling the resonance using a
sine wave shape. The filter being used is a resonant filter.

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drum beat4 res [Link]

When using a decent delay unit like the Ohm Boyz you can create so many
textures that, as usual, the only limitation is your creativity.
Here are a few more examples of using delay creatively on a drum beat using
drum beat2 [Link]

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drum beat2 destroyed [Link]

This mad effects is achieved by using varying Tap, delay line and filter
modulation timings.
The following is even more extreme but timed to produce a more musical
pattern.

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Chapter3:Delay

drum beat2 [Link]

Nothing too complex here as we are, again, only dealing with a single Tap, delay
line and modulating the filter cut-off and resonance. However, by paying close
attention to the timing divisions we can express the beat in a multitude of ways
adding new textures and colours.
The next example is processed such that it sounds as if there are two underlying
beats playing at different tempos.

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Chapter3:Delay

drum beat2 different [Link]

By varying the timing divisions and modulating the frequency cut-off and
resonance with a ramp up waveform we can emulate different tempo structures.
Using ramp waveforms to modulate frequencies and timing divisions we can
create speeding up/down tempo effects or filter builds/drops and so on. The
waveform selection is always critical when making modulating decisions.
Let us end these particular examples with a complete mangling of the drum
beat.

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Chapter3:Delay

drum beat2 [Link]

Using the LFO ramp up waveform to modulate a host of destination, delay line,
frequency cut-off and resonance and Tap delay time we can almost eliminate
any trace of the original beat and replace it with a sound effect. If you look at
the images of the drum beat before and after you will note the extent of the
sonic mangling.
drum beat2 dry

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Chapter3:Delay
drum beat2 mangled

Delay is also very useful on single hits like a kick drum or snare.
big [Link]

big kick1 delay [Link]

A very simple effect with synced time difference between left and right channels
affording a nicely panned delay effect. With additional filtering and cross feeding
left into right channels and vice versa a new texture is created.
Another great way to change the texture and movement of a sound for creative
use is to work with the Tap (pre delays) and pan assign different time values for
them with different pan settings. Working with a wave modulator (LFO) to
control filter frequency spread across the pattern will give endless combinations

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Chapter3:Delay
of colour to a sound. Using the same kick sample and working with the following
settings we can achieve an alternating panned sound that can lend itself to
interesting sound design work.

big kick1 panned [Link]

Using all 4 Taps and varying their pan positions and levels coupled with a sine
wave LFO modulating the frequency of a LPF we can get a very interesting effect
that pans and abates over time.
rim snare [Link]

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Chapter3:Delay

rim snare [Link]

Again, using synced delay time differences with a lot of feedback (feeding the
output back into the input channel) gives the dub effect so commonly found on
mixes. The more the output is fed into the input the more dramatic the effect.
Delays are also very useful when you want to give both depth and space to a
sound without having to use reverb. The problem with reverb is that when used
on low frequencies it tends to smear the frequencies thus making it sound
mushy and undefined. Delay can be a solution in this instance. A good example
of using delay to give depth, movement and space without compromising the
low frequencies is to treat a bass line that has a lot of low frequency content.
acoustic bass [Link]
I decided to use an acoustic bass pluck as this is usually much harder to treat
than a synthetic bass sound.

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Chapter3:Delay

acoustic bass ultra [Link]

The trick is to use short delay times to create an echo type of effect but to rolloff the high frequencies. This allows the delay to thicken the sound without it
sounding like a doubled or chorused effect. I have used synced time variances
but this can make the delayed sound lie in a biased state (more towards one
side of the stereo field) as the time difference will be heard on the other
channel. This can be fine tuned with shorter delay times but for this example I
quite like the overall effect.
Percussive lines can also benefit greatly with the use of delays. A shaker loop
that is static can not only be livened up and widened but other patterns can be
created from it by adjusting left and right channel delay times.
shaker dry 80 [Link]

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Chapter3:Delay

shaker dry 80 bpm wide [Link]

Using very short synced timing divisions on each channel the delay adds width
and a little movement to the shaker line.
To take the above example further: changing the left and right divisions to
and respectively we can create a slight offbeat and add thickness and
movement to the shaker pattern
shaker dry 80 bpm delay [Link]
Using delay is not simply about repetitions but about shaping the repetitions to
create a more musical outcome. Sadly, most producers are not aware of the
power of delays and only use them to add a decay and space to a mix or
elements within a mix. But they can be used so creatively that it is a shame to
demote them to simple repetition duties. By using a modulator and having
control over the Taps we can get very creative with delays as has been covered
in this chapter.
In the next few examples I will use the piano [Link] sample and try to make
it a little more interesting by using a single delay engine and one Tap but
varying the modulator and filter with emphasis on the synced time variances.

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Chapter3:Delay

piano ohm std [Link]

The delay line is set beat division for the delay whilst the Tap delay is set at 0.
This allows for the nice rolling along effect. The filter frequency and resonance
are modulated by the sine wave LFOs.
The next example is a variation on the above and features the delay line set at 1
beat divisions for the delay, a resonant filter modulated (filter frequency and
resonance) by a triangle shaped LFO. Feedback is a little longer and the level of
the delay line is matched with the level of the pre delay.
piano ohm musical [Link]
Later on in this book we will cover more creative examples using multi effects.
Using delay alone is wonderful, as is with reverb or any other single effect, but
things really start to come alive when effects are used together either in series
or parallel to create complicated and evolving sonic scapes. Using modulators to
control different aspects of effects completely changes how we use and perceive
effects. The more control over an effect or effects the more creative and
satisfying the results.
In the next example we will create a WAH type of effect.

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Chapter3:Delay

piano [Link]
Using a single delay line and only one Tap we can get really creative. The trick
here is the timing of the LFO (period) and how it oscillates. Keeping the Tap at
division and the delay line also at division but changing the period of the
LFO to 1 to modulate the filter frequency and the period of the second LFO to
a whole beat division to control the resonance we can emulate the wah-wah type
of effect.
It doesnt end there either. We can emulate a reversed effect by adjusting the
gains of the Taps and delay and using tiny timing variances to fool the mind into
thinking the effect is actually reversed. This is called pseudo reverse. In the
next few examples I will create a reverse effect using division timing variances
and in the second example I will hone the settings a little more to create a more
musical effect that seems to be pseudo stereo but also moves dynamically.

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Chapter3:Delay

piano pseudo reverse

The delay line time is set at and the Tap delay at 1 whole division, the same
as the LFO sine wave modulator period.
In the next example the Tap and delay line times are the same but the LFO
period is different. This allows for a tempo change effect and a reverse effect
combined in one.

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piano tempo and reverse [Link]

Finally, here is a strange but really useful reverse effect using the delay unit and
I particularly like this one because it is both weird but strangely hypnotic.

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Chapter3:Delay

piano reverse [Link]

The Tap tempo delay (1 whole division) is modulated by the LFO using a ramp
up waveform (1 period). The delay line is at timing and modulated by the LFO
but using a sine waveform shape (1 period). The filter frequency is also
modulated by the LFO using a sine wave shape.
Feel free to experiment as much as you like as this really is the only way to burn
into your memory not only the techniques and principles related to delay units
but also the outcomes. I find a combination of technical data, images and audio
examples are the best way to remember a process.

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Chapter4:Chorus

Chapter 4 Chorus
With effects such as chorus, flanging, phasing and vibrato pitch modulation is
used along with delay. With chorus, which we will concentrate on for this
chapter, an equal mix of the wet and dry signal is used with the wet signal being
delayed and pitch modulated. The pitch is modulated using an LFO as the source
and both the depth and rate are used to create varying colours or textures of
the effect. The modulation depth is defined as intensity which is the range
between the maximum and minimum values. The rate determines the speed at
which these maximum to minimum shifts occur.
Too much depth when using chorus can sound as if the output is detuned
whereas too little depth can make it sound less and less pronounced. Rate can
have a dramatic effect on the chorus as slow rates create a more undulating and
smoother effect whereas fast rates will create a speeded up wobble effect.
The relationship between rate and depth is very important and it is about finding
a good balance between the two when processing sounds. The modulator shape
is also critical as the shape denotes the periodic (unless chosen otherwise)
cycling of the modulation destination. A sine wave shape will give a smoother
cyclic effect, and is the most common LFO shape for chorus and flanging, as
opposed to a square or pulse waveform which will have an extremely distinct
effect as it cyclically switches between two delay times.
Whereas flanging will take the output and feed it back into the input (feedback)
chorus does not adopt this process creating a far more subtle shimmering effect.
We will cover flanging in another chapter.
Chorus is often used on guitars, basses (mainly acoustic) and keyboard sounds
like electric pianos and so on. Although chorus can thicken and widen a sound it
can also push it into the background, so you need to be wary of the rate and
depth of modulation particularly when using it for these purposes as opposed to
using it for a specific colour. For this reason alone we rarely use chorus on
vocals although it can be used to thicken specific backing vocals, but to be
honest, double tracking, reverb and delay are better options.
Stereo chorus is even more interesting and dynamic and works by inverting the
polarity of the delayed signal and combining it with the dry signal in one channel
whilst keeping the polarity the same for the delayed and dry signals in the other
channel. The result is interesting in that one channel will have frequency peaks
whereas the other channel will have frequency dips.
Let us start with some basic examples of applying chorus and a good one to our
teeth into is an acoustic bass. First, I will use a delay plug-in and use the LFOs
to modulate the delay times which creates a sweeping up and down effect in
pitch.

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acoustic bass [Link]

acoustic bass ohm [Link]

The Tap delay times are modulated by LFO shaped waveforms with varying
periods. This gives the chorus effect and we can vary the timings for the
modulation to attain heavier or lighter chorus effects.
But the most common way to apply chorus is to simply use a dedicated chorus
plug-in/effect. Using the same bass sample let us use the Blue chorus plug-in
which is as simple as you can get in terms of features and control.

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acoustic bass blue [Link]

The Mix is a standard wet/dry fader and I have left it pretty even between the
two. Speed refers to the rate in other chorus units and is measured in frequency.
Depth refers to modulation depth (intensity). I have selected a short delay but
not too short so as to create a doubling effect but short enough not to have
another distinct follow on sound (echo or repeat). Width controls the stereo
width of the effect and can be achieved in a number of ways depending on the
effect unit being used.
In the next 2 examples I am going to cover the relationship between rate and
depth and I will exaggerate the effect so you can hear what a difference it
makes. I will use the same bass sample but will change the chorus plug-in to the
Kjaerhus Audio Classic Chorus which is free.

acoustic bass kjaerhus [Link]

Keeping a short delay time but using maximum rate and depth settings the bass
now sounds very wobbly and is detuning. The relationship between the rate and
depth values is very important in shaping a chorus effect. But by altering the
rate only to a slower value the sound starts to sound far smoother and more
musical although it still moves across the stereo field.

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acoustic bass kjaerhus slow [Link]

Slowing the rate down, whilst keeping the depth at maximum, has had a
dramatic impact on the chorus effect. It is good to always tailor the effect to the
sound using both your ears and some data values. Extreme settings are great
for creating specific effects or colours but subtle settings will afford you
smoother and more musical results.
Using different delay times, rates and depths we can create a whole host of
textures.
Using the following keyboard line electric keys [Link] we can have some fun
creating new colours.
electric keys [Link]

electric keys kjaer short [Link]

Using the tiniest delay time and mainly playing with the rate and depth we can
enhance the keys line to sound dynamic yet maintaining low frequency colour.

electric keys kjaer high [Link]

By keeping the same settings bar the rate, which has been increased, the chorus
effect is far more evident and yet still maintains the low frequency content
simply because it is doubling up the sound as opposed to separating it with a
higher delay time.

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electric keys kjaer higher [Link]

By selecting a higher delay time but keeping a low rate and depth the keys line
now sounds clearer and slightly doubled. However, the chorus effect is still
evident and sounds brighter due to the delay between the dry and wet mix.
Working with a brighter electric piano line (DX) we can add dynamic movement
and a glittering shimmer at the top end. I will use the Blue Chorus for this
particular example.

keys [Link]

keys dx1 blue [Link]

Working with average speed (rate) and depth but with a little more delay time
the DX line shines through and sounds vibrant and glossy.
Using the same DX line and Blue Chorus we can create a more detuned and
slightly heavier chorus effect.

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keys dx1 blue heavy [Link]

I have deliberately set a high depth and a faster speed (rate) with a slightly
longer delay time so as to attain an obvious chorus effect but with detuning
qualities which are evident to the point whereby some single notes sound out of
time with the main chords.
We can also use chorus to thicken, widen or deepen vocal lines.
I will use the experiment with [Link] sample and use different settings
to attain different results.
experiment with [Link]

experiment with reverb chorus [Link]

The real action here takes place with the delay time variance, almost no rate and
very little depth. By having the mix more towards the direct source and applying
the delay time the effect is that of a more wider/spread vocal.
We can even emulate a close reverb effect by using chorus.

