How to Make a Noise: Analog Synthesis

How to Make a Noise: Analog Synthesis by Simon Cann

Published by
Coombe Hill Publishing
33 Melrose Gardens
New Malden
Surrey KT3 3HQ
United Kingdom

coombehillpublishing.com

Copyright © Simon Cann 2005, 2007, 2011

All rights reserved. No part of this publication may be reproduced in any form or by any means without the prior permission of the publisher.

While every effort has been taken to ensure accuracy, the author and the publisher make no representations about the accuracy of the information contained in this book and cannot accept responsibility for any errors or omissions, or for any loss based on the content of this book.

Massive is part of Komplete and was created by Native Instruments GmbH.

FabFilter Twin 2 was created by FabFilter Software Instruments, a company founded by Frederik Slijkerman and Floris Klinkert.

This book has not been endorsed by any of the corporations or individuals listed. Equally, neither the publisher nor the author endorse or offer any support for any of the products featured in this book.

Front cover background image © Brunoil | Dreamstime.com

ISBN 978-0-9554955-4-0

20110930-1.0

About the Author

Simon Cann is a writer based in London. He is the author of a number of music- and business-related books.

His music-related books include: Cakewalk Synthesizers: From Presets to Power User, Becoming a Synthesizer Wizard: From Presets to Power User, and Jumpstart Your Music Career. His business-related books feature the experiences of international entrepreneurs who have built successful companies in the hottest business locations around the world.

Connect and Keep Up-to-Date

You can find out more about Simon and keep up-to-date in many places:

Read more about Simon at his website, simoncann.com.

Check out his other music-related books and videos at his Noise Sculpture website, noisesculpture.com.

Find and Circle him on Google+, profiles.google.com/simon.cann.

Keep up to date with his publications by joining his mailing list.

Follow @simonpcann on Twitter.

Like the Noise Sculpture page on Facebook, facebook.com/noisesculpture.

If you do the RSS thing, subscribe to his news feed.

Introduction

Welcome and thank you for buying this book.

How to Make a Noise: Analog Synthesis introduces the tools and techniques so you can create and control sounds using analog synthesis.

Creating and controlling sound—or sound design as it is often called—is not simply about making fun and/or cool sounds. It's about finding a sound that works in context. If you draw a parallel with a symphony orchestra, sound design is about choosing (in other words, programming) the right instrument and ensuring the appropriate playing technique is adopted in order to create the tonal effect the composer (you) wants. In other words, sound design is all about trying to find and create the appropriate medium to express your ideas.

This book is about analog synthesis techniques—it is not about analog synthesizers. The vast majority of synthesizers available today (including the synthesizers featured in this book) are digital, but still many (probably most) synthesizer sounds are created using traditional analog, subtractive techniques.

Although they are illustrated in this book with two specific synthesizers, these techniques can be applied to virtually any hardware or software synthesizer.

And by the way, this book looks at sounds. It is not "the missing manual" for either of the featured synthesizers (since the synths come with very good manuals). It is about how to create and control sounds, and in that context, it doesn't claim to show you how to use every control on every synthesizer.

Instead, it is intended as a ground-up learning experience, in other words, we'll start creating sounds from first principles rather than deconstructing existing sounds. At all stages I would encourage you to try things out and get your hands dirty: there is no need for precision—get out there and create the sounds that sound good to you!

Understanding Synthesis

Synthesis and sound creation comes in many forms.

A key to creating sound with any synthesizer is to understand the strengths and the limitations of your chosen tool (and by implication, the strengths and limitations of the synthesis techniques available on your chosen synthesizer). There's no point in using the wrong tool for a job.

When it comes to synthesizers, there are three main choices for creating sounds:

analog synthesis

frequency modulation synthesis, and

sample-based synthesis.

Let's briefly look at these in turn.

Analog Synthesis

This book looks at analog or subtractive synthesis. This is the type of synthesis used in many classic synthesizers such as the Moogs (Mini Moog, Moog Modular, Taurus, Little Phatty), the Oberheim synths (OB-8, OB-X), the Rolands (SH-1000, Juno-6, Jupiter-4, and Jupiter-8), Prophet-5 from Sequential Circuits, and the ARPs (ARP 2500, Arp 2600).

You will hear people describe analog synthesizer sounds as warm, rich, simple, and powerful. They are all of these and more.

Analog synthesis works by taking a sound source and shaping it. These techniques are fast and powerful, and if you are prepared to invest the time you will find complexity and delicacy.

As you will see from this book, analog synthesis techniques are a great place to start when you're developing a firm grasp on creating and controlling sounds. Once you have a grasp of analog synthesis techniques, then you can think about other types of synthesis.

Frequency Modulation Synthesis

Frequency modulation synthesis (or FM synthesis as it is more often called) adopts a completely different approach to creating sounds when compared with either analog or sample-based synthesis.

Instead of starting with a bright sound and making it dull and/or shaping it with a filter, FM synthesis starts with a simple wave—usually a sine wave—and distorts it (usually with another sine wave). The interaction between the two (or more) waves creates and shapes the sound over time.

The sound from FM synthesis is often characterized as being harsh or cold. It can be, but more significantly, FM synthesis is capable of creating a broad, detailed, highly nuanced range of tones which can be controlled with an exceptional level of detail.

In some ways FM seems complex since it offers an infinite range of possibilities and permutations. However it is this range of choices which offers the broad array of tonal nuances. These different sonic characteristics make FM synthesis an ideal complement to work alongside analog sounds.

Sample-Based Synthesis

Sample-based synthesis offers the facility to recreate the sound of real instruments with a high degree of sonic fidelity and musical control. It then allows the musician to go further and create super-real instruments in that the sampled instrument never goes out of tune, never has a bad microphone located in the wrong position, and can have effects (such as EQ and compression) built-in to the sound.

Beyond that, sample-based synthesis can take raw waveforms, then twist and manipulate them in the digital domain and by applying a range of analog-like tools and techniques to create wholly new sounds. And of course, sample-based synthesis can be used to create and control loops as well as other drum and percussion elements.

Frequency modulation synthesis and sample-based synthesis are covered in separate books which are discussed in greater detail in the Further Resources section.

The Featured Synthesizers

This book is about creating sounds with analog synthesis.

To make the link from the page to the synthesizer, the book shows the practical application of the techniques with two synthesizers: Twin 2 and Massive. These synthesizers are used for illustration: you don't need to own them, but it might help. Even if you don't own the synthesizers, the principles set out in this book can be applied to virtually any other (hardware or software) synthesizer.

Twin 2

Twin 2 was created by FabFilter Software Instruments, a company founded by Frederik Slijkerman and Floris Klinkert (see Figure 1).

Figure 1 Twin.

It is a powerful three-oscillator, twin filter synthesizer, with an extremely high quality sound and a very flexible modulation system. In many ways, the synthesizer has the feel of a vintage analog machine brought up-to-date and rendered in software.

Massive

Massive was created by Native Instruments (see Figure 2).

Figure 2 Massive.

It is a huge-sounding three-oscillator, twin filter synthesizer. While its analog synthesis model underlies its architecture, Massive is very much a modern synthesizer with a huge array of sonic options.

These two synthesizers are used to illustrate specific points. Both synths are capable of much more than I am demonstrating in this book. You can read more about them and their features in the section Architecture of the Featured Synthesizers.

There are other great synthesizers out there—the techniques set out in this book can be applied equally to many of them, but these two have been chosen to illustrate the techniques covered by this book. They were selected because: (1) they are tools of the highest quality (2) they are popular and widely used (3) they sound great, and (4) they are available in PC and Mac versions. And just to reiterate the disclaimers at the start of this book, the developers of these synthesizers have not been endorsed this book and equally, neither the publisher nor the author endorse or offer any support for any of the products featured in this book.

While this book is about analog synthesis, these synthesizers are anything but analog. A key benefit of these synths (when compared to analog hardware) is their semi-modular structure—you can build what you want, however you want it, giving you the sound and control you want.

These synthesizers also add many functions that are not available on conventional analog hardware. For instance:

Multi-stage envelopes.

Tempo-synced modulators.

A wide range of filters (and yes, some of the big modular synths might have a range of filters, but few have the range of these featured synths).

And while I'm in a mood for disclaimers and clarity, this book is about how to make sounds—there is no discussion about the relative merits of the two synthesizers (especially when their features overlap), nor about the fidelity of any hardware that has been modeled in the creation of any of the featured synthesizers.

Format of this Book

This book is only available in electronic form and has been designed, and optimized, to be read on electronic reading devices such as the Kindle, nook, and iPad, as well as the desktop version of any reading tools.

The graphics have been have been optimized for the intended reading medium and are in color. These color images render on a grayscale eInk screen without any problems. The layout (including the images) has been designed to work on a wide range of screens including a standard 6" eInk screen, a phone-sized screen, and an iPad-sized screen.

There are, of course, a few limitations with this format—perhaps the biggest relates to the graphics. Many electronic readers can only deal with graphics of a certain file size. Accordingly the graphics have been squeezed to extremes. This is not a problem for the line drawings (such as the images of the envelopes), but it is more of a challenge for the screenshots (which are reduced in terms of pixel size as well as file size). With this extreme compression, some of the very fine details simply do not render as clearly as I would like on an eInk screen.

You will also see that I have included clickable links to external websites (which will require an internet connection). To differentiate between the internal and external links, external links have a small image to their right, thus.

Navigating this Book

Throughout the book there are clickable links to make it easier for you to jump to a specific point. This means that you can go straight to the section you are looking for, and if you don't like the order in which I've put the chapters, then you can read them in the order you want to and hopefully there are links to any earlier material that is necessary to get a full understanding in the later chapters.

And on the subject of the order that the information is presented, the book is structured as follows:

Chapter 1: Getting Started with Synthesis. Chapter one introduces you to the basic principles of synthesis and the architecture of the featured synthesizers.

Chapter 2: Sound Sources. Chapter two looks at the sound sources and how to deploy those sources to create the broadest sonic palette.

Chapter 3: Filters. Chapter three looks at filters which shape the raw sonic output from the oscillators to create the sound you hear.

Chapter 4: Making the Sound Dance. Chapter four looks at modulation: the process by which all of the elements are controlled to create the final sound.

Chapter 5: Envelopes. Chapter five looks at envelopes, the automatic level controls which shape the sound in real time.

Chapter 6: LFOs and Specialized Modulation Sources. Chapter six looks at low frequency oscillators and other tools to help animate your sound.

Chapter 7: FX. Chapter seven looks at the effects units that are available to the musician and how they can be used to create and shape sound.

Chapter 8: Sculpting Sounds with Synthesizers. Chapter eight looks at the practicalities of creating sounds using analog synthesis techniques.

Further Resources. If you're wondering what to do after you've read this book, then the further resources section will give you some ideas, including directing you to some free books and videos.

OK. That's enough introduction: let's get on with the music.

Chapter 1: Getting Started with Synthesis

While analog synthesizers are all different, they do essentially the same task: create and shape sound.

The options to shape a sound are determined by the tools that each synthesizer makes available, the signal flow within the synthesizer, and the level of control over each element that the musician is given. In short, the architecture of the synthesizer is fundamental to a synthesizer's sound creation possibilities, so let's start by looking there.

Synthesizer Sound Architecture

There are many factors in a sound's design, but in essence a sound will usually have three main elements present in its sound (see Figure 1.1):

the sound generator (an oscillator)

the filter (to shape the tone), and

the volume control (to control the level of the sound over time).

Figure 1.1 The basic signal flow in the sound creation process.

Once the sound has been created, it is then "shaped" over time—both in terms of the tone and the volume—to provide a (hopefully) pleasing tone (and if not a pleasing tone, then hopefully the tone that is created will be one that you want to create).

This type of synthesis—which is seen in every analog synthesizer and analog-emulating synthesizer—is usually called subtractive synthesis: the sound generator creates a sound and the filter "subtracts" elements to reach the final sound.

Once the basic sound has been created, a range of control sources are employed to effect tonal changes over time. These control sources include:

envelopes, and

oscillators (often low frequency oscillators or LFOs).

The term that is applied when one device controls (or changes) another is modulation. So if an LFO controls a filter, perhaps to create a wah-wah type effect, then we would say that the LFO modulates the filter (or more accurately, the LFO modulates the filter's cut-off frequency).

Don't worry if this sounds a bit confusing—this is just a brief introduction to the workings of synthesizers; we will discuss the various elements in greater detail as we progress.

When we start looking at the synthesizers in some detail, it may appear that they have similar features and similar signal flows. You might logically expect that the sounds they produce will be the same. In practice, this is not the case. In synthesizers (both those featured here and the others synthesizers you will come across):

apparently similar sound sources will create very different sounds

apparently similar filters will work in different ways, and

the interaction between various elements will differ giving markedly different sounds.

In short, different synthesizers sound different. I would encourage you not to try to compare and contrast the differences, but to listen to how the synthesizer you are using sounds, and listen to how the tools you have available help you to shape the tone to get the sound you want.

Architecture of the Featured Synthesizers

Let's look at the architecture of the featured synthesizers and think a bit about what the design means for sound creation possibilities.

And just to remind you: this book focuses on sound creation, so I'm not intending to tell you about all the features of these synthesizers.

Twin Architecture

The heart of Twin is three oscillators and two filters.

Each oscillator has six waveforms (including two noise options). Oscillator one and oscillator two each has its own separate (stereo) audio channel—oscillator three then has its output split between the two oscillators' audio channels.

There are two filters, which can be set as low-pass, high-pass, or band-pass filters with a curve of 12dB/octave, 24dB/octave, or 48dB/octave (and if this all seems confusing, don't worry—it's a good thing and will be explained in more detail later in this book).

There are then 11 filter type options—each different filter has a different tonal response.

The filters can be arranged in parallel (to sculpt two separate tones), in series (so that the output from filter one is then sculpted further by filter two), or on a per-oscillator basis so that the separate outputs from oscillators one and two (with any element of oscillator three) is then processed separately. Each filter works in stereo so you can pan the oscillators across the audio field.

The output from the filters is mixed, but then split. One feed goes straight to the output and the other goes to the delay section.

In the delay section, there are two delay units—one is applied to the left audio channel and the other to the right audio channel. While it may seem that Twin doesn't offer much in the way of FX, as you will see when we get to Chapter 7: FX, this delay arrangement offers many options.

Having passed through the delay lines, each (delayed) signal then passes into another filter bank. Like the main filters, these can be arranged in series (so the delayed signals are mixed and then pass first into filter one after which the output from that filter is then sculpted further by filter two), in parallel (to sculpt two separate tones), or on a per-delay basis so that the separate output from each delay line is then processed separately.

After passing through the delay line filters, the signal goes to the output. At the output, the dry and processed signals are separately mixed and balanced in the stereo panorama.

In addition, Twin offers a (seemingly) limitless supply of modulators. If you want to modulate anything, due to the semi-modular design, all you need to do is call up a modulator and assign it to a destination. You can even assign modulators to modulate other modulators (so for instance, you can set an envelope to modulate the extent to which an LFO modulates its destination).

As well as conventional envelopes and LFOs (which, as you will read later, are both over-specified) Twin offers an X-Y Controller, an Envelope Follower (so you can use external audio signals to control a level), and a wide range of MIDI control and automation options.

Massive Architecture

Like Twin, the heart of Massive is three oscillators and two filters. But after that, the two start to diverge...

Each oscillator has a large range of waveforms—some are analog-emulating, but most are digital. The waveforms are not single waves, but a whole range of similar waves (called wavetables)—the advantage of this approach is that you can change your sound by tweaking the source wave in real-time. As the waves are related, you can find a similar, but different, sound. There is a modulation oscillator; however, since this book is about analog synthesis techniques, I won't be looking at the sonic options that it offers.

There is a separate noise generator which has a range of noise-creating options.

The output from each wave and the noise generator can then be balanced between the two filters. There is a third option and one of the waves (or the noise generator) can also have its unfiltered signal mixed directly before the master EQ section and then output.

Like Twin, Massive has two filters which can be set as low-pass, high-pass, or band-pass filters with a curve of 12dB/octave, 24dB/octave, or 48dB/octave. In addition, Massive has a range of other more specialized filters.

The filters can be arranged in parallel or in series, and the two options can be balanced to give shades of sonic difference.

After passing through the filters, the signal passes through two FX units, one EQ block, and to the output.

At various points (before and after the filters, and after the filters output has been mixed) FX units can be inserted into the audio chain. At various points, the signal can also be tapped and fed-back into the audio chain in order to warm-up/roughen-up the sound.

Massive has four envelopes and four LFOs, and in addition there are a range of other modulation sources and controls that can be applied. The LFOs can also be repurposed for specialized tasks, becoming Performer and Stepper modulation sources (which are essentially highly tweakable LFOs).

Examining the Elements

In the next few chapters we're going to focus on each element in the sound creation process.

This is a slightly unnatural approach since you're unlikely to only tweak one parameter when you're creating sounds. In practice, you will probably keep making minor adjustments to all the functions as you fine-tune your sound. However, I want to make sure we have a clear understanding of what every element brings to a sound. Then we can move on and look at how to apply those principles, and how to structure the elements, in order to create sounds

These next few chapters will therefore be slightly abstract in that we're not looking at completed sounds, but I encourage you to get your hands dirty and try things out with whichever synthesizers you have to hand.

Chapter 2: Sound Sources

We've looked at synthesizer architecture. Let's now move to the beginning of the audio chain and look at the sound source.

In analog synthesizers, a sound is created by an oscillator—these are single-cycle waves which keep repeating. In today's digital synthesizers, the sound sources are digital recreations of analog oscillators. Without any treatment, most of these waves (whether digital or analog in origin) sound either harsh or uninteresting.

Basic Wave Shapes

There are some wave shapes which are common to virtually every synthesizer that has ever been produced. Twin (see Figure 2.1) and Massive both offer all of the basic wave shapes. Massive also offers a much wider range of specialized waves which I will discuss briefly.

Figure 2.1 The oscillator block in Twin where the waveforms can be selected. You can see on the left-hand side that the selected waveform is displayed.

