Apple Logic Pro 7 Plug In Reference Ref

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Logic Pro 7
Plug-In Reference

 Apple Computer, Inc.
© 2004 Apple Computer, Inc. All rights reserved.
Under the copyright laws, this manual may not be
copied, in whole or in part, without the written consent
of Apple. Your rights to the software are governed by
the accompanying software licence agreement.
The Apple logo is a trademark of Apple Computer, Inc.,
registered in the U.S. and other countries. Use of the
“keyboard” Apple logo (Option-Shift-K) for commercial
purposes without the prior written consent of Apple
may constitute trademark infringement and unfair
competition in violation of federal and state laws.
Every effort has been made to ensure that the
information in this manual is accurate. Apple Computer,
Inc. is not responsible for printing or clerical errors.
Apple Computer, Inc.
1 Infinite Loop
Cupertino, CA 95014-2084
408-996-1010
www.apple.com
Apple, the Apple logo, Aqua, Final Cut, Final Cut Pro,
FireWire, iBook, iMac, iPod, iTunes, Logic, Mac,
Macintosh, Mac OS, PowerBook, Power Mac, Power
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Other company and product names mentioned herein
are trademarks of their respective companies. Mention
of third-party products is for informational purposes
only and constitutes neither an endorsement nor a
recommendation. Apple assumes no responsibility with
regard to the performance or use of these products.

Contents

Preface

9
10

Introducing Logic’s Plug-ins
About This Manual

Chapter 1

13
13
16
19
20
21

Basics
Using Plug-ins
The Plug-in Window
Plug-in Settings
Plug-in Automation
Plug-ins From Other Manufacturers

Chapter 2

23
23
24

Instruments and Effects
Effect Plug-ins
Instrument Plug-ins

Chapter 3

29
29
32
33
37
38
38
38

Equalizer
Channel EQ
Linear Phase EQ
Match EQ
Fat EQ
Silver EQ
DJ EQ
Individual EQs

Chapter 4

39
39
42
42
43
45
45
47
49
50
52

Dynamic
Compressor
Silver Compressor
Expander
Noise Gate
Silver Gate
Enveloper
DeEsser
Limiter
Adaptive Limiter
Multipressor

Chapter 5

57
57
62
62
63
64
65
66

Distortion
Guitar Amp Pro
Distortion
Overdrive
Bitcrusher
Clip Distortion
Phase Distortion
Distortion II

Chapter 6

67
67
70
72
79
92
92

Filter
AutoFilter
Fuzz-Wah
EVOC 20 Filterbank
EVOC 20 TO
High Cut/Low Cut
High Pass/Low Pass Filter

Chapter 7

93
93
94
96

Delay
Sample Delay
Tape Delay
Stereo Delay

Chapter 8

97
97
98
99
99
101
104
105
106
106
107

Modulation
Modulation Delay
Chorus
Flanger
Phaser
RingShifter—Ring Modulator/Frequency Shifter
Tremolo
Ensemble
Rotor Cabinet
Scanner Vibrato
Spreader

Chapter 9

109
109
110
111
114
115

Reverb
AVerb
SilverVerb
GoldVerb
PlatinumVerb
EnVerb

Chapter 10

117
118
119

Convolution Reverb: Space Designer
Using Space Designer
Space Designer’s Parameters

Contents

137
138

Creating Impulse Responses
About Convolution

Chapter 11

143
143
144
145
147
150
152
153
154

Special
Spectral Gate
Pitch Shifter II
Vocal Transformer
Pitch Correction
SubBass
Denoiser
Exciter
Stereo Spread

Chapter 12

157
157
158
159
160
161
163
166
166

Helper
Test Oscillator
Tuner
Gain
I/O
Direction Mixer
Multimeter
Correlation Meter
Levelmeter

Chapter 13

167
167
168
168
169
169

Vocoder—Basics
What Is a Vocoder?
How Does a Vocoder Work?
How Does a Filter Bank Work?
Analyzing Speech Signals
Tips for Better Speech Intelligibility

Chapter 14

175
175
176
190

The EVOC 20 PS
Using the EVOC 20 PS
EVOC 20 PS Parameters
Block Diagram

Chapter 15

191

Vocoder History

Chapter 16

195
195
196
196

Synthesizer Basics
Analog and Subtractive
What Is Synthesis?
Subtractive Synthesis

Chapter 17

201
201

EFM 1
Concept and Function

Contents

202
202
204
205

Global Parameters
FM Parameters
Modulator and Carrier
The Output Section

Chapter 18

207
207

ES M
Parameters of the ES M

Chapter 19

209
209

ES P
Parameters of the ES P

Chapter 20

213
213

ES E
Parameters of the ES E

Chapter 21

215
215

ES1
Parameters of the ES1

Chapter 22

223
223
225
283

ES2
Concept and Function
The ES2 Parameters
Tutorials

Chapter 23

301
301
302
309
325
332
340
341

Ultrabeat
The Structure of Ultrabeat
Overview of Ultrabeat
The Synthesizer Parameters
Modulation
The Step Sequencer
Importing Sounds
Tutorial: Creating Drum Sounds in Ultrabeat

Chapter 24

355
356
359
360
362
373
379
384
392
400
408
409
426

Sculpture
The Synthesis Core of Sculpture
Sculpture’s Parameters
Global Parameters
String and Object Parameters
Processing
Post Processing
Modulation Generators
The Control Envelopes
Morph
MIDI Controller Assignments
Programming: Quick Start Guide
Programming: In Depth

Contents

Chapter 25

453

KlopfGeist

Chapter 26

455
455
456
460
477
481
482

EVB3
Concepts and Function
MIDI Setup
The EVB3 Parameters
MIDI Controller Assignments
Additive Synthesis With Drawbars
A Short Hammond Organ Story

Chapter 27

485
485
486
502
503

EVD6
The EVD6—Concept and Functions
Parameters of the EVD6
Controlling the EVD6 via MIDI
A Brief History of the Clavinet

Chapter 28

505
505
506
513
516

EVP88
The EVP88—Concept and Functions
Parameters of the EVP88
The E-Piano Models Emulated
EVP88 and MIDI

Chapter 29

519
520
526
531
568
570
571

EXS24 mkII
Using Instruments
File Organization
Sample File Import
EXS24 Key Commands
A Brief History of Sampling
MIDI Controller List

Chapter 30

573
573

GarageBand Instruments
About GarageBand Instruments

Chapter 31

575

External Instrument

Glossary

577

Index

599

Contents

Preface

Introducing Logic’s Plug-ins

The professional Logic music and audio production
software features a comprehensive collection of powerful
plug-ins.
These include; innovative synthesizers, high quality effect plug-ins and authentic
recreations of vintage instruments. Logic also supports the use of Audio Unit plug-ins
in Mac OS X and also supports TDM plug-ins for users of TDM systems.
Given a fast enough computer, you could conceivably arrange and mix an entire song
using several software instruments, such as Logic’s ES1, ES2, EVP88, or EXS24, amongst
others. These instruments have the added benefits of superior sound quality and
timing as the audio signal never leaves the digital domain, and you can freely edit
these software instrument parts, change the tempo and more, right up to the final mix.
Don’t worry if you’re unfamiliar with the terminology used here—this manual will
explain everything. It covers all of the general things you need to know about plug-ins
and will introduce you to the individual effects and instruments and their parameters.
We’ve included a few tutorial chapters, which will explain how to program sounds
using several of Logic’s instrument plug-ins.
Using plug-ins is much easier if you are familiar with some of Logic’s basic functions.
You should be acquainted with Logic’s Audio Mixer before going further. Information
about it can be found in the Audio Mixer section of the Logic reference.
The Bounce buttons found on the Master Audio Objects allow you to write submixes of
plug-in tracks—as an audio file—to disk at any time. For details please refer to the
Logic reference.
Whatever you play on your instruments can be recorded by simply pressing Logic’s
Record button. Your performances can be freely edited in any of Logic’s MIDI editors.
Further details about this can be found in the Logic reference

Logic’s plug-ins include the following features:
• Real-time processing of audio.
• Support for sample rates up to 192 kHz.
• Altivec optimizations for the Power Macintosh G4 and G5 processors which increase
the number of software effects and instruments that can be run simultaneously.
• A sophisticated, intuitive, real-time graphical editing interface for most Logic plugins.
• A consistent window interface for Logic, Audio Unit and TDM plug-ins.
• The ability to save and load individual plug-in effect and instrument settings or
entire channel strip configurations, including those from Apple’s GarageBand
application.
• Almost all plug-in parameters can be automated via Logic’s total recall mix
automation system.

About This Manual
This guide covers all areas of plug-in usage in Logic. All plug-in parameters are
discussed in detail.
The Basics section discusses the most essential aspects of plug-in usage, the Plug-in
window interface and global plug-in commands and menus.
The Instruments and Effects chapter covers the differences between effect and
instrument plug-ins.
Ensuing chapters discuss the parameters of individual plug-in effects and instruments.
The instrument chapters include a number of tutorials that will help you to make the
most of your new instrument.
The Onscreen Help system—accessible from Logic’s Help menu—is fundamentally the
Reference Manuals in electronic form. It has the advantage of being at your fingertips
when you need it, and is also searchable.
Even if you’re the type who just doesn’t like reading manuals, we ask that you read the
next section. It will provide you with essential information on the basic use of Logic’s
plug-ins.
Please note that all topics described herein were accurate at the date of printing. For
up to date information on changes or additions made after printing, please refer to the
Late Breaking News on the Logic DVD, and/or to the Update Info, included with each
Logic update.

Preface Introducing Logic’s Plug-ins

Conventions of this Guide…
Before moving on to the Basics section, we’d like to cover the following conventions
used in this manual.
Menu Functions
For functions that can be reached via hierarchical menus, the different menu levels are
described as follows: Menu > Menu entry > Function.
Important Entries
Some text will be shown as follows:
Important: Information on function or parameter.
These entries discuss a key concept or technical information that should, or must, be
followed or taken into account. Please pay special attention to these entries.
Notes
Some sections provide additional information or tips that will assist your use of the
effect or instrument plug-in. These are displayed as shown below:
Note: Information on function or parameter.
Key Commands
Several plug-in functions can be activated or accessed with key commands—computer
keyboard shortcuts. The key commands mentioned in this guide are based on the
standard Key Command Set, assigned by the Logic Setup Assistant. Where possible, we
have also included the standard Key Commands for PowerBook users. These are based
on the PowerBook Key Command Set, assigned in the Logic Setup Assistant.

Preface Introducing Logic’s Plug-ins

Basics

This chapter covers all important steps required for plugin use in Logic.
The steps include:
• Inserting, deleting, and bypassing plug-ins.
• Operating plug-ins in the Plug-in window.
• Managing plug-in settings.
• Automating plug-ins.

Using Plug-ins
Inserting and Deleting Plug-ins
Plug-ins can be either; software instruments, which respond to MIDI note messages, or
audio effects, which do not respond to MIDI note messages.
• All plug-ins can be added via the plug-in menu of an Audio Object.
• Effect plug-ins can be inserted into the Insert slots of all Audio Objects.
• Software-based instruments can only be inserted into special Audio Objects, called
Audio Instruments. These Audio Instrument Objects have a special Instrument slot,
directly above their Output slots.

To add a plug-in:
1 Click-hold on an Audio Object’s Insert/Instrument slot.

2 The plug-in-menu appears, showing all available plug-ins. Move the mouse through
the different levels of the hierarchical menu and choose a plug-in name, then release
the mouse button.

The Plug-in window is launched automatically. If you do not want the Plug-in window
to open automatically after insertion, uncheck the Preferences > Audio > Display > Open
Plug-in window on insertion preference.
You can open a closed Plug-in window by double-clicking on an assigned Insert/
Instrument slot.
You can set all plug-in parameters in the Plug-in window. For further information
please read “The Plug-in Window” on page 16. Closing the Plug-in window leaves the
plug-in active.

Chapter 1 Basics

To remove a plug-in:
1 Click-hold the corresponding Insert/Instrument slot.
2 The plug-in menu is opened. Select the No Plug-In menu option.

Inserting Mono/Stereo Plug-ins
You can insert mono and stereo effects into Logic’s mono objects. If you use a stereo
effect in a mono object, the plug-in menu is limited to stereo effects from this insert
point onwards.
Note: In general, stereo effects require twice as much processing power as their mono
counterparts.
In stereo objects, the plug-in menu only shows effects with stereo inputs and stereo
outputs. If you hold the Option key while opening the plug-in menu on stereo objects,
you can also select mono effects.
Logic automatically inserts conversion modules (in the background) to handle stereo
→ mono and mono → stereo transitions. This enables you to use plug-ins in any order.
Please keep the following in mind when doing so:
• These conversion modules require extra processing power.
• During a stereo → mono conversion, all spatial information is lost.
• During a mono → stereo conversion, no spatial information is added—the same

mono signal is sent to both outputs.

Bypassing Plug-ins
If you want to deactivate a plug-in, but don’t want to delete it, you can bypass it.
Bypassed plug-ins do not drain system resources.

To bypass a plug-in:
Option-click the appropriate plug-in insert/instrument slot on the desired Audio
Object.
The insert slot of the bypassed plug-in turns from blue to gray, indicating that the plugin is currently bypassed.
You can also use bypass a plug-in from within the Plug-in window. Further information
on this can be found in the following section.

Chapter 1 Basics

The Plug-in Window
Hands-on operation of plug-ins is performed in the Plug-in window. This window
allows access to all plug-in parameters. The Plug-in window can be opened by doubleclicking on the blue plug-in label on an Audio Object. Each instance of a plug-in has its
own Plug-in window, allowing each to have discrete settings.

Operation of Built-in Plug-ins
Adjusting Parameters

To toggle a Plug-in window’s buttons:
Click on the button. It toggles to the next/previous option, or will be enabled/disabled.
To adjust a slider:
Click-hold anywhere on the slider and drag up/down or left/right.
To adjust rotary knobs:
Click-hold on the center of the rotary knob and drag the mouse up and down. You can
also move the mouse in a circular motion. Fine-tuning of values is easier when using a
larger radius for this circular motion.
To adjust numerical panels:
Click-hold on the panel’s numerical value and drag up/down. If there are up/down
arrows alongside such panels, you can use them to increment/decrement the value by
one step.
Note: You can reset any parameter to its default value by Option-clicking on it.
Note: If you hold Shift before clicking and moving a control, its value can be finetuned.

Common Plug-in Window Parameters
The gray area at the top of the Plug-in window is common to all Logic plug-ins. It offers
a number of important functions for plug-in use.

Link
The button to the extreme left (with a chain on it) is called the Link button. If the Link
button is switched on, a single Plug-in window will be used to display all opened plugins. Each time you launch a new plug-in, the window will update to reflect the new
selection. By default, the Link button is switched off, allowing you to open several Plugin windows simultaneously. This is handy if you want to compare the settings of two
plug-ins or adjust several plug-ins at the same time.

Chapter 1 Basics

When changing the Arrange track, an open Plug-in window will update to display the
corresponding slot’s plug-in on the newly-selected track. As an example, if the ES1 was
loaded on Audio Instrument channel 1, and an EXS24 instance was loaded on Audio
Instrument channel 1, switching between these tracks would automatically update the
Plug-in window to show the ES1/EXS24, respectively.
Bypass
The Bypass button allows a plug-in to be deactivated, but not removed from the insert/
instrument slot. You can also bypass the effect directly in the Audio Object by Optionclicking on the corresponding insert slot.
Settings Menu (Arrow)
Clicking the Arrow to the right of the Bypass button accesses the Settings menu.
Further information on this can be found in “Plug-in Settings” on page 19.
Switching the Contents of the Plug-in Window
You can reassign any open Plug-in window—in two different ways—via the two pulldown menus to the right of the Settings menu (the Arrow):
• Use the upper pull-down menu (Track 1 in the diagram) to switch the Plug-in
window between all channels that use the same plug-in. If you have inserted the
EVB3 on tracks 1 and 6, for example, you can switch between these channels and
adjust the parameters of each EVB3 instance, respectively.
• In the lower pull-down menu you can switch between the plug-in slots of the
selected channel. As an example, if a particular channel uses an Equalizer and an
EVB3 plug-in, you can switch the Plug-in window between these plug-ins.
Editor—Controls View
The plug-in parameters can be viewed in two forms: Controls and Editor. The Editor
view shows the plug-in’s graphical interface, if it offers one. The Controls view displays
all plug-in functions as a set of horizontal sliders, with numerical fields to the left of
each parameter. These fields are used for both the display and entry of data values.
To switch the view modes:
1 Click-hold the Editor button in the gray area at the top of the Plug-in window.

2 Choose Controls from the pull-down menu.

Chapter 1 Basics

Some Logic plug-ins may have additional parameters that don’t show up on the Editor
control panel. This is indicated by an additional 001/011 button next to the Link button.

Activate this button to reveal sliders for the extra parameters at the bottom of the Plugin window.
Plug-ins With Side Chain Input
All plug-ins that support side chain inputs, feature an additional Side Chain pull-down
menu in the gray area at the top of the Plug-in window. This facilitates the routing of
any Audio track, Input channel or Bus Object into the plug-in via a side chain.
You can also route an Instrument channel as side chain signal, if you follow
these steps:
1 Create a Send, using a Bus on the Instrument channel.
2 Choose the selected Bus as a Side Chain input for the plug-in.
Once the Side Chain input is selected, the plug-in processes the audio of the channel it
is inserted in, using the trigger impulses provided by the Side Chain. The signal peaks
of the Side Chain input, combined with the Threshold parameter of the plug-in,
determine when the plug-in is triggered.
Examples for Side Chaining
• A sustained pad sound is sent through a noise gate, which is triggered by a drum

track being used as the Side Chain input signal. This results in a rhythmic pad sound
which follows the signal peaks of the drum track.
• A noise gate inserted into a bass guitar channel is triggered by the kick drum track
via the Side Chain. This can “tighten” the timing of the bass guitar, as it follows the
kick drum signal.
• Side Chains can also be used to blend a music mix with a voice-over. To achieve this,
the mix needs to be routed through a compressor which, in turn, is side chained,
using the voice-over track. In this type of setup, the music becomes softer when the
narrator is speaking, and louder, when not. The effect is also known as ducking.
Please note that in order for this to function, the automatic gain make-up or Auto
Gain (if applicable to the compressor plug-in) must be disabled.

Chapter 1 Basics

Plug-in Settings
Logic’s plug-ins ship with a library of ready-to-play preset sounds, known as Settings.
These Settings can be found in the Logic > Plug-In Settings subfolder, following the
installation procedure.
Note: It is strongly recommended that you do not attempt to change the Logic > Plugin Settings folder structure. Within the Plug-in Settings folder you are, however, free to
sort your settings into sub folders. This folder structure is reflected in a hierarchical
menu, shown each time you load a plug-in setting.
All current plug-in settings are stored with the song file, and are automatically recalled
the next time you load the song. You can also recall and save individual settings via the
Settings menu functions. The Settings pull-down menu can be opened by clicking on
the Arrow in the gray area at the top of the Plug-in window.

Functions of the Settings Menu
In the gray area at the top of each Plug-in window is an Arrow button. Clicking on it
opens the Settings menu, which features the following functions:

Copy Setting
Choose this entry to copy all parameter settings into a special Settings clipboard, which
is independent from the global Logic clipboard.
Paste Setting
If you have opened a plug-in of the same type (two SilverVerb instances, for example),
you can use this command to paste the parameter set from one to the other via the
Settings clipboard.
Save Setting
This allows you to name and save a setting.
Note: If you save a Setting with the name of #default in a plug-in’s Settings folder, it will
be loaded as the default plug-in Setting.

Chapter 1 Basics

Load Setting
This function can be used to load a setting. The file selector box only shows settings for
compatible plug-in types. Each plug-in has its own set of parameters, and therefore its
own file format.
Note: Proprietary plug-in-settings created in Logic for Windows can be read by Logic
for Mac OS, and vice versa. Plug-in settings files created on the Mac must be saved with
a .pst file extension in order for them to work in Logic for Windows.
Note: Some plug-ins allow you to load Settings files by dragging and dropping them
from the Finder. This poses a problem as float windows will disappear once Logic is “in
the background”, and the Finder becomes the active application. To circumvent this
issue, you can hold Option when inserting a plug-in, making it a non-floating window.
Next/Previous Setting
These functions allow you to load the next/previous setting in the folder. You can also
make use of the Next/Previous Plug-In Setting (or the Next/Previous Plug-In Setting or EXS
Instrument) key commands. These are not set by default, so you will need to assign
them. Once assigned, you can simply press the appropriate key command to step
forwards/backwards through your plug-in settings. In Logic Pro, Previous/Next Setting
can be assigned to almost any MIDI message, such as Control Change or Program
Change commands.

Settings of other Manufacturers
Logic can read the most common settings files used by Audio Unit plug-ins.

Loading and Saving Multiple Plug-ins
Logic’s Mixer windows allow you to save and load multiple plug-ins (inclusive of their
Settings files) via the arrow pull-down menu alongside the word Inserts on channel
strips. The entire channel strip can be stored and recalled for use on any suitable Audio
Object, allowing common chains of effects such as Reverb, Chorus, and Delay to be
loaded far more quickly than individually inserting each plug-in. Further details can be
found in the Logic reference.

Plug-in Automation
Almost all Logic plug-ins can be fully automated, which means that you can record,
edit, and play back almost any movement of any knob, switch or fader in any plug-in.
For more information, please read the Automation chapter in the Logic reference.

Chapter 1 Basics

Plug-ins From Other Manufacturers
Audio Unit Support
Correctly installed third-party Audio Unit plug-ins (Effects and Instruments) can be
used in Logic. Clicking on an Audio object insert/instrument slot will launch the
hierarchical Plug-In menu. A separate Audio Units submenu displays all installed Audio
Unit plug-ins.

TDM Plug-ins
Users of a Digidesign TDM system can utilize TDM plug-ins in Logic.