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experiment with reverb chorus [Link]

Again, the trick here is to play with the delay time, fine and rate keeping low
depth values. And finally I would like to end with a flange example, which nicely
leads us to the next chapter.

experiment with reverb chorus [Link]

We will cover the subject of flange in the next chapter but I wanted you to get a
taste of what can be achieved with a simple chorus. The sound itself moves
nicely across the stereo field and sounds thicker and wider. Maximum depth
coupled with a healthy amount of delay but with negligible delay time creates
this type of effect.
Chorus is a very common and popular effect as it is extremely versatile on most
sound sources and can be used to create so many different textures that it has,
and will always be, one of the go to effects for producers.

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Chapter 5 - Flanging and Phasing

Flanging
A flanger is another modulation effect whereby the delayed signal, which is
added back to an equal amount of the dry signal, is modulated by using a LFO. If
you add the output back into the input (feedback) you get resonance with the
comb-filter effect.
Comb-filtering occurs when the delayed signal is combined with the dry direct
signal. The comb filter creates peaks and troughs in the frequency response. If
the polarity of the dry signal is the same as the delayed signal we call this
positive flanging and if the polarity of the delayed signal is opposite to the
polarity of the dry signal we call this negative flanging.
If the feedback is greater then you get what we call resonant flanging. The
more feedback applied the more resonant the effect. This is a bit like increasing
the resonance on a normal filter.
Feedback also has phase. If the feedback is in phase then it is called positive
phase. If the feedback is out of phase it is called negative feedback. Negative
feedback has odd harmonics whereas positive feedback has even harmonics.
Some flangers will have a phase parameter to control the negative and positive
phase and this has quite a dramatic effect on the overall effect. The best way of
explaining this is using examples. Additionally, the phase controls can also alter
the degree rates and therefore have a dramatic impact on the way the effect is
output.
Phase occurs when two signal are time shifted. In other words, one signal will
be offset by a specific amount to another. Basically, the phase is the progression
along the cycle of the waveform determined as degrees. A 360 degree phase is
one complete cycle of a waveform where 0 degrees is the start of the cycle. If
two signals are aligned exactly in time and location then they are deemed to be
in phase. If they are not then they are deemed to be out of phase.
Short of going into an epic debate about the subject of phase it is probably best
to explain what happens when signals are either in or out of phase. I will use a
basic sine wave in stereo and then show you what happens when it is in phase
and then 180 degrees out of phase (total phase cancellation).

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sine wave in [Link]

Both sine waves are in phase and aligned perfectly.

By moving the other sine wave by 180 degrees (inverting), which is halfway
along the cycle, we are able to see what happens when they are summed. This
waveform is now 180 degrees out of phase. That means it has moved 180
degrees (upside down or inverted). In other words the peaks of the cycle
coincide identically with the troughs of the other cycle. If I now sum these two
channels to one mono output I should get silence (cancel out). This is called
total phase cancellation.

Once the channels are summed you get the following; total phase cancellation.

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sine wave [Link]

Because total phase cancellation has taken place the result is simply silence.
If you vary the amount of phase by degrees you get partial phase cancellation
and so on whereby some frequencies are cancelled. This teaches us what
happens with phase and how varying amounts of it affect the result. In terms of
using phase in effects the results can be honed to taste depending on how much
and what type (positive/negative) are used.
One of the simplest and best flanger vsts is the Blue Flanger as it has a
dedicated Phase switch which at normal is positive and at inverted is negative.
Positive phase will impart a more metallic effect whereas negative (inverted)
phase will hollow out the sound and make it thin. Additionally, the amount of
phase can be altered to provide different textures to the effect. The modulation
is controlled by a LFO which can be set, in the case of the Blue Flanger, to be
either a sine waveform shape or a triangle waveform shape. This may all sound
a bit confusing but with the following examples I am hoping things will become
much clearer.
drum beat2 [Link]

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drum beat2 blue flange normal [Link]

This is a basic example of using a flanger with a 50 wet/dry mix, heavy feedback
so at to accentuate the effect but keeping the phase both normal and at 0
degrees. If you consider that 180 degree phase is total cancellation then 0
degrees denotes in phase. If I now invert the phase to a negative phase then
you will hear the hollow type of effect discussed earlier.

drum beat2 blue flange inverted [Link]

The flange effect is almost abated and the kick has hollowed out considerably.
Let us go for a heavy flange effect using the same drum line.

drum beat2 blue heavy [Link]

Using a faster speed for the modulator, which I have changed to a LFO sine

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wave shape, normal phase but at 180 degrees I am able to create the well
known but heavy flange effect.
Finally, let us use the Kjaerhus Flanger to create a dynamic flange effect over
time.

drum beat2 kjaer crazy [Link]

This effect has been created by using maximum depth and delay time variances
but keeping a wet/dry mix at 50%. As this is a dedicated flanger vst the
modulation is set and the timing variances are governed by the delay time and
rate.
Phasing
Phasing behaves in much the same way as chorus and flanging but with shorter
delay times and the effect itself is far more subtle. Because phasing works with
shorter delay times than chorus and flanging the impression is that there is only
one sound as opposed to two distinct sounds and because the LFO modulates
frequencies (peaks and trough, also known as notches) the filtering effect is
quite pleasing.
Unlike flangers which use delay lines phasers use all pass networks/stages
which are all pass filters and by adjusting the frequencies, and phase, of the
stages different phase effects can be created. By going back to the principles of
phase earlier in this chapter we can then see how certain frequencies cancel
each other out and therefore create different textures.
Phasers can be used to add swirls to sounds or to completely maul a sound into
something else. Because it is a subtle effect it doesnt mean you cant extreme it
for sound design purposes. As with all effects and dynamics it is up to the
individual to explore and experiment.
Let us start with a nice but static pad sound:
pad [Link]
We can now add dynamic movement to the pad sound simply by applying the
phaser.

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pad [Link]
Using 10 stages, very little feedback and low rate we can emulate movement
across the pad sound simply by altering the phase of the stages and processing
the phase relationships.
We can also move the effect across the stereo field by selecting the right
frequencies, notches and modulating using the LFO and in this instance set to a
triangle shape waveform. By using fewer stages the sound is not as dense and
evolving and comes across as far more subtle.

pad phaser [Link]

Keeping low frequencies, 4 stages for simplicity and sparser effect and adjusting
the notch intervals with a triangle LFO shape waveform we can create a nice
movement across the stereo field with a sweet swirling effect.
And just for fun let us create a nice little wobble effect using the same pad line.

pad phase [Link]

Lowest stage count plus a heavy feedback timed with a high rate gives us this
interesting wobble effect. A good way to see what happens in terms of notches

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and their intervals is to study the next two images which show the pad sound
dry and affected with the settings above.
pad dry waveform

pad wobble waveform

The notches and intervals are clearly visible in the wobble waveform. Of course,
this isnt just about the notches but also how the phase is displayed with specific
frequencies getting cancelled. The frequency response is very different to the
dry version.
Now we can start to have some fun using a keyboard sound:
electric keys [Link]

electric keys phase [Link]

Maximum feedback with 4 stages and filter adjustment creates this really nice
metallic effect. Lets make the keyboard cry!

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electric keys phase [Link]

If you compare the two images above, one for the weird and the other for the
cry, you will notice that the only two parameters that I have adjusted are the
stages and filtering frequencies. The effect is a sweet but wailing one.
And finally we will end with an example that displays one of the most common
uses for phasing; electric guitar.
tele 2 [Link]

tele 2 phased
A standard phasing effect that sounds more like tremolo than phasing. I am sure
you recognise this effect particularly from the earlier days when guitarists were
limited with the types of effects available to them. Low depth, no feedback but a
high rate and midway count of stages gives this distinct effect.
Flangers and phasers have many uses in both live and mix environments and
although I have a preference for phasers when it comes to washy types of
sounds or shimmering effects I do love to abuse the flanger for specific sound
design projects, but as with all effects and dynamics, experiment and abuse to
taste. However, both of these effects really come alive when used in conjunction
with other effects which we will cover in the final chapter.

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Chapter 6 Distortion
When we talk about distortion the image, invariably, conjured up is that of a
guitarist thrashing his guitar with acres of overdrive. In this chapter I am more
interested in covering harmonic and non harmonic distortion in subtle ways
using non linear systems rather than using a specific overdriven effect like guitar
distortion or a fuzz box etc.
In an analogue system overdriving is achieved by adding a lot of gain to a part
of the circuit path. This form of distortion is more commonly related to
overdriving a non linear device. But it doesnt end there as any form of alteration
made to audio being fed into a non linear device is regarded as distortion even
though the term is quite a loose one and not too helpful. The idea is to create
harmonic distortion and this is the area I want to explore in this chapter.
Harmonic distortion means that additional harmonics are added to the original
harmonics of the audio being fed. As all sound carries harmonic content, and this
is what defines its timbre, then it makes sense that any additional harmonics will
alter the sound quite dramatically. Harmonic distortion is musically related to the
original signal being treated and the sum of the added and original harmonics
make up the resultant harmonics. The level and relative amounts of the added
harmonics give the sound its character and for this we need to look at the two
main types of harmonic distortion: odd and even order harmonics. The exception
to this is digital distortion which sounds unpleasant and the reason for this is
that the digital distortion is not harmonically related to the original signal.
Harmonics are simply multiples of the fundamental frequency of a sound and the
addition of harmonics within a sound define the sound timbre and character.
Even order harmonics are even multiples of the source frequency (2, 4, 6, 8 etc)
and odd order harmonics (3, 5, 7, 9 etc) are multiples of the source frequency
(fundamental).
Even order harmonics (2, 4, 6 etc) tend to sound more musical and therefore
more natural and pleasing to the ear and higher levels of this can be used as the
ear still recognises the musical content. Odd order harmonics tend to sound a
little grittier, deeper and richer and higher levels of this cannot be used as
abundantly as even order harmonics as the ear recognises the non harmonic
content and it results in an unpleasant effect. But there are uses for both and
depending on how the harmonics are treated some wonderful results can be
achieved.
The following examples are very well represented using Christian Buddes free
Christortion plug-in. This plug-in simply excites the different harmonics of the
input signal and is a great tool for displaying the various processes whilst
providing an audio reference too.

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The idea is to have detailed control over the distortion by using the different
sliders for the harmonics. There are ten sliders. The first slider is a DC gain slider
and we wont be using that for now. The next slider controls the gain of the
fundamental frequency and the ensuing eight sliders control the gains of the
harmonics, and each slider has a phase invert function (switched on by using the
checkboxes). The GUI below the sliders displays the effect of the plug-in on a
sine wave and is used purely for referencing the shape. The graph in the
mountain on the right displays the harmonic shape and gain when a slider is
affected.
I will run through some simple examples using even and odd harmonics so you
can see and hear the differences. I will also include some mixed examples using
a combination of different harmonics. Where required I might phase invert as
this has a dramatic effect on the sound. I will start off by using the even and odd
harmonics without the fundamental so you can hear the dramatic differences
between them.
Let us start with:
electric keys [Link]

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Harmonic 1 odd

electric keys harm1 [Link]

A nasty odd harmonic that sound more atonal than anything else. No
fundamental is being used so that you can hear how the harmonics sound on
their own. The following example has 3 odd harmonics being used, harmonic 1,
harmonic 3 and harmonic 5.

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Harmonics 1, 3 and 5 odd

electric keys harm1 3 5 [Link]

The shape of the 3 odd harmonics are clearly evident in the display above and
the audio file represents a good result of the combination of the three odd
harmonics.
Now we will concentrate on the even harmonics.