Before we go any further, a quick suggestion: please do not try to make any assumptions about how a wave will sound based on how the wave looks—just because a wave looks spiky on an oscilloscope (or in an image I have drawn), that doesn't mean that it will have a spiky sound (especially after it has been through a filter). There is only one way to really understand how a wave sounds, and that is to listen to it.

Also, please don't get too hung up with how a wave with the same name will sound different on different synthesizers. Don't concern yourself with labels, but instead, do concern yourself with how individual waves in different synthesizers sound and how you can use each wave in an appropriate context.

As a side note, as we will read later (in Chapter 6: LFOs and Specialized Modulation Sources), these basic wave shapes are very often used for low frequency oscillators.

Sawtooth Wave

The sawtooth wave (see Figure 2.2) may be the closest that we will come to a general purpose wave. It is used by both of the synthesizers featured in this book and on virtually every other synthesizer.

Figure 2.2 A sawtooth wave. In this case, a rising sawtooth wave.

The sawtooth wave gives a bright sound which is often used as the basis for brass and string sounds as well as general "fat" synthesizer sounds (such as stabs and basses). It is rare, but not unknown, to hear a raw sawtooth wave. Because of the harsh, bright quality of the wave it is usually filtered.

While a sawtooth wave is a bright wave, filtering can add warmth and depth to a sound. The sawtooth lends itself well to filtering—its bright character can be easily shaped allowing you to create a unique sound.

However, even with filtering the wave often tends to dominate a broad proportion of the sound spectrum. This may not be a problem, but if your arrangement contains several patches based on sawtooth waves, you may find your mix starts to get muddy. One solution here is to apply two filters in series when you create a sound—one to shape the sound and the other (a high-pass filter) to cut out some of the muddiness. We'll look at this technique in Chapter 8: Sculpting Sound with Synthesizers.

Square and Pulse Waves

The square waveform (see Figure 2.3) has a hollow quality and is often used to create "woody" or "reedy" tones such as those found in woodwind instruments. It is also frequently used in bass sounds, either on its own or to fatten up a sound, often acting as a sub-oscillator (in other words, a note pitched below the fundamental which is felt more than it is heard).

Figure 2.3 A square wave.

Unlike the sawtooth wave, when the square wave (and to an extent the pulse wave) is filtered, the underlying character of the wave tends to shine through. Accordingly, filtering will often make the sound thinner or less bright, rather than creating a new sound. However, this is very much dependent on the characteristics of the chosen filter.

A pulse wave (see Figure 2.4) is like a square wave, but the two sides of the wave are not balanced. The tone varies according to the width of the pulse, giving a range of tones going from sounding similar to square wave, through becoming increasingly thinner and more nasal, to ending with noise.

Figure 2.4 A pulse wave.

Some synthesizers offer a pulse wave option, however, Twin does not. Instead, it offers a square wave where the pulse-width can then be modulated, either manually (by adjusting the middle knob, see Figure 2.1) or by attaching a modulation source (such as a low frequency oscillator) to constantly shift the pulse-width, which often gives a fatter tone.

Pulse-width modulation (often called PWM) is a technique most associated with square and pulse waves. With a square wave, the positive and the negative phases of the wave are balanced. When the pulse-width is modulated, this balance changes to give a different shaped wave. The different waveforms are not simply different shapes on an oscilloscope, but contain different spectral components, hence their different tone.

Sine Wave

The sine wave (see Figure 2.5) is perhaps the most basic element in a sound. It is the purest form of tone you can have—it consists of the fundamental note and has no overtones. If you run a sine wave through a filter, there are no overtones to filter out—therefore the only effect that a filter would have would be to reduce the volume of the note itself. If you put any sound through a low-pass filter, as you take out the harmonics it will tend to sound like a sine wave.

Figure 2.5 A sine wave.

On its own a sine wave can sound quite dull and is not often a first choice for programming. However, as a waveform it is often used to thicken up patches. Where a waveform sounds weak on its own adding a sine wave can give a depth to a patch and add a roundness/fullness to the sound.

Sine waves are also often added to bass patches to give a sub-sonic, foundation shaking, quality. If you are doing this, please check the patch on full range monitors and take care not to blow your speakers.

In some of the original analog synthesizers, it was difficult to generate a pure sine wave so it was common to implement a triangle wave instead. The triangle wave is easier to generate using analog components and gave a sound which was "close enough".

Triangle Wave

A triangle wave (see Figure 2.6) gives a sound that is slightly less reedy or perhaps less sharp than a square wave. If you want to stretch a point, you could alternatively think of the triangle wave as being like a sine wave but somewhat sharper in its output.

Figure 2.6 A triangle wave.

A triangle wave is often used in an LFO as a modulation source since it can give smoother results in some contexts.

Noise

A noise generator (see Figure 2.7) is useful for creating unpitched sounds, for instance, it is often filtered and used for wind effect, to simulate breath (perhaps when emulating a flute), and in percussion sounds.

Figure 2.7 Noise.

Noise comes in different colors (white, pink/red, brown, and gray). The colors denote the balance of harmonics in the wave. The practical effect of this is that white and gray noise are the brightest, then pink/red, and finally brown.

Twin offers noise (white and pink) as a wave shape option. Massive (see Figure 2.8) has a separate noise generator (in addition to the three oscillators) with a wide range of noise sources which can then be tweaked further.

Figure 2.8 The noise generator section in Massive.

Complex Wave Shapes and Variations

We've now looked at the basic waves.

One of the reasons for choosing Twin to illustrate some of the principles in this book is that it only has a limited number of waves (indeed, it only has the waves mentioned so far). While this may appear limiting, it does not need to be—as you will see when we reach the next section (Combining Sounds) the three waveforms can be combined to create a broad palette of sounds.

While Twin has fewer waveforms than many other synthesizers, its approach does have advantages, primarily it is fast—you can always get want you want, quickly, and this speed gives a focus in programming.

Like virtually every other synthesizer on the market, Massive offers a range of different waves. As well as offering a wide range of waves, each wave is actually a set of waves—when you load one wave, you're actually loading a whole set of waves (a wavetable), each of which is similar but subtly different. By changing the selected wave from the wavetable while the sound is playing you can achieve noticeable tonal shifts.

To access the different waves in the wavetable, you can spin the Wt-position knob in the oscillator section (see Figure 2.9).

Figure 2.9 The oscillator section in Massive.

Wavetables of this type are comparatively uncommon (and they definitely never appeared in analog synthesizers—they are something of the digital era). However, analog synthesis techniques can be applied when this form of oscillator is used as a sound source.

Combining Sounds

Apart from one or two synthesizers, every synthesizer (including both synths featured in this book) allows you use more than one waveform at a time. A single oscillator has a limited range of options—if you want a broad tonal palette from which to create your sounds, then you need to start combining oscillators.

There are many reasons for sound combining (including to create new tones, to augment a weak tone, to get a fatter sound, or simply to get a smoother sound) and there are many ways that sounds are combined, for instance two similar sounds could be doubled or two different sounds can be layered to create a wholly new sound.

Whatever the reason, sounds created with multiple oscillators will generally fill more of the sound spectrum. Accordingly, care needs to be taken to ensure that these sounds do not come to dominate a mix (unless that is the intention).

There are many ways that sounds can be combined to provide different sounds:

Waves can simply be doubled.

One wave can be set at a different pitch.

The phase of one wave can be different from another.

A different waveform can be selected.

A combination of any of the above.

Let's look at these options in a bit more detail.

Doubling Oscillators

The simplest way to combine oscillators is to take one oscillator and clone its settings to a second oscillator. Simply cloning the oscillator may not do much more than increase the volume. However, the addition of a second oscillator does give many more sonic opportunities.

Oscillator Phase

When we talk about the "phase" of an oscillator we generally mean the position in a wave's cycle (see Figure 2.10). With one oscillator, the phase matters little, however with two oscillators the effect of phase can be dramatic.

Figure 2.10 The phase of an oscillator. The blue sine wave has reversed polarity (when compared to the red sine wave) and the dotted gray triangle wave is 180 degrees out of phase.

If you have two waveforms that are totally in phase then you will get reinforcement of the signal. Reinforcement is perceived as an increase in volume. If your waves are out of phase then you will get interference and cancellation. The effect of the cancellation depends on the individual waves: if you have two sine waves that are 180 degrees out of phase, then you will get total cancellation.

Partial cancellation caused by putting waves out of phase is generally perceived as a change in tone, usually making the sound thinner and sharper. This may be a great result if you are after a plucked sound or a more metallic sound—it may be less impressive if you are after a really fat sound.

The phase of a wave can be synchronized with the start of a note (in other words, the key strike). This means that phase of every note from each oscillator will be consistent each time a key is struck—so if you have set two waves to be slightly out of phase, the tone this creates will be consistent for each key strike.

You can synchronize the start point of a wave in Twin by clicking on the MIDI symbol to the right of the waveform selector drop-down in each wave. You will only hear an effect (which you might perceive as a tightening of the sound) if you sync more than one wave. If you do not sync the waves, then their starting phase will drift and you will hear cancellation.

You can't set the starting phase of a wave in Twin, but you can in Massive by accessing the Osc section in the Center Window (see Figure 2.11).

Figure 2.11 The Osc block in the Center Window of Massive where you can set the phase for each wave (in the Oscillator Phases section in the middle).

Neither of the featured synthesizers allows direct access to tweak the polarity of a wave.

Oscillator Detuning

Another common technique used with doubled oscillators is to detune one of the oscillators (or to detune each of them but in different directions). Subtle detuning can give a natural chorusing effect which is perfect for creating fullness/roundness/smoothness or just adding fatness to the sound.

There is a balance to be struck when using this technique—if the detuning goes too far then the resulting sound can become flabby and/or out of tune. This may be a good or bad thing, depending on the effect you're after.

Multi-Oscillators

So far we have looked at doubling oscillators. However, there is no reason why you shouldn't triple or quadruple oscillators (assuming you have that number available) and separately detune each oscillator.

Clearly there is a limit to how many oscillators you can call up—for both of the featured synths, that limit is three. However, many synthesizers allow you to get past limitation by using a multi-oscillator. These work by giving a large number of oscillators (for instance Twin allows 32), each slightly detuned. The net result is a massive sound (no pun intended) without using all of your available CPU resources.

Different synthesizers implement this function (if they offer it) in different ways. In Twin, this functionality is applied to all oscillators (in other words, it's not on a per-oscillator basis). There are four controls (located across the bottom of the interface) that control this sound (see Figure 2.12):

Figure 2.12 Implementing multi-oscillators in Twin. By the way, the Spread and Stereo knobs pop-up (as displayed) when you click on them.

Voices. The Voices drop-down determines the total number of notes that may be played simultaneously. This number is the product of the number of keys held and the number of oscillators (controlled by the Unison knob).

Unison. The Unison drop-down determines how many oscillators there are in this multi-oscillator, so if you set this to 12 and hold down a key, then you will have twelve slightly detuned oscillators under your finger (provided you have set the number of voices to 12 or more).

Spread. The Spread control determines the maximum detuning of the most detuned oscillators within the multi-oscillator—the tuning of the other oscillators will then distributed between these two extremes. Set this too low and the sound will be thin and buzzy; too high and it will sound out of tune. Somewhere in the middle and you'll find that fat sound.

Stereo. The Stereo control doesn't change the sound, but it does allow you to spread the voices across the stereo panorama.

You can access similar functionality in Massive in the Voicing section of the Center Window (see Figure 2.13).

Figure 2.13 The Voice block in the Center Window of Massive where you can set the multi-oscillator parameters.

Layering Oscillators

There is a fine difference between layered and doubled oscillators—I am only using different terms so that I can be clear about the different concepts.

With layered oscillators, you take two (or more) oscillators acting together. However, the oscillators are different, so for instance, you may combine a saw and a square wave. Alternatively you may combine two square waves, one being an octave higher than the other.

The combinations may or may not be detuned—that is all a matter of taste, however, if the oscillators are split over octaves, the effect of mild detuning from perfect octaves is not particularly noticeable.

For a simple practical example of layered oscillators, take a sawtooth wave and a square wave. On their own, both waves have a certain sound: both are quite bright and in certain circumstances this may mean that some richness and some depth of tone is lacking. You could double either of these individual waves to get a richer tone. Alternatively, you could layer these two oscillators.

Layering gives a new tone that is neither sawtooth nor square—the layered sound still has the brightness of the two components, but perhaps more weight. If you then want to thicken things up considerably, you drop the square by an octave.

There are many options for layering waves. Take a look at Figure 2.14 which shows a sawtooth wave and a square (both of the same pitch being added together).

Figure 2.14 When a sawtooth wave and a square wave are added together, the result looks like a combination of the two.

Now look at Figure 2.15 which shows two sine waves being added: one being pitched an octave above the second.

Figure 2.15 When a sine wave, and a sine wave pitched an octave higher than the first are added together, the result is a wholly new wave.

Lastly, look at Figure 2.16 which shows two sawtooth waves of the same pitch being added: one is 180 degrees out of phase when compared to the other. As you can see the end result is a sawtooth wave, but that resulting wave is pitched one octave above the two combined waves.

Figure 2.16 When two sawtooth waves—one 180 degrees out of phase—are combined, the result is a sawtooth wave, but raised by an octave.

Layering Sounds

In creating sounds you aren't limited to layering/doubling your oscillators—you can also layer whole sounds.

If you split your oscillators and process them separately through different filters then you can create two completely different sounds. These two sounds can then be layered to create a new sound. This is a different approach to layering oscillators and will give you another dimension to your sound.

As you will see when we look at Chapter 4: Making the Sound Dance you can control these sounds in real-time, for instance, you can mix them using an X-Y controller.

Hard Syncing Oscillators

One other way to layer oscillators is to hard sync them.

With hard sync, one oscillator is slaved to another oscillator. The slave oscillator will restart its phase each time the master oscillator restarts its phase.

If the two oscillators are pitched at the same level, the effect is comparatively mild. If the two oscillators are pitched differently, then one oscillator will complete its cycle before the other. This means that the slave will be part way through its phase when it is retriggered.

This can result in a very hard sound that is often used for creating cutting lead sounds.

You can engage this function in Twin on a per-oscillator basis. When the oscillator controls are open, the third knob in the oscillator bank controls the pitch of the hard synced oscillator. (When the knob is set to 1 the effect is switched off.)

Combining Waves in Practice: Creating New Tones

That's enough theory; now it's over to you. Let's me suggest a few first steps and talk about what you might hear.

Before you can build sounds you need to understand your raw material—in other words, you need to listen to the raw sounds coming out of your synthesizer and be able to hear the sonic nuances when those sounds are combined with other sounds, and when those resulting sounds are filtered. In other words, you need to have your own musical equivalent of paint color charts when you want to redecorate your home.

If you've ever used these charts, you will know that on their own each color looks fine—it is only when you put the color next to a similar color that you can see the similarities and the differences. And as with color charts, when you have a range of tones, there is a dilemma about choosing the right one.

So fire up your favorite synthesizer, and let's get going!

In auditioning the combinations I suggest below, listen for these elements:

the added weight of combining two oscillators

whether the combination creates a completely new tone, and

whether the component parts can be separately identified.

Also pay attention to listen to how the combinations react to filtering, especially when one oscillator is pitched above the other. In this case, with a low-pass filter the higher pitched oscillator will be disproportionately attenuated (which may or may not be a good thing).

Raw Waves

Before you do anything, first listen to your raw waves with no filtering. Get to hear what a single wave on its own really sounds like and only then listen to how they react to filtering.

Doubled/Layered Wave, Same Octave

Once you really know how your waves sound:

double the waves, so you have two of the same waves playing together, and

layer the waves, so for instance, listen to a sawtooth and a square wave in combination.

When the waves are layered, adjust their relative levels so one is more dominant than the other, then change the levels so the other dominates.

Listen to the waves with and without filtering, and if your synth has the facility, listen to the waves in and out of sync and with the phase adjusted.

If you are layering a sawtooth and a square wave, they will usually combine well. When the combination is filtered, to my ear the sawtooth is filtered more heavily.

If you're able to adjust the phase, notice how the sound becomes thinner as the phase is changed. Pay particular attention when the phase difference is near to—but not exactly—180 degrees.

Layered Waves, Different Pitches

When you're happy with the sounds from layered waves, start to introduce pitch differences:

initially split the waves over one or more octave intervals and use the same waveform, then

use the same intervals but combine different waveforms (square and sawtooth always work well), and finally

combine waves (the same or different shape) with different pitches (the interval being set in semitones, for instance, seven semitones—a major fifth).

You will hear that some combinations give a single coherent sound. With a bit of level balancing and maybe some gentle filtering, you will often be able to find this coherent sound.

As the interval becomes greater, it will be harder to find a coherent sound. Very often you will be able to hear two distinct elements to the sound although those elements will not necessarily be the source sounds. And by the way, there's no right or wrong here—it doesn't matter whether you achieve a coherent sound or if there are two clearly audible elements. What matters is that you hear what's going on.

When you use non-octave intervals, you will probably want to drop the level of the re-pitched oscillator unless you are after a clangorous sound. However, you will also find that some intervals (for instance fifths—in other words seven semitones above, or an octave plus seven semitones) will sound best.

As I've already mentioned, the other behavior that you will probably notice is that will greater pitch differences, the effect of the filter on the higher pitched oscillator will be much more noticeable.

Combining Waves: In Summary

As I hope you have heard, combining two waves will create a range new sounds that you can control to broaden your sonic palette.

To summarize, when mixing two waves together:

The waves don't need to be the same volume—one can dominate and one can be softer with the combination. Varying the mix of two waves will vary the resulting tone.

You can have any pitch combination. The easiest pitch combinations are based on octave steps (in other words you can set one wave an octave or two octaves, and so on, above or below the other). However, there's no reason why you shouldn't use other intervals, for instance, a major fifth (seven semitones) difference often works well.

Some combinations work well and give weight to the sound and/or a new tone. Other combinations don't work so well. Take some time to listen to what sounds right to you.