Chapter 1 Basics

Instruments and Effects

This chapter explains the difference between effect and
instrument plug-ins.
Instrument plug-ins respond to MIDI note messages, effect plug-ins do not. Therefore
instrument plug-ins can only be inserted into special Audio Objects, called Audio
Instruments.

Effect Plug-ins
Logic’s effects can be installed into all insert slots of all Audio Object types (See
“Inserting and Deleting Plug-ins” on page 13.). This allows processing of all audio and
instrument signals.
There are two ways of sending audio to effects: via an insert, or via a bus (also known
as an “aux send”).

Insert Effects
With insert effects, all of the signal is processed. This means that 100% of the signal
flows through the effect. This is suitable for equalizers or dynamic effects. This also
typically applies to pan knobs and faders.
If you have enough processing capacity, you can use up to 15 insert effects per audio
object. An extra blank insert is created, as soon as all the currently displayed inserts are
used, up to the maximum allowed.

Bus Effects
When you use bus effects, a controlled amount of the signal is sent to the effect. Buses
are typically used for effects that you want to apply to several signals at the same time.

Within Logic, the effect is placed in an insert slot of a bus object. The signals of the
individual tracks can each be sent to the bus, controlled by a Send knob.

The audio signal is then processed with the effect, and mixed with the stereo output.
The advantage of this “bussed” approach, over inserting effects on tracks, is efficiency.
This method allows as many tracks as you like to be processed by one inserted plug-in,
massively saving CPU power when compared to insertion of the same effect directly
into multiple tracks.
For computationally-intensive effects such as reverb, it’s always advisable to insert
them into a bus. Chorus, Flanger, and Delay effects should also always be inserted into
a bus, if they are going to be used on more than one track.
In some cases, it may make sense to patch an effect such as a delay, directly into the
insert of an individual track. There are no restrictions in Logic as to where effects may
be used.

Instrument Plug-ins
The Audio Instrument Object Type
Unlike effect plug-ins, instrument plug-ins respond to MIDI note messages. Instrument
plug-ins can only be inserted into special Audio Objects, called Audio Instruments.
Audio Instruments feature a special instrument slot, directly above their Output slot.
An Audio Instrument is an Audio Object with its Channel parameter switched to one of
the Instruments. Any audio object can be switched to operate as an Audio Instrument,
by changing this parameter (Channel) in the Object Parameter box.
To create an Audio Instrument Object:
1 Open Logic’s Audio Mixer, by choosing Audio > Audio Mixer.
2 In the Audio mixer window select New > Audio Object to create a new Audio Object.

Chapter 2 Instruments and Effects

3 Double click the newly-created Audio Object icon, so that the (grayed out) channel
strip appears.

4 Now, go to the Object Parameter box, and set the Channel parameter to an Instrument.
The generic Audio Object will now operate as an Audio Instrument, allowing you to
insert any Instrument plug-in into the instrument slot.
The default song—the song that opens if you move the Autoload Song away from the
Logic folder—features a number of ready-configured Instruments, that can be accessed
via the Track Mixer or Audio Mixer.
The output signal of a software instrument plug-in is fed into the input (the instrument
slot) of the Instrument channel strip, where it can be processed via inserted plug-ins
and/or sent to busses.
Logic supports up to 64 discrete Audio Instruments. The number of instrument
instances which can be run simultaneously is dependent on the availability of
computer processing resources.
Following the insertion of an instrument, the Audio Instrument Object can be used just
like a MIDI track in the Arrange window. The Audio Instrument Object can also receive
MIDI notes from standard MIDI instrument objects via Environment cables. This is
useful for creating layered sounds with “real” MIDI instruments and virtual instruments.
Please note that the Options > Preferences > MIDI > Use Unified Virtual and Classic MIDI
Engine setting needs to be switched on for these features to work.
When an Audio Instrument track is selected, it is ready to be played in real-time and
consequently produces some system load. Normally, Logic releases system resources
used by the Audio Engine when the sequencer is stopped. This is not the case,
however, if an Audio Instrument track is selected in the Arrange window, and is
therefore available for real-time playing. Selecting a MIDI track or a standard Audio
track exits this Audio Instrument “stand by” mode, and releases reserved system
resources when the sequencer is stopped.
Note: Muting an Audio Instrument track in the Arrange does not reduce system load.

Chapter 2 Instruments and Effects

Logic’s Bounce function allows the entire Audio Instrument track to be recorded as an
audio file. This “Bounced” audio file can then be assigned (as an audio region) to a
standard Audio track, allowing you to reassign the available processing (CPU) power for
further synthesizer tracks. For details, please refer to the Bounce chapter in the Logic
Reference manual.
You can also make use of the Freeze function to capture the output of an Audio
Instrument track, again saving processing power. For details please refer to the Freeze
section, in the Logic Reference manual.

Accessing Multiple Outputs
Logic supports the multiple outputs of the EXS24 and all Audio Unit (AU) compatible
instruments. In addition to the Mono and Stereo submenus of the Audio Instrument
plug-in menu, a Multi Channel submenu lists all Instruments that offer multiple outputs.
A plug-in needs to be inserted from the Multi Channel submenu, in order to access its
individual outputs.

Note: Not all plug-ins (both Logic and third-party) are multi-output capable. If the
Instrument does not appear in the Multi Channel submenu, it is not equipped with
multiple output facilities.
The first two outputs of a multiple output instrument are always played back as a
stereo pair by the Instrument channel in which the plug-in is inserted. Additional
outputs (3 and 4, 5, and 6, and so on) are accessed via the Aux Objects.

Chapter 2 Instruments and Effects

Software Instrument Pitch
The Song Settings > Tuning > Software Instrument Pitch > Tune parameter remotely
controls the main tuning parameter for all software instruments (plug-in synthesizers,
such as the ES1 or EXS24 sampler and others) by ±100 cents.

Note: Some instruments do not recognize this remote command.

No Hermode Tuning
Logic allows all software instruments to be globally tuned to different tempered scales,
including Hermode Tuning (see the Tuning section of the Logic reference manual for
details). There may, however, be occasions where you want individual Software
Instruments to be exempt from this global tuning system.
When File > Song Settings > Tuning > Hermode Tuning is active, a No HMT checkbox is
visible in the Object Parameter boxes of all Audio Instrument channels. Simply click in
this box to prevent the selected software instrument from following the global
Hermode Tuning scale.
Any software instrument with an active No HMT checkbox will be played back at equal
temperament.

Chapter 2 Instruments and Effects

Equalizer

This chapter covers all Logic equalization effects.
Equalizers allow you to increase or decrease the level of
selected components in the overall audio spectrum.
Logic’s built-in equalizers include the Channel EQ, Linear Phase EQ, Match EQ, Fat EQ,
Silver EQ, DJ EQ, High/Low Pass Filters, High/Low Cut EQ, Parametric EQ and High/Low
Shelving EQ plug-ins.

Channel EQ

The extremely high-quality Channel EQ offers eight frequency bands and an
integrated FFT analyzer.

EQ Parameters
The Band Type buttons above the display activate the Channel EQ’s bands individually;
inactive bands do not use any computer resources.
Band 1 is a lowpass filter and band 8 is a highpass filter.
Note: The Q-parameter of band 1 and band 8will have no effect when using a slope of
6 dB/Oct.

Bands 2 and 7 are defined as shelving equalizers.
Note: When the Q parameter of band 2 and 7 is set to an extremely high value (to 100,
for example), the equalizers only apply to a very narrow band, and can work in a
fashion that is similar to notch filters.
Bands 3to 6 are bandpass filters.
You can set the band parameters either in the parameter area or directly in the central
EQ display. Move the mouse horizontally over the display. When your mouse cursor is in
the access area of a band, its individual curve and parameter area will be highlighted
and a pivot point appears. When you click-hold the mouse button directly on the
(illuminated) pivot point of a band, vertical movements (up/down) will change its Q
value. Horizontal movements (left/right) change the Frequency of the band. When you
click-hold the mouse button on the display background, horizontal movements will
again change the Frequency of the band. Vertical mouse movements will change the
Gain of band 2 to 7. The slope values of the highpass and lowpass filters (bands 1 and
8) can only be changed in the parameter area below the graphic display. Click-hold on
the parameter: Moving up increases, and down decreases, the value.
After boosting or cutting frequency bands, you can use the Master Gain fader to
readjust the output level of the Channel EQ.

Channel EQ—Analyzer
The precision analyzer of the Channel EQ uses Fast Fourier Transformation (FFT) to
show the energy of all frequency components of the signal. The central display of the
Channel EQ fulfills multiple display functions: it shows both the curve of the FFT
analyzer and the EQ curve. An identically scaled frequency axis is shown for both. This
allows you to easily recognize unwanted frequencies in the analyzer curve, while using
the EQ to edit them accordingly.
A click on the Analyzer button activates/deactivates the FFT analyzer. The display
directly under the button determines the location of the Analyzer. You can switch the
Analyzer pre EQ or post EQ (default) in order to compare the original signal with your
edits.
Click into the display to open a pull-down menu that defines the resolution of the FFT
analyzer—or more accurately, the number of frequency bands. The higher the precision
of measurements, the more CPU power is needed.
High resolutions are necessary whenever you need reliable results in the very low bass
frequency area. The bands derived from FFT analysis are divided in accordance with the
frequency linear principle—non-technically, this means that there are far more bands
in the highest octave than in the lowest.

Chapter 3 Equalizer

Use the Scales to the left and right of the EQ display, to change the vertical scale of the
EQ and analyzer curves.
To increase the resolution of the EQ Gain parameter (dB Warp) in the most interesting
area around the zero line, click-hold in the green dB Scale on the left side of the
graphic display, and move the mouse up. Moving the mouse down, will decrease the
parameter value. The overall range is always ±30, but small values will be easier to
recognize.
As soon as the Analyzer is activated, you can change the Analyzer Top parameter, which
alters the scaling of the FFT analyzer, on the right side of the graphic display. The visible
area represents a dynamic range of 60 dB, but by click-holding and vertically dragging,
you can adjust the maximum value between +20 dB and −40 dB. The Analyzer display
is always dB-linear.
Two additional Analyzer parameters are available via the 001/100 view. Analyzer Mode
allows you to switch between Peak and RMS. Analyzer Decay allows you to adjust the
decay rate (in dB per second) of the Analyzer curve (peak decay in Peak mode or an
averaged decay in RMS mode)
Note: The FFT analyzer needs additional CPU resources. In fact, resource consumption
increases significantly at higher resolutions! We recommend that you disable the
Analyzer or close the Channel EQ window after setting the desired EQ parameters. This
will free up CPU resources for other tasks.

Using the Channel EQ as the Default EQ
The Channel EQ replaces the Track EQ of older Logic versions. It is inserted into the first
available insert slot by double-clicking the EQ area on the upper portion of mixer
channel strips. This area will change to a thumbnail view of the Channel EQ display. The
thumbnails provide an overview of the EQ settings used in each individual channel.

Chapter 3 Equalizer

Linear Phase EQ

The extremely high-quality Linear Phase EQ plug-in is almost identical to the Channel
EQ. With the exception of the different name and a few different colors, it uses the
same familiar eight-band layout, and method of operation, as the Channel EQ.
Under-the-hood, however, the Linear Phase EQ uses completely different technology
which preserves the phase of the audio signal 100%—even if the wildest EQ curves are
applied to the sharpest signal transients!
As with all good things in life, there is a catch. The Linear Phase EQ uses more CPU
power than the Channel EQ. Another factor is the inherent amount of latency
introduced by this technology. Logic’s plug-in delay compensation will successfully
prevent the worst of these latency artefacts in mixdown situations—but don’t even
think about playing software instruments live when using the Linear Phase EQ.
As the parameters of the Channel EQ and Linear Phase EQ are almost identical, you
may freely copy settings between them. For more information on the parameters of
the Linear Phase EQ, read up on the “Channel EQ” on page 29.

Chapter 3 Equalizer

Match EQ

The Match EQ plug-in allows you to “match”, and transfer the frequency spectrum from
one signal to another, or to store it as a spectral template file. In this way, you can
acoustically match the sound of various songs for an album, or impart the “sound” of
any reference source onto your own recordings. The alignment of signals is automatic,
but you can also manually draw or modify the filter curve to alter the sound as
required.
Note: Match EQ acoustically matches two audio signals. It does not, however, match
any dynamic differences in the two signals.

Description of the basic functions
Match EQ is a learning equalizer that reads the frequency spectrum of any reference
source, including: the input signal, an audio file, or a template. Alternatively, you can
load a setting file or import the settings of another Match EQ instance via a copy and
paste operation.
You can analyze the average audio spectrum of the track the plug-in is assigned to or
load another setting file or template. By matching both spectra, a filter curve is
generated. This generated curve adapts the track signal to match the sound of the
template. If required, you can modify the filter curve by boosting or cutting gain in
different frequencies, or inverting it. Further to this, you can manually modify the curve
by creating a virtually infinite number of peak filters, and adjust them as required. In
this way, you can draw your own filter curve to optimize the sound as required. The
internal analyzer allows you to visually check the frequency spectrum of the original
data and the resulting curve, making manual corrections at specific points within the
spectrum easier.

Chapter 3 Equalizer

Parameters
The View pull-down menu allows you to select the type of information shown on the
analyzer display in the center. The following options are available:
• Automatic: Depending on the selected function, the analyzer view is automatically
toggled between the three following options.
• Template: The analyzer display shows the average frequency curve, which is
generated by analyzing the input signal or loading a template.
• Current Material: The analyzer visualizes the average frequency curve, which is
generated by analyzing the track signal or loading a Setting file or template.
• Filter: The analyzer displays the filter curve, which is generated by matching the
Template and the spectra of the Current Material.
Independent of the selection, the analyzer can be activated/deactivated via the On/Off
button. The Analyzer Position pull-down menu allows you to place the analyzer tap
before (Pre: unchanged) or behind the filter circuit (Post: behind the Match EQ).
Note: Deactivating the analyzer frees up processing power for other applications.
On stereo channels, the view mode is configured via the lower View toggle menu. You
can select whether the analyzer displays both audio channels via separate curves (L&R)
or the summed maximum level (LR Max).
You can manually modify the filter curve generated via matching the Template with the
Current Material. The buttons in the Select section let you choose whether the
modifications are applied only to the left, right, or both channels.
You can refine this selection via the Channel Link slider. If the slider is set to 1.0, the L
and R buttons for the single channels will have no effect, because both channels are
represented via a common EQ curve. At the minimum value of 0.0, two separate filter
curves are displayed, each of which can be selected for editing via the L and R buttons.
The intermediate settings of the Channel Link slider allow you to blend these extreme
values as required. As a result, any modification to either of the filter curves is
transferred to both channels, dependent on the Channel Link setting.
Note: In the mono version of the plug-in, the parameters in the View Mode, Select, and
Channel Link sections have no effect.
The Template and Current Material buttons perform the spectral analysis of the audio
signals, and match the resulting curves. Clicking the Learn button in the Template
section starts and stops measurement of the average frequency spectrum in the
reference signal.
Clicking the Learn button in the Current Material section starts and stops measurement
of the average frequency spectrum in the audio material of the track.

Chapter 3 Equalizer

Note: Audio files can also be dragged onto the Template or Current Material Learn
buttons to generate template or current spectra. A progress bar displays the progress
of the analysis process.
If you right-click (or Control click) on either of the Learn buttons, a context menu opens.
This menu allows the spectrum of the template or the track signal (Current Material) to
be:
• cleared
• copied to the Match EQ clipboard, which is common to all Match EQ instances in the
current song.
• pasted from the Match EQ clipboard to the active instance.
• loaded from a stored Setting file.
• generated from an audio file (chosen in the File Selector). This is done by choosing
the Generate Template/Current Material Spectrum from audio file option, and selecting
an appropriate file in the file selector that appears. A progress bar displays the status
of the analysis process.
Note: When you activate the Learn button in either the Template or Current Material
section, the View parameter is set to Auto, and the analyzer will display the current
status of the spectral analysis, indicated by a progress bar.
The Match button in the Current Material section allows you to write the differences
between the learned or loaded Template and the learned or loaded spectrum of the
Current Material to a filter curve. Differences in gain are automatically compensated for,
with the resulting EQ curve referenced to the 0 dB line.
The filter curve is updated automatically each time a new template or current material
spectrum is learned or loaded, when the Match button is enabled. You can toggle
between the matched (and possibly scaled and/or manually modified) filter curve and
a flat response by activating/deactivating the Match button.
Note: Each time a new audio spectrum is matched—either by loading/learning a new
spectrum while Match is activated or by activating Match after a new spectrum has
been loaded—existing manual modifications are discarded, and Apply is set to 100%.
Basically, only one of the Learn buttons may be active at a time. As an example, if the
Learn button in the Template section is active and you press the Learn button in the
Current Material section, the analysis of the template file stops, and the current status is
used as the spectral template. Analysis of the track (Current Material) will then begin.
Note: If you have manually modified the filter curve, the original (or flat) curve can be
restored by Option-clicking on the background of the analyzer display. A second
Option-click restores the most recently modified curve.

Chapter 3 Equalizer

The filter curve can be edited via the Smoothing slider. At a value of 0.0, the filter curve
is applied to the track signal without any changes. At all other Smoothing settings, the
filters are smoothed at a constant bandwidth. A value of 1.0, for example, means that all
filters have a constant bandwidth of one semitone that is used to smooth the notchlike filters in the curve. A bandwidth of: four semitones (a value of 4.0—or a major
third), an octave (a value of 12.0) and two octaves with the maximum setting (24.0).
Note: Smoothing does not affect any manual modifications of the filter curve.
The Apply slider exaggerates (101% to 200%), reduces (99% to 1%) or inverts the peaks/
dips (−1% to −100%) the effect of the filter curve on the track signal. At a value of 100%,
the signal is aligned to the curve without any changes.
The Phase toggle menu switches the operational principle of the filter curve.
• The Linear option prevents processing from altering the signal phase. At the same
time, the latency of the plug-in will increase.
• The Minimal option alters the signal phase, but latency is reduced.

Manual Modifications
You can graphically edit the matched filter curve directly in the display. Just click at any
point within the filter curve to create a new peak band. You can shift the peak
frequency for this band (within the entire spectrum) by dragging the mouse
horizontally. Vertically moving the mouse allows you to set the gain of this frequency
band (range: −24 to +24 dB). The Q-factor of the filter is set by the vertical distance
between the click point and the curve. By clicking on the curve, the maximum Q-value
of 10 (for notch-like filters) is used. Clicking above or below the curve decreases the Qvalue. The further you click from the curve, the smaller the value (down to the
minimum of 0.3).
• The Q -factor can be changed continuously by pressing Shift and moving the mouse
up/down while keeping the mouse button pressed.
• If Option is hold while releasing the mouse button, the modification is cancelled.
Note: The current values are shown in a window within the display while the left
mouse button is held down.
The colors and modes of the dB scales on the left and right of the display are
automatically adapted to the active function. If the analyzer is active, the left scale
displays the average spectrum in the signal, while the right scale serves as a reference
for the peak values of the analyzer. Basically, the analyzer visualizes a dynamic range of
60 dB. The displayed range can, however, be shifted between the extreme values of
+20 dB and −100 dB by click-dragging on the scale.

Chapter 3 Equalizer

If the resulting filter curve is displayed, the left scale—and the right, if the analyzer is
inactive—shows the dB values for the filter curve in an appropriate color. By clickdragging on one of the scales, the overall gain of the filter curve is adjusted in the
range from −30 to +30 dB.

Fat EQ

The high-quality Fat EQ offers up to 5 fully parametric bands—buttons 1 through 5
activate these individually; inactive bands do not drain your computer’s resources.
The icons above the graphic display let you determine whether Band 1 acts like a
highpass filter or a low shelving EQ. Similarly, Band 5 can be switched back and forth
between use as a lowpass filter and a high shelving EQ. Bands 2 and 4 can be switched
from their normal operating mode (as fully parametric bandpass filters) to low or high
shelving EQs. The center band (number 3) always operates as a fully parametric
bandpass filter. The shelving filter’s slope characteristics for bands 2 and 4 are
adjustable via the Q parameter.
The area directly below the graphic display (depicting the frequency response curve) is
used to select the frequency for the individual bands. Simply click on the number, and
change the value with your mouse. You’ll be able to hear an individual frequency better
if you turn it up by rotating the Cut/Boost knob located below it clockwise.
The same holds true for any frequency that you want to attenuate. Once you’ve located
the frequency that you’re hunting for, you can back off the Cut/Boost knob level, and
set it to the desired value. Use the Q (bandwidth) parameter located in the lower
display to determine the extent that the band influences neighboring frequencies.

Chapter 3 Equalizer

At low Q values, the EQ influences a wider frequency range, and at high Q values, the
effect of the EQ band is limited to a very narrow frequency range. Please bear in mind
that your perception of an attenuated or boosted frequency depends heavily on the Q
parameter: If you’re working with a narrow frequency band, you’ll generally need to
cut or boost it more drastically to notice a difference.

Silver EQ
The Silver EQ contains one High Shelf, a Parametric and one Low Shelf filter with the
corresponding parameters. More on each of these is found in the Individual EQ’s
section below.

DJ EQ
The DJ EQ combines Low and High Shelving Filters with a fixed frequency, and one
Parametric EQ with its attendant parameters. More on each of these is found in the
Individual EQ’s section below.
The special feature of the DJ EQ is that it allows the gain of the filters to be reduced
down to −30 dB.

Individual EQs
Parametric EQ
The Parametric EQ offers the following three parameters:
• Hz: Center frequency
• dB: Cut/Boost
• Q: Quality
A symmetrical frequency range on either side of the center frequency is boosted or cut.
You can adjust the width of this frequency range with the Q control.

High Shelving EQ/Low Shelving EQ
• The Low Shelving Equalizer only affects the frequency range below the selected

frequency.
• The High Shelving Equalizer only affects the frequency range above the selected

frequency.

Chapter 3 Equalizer

Dynamic

This chapter introduces Logic’s Dynamic plug-ins.
This includes the Compressor, Silver Compressor, Expander, Noise Gate, Silver Gate,
Enveloper, DeEsser, Limiter, Multipressor, and Adaptive Limiter plug-ins.