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Harmonic 2 even

electric keys harm 2 [Link]

A far smoother sound and the harmonic display is as it should be, a sine wave
shape displayed as a 2nd harmonic even.
Harmonic 2, 4 and 6 even

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electric keys harm 2 4 6 [Link]

A combination of the even harmonics (2, 4 and 6) allows for a fuller sound but
maybe not as nice as we would like. The display shows the harmonics response.
The trick is to mix odd and even harmonics to attain the type of harmonic
distortion we want.
We can play around with different harmonics and use phase inversions to further
develop the sound, as is seen with the bass examples;
acoustic bass [Link]
In this example we change the bass sound and by applying harmonic distortion
with phase inversions we can dirty up the sound.

acoustic bass harmonics [Link]

A combination of odd and even harmonics, along with the fundamental, and with
select inversion the sound comes across as strong and dirty but still quite
musical and useful.
A little harmonic distortion works nicely on drum beats too.
drum beat2 [Link]

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drum beat2 harmonics [Link]

Using the harmonics with the fundamental and just varying the levels and
leaving phase alone we can add some nice distortion to the drum beat which
accents different frequencies and livens it up.
Emulating harmonic distortion with plug-ins is not as easy as it seems as there
are so many mechanical variables to take into account with a non linear system
that it would be almost impossible to replicate (algorithms) the process and
behaviour. However, that does not mean that plug-ins cannot be used for the
addition of harmonic distortion. They can, but they are predictable.
The two types of process I want to look at are re-amping, tube/valve (active
gains stages) and tape saturation.
Re-amping
Originally, guitar amps were used to drive the signal which would result in higher
(upper) harmonics being created. But because the guitar was then amped to a
speaker the frequency response of the speaker would filter out the higher
harmonics thus creating a more pleasing effect. The technique of re-recording
the speaker output back into the mix is called re-amping and this technique is
now emulated in software.
If a plug-in that is designed specifically for re-amping is not used and a
distortion plug-in is used instead to try to create the same effect then a low-pass
filter has to be used post distortion to filter and attenuate the upper harmonics.
With this technique in mind a very good plug-in I have found that works really

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well for re-amping is the Softube Vintage Amp Room which emulates various
amps and the re-recording of them.
clean tele [Link]
The beauty of the Vintage Amp plug-in is that the microphone can be moved to
create different re-amping textures as both distance and angles can be altered.
By using different types of speaker cabinets with different amps and mic
positions the choice of textures are endless.

clean tele 2 reamp [Link]

I have moved the microphone more central and the tone can clearly be heard as
warmer and fuzzier, all from a simple re-amping process.
By playing around with the various settings and mic position we can add low end
and presence and some lovely distorted undertones on the lower frequencies.

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clean tele 2 reamp [Link]

A warm yet distorted effect takes this electric guitar line into a different world of
dirtiness.
The next example will be out and out overdriven distortion.

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clean tele 2 reamp [Link]

Using a different speaker/amp combo and having the mic up close and personal
with adjustment on the bass and treble we can effect a really nice distortion
without too much brittle crunch. As a great number of speaker cabs (cabinets)
roll off the high end at about 4 kHz onwards the result is a lot more pleasing and
manageable.
We dont have to stay with guitar sounds. We can work with specific bass amps
and speaker combos and treat the low end.
bass [Link]

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bass reamp [Link]

The above plug-in is another lovely product from Softube and is specifically
designed for bass re-amping. Being able to mix between the D.I. (direct inject),
the amp/speaker and mic position we can get some incredible textures.
Additionally, the D.I. has basic tone controls and a limiter and armed with the
phase control this particular plug-in is extremely versatile.
In the above example you can hear the lovely bounce and depth of the bass line.
Using the same bass line we can create new and exciting colours.

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bass reamp [Link]

A really sweet and bouncy distorted bass effect using a blend of limiting and fat
tone.
We can also completely change the texture of the bass line and turn it into more
of an acoustic sound rather than a deep sub.

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bass reamp [Link]

By inverting the phase and adjusting the tone controls we can emulate an
acoustic texture using the same low end sub bass line.
Re-amping is a common technique and one that can reap benefits if you are
after a more organic sound. Much as using room ambient recordings as spatial
reverbs re-amping also captures both the devices and rooms.
Tube/valve (active gain stages)
An active gain stage is another term for an amplifier and the fact that the
amplifier raises the gain of the input signal means that there is an element of
distortion associated with the process and result. Valves, or tubes, fall into this
category. In this instance, the way active gain stages work are actually not that

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complex; the output is controlled by the power (via the power supply) that is
driven through the valve (grid) by the input signal. The higher the input signal
value the more power is driven through the valve via the power supply. This will
invariably account for variances, no matter how small, between the input and
output stage.
Tubes have always been associated with warmth because of this process but
dont let this fool you as some poorly designed systems work against you instead
of for you.
Generally, the most common valves used are triodes (12AX7) which are mainly
used for mic and line level gain stages, and pentodes which are mainly used for
amplifier stages. However, this is not set in stone as they can be alternated and
used in any system and is dependent on the topology design and application.
Triodes tend to produce both even and odd harmonics whereas pentodes tend to
produce odd harmonics. The choice of valve is important when dealing with
distortion as the harmonic content is reliant on the type used. Of course, the
design of both the circuit topology and powering comes into the equation but I
dont want this to be an epic journey into the world of electronics. I want to
simply explain the basic differences between the types of valves and how the
gain stages work as you will come across tube/valve selections in some vst plugins and it helps to know what you are dealing with and how the plug-in will
behave.
Suffice to say that running any audio into a gain stage device will create its own
character at the output stage and whether the choice comes down to using
valves, solid state etc is purely dependent on what works best. As this chapter is
about tube/valve I will stay with using valve emulation plug-ins and reserve solid
state for another day simply because we wont be using any solid state
processing. However, it is important to state that running audio through a solid
state device will add its own sonic character onto the output signal and this is
why it is common practice to run audio through pre amps and the like .
Running audio through valves can be a very subtle effect unless the valves are
driven.
Let us explore how valves behave when used both subtly and in anger.
Using the electric keys file let us run through a few examples of tube processing
using Nomad Factorys E-Tube/Tape Warmer plug-in.
electric keys [Link]

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electric keys nomad lf [Link]

A simple and useful plug-in the E-Tube/Tape Warmer has two 12AX7
valves/tubes and a tone control which adjusts the frequencies in the signal from
low to high. Using subtle amounts of both vales and working more at the low
end frequencies I am able to highlight the frequencies I want to emphasise and
this is how the warm and slightly fuzzy effect is achieved.

electric keys nomad hf tube [Link]

Driving both valves a little more and emphasising the higher frequencies still
affords a nice and pleasant texture but with more accurate low end.

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electric keys nomad tubes [Link]

Driving both valves even further but keeping the tone control flat you can clearly
hear the valve driven effect that is now bordering more on distortion. However,
driving the valves can be really useful particularly when working with low
frequency sounds although you do need to find a good balance between the
input and output stages.
bass [Link]

bass nomad tube [Link]

By using only one valve (Tube A) that is driven and working the lower

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frequencies and making sure to balance the input and output stages the bass
sound is now more fuzzy as it displays classic tube drive. It still sounds thick and
warm but has a lovely analogue colour to it.

acoustic bass nomad [Link]

With both valves set equally and gentle low frequency emphasis and careful
input/output balancing we can make an acoustic bass line sound really thick and
warm.
Another useful tube plug-in is the Voxengo Tube Amp. This plug-in works off the
same design concept as the Tube Warmer using a triode based valve with bias
(grid) and overdrive. This particular plug-in emulates the active stage found in a
pre amp and is mainly for dirtying sounds and adding subtle warmth.
I will use the drum beat2 [Link] and process it specifically for a darker and
bigger sound.

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drum beat2 voxengo dark [Link]

All I have done with this simple and basic plug-in is to up the drive to the input
signal and pushed it into a light tube bias but with heavy low-pass filtering and
balanced it with the output gain. The result is a dark and narrow effect but one
that can be extremely useful in attaining varying textures.
But, we can also attain a nice and fuzzy warm texture as demonstrated with the
next example using electric keys [Link]

electric keys voxengo tube [Link]

With almost no low-pass filtering, small bias and a little brisk with the input drive
we can get a nice, warm yet subtle effect which livens the keys line quite nicely.
Using tubes/valves can be both interesting and exciting as it is not just about
adding harmonic distortion or driving to get warmth, it is also about creating
new sonic textures to liven up existing sounds or mixes. I have, on occasion,
used valves to warm up a mix ever so slightly or strapped a decent tube vst
across the master bus of my DAW and mixed to the bus when trying to achieve
a special tonal colour.
This is a pleasing way to add texture to any sound and if you can get a grasp of
harmonics and how they behave then you can really shape your sounds
dramatically and with authority.
Tape Saturation and Tape Speed
Although tape saturation has been used for years it has now really come into its
own as an effect as so many people want the warmth of analogue in their digital
recordings. Basically, the digital age has made this form of distortion even more
attractive and sought after and many manufacturers now provide tape saturation
plug-ins.

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When driven hard tape responds with gentle distortion, and magnetic tape has
frequency dependent saturation. To understand what happens when driving into
analogue tape it helps to understand how tape behaves. Tape recording is
another non-linear process as there are so many factors governing the process
and result; record and playback heads, tape speed, tape width, tape qualities,
phase and high frequency bias and so on. In other words the process cannot be
linear with so many factors that contribute to both the process and result and
because of these; varying results are obtained even though all the necessary
record and playback criteria are met.
This, although sounding negative, can actually work in our favour and it is this
lack of linear response and inclusion of subtle inaccuracies that lends itself to the
desired process. Tape has the general characteristics of low frequency distortion
(harmonic) and irregular phase responses and if the signal is driven into the
magnetic tape then the dynamic range is compromised and the magnetic
saturation effect affects the high frequency content. The limit of analogue tape is
the saturation. Because tape is magnetic the more signal that is driven into the
system the more magnetization of the tapes magnetic content.
Driving beyond this limit means that the magnetic content, or magnetic
particles, are exhausted and therefore the reproduction of the driven signal
starts to exhibit the qualities stated above. When magnetic particles start to run
out saturation compression takes over as there arent enough magnetic particles
left to store a magnetic field. When the magnetic field in tape has not reached
the minimum threshold to be effective it is called hysteresis. Hysteresis can be
overcome by using Bias. Bias is the introduction of a high frequency, high
amplitude sine wave that is mixed in with the input signal prior to reaching the
record head. This then excites the magnetic particles to produce a stronger
magnetic field. If the magnetic field is too strong then not enough magnetic
particles are active in capturing the driven input signal and this results in
saturation. By altering the bias and hysteresis of tape we can affect different
playback and processing qualities and thus affect the saturation qualities.
Tape speed is another area that is critical in how sound is processed through the
record and playback heads. Controlling tape speed is a huge factor at the
playback stage as any variance will cause specific effects like wow and flutter.
Wow is a variation in pitch (caused by the playback process) over a very short
duration and rarely desirable but flutter, which is also a variation in pitch but
over a much shorter timeline, can be desirable. The flutter causes frequency
modulation which results in sidebands (if you modulate the frequency of one
signal with another the result will be a sum and difference of sidebands) which
are added to the original signal and these will be governed by the flutter rate.
The flutter effect is perceived as a thickness of the sound and therefore can be
very desirable.

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Lets start with some examples using another decent free plug-in called Ferox
which is a magnetic tape simulator.

electric keys ferox subtle [Link]

Using little saturation and a low hysteresis value the sound comes across as
fuller and warmer without the noticeable heavy harmonic distortion we have
experienced in other examples.

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electric keys ferox sat [Link]

With more saturation and no hysteresis the sound comes out fuller and
exhibiting some nice harmonic distortion.

electric keys ferox phase [Link]

Using the mix function whereby the treated signal and the direct signal are
output together, and heavy saturation and hysteresis, we can get a lovely
analogue phaser effect; lush and warm.

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electric keys ferox tp [Link]

By altering the feedback, tape speed and using heavy saturation we can emulate
the thick and lovely doubled effect.
With this plug-in my best suggestion is to play around as much as possible until
it parameter selections become second nature.
I would like to end with some examples using PSPs MixSaturator2 which has a
number of valve and tape options. To be honest this is one of my favourite plugins as it can be used so subtly that it is great for mastering, and aggressively to
create new sonic textures on just about anything.
Starting with:

drum beat2 [Link]

drum beat2 psp [Link]

The real power of this plug-in is the shape function as it scrolls between valve,
tape and digital curves. Using tape2 and a low level of saturation but working
with the frequencies we can create a slightly distorted low end but not an
aggressive one. All the frequencies are nicely represented but the low end has
had some excessive treatment which gives it a nice fuzzy but clean sound.

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drum beat2 psp [Link]

Selecting valve1 for the shape gives us a smoother curve and with some mixing
of both low and high frequencies we can get a smoother and more fluid sound.
But we can also drive the audio and create a harsher and more distorted yet
useable result. In the next example we will highlight the clap in the drum beat
and make it a little more aggressive.

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drum beat2 psp driven [Link]

Using a higher level of saturation and selecting tape1 for the curve plus boosting
the higher frequencies results in this more aggressive sound.
We can also drive and shape bass sound:
bass [Link]

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bass psp [Link]

Selecting valve3 as the shape, which is the most aggressive of the three valve
selection, with a healthy amount of saturation plus working only with low
frequencies we can get a really nice, thick, distorted sound. In the next example
we will treat the bass to sound far more rounded and warm with a nice bounce
to it.