Some combinations work together to give a single coherent tone. Others—particularly those based on large pitch differences—give two (or more) clearly identifiable sounds. Very often these separate sounds are distinct from the component waves that are being mixed.

There are no rules, so I would encourage you to play and listen.

Starting to Shape

This chapter has looked at the raw sounds that are available to you. Let's move on and look at how those sounds can be shaped with filters.

Chapter 3: Filters

We've looked at synthesizer architecture and the main sound sources. Now we're going to look at how to sculpt your sounds.

At its most basic you can think of the filter as being a tone control. A low-pass filter is like a treble control—turn it down and the sound gets "duller". However, a filter can do much more for you.

There are two key differences between a tone control and a filter:

Filters work on a per-note basis to individually sculpt each note. This is important because...

Filters can be animated in real time. If you think of a piano—when a note is struck it is bright, but the tone decays over time. By having filters on a per-note basis, you can hold a note and trigger a new note—the new note will have its own filter giving a much brighter sound which would not be possible if the note needed to use the other decaying note's filter settings.

There are many different types of filters, but the most common is a low-pass filter, so we'll look at the workings of that type of filter first.

Filter Parameters

Filters have two main control parameters: cut-off frequency and resonance, see Figure 3.1.

Figure 3.1 A low-pass filter with a resonant peak.

Cut-Off Frequency

The cut-off frequency is the point at which the filter starts to have effect. So if you have a low-pass filter and set the cut-off frequency to 8kHz, the sound spectrum above 8kHz will be progressively reduced. If you are using a high-pass filter (which is discussed shortly), sounds below the 8kHz cut-off frequency will be reduced.

Resonance

Resonance adds some bite to a filter. It works by boosting the sound spectrum around the filter cut-off frequency as Figure 3.1 illustrates.

Used in moderation, the effect is subtle and can make a sound appear brighter and/or slightly thinner (or less fat if you prefer). When used to the extreme the effect is noticeable—most dance records use filter sweeps with high levels of resonance to exaggerate the effect of the filter sweep.

At very high resonance settings, some filters can exhibit quite extreme behavior, so take care with this parameter and turn down the output if you don't want to burn your ears off.

As I'll discuss later when we start looking at modulation, turning up the resonance also adds some "squelch" to a sound.

Filters on the Featured Synthesizers

Twin and Massive both have two filters.

Filters in Twin

Twin has a gorgeous user interface allowing you to see what is going on (see Figure 3.2). It also has a practical purpose: you can tweak the graphics (as well as twisting the knobs).

Figure 3.2 The filter block in Twin. (Ignore the Master Tune and Portamento knobs: they're nothing to do with filters.)

There are three filter types (low-pass, high-pass, and band-pass, all of which are discussed in this chapter) and there are three selectable cut-off slopes available for each filter. Each filter offers a range of filter models giving different sonic responses and creating a much broader sonic palette.

Filters in Massive

As you can see from Figure 3.3, the filter block in Massive has a less helpful interface, but as long as you know what you're doing, that's not a problem.

Figure 3.3 The filter controls in Massive.

Massive offers a range of different filter types, including some which are not available on Twin (although Massive does not offer a wide range of sonic responses in the way that Twin does).

Signal Flow through the Filters

Both synthesizers allow you to arrange your filters in series, in parallel, and on a per-oscillator basis (which can present some challenges since each synth has two filters, but three oscillators).

When the filters are arranged in parallel (or on a per-oscillator basis), the sound-shaping undertaken by the filters is achieved separately. This mean you can create two distinct tones.

By contrast, when filters are arranged in series, the output from the first filter is fed into the second where it is shaped further.

Neither option is better, they simply give different sonic results.

Massive is slightly unusual and also offers a compromise between series and parallel so that you can:

mix the parallel and the series sounds together, and

split the output from the first filter (to mix it with the series/parallel output from filter two).

In other words, as well as having the series and parallel sounds, you can also find all the combinations in between.

To achieve this level of detail in Massive:

in each oscillator, the F1/F2 slider should be balanced in order that some signal is sent to both filters, but if you want the filters in series, then the signal should be sent to F1 only

the F2 slider (to the left of the filter block) selects between series and parallel (check back to Figure 3.3), and

the Mix slider (to the right of the filter block) selects between the output from each filter.

This level of control is not common on many synthesizers. However, you could readily replicate the signal path with a modular synthesizer.

Filter Types

Let's have a look at the main filter types that are usually found on synthesizers in a bit more detail.

Low-Pass

The low-pass filter (or if you prefer, high-cut filter) allows low frequencies to pass through it, see Figure 3.4.

Figure 3.4 A low-pass filter.

When a low-pass filter is fully open, all frequencies can pass through it.

As the filter is closed, it progressively allows less sound to pass through—you will hear this as the sound becoming duller as the higher frequency elements of the spectrum are filtered out. When the filter is nearly completely closed only the very lowest elements of the frequency spectrum can pass— this sound will be close to the sound of a very low-pitched sine wave.

The effect that the low-pass filter will have on a sound will vary depending on the source waveform you have selected. If you choose a sine wave, then the effect of the filter will be limited. The sine wave comprises only the fundamental frequency, therefore if the filter cuts this, it cuts the whole sound. However, if you choose a sawtooth wave which has a lot of high frequency information, then the low-pass filter will have a much greater perceived effect on the sound.

However, don't think that if you're using sine waves you won't want to use a filter. For instance, if you are playing chords with a sine wave patch then the individual sine wave notes will interact to produce frequencies beyond the range of the individual sine waves—the filter will affect these frequencies.

High-Pass

The high-pass filter (or low-cut filter) is the reverse of the low-pass filter—it allows high frequencies to pass and progressively cuts out the lower frequencies, see Figure 3.5.

Figure 3.5 A high-pass filter.

As well as sound shaping, high-pass filters have another use: to filter out the junk in the lower end of the mix spectrum. How often have you listened to a track and found it sounds muddy or dull? That could be too much unfocused bass. You only get so much dynamic range and without filtering you may be filling your low end needlessly. This means that the key elements—your bass and kick—can't shine.

While high-pass filtering may be noticeable if you play a patch on its own, in a mix any change to the tone of a sound is unlikely to be noticeable except with more extreme cuts. However, the net result of the low end filtering may be to give a cleaner/fuller low end when the bass elements are allowed to come through.

One great use for a high-pass filter (with resonance) is to boost the fundamental tone of a patch. You can do this by tuning the resonance to the fundamental frequency of a note and engaging key tracking so that the resonance boost then follows the played note. If you arrange your filters in series, the second filter can then undertake the main sound-shaping duties.

Band-Pass

The band-pass filter acts like a combination of a low-pass filter and a high-pass filter by cutting the frequency spectrum at both ends to only allow a narrow band of sound to pass, see Figure 3.6.

Figure 3.6 A band-pass filter.

The cut-off frequency control determines the center frequency where the full signal is allowed to pass— from that point outwards, the spectrum of sound is progressively cut. The resonance control has the effect of boosting the signal around the cut-off frequency.

At extremes of frequency, the band-pass filter will sound similar to either a high-pass or a low-pass filter.

Band-Pass filters tend to take a lot of energy out of the signal so you may have to boost the level after the filter.

Notch

If the band-pass filter is the equivalent of burning the candle at each end, then the notch filter equates to burning it in the middle. The notch filter cuts the frequencies at its current value, see Figure 3.7.

Figure 3.7 A notch filter.

The notch filter can be used for effect and it can be used surgically in the mix. If you're trying to mix two sounds and they don't sit well together it may be that they're both trying to operate in the same frequency range. In this case, you can "notch out" one of the sounds to allow the other to sit properly—you can do this in your patch design or, as many mixing engineers do, with EQ in the mix.

Comb Filters

Comb filters work by adding a slightly delayed version of a signal to itself. This causes phase cancellations and can give a slightly "chorused" or metallic type of sound.

The spectrum produced by these filters looks like a comb (see Figure 3.8), hence the name.

Figure 3.8 A comb filter.

Filter Slopes

A low-pass filter progressively reduces the volume of a sound above the cut-off point. The rate at which the sound wave is reduced above the cut-off frequency is determined by the slope of the filter. The slope reflects how aggressively the filter attacks the sound it is filtering.

If you have a 6dB/octave filter (which is quite a subtle filter), it will reduce the level of the sound source by 6dB at one octave above the cut-off point, 12dB at two octaves above the cut-off frequency, 18dB at three octaves above the cut-off.

A 12dB/octave filter (sometimes called a 2-pole filter) will reduce the level of the sound wave's volume by 12dB for each octave above the cut-off point.

A 24dB/octave filter (sometimes called a 4-pole filter) will reduce the level of the sound wave's volume by 24dB for each octave above the cut-off point.

You will see some different filter slopes for a fixed cut-off frequency in Figure 3.9. This is an image drawn by hand and so is not to scale, however, you can see the differing result of the filter slopes. As you would expect, high-pass filters cut at the same rate, but below the cut-off frequency.

Figure 3.9 Different filter curves (note, these are not drawn to scale).

Twin and Massive include a range of 12dB/octave, 24dB/octave, and 48dB/octave filters.

Using Filters in Practice

One of the key issues for making a filter sound realistic is the modulation that is applied to it.

A filter which is not modulated will be, and will usually sound, static. The human ear is attuned to natural instruments which have a constantly shifting character and will expect similar characteristics in synthesized sound.

Modulation is discussed in greater detail in the next chapter, but here's a preview of some of the more common filter modulations.

Controlling Filter Cut-Off

When using a filter to shape tone, the main (but not the only) modulation controls are:

Envelopes. Envelopes provide an ideal control source to shift the filter over time.

Velocity. Velocity allows a filter to mimic the properties of natural instruments so that the sound gets brighter with more forceful playing. As well as allowing velocity to directly control the brightness, you can also set velocity to change the extent to which an envelope modulates the brightness (in other words, you can use envelopes and velocity together).

Filter as a Volume Control

A filter can also act as a volume control—the more a filter closes, the more it attenuates the amplitude of the sound. You can use this to your advantage by setting an envelope to completely close a filter.

Key Tracking and Filters

You may want to use key tracking when setting the filter to mimic the behavior of real instruments.

If you are designing a sound to reflect the behavior of an acoustic instrument, a single cut-off point may be unnatural: the high notes would be too dull and the low notes to bright. To remedy this you could use key tracking with the filter: this would open the filter at higher frequencies and close it at lower frequencies giving a more natural response.

If you want, you can have an unnatural response where the filter closes more at higher pitches.

Controlling Resonance

Resonance is generally used in four main ways:

to emphasize the filter cut-off, and in particular, to emphasize a changing filter-cut-off frequency (which can be controlled by an envelope or by tweaking the parameter)

to give a brighter sound

to give a "thinner" sounds—this is partly a result of making the sound brighter, and

to create an effect, whether that be making the filter scream or giving sounds a really squelchy character.

Controlling resonance with an envelope or MIDI velocity will allow for more subtle nuances to be introduced into a sound.

Adding Some Animation

Now you've got an idea about some of the specific filter modulations you can perform, let's move forward and look at the subject of modulation in greater detail.

Chapter 4: Making the Sound Dance

We've now looked at oscillators and filters, and briefly touched on how they work together and interact. However, all of the sounds we've made so far have been static (unless you have actively adjusted any control as you held a note).

In order to create an interesting sound, the parameters shaping the sound need to be automatically adjusted in real-time as a note is played.

The technique to achieve this movement—which will then animate the sound—is called modulation. Modulation goes to the heart of creating sounds. It is the technique by which a synthesizer creates the varied nuances of sound. Without modulation sounds may be rich, thick, fat, bright or delicate. However modulation will take a basic sound and animate it, giving it life in the hands of a skilled player.

Modulation: The Concept

The concept of modulation in a synthesizer is simple—something is changed by something else, and usually in real time.

Conventionally, a "source" will modulate a "destination". So if we think about vibrato, a low frequency oscillator would act as the source and the pitch would be the destination.

If you then want more detailed expression, then the extent by which the source modulates the destination can be determined by a "control" (which in turn would be another modulation source, but its modulation destination would be the amount of modulation, rather than the destination).

So staying with our vibrato example, if we have a low frequency oscillator acting as the source to modulate the pitch of a note, then you could set a modulation wheel to control the amount of vibrato in real-time.

Modulation in the Real World

Before we get to the detail, let's think about modulation in the context of an acoustic instrument such as the piano. Taking an over-simplistic view, the factors that have an effect on the sound are:

velocity

pitch, and

time.

Effect of Velocity on a Sound

Velocity is another way to describe loudness, which in the case of a piano would be determined by how hard a key is hit. With a piano, the velocity (the loudness) of a note has an effect in three main areas:

the volume—there is a direct link between how hard a note is struck and its volume

the tone—the harder a key is hit, the brighter the tone of the resulting note, and

the sustain time of the note—the louder the note when it is struck, the longer it will sustain.

Effect of Pitch on a Sound

Clearly the piano keyboard controls the pitch of the note played. However, there are other factors affected by the pitch:

the tone—the higher the note, the brighter it sounds, and

the sustain time of the note—the higher the note, the less time that it sustains.

Effect of Time on a Sound

The length of time that a note is held is also a factor in the tone of a piano. Over time:

the sound will get duller (however bright it may have been initially), and

the volume will decay until it finally reaches zero.

Combining the Effect of Velocity, Pitch, and Time

A piano note is far more complex than may be suggested here, however, in broad terms you can see that:

The initial volume of the note is affected by one factor only—velocity. After that, the player has no control over the loudness of the note as it decays naturally.

Tone is affected by three factors—the initial velocity of the note, the pitch of the note, and the length of time it is held. However, the tone increases in line with the increase in pitch (so the higher the note, the brighter it becomes), but the sustain time decreases in line with the increase in the pitch (so the higher the note, the less time it sounds for).

Sustain time (and also volume over time) is affected by two factors—the starting volume (which is in turn controlled by the initial velocity) and the pitch.

To mimic some of the characteristics of an acoustic instrument, we're going to need to use modulation to control the synthesizer.

Modulation Destinations

We will approach modulation in a slightly backwards way and look at modulation destinations before we look at modulation sources. Once we understand what can be modulated, we can work back to choose the best source to create the modulation.

I should point out that the following list of modulation destinations is, of course, not complete: it only covers the main destinations that you are likely to want to control. Different synthesizers will offer different amounts of control, so on any specific synth you may be able to modulate more or fewer functions than are set out here.

Volume

Volume is likely to be one of the main destinations you modulate. You may want to:

control the volume of your whole patch—for instance to make the patch touch sensitive or to add a tremolo effect, or

control the volume of individual oscillators (before or after the signal passes through a filter) so you can cross-fade between two sounds.

Filter

The two main modulation destinations in the filter block are:

the cut-off frequency, and

the resonance.

Often musicians will want to control both in real time—the cut-off frequency to make a sound brighter or duller and the resonance to add some sharpness and bite to the sound (or to add some squelch).

Pitch

The effects that can be obtained by modulating pitch include vibrato and trills as well as weird special effects. Subtle pitch modulation is also important when trying to mimic the properties of natural instruments—for instance a slight and very short rise in pitch at the start of a note will often help recreate part of the attack of a brass instrument or a plucked string instrument.

Vibrato Speed

When a musician applies vibrato to a real instrument, both the speed and the depth of the vibrato increase—it therefore makes sense that LFO speed should be a modulation destination (and the depth of the effect should also be controllable).

Pulse-Width

A fast way to change the tone of an oscillator or to fatten up a sound is to play with the pulse-width modulation.

Pan

The classic use for panning is auto-pan type effects where the sound moves regularly between the left and right channels. There are however more subtle uses of panning, for instance panning can be related to the pitch of a note with low notes panned towards the left and higher notes towards the right as if you were sitting in front of a piano.

FX

The level of any FX units can be modulated to give prominence to an effect. If you want to get more complicated, you may want to change FX controls in real time—for instance increasing chorus depth as vibrato increases, or simply tweaking the delay time.

Wavetable Position

Massive offers a range of wavetables. Instead of single wave source, by using a wavetable and making the wavetable position a modulation destination you can step (or sweep) through a range of different waves in order to create a shifting tone.

Envelope Speed

I've already talked about key tracking (using pitch as a modulation source) and I'll talk about it in greater detail in the next section (Modulation Sources). One frequent modulation destination for key tracking modulation is the envelope, and in particular, the speed of the envelope (even for entirely synthesized sounds) where key tracking can give the sound a more natural quality.

As a side note, Massive allows you to apply key tracking to a whole envelope where Twin applies the modulation on a per-parameter basis (so for instance you can modulate the attack time but leave the decay time unaffected).

Modulation Sources

Most synthesizers offer and array of modulation sources, each of which operates in a different manner. There are also external sources which can act as modulators—such as MIDI velocity, the pitch bend wheel, and the modulation wheel. These sources can then be used directly as modulation sources or as controllers to control the amount by which another modulation source is modulating a destination.

The modulation sources (or controllers) fall into four main categories:

Single-shot sources. An example of a single-shot modulation controller is an envelope—it begins, executes its level changes, and then ends (after perhaps sustaining for a while). Envelopes are discussed in greater detail in Chapter 5: Envelopes.

Continuous sources. An example of a continuous source is a low frequency oscillator—the modulation source repeats its cycle continually modulating the destination. LFOs are discussed in greater detail in Chapter 6: LFOs and Specialized Modulation Sources.

Fixed. Some modulation sources, while being variable, are fixed once a note has been played. The main example of this is velocity—once a note has been struck, the velocity is fixed.

Variable. Other modulation sources are constantly variable after the note has been struck. One example of a variable source is the modulation wheel which can be continuously tweaked in real-time.

A second aspect to think about in relation to modulation sources is polarity:

Unipolar. Some sources (for instance velocity, the modulation wheel, and conventional envelopes) are unipolar. This means that their output within a single cycle is either positive or negative.

Bipolar. By contrast, bipolar sources (such as LFOs and the pitch wheel) are capable of generating both positive and negative signals within the same cycle.

Let's have a look at some of the modulation sources in a bit more depth (as already noted, envelopes and LFOs, including more specialized modulation sources, are discussed in subsequent chapters).