Compressor

A compressor tightens up the dynamics of a signal. This means that the difference in
levels between loud and soft passages is reduced. This “evening out” of the loud and
soft passages means that the peak level remains pretty constant, and the overall
loudness—the perceived volume—of a track is increased. Next to an EQ, a compressor
is your most valuable sound-shaping tool when mixing. A compressor is a universal
effect, it has a virtually unlimited range of applications. You should definitely exploit it
for vocal tracks, but a compressor can also often work wonders for entire mixes. When
you use a compressor, be sure to route the entire signal through it, by inserting it
directly into channels. It should only be used in a bus when you want to compress a
group of tracks (a drum kit, for example) simultaneously, and by the same amount.
Again, these tracks (individual drums in a kit, for example) should be routed to the bus
in their entirety, as opposed to using Send knobs to route just parts of each signal to
the bus. You do this by selecting the appropriate bus as the output destination for the
tracks that you wish to compress.

Logic’s Compressor was designed to emulate the response of the finest analog
compressors. It follows the following principle: When a signal exceeds the defined
Threshold level, the compressor actually alters the response, so that it is no longer
linear. What happens is that all levels that exceed the Threshold are attenuated by the
value set with the Ratio slider. A ratio of 4:1 means that an incoming level that is 4 dB
louder than the Threshold level is dampened, so that it comes out the other end of the
compressor with a level that is just 1 dB above the Threshold level. On the flip side, if
you route in a signal that is loud enough to double the output level of the compressor
(+6 dB), the input signal would need to have a level 24 dB greater than the Threshold
level. This tells us that a compression ratio of 4:1 is a fairly drastic manipulation of the
original signal’s dynamics. Given that the compressor lowers levels, the volume of its
output signal is normally lower than that of the input signal.
To compensate for this decrease in levels, the output of the compressor is equipped
with a Gain slider. Auto Gain automatically sets the level of amplification to a value
equivalent to the “sum of the threshold value minus the threshold value divided by the
ratio” or put less confusingly T—(T/R). This function ensures that a normalized input
signal is amplified so that the output signal is also normalized, regardless of the values
set for Threshold and Ratio—provided you are dealing with relatively static signals. Use
the Attack and Release knobs to shape the dynamic response of the compressor. Attack
determines the amount of time it takes for the compressor to react to signals that
exceed the Threshold. At higher values, the compressor does not fully dampen a signal
until it runs through its Attack phase. This type of setting ensures the original attack, for
example the sound of a pick or finger striking a guitar string, remains intact or clearly
audible. If, on the other hand, you want to maximize the level of a master signal, set the
Attack knob to low values, ensuring that the compressor responds more swiftly. Release
determines the amount of time it takes for the compressor to stop dampening louder
passages, once the signal level falls below the Threshold level. If the compressor
generates an ugly pumping sound, adjust the Release knob accordingly.

Chapter 4 Dynamic

When you have configured a compressor so that it dampens the signal at or above the
Threshold value by the predetermined Ratio, while the level just below the Threshold is
routed through at a 1:1 Ratio, an audio engineer would term the compression as hard
knee. In many cases, however, you’ll come up with a better sounding track by using a
more gradual transition from the 1:1 Ratio below the Threshold, to the Ratio that you
entered for levels above the Threshold. In this scenario, the characteristic curve is not as
radical—it rises gradually from the bottom left to the top right, as seen in the graphic
display. This type of compression is called soft knee. The Knee slider lets you
incrementally select anything from hard to soft knee. This wide range of options
provides you with the tools you need to shape the sound as you like; whether you
want to radically maximize loudness with absolutely no regard for the original
dynamics (hard), or are going for the more musical compression that acoustic
recordings typically require (soft). Keep in mind that Knee only controls the shape of
the compression, not its intensity; use the Threshold and Ratio sliders for this purpose.
Incidentally, the Gain Reduction Meter indicates the intensity of compression used to
tighten up the original signal. This feature is a great help, particularly if you’re not
experienced with using compression. Keep an eye on it to make sure that you’re not
overly compressing your tracks.
When the compressor has to decide whether or not the level exceeds the Threshold (or
if the level is getting close to the Threshold, for soft knee compression), it can analyze
either the peak or RMS level. The latter value is a better indication of how humans
perceive loudness. When you use the compressor primarily as a limiter, select the Peak
button. When you’re compressing individual signals, use of the RMS button will often
deliver better, more musical results.
If you activate Auto Gain and RMS simultaneously, the signal may be saturated. If you
hear any distortion, switch Auto Gain off, and enter a suitable gain level manually.
The Output Clip parameter limits (clips) the output to 0 dB, via the OFF/SOFT/HARD
settings. This setting is only available in the Controls view.
Note: Despite all of these handy tips for tweaking sounds, you should always keep one
thing in mind—there are no hard and fast rules. Use your own taste and ears. If it
sounds good, it is good.

Chapter 4 Dynamic

Silver Compressor

The Silver Compressor is a simplified version of the Compressor. It is limited to
Threshold, Attack, Release, and Ratio controls.

Expander

The Expander is similar to the Compressor, with one fundamental difference—it
increases, rather than reduces, the dynamic range above the Threshold.
The Ratio slider features a value range of 1:1 to 0.5:1. This means that the Expander is a
genuine upward expander (as opposed to a downward expander that increases the
dynamic range below the Threshold). You can use this effect to emphasize the
transients of highly compressed signals. This spices up the sonic image, making it
sound livelier and fresher.
Please bear in mind the fact that you will perceive the signal as being softer, even
when the peak level remains the same. In other words, the expander decreases
loudness. If you manipulate the dynamics of a signal fairly radically (depending on the
Threshold and Ratio values), you’ll find that you’ll need to reduce the level via the Gain
slider to avoid distortion. In most cases, Auto Gain will take care of this for you.
Please check the Compressor section for details on the various parameters.

Chapter 4 Dynamic

Noise Gate

Ordinarily, a noise gate suppresses unwanted noise that may become audible during a
lull in the signal. You can, however, also use it as a creative sound-sculpting tool.
Here’s the basic principle behind a noise gate: Signals that lie above the Threshold are
allowed to pass unimpeded (open gate). Anything below the defined Threshold
(background noise, crosstalk from other signal sources and so on) is fully muted (a
closed gate). In other words, the Threshold slider determines the lowest level that a
signal must be at, in order to open the gate—it separates the wanted or useful signal,
from the unwanted or noise signal.
The Reduction slider allows you to control the intensity of noise suppression. As a rule,
you should set it to the lowest possible value and leave it there, to ensure that the gate
closes completely. If you prefer, you can select other values, thus reducing the noise
signal less dramatically. As an alternative, you can actually boost the signal by up to
20 dB.
The three rotary knobs (at the top) influence the dynamic response of the noise gate. If
you want the gate to open extremely quickly, say for percussive signals such as drums,
set the Attack knob to the lowest value by turning it as far as it will go counterclockwise. If the signal fades in a bit more softly, as is the case with string pads and the
like, a noise gate that opens too quickly can wreak havoc with the signal, causing it to
sound unnatural.
For this type of sonic scenario, set the Attack knob so that the gate emulates the attack
of the original signal. Much the same holds true for the Release phase of signals. When
you’re working with signals that fade out gradually or have longer reverb tails, you
should turn the Release knob up, allowing the signal to fade naturally.
The Hold knob determines the minimum amount of time that the gate stays open. This
knob avoids the dreaded chattering effect caused by a rapidly opening and closing
noise gate. The Hysteresis slider provides another option for avoiding chatter, without
needing to define a minimum Hold time.

Chapter 4 Dynamic

Let’s back up a bit for a brief explanation: Noise gates often begin chattering when the
level of a signal fluctuates slightly, but very rapidly, during the attack or release phase.
Instead of clearly exceeding or falling short of the Threshold value, the signal level
hovers around the Threshold. The Noise Gate then rapidly switches on and off to
compensate, producing the undesirable chattering effect. If you were able to tell the
Noise Gate to open at the determined Threshold level and remain open until the level
drops below another, lower, predefined Threshold level, you’d be able to avoid
chatter—as long as the sonic window formed by these two Threshold values is large
enough to contain the fluctuating level of the incoming signal.
This is exactly what the Hysteresis feature enables you to do—the value determined by
the Hysteresis slider is actually the difference between the Threshold values that open
and close the gate. This value is always negative. Generally, −6 dB is a good place to
start.
If you’re dealing with audio material featuring extremely sensitive transients, or attack
phases that are critical to the overall sound, you may find it beneficial to have the Noise
Gate open up a tad before the useful signal fades in. This is what the Lookahead slider
is designed for. The program analyzes the signal level ahead of time, and anticipates
the point at which it can open the gate before the signal actually reaches the Threshold
value. When you choose to use this feature, please make sure you set the Attack, Hold
or Hysteresis controls to appropriate values.
When you’re working with noise gates, you’ll run across scenarios where the useful
signal and the noise signal have levels that are near enough to be perceived as
identical. A typical example is the crosstalk of a hi-hat—its signal tends to bleed into
the snare drum track when you’re recording a drum kit. If you’re using a noise gate to
isolate the snare, you’ll find that the hi-hat will also open the gate in many cases. To
avoid this effect, the Noise Gate offers Side Chain filters.
When you press and hold the Monitor button, you can audition the Side Chain signal.
You can then set the filters to only allow frequencies that contain a particularly loud,
useful signal to pass. For this example, we’ll use the Noise Gate’s High Cut filter—that
only allows the bottom end and mids of the snare to pass, and cuts the higher
frequencies of the hi-hat. When you switch Side Chain Monitoring off, it will be much
easier to set a suitable Threshold level. This will be a value that is only exceeded by the
level of the louder useful signal—the frequencies that make up the snare’s
fundamental tone, in our example. Put simply, the Noise Gate only allows the sound of
the snare to pass. Should the need arise, you can follow much the same procedure to
isolate a kick or snare drum within an entire mixdown.

Chapter 4 Dynamic

Silver Gate

The Silver Gate is a cut-down version of the Noise Gate. It is limited to Threshold,
Lookahead, Attack, Hold, and Release controls.

Enveloper

The Enveloper is an unusual tool that lets you shape transients—the attack and release
phases of signals. No other type of dynamic effect (such as a compressor or expander)
can achieve similar results—and these results can be quite impressive indeed.
The most important Enveloper controls are the two Gain sliders that govern Attack (left)
and Release (right). In the center position, the signal remains unprocessed. If you turn
the gain up, the attack or release phase is emphasized. If you turn it down, the
corresponding phase is attenuated. As an example, boosting the attack lends a drum
sound more snap, or amplifies the sound of a guitar string being plucked or picked.
When you cut the attack, percussive signals fade in more softly. You can also mute the
attack, making it virtually inaudible. This facilitates a range of interesting manipulations.
Another handy application for this plug-in is maintaining friendships—it allows you to
mask the poor timing of accompanying instruments, rather than tell your pals that they
“groove like a bunch of accountants at the office Christmas party”.

Chapter 4 Dynamic

Emphasizing the release also boosts the amount of any reverb on the affected track.
Conversely, when you tone down the release phase, tracks originally drenched in
reverb end up sounding drier. This effect is particularly useful when you’re working
with drumloops, but there are, of course, many other applications. Let your imagination
be your guide.
When using the Enveloper, you should set the Threshold to the minimum value and
leave it there. Only when you seriously crank the release phase, thus boosting the noise
level of the original recording, should you turn the Threshold slider up a little. This limits
the Enveloper to only influencing the useful signal.
Drastic boosting or cutting of the release or attack phase may change the overall level
of the signal. The Out Level slider allows you to compensate for this effect.
The Time parameters for the attack and release phase (2 knobs below the graphic
display) enable you to access the time-based intervals that the plug-in interprets as the
attack and release phases. Generally, you’ll find values of around 20 ms (attack) and
1500 ms (release) are fine to start with. Adjust them accordingly for the type of signal
that you’re processing.
Similar to its Noise Gate counterpart, the Lookahead slider allows you to define values
that tell the Enveloper to anticipate what the signal will do in the very near future.
Normally, you won’t need to use this feature, except possibly for signals with extremely
sensitive transients. If you do decide to use Lookahead, you may need to adjust the
attack time accordingly.
To give you a better insight into the true nature of the Enveloper, here’s a quick look at
how it works: It is equipped with two internal envelope followers. One follows the
amplitude of the input signal directly, whereas the other follows all changes generated
by the variable delays (individually adjustable for attack and release). The difference
between the two envelope followers is used to boost or cut the original signal by way
of the corresponding Gain sliders (also individually adjustable for attack and release).
In contrast to a compressor or expander, the Enveloper operates independently of the
absolute level of the input signal—provided the Threshold slider is set to the lowest
possible value.

Chapter 4 Dynamic

DeEsser

A DeEsser is a signal processor used for the rejection of hissing, or sibilant noises. This is
why it is called a “DeEsser”, and occasionally an “S”-Suppressor. You can, of course, reject
sizzling frequencies with an equalizer, but a DeEsser only rejects this high frequency
band for as long as a threshold level is being exceeded in a specific frequency band.
This “dynamic” ability is why the sound doesn’t get darker when no “sizzling”
consonants are present in the signal. A DeEsser is a frequency-specific compressor,
designed to only compress a particular frequency band within a complex full band
signal. It features extremely fast attack and release times.
In the Logic DeEsser, the dynamic rejection does not necessarily need to take place in
the same frequency range that’s being analyzed. Rather, the DeEsser performs a gain
reduction in the frequency band displayed in the lower window for as long as the level
exceeds a threshold (which falls within the frequency range) displayed in the upper
window.
Note: Please don’t confuse a DeEsser with an effect known as a “Vocal Stressor”. The
latter reduces the gain of the entire range when the level exceeds a threshold defined
in a given frequency range. This type of processing can be achieved with any
compressor with a high pass filter or EQ inserted in its side chain.
The Logic DeEsser does not make use of a frequency dividing network (a crossover,
utilizing low and high pass filters). Rather, it is based on a subtraction of the isolated
frequency band, leaving the phase-curve untouched.
The DeEsser is especially important in FM radio applications, because sharp S-type
consonants can cause harsh intermodulation distortion noises. The need for this
depends very much on the language spoken: English has fewer of these consonants
than German or Spanish, for example.

Chapter 4 Dynamic

Parameters of the DeEsser
Detector Frequency
This parameter defines the frequency band the DeEsser acts upon. It’s not necessarily
the same band that will be reduced.
Detector Sensitivity
This parameter defines the threshold level that needs to be exceeded (around the
Detector Frequency), in order to reduce the level around the Suppressor Frequency.
Monitor
Activation of this switch allows you to monitor the Side Chain signal used by the
DeEsser. If you want to reduce sizzling noises, listen to the input signal, and set the
Detector Frequency in a way that makes the sizzling frequency range more prominent.
(If you find that you like this filtered sound, combine a highpass and lowpass filter—in
order to construct a bandpass—as this approach uses less processing power).
Suppressor Frequency
This parameter defines the frequency band that is reduced when the Detector
Frequency Sensitivity threshold is exceeded.
Strength
Strength sets the amount of gain reduction around the Suppressor Frequency.
Smoothing
Smoothing controls the reaction speed of the gain reduction start and end phases. It’s
a combination of attack and release time parameters, as known from compressors.

Chapter 4 Dynamic

Limiter

The Limiter is also a standard effect for processing a summed stereo signal. It is
normally used for mastering.
You could say that a limiter is a compressor with an infinite compression ratio. The
Limiter restricts dynamics to an absolute level. Any input level that exceeds the
Limiter’s threshold (Gain) will be output at this “limited” level, no matter how much
higher the original signal level may have been. The fact that there is a level that the
signal cannot exceed is the distinguishing characteristic of a limiter, when compared to
a compressor.

Parameters of the Limiter
Gain
Most analog limiters would have a “Threshold” control (like that of the Multipressor),
rather than a “Gain” control. This sets the level at which the Limiter will begin to work.
As the Limiter is digital, and is normally is applied as the very last mastering tool, we
can presuppose that:
• the input signal sometimes reaches 0 dB, but does not exceed this value, and
• that the Limiter is being used to raise the signal’s overall volume. This is the reason
why you find a Gain control here—to set the desired level of gain for the signal.
The Limiter is designed in such a way that if set to 0 dB Gain and 0 dB Output Level, it
doesn’t work at all—on normalized regions. If there should be a region that clips (red
gain line), the Limiter—using its basic settings—reduces the level before clipping can
occur. (This does not apply to data that was clipped during recording).
Lookahead
Lookahead determines how far the processor looks into the future, in order to react
earlier (thus better) to peak volumes. Unlike stand-alone processors, this function does
not apply a general signal delay, as the Limiter is not limited to seeing the signal in
real-time.
Set Lookahead to higher values, if you want the limiting effect to take place before the
maximum level is reached.

Chapter 4 Dynamic

Release
Here, you can set the time required by the Limiter (after limiting) to release the effect.
Output Level
This simple volume control sets the desired maximum level of the Limiter’s output
signal.
Softknee
Activate the Softknee button to produce a softer transition from no limiting to full
limiting.
If switched off, the signal will be limited (following a linear curve) absolutely and
exactly when a level of 0 dB is reached.
If switched on, the transition to full limiting is non-linear, meaning softer. The limiting
of the signal will start before a level of 0 dB is reached. This will avoid distortion
artefacts occurring when strong limiting is used without softknee.
Graphic Display
The graphic display shows the reduction of the level (starting from 0 dB downwards).

Adaptive Limiter

With its Compressor (see “Compressor” on page 39), Multipressor (see “Multipressor” on
page 52) and Limiter (see “Limiter” on page 49) Logic features several extremely
versatile options for increasing perceived volumes. A further tool which can be used to
increase the perceived level of signals is the Adaptive Limiter plug-in. In the world of
analog processors, it could be more closely compared to a clipper, rounding and
smoothing only harsh level peaks, rather than to a VCA-type limiter. It allows you to
achieve maximum gain, without having to fear exceeding 0 dBFS. The Adaptive Limiter
may slightly color the sound. An effect most similar to an amplifier when driven hard.

Chapter 4 Dynamic

Following the Adaptive Limiter process, tracks normalized to 0 dB appear to sound
about 2 dB louder, depending on the source signal. As with other Logic dynamic
processes, this plug-in also features a lookahead facility (the Lookahead parameter can
be set in Controls view, allowing the Adaptive Limiter to look into the future. The
Adaptive Limiter reacts to level peaks in signals streaming from the hard disk before
they are played back, and delaying the monitored signal. The typical use of the
Adaptive Limiter is in the summed mix. It is placed after the Multipressor and before
Gain, in order to produce a CD of maximum loudness. As the Adaptive Limiter
compresses signals, it can produce results which sound louder than those resulting
from normalizing in the Sample Editor.
Start the process by adjusting the Input Scale, just as you would set a mixing desk’s
Gain parameter, or a digital recorder’s recording level. The parameter behaves much
like a Gain control, but its purpose is to adjust the input level, which must never exceed
0 dBFS. Adjust the Gain parameter to musically control the internal process of peak
smoothing and gain increase.
Out Ceiling reduces the output level of the process in very fine steps within a range of
only 2 dB. This is no threshold control, just a simple output gain.
The Mode menu in the Adaptive Limiter’s Controls view allows you to choose between
two different forms of peak smoothing. If you choose:
• OptFit, the signal will be limited by following a linear curve. This form of peak
smoothing allows signal peaks that exceed 0 dB.
• NoOver ensures that the signal never surpasses 0 dB, avoiding distortion artefacts.
Remove DC (Controls view) activates a highpass filter that removes direct current from
the signal. When using poorly constructed audio hardware, direct current (DC) can be
undesirably layered over the audio signal.
The Margin display shows the maximum measured level. To reset, click the Over lamps.

Chapter 4 Dynamic

Multipressor

The Multipressor (an abbreviation for multiband compressor) is the epitome of an
audio mastering tool. It’s a pretty complex tool; good sounding settings require quite a
lot of listening experience.

Functional Principle of Multi-band Compressors
The multi-band compressor splits the incoming signal into two to four different
frequency bands before applying compression. These frequency bands are then
compressed independently. After compression, the frequency bands are mixed back
together.
The aim of independent compression of different frequency bands is to reach high
compression levels on the bands that need it, without the pumping effect that is
normally heard at high compression levels.
Much higher Ratios, and therefore, a much higher average volume is possible before
the unwanted artefacts of compression will be heard.
Downward Expansion
Strong multi-band compression allows you to raise the overall volume level—resulting
in a dramatic increase of the existing noise floor. Downward expansion allows you to
reduce or suppress this noise. Each frequency band features a downward expander.
This works as the exact counterpart to the compressor: while the compressor
compresses the dynamic range of the higher volume levels, the downward expander
expands the dynamic range of the lower volume levels. With downward expansion, the
signal will be reduced in level when it is lower than the defined Threshold level. The
effect can be compared to a noise gate, but rather than simply cutting off the sound, it
smoothly fades the volume using an adjustable Ratio.

Chapter 4 Dynamic

Multipressor Parameters
Bands
This parameter (on the right side) determines the number of independently
compressible frequency bands, and has a crucial impact on the amount of computing
power needed for the effect. Classic multi-band compressors use three Bands.
Lookahead
Lookahead (just below the Bands parameter) determines how far the processor looks
into the future, in order to react earlier (thus better) to peak volumes.
Set Lookahead to higher values, when the Peak/RMS control (see below) is set further
towards RMS.
Peak/RMS
Adjusting the control between Peak (full left) and RMS (Root Meantime Square; full
right) is dependent on the type of signal you would like to compress. An extremely
short Peak detection setting is suitable for compression of short and high peaks of low
power, which do not typically occur in music. The RMS detection method measures the
power of the audio material over time and thus works much more musically. This is
because human hearing is more responsive to the overall power of the signal than to
single peaks. As a basic setting for most applications, the centered position is
recommended.
Attack
Allows you to define the time (in milliseconds) required before compression is faded in.
Release
Sets the time required by the compressor (after compression) to release the effect. Just
as with the other values, the best setting for this parameter depends greatly on the
material to be compressed.
Multi-band Graphic
The graphic editor to the left of the Multipressor displays several settings, both
graphically and numerically.
• Crossover Frequencies
The crossover frequencies (vertical borders) between the bands are variable. To
change a band’s crossover point, grab the borders directly within the graphic and
move them to the left or to the right. The frequency is displayed numerically at the
bottom of the graphic.
• Absolute Volume
The horizontal line in the middle of each band displays its current level
(default: 0 dB). By grabbing the area below this line and moving it up and down, you
can set the absolute volume level of the corresponding band. The level is displayed
numerically at the bottom of the graphic. This ability allows the Multipressor to serve
as a basic equalization tool, dependent on how the crossover frequencies are set.