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bass psp [Link]

Less saturation but 100% wet and low frequency management gives us a really
rich, warm and rounded sound. I will now process the electric keys [Link]
and warm it up and allow for some nicely coloured distortion.

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electric keys psp [Link]

Valve2 affords us a nice curve/shape and with lots of saturation but with a
balanced frequency spread we get this lovely, warm and fuzzy texture.
I will end this chapter with two more examples using PSPs Vintage Warmer2.
This plug-in contains a combination of limiting, compressing, valve/amps and
tape recorder emulations/processing and sounds lovely. As we have covered
most aspects of distortion I think it only right to include a couple of aggressive
examples of guitar treatment using the clean tele [Link].

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clean tele 2 psp vwrm2 [Link]

Some serious brick wall limiting with overdriving and altered tape speed gives us
this really heavy and driven guitar texture. But we can be gentler and simply
breathe life into a guitar line without having to go on an extreme excursion.

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clean tele 2 psp vwrm2 [Link]

Using the fastest tape speed, some limiting and overdriving yet balancing the
frequencies we can create a warmer and thicker effect that has more life in it.
I have really enjoyed writing this chapter as it has given me an opportunity to
try various plug-ins and have some great fun in the process. Once you get your
head around the technical terms, theories and processes it ultimately comes
down to one thing; HAVING FUN! So, I implore you to take this attitude with
anything you do that is music related otherwise it becomes a chore and not
playtime....

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Chapter 7 Filters
When we think of filters we think of equalisation and how the two are so easily
interchanged and linked to each other. If anything, this direct relationship has
led to a lot of confusion. I have covered the subject of equalisation in one of my
other books Sound Equalisation Tips and Tricks so I do not want to explore how
equalisation works and where filters feature in its makeup. What I do want to
concentrate on is the synthesizer type of filter and how it has now become an
incredibly useful tool and one of my favourites sound manipulation tools. I will
cover the basic types of filters but will go into a great deal of detail when coming
to using them in working examples.
Let us briefly look at the terminology used and what they mean along with the
most basic filter types.
Cut-off frequency
This is the point (frequency) at which the filter begins to filter (block or cut out).
The filter will lower the volume of the frequencies above or below the cut-off
frequency depending on the type of filter used.
Attenuation
This lowering of the volume of the frequencies, is called Attenuation. In the
case of a low-pass filter, the frequencies above the cut-off are attenuated. In the
case of a hi-pass filter, the frequencies below the cut-off are attenuated.
Resonance
Boosting the narrow band of frequencies at the cut-off point is called resonance.
Also known as Q and bandwidth, in effect, he higher the resonance, the
narrower the bandwidth.

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Q
Also known as width of the filter response, this is the centre frequency of
the bandwidth and is measured in Hz. Also know as bandwidth and resonance.
A high Q value denotes a narrow filter width (bandwidth). A low Q value denotes
a wide filter width (bandwidth).
This is actually a very important piece of information because with the Q control
alone you can make your audio sound high and brittle or warm and musical. This
does not mean that you must use low Q values all the time, in the hope of
attaining warmth, but you must understand what frequencies need filtering. If
your intent is to use EQ as a musical tool, then be aware of what the Q value can
do to audio. For creative EQ, this is a weapon often ignored.
Slope
The rate at which a high or low frequency EQ section reduces the level above or
below the cut-off frequency is termed as the Slope and the shape and
parameters are denoted as dB per octave and are usually: 6, 12, 18 or
24dB/octave. Slope also determines the characteristic of the filter and can range
from smooth to extreme (gentle to aggressive).
Pole
You will often come across the terms 2 pole or 4 pole. This refers to the number
of circuits filters used to attenuate the signal with each pole referring to a value
of 6db.

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Low pass filter

Hi pass filter

In the low-pass filter diagram the frequencies below the cut-off are allowed to
pass through whereas the frequencies above the cut-off are attenuated.
In the hi-pass filter diagram the frequencies below the cut-off are attenuated
and the frequencies above the cut-off are allowed to pass through.

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Band pass filter

This is a great filter. It attenuates frequencies below and above the cut-off and
leaves the frequencies at the cut-off. It is, in effect, a low-pass and a hi-pass
together. The great thing about this filter is that you can eliminate the lower and
higher frequencies and be left with a band of frequencies that you can then use
as either an effect, as in having that real mid-range type of old radio sound, or
use it for isolating a narrow band of frequencies in recordings that have too
much low and high-end.
Try this filter on synthesizer sounds and you will come up with some wacky
sounds. It really is a useful filter and if you can run more than one at a time,
and select different cut-offs for each one, then you will get even more
interesting results. Interestingly enough, band-pass filtering is used on formant
filters that you find on so many soft synths, plug-ins, synthesizers and samplers.
Notch Filter also know as Band Reject Filter
The inverse of a band-pass is the notch filter.
This is a very potent EQ/filter. It can home in on a single frequency band, and
cut/boost it. Used specifically for problem frequencies, the notch can be one of
the most useful filters. This is the exact opposite of the band-pass filter. It
allows frequencies below and above the cut-off and attenuates the frequencies
around the cut-off point.
In terms of the diagram shown for band-pass filtering, the area in between the
two arrows is rejected (cut out) as opposed to allowed to pass through and the
remaining frequencies below and above the cut-off are allowed to pass through.
This is the exact opposite of band-pass filtering.
In the early days of analogue synthesizers we were lucky we had a single lowpass filter and there was a good reason for this. The low-pass filter allowed the
fundamental frequency to pass through unless the filter was closed completely

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(cut-off taken down to 0). For this reason alone it was the most commonly used
filter.
For the purposes of this book we do not need to look beyond the filter types and
how they behave. However, as this book is about effects, we will explore how a
filter can be used as an effect as opposed to simply filtering frequencies and the
best way to do this is to use modulators to affect the varying filter parameters.
The most basic form of affecting a filter is to use an envelope to shape the
filters response. An envelope denotes a shape and the most common envelope
parameters are ADSR (attack, decay, sustain and release); please reference the
chapter for modulators for more information on ADSR. Using an envelope and
targeting the filter we can shape the behaviour of the filter over time. The
following example uses a simple ADSR to shape filter1.
Starting with a simple waveform from Strobe it is easy to demonstrate how the
filter opens using an ADSR (envelope) and in this case it is the mod env
(modulation envelope) much as we did in the chapter on modulation.

strobe tone [Link]

A combination of saw and sine waveforms are run through the filter with no
modulation of any sort. The next step is to assign the mod env to modulate the
filter cut-off.

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strobe tone mod env cut [Link]

The mod env is shaped to start with a slow attack and the filter cut-off responds
to the shape of the envelope. Although a similar example has been covered in
the modulation chapter I thought it prudent to create a similar example because
the next step is to modulate the resonance and the two examples together
should help in understanding the workflow.

strobe tone [Link]

The mod env is not modulating the filter resonance and this can be heard quite
distinctly.

strobe tone res [Link]

The LFO is set to a sine shape and it is used to modulate the filter resonance.
Quite a distinct sound.
That covers the basic envelope modulators for filter cut-off and resonance. Let
us now take this into a sound design concept and have more dynamic movement
and variety.
pad [Link]
A nice pad without any filter shaping that goes through its natural progression

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using basic effects. Now, let us reshape the filter behaviour using the filter1
ADSR of a wonderful vsti called Albino made by Rob Papen.

pad filter [Link]

The filter stays closed and then opens up to allow the filter characteristics to
pass through. This gives the sound a nice swelling texture. The area we are
concerned with is the filter envelope on the left of the GUI. Here is the envelope
parameter in isolation.

A long attack value with a full decay and sustain and a midway release gives this
wonderful swelling effect. Because the attack starts later this means that the
filter opens up later. Using envelopes to shape filters is one of the most useful
ways of creating movement. We can also create the same effect opening and
then closing just by using the filter ADSR.

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pad filter adsr open [Link]

By removing the sustain and shortening the decay time we have the filter
opening for a short period, then closing as the decay leads into a non-existent
sustain.
Lets do one more example using the filter ADSR to shape the filter on pink
noise.
pink [Link]

pink noise filt [Link]

I have included the whole GUI here so you can evaluate the filter selection and
the ADSR. The filter is a low-pass 24dB with the cut-off set at 0 with no tracking
or resonance and the filter is wholly controlled by the filter1 ADSR; the env
(envelope) function controls the mix of the audio going through directly and
using the filter ADSR envelope, a bit like a wet/dry mix control. I have it set to

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completely use the filter1 ADSR envelope. We can take this a step further and
have the ADSR sweep the filter resonance as well as opening the filter.

pink noise filt [Link]

The resonance is now being affected by the filter1 ADSR and using the same
settings as the previous example the time taken for the filter to open and the
resonance to be swept is governed by the attack parameter of the filter1 ADSR.
However, why end there? Lets assign a LFO to modulate the filter panning and
assign another modulator to control the rate/speed of the LFO. Basically, this
means that the frequency of the LFO will determine the speed of the filter
panning and by having another modulator, in this case filter2 env, control the
LFO rate we can create a really interesting effect whereby the filter pan will
progressively speed up and pan faster and faster.

The image above shows the settings for LFO1 using a triangle shaped LFO with a
peak frequency setting that will be achieved when the filter2 ADSR sweeps
through the frequencies by modulating the LFO rate.

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The filter 2 envelope (F2) shows a long attack value and the modulation matrix
on the right shows how the source and destinations have been assigned with
LFO1 controlling the filter1 panning but with the filter2 envelope controlling the
rate/speed of LFO1. The filter2 envelope gradually speeds up the LFO speed
which in turn speeds up the filter panning.
pink noise filt pan [Link]
With the next example I am going to load up one of the presets from my Dark to
Light sound bank I created for Albino. The filter pans are controlled by two
different LFOs and the filter resonance is controlled by velocity. Additionally, the
filter is set to a band pass and a mix of both the direct signal and filter1
envelope shape the filter response.

albino fil res and [Link]

Although not complicated, the use of source modulators controlling
predominantly the filter cut-off and resonance we are able to create a really
dynamic and ambient texture. It is the selection of the band pass filter that gives
this particular sound its very resonant edge.
By layering two sounds and using different modulators to control varying filter
and pitch destinations we can create large evolving sounds. I will use only the
filter and mod screen captures for the next example otherwise I will end up with
page after page of routings.

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Evolving pad layer1

Evolving pad layer2

evolving pad [Link]

LFOs have been used for filter panning, filter resonance, oscillator pitches and
symmetries. By varying the LFO shapes and frequencies, and creating filter
envelopes, we can control varying destination parameters to create a complex
evolving sound over time.
It is now time to get a little more adventurous and use a dedicated filter device
and use different types of modulators to control varying parameters on the filter.
Using drum beat2 [Link] and the Blue Filter we can assign a LFO to
modulate the filter frequency and create some dramatic and interesting effects.

drum beat2 bl lfo [Link]

If you look at the image above you will see that 4 poles have been selected with
an extremely high slope, and a triangle shaped LFO has been used as the source
modulator with freq (frequency) as the destination, and because the mix is set
to fully wet (filter) we get that lovely squelchy and bouncy effect. This is a nice
and simple vst that can be used to modulate a filter but I like to go even more

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extreme and for the purposes of serious mangling I cannot think of a better vst
than Camel Audios Camel Phat3.
Lets go a little extreme here and use a source modulator, in this case a square
shaped LFO, to modulate the cut-off frequency of a low-pass filter.

drum beat2 camel mad [Link]

The lower middle section of the GUI shows that the LFO is switched on and the
shape is a square and the target is the MMFilterCutoff (multi-mode filter cutoff). On the right of that is the multi-mode filter and I have selected a low-pass
filter with an extremely low cut-off value with very high resonance and you can
see the basic envelope follower (env) that is reliant on the attack and release
values when fine tuning the incoming audio signal.
The env knob is an amount/intensity knob (the more you turn it clockwise the
higher the env amount/intensity). The mix has been set to fully wet. The sound
has completely changed and the loop information has been smeared by the LFO
rate and depth amounts. Truly mad, truly lovely.
In the following example I am going to stay with the same drum loop but thin it
our using a high-pass filter but with the right modulation settings I can take this
a step further and make it musical and the real secret here is in the treating of
the kick drum and that has everything to do with timing.

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drum beat2 camel [Link]

This time a triangle shape LFO is used to modulate the high-pass filters cut-off.
Leaving the attack and release values short with a small amount of env we can
get the drum loop to sound bouncy and squashed. The LFO depth and rate are
relatively high but not high enough to cause complete timing mayhem.
How about routing the LFO to modulate the env amount whilst using a band pass
filter? This will make for an interesting sound.