Velocity

When creating sounds, velocity is often used to control:

the loudness of a note

the tone of a note by modulating the filter cut-off, and

the (volume or filter) envelope attack time (if the synthesizer allows you direct control over this parameter, which both of the featured synthesizers do).

Velocity can either be applied directly to modulate the destination, or it can be used as a modulation source controller to control another source (for instance, an envelope parameter) which, in turn, is modulating the destination.

Pitch Bender

The pitch bender is often hardwired to the pitch bend function. However, most synthesizers also allow the pitch bend signal to be assigned to control other functions.

Modulation Wheel

Conventionally the modulation wheel is wired to the vibrato depth and so acts as a modulation controller.

However, it doesn't have to be so—the modulation wheel can also be used as a modulation source (for instance, to control the filter cut-off frequency).

Key Tracking

Key tracking (or key scaling) is the process where changes are made according to the pitch of a note.

For example, if you pluck the bottom E string of an acoustic guitar, the sound will last for over 10 seconds. However, if you play the top E string at the highest fret, the sound will barely last for 3 seconds. With key tracking we can seamlessly control the length of an envelope to reflect the behavior of the acoustic instrument.

Key tracking can be applied in many ways—for instance, the cut-off point of a filter can be controlled by key tracking so that higher notes are brighter than lower notes.

Most synths allow you to set pitch as a modulation source. This will then apply the modulation in a linear fashion across the keyboard. Some synths (Massive is one example) allow you much finer grained control. Continuing with Massive as an example, it allows you to manage the key tracking response by creating up to 16 breakpoints across the keyboard (see Figure 4.1). Where simple pitch-to-destination key tracking provides a linear response, with Massive you can create and control the exact response you want.

Figure 4.1 A unique key tracking response controlling the filter cut-off in Massive.

If you're looking for a quick, but slightly more practical, example of key tracking, then call up a noise source and run it through a low-pass filter. Put the cut-off frequency to a level that the noise is just heard and push the resonance amount to about halfway. Then set pitch as the modulation source and the filter's cut-off as the destination—as you increase the modulation depth, you will reach a position where the cut-off frequency follows the played note. When you reach this point, you will be able to play tuned noise.

The most obvious use for key tracking is to get the pitch of notes right—in other words, you want to know that when you hit a C key that a C note will sound or when you hit an E key that an E note will sound, and so on. This implementation of key tracking is so obvious that we often don't think about it. However, you can override this default assignment by setting key pitch as the modulation source and oscillator pitch as the modulation depth.

As with the filter, Massive also allows you to get away from the linear pitch response and tailor the pitch across the keyboard by creating up to 16 breakpoints across the keyboard.

X-Y Controllers

X-Y Controllers (see Figure 4.2) allow you to control two destinations from one controller—for instance, you could control a filter's cut-off frequency and the filter's resonance simultaneously.

Figure 4.2 Two of Twin's X-Y controllers (the one on the right is being adjusted—the one on the left has been set and continues to have effect).

Twin allows you any number of X-Y controllers.

X-Y pads can work as a modulation source or as a modulation source controller.

Aftertouch

If your keyboard (or other MIDI controller) gives aftertouch messages, then the pressure on the keys while a note is sustained can be used to control your patch. You could, for instance, use aftertouch to control volume to give a swell to a held string chord.

Aftertouch can either be applied directly to modulate the destination, or it can be used as a modulation source controller to control another source which is modulating the destination.

Expression Pedal

Often the expression controller is hardwired to a synthesizer's volume. As with the modulation wheel, it can also be used as a modulation controller.

MIDI Control Change

Lastly, MIDI Control Change messages (which are generated by external sources) can be used as a source or as a source controller.

Setting Up Modulations

Every synthesizer will set up its modulations in a different way. While the individual knobs to be clicked will change, the basic process is usually similar: select the modulator, connect it to the destination, adjust the amount of modulation, and then add a control (if necessary).

By way of illustration, let me show you the process in Twin to modulate the cut-off frequency in filter one with an envelope, and further to control the extent to which that envelope modulates the cut-off with MIDI velocity. The process goes something like this:

First, engage the filter and set its basic parameters (filter type, cut-off frequency, and so on). Clearly until the modulation is set up, any settings here will only be temporary.

With the filter set, the next step is to call up a modulator—in this case an envelope generator—by clicking on the large plus sign on the source bar (the central bar running across Twin's interface) and selecting New Envelope Generator from the pop-up menu. The envelope will then appear in the modulators section at the bottom of the interface.

To connect the modulation source (the envelope) to its destination (filter one's cut-off frequency) there are two options. You can click the large plus sign above the envelope to add a modulation slot—this slot can then be attached to the destination by clicking to the right of the slot to open the drop-down menu from which the appropriate destination can be selected. However, there is a much swifter and more intuitive option—click on the target (top-left of the envelope) and drag to the destination (the cut-off frequency knob) as Figure 4.3 shows.

Figure 4.3 Attaching a modulation source to a modulation destination by drag-and-drop in Twin.

Once you have attached the modulator you can set the envelope parameters. The depth of modulation—in other words, the amount by which the envelope modulates the cut-off frequency—is set by the slot slider above the envelope (labeled Filter 1 Freq in Figure 4.4). This slider and the filter's cut-off frequency should be balanced to create the desired sound. If you look at the cut-off knob in Figure 4.4, you will see that there is now a letter M to the top-right of the knob. This indicates that the parameter is being modulated. If you click on the letter M it will highlight the modulation sources.

Figure 4.4 Adjusting the modulation parameters in Twin.

If you want to remove the modulation, click on the cross at the end of the depth adjuster slot.

This modulation assumes a positive polarity (in other words, it assumes that you want to open up the filter with the envelope). If you want to close the filter (so that the sound gets duller rather than brighter) click on the plus sign on the left of the depth slot. This will change to a negative sign indicating that the output is negative.

The final step is to add velocity control. This will control the depth of the modulation by the envelope—at low velocities the envelope will have little effect, but at high velocities, the envelope will have a significant effect on the filter's cut-off. As before, to call up a modulator—in this case a MIDI source—click on the large plus sign on the source bar and select New MIDI from the pop-up menu. The new modulator will then appear.

In the MIDI source modulator, there are two drop-down menus—we're interested in the top one—from which we select Velocity. As a side issue, the bottom drop-down selects the curve—in other words the rate at which this modulator affects its destination.

The process to connect this MIDI source is the same as for connecting the envelope: drag-and-drop. However, is this instance, the destination is the envelope generator depth slot (in other words the slot above the envelope labeled Filter 1 Freq). Once attached, the interface should look like Figure 4.5.

Figure 4.5 Velocity set to control the depth by which EG2 modulates the filter cut-off.

All that is left is to tweak and play.

Controlling the Parameters with Hardware and Automation

You can also control your synthesizer—both to adjust individual parameters and to control the extent of any modulation—with other tools. The most common tools are hardware (for instance, a set of control knobs on your keyboard) and automation through your sequencer host.

The precise procedure to set up these linkages will depend on many factors (specifically, your host, your hardware, and your synthesizer), but let me quickly introduce the main issues.

MIDI Learn

The easiest way to control your synthesizer with your hardware is to use MIDI learn (provided your synthesizer offers this functionality). The process to engage MIDI learn differs for each synthesizer, but it usually goes something like this:

Click on the MIDI learn button (in Twin this is located on the left-hand side of the bottom bar).

Move the target parameter with your mouse.

Move the control knob on your hardware.

Switch off MIDI learn (which is usually a case of clicking the MIDI learn button for a second time).

Move the hardware to confirm that the routing has been learned. As you move your hardware control, the corresponding software control should move—if it doesn't, then you probably have a problem.

Automation

Most synthesizers expose their automatable parameters. Assigning the parameter to be automated shouldn't be any more difficult than finding the right parameter (often there's a long list) and drawing an automation curve.

More Modulation Sources

Now we've looked at the principles of modulation, let's move on and look at one of the most common modulation sources in more detail: envelopes.

Chapter 5: Envelopes

It's very easy to miss the importance of envelopes—they aren't very interesting and they don't make a noise. However, an envelope can have a more dramatic effect on a sound than a simple waveform tweak. Not only that, but an envelope can have a huge impact on a sound's character, and fundamentally, sound design is all about creating character.

So what is an envelope— Well, in short, it is a level that changes over time. Envelopes don't make any sound, but what they do is to change other parameters that affect the sound—such as the pitch, the filter, and the volume—as the note is played. This in turn animates the sound.

Envelopes are a one-shot modulation source. When triggered, the envelope begins, executes its level changes, and then ends. If you want continuous modulation, then rather than choose an envelope, an LFO (LFOs are discussed in the next chapter), may be a better choice of tool.

Volume Envelopes

The most immediate use for envelopes is controlling volume.

Think of a note played on a piano. When a key is struck, the note goes from silence to the maximum volume instantaneously. From this peak, there is an immediate reduction in the volume and then the note reaches a level from which it gradually fades to nothing. A picture of the volume of a piano note over time is set out in Figure 5.1.

Figure 5.1 A (simplified) volume envelope of a piano note.

Now if you think about a violin note which slowly fades in, stays at its maximum volume until the note ends, and then gently decays, this volume level may look like the image in Figure 5.2.

Figure 5.2 A typical sustained string envelope.

What Else Can Envelopes Do?

An envelope changes a level over time, but its effect will depend on how it is applied. For instance:

An envelope could control volume. Depending on the architecture of the sound, the envelope may control the level of an individual oscillator or the level of a whole patch.

An envelope may control a filter. If an envelope controls a filter, it will (usually) control the cut-off frequency and so change the brightness over time (usually making the sound duller over time, in other words, the sound will start bright and then get duller).

An envelope can also modulate pitch. A common use for this would be to impart a short (and subtle) pitch wobble at the start of a note to give the sound more emphasis. This technique can be used to create drum and percussion sounds.

As we saw in Chapter 4: Making the Sound Dance, these are not the only uses for envelopes.

Different synthesizers are designed in different ways and have different envelope options. Let's look at some of the more common types of envelope.

AD Envelopes

The simplest form of envelope is the attack-decay (or AD) envelope, see Figure 5.3.

Figure 5.3 The AD envelope.

There are two controls in the AD envelope:

A: Attack time. The attack time is the time it takes for the level to reach its maximum after the note is triggered. Using the example of a piano volume, the attack time would be zero—in other words it would take no time for the level to go from nothing to the maximum volume. For a string-type swell the attack may be slower.

D: Decay time. The decay time controls how quickly the level drops (to zero) after it has reached its maximum at the end of the attack phase). Again, using the example of the piano, the decay time would be comparatively slow (although with a piano, the decay of the level would not be linear).

When this envelope is used as a volume envelope, once attack phase and the decay phase have finished, the note dies. The note does not sustain in the manner that an organ note can sustain. This can make the envelope useful for percussion sounds but it may have limitations when trying to create string-like timbres.

Massive uses an AD envelope (which it calls an Internal Envelope) in its Osc section (see Figure 5.4) and LFO section. These envelopes have limited applications and are separate from the four main envelopes in Massive (which are more flexible and are discussed in greater detail later in this chapter).

Figure 5.4 The Osc section in Massive with its AD envelope (to the right).

ADSR Envelopes

The ADSR envelope (see Figure 5.5) is the classic envelope. A variant of this envelope is used in both Twin and Massive (and we will get onto the details of the differences shortly).

Figure 5.5 The ADSR envelope.

There are four main controls in the ADSR envelope:

A: Attack time. The attack time governs the time it takes for the level to reach its maximum after the note is triggered. Using the example of a piano, the attack time would be zero—in other words it would take no time for the level to go from nothing to the maximum volume. For a string type sound the attack may be slower.

D: Decay time. The decay time controls how quickly the level drops (to the sustain level) after it has reached its maximum at the end of the attack phase. Again, using the example of the piano, the decay time would be fast, but it would be longer than the attack time.

S: Sustain level. The sustain level is the level of the envelope while a key is held. This level stays constant until the key is released.

R: Release time. The release time is the time it takes the level to decay to zero after a key is released.

You will notice that with this envelope:

once the sustain part of the envelope has been reached the note sustains indefinitely—it does not decay to zero until the key is released, and

there is no function in the envelope to determine for how long the note sustains (the only control over this is by releasing the key).

Another weakness with this type of envelope (and this applies to all envelopes) is that real sounds do not necessarily increase or decrease in a linear manner. Take the example of a slow swelling violin—in practice the attack of the note is likely to have two phases:

first the note will go from nothing to a very quiet level very quickly, then

the note volume may increase more slowly.

This behavior is not easy to replicate with a synthesizer envelope.

Twin Envelopes

Twin has a (virtually) unlimited supply of envelope generators (see Figure 5.6)—if you want an envelope, all you have to do is call one up and assign its function.

Figure 5.6 An envelope generator in Twin.

There are six main envelope controls in Twin (see Figure 5.7) which are adjusted by dragging the six corresponding handles:

Figure 5.7 The DADSHR envelope.

D: Delay time. The delay time is the time after the note is triggered for which the level of the envelope remains at zero, in other words, the time for which the envelope has no effect. At the end of the delay time, the attack phase begins immediately. The delay time control does not appear on the main envelope generator in Twin.

A: Attack time. The attack time is the time it takes for the level to reach its maximum after the delay phase has ended.

D: Decay time. The decay time controls how quickly the level drops (to the sustain level) after it has reached its maximum (at the end of the attack phase).

S: Sustain level. The sustain level is the level of the envelope while a key is held. This level stays constant while the key is held and through the hold time.

H: Hold time. The hold time is the time after the key is released that the level remains at the sustain level before it enters the release stage. If the delay time and hold time are set to zero, then the envelope performs in the same manner as a conventional ADSR envelope.

R: Release time. The release time is the time it takes the level to decay to zero after the key is released and the hold time stage has completed.

You may also notice that there are some other controls in the envelope beyond the DADSHR handles:

On the top left corner is a target. You can drag-and-drop from here to set the modulation destination. Alternatively, hit the plus sign above the envelope generator which will bring up a depth slider from which you can manually select the modulation destination.

On the left-hand side are two buttons MIDI and side-chain—this is an either/or choice. Either the envelope is triggered when a key is hit (in other words, when there is a MIDI event) or it is triggered when an external audio signal is fed into the envelope. If this side-chain option is selected, then a threshold knob appears to adjust the level of the incoming signal.

On the right-hand side are the range buttons. When the top option is selected, the envelope behaves in a conventional manner and outputs positive level changes. When the lower option is selected, the envelope outputs positive and negative level changes—the sustain level is then set to zero.

Massive Envelopes

Massive has four main envelopes (see Figure 5.8).

Figure 5.8 An envelope in Massive.

While these envelopes can mimic the behavior of a conventional ADSR envelope, there are far more knobs (see Figure 5.9) giving detailed levels of control over your sound:

Figure 5.9 Massive's DADSR envelope.

D: Delay time. The delay time is the time after the note is triggered for which the level of the envelope remains at zero, in other words, the time for which the envelope has no effect. At the end of the delay time, the attack phase begins immediately.

A: Attack time. The attack time is the time it takes for the level to reach the level set by the (attack) level knob after the delay phase ends.

D: Decay time. The decay time controls how quickly the level drops to the (decay) level, (which is equivalent to the sustain level in an ADSR envelope) after it has reached its maximum at the end of the attack phase.

R: Release time. The release time is the time it takes the level to decay to zero after the key is released.

So far, the envelope is fairly conventional—the key difference from a conventional envelope, beyond the delay stage, is that the attack level can be manually adjusted. But there is a lot more that the envelope can do during its sustain period—instead of simply holding a note, it can behave in a manner akin to an LFO. Figure 5.10 shows the controls; let's have a look at what these do:

Figure 5.10 The sustain loop controls in Massive's envelope.

Loop. Under the two Morph drop-downs, there is a Loop control. When this is set to Off, the envelope behaves like a conventional ADSR envelope (in other words, when the envelope reaches this point and the key is held, the level remains constant). When set to Inf, the loop will continue indefinitely. When set to a number, the loop will play that number of times.

Level. The end level of the loop is set by the level control. This means that the level of the loop may increase or decrease between the decay level and this level. Since these two levels may be different, the loop does not cycle in the conventional manner, instead, it plays forwards and then backwards, and then repeats. Accordingly, when the loop number is set, the envelope may end at either point, depending on whether the loop is set to an even or an odd amount.

SLoop. The SLoop control adjusts the time over which the sustain loop plays.

Morph. At its most basic, the sustain loop will adjust its level between the decay level and the sustain loop level. However, that transition between the two points does not need to be linear. The two Morph drop-downs offer a range of different varying level paths—you can select one or both. If you select both, then the Morph knob will morph between the two (and if you only select one, this knob will morph between the chosen path and the linear path.

Key Tracking and Envelopes

I want to mention key tracking again, briefly. This concept is used in several areas. Essentially key tracking means dynamically changing an element of the sound depending on the pitch. So if we think about the volume envelope of a piano, as the notes get higher they sustain for a shorter period and conversely, as the notes get lower they sustain for longer.

If we want to replicate this behavior on a synthesizer we would use key tracking.

You can manually set this function in Twin by modulating each step of the envelope generator (in particular, the attack time, decay time, and release time). In Massive it is more straightforward—the KTR slider which is to the left of the envelope (check back to Figure 5.8) controls this function.

More Specialized Modulation Sources

A lot of nonsense is talked about sound sources—and don't get me wrong, sound sources are important—however, if you use your ears, I think you will find an envelope can change a sound just as much as changing an oscillator.

Now we've looked at envelopes let's move on and look at a more complicated source in more detail: LFOs and the specialized modulators.

Chapter 6: LFOs and Specialized Modulation Sources

A low frequency oscillator (or LFO as it is more frequently called) is a level that changes over time. As with envelopes, LFOs don't make any sound, but what they do is to change other parameters that affect the sound—such as the pitch, the filter, and the volume—as the note is played. This in turn animates the sound.