Chapter 4 Dynamic

• Threshold Display

The horizontal lines (up to three) in the lower area of the window represent the
Threshold values for Compression (upper line), Expansion (middle line), and
Reduction (bottom line). You can set these values by using the controls of the same
name (see below).
Note: You can select the frequency band that you wish to edit by clicking in the lower
section of each band.
Comp. Ratio
This, in conjunction with Compression Threshold, is the central parameter for
compression. The Comp. Ratio determines the strength or rate of level reduction in the
range you want to compress. In most cases, the most useful combinations of these two
settings are either 1) low Compression Threshold and low Comp. Ratio or 2) high
Compression Threshold and high Comp. Ratio.
Exp. Ratio
This, in conjunction with Expansion Threshold, is the central parameter for controlling
downward expansion. It determines the strength of expansion applicable to the range
that you wish to expand.
Graphic Curve
The graphic curve in the middle of the Multipressor shows the Ratio between input
level (horizontal scale) and output level (vertical scale) of all bands. The colors
correspond to the colors of the frequency bands in the left graphic area. Adjustments
to the Ratio and Threshold controls allow you to change the curvature of the selected
frequency band.
Thresholds
• Compression
Here, you can set the minimum level at which the compressor will begin to work. If
the control is set all the way to the right (0 dB), the entire compressor section of the
Multipressor is off-line. The further you move the control to the left, the lower the
level above which the compressor will work.
• Expansion
Here, you can set the maximum level at which the expander should work. If the
control is set all the way to the left (−50 dB), expansion will only occur on signals that
fall below this level. (The Exp. Ratio can be set to a minimum of 1,2:1; below −50 dB
the expansion always takes place at this low ratio.) The further the control is moved
to the right, the higher the level below which the expander will work.
Reduction
Allows you to define the amount of noise level reduction (this is not a threshold value).
If you move the control all the way to the left, the reduction will be at its maximum
value (−50 dB). If the control is set all the way to the right, no reduction will occur.

Chapter 4 Dynamic

Level Meter
In the Level Meter to the right, you may monitor either; the change of level caused by
compression, or the output volume of each band, depending on whether you have
selected Gain change or Output (see below). You can individually switch the bands on
and off, in order to listen to single bands, by using the switches below the meters. If the
switch below a band’s meter is lit (light green), the band will be audible. If a switch is
unlit, the band is muted.
Gain Change/Output
The Gain change and Output buttons can be used to switch the operating mode of the
meters.
• If Gain change is selected, the meters indicate the strength of level reduction (to the
audio material) by the compressor.
• If Output is selected, the meters show the absolute output level of the corresponding
frequency band.
Master Gain
Allows you to reduce the overall gain increase (or increase the overall gain reduction),
resulting from the Multipressor’s settings.

Chapter 4 Dynamic

Distortion

This chapter introduces you to Logic’s distortion effects.
This includes the Distortion, Overdrive, Bitcrusher, Clip Distortion, Phase Distortion,
Distortion II, and Guitar Amp effect plug-ins.

Guitar Amp Pro

The Guitar Amp Pro plug-in simulates the sound of several famous guitar amplifiers
and a number of cabinets/speakers. You can process guitar signals directly within Logic,
allowing you to reproduce the sound of high-quality guitar amplification systems.
Guitar Amp Pro can also be used for experimental sound design and processing. You
can freely use the plug-in on other instruments, as desired—applying the sonic
character of a guitar amp to a trumpet or vocal part, for example!
Guitar Amp Pro offers a range of Amplifier, Speaker, and EQ models that can be
combined in a number of ways. The EQ models are equipped with the Bass, Mid, and
Treble controls typical of guitar amplifiers. Miking can be switched between two
different microphone types and positions. To round out the complement of controls,
Guitar Amp Pro also integrates two classic guitar effects, namely Vibrato and Tremolo.

The Guitar Amp Pro panel is divided into three areas. The upper part of the raised Yshaped user interface contains the Amplifier parameters section. The bottom of this
panel houses the Effect and Output section of the plug-in. The Speaker sections to the
left and right provide access to the miking parameters of the virtual speaker.

Amp Section
In the upper area of the Amp section, you will find three pull-down menus. These allow
you to set up your guitar amp as required, by selecting the appropriate Amp, Speaker,
and EQ model(s).
Eleven different amplifier models can be accessed via the Amp menu to the left.
• UK Combo 30W—neutral sounding Amp model, well suited for clean or crunchy
rhythm parts.
• UK Top 50W—quite aggressive in the high frequency range, well suited for classical
rock sounds.
• US Combo 40W—clean sounding Amp model, well suited for funk and jazz sounds.
• US Hot Combo 40W—emphasizes the high mids of the frequency range, making this
model ideal for solo sounds.
• US Hot Top 100W—this Amp model creates very fat sounds, even low Master settings
result in broad sounds with a lot of “oomph”.
• Custom 50W—with the Presence parameter set to 0, this Amp model is well suited for
smooth fusion lead sounds.
• British Clean—simulates the classic British Class A combos which have been
continuously produced since the 1960s to the present, without any significant
modification. This model is ideally suited for clean or crunchy rhythm parts.
• British Gain—reproduces the sound of a British tube head, and is synonymous with
rocking, powerful rhythm parts and lead guitars with a rich sustain.
• American Clean—emulates the traditional full tube combos used for clean and
crunchy sounds.
• American Gain—emulates a modern Hi-Gain head, making it suitable for distorted
rhythm and lead parts.
• Clean Tube Amp—emulates a tube amp model with very low gain (distortion only
when using very high input levels or Gain/Master settings).
The Speaker menu at the top right provides access to 15 speaker models.
• UK 1×12 open back—Classic open enclosure with one 12" speaker, neutral, wellbalanced, multifunctional.
• UK 2×12 open back—Classic open enclosure with two 12" speaker, neutral, wellbalanced, multifunctional.
• UK 2×12 closed—Loads of resonance in the low frequency range, therefore well
suited for Combos: crunchy sounds are also possible with low Bass control settings.
• UK 4×12 closed slanted—when used in combination with off-center miking, you will
get an interesting mid frequency range; therefore this model works well when
combined with High Gain amps.

Chapter 5 Distortion

• US 1×10 open back—Not much resonance in the low frequency range. Suitable for

use with (blues) harmonicas.
• US 1×12 open back 1—Open enclosure of an American lead combo with a single 12"

speaker.
• US 1×12 open back 2—Open enclosure of an American clean/crunch combo with a

single 12" speaker.
• US 1×12 open back 3—Open enclosure of another American clean/crunch combo

with a single 12" speaker.
• US broad range—Cabinet simulation of a classic electric piano speaker.
• Analog simulation—Internal speaker simulation of a well-known British 19" tube

preamplifier.
• UK 1×12—A British Class A tube open back with a single 12" speaker.
• UK 4×12—Classic closed enclosure with four 12" speakers (black series), suitable for

Rock.
• US 1×12 open back—Open enclosure of an American lead combo with a single 12"

speaker.
• US 1×12 bass reflex—Closed bass reflex cabinet with a single 12" speaker.
• DI Box—This option allows you to bypass the speaker simulation section.

The EQ models in the EQ menu refer to the simulated Amp models. Accordingly, the
British 1, British 2, American, and Modern EQ models are available for the British Clean,
British Gain, American Clean, and American Gain Amp models. You can, however,
combine any Amp model with any EQ model, as required.
Directly below the EQ menu, you will find the Bass, Mid, and Treble controls. Use of
these knobs allows you to adjust the frequency ranges of the EQ models as desired.
Presence is an additional high frequency control which exclusively affects the output
stage (Master) of the Guitar Amp Pro plug-in.
The Link buttons between the menus link your menu selections. To explain, if the Link
button between the EQ and Amp menus is active (yellow), selecting an Amp model will
automatically load the corresponding EQ model. As mentioned earlier, you can,
however, assign any other EQ model to the selected amp—via the EQ menu.
If the Link button between the Speaker and Amp menus is active, selecting an Amp
model will automatically load the corresponding Speaker model. The cabinet
assignment can be changed via the Speaker menu, as required.
To the far left of the Amp section, you will find the Gain knob, which controls the preamplification of the input signal. This control has different effects, dependent on the
selected Amp model. As an example: A maximum Gain setting produces a powerful
crunch sound when used in conjunction with the British Clean Amp model, but the
same Gain setting results in a heavy distortion—suitable for lead sounds—with the
British Gain or Modern Gain Amp models.

Chapter 5 Distortion

To the extreme right of the Guitar Amp Pro GUI, you will find the Master knob, which
controls the output volume of the amplifier (to the speaker). Typically, in tube
amplifiers, an increase in the Master control level produces a self-compressed and
saturated sound, along with increased level, resulting in a more distorted and powerful
amp signal. In the analog domain, this results in an extreme increase in loudness. In
Guitar Amp, the Master control influences the sonic character, with the output level
being set with the Output parameter (see below).

Speaker Section
Following the selection of a Speaker type, you make further adjustments to the miking
parameters in the Speaker section.
The Centered and Off-Center buttons allow you to switch between two different
microphone positions.
Centered aligns the microphone to the center of the speaker cone. This position is also
called On Axis because the microphone capsule is approximately on the same axis
(aligned) with the center of the speaker. In this position, the speaker sounds more full
and powerful, making this setting suitable for Blues or Jazz guitar tones.
Off-Center aligns the microphone to the edge of the diaphragm. This placement is
called Close Edge or Off Axis. The end result is an amplifier signal that is much brighter
and sharper, but a little thinner. This position is more suitable for cutting rock or typical
rhythm ’n’ blues guitar tones.
The microphone Type, in conjunction with microphone placement, is equally essential
for designing the required speaker sound.
When the Condenser button is active, a studio condenser microphone emulation is
used for miking. The sound of condenser microphones is fine, transparent, and well
balanced.
The Dynamic button switches to a dynamic cardioid microphone emulation. This
microphone type sounds brighter and more cutting, in comparison to the Condenser
model. At the same time, the lower Mids become less distinctive, making the Dynamic
model more suitable for miking rock guitar tones.
Note: In practice, combining both microphone types can sound very interesting.
Duplicate the guitar track, and insert Guitar Amp Pro as an insert effect on both tracks.
Select different microphones in both Guitar Amp Pro instances, while retaining
identical settings for all other parameters, and mix the track signal levels. You can, of
course, choose to vary any other parameters, as desired.

Chapter 5 Distortion

Effect Section
The Effect section of Guitar Amp Pro contains the Tremolo and Vibrato effects—
essentials in any guitar rig, and the Reverb.
Note: The Effect section is placed before the Master control in the signal flow, and
therefore receives the preamplified signal (pre-Master).
In order to configure the Effect section, you must activate it via the On/Off buttons,
found to the lower left of the FX and Reverb panels. When the respective effect section
is active, the border of the On/Off button is highlighted.
In the upper middle portion of the FX section, you’ll find the Effect menu, which allows
you to select between the Tremolo and Vibrato effects. Tremolo modulates the
amplitude (and therefore the volume), while Vibrato modulates the frequency (and
therefore the pitch) of the signal. The intensity of the modulation is determined by the
Depth parameter. Speed controls the modulation speed in Hz. Lower settings will
produce a smooth and floating sound, with higher settings leading to a rotor-like
effect. You can perfectly synchronize the modulation speed to the song tempo, if
desired. To do so, simply press the Sync button, found beside the Speed control. Once
synchronization mode is activated, the control range of the Speed control will display
various musical values. Set the Speed control to the desired value, and your Guitar Amp
Pro modulation will be perfectly synchronized to the song tempo.
The Reverb portion of the Effect section contains two controls. Level determines the
amount of reverb signal applied to the pre-amplified signal. The pull-down menu to
the right allows you to select one of three different Spring reverb models.

Output
The Output knob serves as a final level control for Guitar Amp’s output.
The Output parameter can be viewed as a volume control “behind the cabinet”, and is
used to set the level that is fed into ensuing plug-in slots on the channel, or into the
channel output.
Note: This parameter is very distinct from the Master control, which serves a dual
purpose—for sound design, as well as controlling the level of the Amp section.

Chapter 5 Distortion

Distortion

This distortion effect simulates the lo-fi dirt generated by a bipolar transistor.
Move the Drive slider up to increasingly saturate the transistor. Generally, the distortion
created by the plug-in tends to increase the signal level, an effect that you can
compensate for with the Output slider. The Tone knob filters the harmonics-laden
distortion signal, delivering a somewhat less grating, softer tone.
The Distortion Eye is watching—it visually represents the Drive and Tone parameter
settings.

Overdrive

The Overdrive effect emulates the distortion of a field-effect transistor (FET). When
saturated, FETs generate warmer sounding distortion than bipolar transistors.
The Drive slider pushes the transistor over the edge and into overdrive. Generally, the
distortion created by the plug-in tends to increase signal levels, an effect that you can
compensate for with the Output slider.
The Tone knob lets you filter the harmonics-laden distortion signal, which delivers an
even warmer sound.
The Distortion effect’s Eye visually represents the settings of the Drive and Tone
parameters.

Chapter 5 Distortion

Bitcrusher

Bitcrusher is the ultimate digital distortion box. You can do all kinds of wild stuff with it,
such as recreate the 8-bit sound of the pioneering days of digital audio, create artificial
aliasing by dividing the sample rate, or distort signals so radically that they are
rendered unrecognizable.
Warning: The Bitcrusher can damage your hearing (and speakers) when operated at
high volumes.
The Drive slider boosts the level at the input of the Bitcrusher. Please note that this
tends to excite the clipping stage located at the output of the Bitcrusher as well.
The Resolution knob allows you to reduce the resolution from 24 bits down to 1 bit.
The number of bits is always an exponent of two. The range of available values is
equivalent to the exponents of two that a given sample rate can handle. As an
example, 65,536 different values are possible for 16 bits, whereas at 8 bits, you’re left
with just 256. The sonic image becomes ever more ragged as the values decrease
because the number of sampling errors increases, thus generating more distortion. At
extremely low bit resolutions, the amount of distortion can be greater than the level of
the usable signal.
The Downsampling slider lowers the sample rate. As an example, at a value of 2
(halved), the original 44.1 kHz signal is sampled at a rate of just 22.05 kHz. At a factor of
10, the rate is knocked all the way down to 4.41 kHz.
The Clip Level slider lets you define the point below the normal threshold that you want
the signal to start clipping. The Mode buttons are used to determine whether the signal
peaks that exceed the clip level are Folded, Cut, or Displaced (check out the graphics
on the buttons and the resulting waveform in the display). The kind of clipping that
occurs in standard digital systems is usually closest to that of the center mode (Cut).
Internal distortion may generate clipping similar to the types generated by the other
two modes.

Chapter 5 Distortion

Clip Distortion

The Clip Distortion plug-in is a non-linear distortion effect that produces unpredictable
spectra. Beyond drastic distortions, it’s well suited for the simulation of warm tube
overdrive sounds.
The best way to learn what effect the various parameters have is to experiment with
them on different signal sources. As a starting point, the following describes what each
control basically does:
The signal is first amplified by the Drive value, which is a simple gain control. The signal
then passes through a highpass filter. The filter’s cutoff frequency is determined by the
Tone control. The actual non-linear distortion process is controlled by the Symmetry
parameter.
Once the signal has been distorted asymmetrically, the signal passes through a lowpass
filter. This filter’s cutoff frequency is determined by the Filter fader. The Mix parameter
combines the effected signal with the dry signal. This mixed signal then passes through
yet another lowpass filter, where the cutoff frequency is controlled by the Sum Filter
parameter. All filters have a slope of 6 dB/Oct.
The last stage of signal processing is a tunable shelving filter. If you set its Frequency to
about 12 kHz, it will behave like a normal treble control, as found in any mixer’s channel
strip or stereo hi-fi amplifier. Unlike such treble controls, this filter allows for boosts or
cuts of up to ±30 dB (Gain parameter). This somewhat unorthodox combination of
serially connected filters allows for gaps in the frequency spectra that can sound quite
good with this sort of non-linear distortion. The clip circuit graphic visually depicts
every parameter, with the exception of the shelving filter controls.
There are two more parameters in the Controls view: Input Gain and Output Gain.
These can be used to raise/lower the input and output signal levels by up to 30 dB.

Chapter 5 Distortion

Phase Distortion

The Phase Distortion plug-in is based on a modulated delay line, much like the wellknown chorus or flanger effects. As opposed to these, the delay time is not modulated
by a low frequency oscillator (LFO), but rather by a lowpass-filtered version of the audio
input signal itself. This is how the signal modulates its own phase position.
In the signal flow of this effect, the parameters do the following:
The input signal only passes the delay line and is not affected by any other process. Mix
blends the effected signal with the original signal. The delay time is modulated by a
Side Chain signal—namely, the input signal. The input signal passes through a resonant
lowpass filter, the Cutoff frequency and Resonance of which can be set with dedicated
controls. You also can listen to the filtered Side Chain (instead of the Mix signal), if you
engage Monitor. The maximum delay time is set with Max Modulation. The amount of
modulation itself is controlled with Intensity.
In the Controls view, there is one more parameter which can’t be seen in the Plug-in
window. It is only valid for the stereophonic version of the effect. Normally, a positive
input value results in a longer delay time. If you engage Phase Reverse (On), positive
input values result in a reduction of the delay time on the right channel only.

Chapter 5 Distortion

Distortion II

The Distortion II plug-in is based on the EVB3’s distortion effect. More information
about its parameters can be found in the EVB3’s “Distortion” section on page 473.
The Distortion II plug-in offers one additional parameter: the PreGain knob. This allows
you to raise the input signal by up to 20 dB or lower it by as much as 10 dB, in order to
provide a broader range of distortion colors.

Chapter 5 Distortion

Filter

This chapter covers Logic’s filter effects.
The filter effects include the AutoFilter, Fuzz-Wah, EVOC 20 FB, EVOC 20 TO, Low/High
Pass Filter and Low/High Cut plug-ins. The EVOC 20 TO is based on a vocoder. Further
information about vocoders can be found in the chapter “Vocoder—Basics” on
page 167.

AutoFilter

The AutoFilter is an extremely versatile, resonance-capable lowpass filter, that offers a
couple of truly unique features. The most important parameters are located to the right
side of the Plug-in window: The Cutoff Freq. knob determines the point where the filter
kicks in. Higher frequencies are attenuated, lower frequencies are allowed to pass
through.

The Resonance knob emphasizes the frequency range surrounding the cutoff
frequency. When you turn the Resonance up sufficiently, the filter itself begins
oscillating (at the cutoff frequency). Self-oscillation is initiated before you max out the
Resonance parameter, just like the filters on the legendary Minimoog. When working
with Resonance, the manner in which the lowpass filter allows frequencies to pass
changes: higher Resonance values cause the filter to cut the bottom end, making the
signal sound thinner. The Fatness parameter compensates for this audio artefact. When
you turn Fatness up to its maximum value, the Resonance setting has no effect on the
response of the frequencies below the cutoff frequency.
The Slope buttons determine the steepness of the lowpass filter: frequencies above the
cutoff frequency are dampened by 6, 12, 18, or 24 dB per octave (in audio jargon, these
are called filters of the 1st, 2nd, 3rd, and 4th order). Even though the 24 dB filter is
largely the component of choice for synthesizer designers, be sure to experiment with
the other options, as they can also deliver pretty hip results. The Distortion Input and
Output parameters allow you to individually control each of the two distortion units—
one pre-input and the other post-output. Although the two distortion modules are
identical, their respective positions in the signal chain—before and after the filter,
respectively—enable them to generate remarkably different sounds.
All other AutoFilter parameters are used to dynamically modulate the cutoff frequency.
These fall into two sections: Envelope (ADSR, Envelope Generator) and LFO (Low
Frequency Oscillator, Modulation Generator).
The Threshold parameter applies to both sections, and analyzes the level of the input
signal. If the input signal level exceeds that of the variable Threshold level, the
envelope and LFO are retriggered. The Modulation slider of each section determines
the intensity of the control signal’s effect on the cutoff frequency.
Envelope: when the Threshold level is exceeded, the control signal is triggered at the
minimum value. Following a variable interval, the length of which is determined by the
Attack parameter, the signal reaches its maximum value. It drops in level during the
interval defined by the Decay value, and ends up at the Sustain value. Once the signal
level drops below the Threshold value, it falls all the way to its minimum value over the
time determined by the Release parameter. If the input signal falls below the Threshold
level before the control signal has reached the Sustain level, the Release phase is
triggered. The Dynamic Modulation parameter lets you modulate the peak value of the
Envelope section, by using the level of the input signal.

Chapter 6 Filter

LFO: the wave shape used for LFO oscillation is determined by the Waveform buttons.
The choices are: descending sawtooth (saw down), ascending sawtooth (saw up),
triangle, pulse wave, or random (random values, Sample & Hold). Once you’ve selected
a waveform, you can shape the curve with the Pulsewidth knob. Use the Frequency
knobs to define the desired LFO frequency: Coarse sets a value between 0.1 and
10,000 Hz, Fine lets you adjust it in smaller increments. The Speed Mod. (Speed
Modulation) knob is used to modulate the LFO frequency independently of the input
signal level. If the input signal exceeds the Threshold level, the modulation width of the
LFO increases from 0 to the value specified for Modulation. You can also define the
amount of time this process takes, by entering the desired value with the Delay knob. If
the Sync button is activated, the waveform is started at 0° as soon as the Threshold is
exceeded.
Whenever you use the AutoFilter as a stereo plug-in, you can determine the phase
relationships of the LFO modulations on the two stereo sides, with the Stereo Phase
knob.
There are five additional parameters in the Controls view of the Autofilter.
The Volume parameter can lower the Volume by as much as −50 dB, allowing you to
compensate for higher levels when using Distortion, for example. If you switch Beat
Sync to On, the LFO is synchronized to the sequencer’s tempo. The speed values
include bar values, triplet values and more. These are determined by the Rate slider
directly below Beat Sync. Use Sync Phase to shift the phase relationship between the
LFO and the sequencer. Dry Signal sets the level ratio/portion of the non-effected (dry)
signal.