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drum beat2 env amt [Link]

The ramp down shape for the LFO gives us a nice instance of a modulator going
downhill for use of a better phrase. This allows us to have varying textures over
time which then repeats itself when retriggered. The band pass filter can be
made to sound very resonant if the bandwidth is made smaller and with the
envelope follower being pushed and controlled by the ramp down LFO we are
guaranteed to have some very interesting results.
We can use filters on evolving sounds or static sound. In the next example I will
reshape a kick drum using both band pass and notch filters.
big [Link]

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big kick1 camel [Link]

The LFO shape is a ramp up (think of going uphill), which is the opposite of a
ramp down and this is routed to control the band pass filters low filter cut-off.
The multi-mode filter is also activated with a very low cut-off and high
resonance. The two together result in this lovely and greasy drum sound.
No example would be complete without processing a snare.
acu snare [Link]

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acu snare camel [Link]

Using a ramp down shaped LFO routed to the multi-mode filter cut-off with
moderate settings for env, resonance and cut-off we can create this nice funky
slap down type of effect.
Using filter modulation on any sound can work wonders and can offer a huge
range of different textures. Using the acoustic bass [Link] sample we can
completely change the tonal quality and afford us a new texture to use in our
productions.

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acoustic bass camel [Link]

Using both filters, band pass and low-pass, we can shape and modulate the lowpass filter cut-off while running the signal through the band pass. Triangle
shaped LFO works really well with this sound and has changed the bass into
another very usable and distinct texture; a great way to synthesize an acoustic
sound into a more synthetic one.
Using the same bass sound we can manipulate it even further.

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acoustic bass camel [Link]

Using a very low rate for the sine shaped LFO we can make the bass wobble and
by driving the resonance on the band pass filter and coupling it with the
modulation over the high-pass filter cut-off we get the aggressive and fat bass
texture.
Lets try another example by reshaping the electric [Link] line into a
completely different colour.

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electric keys camel [Link]

The settings speak for themselves but what is important is the use of a sine
shaped LFO because it is this smooth cyclic shape that makes the tremolo effect
sound so fluid.
I will end this chapter with a few more preset examples taken from my Albino
sound bank as they provide a good insight into what can be achieved with
modulators controlling filter cut-offs, pans and resonances.

albino filter [Link]

Only one filter is being used here, band pass, but the real heart of this preset is

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the use of LFOs to modulate oscillator pitch, filter cut-off and resonance, and a
single LFO used to control the rate/speed of another LFO. The result is a
dynamic sound effect.
I mentioned earlier that anything, within reason, could be used as a source
modulator and one of the most potent is an arpeggiator. An arpeggiator is the
equivalent of a sequence of notes, chords or modulator being played back at a
defined speed and division. Think of it as a short repeated sequence.
The following are a selection of arpeggiator pre3sets that have the arpeggiator
modulating filter cut-offs, resonances, pitch and so on. I wont list each and
every one but provide a screen shot of the mod matrix so you can ascertain
what is routed where. Never ignore the power of an arpeggiator as anything that
can be used as a source modulator that varies over time has to be a really useful
tool.
Here is a nice little scratch effect using only noise and a simple mod matrix
where two filters are used and two arpeggiators are routed to control the cutoffs of each at varying time divisions. Additionally, two LFOs are used to
modulate the filter pans.

albino filter [Link]

Here is another that uses an arpeggiator to control filter cut-off. The use of
additional LFOs to control pans makes this an interesting repeated sequence.

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albino filter [Link]

We can also create interesting sweet sequences by assigning the arpeggiator to
modulate the cut-off at varying values.

albino swt [Link]

The next example is actually a cute variation of a chord using one LFO to control
the filter pan and the arpeggiator to control the filter cut-off and symmetry.

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albino weeping [Link]

Lets end with a throbbing synth line that uses the arpeggiator to control filter
cut-off and symmetry.

albino throb [Link]

Filters and their modulation have been firm favourites with sound designers for a
long time and with the advent of dedicated filter devices the realms of
production have now embraced this wonderful process.

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Vocoder
Vocoders have been around for a long time and there has now been a
resurgence of this effect/process. I was in two minds as to which chapter to put
the vocoder section in but came to the conclusion that as we are dealing with
filters we might as well go the whole way and include vocoders as an effect
within this chapter.
A vocoder allows the sound character of the source (modulator) to affect a
destination sound (carrier). In other words, the characteristics of one sound are
used to modulate the characteristics of another sound. The modulator is fed
through a bank of filters that analyse the response and frequency characteristics
and this in turn controls the level of carrier signal which is also fed through a
bank of filters (the level of signal going into each filter is controlled by the
modulator). The more filter bands the more control and definition over the
signal.
The classic case of using a vocoder is to use a voice (fed through a microphone)
to modulate the carrier which can be anything that has a nice sustained
character and pad sounds came top of the list. However, today we use vocoders
with varying modulators and carriers and create some of the most innovative
and interesting sounds.
For the next few examples I will use vocal samples and modulate them with
Midi. I would like to take this opportunity to thank Tanikye and GAM for use of a
variety of vocal lines sung by Tanikye.
tanikye vocals [Link]
A not too complex vocal line sung dry.

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Using Cubase SX3 as the DAW and routing the vocoder on the Midi output
channel and using it as an insert on the audio channel where the vocal sits we
can modulate the vocal line by using Midi; in this instance I am playing two
minor chords and have used the standard vocoder in Cubase set to play 7ths.
vocoder [Link]
Using the various filters and assigning specific parameters to further modulate
the carrier we can create really evolving sounds and textures. By using the
minor chords to modulate the carrier we can literally change the harmonic
content of the carrier.
We can also modulate the carrier using an audio file. We are not limited to just
trigger by Midi. I will use a drum beat to modulate a pad sound to create a
completely new texture, and one that I quite like as it has kept the vocal aspect
of the filters.
We will use the following two audio files:
vocoder drum beat 140 [Link]
vocoder pad 140 [Link]

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vocoder audio to [Link]

The drum beat is the modulator and the pad sound is the carrier. You can still
hear some of the drum elements and the varying triggered levels that go into
the filter bank are quite evident.
Using the same setup and playing around with the vocoder parameters we are
able to create a more musical line without the drums being evident.

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vocoder audio to audio [Link]

Extending the bandwidth and setting minimum and maximum frequency ranges
and using the noise parameter we can create this lovely harmonic texture. If
look at the bottom right of the GUI you will see that I have selected 10 filter
bands to play with. I will use one more example of this setup but alter the noise
parameter and drop down to 8 filter bands and you will notice the difference in
texture and definition.

vocoder audio to audio [Link]

A slightly less detailed but meatier texture that sounds as if it could sit behind a
beat and add some colour to a mix.
In the following example we will use the vocal line to modulate the pad line
(carrier) and apply a lot of filtering with the existing reverb to create a
whispering effect.

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vocoder audio to audio [Link]

A really filtered and breathy effect by having the vocal line modulate the carrier
which is a pad sound. The use of 24 bands of filters makes all the difference here
as more detail and warmth can be applied as there are more filters to use. The
bandwidth and frequency ranges give us additional control over the behaviour of
the modulator and carrier.
It is also possible to have both the modulator and carrier signals playing
together by using the talk thru function. Using the following as the carrier and
modulating it with Midi:

vocoder bass [Link]

A 94 BPM bass line with no effects applied.

vocoder talk [Link]

Dirty yet dynamic and this is achieved with selecting the desired min and max
frequencies whilst mixing it with the talk thru function.
The final example uses the more traditional vocoder sound but is mixed with the
original to allow for a slightly thicker and more interesting texture.

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vocoder talk thru [Link]

This is a little more like the talk box type of effect and features a narrower
frequency range but with a full bandwidth. The talk thru allows the mixing of the
modulator signal with the carrier and a little high thru brings in the more raspy
flavour.
This chapter has dealt with the subject of filters and their manipulations using
varying source modulators but nothing is more educational and fun than you
exploring this wonderful process and trying different source modulators to
varying destinations.

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Chapter 8 - Pitch Manipulation

Although pitch shifting and pitch correction fall into the corrective realms of
production it is true to say that they have become creative and musical tools. To
be honest we have been manipulating pitch for years in sound design and in
production techniques; from detuning and layering sounds to thicken and
dynamically enhance them to altering pitch in real-time to create a specific
effect.
However, modulating pitch has now become a craft in itself and we have come a
long way from simple sonic shaping to creating a completely new form of music
for search of a better word. And the best example of this is the treatment of
vocals into a new musical texture that has now become synonymous with so
many commercial hits.
Starting with the most common and simple of the pitch manipulation processes:
Pitch shifting/bending
This process involves moving the pitch of a sound to another pitch by semitones
or octaves. The audio signal is dissected into small slices. If the pitch is to be
decreased (pitching down) then the slice/s are lengthened and vice versa for
pitching up. The accuracy of the process is dependent on a number of factors
and is governed by how well the algorithms are coded. If the coding is poor the
pitch shifting will sound unnatural and grainy. Huge pitch shifts often result in
unnatural results and sound so. It is often best to stick to small pitch shifts and
working in semitones, or cents, yields more fluid results. Pitch shifting in realtime is the equivalent of pitch bending as the pitch travel is defined by a
modulator.
Using the following vocal file, tanikye [Link], we can experiment with
pitch shifting and bending.
tanikye [Link]
Using Sound Forges Pitch Shifting effect we can shift the vocal up by a fifth.

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tanikye vocals pitch up [Link]

That is as simple a pitch shifting example as you will ever get; a simple pitching
up to create a 5th. Of course, we are not too concerned about the Mickey Mouse
sounding result and the real interest is in how well the effect is processed. In
this case it is not too bad and doesnt suffer from too much graininess.
Pitch shifting can also me modulated, as we covered earlier in the chapter on
modulators, both in real-time and with a modulator, but its most common use is
to shift the pitch of a signal to another pitch. Manipulation is far more interesting
and pitch bending is a very common process and I am sure you have used it
many times when using the pitch wheel on your keyboard/synthesizer to alter
the pitch of a sound in real-time. The following is an example of pitch bending
up and down using Albino with the bend range set to one octave (12 semitones).

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albino pitch bend [Link]

The image above shows the pitch bend range set at one octave (12 semitones)
and the audio example shows how the bending is performed; fluid and dynamic.
Earlier, we used LFOs to manipulate pitch. Let us do the same with the same pad
sound but this time we will shape the LFO to exhibit a more sound effects type
of colour over the sound.

albino lfo3 [Link]

The LFO is sine shaped and at a very low frequency so it can modulate the pitch

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(mod matrix) up and then down and so on using its cyclic qualities. The pad
sound has now turned into a siren type of sound.
We can also pitch shift using an envelope. I will be using Sound Forges pitch
bending envelope generator to create a crazy envelope to manipulate the pitch
across 2 octaves (1 octave up and 1 octave down).

tanikye vocals pitch [Link]

The envelope shows how the pitch drops then rises then drops and so on,
creating this crazy pitch modulation. Of course, I am using extreme shapes to
show what can be done but with sensible shaping some excellent and musical
textures can be achieved and we will cover these techniques using Antares Auto
Tune.
A variation of using an envelope to modulate pitch is as follows; using the same
pad sound and the modulation envelope in Albino to modulate the main pitch.

albino mod env main [Link]

The mod env is displayed on the left hand side of the GUI with the shape
denoting a very long attack, decay and sustain, with a midway release. You can
hear how the envelope controls the main pitch (mod matrix on the right shows

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the routing to pitch over 2 octaves) and because of the shape of the envelope
the pitch rise is gradual and fluid.
The next example uses an arpeggiator to control the main pitch of a sound.
albino dry [Link]
A drum sound created in Albino playing back a sequence. However, when we
assign the arpeggiator to control the main pitch and over 2 octaves down then
we get an interesting colour.

albino arp pitch [Link]

The arpeggiator is modulating the main pitch with additional modulation from
various envelopes to create the drum sound from the existing waveforms and I
have also routed the modulation wheel to further control the main pitch. This
allows for some interesting variations and textures.
Pitch correction/tuning correction
The difference with tuning correction, as opposed to pitch shifting, is that the
pitch of the audio is analysed and then compared to a pre defined scale, be it
chromatic or any user defined scale. In other words, there is now a reference to
tune the incoming audio to.
The more extreme the referencing scale the less accurate the analysis. However,
this can actually be a good thing as we now have famous tuning effects that are
featured across so many commercial releases and the creation of these extreme
tuning effects has become, dare I say it, an art form in itself. Some may
disagree with me but as far as I am concerned anything that can be used to
change the sonic texture of a sound is another tool that can be used creatively.
Although in the next chapter we will be using effects as creative tools, and
tuning correction will be featured heavily, I would rather stick with corrective
measures for this chapter whilst delving into tuning practices, and for the
following examples we will explore the basic features of Antares Autotune which
has now become synonymous with the Cher or T-Pain effect.