A low frequency oscillator is an oscillator which oscillates at a low frequency—for our purposes it usually oscillates below the audio threshold. In addition to working as a vibrato source, the LFO is also often used to modulate:

volume, to give tremolo effects, and

the filter cut-off, to give wah-wah or gating/stuttering type effects—especially when a square or pulse wave is selected as the modulation source.

You can, of course, modulate many other destinations with an LFO.

Envelopes are a one-shot modulation source. When triggered, the envelope begins, executes its level changes, and then ends. An LFO is an oscillator and so repeats its cycle continually modulating the destination. Because of this repetition, the effect of LFO modulation is often more prominent in sounds.

Main LFO Controls

The main controls on any LFO are wave shape and depth.

Wave Shape

Most synthesizers offer several LFO shapes; these are usually the same basic wave shapes that are available for audio oscillators.

The most frequently used LFO shapes are:

Sine. The sine wave produces an even, rounded modulation source.

Triangle. The triangle wave has virtually the same effect as the sine wave, but the changes are constant. This may give a crisper sound when compared to the sine wave. Both the sine and triangle are ideal waves for vibrato, tremolo, or wah-wah effects.

Sawtooth. Depending on the polarity of the wave, the sawtooth provides either a rising modulation source which drops abruptly after reaching the maximum value or a falling modulation source, rising to a peak and then gently falling. The wave is often used for modulating the filter cut-off.

Square. The square wave is either at its maximum or its minimum. This wave tends to be used as an LFO to give rhythmic effects. It can also be used to modulate the pitch to give trill type effects.

Random. Some synths offer a random or more commonly, a sample-and-hold wave (see Figure 6.1). The output from the a random wave is a random value, and for a sample-and-hold wave, the level of the wave steps from level to level—there is no smooth transition meaning that this is an excellent choice for creating rhythmic effects.

Figure 6.1 A sample-and-hold oscillator which is especially useful for creating rhythmic effects.

These are the most common shapes—many synthesizers offer a much wider range of options, and as we'll read, many allow you to manipulate the wave to create even more sonic effects.

Frequency

The frequency control affects the speed of the LFO. If an LFO is used as a vibrato source, then the frequency control will affect the speed of the vibrato. Most LFOs have a range of around 0.1 cycles per second to about 20 cycles per second.

It is also very common to allow an LFO frequency to be synchronized with the tempo of a track so that the time it takes the LFO to complete a whole cycle is a subdivision of the track's tempo. This option is often used in conjunction with rhythmic effects. Both Twin and Massive offer this functionality.

Depth

The depth setting of an LFO controls the extent to which the LFO affects the modulation destination.

When using an LFO to introduce vibrato you will probably have to apply less vibrato than a player would apply to a natural instrument. LFO vibrato usually has a fixed frequency (but this too can be modulated) and so the effect will quickly sound mechanical. Reducing the depth of the effect helps to stop some of the mechanical feel becoming too prominent.

This setting needs to be carefully balanced so that an appropriate range of its effect is controlled. For instance, if you are using an LFO to give some vibrato, you probably want a range which is less than (plus or minus) one semitone.

Sync

When a note is triggered, any LFO that is applied to that note could be at any stage of its cycle. Usually that's not a problem—indeed, this random element (with a free running LFO) may be desirable for ensemble type sounds—however, there will be times when you want to control where the LFO is in its cycle at the point when a note is triggered. For instance, if the LFO is modulating pitch, then you are likely to want to know—with certainty—the starting pitch when the note is triggered.

Sync is also an issue when you start adjusting the phase of an LFO.

Phase

For any control over an LFO's phase to be meaningful, you will want to synchronize your LFO (otherwise you're simply adjusting something that is random in order to create something that is still random, but random in a different way).

By combining sync and phase, you can set an LFO to start playing at any specific point in its cycle when a note is triggered.

Monophonic/Polyphonic LFOs

LFOs can be monophonic or polyphonic.

Monophonic LFOs

With monophonic LFOs, a single oscillator is used whenever that LFO is invoked. So if you are using an LFO to add some vibrato to a pad, when you play a chord all of the notes will have the rise and fall of their pitch synchronized. The effect may not be particularly natural in this instance.

Polyphonic LFOs

With a polyphonic LFO, each separate note has its own LFO which will start its cycle at a different time from any other LFO (assuming that you haven't engaged the sync function, which would defeat the purpose of using a polyphonic LFO).

Using our pad example again, using a polyphonic LFO to modulate a chord will mean that some notes will have their pitch raised while others are having their pitch flattened. This may give a more desirable result for a pad, but is less use for rhythmic sounds.

Wacky Waveforms

We've looked at the basic waveforms. Let's move on and have a look at some of the more unusual waveforms, or more importantly, some of the more interesting implementations of LFOs.

All synthesizers have their own unique approach to implementing waveforms, so for this section, I will use Twin and Massive as examples of some of the interesting implementations.

Twin LFO Balance

In Twin, around the LFO frequency control is the balance control ring—this affects the shape of the LFO.

Turned fully to the left and a falling sawtooth wave is output as a modulator; in the center, a triangle wave is output; and fully to the right, a rising sawtooth is output. In between these extremes the shape morphs between the three points.

This arrangement gives both a falling and a rising sawtooth as well as offering many options between these two extremes.

The Glide knob can further mess with the LFO shape, adjusting the sharpness of the transition from maximum to minimum. In practice this can give the LFO waveform more of a square wave character.

The Glide knob will also affect transitions in hand-drawn LFOs. On which subject...

Twin Hand-Drawn LFOs

Twin allows you to draw your own LFOs.

On the right of the LFO block is a plus tab (see Figure 6.2). Click on this to add node points (up to 16). Once the nodes have been added, they can be tweaked (either dragged-and-dropped or the value knob controls the selected node).

Figure 6.2 A hand-drawn LFO in Twin.

The balance knob continues to work—emphasizing one side or the other of the wave, and the glide knob behaves as expected to control the transition. In addition, the glide knob to the right of the hand-drawn LFO acts in a similar manner to the master glide knob, except that it applies to individual transitions, not all transitions.

The small button with the image of a keyboard is the LFO snap button. If you engage it, the value of each node is snapped to the nearest position on the keyboard. This allows you to use the LFO as a sequencer—each note will then be correctly tuned.

By the way, if you want to remove a node point, select it (click on it) and click on the minus tab to the right of the LFO.

Morphing LFOs in Massive

Massive, as you might expect, has a rather unique take on LFOs.

It has the usual range of LFOs (sine, sawtooth, square, and triangle), and you can play the usual games (sync, mess with phase, polyphonic LFOs, and so on). And of course, it has a range of other LFO shapes to give some really interesting sonic textures.

But it doesn't stop there... Instead of offering a single LFO (monophonic or polyphonic), each LFO has two LFO sections (see Figure 6.3) and you can cross-fade between these two waves to create an even broader range of effects.

Figure 6.3 The LFO block in Massive with two cross-faded LFO waves selected.

Specialized Modulation Sources

Beyond the LFOs, Massive offers two specialized modulation sources intended for rhythmic effects: Stepper and Performer. To access either of these, open an LFO and from the drop-down under the top row of green LFO buttons, you can select the appropriate tool.

Stepper

The stepper (see Figure 6.4) is a modulation source with a number of steps (up to 16). Each step is evenly spaced (which makes this tool particularly effective when the steps are synchronized to the beats of a bar.

Figure 6.4 The stepper in Massive.

The level in each step can be set individually (hence each beat can have an individual level reinforcing the rhythmic element). The stepper is then attached to a modulation destination (typically a filter cut-off or a volume amplifier) and off you go.

A similar effect can be achieved in Twin using the LFO (check back to Twin Hand-Drawn LFOs). Indeed, Twin has more flexibility in some ways since you can control the level changes between each step. However, Twin's LFOs do not offer the same flexibility that is available in Massive's performer.

Performer

The performer (see Figure 6.5) is similar to the stepper in that there are a number of steps (again up to 16) which can be assigned to a range of modulation destinations.

Figure 6.5 The performer in Massive (with the Load Curve option selected—this is used to select the curve at each step).

The key differences between the two are:

With performer the level of each step is a curve (in other words, the level can change during the step).

There are two rows, so you can set up two completely different patterns and morph between the two.

Now Polish!!

The LFO is the last basic element that I'm going to look at. In the next chapter we'll move on and look at how to add some polish with effects.

Chapter 7: FX

We've looked at the elements used to create a sound using analog synthesis techniques. Now we'll look at how to polish those sounds and how to extend your sonic palette further using effects units.

The FX units you find on today's synthesizers are generally used in two ways:

as an integral part of the sound for instance, a distortion unit, or

to enhance a sound, for instance, reverb to give a sound some space and depth.

We are going to look at both options for shaping your sound.

Deployment of FX

FX units are deployed in one of two ways: as insert FX units or as send FX units. There is a practical difference between the two.

Insert FX

With insert FX the whole of the audio signal passes through the FX unit (see Figure 7.1).

Figure 7.1 The signal flow through an insert FX unit.

If you want to add a distortion effect, then you are likely to want the whole signal to be distorted by the fuzz box. EQ and compression are other typical examples of insert FX.

Master FX

Some synths (for instance, Massive) offer master FX units. There is nothing intrinsically different about these units or the signal architecture: they are insert FX units located immediately before the synth output so that the processing is applied to the whole signal.

Send FX

With send FX, you send part of your signal to the FX unit and then mix an effected signal back with the dry sound (see Figure 7.2).

Figure 7.2 The signal flow with a send FX unit.

Examples of FX units that are grouped under the send FX heading are modulation effects (chorus and so on), delays, and reverbs.

Distortion

Most people are used to hearing distortion used in an extreme way. However, depending on the amount of control you are given over the distortion, it can be used in a much more subtle manner.

Distortion units also come in different flavors—in broad terms, digital distortion and analog-emulating distortion.

Old School Overdrive

If you've looked at Twin, you'll notice that it doesn't have a distortion unit. Indeed, it only has one FX unit, the delay. However, like many other synths, it is capable of creating a very broad range of overdrive and distortion sounds.

Twin has 11 different filter models, each of which reacts differently to the signal output from the oscillator section. Some of those filters will mimic the behavior of overdriven analog gear. So, for instance, if you push a sawtooth wave into the Extreme or Raw filter, you will hear an overdriven sound. This is a light, subtle distortion, but is highly effective for creating a spectrum of warmer, richer tones but stopping short of creating the effect heard with a highly driven distortion unit.

The effect is very much dependent on a number of interacting factors, specifically:

the output from the oscillators (the waves chosen, whether the waves are synced, and the output level of the waves)

the filter type, and

the settings on the filter.

While this approach gives a much less obvious distortion sound, it is more subtle and gives a more controllable tone that dedicated distortion units struggle to achieve.

Analog-Emulating Distortion

Analog-emulating distortion is (usually) designed to give a gentle overdrive (usually a level of distortion beyond filter overdrive, but not reaching out-and-out shredding distortion). At lower levels its effect is heard as a warming/fattening of the sound, rather than as distortion.

Most distortion units have fairly simple controls, usually with only a drive control and occasionally with a tone control.

Massive has a number of overdrive/distortion units, each has a different character distortion. In addition to the drive knob, Massive also has a dry/wet mix knob so that the clean and distorted signals can be mixed.

Digital Distortion

By contrast to analog-emulating distortion, digital distortion is (often) quite harsh and cold, lacking the warmth/fattening of digital distortion. You will find these units with a number of labels; typically they are called digital distortions, decimators, lo-fi, bit crushers, and sometimes alias units.

To understand what these units are trying to achieve, you need to understand digital audio. There are two factors that determine the audio fidelity of a waveform generated by a synthesizer (or any other digitally generated sound for that matter):

the bit rate—the number of times that the signal is generated per second, and

the sampling rate—the number of levels of loudness that are allowed.

The two factors in combination affect the quality of a digital waveform. Take audio on a CD as a practical example of this: the sound you hear has 65,536 (2^16) levels of loudness and each level of loudness is recalibrated 44,100 times per second. If you reduce the bit rate and the sampling rate of a CD, then the quality would fall as Figure 7.3 shows.

Figure 7.3 As the bit rate and/or sample rate are cut, the resolution of the wave—in other words, its audio fidelity—will fall. This drop in fidelity is perceived as distortion which, with care, can be controlled for creative effect.

Digital distortion units will allow you to control (within the FX unit) either or both of the bit rate and the sampling rate, and sometimes also the tone of the output. Generally the distortion that these units create is pretty harsh and hard to control, but if you want a really brash overdrive, these are the units to head for.

If you do want more control (and your synth allows you to move the effect within the signal chain in the way that Massive does) then put the digital distortion unit before a filter, so that the filter can smooth off some of the rough edges.

Compression

Compressors were originally used to prevent an analog signal overloading an analog input, and indeed, compressors still are used for this purpose.

In essence, a compressor is an automatic gain control: when a signal gets to a certain level the compressor restricts how much louder the signal can get (see Figure 7.4).

Figure 7.4 A compressor controlling an input signal.

Most compressors allow you to control:

Threshold. The threshold is the level above which a signal is attenuated.

Ratio. The compression ratio controls the extent to which a signal above the threshold is allowed to increase in level. So for instance, with a 4:1 ratio if the input signal increases by 4dB above the threshold, the output signal will only increase by 1dB above the threshold.

Gain. When a signal has been compressed, it will be quieter than the input signal. The make-up gain allows the output level to be rebalanced to an appropriate volume.

In practice, you're quite unlikely to want to use the compressor as a volume control and instead will use it as an effect where it can help to:

make a sound fatter and/or smoother

enhance the perceived loudness of the sound, and

give the sound more punch.

The disadvantage of using a compressor is that the resulting sound can tend to dominate a mix. With more extreme compressor settings, the sound will usually have a more uniform level and will fill a broader frequency range. Both of these factors can make it harder to mix a highly compressed synthesizer sound. Another downside of compressor abuse is that the resulting sound tends to be less interesting—the delicate harmonic shifts of a patch tend to get covered up as all of the harmonics take a similar level.

EQ

The purpose of equalization is to make certain elements of the sound spectrum louder and other elements quieter. You can use EQ creatively or surgically. However, remember that you still need to get your sound to sit in a mix, so go carefully with the EQ, and try not to use the effect to cover the faults of poor programming. Also, remember that you may not hear the full effect of any EQ tweaks unless you are using full-range speakers.

The most flexible form of EQ is the parametric EQ where you can set:

the frequency to be boosted or cut,

the amount of boost or cut, and

the bandwidth (or Q) of the boost or cut. The bandwidth determines how wide the boost or cut is, much like a constantly variable slope control for a filter.

A simplified form of EQ is the shelf EQ. A shelf EQ will cut or boost:

the whole signal above a specified frequency in the case of a high shelf, or

the whole signal below a specified frequency in the case of a low shelf.

Take a look at Figure 7.5 which shows a low shelf EQ (on the left), a high shelf EQ (on the right), and a parametric EQ in the middle.

Figure 7.5 A low shelf, parametric, and high shelf EQ.

Some more simple EQ controls allow you to tweak what is labeled as tone. These are usually high shelf EQ units and the tone control allows you to set the amount by which the signal is cut. It is rare in a simple tone control, but occasionally some units also allow you to boost the signal.

Filter

We looked at filters in Chapter 3: Filters.

Many synthesizers include filters in the FX sections (as one example, Twin includes filters). These usually operate in the same manner as those described in Chapter 3, except often (but not always) they apply to the whole sound, not to individual notes (meaning that, for instance, you may not be able to apply individual modulations).

The other issue about filters as FX units is their location in the signal chain. Often you can place them at a more appropriate point than the main filters, for instance, you can place the FX filter immediately after a distortion effect.

Modulated Delay Effects

Modulated delay effects came to the fore around the time that analog synthesizers moved to the mainstream.

These units take many different forms (chorus/flanger/phaser, and so on) but they all do a similar thing: in essence they mix a delayed, pitch-modulated signal with the original signal to give a fuller/warmer/richer/whooshier (insert whatever description you want) sound. The delay time for the delayed signal is so short that you don't hear it as an echo, simply as a cohesive changed tone.

While the operating principles are the same, the sound the different units produce can be markedly different. Although the delays are so short the delayed signal cannot be separately identified, the delay time has a major effect on the sound.

Used sparingly, these effects can give an element of spark or richness to a sound. Used to excess and you will create a sound that quickly becomes passé.

Phaser/Flanger

Words cannot readily describe the difference between the sound produced by a phaser and the sound produced by a flanger, however, to my mind a phaser gives a more nasal sound where a flanger gives a more whooshy type sound. One of the reasons for the phaser sounding more nasal is that the delayed signal passes through a notch filter (which has its frequency modulated).

Flangers and phasers are often used on percussion sounds and with plucked-type sounds (such as guitars and electric pianos). The sound is very distinctive and is hard to use in a subtle way, so you are likely not to want to use the effect too frequently. The sound also has a 1970s retro quality to it—whether you love or hate that era is likely to have an influence on how you feel about phasing and flanging.

The main controls you are likely to find on a phaser or a flanger may include:

Delay. The delay control will adjust the time of the delayed signal.

Feedback. The feedback control sets the number of times the delayed signal is repeated. This has an effect on the perception of depth of the effect.

Rate. The rate control adjusts the frequency of the low frequency oscillator modulating the pitch of the delayed signal.

Depth. The depth control adjusts the extent to which the low frequency oscillator modulates the pitch of the delayed signal.

Chorus

Chorus units also mix a delayed signal with the original signal, however, they use a longer delay than would be typical with a phaser or flanger. This delayed signal is then modulated by a low frequency oscillator.

The result can be a much more lush/smooth sound—at extremes however, it can sound like vibrato played too fast or just bad tuning.

Used in moderation, the effect can either add sparkle or thicken up a sound without noticeably changing the sound's characteristics.

The main controls you are likely to find on a chorus are similar to those on a phaser or a flanger and may include:

Delay. The delay control will adjust the time of the delayed signal.

Feedback. The feedback control sets the number of times the delayed signal is repeated. This has an effect on the perception of depth (or out-of-tuneness) of the effect.