Chapter 6 Filter

Fuzz-Wah

The Fuzz-Wah effect is the standalone plug-in version of the EVD6’s Wah effect. It
incorporates additional compressor and distortion (Fuzz) facilities, and features some
additional parameters over the integrated EVD6 Wah. These are outlined below.

Parameters of the Fuzz-Wah
FX Order

This parameter allows to you select the order in which the Fuzz/Wah effects are placed.
Choices are: Fuzz –Wah or Wah–Fuzz.
Wah Mode
There are simulations of several classic wah effects, as well as some basic filter types
available. Available models are: off, ResoLP, ResoHP, Peak, CryB, Morl1, Morl2.

Wah Level
Can be used to adjust the level of the wah-filtered signal, relative to the original level.
Also see the Auto Gain section below.
Auto Gain
While sweeping through the main formants of the input signal, the output level of the
Wah may vary wildly, which is not always desirable. Activating the Auto Gain parameter
will automatically compensate for this side-effect. Range: on/off

To hear the difference Auto Gain can make:
• Switch Auto Gain to on.
• Raise the effect level to a value just below the mixer’s clipping limit.
• Make a sweep with a high relative Q setting.
• Now switch Auto Gain to off, and repeat the sweep.

Chapter 6 Filter

Warning: Please take care while doing this, or your ears and speaker system may be
damaged.
Relative Q
The quality of the main filter peak can be increased/decreased, relative to the model
setting, thereby obtaining a sharper/softer wah sweep. When set to a value of 0, the
original setting of the model is active. Range: −1.00 to +1.00 (0.00 is the default)

Pedal Range
Common MIDI foot pedals have a much larger mechanical range than most classic Wah
pedals.

The exact sweep range of the wah filter effected by the MIDI foot pedal is set with the
Pedal Range parameters. The highest and lowest possible value reached by the pedal is
graphically represented by a gray bracket around the Pedal Position fader (see below).
The left and right limit is set by clicking and moving it with the mouse. Additionally
both values can be moved simultaneously by clicking in the center of the bracket and
moving it to the left or right.
Pedal Position
This parameter represents the current position of the Wah pedal.
To control and automate the Pedal Position via an external MIDI controller for example
a MIDI pedal, your Logic environment has to be prepared accordingly. For more
information please read “Controlling the EVD6 via MIDI” on page 502.
AutoWah Depth
In addition to using MIDI foot pedals (see above), the wah effect can be controlled
using the Auto Wah facility. The sensitivity of the Auto Wah can be set with the Depth
parameter. Range: 0.00 to 100. (See also the “Envelope (Depth)” section, from page 500
onwards.)

Chapter 6 Filter

AutoWah Attack/Release
These parameters allow you to define how much time it takes for the Wah filter to open
and close. Range (in milliseconds): 10 to 10,000

Comp Ratio
The Comp Ratio of the integrated compressor can be adjusted between 1:1 (no
compression) and 30:1. The Compressor is tied to the Fuzz effect, and always precedes
it. As such, the FX Order parameter is very important for placement of the Compressor
in the effects chain.

Fuzz Gain
Controls the level of Fuzz (distortion). Range: 0 dB to 20 dB.
Fuzz Tone
The integrated Fuzz effect can be adjusted, tonally, with this parameter. Range: 2000 Hz
to 20,000 Hz

EVOC 20 Filterbank
The EVOC 20 FB consists of two formant filter banks, which are also used in Logic’s
EVOC 20 PS vocoder. More information on the filter banks are found in “How Does a
Filter Bank Work?” on page 168.
The input signal runs through both filter banks in parallel. Each bank features
independent volume faders for each band, allowing levels to be set freely—ranging
from unchanged through to silence. The latter completely suppresses the selected
formants in the overall sound spectrum. Use of the Formant Stretch and Formant Shift
parameters provide total control over the position and width of the filter bands. In
addition, you can also crossfade between the two filter banks.

Chapter 6 Filter

Parameters of the EVOC 20 FB

The EVOC 20 FB interface is divided into three main sections. These are the Formant
Filter, Modulation, and Output areas.
The Formant Filter Area
The Formant Filter Window
The Formant Filter window is divided into two sections by a horizontal line. The upper
half applies to the Filter Bank A, and the lower half to the Filter Bank B.

The individual vertical bars in each bank of settings are faders which represent the level
of a particular frequency band/formant. To adjust each fader, simply click-hold on the
desired bar and drag up or down.
Complex bar curves are easily created by “painting” them in: Click and hold the mouse
button next to a bar on the blue or green portion of the background, and drag left or
right over the bars within the editing field. The length of the bars will be adjusted in
accordance with the mouse movement. This method makes editing multiple frequency
band levels quick and convenient.
Bands
The Bands parameter determines the number of frequency bands used by the
EVOC 20 FB. It ranges from 5 to 20.

Chapter 6 Filter

Note: Increasing the number of bands also increases the processor overhead.
High/Low Frequency
The blue bar shown just beneath the EVOC label is a multi-part control which is used to
determine the lowest and highest frequencies allowed to pass by the filter. Frequencies
which fall outside these boundaries will be cut. All filter bands are distributed evenly
across the range defined by the High/Low Frequency values.
• To adjust the Low Frequency value, simply click-hold on the silver slider to the left of
the blue bar, and drag to the right (or left). The value range is 75–750 Hz.
• To adjust the High Frequency value, simply click-hold on the silver slider to the right
of the blue bar, and drag to the left (or right). The value range is 800–8000 Hz.
• To adjust both sliders simultaneously, click on the area between the slider halves
(directly on the blue bar) and drag to the left or right.
• You can make changes to the High/Low Frequency values directly by using your
mouse as a slider on the numerical entries—80 and 8000 Hz in the diagram.
Lowest/Highest
These parameters can be found in the two small switches on either side of the Formant
Filter window. These switches determine whether the lowest and the highest filter
bands are bandpass filters (just like all the bands between them), or whether they act
as lowpass/highpass filters, respectively. Click once on them to switch between the two
curve shapes available.
• In the Bandpass setting, the frequencies below/above the lowest/highest bands are
ignored.
• In the Highpass or Lowpass setting, all frequencies below the lowest (or above the
highest) bands will also be treated.

Slope
The pull-down menu Slope determines the amount of filter slope applied to all filters of
both filter banks. Choices are 1 (filter attenuation of 6 dB/Oct.) and 2 (filter attenuation
of 12 dB/Oct.): 1 sounds softer, 2 sounds tighter.
Boost A/B Controls
The Boost A and Boost B knobs allow an increase or cut in the overall gain of the A and
B filter banks. Their range is ±20 dB. To adjust, click-hold and drag up or down with the
mouse.
Note: You will need these controls, as the filter bank achieves its sounds by turning
down the level of one or more filter bands. To make up for the resulting energy loss,
use Boost.

Chapter 6 Filter

Note: Boost is also quite handy to adjust the levels of both filter banks to each other, so
that using Fade A/B (see below) leads only to a sound color change, but not to a level
change.
Fade AB Control
The Fade AB crossfades between the A and B filter bank. At its extreme top or bottom
position, you will only hear one of the filter banks.
Formant Shift

Moves the position of all bands in both filter banks up and down the frequency range.
Note: You can jump directly to the values −0.5, −1, 0, +0.5 and +1.0 by clicking on their
numbers.
Resonance

Resonance is responsible for the basic sonic character of both filter banks: low settings
give it a soft character, high settings will lead to a more snarling, sharp character.
Increasing the value for Resonance emphasizes the middle frequency of each frequency
band.
Modulation Parameters

The Modulation (LFO) area controls the Formant Shift and Fade A/B parameters of the
EVOC 20 FB. It allows synchronous/non-synchronous modulation in bar, beat (triplet) or
free values.
• LFO Shift, on the left-hand side, controls Formant Shift modulation of the filter bands.
• LFO Fade controls the Fade AB parameter.

Chapter 6 Filter

Waveform
The Waveform switches allow the selection of the waveform type used by LFO Shift and
LFO Fade. A selection of Triangle, falling and rising Sawtooth, Square up and down
around zero (bipolar), Square up from zero (unipolar), a random stepped waveform
(S&H), and a smoothed random waveform is available for each LFO.

Intensity
The Intensity sliders control the amount of Formant Shift and Fade A/B modulation by
the respective LFOs.
Rate
These knobs determine the speed of the modulation. Values to the left of the center
positions are synchronized with the sequencer’s tempo and include bar values, triplet
values and more. Values to the right of the center positions are non-synchronous and
displayed in Hertz.
Note: The Formant Shift and Fade LFO modulations are the keys to the most
extraordinary sounds of the EVOC 20 FB: Make sure to set up totally different or
complementary filter curves in both filter banks. Use rhythmic material like a drumloop
as an input signal. Set up tempo-synchronized modulations—with different Rates for
each LFO—for Formant Shift and Fade A/B. And then try a tempo-synchronized Tape
Delay after the EVOC 20 FB. You will end up with unique rhythms.

Chapter 6 Filter

Output Section

Overdrive
This switch enables/disables the Overdrive circuit of the EVOC 20 FB.
Note: To actually hear the Overdrive effect, you may need to boost the level of one or
both filter banks.
Level
The Level slider controls the level of the EVOC 20 FB’s output signal.
Stereo Mode
This pull-down menu determines the input/output mode of the EVOC 20 FB. Choices
are: m/s—mono input to stereo output and s/s—stereo input to stereo output.
• The Stereo Mode should be set to m/s if the signal going into the EVOC 20 FB is
monophonic, for example a mono audio track.
• Stereo/stereo (s/s) is the preferred setting for stereo input signals. In this case, the
stereo signal is processed by separate filter banks for the left and right channels.
When using the m/s Mode on stereo input signals, the stereo signal is first summed
to mono before it is passed on to the filter banks.
Stereo Width
Stereo Width distributes the output signals of the filter bands in the stereo field.
• In the left position, the output of all bands are centered.
• In the center position, the output of all bands ascends from left to right.
• In the right position, the bands are output evenly on the left and the right channel.
The stereo/stereo mode (s/s) uses one A/B filter bank per channel. The positioning of
the frequency bands correspond to that described above, but the bands of each filter
bank ascend in opposing directions, from left to right.

Chapter 6 Filter

MIDI Controllers Received
The following tables show the CC numbers used when the following MIDI preference is
active: Options > Settings > MIDI Options > (Version 4.x behavior).
Filter Bands

Boost

Band Level

Bank A

Bank B

Band 1

CC #64

CC #96

Band 2

CC #65

CC #97

Band 3

CC #66

CC #98

Band 4

CC #67

CC #99

Band 5

CC #68

CC #100

Band 6

CC #69

CC #101

Band 7

CC #70

CC #102

Band 8

CC #71

CC #103

Band 9

CC #72

CC #104

Band 10

CC #73

CC #105

Band 11

CC #74

CC #106

Band 12

CC #75

CC #107

Band 13

CC #76

CC #108

Band 14

CC #77

CC #109

Band 15

CC #78

CC #110

Band 16

CC #79

CC #111

Band 17

CC #80

CC #112

Band 18

CC #81

CC #113

Band 19

CC #82

CC #114

Band 20

CC #83

CC #115

Bank A

CC #84

Bank B

CC #116

Bands

CC #85

FF Low Freq

CC #88

FF Hi Freq

CC #89

Fade A/B
Formant Filter

Chapter 6 Filter

CC #117

Formant Shift

CC #90

FF Resonance

CC #91

Slope

CC #92

FF Low/Bandpass Select

CC #119

FF High/Bandpass Select

CC #120

Filter Bands

Band Level

Bank A

LFO 1 (Shift)

LFO 1 Rate

CC #93

LFO 1 Waveform Select

CC #94

LFO 1 Intensity

CC #95

LFO 2 Rate

CC #121

LFO 2 Waveform Select

CC #122

LFO 2 Intensity

CC #123

Level

CC #124

LFO 2 (Fade)

Output

Mono/Stereo Select

CC #87

Stereo Width

CC #86

Bank B

EVOC 20 TO
The EVOC 20 TO is a vocoder equipped with a monophonic pitch tracking oscillator,
hence the TO in its name (more information about vocoders can be found in the
chapter “Vocoder—Basics” on page 167). Non-technically, this allows the EVOC 20 TO to
follow (track) the pitch of a monophonic incoming audio signal. If the audio signal is a
vocal melody, for example, the individual pitches of the sung notes will be tracked and
mirrored by the Synthesis engine.
The EVOC 20 TO derives its synthesis signal from its monophonic tracking oscillator.
Alternatively to the tracking oscillator, the EVOC 20 TO can use a freely selectable audio
signal as the synthesis signal.
Note: For good pitch tracking, it is essential that the signal is monophonic (one pitch at
a time) and as dry as possible. Absolutely avoid background noises. As an example,
using a voice already processed with even a slight reverb will deliver pretty strange
results. The results will be even stranger when signals with no audible pitch are used—
such as drumloops. The resulting artefacts might, however, be exactly what you’re after
in some situations.
It should be noted that the EVOC 20 TO uses a Side Chain, allowing the use of another
track as the analysis and/or synthesis signal. In the gray area at the top of the Plug-in
window, click-hold on the Side Chain pull-down menu, and select the desired Audio
track. In the Mixer, adjust the volume levels of the EVOC 20 TO and the audio track used
for the Side Chain to taste.
The EVOC 20 TO is not limited to pitch tracking effects. It can vocode a signal by itself,
making it very useful for unusual filter effects. Try this with different Resonance, Formant
Shift and Formant Stretch settings.
As both analysis and synthesis input signals are freely selectable, you can even vocode
an orchestra with train noises, for example.

Chapter 6 Filter

The EVOC 20 TO can be used in the insert slots of Audio, Audio Input, Bus, Master, and
Audio Instrument channels.
The signal path of the EVOC 20 TO is shown in the block diagram on page 190.

Parameters of the EVOC 20 TO

The EVOC 20 TO interface is divided into five main sections. From left to right, these are
the Analysis/Synthesis, Formant/Filter, Modulation, Unvoiced/Voiced (U/V) Detection and
Output areas.
Analysis In Parameters
Attack
The Attack parameter determines how quickly each envelope follower coupled to each
Analysis filter band reacts to rising signals. Longer Attack times result in a slower
tracking response to transients of the Analysis input signal.
Note: A long Attack time on percussive input signals (a spoken word or hi-hat part, for
example) will translate into a less articulated vocoder effect. Set Attack as low as
possible to achieve precise articulation.
Release
The Release parameter determines how quickly each envelope follower coupled to each
Analysis filter band reacts to falling signals. Longer Release times make transients of the
Analysis input signal sound for a longer period of time at the Vocoder’s output.

Note: A long Release time on percussive input signals (a spoken word or hi-hat part, for
example) will translate into a less articulated vocoder effect. Release times that are too
short result in rough, grainy vocoder sounds. Release values of around 8 to 10 ms have
proven to be useful starting points.

Chapter 6 Filter

Freeze
The Freeze button holds the current analysis sound spectrum indefinitely.
The “frozen” Analysis signal can capture a particular characteristic of the source signal,
which is then imposed as a complex sustained filter shape on the Synthesis section.
Using a spoken word pattern as a source, for example, the Freeze parameter could
capture the attack or tail phase of an individual word within the pattern—the vowel a,
for example.
With Freeze engaged, the Analysis filter bank ignores the input source until it is
disengaged.
Another use of the Freeze parameter (which can be automated) could be to
compensate for people’s inability to sustain sung notes for a long period without
taking a breath. If the Synthesis signal needs to be sustained, when the Analysis source
signal (a vocal part) isn’t, Freeze can be used to lock the current formant levels (of a
sung note), even during gaps in the vocal part—between words in a vocal phrase.
Note: When the Freeze parameter is used, the Attack and Release parameters have no
effect.
Analysis In (pull-down menu)
This pull-down menu determines the Analysis signal source—Track or Side Chain. To
switch between them, use the mouse as a slider and drag vertically.

• Track—allows you to use the audio track, into which the EVOC 20 TO is inserted, as

the Analysis signal.
• Side Chain—allows you to use another audio track as the Analysis signal. The

selection of the actual Side Chain source track is achieved by click-holding on the
Side Chain pull-down menu in the gray area at the top of the Plug-in window.
Note: If no Side Chain track is assigned in the Side Chain pull-down menu, the track’s
signal will be used.
Synthesis In Parameter
Synthesis In (pull-down menu)
This pull-down menu determines the Synthesis signal source—Osc(illator), Track or Side
Chain. To switch between them, use the mouse as a slider and drag vertically.
• Oscillator—allows you to use the built-in monophonic tracking oscillator. The
oscillator tracks the pitch of the Analysis input signal. Selection of the Oscillator
activates the other parameters in the Synthesis section. If Osc is not selected, the FM
Ratio, FM Int and other parameters in this area have no effect.

Chapter 6 Filter

• Track—allows you to use the audio track, into which the EVOC 20 TO is inserted, as

the Synthesis source signal.
• Side Chain—allows you to use another audio track as the source material for the

Synthesis section. Selection of the Side Chain track is achieved by click-holding on the
Side Chain pull-down menu in the gray area at the top of the Plug-in window.

Note: If no Side Chain track is assigned in the Side Chain pull-down menu, the track’s
signal will be used.
Bands
The Bands window determines the number of frequency bands used by the
EVOC 20 TO. It ranges from 5 to 20. Adjustments are made by using the mouse as a
slider. The greater the number of bands, the more precisely the sound can be reshaped.

Note: Increasing the number of bands also increases the processor overhead.
The Tone Generator of the Tracking Oscillator
Depending on the position of the FM Int control, the tracking oscillators delivers either;
a sawtooth wave or the signal of an FM tone generator.
The FM tone generator consists of two oscillators, each generating a sine wave. The
frequency of Oscillator 1 is linearly modulated by Oscillator 2. This deforms the sine
wave of Oscillator 1 to a waveform with rich harmonic structure. Its harmonic structure
depends on the modulation intensity and the frequency ratio of both oscillators.
Tune
Coarse Tune offsets the pitch of the oscillator in semitones by up to ±2 octaves.

Fine Tune: The default value is concert pitch A = 440 Hz. The range is from 425.00 to
455.00 Hz.

Chapter 6 Filter

FM Int

This knob selects the basic waveform and controls the intensity of FM modulation.
• If set to 0, the FM tone generator is disabled, and a sawtooth wave is generated
instead.
• If set to values higher than 0, the FM tone generator is activated. Higher values result
in a more complex and brighter sound.
FM Ratio
The FM Ratio (value range 0.5 to 3.5) knob defines the ratio between the Carrier and
Modulator frequencies—the frequencies of Oscillators 1 and 2. This setting defines the
basic character of the sound.
With even-numbered values or their multiples, harmonic sounds are produced. With
odd-numbered values or their multiples, inharmonic sounds are produced, which we
perceive as being metallic sounding.
• An FM Ratio of 1.000 produces results resembling a sawtooth waveform.
• An FM Ratio of 2.000 produces results resembling a square wave with a pulse width
of 50%.
• An FM Ratio of 3.000 produces results resembling a square wave with a pulse width
of 33%.
FM Ratio is only relevant if FM Int is not set to 0.
Pitch Quantize
The Pitch Quantize, Root/Scale and Max Track controls, in conjunction with the piano
keys of the onscreen keyboard, control the automatic pitch correction facility (Pitch
Quantize) of the tracking oscillator. Pitch Quantize, in conjunction with the Root/Scale
and Max Track parameters, can be used to constrain the pitch of the tracking oscillator
to a scale or chord. This allows subtle or savage pitch corrections, and can be used
creatively on unpitched material with high harmonic content, such as cymbals and
high-hats. To use pitch quantization, the Strength parameter must be set above a value
of zero, and at least one of the onscreen keyboard keys needs to be activated.

Chapter 6 Filter

• Strength—determines how pronounced the automatic pitch correction is.
• Glide—determines the amount of time the pitch correction takes, allowing sliding

transitions to the quantized pitches.
Root/Scale
The Root and Scale parameters, in combination with the onscreen keyboard, define the
pitch(es) that the tracking oscillator is quantized to.

• If you click on the value shown below the word Scale, a pull-down menu opens. Here

•
•

•

you can select a scale or chord. See the listing of preset scales and chords shown
alongside.
Root selects the root key of the respective scale or chord. The Root parameter is not
displayed when chromatic or user is selected.
Any combination of keys can be activated by clicking on the notes of the onscreen
keyboard. Activated keys are illuminated. To disable any active notes on the
keyboard, simply click on the note a second time. The Scale display will change to
user as soon as any key is edited.
The previously displayed scale or chord is used as the starting point when creating a
user scale. This allows you to select a preset scale or chord, and then modify it by
clicking on the notes of the onscreen keyboard.
The last edit will be remembered. You can select a new preset scale or chord, and as
long as you don’t make any changes you can always jump back to the previously set
user scale.

As with all Logic plug-ins, the Root and Scale parameters, and the keys of the onscreen
keyboard can be automated.

Chapter 6 Filter

Max Track
This parameter cuts the high frequencies of the analysis signal, making the pitch
detection more robust. Should the pitch detection produce unstable results, reduce the
Max Track parameter value to the lowest possible setting.
Formant Filter
The Formant Filter Window
The Formant Filter window is divided into two sections by a horizontal line. The upper
half applies to the Analysis section and the lower half to the Synthesis section. Changes
made to the High/Low frequency parameters, the Bands parameter or the Formant
Stretch and Formant Shift parameters will result in visual changes to the Formant Filter
window. This provides you with invaluable feedback on what is happening to the signal
as it is routed through the two formant filter banks.