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Let us start with an edit of the Tanikye vocal and correct the odd tuning
problem.
tanikye vocal [Link]
Let us use Autotune to analyze the incoming pitch data and reference it to a C
Major scale.

Let me very briefly explain the front end of the GUI relevant to the ensuing
exercises. The input type selects the type of voice or instrument needing to be
processed and in this instance, as it is a female vocal, I have selected Soprano.
Tracking has been left at default. Key and scale are the referencing scales. If the
vocals were sung in A Minor then I would have selected A as the Key and Minor
as the scale.
However, as this particular Tanikye vocal is sung in C Major I have selected C as
the Key and Major as the Scale in the central area of the GUI. I can have the
Key as C and leave the Scale as chromatic and then define it further in the
central area or select it at source. Using it in the centre area allows me to
toggle between different settings. I have left the Vibrato area alone as we dont
need it for these exercises. Everything else has been left at default as the
current settings are good enough for the type of correction we need.
As Tanikyes vocals rarely waver off pitch these subtle settings are all we need to
correct certain notes etc. Currently, the Correction Mode is set to Auto.
However, we will now analyse and record the vocal and edit it in Graphical Mode.

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As you can see, the audio has now been analysed and recorded with the pitch
being tracked. The centre white area denotes the audio waveform and the little
squiggles denote the pitch information (zoomed in below). This area is the Pitch
Graph display with pitch being displayed by the vertical axis and time being
represented by the horizontal axis.

The next step is to import the Auto Correction mode settings from the front end
and this will show us what corrections Autotune has found.

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There are now two sets of line squiggles and one is the detected incoming pitch
data and the other is the corrected data that Autotune has provided through
the front end settings analysis.
Now that we have these two useful bits of information we can process them in a
number of ways available to us via the Tools section. I am not going to provide
Autotune tutorials here but rather delve in and explain how the tuning process is
accomplished and how we can manipulate the data to be either corrective or
creative.
Autotune has done its best to correct the audio but there are a few anomalies
that need manual tuning. By doing this ourselves it will allow you to see the
processes and power of this software as it is with manual tuning that we can
explore some crazy effects. This will be covered in the next chapter but for now
let us use it as a corrective tool.

Lets concentrate on the grey area I have highlighted as the first port of call for
correcting some wayward notes. I will use the Make Note tool and realign some
of the wayward notes to the nearest semitones as Autotune has
sharpened/flattened some notes too much.

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The boxes you see are areas I have selected to correct using the Make Note
tool. We can go along the audio file and make these types of corrections until we
are happy with the overall result.
tanikye vocal [Link]
The corrections are quite subtle but evident as they move along without too
many anomalies.
I will end with one more vocal correction and leave a note or two for you to play
with using a double tracked vocal.
tanikye vocal [Link]

tanikye vocal bv [Link]

The bv vocals sound a lot smoother and more fluid.

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Time Stretching (compression/expansion)
Time stretching is the process used to alter the speed or length (duration) of an
audio file without altering its pitch; in effect, the opposite of pitch shifting. The
process involves adding or subtracting slices to make the audio length longer or
shorter. Time compression shortens the audio length and time stretching
lengthens it.
This is an extremely useful tool as it allows us to mix and match varying tempos
of audio files to the same tempo without having to alter their pitches and is one
of the favoured tools when creating mashups (where two songs are mixed
together to create a new remix). Of course, the better the algorithms used for
the process the better the time stretching result. Additionally, when we come to
the point whereby time stretching starts to break down as a process we start to
enter the realms of audio mangling.
Lets run through a few examples using Propellerheads Recycle which is a slicing
(chopping) tool that also has some really useful processing features of which
time stretching is one of the more notable ones. We will use a basic 80 BPM
drum beat.
bifta 80 bpm [Link]

bifta 80 bpm drums [Link]

Although Recycle is a slicing software it has a very usable time stretching
feature. By selecting the original BPM of the audio followed by setting the
number of bars used for the audio length and selecting enough slices so the
software can perform the time stretching adequately we can then select an

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output BPM (in this case 140 BPM). The result is not too bad at all. Recycle is
actually quite good at time stretching and if you select the right global
parameters you can create new time based textures (we will cover these in the
next chapter). To show how well Recycle can time stretch lets go even further
with the time stretching and push the output BPM to 255 BPM which would break
down on most software.

bifta 80 bpm drums 255 [Link]

Even at 255 BPM the audio file and time stretching process has not broken
down. Some softwares are only good with time stretched BPMs matched closer
to the original tempo. Others perform extremely well especially if a good code is
used as in the case of Wavelab whereby the Dirac processor takes over and
provides a higher quality of stretching.

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bifta 80 bpm drums 140 bpm [Link]

I have used pretty much the same settings as in Recycle (bars, original tempo,
output tempo etc) and the result is very good.
We can, of course, go the other way and slow down the output tempo.
not sure 105 bpm [Link]

not sure 105 bpm drums [Link]

In this instance we have slowed down the original tempo from 105 BPM to 80
BPM and the result is still very good.

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I mentioned earlier that some softwares cannot time stretch too well and
therefore you end up with anomalies. But what if we wanted to keep those
anomalies in the context of sound design? Or, what if we used extreme settings
to create some crazy effects?
Using tanikye [Link] and Cubase SX3s time stretching function we can go
wild with the settings and create some interesting new textures. For the
purposes of file sizes I will truncate the audio files so that you dont end up with
a huge file that drags on forever.

cubase tanikye time [Link]

What we have done here is to stretch 17 odd bars of the vocal line to fir into 55
odd bars; that equates, according to the software, a 383% stretch. I have used
the real-time algorithm and selected vocals as the preset so that I can get the
best possible result. Algorithm selection is as important, if not the most
important, as any other parameter selection.
Let us now time stretch/compress the same vocal line from 17 odd bars into 4
bars.

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cubase tanikye time [Link]

You can hear how extreme this now sounds with crazy warbling everywhere.
I hope this chapter has conveyed how important and fun pitch manipulation can
be, be it for corrective or creative uses. In the final chapter we will incorporate
these processes to create more colourful and interesting textures

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Chapter 9 - Creative Effects

This final chapter is all about the creative process using effects. All the preceding
chapters have been break downs of the effects that will be used in this chapter,
plus a few more bonus effects that didnt warrant entire chapters to explain
their usage.
To be creative with effects we need to have the power to chain effects together
in series or in parallel. Feeding one effect into another is a great way to morph
and mangle sounds. Using two or more simultaneous effects in parallel also
affords us huge scope for sonic manipulation. Using more than one effect, be it
in series or parallel, is termed as a multi effect.
We will cover lots of examples using different effects in series and in parallel and
explore the differences between the two and what types of results can be
achieved.
To be honest, I have really been looking forward to this chapter as it allows me
to share some tried and tested processes with you but, more importantly, it also
allows me to explore some new creative processes that have come about from
experimentation. I hope that by sharing this obsession with you it will allow you
to see effects in a different light and not just as a corrective or standard tool
when mixing but as a sound designers pool of tools that can be explored and
exhausted.
There are many ways of chaining effects together and this usually involves
inserting one effect after another in the DAW or using auxiliary sends/returns to
access additional effect like reverbs. However, I am prone to using specific vstis
that have a comprehensive effects section that can be used within the vsti but
even better is a modular system whereby you can slot effects into any order that
suits you; a bit like inserting effects on channel inserts and the routing them to
each other. The best modular system I have come across is Native Instruments
Reaktor.
But before we start with the more interesting multi effects let us cover a cover of
examples of running effects in series and in parallel.

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filters in [Link]
The image above displays the oscillators used and the filters adopted in series. I
have duplicated the oscillators using identical settings but with oscillator 1
assigned to filter 1 and oscillator 2 assigned to filter 2. Filter 1 is using a lowpass filter at 24 dB slope and filter 2 is using a comb filter. The two oscillators
playing together are using filters 1 and 2 in parallel. In other words, filter 1 acts
separately to filter 2 and vice versa and both filtered sounds can be heard sitting
one on top of the other.
The following is the same setup but the filters are in series.

filters in [Link]
If you look at the top right hand section where the Amp parameters lie you will
see that I have routed filter 1 to filter 2 (F2). I have also routed oscillator 2 to
filter 1 (F1). The two oscillators now feed into filter 1 which in turn feeds into
filter 2. This is called in series. The filter settings of filter 1 shape the sound of
the two oscillators. This in turn is then filtered again by filter 2 using the result
of filter 1. This example is, of course, unexciting but it does show how the sound
changes when using filters in parallel and in series.
This time I will use the same oscillators and run them both parallel and in series.
I will alter the filter attack in the filter envelope so that filter 1 climbs to the filter
cut-off slowly. I will be using a low-pass filter for filter 1 and a comb filter for
filter 2.

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filters in parallel fil [Link]

Oscillator 1 is going into filter 1 which is a low-pass filter. Oscillator 2 is going
into filter 2 which is a comb filter. The waveform displays how the filter 1
envelope has been shaped with a later attack than filter 2s envelope. The
display shows how filter 1 is triggered first with filter 2 coming in later. This
usage of filters in parallel gives us a thick sound as both filters can be heard
together. However, this changes when we assign the oscillators to feed into the
filters in series.
Using the exact same settings as above but flicking oscillator 2 to feed into filter
1 which then feeds into filter 2 (series) the sound changes dramatically as one
filter morphs into another and therefore both filters are heard as one as opposed
to being heard together.

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filters in series fil [Link]

Not only can we heard the attack altering the filter response but we can see how
the filtered sound behaves by looking at the waveform display above. We now
have a low-pass filter feeding into a comb filter and the filter 1 attack allows for
a more dynamic filter morphing.
With the next example we will stay with 2 oscillators, but use sine waveforms,
and run two filters in parallel and have two different LFOs controlling the filter
pans. We have covered modulators and specifically LFOs so this shouldnt be
hard to understand.
sines [Link]
A basic 2 oscillator preset using sine waves with no filter pans but each oscillator
being fed into a separate filter (parallel).
Now we will use two LFOs with each one modulating each filter pan. The idea is
to have each filter alternately pan to the opposing side of the stereo field (left
goes to right while right goes to left).

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sines lfo filter [Link]

The filter pans movement is now quite audible and shows how easy it is to
create a simple modulation that can take a static and uninteresting sound and
liven it up.

sines lfo filter pans [Link]

This is the exact same preset but with the filters being run in series and
modulated by the same LFOs. You can see the differences between the two by
studying the waveforms and, of course, by listening to them.

Lets try some mangling of drum sounds.

dirty kick [Link]
Nothing too special here. A standard 808 kick drum sample. But what happens if
we use a combination of effects chained in series and fed from one device to
another? Let me introduce you to the Banaan Electrique from Reaktor 5.

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dirty kick [Link]

The kick sample is now filthy and full of character and this was achieved by
using a gate which output to a tube compressor then a fully detailed amp
simulator with pre EQ, amp, speaker and post EQ sections, followed by a reverb
and then into a stereo compressor. Using the same kick we can create a rounder
and warmer tone with the following routings and parameter selection.

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dirty kick2 [Link]

The same routings as above but with a delay device instead of a reverb device.
Additionally, please have a look at the varying parameter settings. You can see
how the top end has been tamed and the delay allows for the movement of the
kick across both channels. Not necessarily what we want with kick drum samples
but these exercises are to show you how a sound can be manipulated with
effects.
Lets play with a dead and uninteresting kick sample and create something a
little epic and dynamic.
dead kick1 [Link]

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dead kick1 [Link]

Almost all devices have been used here bar the phaser/flanger device.
This time lets really mangle a drum beat and I will use each device in isolation.
bifta drums 80 bpm [Link]

bifta drums 80 bpm gate [Link]

I have disengaged all the other devices bar the input, gate and tube compressor.
As you can hear I have kept it quite full and not coloured extreme as the
ensuing devices will mangle the sound. I have played around with the gate so as
to allow most of the sound through without reshaping it too much. The tube
compressor has been used very gently with no heavy ratios or late attack times.
The idea is to accommodate for every stage of processing otherwise things can
get a little hectic and levels will fly all over the place. I tend to choose which
modules/devices I want to use and have them all active whilst making
adjustments as I can gauge every step by referencing against the master outs

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making sure not to clip. But in cases like this, or whenever I want to use a
specific process, I will work with the device that is required and then shape the
sound for the next stage of processing.

bifta drums 80 bpm amp [Link]

The chorus/tremolo device is not required and therefore turned off. However,
the amp and speaker section and pre and post EQs have been used and adjusted
to give the quieter signal that you can hear. The reason for this is that there will
be other stages of processing and gain management is an integral part of all
processes and this has to be accommodated for.

bifta drums 80 bpm phase [Link]

The delay device is turned off but the phaser/flanger device is used gently to
allow for the movement of the sound. By using a sensible rate allows for the
sound to move slowly without exhibiting the warbling type of effect that can be
heard when using a fast rate.
Finally we add reverb and stereo compression.

bifta drums 80 bpm all [Link]

The trick with the reverb here is to take away some of the really wet effect and
tame it with some low-pass and hi-pass filtering. The stereo compressor brings
the final level back up and shapes the sound to be a bit crispier and stronger.
Lets move on with a drum beat being run through another Reaktor device,
Anima.