Rate. The rate control adjusts the frequency of the low frequency oscillator modulating the pitch of the delayed signal.

Depth. The depth control adjusts the extent to which the low frequency oscillator modulates the pitch of the delayed signal.

Tone. The tone control will adjust the tone of the delayed, modulated signal to take away any harshness or metallic ringing.

Modulation Effects with Twin

Twin does not offer any sort of modulated delay effects: it has a "plain" delay. However, it is quite possible to create an array of modulation effects with Twin.

Twin has a number of delay options—if you select ultra short and modulate the delay time with a low frequency oscillator, then you will create a chorus-like effect. While there is no way to directly modulate the pitch of the delayed signal, modulating the delay time has a similar sonic effect.

As Twin offers two delays, you can set a separate LFO to modulate the time of the second delay to create a richer sound.

A General Word of Caution about all Modulation Effects

All modulation effects tend to have both a brightening and a thickening effect on a sound. This means that the effected sound can dominate the sound spectrum to a greater extent than the dry sound. Be judicious in your use of these effects if you are applying them to a sound that is intended to be used in the background—it may become more of a feature than you intended. If you are applying modulation effects to a background sound, then look at your arrangement and consider thinning out the arrangement to ensure the modulation effect does not become dominant.

Delay

Virtually every synthesizer with built-in effects will offer some sort of delay.

Usually there will be two ways to set the delay time:

The delay may be set as an absolute amount, usually defined in milliseconds.

Alternatively, the delay time may be linked to the tempo of the track, being defined as a fraction of the beat (for instance, with delay of 1/4, the signal is delayed by a quarter note).

Often several different types of delay unit are available—the most common are mono, stereo, and cross delay.

Mono Delay

A mono delay adds a simple delay to the input signal. A mono delay can be achieved in a stereo delay unit if the delay time is equal for both channels.

Twin offers two mono delay lines (although there is the facility to cross feedback, see Cross Delay below).

Stereo Delay

With a stereo delay, there are two separate delay lines which may (or may not) have different delay times. Conventionally these delay lines are hardwired left and right.

Cross Delay

Like a stereo delay, a cross delay has two channels. However, the output from the first delay is fed into the second and the output from the second is fed into the first. This can give the effect of spreading the delays across the stereo spectrum or ping-ponging the delays between the two channels.

Common Delay Controls

The most common controls that all delay units have are:

Delay time. As noted, delay time is often (but not always) synchronized to the host's tempo and expressed in terms of beats.

Feedback. Feedback controls how many times each delay repeats (quite literally, it controls how much of the delayed signal is sent back to the input).

EQ, damping or a filter. EQ, damping, and a filter may be used to control the sound of the echo. Typically these controls will be used to cut the high frequency elements of the delayed signal to make the sound duller, resulting in a more natural sounding echo.

Using Delay Lines

When adding delay, you need to consider what you are trying to achieve. If you are adding the slightest touch of slap-back just to give a bit of spark and a bit of space, then make sure you have done that. If you want to add a bit of echo, make sure that the delays are duller than the original source. If you are after an effect, then, as with the modulation effects, moderation is key.

Like the modulation delay effects (and reverb), you also need to be cautious about how much space in the sound spectrum an echo will fill. There are several ways to tame the echo so that the sound doesn't dominate. The two most common are by reducing the feedback and cutting the volume of the echo, and filtering (both high-pass and low-pass) the delayed signal so that the echo fills a narrower portion of the sound spectrum.

While an echo may dominate the sound spectrum, I often find that it dominates less than reverb while giving more character to the sound than reverb. I also find echo brighter and less mushy. For this reason I will often use echo rather than reverb when building patches. However, this is very much just a preference of mine.

Reverb

Reverb can give a sound some spatial context.

With reverb units, there is significant interaction between the controls, so even more than normal it is necessary to balance several parameters when shaping the sound. The most common controls offered by reverb units are:

Pre Delay. This controls the time between the original signal and the reverb taking effect. Smaller values give the perception of a smaller room while longer delays give the perception of larger rooms.

Size (or Room Size). This controls the size of the room being simulated by the reverb unit and has a direct effect on the decay time of the reverb (in other words, the time for which the reverberation effect sounds). Occasionally some units will offer a separate decay control to adjust the decay time.

Damp. Damping works by cutting the higher frequencies in the reverb signal. One of the reasons why electronic reverb sounds artificial (that is clangy and metallic) is because it has too much high frequency information. The damping control can help to create a more realistic simulation.

Width. If a width control is present, it will adjust the stereo width of the reverb.

Using FX in Your Own Sounds

We've now looked at the sonic elements and the FX units to polish and enhance our sounds. It's time to move on and start creating sounds.

Chapter 8: Sculpting Sounds with Synthesizers

When it comes to creating sounds using analog synthesis techniques, there are no hard-and-fast rules. Taste is more important than theory and your taste will almost certainly be different to mine. Accordingly, this chapter is intended only as an introduction, a starting point, to synthesizer sound design. You will notice that there are no patches accompanying this book—that is intentional, this chapter is all about using your ears and finding what sounds right to you rather than replicating what sounds good to me.

There is no magic formula to creating a sound. However, I would suggest a few basic steps before you jump in:

Understand what each knob does.

Understand the signal flow in your synthesizer and how each element acts and interacts with the other elements.

Have a definite aim about the sound you're trying to create—don't just fiddle with the knobs and hope something useful comes along.

Once you have these basics sorted, I suggest a two stage approach:

Get something that is "alright". In other words, create a sound that is functional and is reasonably close to where you want to end up—you should be able to get to this point quickly.

Then undertake the detailed tweaking. This is the time to make sure the sound is exactly right for your needs—and will be the time consuming part of the programming.

Now you know how your synthesizer works and you have a grasp of what all the knobs do, so how do you get the basic sound together— For me, I then tend to think in terms of the main food groups.

Main Food Groups

Most sounds can be characterized by two key elements—their brightness and the envelope (primarily the attack of the envelope, but often you need the decay and sustain level to be right for the contrast of the attack to be clear). Once these two elements have been addressed, any tonal nuances after that may be affected by a range of factors.

Brightness

If you are trying to categorize the brightness of a sound, it will generally fall within one of four descriptions:

very bright

bright

dull, or

very dull.

You can try to subdivide further, but realistically, it is already quite difficult to try to find the line between these four categories.

Attack

If you try hard, you can find four categories of attack time:

fast—percussive (for instance a xylophone hit or a guitar string being plucked)

fast—but not percussive, for instance an organ

medium, and

slow

Again, you can try to subdivide the categories further, but it is a fairly pointless exercise.

Getting the Combination Right

Once you understand what you want in the way of brightness and envelope, getting the right sound is dependent on the right elements being drawn together. For instance, here are some combinations may want to look at:

To Get Brightness

To get brightness, look at the oscillator (or oscillators) and filter combination. If they don't work together, you will either get a shrill sound, or a thin sound, or a dull sound, but never a bright sound. For instance, if you are creating a sound based on a square wave, you may find it preferable to choose a 12dB filter. A 24dB filter may work, but to my ear the steeper filter can rob the square wave of much of its character which you may want to keep if you are trying to create a bright patch.

To Get Richness

One of the key elements to a rich sound is its movement. You can achieve this with careful selection of your sound source (which will usually involve several slightly detuned oscillators and perhaps some delicate use of a sub-oscillator). Alternatively, you can just slap on loads of chorus. My preference is for the former option.

To Get Attack

Most obviously, it is important that the envelope opens quickly to get some attack in a note. Depending on the tonal quality you are after, it is often important that the note then decays quickly to the sustain level—this will give you a percussive envelope reminiscent of a plucked guitar string or similar.

However, to ensure that the attack is emphasized, another significant factor is the waveform. Some waves can sound slower than others. For instance, all other factors being equal, a square wave can sound like it has a faster attack than a sawtooth wave. In general bright sounds (especially aggressive, distorted bright sounds) are perceived as having more attack.

To Get Warmth

A warm sound is subtly different from a rich sound: often it is much less bright. Warmth is also a comparative term—if a colder sound starts a patch and is then washed over by a warm sound, the contrast can give the listener the illusion of even deeper (hotter?) warmth.

Any warm sound will probably have a slow attack and a slow release. There must be thickness to the sound which will often come from detuned oscillators.

To Get Punch

It is hard to get some punch into a sound without adding too much brightness. First you need a fat sound source, then you need a fast attack and finally you need a hefty filter. The filter can be modulated with the envelope so that the attack is emphasized but the brightness is controlled.

Building Patches

Before we look at how to create sounds, let me offer a few last considerations.

Designed For Purpose

Any sounds you create should be created for a specific purpose (even if that purpose is just to have fun or to hear how something sounds). If you are creating sounds for a specific purpose—for instance, to work within the context of other parts—you stand a far better chance of getting the right sound which then fits in the mix.

Although you might have designed a sound for a specific purpose, that does not prevent you from using it in another context. Indeed, you may find that, for instance, a bass sound is a good start point for creating a stab sound, and so rather than starting from a plain sound you can simply tweak an existing sound.

Context of a Mix

It doesn't matter whether the sound you create is good or bad when you listen to it in isolation—it matters whether the sound works when you deploy it in context. In other words, it only matters how it sounds in the mix.

When you are looking at how sounds work within the context of a mix, ideally each sound will have its own place in the frequency spectrum, and will be clear and distinct. You may not always want that clarity, however, you will be more likely to have your mixes criticized for being "muddy" than for being "too sparse".

Choose the Right Tools

When you are programming take some time to make sure you choose the right tool.

While I would never discourage you from trying, there are some tools that are better suited to certain jobs, and sometimes, analog synthesis is not the right answer. For instance, although it is quite possible to create bell-like tones using analog synthesis, I think you will get better and more consistent results by using frequency modulation synthesis. Equally, if you want to recreate the sound of a piano, don't waste your time trying to choose the right oscillator/filter combination and tweaking envelopes—use a set of piano samples.

More importantly, choosing the right tool is also a function of having the knowledge to control that instrument. The only way to learn how to use an instrument is to keep using it over a long time—don't keep buying new instruments, pick one synth and learn every aspect of its character.

Emulating Real Instruments

When it comes to creating sounds, a first step for many people is to attempt to recreate the sonic textures of real instruments.

I don't like recreating real instruments with analog synthesis techniques for a simple reason: there are better options. If you want to create a piece that uses real instruments, then there are two options if you want to something that will not be noticeable to the listener:

Use real instruments. If you want the nuance of a piano, then using a piano will give you the best sound—there's no way that a synthesizer will sound better than the real thing.

If the real thing is impractical, then use a set of high quality samples.

Another reason for not trying to replicate real instruments with analog synthesis is that it is impractically difficult. With a real instrument the musician has a virtually limitless tone palette. While synthesizers have an incredibly wide tone palette, you can realistically only create one sound at any one time (or if you use some modulation, a limited range of sounds). What you cannot readily do with a synthesizer is replicate the choices that are open to a player—for instance, you cannot replicate a simple change from a bowed note to a plucked note as a violin player may make. The only way to create these ranges of tones is to synthesize each one separately... and that is where using a synthesizer becomes less practical, especially if you are trying to synthesize a whole performance.

A correlation to the range of tones that any one instrument creates is that real sounds are usually created by a range of instruments. If you look at an orchestra, there isn't a single "string" instrument. Instead, there are a range of violins, violas, cellos, and basses, each one of which has its own unique range of tones. It is the combination of these tones that gives the final sound that we hear as "strings".

That being said, some synthetic sounds can work where a real instrument (or group of instruments) would usually be heard. For instance, a string section or a brass grouping can be synthesized. This approach will work, provided the sound is used with some care—if you are creating a string sound then you will often be able to create pads (in other words, chord sections which don't move too much). Trying to create a fast moving string section is less likely to be successful.

While I'm not an advocate of reproducing real instruments, for descriptive purposes, it's often very useful to reference real instruments, and the characteristics of real instruments, when trying to describe sounds, and so I will make these references (as you will read).

Playability

Finally, any sound has to be highly playable. The playability of the sound is one aspect that separates the average sound from the professional. Use your knowledge of how the sound will be deployed to set the modulation sources.

For instance, if the sound is going to be played by a live musician, then you might ensure that velocity has an effect and that the modulation wheel and expression pedal both offer a wide range of control. However, if the MIDI track controlling the sound is going to be programmed, then you might want to make sure that external controllers, such as track envelopes may have a significant effect, so for instance, you may add an additional filter which has little effect in the normal course but can be readily controlled by a track envelope.

Building Basses

Let's start by looking at some bass sounds.

Simple Bass Sounds

The simplest—and in some ways the most effective—bass sounds can be created with a single oscillator, one filter, and an envelope. With these three elements it is virtually impossible to fail to create an effective bass sound, however, the sound that is created might not necessarily be the sound that you are after, so we'll look at other options in a moment.

For simple bass sounds, I usually start with a basic wave and build, but there are a wide range of factors which will affect the sound:

Waveform. There are two obvious waveform choices: a sawtooth wave and a square wave. A sawtooth wave will usually give you a thicker sound and a square wave will give you a more hollow sound—choose whichever is right for the sound you're after.

Cut-off frequency. The cut-off frequency affects the brightness of the sound, and as you turn down this control, it also thickens up the sound.

Filter slope. The filter slope will have an effect on the character of the sounds. In broad terms, a shallower slope (for instance a 12dB/octave filter) will have a brighter/snappier character where a steeper slope (such as a 48dB/octave filter) will have a thicker sounds. However, these general characteristics are very much dependent on the interaction with the other factors.

Resonance. The resonance control can make the sound thinner, but it can also make it brighter giving more emphasis to each note, particularly when coupled with a filter envelope. At extremes, resonance can get rather annoying.

Envelope destination. When building a basic bass sound, the logical destination for the first envelope is the filter's cut-off frequency. However, this is not the only choice, for instance, you can attach the envelope to control the volume of the patch, but if you do this, then the sound will be far more static (as there will be no movement in the filter).

Envelope depth. Assuming the (only) envelope controls the filter's cut-off frequency, then the envelope depth will have a major effect on the tone. This will give sonic results in two areas. First, each note will be articulated more clearly as the depth increases, and second, if the resonance control is set at a reasonable level, then the envelope depth will control the amount of "squelch" that the sound gives.

Envelope attack time. Assuming the envelope is controlling the filter's cut-off frequency and the envelope depth is set at a moderate level, then the envelope attack time can have a significant effect on the character of the note. A few milliseconds here or that at the start of the envelope can make a significant change to the attack of the note.

Envelope decay time. This control is quite subtle and only has any real effect when the envelope depth and the resonance is at a reasonable level, when its effect is more of a gentle tonal shift controlling how the squelch decays. As the sustain level is cut, the envelope decay time will have more significance (since the change between the peak and the sustain level will be greater).

Simply jumping in and twisting the knobs may not be the best way to design a sound, but for a basic bass sound, it's not necessarily a bad approach if you're trying to increase your understanding of how your synthesizer sounds and how the elements interact.

Once you have a basic sound, then you can work to refine it. As well as tweaking the filter and the envelope parameters, you might also want to add some portamento—set the synth to monophonic mode (so that it only plays one note at a time) and play with the portamento so that the pitch of each note slides from the previous.

Fat Bass Sounds

Once you have a single oscillator/single filter bass sound, you might want to think about something with more complexity and greater weight.

Before we start fattening things up, a word of caution: the single oscillator/filter sound works because of its limitations. It is not excessively fat or deep, so it will sit in a mix, and is nearly perfect for busy bass lines. As you fatten things up, the bass will increasingly dominate the mix and lead to a muddier sound. Accordingly, if you want to use a fat bass sound, ensure something else is removed from the low-end of the mix to make room for the sound.

Clearly the choice of wave is going to have an effect—it is much easier to thicken a sawtooth wave than a triangle wave—so make sure you have an appropriate wave before you start tweaking.

The simplest way to fatten up a bass sound is to add a second oscillator using the same waveform, and to detune that wave by a few cents (100 cents = 1 semitone). As a general rule of thumb, the more the waves are detuned (by reference to one another) the thicker the perceived sound. However, at a certain point the two waves will start to sound detuned, rather than fat.

You can add a third (and a fourth, if available) oscillator. If you are going to use three oscillators, then tune one up and one down, leaving one oscillator at the original pitch.

You can also thicken your sound by using a sub-oscillator which I will discuss shortly in the Sub-Oscillators section.

Once you have thickened your sound, then you should start to look at the envelopes.

Bass Envelopes

The bass sounds described so far have envisaged fast attack times (for the volume envelope) and swift release times. You will have noticed that I have only mentioned the volume envelope in passing—the only real effect on the volume envelope has been achieved through the filter cut-off envelope (and since that has been quick, the volume envelope has been swift).

However, for many sounds you will want a different envelope shape. For instance, if you're trying to create a dub-like bass sound or a sustained string bass line, a slower attack time and slower release time may be appropriate for the volume envelope. If you are going to apply a volume envelope, then (provided your synthesizer allows this option, and it almost always does) you should call up a separate envelope to control the volume.

Once you have two envelopes—one for the volume and one for the filter—think about (and listen to) how the envelopes interact with each other. To give you one example, if you slow the envelope controlling the filter cut-off frequency, so that its attack time is slightly slower than the volume envelope, you can get something of a "mwah" sound or a "spit" (depending on the relative envelope settings, the choice of waves, and the filter settings). These small tweaks can make significant changes to the character of the sound leading to a more interesting bass line.

Sub-Oscillators

A sub-oscillator is an additional oscillator added (usually) one, although sometimes two, octaves below the main oscillator (or oscillators). They perform two main functions:

First, they change the tone of the sound, usually giving it more thickness.

Second, they add more weight to the sound. This can be a subtle addition to the solidity of a note or it can be massive sub-sonic floor-shaking bass.

These two changes are usually mutually inclusive, so you will add weight and change the tone of the sound by adding a sub-oscillator.

We looked at the effect of mixing two oscillators together to create a new wave sound in the Layering Oscillators section. A sub-oscillator can create a new sound; however, that is not necessarily the aim of a sub-oscillator:

If your aim is simply to create a more solid bass sound without altering the tone to too great an extent, then try a sine wave as your sub-oscillator.