High/Low Frequency
The blue bar shown just beneath the EVOC 20 TO logo is a multi-part control which is
used to determine the lowest and highest frequencies allowed to pass by the filter
section. The length of the blue bar represents the frequency range for analysis and
synthesis. Frequencies of any audio input which fall outside these boundaries will be
cut. All filter bands are distributed evenly across the range defined by the High/Low
Frequency values.
• To adjust the low frequency value, simply click-hold on the silver slider to the left of
the blue bar, and drag to the right (or left). The value range is 75–750 Hz.
• To adjust the high frequency value, simply click-hold on the silver slider to the right
of the blue bar, and drag to the left (or right). The value range is 800–8000 Hz.
• To adjust both sliders simultaneously, click on the area between the slider halves
(directly on the blue bar) and drag to the left or right.
• You can make changes to the High/Low Frequency values directly by using your
mouse as a slider on the numerical entries—80 and 8000 Hz in the diagram.

Chapter 6 Filter

Lowest/Highest
These parameters can be found in the two small fields on either side of the Formant
Filter window. These switches determine whether the lowest and highest filter bands
are bandpass filters (just like all the bands between them), or whether they act as
lowpass/highpass filters, respectively. Click once on them to switch between the two
curve shapes available.
• In the bandpass setting, the frequencies below/above the lowest/highest bands are
ignored on analysis and synthesis.
• In the highpass (or lowpass) setting, all frequencies below the lowest (or above the
highest) bands will be considered for analysis and synthesis.
Formant Stretch
This parameter alters the width and distribution of all bands in the Synthesis filter bank,
extending or narrowing the frequency range defined by the blue bar (Low/High
Frequency parameters) for the Synthesis filter bank.
If Formant Stretch is set to 0, the width and distribution of the bands in the Synthesis
filter bank is equal to the width of the bands in the Analysis filter bank. Low values
narrow the width of each band, while high values widen it. The control range is from
0.5 to 2 (as a ratio of the overall bandwidth).
Note: You can jump directly to a value of 1 by clicking on its number.
Formant Shift

Chapter 6 Filter

Note: Formant Stretch and Formant Shift are especially useful if the frequency spectrum
of the Synthesis signal does not complement the frequency spectrum of the Analysis
signal. You could create a Synthesis signal in the high frequency range from an Analysis
signal which mainly modulates the sound in a lower frequency range, for example.
Resonance

Resonance is responsible for the basic sonic character of the Vocoder: low settings give
it a soft character, high settings will lead to a more snarling, sharp character. Increasing
the value for Resonance emphasizes the middle frequency of each frequency band.
Note: The use of either, or both, of the Formant Stretch and Formant Shift parameters
can result in the generation of unusual resonant frequencies when high Resonance
settings are used.
Modulation Parameters
The LFO can modulate …
• the frequency (Pitch) of the tracking oscillator (vibrato) or
• the Formant Shift (Shift) parameter of the Synthesis filter bank.
It allows synchronous/non-synchronous modulation in bar, beat (triplet) or free values.

Wave
The Wave switches allow the selection of a waveform type to be used by the LFO. A
selection of Triangle, falling and rising Sawtooth, Square up and down around zero
(bipolar, good for trills), Square up from zero (unipolar, good for changing between two
definable pitches), a random stepped waveform (S&H), and a smoothed random
waveform is available. Simply click on the appropriate button to select a waveform
type.

Intensity
The Intensity sliders control the amount of Formant Shift and Pitch modulation (Vibrato)
by the LFO.

Chapter 6 Filter

Rate
This knob determines the speed of the modulation. Values to the left of the center
positions are synchronized with the sequencer’s tempo and include bar values, triplet
values and more. Values to the right of the center positions are non-synchronous and
displayed in Hertz (cycles per second).
Note: The ability to use synchronous bar values could be used to perform a formant
shift every four bars on a one bar percussion part, which is being cycled. Alternately,
you could perform the same formant shift on every eighth note triplet within the same
part. Either method can generate interesting results, and can lead to new ideas, or a
new lease of life on old audio material.
U/V Detection
Please refer to “Analyzing Speech Signals”, from page 169 onwards, for an explanation
of the U/V Detection principle.

Speech intelligibility is highly dependent on high frequency content, as human hearing
is reliant on these upper-end frequencies to determine syllables within words. Bear this
fact in mind when using the EVOC 20 TO, and take care with filter frequency settings in
the Synthesis and Formant Filter sections.
To aid intelligibility, it may be worthwhile using equalization to boost particular
frequencies in the mid to high frequency range, before processing the signal with the
EVOC 20 TO. Please see the “Tips for Better Speech Intelligibility” on page 169 for
further information.
Sensitivity
This parameter determines how responsive the U/V detection is. By turning this knob
to the right, more of the individual unvoiced portions of the input signal are
recognized.
When high settings are used, the increased sensitivity to unvoiced signals can lead to
the U/V source—determined by the Mode parameter—being used on the majority of
the input signal, including voiced signals. Sonically, this results in a sound that
resembles a radio signal which is breaking up and contains a lot of static or noise.

Chapter 6 Filter

Mode
Here, you select the sound source(s) which can be used to replace the unvoiced content
of the input signal. Possible settings are Off, Noise, Noise + Synth, or Blend.

• Noise—uses noise alone for the unvoiced portions of the sound.
• Noise + Synth—uses noise and the synthesizer for the unvoiced portions of the

sound,
• Blend—uses the Analysis signal after it has passed through a highpass filter, for the

unvoiced portions of the sound. This filtered analysis signal is then mixed with the
EVOC 20 TO output signal. The Sensitivity parameter has no effect on this setting.
Level
Level controls the amount of the signal (Noise, Noise + Synth, or Blend) used to replace
the unvoiced content of the input signal.

Warning: Care should be taken with this control, particularly when a high Sensitivity
value is used, to avoid internally overloading the EVOC 20 TO.
Output Parameters
Signal

This pull-down menu offers the choice of Voc(oder), Syn(thesis) and Ana(lysis). Selection
of one of these determines the signal that you wish to send to the EVOC 20 TO’s main
outputs. To hear the vocoder effect, Signal must be set to Voc. The other two settings
are useful for monitoring purposes.

Chapter 6 Filter

Level
Level controls the volume of the EVOC 20 TO’s output signal.
Stereo Mode

This pull-down menu determines the input/output mode of the Synthesis filter bank.
Choices are: m/s—mono input to stereo output and s/s—stereo input to stereo output.
Note: The Stereo Mode should be set to m/s if the signal going into the Synthesis filter
bank is monophonic or Synthesis In is set to Osc.
Note: Stereo/stereo (s/s) is the preferred setting for stereo Synthesis input signals. In
this case, the stereo signal is processed by a separate filter bank for the left and right
channels. When using the m/s mode on stereo Synthesis input signals, the stereo signal
is first summed to mono before it is passed on to the Synthesis filter bank.
Stereo Width
Stereo Width distributes the output signals of the Synthesis section’s filter bands in the
stereo field.
• In the left position, the output of all bands are centered.
• In the center position, the output of all bands ascends from left to right.
• In the right position, the bands are output—alternately—on the left and right
channels.
Note: The stereo/stereo mode (s/s) uses one filter bank per channel. The positioning of
the frequency bands correspond to that described above, but the bands of each filter
bank ascend in opposing directions, from left to right.

Chapter 6 Filter

MIDI Controllers Received
The following tables show the CC numbers used when the following MIDI preference is
active: Options > Settings > MIDI Options > (Version 4.x behavior).
Sidechain Analysis

Synthesis

Pitch Quantize

Formant Filter

LFO

U/V Detection

Output

Chapter 6 Filter

Attack

CC #79

Release

CC #80

Freeze

CC #78

Input Mode

CC #65

Bands

CC #66

FM Ratio

CC #86

FM Intensity

CC #87

Coarse Tune

CC #85

Fine Tune

CC #105

Strength

CC #90

Root Scale Key

CC #103

Root Scale Presets

CC #104

Max Track

CC #88

Glide Time

CC #89

FF Low Freq

CC #67

FF Hi Freq

CC #68

Formant Shift

CC #69

Formant Stretch

CC #74

FF Resonance

CC #75

FF Low/Bandpass Select

CC #76

FF High/Bandpass Select

CC #77

LFO Rate

CC #70

LFO Waveform Select

CC #71

Shift Intensity

CC #72

Pitch Intensity

CC #73

Sensitivity

CC # 83

Mode

CC # 82

Level

CC # 84

Signal Out Select

CC #106

Level

CC #107

Mono/Stereo Select

CC #108

Stereo Width

CC #81

Root/Scale kyb.

CC #91

CC #92

CC #93

CC #94

CC #95

CC #96

CC #97

CC #98

CC #99

CC #100

CC #101

CC #102

High Cut/Low Cut
• The Low Cut filter attenuates the frequency range below the selected frequency.
• The High Cut filter attenuates the frequency range above the selected frequency.

High Pass/Low Pass Filter
• The High Pass Filter affects the frequency range below the set frequency. Higher

frequencies pass through the filter. You can use the High Pass Filter to completely get
rid of the bass range below a selectable frequency.
• The Low Pass Filter affects the frequency range above the selected frequency. Lower
frequencies pass through the filter. You can use the Low Pass Filter to completely get
rid of the treble range above a selectable frequency.

Chapter 6 Filter

Delay

This chapter describes Logic’s delay effects.
This includes the Sample Delay, Tape Delay and Stereo Delay plug-ins.

Sample Delay
This plug-in allows the simple delaying of a channel by single sample values. The stereo
version of the plug-in provides separate controls for each channel. This plug-in (when
used in conjunction with the phase inversion capabilities of the Gain plug-in) is
particularly suited to the correction of run-time problems that may occur with multichannel microphones.
Every sample (at a frequency of 44.1 kHz) is equivalent to the time taken for a sound
wave to travel 7.76 millimeters. Looked at differently: If you delay one channel of a
stereo microphone by 13 samples, this will emulate an acoustic (microphone)
separation of 10 centimeters.

Tape Delay

The Tape Delay simulates a vintage tape echo device, although with some very useful
features that such old devices never offered. The first of these is that it’s delay settings
are variable in musical increments. It is equipped with a highpass and lowpass filter in
the feedback circuit, as well as a circuit that simulates tape saturation effects. This plugin is ideal for the dub delays invented by Jamaican toast masters, and used in many
styles of music today.
Switching the Sync button on forces the plug-in to use the internal tempo of the
sequencer. Tempo information is updated in the plug-in window when you open it,
and every time you subsequently execute a mouse operation. The plug-in can even
handle tempo changes. The Tempo parameter field serves solely to display the current
bpm value—you can’t use it to change the tempo of the sequencer.
When you want to create dotted note values, move the Groove slider all the way to the
right to 75%; for triplets, select the 33.33% setting. Note that all intermediate values are
possible. You can view the current delay value in the Delay parameter field.
Disengage Sync if you would like to adjust the delay time independently of the song
tempo (or change the song tempo without changing the delay time). In this mode, the
bpm or ms values can be altered freely by clicking in the Tempo parameter field, while
dragging up or down with the mouse. Note when changing the ms values using the
left portion of the Delay parameter field, the ms values will increment in large steps,
while using the right portion of the field will increment the ms values in small steps.
As you might expect, the Feedback slider determines feedback intensity; in other words,
the amount of delayed and filtered signal that is routed back to the input of the Tape
Delay. When you set it to the lowest possible value, the Tape Delay generates a single
echo. If Feedback is turned all the way up, the echoes are repeated ad infinitum. Keep
in mind that the levels of the original signal and its taps (echo repeats) tend to add up,
and may cause distortion. This is where the internal tape saturation circuit comes to the
rescue—it can be used to ensure that these overdriven signals sound good.
The Freeze parameter captures the current delay repeats and sustains them until the
Freeze parameter is released.

Chapter 7 Delay

You can shape the sound of the echoes, using the on-board highpass and lowpass
filters. Although these filters are fairly flat, they’re not located post-output. They are
located in the feedback circuit, meaning that the effect achieved by these filters
increases in intensity with each repeat. If you’re in the mood for an increasingly muddy
tone, move the High Cut filter slider towards the left. For ever thinner echoes, move the
Low Cut filter slider towards the right.
The Mix slider determines the balance between the original (dry) signal and effects
(wet) signals. If you’ve inserted the Tape Delay in an individual track, you’ll generally
find that settings of up to 50% are desirable. If the Tape Delay is patched to the insert
of a Bus channel, and you’re routing the signals of a track to the plug-in with the Send
controls, you should set the Mix slider to 100%, and leave it there.
If you’re unable to hear the effect, even though you’ve set up a suitable configuration,
be sure to check out not only the Mix knob, but also the filter settings: Move the High
Cut filter slider to the far right, and the Low Cut filter slider to the far left.
The Tape Delay includes an LFO for delay time modulation. Use it to produce very
pleasant and special chorus effects, even on long delays. The LFO produces a triangular
wave, with adjustable speed and modulation intensity, that can be evened out with the
Smooth parameter. This also smoothes the Flutter. Flutter simulates the irregularities of
tape transport speeds used in analog tape delay units, and is also adjustable in speed
and intensity.
There are three further parameters in the Tape Delay’s Controls view. The Dry and Wet
sliders can be used to control the original and effect signal amounts individually,
independently of the Mix parameter. Distortion Level can lower the distorted signal
(tape saturation) level by up to 20 dB.

Chapter 7 Delay

Stereo Delay

The Stereo Delay works much like the Tape Delay, which is why we’ll skip the general
info, and take a closer look at the differences between the two. There is just one Stereo
Delay (s/s), hence the stereo input and output. You are free to use the Stereo Delay for
monaural tracks or busses, when you want to create independent delays for the two
stereo sides. Please bear in mind that if you use this option, the track or bus has two
channels from the point of insertion forward. Unlike the Tape Delay, the Stereo Delay
does not feature a circuit that replicates tape saturation.
You can set the Delay (using Note buttons and Groove sliders), Feedback, and Mix
values separately for the two sides. The High Cut and Low Cut sliders, however, apply
equally to both sides. In addition, the plug-in features a Crossfeed knob for each stereo
side. It determines the feedback intensity—or the level at which each signal is routed
to the opposite stereo side.
There are ten additional parameters in the Stereo Delay’s Controls view.
If you would like to adjust the delay time independently of the song tempo, select ms
in the Delay Unit pull-down menu. You can use the Left Delay and Right Delay sliders just
above the Delay Unit pull-down menu to set the delay time in milliseconds. Left Input
and Right Input determine the input signal for the two stereo sides. You can choose
between Off, Left, Right, L+R, L−R.
Selecting the Inv option in the Phase Left FB and Phase Right FB pull-down menus allows
you to invert the phase of the corresponding channel’s feedback signal. The inv option
is also available in the Phase L→R FB and Phase R→L FB pull-down menus, where it can
be used to transfer the inverted feedback signal of the left/right channel to the right/
left channel. The Tempo Freeze parameter captures the current delay time and sustains
it until the Freeze parameter is released.

Chapter 7 Delay

Modulation

This chapter introduces Logic’s modulation effects.
This includes the Modulation Delay, Chorus, Flanger, Phaser, Ring Modulator, Tremolo,
Ensemble, Rotor Cabinet, Scanner Vibrato, and Spreader plug-ins.

Modulation Delay

As its name implies, the Modulation Delay generates effects such as flanging or chorus,
based on modulated short delays. It can also be used—without modulation—to create
resonator or doubling effects.
The modulation section consists of two LFOs, with variable frequencies (0 to 20 Hz). The
balance between these two is determined by the LFO Mix slider. Use the Width slider to
enter the desired modulation width. When the Width slider is set to the far right
position, delay modulation is switched off completely. The Vol.Mod. (Volume
Modulation) slider determines the intensity of amplitude modulation (Tremolo). The
Constant Mod. (Constant Modulation) button lets you do just that—ensure that the
modulation width remains constant, regardless of the modulation rate. When this
feature is switched off, higher modulation frequencies reduce the modulation width. In
simple delay circuits, a delay modulation would normally also modulate the pitch of
the signal. Use the Anti Pitch button to ensure that the pitch of the modulated signal
remains constant. This is exactly how high-end chorus and flanger effects work.

Set the basic delay time with the Flanger-Chorus knob. Set to the far left position, the
Modulation Delay puts on its flanger cap. As you move towards the center position, it
thinks it’s a chorus. As you move the knob closer to the far right position, you will hear
clearly audible delay taps. This latter type of setting is generally used without
modulation (Width = 0), for doubling effects.
The Stereo Phase knob defines the phase of the modulation between the left and right
stereo sides. At 0°, the extreme values of the modulation are achieved simultaneously
on both sides, at 180°, the extreme values opposite each other are reached
simultaneously.
The Feedback slider determines the intensity at which the effect’s signal feedback is
routed to the input. If you’re going for radical flanging effects, enter a high Feedback
value. If simple doubling is what you’re after, you won’t want any feedback at all. The
Mix slider determines the balance between dry and wet signals.
The Controls view offers six further parameters:
If you set True Analog to on, an additional all-pass filter is switched into the signal path.
An all-pass filter shifts a signal’s phase angle, influencing its stereo image. Use Analog
Left and Analog Right to control the way that the allpass filter affects each of the stereo
channels.
The Speed LFO 1 R and Speed LFO 2 R sliders allow independent modulation rate settings
for LFO1 and 2 (for the right stereo channel). These parameters only work if the Free
option is chosen in the Stereo pull-down menu. With Stereo set to Link, the modulation
rates of the left and right stereo channels are tied to each other, and rates are set by
the LFO controls in the Plug-in window. In this situation, the Speed LFO 1 R and Speed
LFO 2 R parameters are non-functional.

Chorus

The Chorus effect is based on a delay line. It’s output is mixed with the original, dry
signal. While the chorus effects delay time is set internally, you can define its
modulation width (Intensity parameter) and modulation frequency (Speed parameter).
The Mix slider determines the balance of dry and wet signals.

Chapter 8 Modulation

Flanger

The Flanger works in a similar fashion to the Chorus, but with a shorter delay time, and
the output signal being fed back into the input of the delay line. Use the Intensity slider
to determine the Flanger’s modulation width. Speed sets the frequency of the
modulation. Feedback determines the amount of the delayed signal that is routed back
into the input. Negative values invert the phase of the routed signal. The Mix slider
determines the balance of dry and wet signals.

Phaser

The Phaser emulates the effect of analog phaser circuits with four to twelve orders (as
in 4th order, 5th order and so on) Use the Order slider to set the desired number of
orders. As a rule, the more orders a phaser has, the heavier the effect. The 4, 6, 8, 10,
and 12 settings put five different phaser algorithms at your fingertips, all of which
replicate the analog circuits that they are modeled on, each designed for a specific
application.
Note: You are free to select odd numbered settings (5, 7, 9, 11), which, strictly speaking,
don’t generate actual phasing. The more subtle comb filtering effects produced by odd
numbered settings can, however, come in handy on occasion.

Chapter 8 Modulation

The modulation section offers two LFOs, featuring individually variable frequencies, and
freely variable mix options (LFO Mix). Additionally, the frequency of LFO 1 can be
modulated by the level of the input signal. Use the Envelope Modulation slider to set
the desired modulation intensity. By staking out the limits of the modulation with its
highest and lowest values, you can determine the modulation width and range. These
high/low limits are controlled by the Sweep Ceiling and Sweep Floor sliders—you can
enter values for them directly in the form of the desired frequency. This value also
determines the maximum intensity of the comb filtering created by the phasing effect.
The Stereo Phase knob is used to define the phase for the left and right channels of a
stereo phaser (s/s). When you’re using a monaural phaser, this parameter is, of course,
meaningless and can’t be set. As the icing on the phasing cake, you can tweak the
Color slider to add just that to the effect. Technically, the comb filtering effect is
amplified via feedback.
There are six additional parameters in the Phaser’s Controls view.
The Mix slider determines the balance of dry and wet signals. Negative values result in
a phase inverted mix of effect and direct signal. The Phaser’s built-in envelope follower
tracks any volume changes in the input signal, generating a dynamic control signal.
This control signal can be used as a modulation source. Dir.-Env-Mod sets the desired
modulation intensity for the envelope control signal. Warmth switches on an additional
distortion effect, which allows the creation of warm overdrive effects. FB Filter can be
used to activate an additional filter section, which processes the feedback signal of the
Pitch Shifter. This filter section consists of a highpass and lowpass filter, where cutoff
frequency can be set with LP Cutoff and HP Cutoff.

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Chapter 8 Modulation

RingShifter—Ring Modulator/Frequency Shifter

Logic’s RingShifter plug-in combines a ring modulator with a frequency shifter effect.
These two related effects are based on modulation of the signal amplitude. Both effects
were popular during the 1970’s, and are currently experiencing something of a
renaissance. The ring modulator, for example, was extensively used on jazz rock and
fusion records in the early 70’s. The frequency shifter was, and still is, found as part of
many modular synthesizer systems. Due to the intricate nature of its hardware, the
frequency shifter was (and remains) relatively expensive to produce, and was therefore
never as widespread as the simpler ring modulator.

Technical Background
The ring modulator modulates the amplitude of the audio input signal using either; the
internal oscillator or a second audio signal. The frequency spectrum of the resulting
effect signal equals the sum and difference of the frequency content of the two original
signals. Its sound is often described as metallic or clangorous.
An elaborate array of allpass filters enables the frequency shifter to separate the sum
and difference signals into two separate audio signals—one carries the audio signal
with its frequency spectrum shifted up, the other with its spectrum shifted down. The
amount of frequency shift is set via the frequency of the internal sine wave oscillator.
Frequency shifting should not be confused with pitch shifting. Pitch shifting transposes
the original signal, leaving its harmonic frequency relationship intact. The frequency
shifter shifts the frequency content by a fixed amount and, in doing so, alters the
frequency relationship of the original harmonics. The resulting sounds range between
sweet and spacious phasing effects to strange robotic timbres.