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drum beat3 115 bpm [Link]

A 4/4 115 BPM drum pattern that we can use to sync LFOs to modulate filter
bands.

drum beat3 anima [Link]

Anima is a filter bank that is modulated by a LFO but its design incorporates
important parameters like control of the bandwidth, phase and cycles (Waves ).
The band pass aspect plus the use of the aforementioned parameters of this
filter bank allows for the creation of some very interesting textures. This is an
excellent device as it is not only really simple to use but the modulation can
account for some manic textures. The most useful parameter is the Y-Width. If
on the bandwidth of the filters can be modulated and if off the phase offsets
are modulated. Waves controls the number of cycles and controls the amplitude
of the band pass filters.
Lets use the bifta drums 80 bpm [Link] and mangle it a little using Anima.

anima bifta drums [Link]

By leaving the W-Width off and using modulation to control phase offsets and
syncing the LFO to the master BPM we can create unusual rhythmic effects.
Another useful tool I use is Cubases very own Step Filter. This is a really funky
little vst as it allows great manipulation of both filter cut-off and resonance and
it works by allowing the user to select steps, much like the steps in an
arpeggiator, which control relative filter cut-offs and resonance. Basically, there
are 16 steps and are displayed horizontally, and the filter cut-offs and resonance
are displayed vertically. You move the steps vertically if you want to change the
filter cut-offs/resonance and horizontally to create the 16 step pattern exactly
like any other step sequencer.

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Lets create a funky, squelchy filtered drum beat using:
drum beat1 [Link]

drum beat 1 step [Link]

I have synced the pattern to the master clock on 1/8 divisions and using band
pass filters. I havent altered the resonance (right hand side of the GUI) but
have altered the filter cut-offs. The base cut-offs and resonances are the lowest
value reference points and by adjusting these values we can get softer or
harsher results.
Lets now adjust the step filter with different resonance values and a couple of
altered cut-offs with a lower cut-off base reference value, and route this through
the Softube TSAR-1 reverb device.

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drum beat 1 step filter2 [Link]

By altering the filter cut-offs and resonances we can create a more dynamic
effect which comes across as more squelchy. Routing the filtered effect through
the reverb with lots of dampening, dark tone selection and high amounts of
diffusion we can get a rounder and darker feel. The result is a dynamic filter
effect with space and depth. BUT, why stop there? Lets run the result through
another device, this time the Ohmboyz delay, and add a nice drum beat shuffle
to it using delay values.

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drum beat 1 step filter2 tsar1 [Link]

The drum beat now has movement and life and the delays add that nice shuffle
feel. I have only used a single delay line with delay timing synced to the
master BPM. The three effects have been routed in series which allows the step
filter to feed the reverb which then feeds the delay.
The way to approach working with effects in either parallel or serial is to make a
conscious decision as to the type of effect you are after. Try to hear the effect
in your head and work backwards to end up with the starting point, a bit like
Subtractive synthesis. I tend to think of deciding how to set up my effects path
as an instance of music in my head; do I want the sound to have morphing
features or do I want the sound to explode with all effects playing
simultaneously? Do I want the sound to evolve over time or do I want to make a
single event statement? It is always helpful to ask yourself these types of
questions as they will lead you to construct a better effects chain and the beauty
is that the order of effects can always be changed in the chain since working
with DAWs nowadays allows for a mouse click to make a change whereas in the
old days we had to rewire the path. Additionally, if an effect is not required then
it can be bypassed by a single click and this allows for such huge flexibility that
you can spend little time in testing different effects and then deciding on which
to use.
Although the above can be a good way to structure your effects goals there is

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nothing to stop you doing this randomly and impulsively. I have found some
great results by accident or by being impulsive and, dare I say it, haphazard in
my approach.
Lets do another example with the same drum beat but this time we will use the
delay routed into the PSP MixSaturator to add some dirty low end saturation.
The delay will use LFOs for modulation and we can get a little creative and have
the delays moving across the stereo field.

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drum beat 1 step filter2 ohm mix [Link]

The drum beat now has saturated low end, a nice LFO modulated delay that
adds movement and a step filter that uses a low-pass filter with altered cut-offs
and resonances which allow some really nice squelchy colours to shine through.
Using the same drum beat let us create a really lovely low end moving effect by
using Reaktors Echomani which is actually a delay device with quite a detailed
and well thought out mod matrix which allows us to control the destinations with
all sorts of LFO control.

drum beat 1 [Link]

By using the LFOs to modulate both the EQ section and the delay values we can

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create a slow throbbing flange effect that lies beneath the beat and adds depth
and a completely different low frequency texture that seems to have a life of its
own. The trick is to time everything to sync nicely so we have two layers of
movement. A little bit of distortion adds the final touches to a very useful sound.
All the parameters above have been covered in earlier chapters and the use of
some simple routing allows for such a huge variety of textures that it would be a
sin not to play around and enjoy the results.
Let us end this drum beat exercise with one more instance of Echomania. This
time we will go for a really heavy metallic flange effect.

drum beat 1 echomania [Link]

The real heart of this particular effect is the use of higher frequencies for the EQ
modulation and phase changes for both the delay and EQ modulations.
Additionally, using extremely short delay times and a single delay engine that
uses healthy amounts of feedback we are able to get the heavily modulated
flange effect that lends itself nicely to a time based colour.
The trick with any sound design project is to find innovative effects to use in
mangling sounds beyond recognition or creating variations of the same sound.
Of course, chaining effects in series or parallel or both is a good start but
manufacturers nowadays are in touch with what is required in terms of shaping
sound and this comes from understanding what the most common processing
requirements are.
OhmForce who designed the OhmBoyz understand that simply offering a basic
stereo delay to users is not enough and control over each and every parameter
with a choice of a bank of modulators is the condensed and sensible approach to
using delays in detail. The same can be said of NI Reaktor; the concept of
chaining a bunch of effects in series and parallel and offering them as single
devices with acres of parameter control is a clever way of having everything in
one box so to speak.
More and more manufacturers are realising that dedicated effects, although
good, are not enough and packaging them with other utilities and processes is a
sure fire way of keeping the workflow streamlined and therefore helping the
creative process and not hindering it. A simple, yet effective, process is one that

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allows the user to tap into all the parameters on one screen and in one effect
and bundling it with relevant effects/processes.
A really useful, and extremely simple, effect I have come across is Reaktors
Instant Repeater. The thinking behind this effect is so simple that it belies belief
and not only is it extremely useful but takes the place of a number of processes
that would have had to be conducted before the fun began. Slicing audio into
steps and then offering huge control over the steps is the equivalent of
throwing in an audio slicer and then trying to modulate the slices in real-time; a
bit like a dedicated sequencer that breaks the audio down into grains and then
modulates them.
Lets have fun using the following vocal line and mangling it with the Repeater:
tanikye vocal [Link]

repeater tanikye [Link]

I have divided the vocal line into 32 parts and have selected repeat times over 4
beats with the length of the audio at full and the start time at 0. This choppy
effect is fun and can be used creatively as an effect in a mix scenario and can
even be more useful if timed and layered with the original dry version. The Att
and Dec functions relate to setting the times for crossfading between the dry
and wet signal sampled as 1% of the loop length (a great way of creating
stuttering effects etc).
Using the same vocal line but with the following adjustments we can create a
funky stuttering effect.

repeater tanikye bv [Link]

All I have done is play with the Att/Dec, altered the beats, divide and repeat
values and the result is this staggered repeated segments/divisions of the
original vocals. This is a great way of slicing audio at varying points and
triggering them without having to actually slice the audio and placing midi
events for the various triggering times and lengths.

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We can go back to basics using a single device with a LFO modulating specific
frequency bands and by using the phase functions we can access the different
parts of the LFO cycle to create some new textures.

filter bank tanikye [Link]

The LFO and its phase are the real functions to attack to create a nice undulating
effect. The frequency bands can be selected and the LFO phase determines the
cycle point for the modulation. By altering the phase of the LFO we can create a
modulated frequency and pan effect.

filter bank tanikye bv [Link]

These types of simple effects can be used for very creative purposes and if you
chain them together you can create far more complex effects.
I want to use one more simple effect that we have covered in an earlier chapter;
pitch shifting. The device I will use is Reaktors Pitch Shifter. The beauty of this
device is that harmonies can be created by using grains and their pans.

pitch shifter [Link]

The Shift L/R are pitching shifting controls for shifting by semitones across the
left and right axis. Grain L/R adjusts the grain size of the algorithm (move left
for big grains and right for small grains and therefore more pitch shifting
accuracy). FB is feedback and Reverse reverses the grains.
The whole point of sonic mangling is to do exactly that, mangle sound. So, why
not use the same Pitch Shifter and transform the female Tanikye vocal into what

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sounds like a devil voice with another voice panned to the right that is equally
unwell.

pitch shifter tanikye [Link]

It is so enjoyable playing with decent plug-ins that offer innovative controls. Try
using the settings above on your audio samples and have fun.
For the next chained set of effects I am going to go a little crazy and use a
vocoder, a saturator, a reverb and a delay. Because I am using a vocoder in
Cubase I have had to create a group and assign the carrier and modulator to
this group. Depending on your DAW I am sure there are other ways of setting up
vocoders. This does. However, allow me to insert effects on different channels;
the carrier, the modulator and the group channels.
I will be using the following audio files:

noise hook 80 [Link]

This will be the carrier signal and can be treated on its own channel.

bifta drums 80 bpm [Link]

This will be the modulator and, again, has its own channel therefore can carry its
own effects separate to the modulator channel.
The group channel, where the carrier and modulator lie routed to the vocoder,
can also carry its own effects and dynamics. The combination of two channels
and a group channel allows us huge scope and diversity in applying effects and
dynamics to any of the channels in parallel and/or series.
Let us start with the modulator channel and take a look at the effects settings:

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voc mod drums [Link]

Using the MixSaturator2 we can tame the low end and shape and boost the top
end. By doing this we are compensating for the final series of effects being
output, so it pays to continually reference against the vocoder bands as the
frequencies selected here have a huge impact on the vocoder and the carrier;
lots of saturation taking place but specific to certain frequencies.
Next, we will run the modulator through a delay effect. I could choose to run the
delay post vocoder but I want to affect the delay itself as this will create a really
sweet texture when vocoded.

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voc mod drums sat [Link]

Working with divisions and using the RF (resonant filter) to add some spice
and movement and using a single delay engine/line we can push the vocoder to
accept the delay as part of the frequency makeup. The audio file may sound
strange to you as it has acres of feedback and is really pronounced but I am
continually referencing against the final output and am shaping this delay whilst
listening to the vocoded result. In isolation it can sound over the top but in
context with using all the other effects it sounds very nice and this is why you
need to always reference against the final output.
The carrier is the noise hook and I have left this alone for now as it is already
pretty dynamic. We can, of course, add any effect we choose as this channel is
separate to the others and can carry its own processing, but try to bear in mind
that any changes to this channel will change the overall result.
The group channel effects are as follows:

voc mod drums group [Link]

A full bandwidth allows the signal to be really dynamic across all the frequencies
and by allowing a little High Thru we can mix the hi-passed signal in with the
vocoded signal thus allowing us to hear some of the modulator along with the
full vocoded signal.
Next, we can add a little reverb to give it some space and life.

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voc mod drums group [Link]

You can see that the reverb settings are not too drastic with a very low wet mix.
The idea is to allow for some high reverb frequencies to shine through without
being too wet or splashed.
The result is a smooth and interesting texture that moves along dynamically
thanks to the delay device and offers a sweet sound that isnt too low end heavy
thanks to the saturator and vocoder settings.
By changing the carrier signal we can impart a whole new texture to the sound,
and for that I will use:
voc mod [Link]

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And the result:
voc mod pad2 [Link]
Just by changing the carrier signal and not altering anything else we get a new
texture that is sweet, warm and full.
Feel free to experiment with different carriers and modulators and when using
the effects that are BPM based make sure that all the timings are synced
otherwise you might find that some elements dont sit together too well. Of
course, you can do whatever you like so long as it sounds musical and right.
Lets now have some real fun treating Tanikyes vocals with Autotune, a vocoder
and a reverb. The idea is to layer the dry vocal line with the vocoded layer which
is then detuned, and then passed through the reverb. However, the Autotune
will be used to sway some notes on the vocal channel.
tan dry ld [Link]
I have taken a segment of the vocal line so you can hear Autotune both in
corrective mode (correcting pitch and tuning anomalies) and in creative mode
(trying to pitch manipulate to get the common T-Pain type of effect).
First, let us take a look at the Autotune settings which is dropped into the vocal
channel (carrier) as an insert.