If your aim is to create a deeper and richer bass, and you can accept some change in the timbre, then try a square wave as the sub-oscillator.

Both waves will give a change in tone—the choice of wave affects how much the tone changes. The third option is to double the existing oscillator wave—depending on the sound you are aiming for, this may give good results.

The other factor affecting the tone change is the level of the sub-oscillator: at lower volumes the sub-oscillator will be less noticeable (and will therefore not add as much weight).

As a side issue, if you want to fatten your bass sound and give it more focus, think about using a high-pass filter.

If you run the sound first through a high-pass filter and then a low-pass filter, the high-pass filter can be used to boost the bass while the low-pass filter shapes the sound. To boost the bass, increase the amount of resonance on the high-pass filter and use key tracking to control its cut-off frequency. When you get the amount of key tracking right (and balanced with the cut-off frequency) the resonance boost will follow the pitch of the played note and each played note will have its fundamental frequency boosted.

One other point to remember when you're adding sub-oscillators: speakers. Listen through full range speakers, otherwise the sound you will end up with will be far too bass-heavy.

Bass Split Octaves

In many ways, a split octave sound is a sub-oscillator construct turned on its head. In other words, the main oscillator is the lower note and the secondary oscillator(s) is (are) set at a higher pitch when compared to the main oscillator. As you would expect, the main difference here is that where the sub-oscillator arrangement is designed to enhance the initial sound, with split octaves, the purpose of the addition oscillator(s) is (are) to change the tone, and generally to change the tone by making the sound brighter with a more complex wave construction (using the principles set out in the Layering Oscillators section).

Typically if you layer one oscillator one or two octaves above another, you will create a classic 1970s bright synth bass sound. You can tame this sound by controlling the level of the higher oscillator and with a low pass filter. The filter will have a disproportionate effect on the higher pitched oscillator which will often make its apparent excesses easier to calm.

Bass FX

Personally, I'm not a big fan of FX on bass sounds—they tend to make the sound mushy and the whole mix sound muddy. However—and slightly counter-intuitively—adding some subtle chorus will often brighten up a bass sound and give it more presence, while not portraying much of the chorusing effect. This brighter sound will, of course, be more dominant and take up more space in the mix.

Anyway, that's enough bass. Let's move on to brass.

Building Brass Sounds

Real brass instruments create a very brash and harsh sound. What people often perceive as brass is a much softer, more restrained tone. As well as being harsh, real brass instrument tend to sound much thinner than many people expect—often people associate brass with a fat sound because they are used to hearing brass as a section with several instruments covering a wide range of the tonal spectrum.

Basic brass sounds are moderately straightforward to create: you take a sawtooth wave and push it through a low-pass filter. From here you can then tweak the volume envelope to create a more realistic sound—in particular, you can recreate the impact when a note is first blown.

To create more "spit" at the start of the note, use an envelope to control the filter cut-off frequency and delay the attack time slightly (by reference to the attack time for the volume envelope).

As a matter of personal taste, I prefer not to add much in the way of FX to brass sounds. However, there are three tweaks you may want to think about:

First, to get a bit more harshness into the sound, think about adding some (subtle) distortion. This will help you achieve a slightly thinner and rougher sound.

I sometimes add some echo (a short delay with only one or two repeats). This tends to add more feeling and life to the sound rather than changing the tone.

You might also want to add a bit of EQ to thin the sound and help it sit in a mix.

Going back to the notion that much of what we hear as brass is actually a brass section—in other words, a group of individual instruments, each different, playing together. If you create a brass-like sound and play a few chords, you may not achieve satisfactory results since you won't get the effect of individual instruments playing together.

There are several approaches you can adopt to achieve greater realism (or rather, to reduce the unrealistic result that comes from having one brass patch):

First, you can create a number of individual patches and play each as a separate part. This replicates how real instrument sections are created.

Another alternative is to create parallel sounds within your patch. For instance, in Twin and Massive you can arrange two single-oscillator/single-filter combinations in parallel, each of which can create a separate sound that can then be placed towards the left and the right in the stereo panorama.

A third option is ignore analog synthesis and to employ a set of samples (this is especially effective if each part has a separate instrument). The benefit of a good set of samples is that many brass-specific playing techniques can be easily accessed giving a more realistic performance. You can read more about sample-based synthesis in How to Make a Noise: Sample-Based Synthesis.

Sticking with the metal theme, let's move from brass to look at bells.

Building Bell-Like Tones

If you want a broad palette of controllable bell and chime sounds then, in my opinion, there are better options than using analog synthesis techniques. For range, controllability, musicality, and tone, I think you will get better results using frequency modulation techniques (which are covered in How to Make a Noise: Frequency Modulation Synthesis).

However, you can still get interesting sounds with analog synthesis, so let's have a look.

The simplest way to create a chime-like tone is to select a triangle wave, set it to a high octave, and set a volume envelope to broadly replicate the envelope of a bell (in other words, a sharp attack, followed by a fairly swift decay time). You can double the octave and slightly detune the second wave to give a bit of "ring" to the sound. You can also detune the second oscillator by larger steps (five semitones can work quite well) to give a richer, more metallic sound.

To get a greater level of sophistication, you can send the waves through a low-pass filter with the resonance set high. If you push the resonance high enough (and your synthesizer allows this) then you will reach self-oscillation of the filter—you will hear this as a sine wave, which, provided you have the envelope right, can give more of a ring to the bell.

You can use quite a lot of FX without the processing becoming too noticeable. The most natural sounding processors are chorus, reverb, and echo.

Chorus will add some thickness and a lot of sparkle to the sound. You can turn the depth (of the LFO modulation) up quite high, but the speed (of the LFO modulation source) should be kept fairly slow.

A short delay time with a low amount of feedback will give some life to the sound and brighten up your bell-like tones.

A small amount of reverb will often help to smooth the sound, giving a more natural bell or chime-like tone.

Building Key-Based Sounds

Let's have a look at some keyboard-type sounds.

As with bell-like tones, let me repeat my warning: analog synthesis may not be the best way to create realistic key-based sounds. However, that alone is not a reason not to create these sounds. Indeed, there are many reason to create sounds that are reminiscent of key-based instruments. These reasons include ease of use, immediacy, the facility to tweak and add FX, and most significantly, the ability to take the character of the sound and mangle it until you have created something entirely new.

So the sounds I am going to produce here are more about character, and less about sonic fidelity.

Piano

The root to create a really quick-and-dirty piano-like sound is a square wave and a sawtooth wave. Load the two and balance the level of the oscillators—to my ear you want slightly more sawtooth than square. This will give you the brightness but without the hollowness of a square wave on its own.

Once you have a basic tone, set a volume envelope to mimic the envelope of an acoustic piano. Staying in the volume envelope, adjust the sustain level so that it is unnaturally high. This might seem odd, but it will compensate for the volume loss through the filter.

Turning to the filter, choose a fairly gentle slope (12 dB/octave or less). Set the resonance to zero, turn the cut-off down low, and set an envelope to modulate the cut-off frequency to the maximum extent.

When the filter is set up, you can adjust the envelope. This should be adjusted in a similar manner to the volume envelope, however, the sustain level should be much lower and the release time should be longer (to avoid unnatural changes during the short release stage before the volume is cut by the volume envelope.

This isn't a great sound—in fact, it's pretty rough—but it might fit in a mix. With a bit of tweaking you can create far more of an electric piano-type sound which may be preferable in a mix.

Clavinet/Harpsichord

You can create a fairly good clavinet or harpsichord-type sound using analog techniques. And when I say "fairly good", I'm talking in terms of sound, not in terms of fidelity of reproduction.

The root of the sound is key-synchronized oscillators with one oscillator pushed out of phase. The further the oscillator is moved out of phase the more of a plucked sound you can achieve—this plucked sound is the root of the clavinet/harpsichord tone.

Once you have the plucked sound, you can then shape it (with a volume envelope) to create the shape of the sound (fast attack, fast decay, low sustain level). If you want to add a bit more shaping, then you can add a filter with an envelope. If you do add the filter, then you probably want to raise the sustain level on the volume envelope to compensate for the level attenuation by the filter.

To add a bit more weight to the sound, you can add another oscillator (with the same waveform at the same pitch). Keep the level of this oscillator low enough that this oscillator doesn't dominate, but high enough that it adds weight and thickness to the tone.

Once you have all the elements in place, tweak to balance the sound.

If you're after a more radical tone change, then try a different waveform—both the sawtooth and square waves give good results. Alternatively, try some oscillator combinations where the waves are at different oscillator settings: two sawtooth waves with the second set two octaves above the first can work well.

Organ

Of course, you could create a realistic organ sound by loading up a sample set. However, if you're working with analog synthesis there is a really quick and natural way to create organ tones. Not only that, but the sounds are far more tweakable and controllable in real-time than they are with sample sets.

The setup is simple:

Switch off the filter.

Load a sine wave into the oscillators (you'll want at least three oscillators here). If you don't have a sine wave, then a triangle wave works perfectly well, in fact, a triangle wave can give a richer sound.

Set the volume envelope to have a very fast attack time, the sustain level set to the maximum, and the release time set to zero.

Tune the three oscillators at octave pitches (so oscillator two is one octave above oscillator one and oscillator three is one octave above oscillator two).

Set the level of oscillator one to mid-way and oscillators two and three to zero.

To create the tone, slowly bring up the level of oscillators two and three. There is no "right" sound here—keep balancing the level of the three oscillators and listen to the tonal variations.

Once you've got an idea about the basic tonal variations, then start playing around:

You don't have to stick with these pitch intervals. You can have any interval you want, but as a start, drop the pitch of oscillator one by an octave (and reduce its level so it doesn't dominate too much) and drop the pitch of oscillator three so that it is a major fifth (seven semitones) above oscillator two. After this, keep playing with pitches and levels.

Slightly detune the oscillators to get that Leslie cabinet vibrato.

Experiment with LFOs to control the level of each oscillator and the pitch of each oscillator. When employing these LFOs, you only want the slightest wobble (you can use a fast LFO, but with a small depth).

Add some distortion—ideally a gentle amount of vintage distortion to give the sound some more warmth and body.

Building Lead Sounds

Lead sounds are intended to feature prominently in a mix without dominating.

The key factor for any lead sound is that it should fit—it should be appropriate within the context of the track where it is played. It can be simple, it can be complex—it just has to fit.

I'm not going to try to describe how to create a lead sound: there are too many options and too many permutations, and very often a plain sawtooth or square wave with a bit of filtering is all that is needed. Instead, let me add a few thoughts about how a sound can fit within the context of a track.

One of the key issues to think about with a lead sound is the extent to which it dominates. This is a fine balance: the sound should cut through, but it shouldn't be so loud that it masks the backing. One way to ensure the sound can be heard, but to stop it dominating is to run the sound through a combination of filters arranged in series.

If you run the sound first through a high-pass filter and then a low-pass filter, then the high-pass filter can cut out the low-end information and the low-pass filter can then shape the sound. As long as you use the high-pass filter judiciously you will notice little effect on the sound when the end result is played within the context of a track.

One quick way to make your lead sound really stand out is to add some distortion. At lower levels this will often bring some presence to the sound without changing the character too radically. Of course, you can add lots of distortion if you really want your sound to stand out, and if you want a balance, then why not set MIDI velocity as a modulation source and the distortion amount as the modulation destination—this will may your sound very playable.

Another area to look at when you are creating lead sounds are the user controls, in particular, the bend wheel, the modulation wheel, and MIDI velocity. The combination of the three will give you control over a wide range of tonal nuances as you play.

One area where you can go wild is in adding FX units. I'll leave these to your imagination.

Building Pads

I made this comment in connection with the Building Lead Sounds section and will make this comment again in the Building Plucks and Stabs section: pad sounds can fill up huge amounts of the sonic range. This can lead to them dominating a mix making it muddy and indistinct, as well as overpowering the other elements of the mix.

In order to avoid this unwanted dominance, the art to creating a great pad is to find a pleasing tone while keeping the sound as thin (or if you prefer, as "un-fat") as possible. As I've mentioned before, one of the best ways to thin a sound is to run a high-pass filter and a low-pass filter in series, and use the high-pass filter to cut out as much of the low-end information as you can (and remember, you can always cut more than you expect when you listen to a sound within the context of a mix). If you don't want to use a filter, then try EQ to cut the low end.

Simple Pads

Many simple and highly effective pad sounds can be created with a single filtered wave. While effective, this approach has a limited tone palette, and may leave you with a sound with little movement.

Turning to the tone palette first, the most straightforward way to broaden the range of tones that is available to you is to use a second oscillator. As discussed in the Layering Oscillators section, this combining approach can create a whole new waveform, giving you a near infinite tone palette to work with.

You will often achieve excellent results choosing a different waveform and/or pitching the second oscillator one or two octaves above the first. Other non-octave intervals may work (for instance plus seven semitones—a major fifth—is often good), but you may need to take extra care to keep the level of the second oscillator at a place where the sound doesn't become too strained.

Also, pay attention to whether you are hearing one coherent sound or two separate tones—neither option is wrong (or right), nor better, but you should be clear about the end result you are trying to achieve. Another issue to remember is that a low-pass filter will have a disproportionate effect on the higher pitched oscillator. This may be useful as it will make it easier to control the higher pitched oscillator without affecting the main oscillator.

On the subject of the filter, often when creating a pad sound you will find that one filter setting does not play well across the whole keyboard. In this case, a bit a key tracking will often help—set the filter cut-off to open more at higher pitches and you will create a more natural sound that has a lighter quality.

When you want to add movement to a pad sound, you have two options:

The first choice is to add an LFO to subtly modulate the pitch of one of the oscillators.

The second choice is to bring is some FX—delayed based modulation effects (such as a chorus, phaser, or flanger will usually work well).

There is, of course, a third option—bring in another oscillator and detune it slightly. And lastly, I'm sure you've already figured that you can add an LFO, some FX, and third oscillator all at once.

Adding Some Control

To move beyond a plain pad, you can introduce some player control. Depending on the context, you have many options:

The modulation wheel is a useful quick-and-dirty control. Often it can be assigned to control the filter cut-off frequency allowing you to tweak the tone in real time.

If you want to control the filter cut-off frequency there are other options. First, MIDI velocity can be set as the modulation source. The downside to this option is that you have no ongoing control—the filter setting is based on the amount of velocity when the note was struck. If you want ongoing control (and your MIDI hardware allows it) then you can use aftertouch to control the cut-off frequency as the sound sustains.

Perhaps a better way to use MIDI velocity is to control the attack time of any envelope. Through setting MIDI velocity as the modulation source and the attack time of an envelope as the destination you can choose whether notes fade in slowly or start immediately by changing how each note is struck.

String Sounds

As with Building Brass Sounds, when creating string sounds, there is an advantage to creating a number of individual string-type sounds and playing separate parts with different timbres. The end result will then sound less synthetic.

Synthetic string tones can be created quite quickly: take a sawtooth wave and pass it through a low-pass filter. The hard work is in refining the tone, and here you have the usual tools:

The volume envelope to create the changes in the level. One area that needs particular attention is the attack phase—unlike many sounds, string sounds benefit from a slower attack time in order to mimic the swell of a real string section (and as noted above, you can control the attack time on a note-by-note basis with MIDI velocity).

The filter envelope can change the tone of the sound. This can work well to emulate the delicate tonal changes of a stringed instrument; however, you should limit the use of filter envelopes if you are using high levels of resonance (since strings don't usually "squelch").

If you are trying to emulate a stringed instrument of any variety, it is almost mandatory (unless you are trying to emulate a solo instrument, in which case, I recommend you use samples or a real instrument) to double oscillators and add a bit of detuning. This gives the effect of several instruments playing at once and both thickens and brightens the tone.

And of course, with any string-based pad, you should think about adding time based modulation delay FX units.

Rhythmic Pads

Pad sounds don't have to simply create sound... they can add rhythm too.

Once you have your pad sound sorted, if you want to add a rhythmic element, turn to the LFO section. You can use any wave, but the most obvious sonic results can be achieved with square and pulse waves.

With the LFO waveform selected, set the LFO frequency at a moderate rate, and set the LFO to modulate the volume of the patch. This will create an effect where the sound is heard during the positive phase of the square wave, but the sound is attenuated during the negative phase giving you a rhythmic pad.

Once you have the effect working, then adjust the tempo of the LFO frequency to match the beat of your track (most synths allow you to synchronize the beat without needing to precisely adjust the LFO).

To take this effect further, set the filter cut-off frequency as the modulation destination. This may give a similar rhythmic result, but there will be tonal changes rather than just volume changes. You should also experiment with different waveforms—the rhythmic effect may be less pronounced, but the tonal changes may be more interesting.

Many synthesizers (such as Twin and Massive) also have specialized modulation sources which create beat-synchronized stepped level changes specifically designed to create rhythmic effects (check back to Specialized Modulation Sources for details of these tools). If you want to create a rhythmic pad with any level of complexity, you probably want to dig into these tools.

When using these tools it's very easy to focus on controlling the volume and filter cut-off to create rhythm, but also pay attention to the gate time of each step (in other words, the length of time for which each step sounds—a step does not need to sustain for the full length of each step). In Massive, when using the performer it is possible to control the gate time on a step-by-step basis; in other synths you may need to put in a bit more work.

And just because the pad creates rhythmic sounds, don't forget the real-time control. For instance, you can control the filter cut-off (or the depth of the modulation of the filter cut-off) with the modulation wheel or with aftertouch (to give two examples) in order to keep some control over the sound.

Building Plucks and Stabs

In many ways, plucks and stabs bring together all of the sound building principles we have looked at so far.

These sounds are, by their nature, short and usually prominent, and ironically, their sound matters less. However, to be effective, their tone needs to be molded in order to get the right plucked or stabbed sound. Equally, these sounds are far more sensitive when it comes to balancing several competing factors (for instance, when it comes to balancing the decay time and the sustain level of an envelope).