Chapter 8 Modulation

101

Modes
The four Mode buttons determine whether the plug-in either operates as a frequency
shifter or as a ring modulator. The frequency shifter offers the Single and Dual settings.
The ring modulator provides the OSC and Side Chain settings.
• Single (Frequency Shifter): The frequency shifter generates a single shifted effect
signal. The position of the large Frequency rotary control determines whether the
signal is shifted up (positive value) or down (negative value).
• Dual (Frequency Shifter): The frequency shifting process produces one shifted effect
signal for each stereo channel—one is shifted up, the other is shifted down. The
position of the large Frequency rotary control (in relation to the 0 point) determines
the shift direction in the left versus the right channel.
• OSC (Ring Modulator): The Ring Modulator uses the internal sine wave oscillator to
modulate the input signal.
• Side Chain (Ring Modulator): The Ring Modulator modulates the amplitude of the
input signal with the audio signal assigned via the side chain of the plug-in.
Note: The internal sine wave oscillator has no effect in the Side Chain mode, and for
this reason, the oscillator frequency controls are not accessible.

The Oscillator
In both the Ring Modulator OSC mode and the Frequency Shifter modes, the internal
sine wave oscillator serves as an amplitude modulator of the input signal. The large
Frequency rotary control is used to set the frequency of the sine oscillator. It can be set
between 0 and ±5,000 Hz in extremely fine increments.
• In the Frequency Shifter modes, this parameter controls the amount of frequency
shifting (up and/or down) applied to the input signal.
• In the OSC mode of the ring modulator, this parameter controls the frequency
content (timbre) of the resulting effect. This timbre can range from subtle tremolo
effects to clangorous metallic sounds.
To optimize adjustment, the scaling of the Frequency rotary control can be switched via
the Lin(ear) and Exp(onential) buttons. The exponential scaling offers extremely small
increments around the 0 point, which is useful for programming slow moving phasing
and tremolo effects. In the Lin(ear) mode, the resolution of the scale is even over the
entire control range.
Further to these options, the oscillator frequency can be modulated with an envelope
follower and LFO (see later). The oscillator is capable of frequency sweeps through the
0 Hz point. The modulation depth for the envelope follower and LFO is set independently, by using the corresponding bipolar slider.

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Chapter 8 Modulation

Delay
The effect signal is routed through a delay, following the oscillator. The Level control
sets the level of the delay added to the ring modulated or frequency shifted signal.
Note: A Level value of 0 passes the effect signal directly to the output (bypass).
The Time control sets the delay value from 0 to 2,000 milliseconds. Activation of the
Sync button synchronizes the delay to your Logic song tempo, in musical note values.

Output
The RingShifter offers a feedback loop, which operates independently of the delay
section, by routing the output of the RingShifter back into its input. Feedback gives the
RingShifter sound an additional edge, and is useful for a variety of special effects. In
combination with a slow oscillator sweep, it produces a rich phasing sound. Comb
filtering effects are created using high Feedback settings with a short delay time
(< 10 ms). Using longer delay times in combination with Feedback creates spiralling
and continuously rising and falling frequency shift effects.
The Stereo Width rotary control determines the breadth of the effect signal in the stereo field.
Note: Stereo Width only affects the effect signal of the RingShifter, not the dry input
signal.
The Dry/Wet rotary control sets the mix ratio of the dry input signal and the wet effect
signal. If required, the Dry/Wet mix can also be modulated with an envelope follower
and LFO. The modulation depth for the envelope follower and LFO is set independently with their bipolar sliders

Modulation Sources
The Oscillator Frequency and Dry/Wet mix ratio parameters can be modulated via the
internal Envelope Follower and LFO. The Oscillator Frequency even allows modulation
through the 0 Hz point, thus changing the oscillation direction.
The Envelope Follower analyzes the amplitude (volume) of the input signal, and uses
this to create a continuously changing control signal—a dynamic volume envelope of
the input signal. This “control signal” can be used for modulation purposes. The Power
button turns the Envelope Follower on or off. The Sens(itivity) slider determines how
responsive the Envelope Follower is to the input signal. At lower settings, the Envelope
Follower will only react to the most dominant signal peaks. At higher settings, the
Envelope Follower will track the signal more closely, but may react less dynamically. Try
to find a suitable median (compromise) setting for the input signal. The response time
of the Envelope Follower is set with the Attack slider. Decay controls the time it takes
the Envelope Follower to return from a higher to a lower value.

Chapter 8 Modulation

103

The LFO is the second modulation source. It is activated/deactivated via its own Power
button. The LFO produces continuously cycled control signals. The LFO waveform can
be shaped as required via the Symmetry and Smooth sliders. The LFO waveform display
provides visual feedback of the resulting waveform. The Rate rotary control sets the
cycle speed of the LFO. Press the Sync button if you want to synchronize the LFO cycles
with the Logic song tempo (using musical note values).

Tremolo

The tremolo effect is a cyclic modulation of the amplitude, resulting in periodic volume
changes of. As opposed to the vibrato effect which can be achieved with the
Modulation Delay plug-in, the amplitude (not the frequency) is the modulated
parameter. You’ll recognize this effect from vintage guitar combo amps (where it is
sometimes incorrectly referred to as vibrato).
The intensity of modulation is set with Depth. Rate defines the speed (frequency) of the
modulation. If Symmetry is set to 50% and Smoothing to 0%, the modulation has a
rectangular shape. This means that the timing of the full volume signal is equal to that
of the low volume signal, and that switching between both states occurs abruptly. You
can define the loud/quiet time ratio with Symmetry, and make it fade gently in or out
with Smoothing. Stereophase defines whether the modulation takes place in phase or
out of phase, when in stereo mode. It can be set to any phase angle. When set to out of
phase (−180º) the balance wanders from left to right. When set to 180º, left and right
channels are altered in volume simultaneously (in phase).
The graphic display is self-explanatory: All parameters, except modulation speed
(Rate), are displayed.

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Ensemble

The Ensemble is like a pitch shifter on steroids—it consists of eight internal,
modulatable pitch shifters. Two standard LFOs and one random LFO enable you to
come up with fairly complex pitch modulations, which—much like a natural chorus
effect—conjure up the impression of an instrumental or vocal ensemble. The
Ensemble’s graphic visually represents the number of voices, and their modulations.
Use the Voices slider to determine how many voices (1 to 8) are generated, in addition
to the original. Please note that the plug-in’s appetite for computer resources increases
proportionally to the number of voices: When you activate eight voices, the Ensemble
requires roughly eight times the CPU power of a pitch shifter.
The two conventional LFOs and the random LFO (which generates random
modulations), each feature a Rate knob that controls frequency, and an Intensity slider
to determine the modulation width.
The Phase knob controls the phase relationship between the modulations of the
individual voices. The value that you select here depends on the number of voices,
which is why it is indicated in percentages rather than degrees. The value 100 (or −100)
is equal to the greatest possible distance between the modulation phase of all voices.
Here, the voices are distributed an equal distance apart over the full 360°.
The Stereo Base slider serves to distribute the voices across the stereo field. When you
set a value of 100%, the stereo base is expanded artificially. Please note that monaural
compatibility may suffer.
In addition to the familiar Mix slider that determines the balance of dry and wet signals,
the Ensemble also features an Effect Volume knob. This lets you determine the level of
the effects signal separately. This feature allows you to compensate for changes in
volume caused by manipulating the Voices parameter.

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Rotor Cabinet

The Rotor Cabinet plug-in is based on the EVB3’s Rotor Cabinet effect section. A
detailed description of it’s parameters can be found in the EVB3’s “Rotor Cabinet”
section on page 474.
Please note: There is no Speed Control parameter on the Rotor Cabinet plug-in. You can
switch rotor speeds manually.

Scanner Vibrato
In its mono-version, the stereo parameters of the scanner vibrato are hidden behind a
transparent cover (right):

The Scanner Vibrato plug-in is based on the EVB3’s Scanner Vibrato effect section. More
information about its parameters can be found in the EVB3’s “Scanner Vibrato” section
on page 462.
The stereo version of the Scanner Vibrato effect features two additional
parameters: Stereo Phase and Rate Right.
If Stereo Phase is set to free, the modulation speed can be set independently for the left
(Rate Left) and right (Rate Right) channels. This allows for quite wild effects, as left and
right modulation are not synchronized with each other.
If Stereo Phase is set to 0 to 360 degrees, the modulation speed for both the left and
right channels is set with Rate Left. Stereo Phase determines the phase relationship
between left and right channel modulations, thus enabling synchronized stereo effects.
Rate Right has no function when in this mode.

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Spreader
The Spreader plug-in widens the stereo spectrum with an effect that is quite similar to
the Chorus effect. The frequency range of the original signal is periodically shifted in a
non-linear way. In comparison to the Stereo Spread effect, the perceived pitch changes.
Use the LFO Intensity parameter to set the modulation width of the Spreader. LFO Speed
controls the modulation frequency. Channel Delay determines the delay time in
Samples. Mix sets the balance of dry and wet signals.

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Reverb

This chapter describes Logic’s reverb effects.
This includes AVerb, SilverVerb, GoldVerb, PlatinumVerb, Enverb, and Space Designer.
Space Designer is Logic’s only convolution reverb, and is described separately in the
“Convolution Reverb: Space Designer” chapter, from page 117.

AVerb

Although the AVerb is based on a simple reverb algorithm, it delivers remarkably good
results.
The actual reverb algorithm is controlled by just four parameters:
• As its name implies, Reflectivity defines how reflective the imaginary walls, ceiling,
and floor will be.
• Room Size challenges your architectural skills—use it to define the dimensions of
simulated rooms.
• Density/Time determines both the density and duration of the reverb.
• Pre Delay determines the delay between the original signal and the reverb tail.
The Mix parameter determines the balance between the effected (wet) and direct (dry)
signals.

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Where high Pre Delay settings tend to generate something similar to an echo, low
values often muddy the original signal. Ideally, you should go for as high a setting as
possible before the plug-in begins generating something that sounds like a tap delay.
With appropriate Pre Delay settings, you can apply relatively generous amounts of
reverb to percussive parts, while retaining definition on the attack portions of the
sounds.

SilverVerb

The SilverVerb algorithm is controlled by just three parameters: As its name implies,
Reflectivity defines how reflective the imaginary walls, ceiling, and floor will be. Room
Size challenges your architectural skills—use it to define the dimensions of simulated
rooms. The graphic display visually represents these parameter settings.
Predelay determines the delay between the original signal and the reverb tail.
Whereas high Predelay settings tend to generate something similar to an echo, low
values often muddy the original signal. Ideally, you should go for as high a setting as
possible before the plug-in begins generating something that sounds like a delay tap.
With appropriate Predelay settings, you can apply relatively generous amounts of
reverb to percussive parts, while allowing the attacks to remain intelligible.
Low Cut and High Cut let you filter bass and treble frequencies out of the reverb tail.
In most cases this will open up your mix. The reason for this is that a long reverb with a
great deal of bottom end generally makes for a flabby mix, and high frequencies in the
reverb usually sound somewhat unpleasant, hamper speech intelligibility, or mask the
overtones of the original signals.
There are four further parameters that are available in the Extra Controls view.
Density/Time determines both the density and duration of the reverb. Small value
settings tend to generate something similar to an echo. High values result in a reverblike effect.

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The Modulation Rate, Modulation Int and Modulation Phase parameters control an
additional modulation delay. It consists of two LFOs with variable frequencies (set with
Modulation Rate). The desired modulation width is set with the Modulation Int slider.
When this slider is set to the far right position, delay modulation is switched off
completely. The Modulation Phase knob defines the phase of the modulation between
the left and right stereo sides. At 0°, the extreme values of the modulation are achieved
simultaneously on both sides, at 180°, the extreme values opposite each other are
reached simultaneously.

GoldVerb

The GoldVerb consists of two sections: Early Reflections and Reverb (diffuse
reverberations). The balance between these two sections is controlled by the Balance
ER/Reverb slider, located above the graphic. When you set this Balance slider to either of
its extreme positions, the unused section is deactivated, maximizing performance.

Early Reflections
This section emulates the original signal’s first reflections as they bounce off the walls,
ceiling, and floor of a natural room. These early reflections are essential to how we
perceive a room. All information about the size and shape of a room capable of being
discerned by the human ear is contained in these early reflections.

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Predelay
Predelay is the amount of time that elapses between the original signal, and the arrival
of the early reflections. In any given room size and shape, Predelay determines the
distance between the listener and the walls, ceiling, and floor. When used with
artificially generated reverb, it has proven advantageous to allow this parameter to be
manipulated separately from, and over a greater range than, what is considered natural
for Predelay. In practice, too short a Predelay tends to make it difficult to pinpoint the
position of the signal. It can also color the sound of the original signal. On the other
hand, too long a Predelay can be perceived as an unnatural echo. It can also divorce
the original signal from its early reflections, which leaves an audible gap. The ideal
Predelay setting depends on the properties or, more accurately, the envelope of the
original signal. Percussive signals generally require shorter Predelays than signals where
the attack fades in gradually. A good practice is to use the longest Predelay possible
before undesirable side effects, such as an audible echo, begin materializing.
Room Shape
Use this slider to define the geometric form of the room. The numeric value (3 to 7)
represents the number of corners it has.
Room Size
Unsurprisingly, Room Size determines the dimensions of the room. The numeric value
indicates the length of its walls—the distance between two corners.
Stereo Base
The Stereo Base parameter enables you to define the distance between the two virtual
microphones that you are using to audition the simulated room. Spacing the
microphones slightly further apart than the distance between two human ears
generally delivers the best results. Of course, more realistic results can be obtained if
you choose to use the distance between two ears located on opposite sides of the
same head.

Reverb
This section generates diffuse reverberation.
Initial Delay
This is the delay between the original signal and the diffuse reverb tail. If you’re going
for a natural-sounding, harmonic reverb, the transition between the early reflections
and the reverb tail should be as smooth and seamless as possible. Basically, what we
said about the Predelay holds true for this parameter:
Set the Initial Delay so that it is as long as possible without a perceptible gap between
the early reflections and the reverb tail.

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Density
This parameter controls the density of the diffuse reverb. Ordinarily, you want the
signal to be as dense as possible. However, less Density means the plug-in eats up less
computing power. Moreover, in rare instances, too great a Density can color the sound,
which you can fix simply by reducing the Density knob value. Conversely, if you select a
Density value that is too low, the reverb tail will sound grainy.
Diffusion (Controls View)
Diffusion sets the diffusion of the reverb tail. Sometimes, the terms “diffusion” and
“density” are confused. The density is the average number of reflections in a given
period of time. The diffusion is the amount of irregularity of the density. High values of
diffusion represent a regular density, with few alterations in level, times, and panorama
position. At low diffusion values, the reflection’s density becomes more irregular and
grainy. The stereo spectrum changes, too.
Reverbtime
Reverbtime is commonly considered as the amount of time it takes for the level of a
reverb signal to drop by 60 dB. This is why the reverb time is often indicated as RT60.
Most natural rooms have a reverb time somewhere in the range of one to three
seconds, a value which absorbent surfaces and furniture reduces. Large empty halls or
churches have reverb times of up to eight seconds, some cavernous or cathedral-like
venues even beyond that.
High Cut
Uneven or absorbent surfaces (wallpaper, wood paneling, carpets, and so on) tend to
reflect lower frequencies better than higher frequencies. The High Cut filter replicates
this effect. If you set the High Cut filter so that it is wide open, the reverb will sound as if
it is reflecting off stone or glass.
Spread
This parameter controls the stereo image of the reverb. At 0%, the plug-in generates a
monaural reverb, at 100%, the stereo base is artificially expanded—which, of course,
makes the reverb sound monumental, but collapses in monaural playback.

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PlatinumVerb

The difference between the PlatinumVerb and the GoldVerb is the former’s enhanced
Reverb section. The Early Reflections sections of the two plug-ins are identical. For more
information, please read the “GoldVerb” section, on page 111. We’ll focus on the
additional features offered by the PlatinumVerb in this section.
The Reverb section of the PlatinumVerb is based on a genuine dual-band concept. This
is to say that the on-board frequency crossover splits the incoming signal into two
bands, which are then processed with reverb in two separate modules.

Parameters of the PlatinumVerb
Crossover
This is the frequency that the two frequency bands are split at, for separate processing.
Low Ratio
This parameter determines the reverb time of the bass band. The Reverbtime parameter
applies to the high band. At 100%, the reverb times for the two bands are identical. At
lower values, the reverb time of the frequencies below the crossover frequency is
shorter. At values greater than 100%, the reverb time for low frequencies is longer.
Both of these phenomena occur in nature. In most mixes, a shorter reverb time for bass
frequencies is preferable. As an example, if you’re using the PlatinumVerb to put reverb
on a drumloop featuring kick and snare drums, a short reverb time for the kick drum
allows you to set a substantially higher wet signal.
Low Level
This knob determines the level of the bass reverb. At 0 dB, the volume of the two
bands is equal. The bass reverb level can be boosted by up to 12 dB and attenuated by
up to 100 dB.

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In the vast majority of mixes, your best bet is to set a lower level for the low frequency
reverb signal. This enables you to turn up the level of the bass instrument—making it
sound punchier. This also helps to counter bottom-end masking effects.
The Controls view offers four additional parameters.
ER Scale allows you to scale the early reflections along the time axis, enabling the Room
Shape, Room Size and Stereo Base parameters to be influenced simultaneously. Dry and
Wet can be used to control the amounts of the original and effect signal individually,
and independently of the Mix parameter. The Diffusion slider is also available in the
GoldVerb plug-in. A detailed description of its function can be found on page 113.

EnVerb

Logic’s EnVerb is based on a rather unusual and innovative reverb algorithm. It has a
unique feature—you can adjust the envelope of the diffuse reverb tail freely. This
provides options that far exceed those of a conventional gated reverb.
The EnVerb algorithm requires a reasonable amount of computing power.

Time Parameters
With a concept as sophisticated as that of the EnVerb, you can well imagine that a
single parameter for reverb time just won’t do the trick.
Original Delay
This parameter enables you to delay the original signal. This delay is only noticeable
when the Mix parameter is set to a value other than 100%. The starting point of the
diffuse reverb tail is not influenced in any way.
A delayed original signal is particularly handy when you want to generate reverse
reverb: Set all envelope parameters to 0 with the exception of Attack and Original
Delay, which you should set to approximately the same value that you want to predelay the given region or track.

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Predelay
This is the delay between the (undelayed) original signal, and the starting point of the
reverb attack phase.
Attack
This is the amount of time it takes for the reverb to climb to its peak level.
Decay
This is the amount of time it takes for the level of the reverb to drop from its peak to
the sustain level.
Sustain
This is the level of the reverb that remains constant throughout the sustain phase.
Hold
This is the duration (time) of the sustain phase.
Release
This is the amount of time that the reverb takes to fade out completely, after it has run
through its sustain phase.

Sound Parameters
The following parameters shape the sound of the reverb. (For more information on
these parameters, check out the in-depth descriptions of the GoldVerb or
PlatinumVerb).
Density
Here, you can set the reverb’s density. Higher values generally sound better.
Spread
Here, you can set the stereo base of the reverb.
High Cut
Here, you can set the high-frequency attenuation for the reverb.
Crossover
Here, you can set the crossover frequency for the Low Level parameter.
Low Level
Relative reverb level of frequencies below the crossover frequency. Although you are
free to turn the level of these frequencies up, in most cases, you’ll get better-sounding
results when you set negative values for this parameter.

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Convolution Reverb:
Space Designer

This chapter introduces you to Logic’s Space Designer
Reverb effect.
Space Designer is a convolution reverb plug-in. Reverberation is generated by means of
a real-time convolution process, using any loaded impulse response (IR) recording
(reverb sample). Put another way, an IR recording of an actual real-world room, a
vintage plate or spring reverb unit, for example. The result is an exceptionally realistic
reverb/room sound.
An impulse response can be viewed as the total echoes (reflections) in a given room,
following an initial signal spike. The waveform display of an IR is also known as a
reflectogram. The IR file is simply an audio file.
Space Designer is able to modify existing impulse responses, providing unprecedented
control over dynamics, timbre, and length via a comprehensive set of parameters such
as envelopes and filters. The graphically-editable envelopes are optimized for reverb
tasks, allowing the creation of ultra-smooth envelope curve shapes.
In addition, Space Designer includes an innovative on-board IR synthesis facility that
incorporates the editing flexibility of the envelopes and filters. Reverbs based on
synthesized IRs provide an incredibly rich and smooth sound.
The sound of the Space Designer reverb never exhibits the shattering or resonances
typical of conventional reverb units—just smooth and rich reverb tails—even when
using extreme settings on the most complex of input material.

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Using Space Designer
Inserting Space Designer
As per all effect plug-ins in Logic, Space Designer can be inserted into any Audio
Object’s insert slot. In doing so, you should take the following information into
consideration:
• Set the Direct Output parameter to a value of 0 (mute), if Space Designer is inserted
into a bus channel. This happens by default in Logic, but should you change it
(accidentally or by design), and things don’t sound right, this is the first thing to look
at.
• When inserted on an Audio Instrument, Audio channel and so on, the Direct Output
level is set to 100%.
• The default Reverb Level is set at −10 dB, regardless of the channel type that Space
Designer is inserted into.
• Note that the Project Manager sees all IRs as simple audio files.

Automation in Logic
As with all Logic plug-ins, a Space Designer effect used on a Logic mixer channel is fully
programmable.
Important: Space Designer can not be fully automated as per most other Logic plugins. This is because Space Designer needs to reload the Impulse Response (and
recalculate the convolution) before audio can be routed through it.
You can, however, record, edit, and play back any movement of the following Space
Designer parameters via Logic’s track-based automation system.
• Stereo Crossfeed
• Direct Output
• Reverb Output

Operation of Space Designer’s Parameters
Space Designer’s envelopes can be adjusted graphically, by click-holding on the nodes
(shown as hollow or filled dots) or envelope curves, and dragging to the desired
position in the Envelope window of the Space Designer’s GUI. The envelopes can also
be adjusted via click-hold and vertically dragging on the various parameter values in
the long horizontal panel below the graphical Envelope window.
There are also some clickable key parameter positions in the GUI—with the slider or
dial jumping to that position immediately. Clickable parameter positions are:
• The three base positions of the reverb input slider—Stereo, Mono, Xstereo.
• The zero mark of the Low Shelving EQ Gain control.
• The HP, BP, 12 dB and 6 dB filter modes.