The input selection is Soprano as we are dealing with a female vocal line, the
key is C with chromatic scaling selected as a major scale and, most importantly,
the retune speed is at maximum with a lot of humanize to take the edge off the
retuning extremes. This will give us the famous Autotune effect.
The group channel where the vocoder sits has the following vocoder and reverb
settings using Cubases bundled vocoder plug-in and Softubes TSAR-1 reverb.

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The vocoder settings are really subjective and down to how you want to
represent the sound and in this instance I was going for a layered effect as
opposed to running the vocoder fully wet and because of this I have selected a
large Talk Thru amount which mixes more of the dry signal with the wet signal
thus allowing for the original dry vocals to come through a lot more and this is
what I term as layering as opposed to fully wet. I have also been careful with
the Bandwidth settings making sure not to have too much of a broad vocoded
range.

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The reverb settings are both subtle and slightly dark. I was not going for a
washy reverb effect but trying to add to the vocoded texture. Between Density,
Tone and Dampening I think this has been achieved. I made sure to adjust the
Pre Delay to allow the signals attack to come through before the reverb
reflections kicked in. Finally, a tiny amount of the wet signal is mixed in with the
dry signal to allow for less reverb wash and more of the dry signal to come
through. These settings allow us to shape the vocoded signal so as to add to the
overall vocal effect.
tan ld bv [Link]
The result is a nicely textured vocal line that uses Autotune both for correction
and colour, the vocoder for layering the vocal line with vocoded harmonies and

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to shape the overall result with some reverb to add depth and space to the
vocoded effect.
Time to move onto one of the earlier vocal samples I created and with this
particular sample we are going to warp and mangle it until it becomes a little
more interesting. The vocal sample I am going to use is:
experiment with [Link]
Nothing too exciting her54e but we can have a little fun destroying it until we
find something usable. So, why not start with some drastic Autotuning that does
the opposite of the usual T-Pain type of effect? Why not change the throat
values and maul it a touch?

experiment with reverb [Link]

Some heavy pitching and stretching effects obtained by transposing an octave
down whilst maximising the throat length. We can now take this to the next step
and mangle it even more using another stage of pitch shifting whilst making sure
we get a nice stereo spread.

experiment pitched [Link]

We used the pitch shifter before and with this double processing of pitches we
get that foreboding devil/demon voices.

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Moving on from using just effects lets not forget, or ignore, one of the most
potent tools at your disposal; automation.
Automation
This process involves automatically changing parameters on any given device or
applying a process through automatic means as opposed to manual means. The
usual and most obvious automated processes are those that involve pretty
standard parameters like volume or pan etc. The idea, and process, is to either
record the volume or pan movement by selecting record in your DAW and then
moving the volume or pan knobs whilst recording or by drawing in (with the
pencil tool) automation process in automation lanes. This data is then read by
the software which plays back the audio with the automated changes.
I will start with a very basic example of pan and volume automation using the
vocal line from above in Cubase.

experiment volume and pan [Link]

The experiment vocal line is in an audio track and by expanding the channel you
get access to the automation lanes. It is in these lanes that you can instigate
automation. I have drawn in a couple of basic automation event. The first one is
for volume and the drawn line moves the volume from 0 to a selected value
(maximum in this instance). The next automation lane automates the panning of
the audio from left to right. You can hear the volume building up and the pan
data moving from left to right.

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These two parameters are the most commonly automated events but the real
power of automation, in terms of sound design, comes into play when the
parameters of an effect or dynamic are automated.
For the next example I am going to use, via a drop down menu for the
automation parameters, the master mix of CamelPhat (filters bank).

From the automation drop down menu I can select whichever parameter I want
to automate from the list of inserted plug-ins. In this example I will select
master mix of CamelPhat. This will allow me to automate the mix element of the
filter. But I am not going to stop there. I will also automate the filter cut-off so
that it closes over time. So, now I have two instances of automation; one alters
the wet/dry mix content and the other automates the filter cut-off.

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experiment camelphat [Link]

Automation lane 1 shows the master mix data and lane 2 shows the filter cut-off
data. You can hear the changes in real-time as the automation lines display
steady changes over time until the end.
I cannot show you the real-time automation movements in a static image but I
will show you the CamelPhat settings in the event you want to replicate the
example.

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Okay, time to go a little mental and have some real fun. The next example will
have 3 sets of automation and each one will be drawn a little manically so as to
create a moving delay across the left and right channels (pan automation) plus
changing the delay time in real-time. I will use a short static sound so you can
hear how the delays behave after the sound has stopped.
big [Link]

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big kick automation [Link]

I have drawn a ramp down shaped automation for the delay time which results
in the delay times being changed over time and in this instance the times get
shorter and therefore the delays speed up. In the next lane I have drawn in a
staggered shape for the level. This allows the levels to change over time and you
can clearly hear this in the rendered audio file. Finally, I have drawn in a crazy
shape to alter the pans over time.
By looking at the above and listening to the rendered audio file you can see how
sound designers can create evolving sounds. A good example for using the
automation settings above is when a sound designer wants to create cannon
bursts or something that drops and bounces around while the volume changes.
Lets take this a step further to the realms of sound design as per the distant
cannon bursts mentioned above.
For this I will be using the CamelPhat, the Rob Papen RP Verb and the
automated OhmBoyz delay and all these processes will be applied to the same
big [Link] sample. I will apply a bit of EQ so as to take out most of the mid
to high frequencies whilst boosting the lop0w frequencies a little.

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big kick automation eq [Link]

The kick now sounds dark and muffled, exactly what we want. I mentioned
earlier how important it is to keep referencing against the resultant output and
in this instance it is really important as we can only gauge what EQ to we need
to apply by auditioning it with all the processes in place.
Next we need to filter the kick even further using CamelPhat.

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big kick automation camel [Link]

This has darkened the kick even more and given it that underlying throb that
sounds so good when run through a reverb and delay. Next, we need to shape
the reverb.

big kick automation rp [Link]

The kick now sounds a little mushy and metallic and I have made sure not to use
a long decay time as I want the delays to have the colour and not the reverb.
The delay comes next and is followed by some automation.

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big kick automation [Link]

Using automation to control the delay level (gradual drop in level over time), the
delay pans (a little biased so as not to be too strict in moving to either side) and

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the filter cut-off to close over time we now have a moving and dynamic sound
that has completely changed from the original single hit static kick sample.
I would like to end this book with a little arrangement that incorporates a
number of the processes outlined in this book.
The audio files I will be using are as follows:
deep sleep 71 bpm [Link]
This drum beat will act as the modulator for the vocoder.
vocoder pad [Link]
This will be the carrier for the vocoder and both the above will be sent to a
group as before.
albino [Link]
This line will be the backdrop to the arrangement.
the beat stretched [Link]
This is the main drum beat but it has been time stretched and there will be some
very serious mangling of this beat to add a new feel and texture to the
arrangement.
the beat [Link]
A little reverb and not much else for this piano line.
Lets run through each process and then mix all the files together to create a
new texture.
The first step is to get the vocoder rolling along nicely with the drum beat (deep
sleep 71 bpm) and the pad (vocoder pad carrier). As these are both sent to a
group and processed it makes sense to look at the processing section.
First off, lets look at the vocoder settings:

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Using a full bandwidth with a tiny amount of High Thru to give it some lift and
bounce and very little else bar the 24 band filter stages. This is then passed
serially into the TSAR-1 reverb.

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A balanced mixture of reverb time, density and dampening allows a smooth

reverb to work its magic. I have used a healthy pre delay to allow the vocoder
attack to come though and by using a small wet/dry mix I am able to tame the
reverb so as not to mush up the frequencies too much. This has been done
because there is still another stage of processing to go and that is the OhmBoyz
delay plug-in.

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The trick here is to make the vocoded line bounce along without the delays
being too obvious and to do this a delay time division works very well and
with the delays being filtered and modulated it takes away the bright delay taps.
The result of the vocoder group channel is as follows:
the beat vocoder [Link]
A bright but bouncy vocoded line will add movement to the arrangement whilst
keeping it melodic.
Let us now time stretch/compress the drum beat (120 BPM) to fit into the 71
BPM track.

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The original BPM of this beat was 120 BPM but we have now
stretched/compressed it to fit into the 71 BPM we require for the arrangement.
Keeping the slices equidistant means we get a smooth process.
This beat is now mangled using the following processes:

Using NI Reaktors Banaan Electrique I have concentrated more on getting the

right colour using the amp and speaker simulators concentrating more on taming
the high end but adding some low end presence and boosting in the mid range. I

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have done this whilst referencing against the next process which is the Fusion
Reflection plug-in, again from NI Reaktor.

Using a lot of echo and chorus and concentrating more on diffusion amounts,
and both have hefty doses of it, I can create a crumbling and vibrant drum beat
effect which barely sounds like a drum beat anymore but sounds like a booming
kick and gated reverb snare. The trick, as always, is to reference against the
other processes in the arrangement. The result is:
the beat drum beat [Link]
Heavily affected and downright dirty this newly mangled drum beat sits nicely in
the arrangement.
Finally, let us look at the albino [Link] settings:

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Using band pass filters and modulating the filter pans with LFOs we can get this
sweet evolving synth keyboard sound that moves dynamically across the stereo
field whilst bouncing along nicely due to the filter and amp ADSRs plus the use
of reverb and delay.
And when you put it all together this is the result:
the beat [Link]

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Chapter 10 - Final Word

I hope you have had as much fun reading this book as I have had writing it!
I thought a lot about this book and how to approach it and present it to you and
I decided that some theory with a sprinkling of technical data plus a ton of audio
and visual examples and exercises were the way to go forward. After all, this
book is about being creative and enjoying yourself and if you master the effects
and processes used then you are ready to explore the diverse and beautiful
world of effects on your own. I am fully aware that some readers would have
liked a thesis containing masses of physics and math but, to be honest, isnt a
little background knowledge relevant to the industry coupled with a hands on
approach the best way to learn a process? I think so and with this in mind I have
included so many audio and visual examples that I think you will be sick of me
for a while. BUT, I have not disregarded providing the technical data required in
understanding both the processes and how to best utilise them and I implore
you not to ignore the background data for each process. Understanding the
technicalities in this industry is even more important than a random hands on
approach as there will be all sorts of situations that require a basic
understanding of the technicalities and principles involved, and this is what I
have tried to present to you.
I tend to find that whenever I teach or get engrossed in tutorials I forget how
fun music is and can be. When it becomes your livelihood you can lose touch
with why you got into this vocation in the first place and it is writing books like
this that grounds me again and reminds me of why I got into this madness in
the first place.
Effects are not just restricted to mixing tools but are the most potent tools used
by sound designers, special effects creators, and film scorers. A film with only
music and dialogue ends up being both static and linear in that you can see
where the script and music are taking you. But throw in a few effects and crazy
sounds, or reshape and represent the music in a different light and you have to
re-evaluate the content of the movie. As long as interest is maintained then the
movie will instil a new thought into your head and an emotion in your soul and
keep the interest going. A good example of this is Scarface. The script, acting
and story are excellent but once the genius of Giorgio Moroder is thrown into the
mix then the film takes on a whole new meaning, well, certainly for me. Moroder
used specific equipment to construct the music for the film but it was his use of
effects that also added to an inspired score. Vangelis score for Chariots of Fire
also moved me musically as the effects used in that film changed the sound
entirely and presented it in such a creative fashion that until today it is still
iconic.
Without effects there is so much you can do with a modulation matrix but when
effects are used creative output is endless. It is only through the use of effects

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and some clever little utilities that we begin to understand the behaviour of
sound and once that is understood then warping it becomes much easier.
Basically, the process of learning how to use effects leads to a cyclic climb of
knowledge in that every bit of information mastered allows you to view the
processes in a different light and that in itself takes you back to step 1. Every
time I learn something new about an effect I end up starting from scratch and
using my new found knowledge to further master and understand the process I
am exploring.
Using effects is without a doubt the most creative and fun part of any sound
based mangling chore. I cannot tell you how bored I get with creating drum
sample after drum sample for libraries. It is only when I can sonically alter the
samples that I get creative and enjoy myself even if the results are never used.
Give yourself a break and have some fun.
Mangle something today!
Thank you for purchasing this book.
Eddie Bazil
[Link]

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