When it comes to balancing the competing factors, there are several issues to look at:

The sound can be fat or thin, or anything in between. However, as a general rule of thumb, the fatter the sound, the shorter the pluck/stab should be. If you don't get the balance right, then you may find the sound dominating the track.

Generally envelopes need to be fairly sharp. Usually the critical phase in the envelope is the decay time (since the attack time will be zero and the sustain level will be set to zero, or at least very low).

The filter envelope does not need to duplicate the volume envelope. Indeed, there is an advantage to using a separate envelope. By delaying the filter envelope's attack time slightly, you can emphasize the start of the note, reinforcing the pluck sound. The pluck sound will be further emphasized if the decay times vary.

To get a sufficiently sharp sound (if that is what you are after) think about incorporating a subtle amount of distortion.

These sorts of sounds work really well with some velocity sensitivity—particularly to control the cut-off frequency.

In the higher keyboard regions stabs and plucks can sound too bright. To address this issue, apply some negative key scaling to reduce the brightness of the patch in the top octaves.

Building Synthetic Drum Sounds

Let's round off this chapter by looking at synthetic drum sounds.

Most synthetic drum sounds have two key elements:

a sound, which is often a processed sine wave (and if you don't have a sine wave, a triangle wave will work), and

some noise, which is usually filtered.

Rather than trying to explain the theory, it's probably easiest if we leap straight in and start making some drum sounds. The intention behind all of the sounds in this section is that they are quite raw—once you have got your head around the concepts I am demonstrating, I suggest you take some time to finesse these sounds and perfect them for use in your own productions. But before we get going, a couple of points:

First, although this section covers drums and percussion, there are several techniques that are applicable to more conventional sounds (for instance, the pitch modulation technique I will show you to get the kick into a kick drum can be widely applied).

Second, the key word in this section's title is "synthetic". The sounds I am going to create here are intended to sound like electronic drums. If you want realistic acoustic sounding drums, then either hire a real drummer or go and buy yourself a set of high quality samples.

However, if you are going to use samples, then you might want to come back and check out some of the techniques here—you will often find that layering samples and synthetic drums can keep the acoustic sound but give the necessary depth and power which can be missing when using samples alone.

All of the techniques set out in this section can be applied to conventional synthesizers. There are also specialized drum tools which can help you create synthetic drum sounds, but unless you understand what you're doing, then these specialized tools can give unpredictable results. Once you've been through this section and have got a grip on synthetic percussion, you might want to take a look at these specialist tools.

One last word: all of the sounds created in this section can be played across the keyboard. When taken out of an appropriate range, some sounds will not sound natural (for instance, kick drums should be played over a lower range). If any sound sounds particularly unnatural (and remember, we're dealing with synthesized sounds) then try it in another key range.

That's enough prevarication... let's create some sounds.

Kick Drum

The essence of a kick drum is a low-pitched sine wave (or triangle wave) so make sure you're listening through a full-range speaker as you program this sound.

To start, select the sine wave and tune it to its lowest setting. To create the sound the sine wave should then be controlled by two envelopes:

A volume envelope gives the characteristic staccato drum sound.

An envelope assigned to modulate the pitch of the envelope gives the sound of the "beater" "hitting" the "drum".

With your sine wave loaded and dropped in pitch, go to the volume envelope and set it so:

The attack time is set to zero.

The decay time is fast, but not zero.

The sustain level is zero. If you have an attack-decay envelope available that will achieve the same result in less time.

The crucial control here is the decay time. If this is too fast, then all you will hear is a click. If it is too slow, then you will hear the low pitched sine wave. Play with this control and listen to the shades of tone it allows you to access. Also notice that you are changing the tone/character of the note without a filter and without modulating the source wave.

Once you have a basic tone that you are happy with, you can add a bit more kick to the sound by setting another envelope to modulate the pitch of the sine wave. Set the envelope with similar settings to your volume envelope:

Attack time set to zero.

Fast decay time, but not zero, and shorter than the decay time in the volume envelope.

Sustain level set to zero.

The purpose of this modulation is to add some punch to the impact, not to hear a changed pitch in the drum. Once you have set the pitch envelope, start to raise the level of modulation depth. If you raise the level too far, you will hear the pitch start to rise—for a kick drum, you're looking for the impact at the start of the kick and a slight tightening of the overall sound.

Once you've got here, you might want to try increasing (but only slightly) the decay time in the volume envelope and balancing the two decay times and the pitch modulation depth.

If you want to add some more control to the sound, then set the MIDI velocity to modulate the pitch depth. With this set, MIDI velocity will affect the tone of the initial beat.

Closed Hats

Synthetic hi-hats are quite straightforward: they are essentially a bit of filtered noise. Let's start by looking at a closed hi-hat.

Virtually every synthesizer will offer a noise source—choose the brightest you can find, ideally choose white noise.

Once you have your noise, a basic closed hi-hat can be created with a volume envelope with:

The attack time set to zero.

The decay time set to a fairly short time (about the decay time of a closed hi-hat).

The sustain level set to zero.

This will give you a workable closed hi-hat. If you then want to play with its character, call up a band pass filter (ideally with a 12dB/octave slope). Set the resonance control to about mid-way and then tune the sound with the cut-off frequency.

To control the tone further, the cut-off frequency in the filter can be modulated (very gently) by an LFO. This can give a subtle variation in tone each time a note is triggered.

If you want more control over the tonal variation, you could again modulate the cut-off frequency, but use MIDI velocity (or pitch) as the modulation source. In this way, you could program variation in your MIDI track. You could, of course, use both of these techniques.

Alternative Noise Sources

If you're not happy with the noise sources your synthesizer offers, or if you just want more tonal variation (which, in the context of hi-hats, will give you a range of different character hats) then you can create your own noise source.

To create your noise source:

Choose a waveform—any wave will do, but I find the square wave gives much brighter, forceful results.

Tune the wave up—maybe raise it about four or five octaves, but depending on the particular synth, you may not want to go as high as you can if you start getting a screamingly high note rather than the effect intended.

Set an LFO to modulate the pitch of the wave.

Select a random or sample-and-hold wave for the LFO. If these waveforms aren't available, then load two LFOs and choose a square or a pulse wave in each LFO.

Set the pitch modulation depth to the highest level you can and the frequency of the LFO to the maximum. If you are using two LFOs, then tune the frequencies to slightly different settings—the combination of the two will give an effect close to a random wave.

And there you (should) have a noise source. You can change the tone quickly by selecting different waves in the oscillator (or by messing with the LFO depth and frequency).

Once you have created your own noise source, use it in your hi-hat sounds.

Open Hats

Having set up the closed hi-hat, the next logical step is the open hi-hat.

There is a simple change to morph the closed hat to an open hat—extend the decay time.

Depending on the noise source you are using, a longer decay time can allow an open hat to dominate the sound to a greater extent than the close hat does. To constrain the open hat, try a steeper filter (for instance, try a 24dB/octave filter rather than a 12dB/octave filter and reduce the resonance. While you're tweaking the filter, you may also want to push the cut-off frequency to adjust the tone.

Although we're looking at synthetic percussion here, listen to your open and closed hi-hats in context. There's no rule that they have to have any sonic relationship in the way that the hats on an acoustic drum kit would (since they would be the same hats...), but listeners will usually expect there to be a link between the two sounds.

Snare Drum

There are two elements to a snare drum sound:

The drum sound.

The snare sound.

It is something of a challenge to create this effect with a synth, but we can get close.

The basic drum sound is created in a similar manner to how we created a kick drum: a sine wave with its pitch modulated by a fast envelope, and with its volume contour mirroring the shape of a real drum. However, unlike with a kick drum, the pitching of the snare can be higher. Set it at a level that feels right to you—with higher pitches you get a thinner drum and at lower pitches you get a flabbier sound.

The snare sound is created with a noise source—white noise works well, or you can add your own noise source.

The snare element should be controlled by a volume envelope, but not by the same volume envelope as the drum sound uses since the snare should have a longer decay (to mimic the ring of the snares).

Once the two elements are in place, their levels should be balanced. I find that the drum element should be set at a lower level than the snare element. Once you set these levels, you may find that you want to tweak the envelopes—in particular the pitch envelope modulating the sine wave—in order to make the snare element more forceful.

Another subtle bit of shaping you can add is to run the noise through a low-pass filter. Choose one with a gentle slope (12dB/octave is a good choice) and boost the resonance. You can then control the snare tone to help it blend with the drum element.

Claps

A synthesized clap can be created with noise—whether white noise or synthetically created noise—as its foundation. The level of the noise can then be controlled with an envelope with similar settings to those we have already used. To give it more focus, the sound can then be passed through a band-pass filter (12dB/octave).

This set up gives a basic sound. It also gives us only one clap. Before we resort to FX units (which really do help with this type of sound), there are two further steps you can take:

First, a second, parallel filter can be introduced. The settings should be similar but with a different cut-off frequency to bring a different tonal quality to the sound.

The second step is to add more oscillators (or rather more noise); ideally each noise source should have a different character to increase the illusion of a group of claps. Each oscillator should have its own volume envelope with similar, but unique settings. Perhaps the biggest difference for each envelope should be the addition of a small amount of delay (before the attack stage). The slight delay will give the impressive of the clap coming at a slightly different time (as would be the case if there were several people clapping).

Cowbell

To create a synthetic cowbell, take two square waves, add a volume envelope (you've got the idea where to start), and tune one wave five semitones above the other.

This gives you the basic sound. To tweak, run the combination through a low-pass filter (12dB/octave) with the resonance pushed quite high. Adjust the cut-off frequency to taste. If you then want to work on the tone a bit more, play with the pitch of the second oscillator.

Syn Drum

If you don't know what a syn drum sounds like, then you weren't around in the 1970s. In this case you probably also missed out on disco, which may be no bad thing.

The basis of this sound is a sine wave which then has its pitch modulated. This pitch modulation is similar to the technique we used to get some punch at the start of a kick drum, except here, the decay time is lengthened so that the pitch fall effect can be clearly heard. The timing here is quite significant: it needs to be longer than simply to give a punch, but not too long (this is drum sound). Take some time to listen to the sound as you tweak the decay time.

Once you have got the pitch envelope right, set up a volume envelope with similar parameters and set the decay time so that the note ends (at the zero sustain level) at the end of the pitch fall. This will give a drum envelope instead of allowing the note to sustain.

Toms

To create an electronic version of toms, take the syn drum sound, and tweak:

First, shorten the decay time in the pitch envelope so that the characteristic syn drum sound is replaced with the classic punch.

Second, shorten the decay time in the volume envelope so that there is no sound heard after the hit.

As you will have realized, this sound is very similar to the kick drum. However, there should be differences:

Toms will be played at a higher range than the kick.

As well as being played at a higher range, you are likely to want to play over a range (perhaps a range of nearly an octave) to reflect the different sizes of toms.

The sound of toms should be rounder than the kick.

Tune to taste and you'll be there.

Drum FX

All of the drum sounds have been created without effects, however, you might want to think about adding some FX both to enhance and sweeten the sounds. In particular, you might want to think about the following:

Some (subtle) overdrive/distortion could add some warmth to the sound and give a slightly vintage vibe.

Compression may help to smooth out the individual drum sounds as well as giving them more presence.

EQ may help to focus the drum sounds.

In particular for the clap sounds, echo may help to give the illusion of a group.

And finally, although they're not FX units, the more specialized step modulation generators can create a great rudimentary drum sequencer.

Building Your Own Sounds

That should be enough sounds to get you started, so go and grab your synthesizer of choice and start making your own noise.

If you're wondering where to find out more about making sounds with synthesizers, then check out the next section, Further Resources, which points you to a range of further materials (including free downloadable books and free sound design videos).

Further Resources

I hope you have found this book interesting and useful. If you want to know more about sound design and making music with synthesizers, then I suggest you head over to my website (noisesculpture.com) and check out my music-related books and videos. If you're browsing the site, I suggest you look at the following books and videos first.

How to Make a Noise Series of Books

There are five books in the How to Make a Noise series (this one, plus the four others listed here). Each book looks at a different aspect of synthesis, focusing on the particular strengths of that technique.

While each book is separate, there is (as you would expect) some overlap (for instance, a filter will always act like a filter, whatever the synthesizer—but the way that the filter is used will vary).

How to Make a Noise: Frequency Modulation Synthesis

(published by Coombe Hill Publishing, ISBN: 978-0955495502, available in electronic format only, noisesculpture.com/fm)

Frequency modulation synthesis came to prominence in the 1980s with the DX7 synthesizer which dominated the sounds in the charts for years.

FM synthesis adopts a completely different approach to creating sounds when compared with either analog or sample-based synthesis. Instead of starting with a bright sound and making it dull and/or shaping it with a filter, FM synthesis starts with a simple wave—usually a sine wave—and distorts it (usually with another sine wave). The interaction between the two (or more) waves creates and shapes the sound over time.

How to Make a Noise: Frequency Modulation Synthesis shows you how to create a broad, detailed, highly nuanced range of controllable tones. These principles can then be applied to any synthesizer with frequency modulation capabilities.

How to Make a Noise: Sample-Based Synthesis

(published by Coombe Hill Publishing, ISBN: 978-0955495502, available in electronic format only, noisesculpture.com/sample)

Sample-based synthesizers can be used to create the sound of a full orchestra or individual instruments. They can replicate the sound of a full drum kit, delicate vintage gear, and specialist instruments. Samplers allow one musician to play the bagpipes, a harp, a lute, and a whole orchestra, all at the push of a button.

As well as offering the facility to recreate the sound of real instruments with a high degree of sonic fidelity, sample-based synthesis also allows real sounds to be used as the basis for new sounds. These sounds can then be manipulated and mangled to create unimagined sounds with genuine complexity and depth.

How to Make a Noise: Sample-Based Synthesis looks at how to create and control sounds—both realistic recreations of real instrument and creative warped tones—and the tools and techniques that are available to help shape your sounds.

How to Make a Noise: iPad Synthesizers Edition

(published by Coombe Hill Publishing, ISBN: 978-0955495502, available in electronic format only, noisesculpture.com/ipad)

The fourth book in the series, How to Make a Noise: iPad Synthesizers Edition doesn't focus on a particular technique, but instead, looks at creating sounds on five synthesizers that are available for the iPad (BassLine, iMS20, iELECTRIBE, ReBirth, and NLogSynth Pro).

Many of the techniques discussed in the three previous books are covered in iPad Synthesizers Edition, but with particular focus to their application to the five featured iPad synthesizers.

How to Make a Noise: A Comprehensive Guide to Synthesizer Programming

(published by Coombe Hill Publishing, ISBN: 978-0955495502, noisesculpture.com/comprehensive)

There is another book in the How to Make a Noise series—the original hard copy (paper) book. The four electronic-only books are parallel editions to this existing book.

How to Make a Noise: A Comprehensive Guide to Synthesizer Programming was first published in 2005 and revised in 2007 and so is now past its first flush of youth. However, it is perhaps the most widely read book about synthesizer programming (and there's a free downloadable PDF version, too).

This book focuses on building sounds (over 300 sounds are built) and looks at building these sounds with six synthesizers (Cameleon 5000, Rhino 2, Surge, Vanguard, Wusikstation, and Z3TA+) which were on the cutting edge when the book was first published, but (in most cases) have now been superseded.

However, the techniques outlined are still applicable and the book also sets out other synthesis options, such as additive synthesis and wave-sequencing (which are not covered in any of the other publications).

Jumpstart Your Music Career

(published by Course Technology PTR, ISBN: 978-1435459526, noisesculpture.com/jump)

You will have noticed that this is not a book about synthesis. However, once you can make sounds, then might to want to make some money, and this is where Jumpstart Your Music Career comes in. This book is for every musician who wants to earn a living—and to sustain a long-term career—making music.

It has never been so easy to have a career in music: the barriers to entry have never been lower, the access to and ways to find fans have never been more straightforward, and the options to generate income have never been more plentiful.

The hard part is developing a strategy that's right for you and then executing it. Jumpstart Your Music Career looks at how you can build and sustain a successful career. It is the book that should be on every musician's bookshelf.

Cakewalk Synthesizers: From Presets to Power User, second edition

(published by Course Technology PTR, ISBN: 978-1435455641, noisesculpture.com/cakewalk-synthesizers)

Cakewalk Synthesizers: From Presets to Power User, second edition is the ultimate guide to understanding and using all of Cakewalk's synthesizers and learning about synthesizer programming.

The book features the complete range of Cakewalk (and rgc:audio) synthesizers, including Rapture, Rapture LE, Dimension Pro, Dimension LE, Beatscape, Session Drummer 2 and Session Drummer 3, Z3TA+, and Pentagon I.

There is also a package of downloadable material to accompany this book. This package includes additional chapters and is available for free to everyone (you don't need to have bought the book). This package also includes sound design master classes from several leading synthesizer programmers and so is well worth checking out.

Becoming a Synthesizer Wizard: From Presets to Power User

(published by Course Technology PTR, ISBN: 978-1598635508, noisesculpture.com/becoming-a-synthesizer-wizard)

Becoming a Synthesizer Wizard: From Presets to Power User is for:

People who are new to synthesis and who want to understand the basics and take their first steps in creating and controlling sounds with synthesizers.

Musicians who are familiar with synthesis, but who want to solidify or expand their knowledge so they can use their tools more fully.

Anyone who wants to build, share, and perhaps sell their own synthesizers.

The book looks at synthesis from first principles. It focuses on the elements of synthesis and the building blocks, and it highlights its examples using four leading modular synthesizers (Zebra 2, KarmaFX Synth Modular, VAZ Modular, and SynthEdit). From these building blocks, the book looks at a range of sound building techniques that can be used with the featured synthesizers, or can be readily applied to any other synthesizer.

The book is also a great introduction to SynthEdit for anyone who wants to create—and then share or sell—their own synthesizers.

Synthesizer Boot Camp

(noisesculpture.com/synthesizer-boot-camp)

Synthesizer Boot Camp is a series of videos which looks at specific synthesizer techniques that can be applied in creating sounds. All of these videos are freely available on my website and can also be accessed on the Noise Sculpture YouTube channel.