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Note that parameter changes occur after the release of the mouse button, and after an
additional grace period has elapsed, which is indicated by a blue bar. The calculation
itself requires a certain amount of processing time—depending on the speed of your
Macintosh CPU(s). During this calculation time, no other parameters can be adjusted. A
(red) progress bar below the Length parameter panel will advise you of the calculation
status.
• Prior to the appearance of the progress bar, a blue grace period bar will appear for
some parameters. This represents the time before any calculation starts. The bar
counts down from right to left, during which time you can adjust the parameter.

Space Designer’s Parameters
The following chapter discusses the parameters of the Space Designer.

Impulse Response Parameters

The Impulse Response section allows you to select or create Impulse Responses.
The Space Designer offers two modes of operation—IR Sample and Synthesized IR.

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IR Sample
When the IR Sample button is initially clicked, an operating system File Selector dialog
will be launched, allowing you to select the desired Impulse Response file from a folder
on your hard disks or CD.
If you have already loaded an IR file, this button is simply used to switch back from
Synthesized IR to IR Sample mode. To change the Impulse Response click the downward
pointing arrow to the right of the button.
The default IR folder is found in the following path: Local: /Library/Application Support/
Apple/Impulse Responses
Simply browse to the desired file, and click OK. Any mono, stereo or split stereo AIFF,
SDII or WAV file can be used.

The downwards pointing arrow to the right of the button accesses the following menu
functions:
• Load IR—loads an IR sample without changing the envelopes.
• Load IR & Init—loads an IR sample and initializes the envelopes.
The name of the loaded IR file and its length are displayed in the Envelope window.
Synthesized IR
Clicking on this button will activate the Impulse Response synthesizer. A synthesized
IR—determined by the values of the Length, Envelope, Filter, and Spread parameters—is
generated.

Subsequent button pushes will randomly generate new IRs with slightly different
reflection patterns. The current IR state will always be saved with a setting, allowing for
an accurate reproduction of the reverberation sound when next loaded.
You may freely switch between a loaded IR sample and a synthesized IR without losing
the settings of the other.

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Sample Rate
This parameter determines the sample rate of the Impulse Response. By default the
current Logic song sample rate is used by Space Designer as well (if the Logic song is
running at 96 kHz, Space Designer uses the same rate). When loading an Impulse
Response, Space Designer automatically converts the sample rate of the IR to match
the current Logic song sample rate—should it be necessary. As an example, this allows
you to load a 44.1 kHz Impulse Response into a Logic song running at 96 kHz, and vice
versa.

Three other options are also available. These are half-divisions of the preceding value—
one-half, one-quarter, one-eighth. As examples:
• If the top sample rate is 96 kHz, the options will be 48 kHz, 24 kHz and 12 kHz.
• If 44.1 kHz is the selected sample rate, the options will be 22.05 kHz, 11.025 kHz and
5512 Hz.
Changing the Sample Rate increases (up) or reduces (down) the frequency response
(and length, see below) of the Impulse Response, and therefore the overall sound
quality of the Space Designer.
By selecting half the Sample Rate, the IR becomes twice as long. The highest frequency
that can be reverberated will be halved. This facility results in a behavior that is much
like doubling every dimension of a virtual room (multiplying a room’s volume by eight).
This can sound great!
Another benefit is that the process requires significantly less processing power, making
half Sample Rate settings the ideal solution for wide, open spaces. Check it out!
Don’t worry too much if the maximum bandwidth of the reverb tail is reduced to
11.025 kHz when you select a sample rate of 22.05 kHz (half of 44.1 kHz). Natural room
surfaces (concrete and tiles, excluded) barely reflect such high frequencies.
The lower sample rates can also be used for interesting tempo/pitch and retro-digital
sounding effects.

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Note: If running Space Designer in a song at 96 kHz (utilizing an Impulse Response
originally recorded at 44.1 kHz), you may want to reduce the IR Sample Rate to 1/2. To
do so, use the Sample Rate Slider in Space Designer. Make sure the Preserve Length
function is enabled. This cuts CPU power consumption by about 50%, without
compromising reverb quality. There is no loss in reverb quality, because the Impulse
Response—originally recorded at 44.1 kHz—will not benefit from the higher song
sample rate of 96 kHz.
Note: Similar adjustments can be made while running in Synthesized IR mode. Most
typical reverb sounds don’t feature an excessive amount of high frequency content. If
you were running at 96 kHz, you would need to make use of some deep lowpass
filtering to obtain the mellow frequency response characteristics of many reverb
sounds. As a different approach, you are better served to first reduce the high
frequencies by 1/2 or even 1/4 using the Sample Rate slider in Space Designer, and
than apply the lowpass filter. This conserves a considerable amount of CPU power.
Length
This parameter determines the length of the Impulse Response (sampled or
synthesized).

All envelopes are calculated as a percentage of the overall Length automatically, which
means that if this parameter is altered, your envelope curves will stretch or shrink to fit,
saving you time and effort.
Note that the Length parameter value can not exceed the actual length of an IR sample.
Also note that longer IRs (sampled or synthesized) place a higher strain on the CPU.
Preserve Length
Activating this button preserves the length of the Impulse Response when the Sample
Rate is changed. If the Sample Rate is halved, the Impulse Response will be doubled in
length. To avoid this, make use of the Preserve Length function. Having said this, feel
free to manipulate these two parameters as you see fit, as it can lead to a number of
interesting results. In any case a lower sample rate will save CPU power.

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Global Parameters
Input (Crossfeed)
This parameter allows a stereo input signal to be:

• processed on both channels, retaining the stereo balance of the original signal—top

of slider,
• to be processed in mono—middle of slider,
• to be inverted, with processing for the right channel occurring on the left and vice

versa—bottom of slider.
• a mixture of stereo to mono cross feeds (in-between positions)

Note: This slider is not available when Space Designer is used as a mono plug-in. When
Space Designer is inserted as a mono to stereo plug-in this parameter does not have a
function.
Direct Output
This parameter sets the direct signal output level—the level of the non-effected (dry)
signal.
Note: Set this to a value of 0 (mute) if the Space Designer is inserted in a bus channel,
or when using modelling IRs such as speaker simulations.

Reverb Output
This parameter sets the output level of the effected (wet) signal.

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Rev Vol Compensation

Rev Vol Compensation (Reverb Volume Compensation) attempts to match the
perceived (not actual) volume differences of Impulse Response files. It is set to on by
default, and should generally be left in this mode, although you may find that it isn’t
successful with all types of Impulse Responses. In such situations, switch it off and
adjust the Input and Output levels accordingly.
Pre-Delay

Pre-delay is the amount of time that elapses between the original signal, and the arrival
of the early reflections. Put another way, it represents the time the effected signal is
delayed in relation to the unprocessed direct signal.
In any given room size and shape, pre-delay determines the distance between the
listener and the walls, ceiling, and floor. When used with artificially generated reverb, it
has proven advantageous to allow this parameter to be manipulated separately from,
and over a greater range than what is considered natural for pre-delay. In practice, too
short a Pre-Delay tends to make it difficult to pinpoint the position of the signal. It can
also color the sound of the original signal.
On the other hand, too long a Pre-Delay can be perceived as an unnatural echo. It can
also divorce the original signal from its early reflections, leaving an audible gap
between the signals. The ideal Pre-Delay setting depends on the properties or, more
accurately, the envelope of the original signal. Percussive signals generally require
shorter Pre-Delays than signals where the attack fades in gradually. A good rule of
thumb is to use the longest Pre-Delay possible before undesirable side effects, such as
an audible echo, begin materializing. More information on the general principles of
reverberation can be found in “So, Just What Is Reverberation?” on page 139.
IR Start

This parameter enables you to shift the playback point into the IR, which will effectively
cut off the beginning of the IR.

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The IR Start parameter can for example be used to eliminate any peaks at the
beginning of the IR sample. It also offers a number of creative options, such as its use
when combined with the Reverse function.
Note: The IR Start parameter is not available in Synthesized IR mode. In the Synthesized
IR mode this parameter is not required as, by design, the Length parameter provides
identical functionality.
Latency Compensation

When activated, this parameter delays the direct signal (in the Output section of the
plug-in) to match the processing delay of the effect signal—this compensation occurs
within the plug-in.
Processing latency is 128 samples at 44.1 kHz, and is higher at lower sample rates.

Envelope Window Parameters

The Envelope window of the Space Designer facilitates a new envelope design based
on Bezier curves, with two curve segments (attack and decay) being used to form a
complete envelope. Note the transparent view of each of the envelope curves. Also
note the small IR Overview. It helps to orientate yourself when zoom is active.
The parameters discussed below are global and will affect the currently selected
envelope, dependent on the envelope mode—Volume, Filter or Density.
Please note that use of the word mode here is designed to simplify your
understanding. Obviously, the Space Designer displays, and uses, up to three discrete
envelopes simultaneously.

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Envelope Window Node Handling
Before we get to the buttons, we’d like to briefly touch on the Envelope window of the
Space Designer GUI.
When first launched, a default synthesized Impulse Response and set of Envelopes are
automatically created.
You will see a few nodes placed around and within the Volume Envelope that is
displayed. These nodes are indicators of several parameter positions/values.
If you look carefully, you will see that some nodes are actually a little larger than others,
that some are filled and others are hollow. To explain:

• The filled parameters are active/selected. Hollow ones are not selected/active.
• The small nodes attached to a line are used for finer adjustments to Envelope curves.

We refer to these as the Curve Form nodes. More information on their use can be
found in the “The Volume Attack and Decay Curve Form Nodes” section, on page 130.
• The large nodes are value indicators of the parameters that appear in the horizontal
box below—Init Level, Attack Time, Decay Time, and so on.
If you click-hold on any numerical value, say the Init Level, and drag your mouse up/
down, the corresponding node will move in the Envelope window. Try this with each
numerical parameter to establish which node is which.
Now, move your mouse cursor over the edge of the node that controls the Init Level,
and you’ll see a pair of arrows. If you move your mouse cursor into the center of the
node, the arrows (and node) are filled. The arrows simply indicate the possible
directions that the node can be moved.
Experiment with each node/parameter in each of the Envelope window modes (see
just below) to get a hang of this. You’ll find that it’s very intuitive to use.
Reset
Here you can reset the currently selected envelope—Volume, Filter or Density—to
default values.

All
Here you can resets all envelopes to default values.

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Envelope Mode Buttons

Clicking on these buttons will switch to the selected envelope mode. The Envelope
window display will adjust accordingly, with the selected envelope being topped in the
superimposed display. The other envelope curves are shown as transparencies in the
background.
• The Volume Envelope is shown in red.
• The Filter Envelope is shown in yellow (the Filter needs to be switched on).
• The Density Envelope is shown in light blue (Synthesized IR only).
Note: You can also switch between envelope modes via the pull-down menu to the left
of the horizontal parameter panel below the Envelope window.
Please note that use of the word mode here is designed to simplify your
understanding. Obviously, the Space Designer displays, and uses, up to three discrete
envelopes simultaneously.
Reverse
A single click on this button reverses the Impulse Response together with its envelopes,
allowing easy creation of reversed reverb effects, and also some unusual sound
processing options. A second click undoes the reverse.

When you reverse the Impulse Response, you are effectively using the tail rather than
the front end of the sample. As such, you may need to use lower or even negative PreDelay values when reversing.
Zoom to Fit/A and D Buttons
The parameter name effectively describes what it does. It zooms in on the Impulse
Response waveform to make use of the entire Envelope window, in doing so the
display will follow any envelope length changes.

Zoom is directly tied to the current envelope mode, so the Zoom to Fit appearance will
be different dependent on the currently active envelope viewing mode.

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The A and D buttons alongside the Zoom to Fit parameter are for the Attack and Decay
portions of the (currently selected) envelope. The A and D buttons are only available to
the Volume and Filter Envelopes. Simply click on the appropriate button(s) to activate
the desired viewing mode.
The small Overview display indicates which portion of the IR file is currently visible in
the Envelope window.

Uncheck all buttons to return to the standard, non-zoomed view.
Zoom to Fit will only result in a visible change when the selected envelope is actually
smaller than the overall Envelope window display.
Volume Envelope

Init Level
This parameter sets the initial volume level (the start level) of the Impulse Response/
Reverb attack phase. It is expressed as a percentage of the full scale volume of the
Impulse Response file. The attack phase is (generally) the loudest point of the Impulse
Response.

The Init Level should be set to 100% to ensure maximum volume for the early
reflections, but obviously use your ears and level meters to ensure that signal levels
and audio quality are optimal.

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Adjusting this up or down can be done graphically in the Envelope window (the large
node on the left), or by click-holding and dragging the mouse cursor up/down on the
numerical entry.
Attack Time

This parameter alters the level of the Impulse Response attack phase over time. It is
expressed in second values, with the maximum possible Attack Time mirroring the
value set by the Length parameter (see “Length” on page 122).
The Attack Time parameter determines the length of time before the decay phase of
the Volume Envelope begins.
Adjusting this left or right can be done graphically in the Envelope window (the large
node at the bottom—on the center line), or by click-holding and dragging the mouse
cursor up/down on the numerical entry.
Note that the combined total of the Volume Attack and Decay Time parameters is equal
to the total length of the Impulse Response file (determined by the Length parameter,
see “Length” on page 122), unless the Decay is shortened.
Decay Time

This parameter alters the level of the Impulse Response decay phase over time. It is
expressed in second values, with the maximum possible Decay Time mirroring the value
set by the Length parameter (see “Length” on page 122).
Adjusting this left or right can be done graphically in the Envelope window (the large
node at the bottom right), or by click-holding and dragging the mouse cursor up/down
on the numerical entry.

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Volume Decay Mode
You have two options—Exp (Exponential) and Lin (Linear).

The output of the envelope generator is shaped by an exponential function during the
decay phase. If the decay phase is set as a straight line, the result is an exponential
function that describes a natural decay.
If set Lin, no shaping is performed.
An Exp curve will result in a much more natural sounding reverb tail.
Click on the desired button to activate the required mode.
End Level
This parameter is used to set the end volume level. It is expressed as a percentage of
the overall Volume Envelope—the default End Level value is 0.000%. At this value, the
reverb tail cuts off abruptly, which is great for gated reverb effects.

Adjusting this up or down can be done graphically in the Envelope window (the large
node at the bottom right), or by click-holding and dragging the mouse cursor up/down
on the numerical entry.
The Volume Attack and Decay Curve Form Nodes
On the curve displayed in the Envelope window, you will find two small nodes in the
attack portion of the overall Volume Envelope curve. A vertical line separates the attack
and decay portions of the envelope. There are also two small nodes in the decay
portion of the envelope—to the right of the vertical line. These are the Volume Attack/
Decay Curve Form node parameters.

The Curve Form nodes are tied to the envelope curve itself, so you can view them as
envelope handles if you wish. Moving the nodes vertically/horizontally will change the
shape of the envelope curve. To change the Curve Form node positions, simply clickhold and drag them.

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The curve shape can also be changed by clicking and dragging on the envelope curve
directly.
Filter Envelope

The entire Filter Envelope graphic, including nodes plus the Filter Envelope parameters
below. The central large node indicates/controls the Attack endpoint (and Decay
startpoint) and Break Level parameters simultaneously. The large node on the righthand edge controls the Decay endpoint and End Level parameters simultaneously. Also
note the smaller nodes in the attack and decay portions of the envelope. These are the
Curve Form node parameters that are used to create curves.
The Filter On switch must be active, or the Filter Envelope controls discussed below will
have no effect. The Filter Envelope only is available if the Filter is switched on.
Init Level
This parameter sets the initial cutoff frequency of the Filter Envelope in Hertz.

Obviously, you should use your ears and level meters to ensure that the filter (and
signal) levels, and desired tonal quality, are appropriate.
Adjusting this up or down can be done graphically in the Envelope window (the large
node on the left), or by click-holding and dragging the mouse cursor up/down on the
numerical entry.

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Attack Time
This parameter determines the time required to reach the Break Level (see below) value.
You may use the Filter Attack Curve Form node parameters to alter the shape of the Filter
Attack curve.

The combined total of the Filter Attack and Decay Time parameters is equal to the total
length of the (synthesized or sampled) Impulse Response (determined by the Length
parameter, see “Length” on page 122), unless the Decay time is reduced.
Break Level
This parameter determines the maximum filter cutoff frequency, the envelope reaches.
It also acts as the break point between the attack and decay phases of the overall filter
envelope. In other words, when this level has been reached after the Attack phase, the
Decay phase will begin.

Note: You can create interesting filter sweeps by setting the Break Level to a value lower
than that of the Init Level.
Decay Time
Determines the time required (after the Break Level point) to reach the End Level value.
You may use the Filter Decay Curve Form node parameters to alter the shape of the Filter
Decay curve.

The combined total of the Filter Attack and Decay Time parameters is equal to the total
length of the Impulse Response file (determined by the Length parameter, see “Length”
on page 122), unless the Decay time is reduced.

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End Level
This parameter is used to set the filter end cutoff frequency. It is expressed as a
percentage of the overall Filter envelope scale. If set to a value of 0, the filter stays
“open” for the remainder of the reverb signal (provided that the filter is configured as
“Low Pass” 6 dB or 12 dB).

The Filter Attack and Decay Curve Form Nodes
On the Attack curve displayed in the Envelope window, you will see two nodes in the
attack portion of the overall Filter envelope curve. The Break Level (a larger node)
separates the attack and decay portions of the envelope. There are also two nodes in
the decay portion of the envelope—to the right of the Break Level node. These are the
Filter Attack/Decay Curve Form node parameters.
Their use is identical to that of the Volume Envelope Curve Form nodes, but obviously
they are used to control the Filter Envelope. Please see the “The Volume Attack and
Decay Curve Form Nodes” section, on page 130, for further information.
Density Envelope

When synthesizing artificial Impulse Responses, Space Designer offers a facility that
approximates the sound of early reflection clusters as part of the overall reverb sound.
This is achieved through use of a ramp envelope to dynamically change the density of
the early reflection simulation.
The Density parameters can be used in conjunction with the Filter to fine-tune the
timbre of your reverbs.

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Please note that the Density Envelope is only available when in the Synthesized IR
mode.
Init Level
This parameter controls the density (the average number of reflections in a given
period of time) of the diffuse reverb. Both the Initial and End Density (Init and End Level)
can be controlled over time through use of the Density Envelope. Lowering the density
levels will result in audible reflections patterns and discreet echoes.

Adjusting this up or down can be done graphically in the Envelope window, or by clickholding and dragging the mouse cursor up/down on the numerical entry.
Ramp Time
This parameter adjusts the length of time elapsed between the Initial and End Density
levels. Adjusting this up or down can be done graphically in the Envelope window, or
by click-holding and dragging the mouse cursor up/down on the numerical entry.

End Level
As above, but controls the reverb tail. If you select an End Level (End Density) value that
is too low, the reverb tail will sound grainy. You may also find that the stereo spectrum
is affected by lower values. Adjusting this up or down can be done graphically in the
Envelope window, or by click-holding and dragging the mouse cursor up/down on the
numerical entry.

Reflection Shape
The range from 0 to 100 determines the steepness (shape) of the early reflection
clusters as they bounce off the walls, ceiling, and furnishings of a virtual space. A value
of 0 results in clusters with a sharp contour and a value of 100 results in an exponential
slope, and a smoother sound.

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This is handy when recreating rooms constructed of different materials. Use of this
parameter, in conjunction with suitable settings for the Envelopes, Density, and Early
Reflection will assist you in creating rooms of almost any shape and material.
Click-hold on the triangle, and slide left/right to adjust.

Filter Parameters
The filter section of the Space Designer provides control over the timbre of the reverb.
Not only can you select from several filter types, but you also have envelope control
over the filter cutoff, independent from the Volume Envelope. Changes to the filter
settings result in a recalculation of the IR, rather than a straight change to the sound as
it plays through the reverb.

Filter On/Off
Switches the filter section on and off. This must be switched on if you wish to make use
of any of the Filter and Filter Envelope controls.
Filter Mode
Switches between four modes. Click on the desired LP (lowpass) 6 dB and 12 dB, BP
(bandpass) or HP (highpass) value. To explain:
• 6 dB (LP)—Bright, good general purpose filter mode. It can be used to retain the top
end of most material, while still providing some filtering/control.
• 12 dB (LP)—Useful where you want a warmer sound, without drastic filter effects. It is
handy for smoothing out bright reverbs.
• BP—6 dB per octave design. Reduces the amount of signal that surrounds the mids
of the input material, leaving the frequencies around the cutoff frequency intact.
• HP—12 dB per octave/two-pole design. This filter reduces the level of frequencies
that fall below the cutoff frequency.
Reso
The Reso (resonance) parameter emphasizes frequencies above, around or below the
cutoff frequency—determined by the selected Filter mode. As you increase the Reso
value, the sound will loose bass and become thinner.
Note: Use the Reso control to add a basic, general color to your reverb, or for drastic
Filter effect sounds.

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Effect Parameters
Low Shelving EQ
As the name suggests, this EQ has a specific frequency that, once set, only allows
frequencies that fall below it to be affected. The Gain control determines the amount of
cut or boost of frequencies below the value set with the Freq parameter (expressed in
Hertz). To adjust these values, click-hold on the applicable parameter field or knob and
drag the mouse up and down.

Stereo Spread
This facility applies additional stereo information to a Synthesized IR.

It has no impact when using an IR sample or when using the Space Designer as a
mono plug-in.
Spread extends the stereo base to frequencies that fall below the frequency
determined by the Xover (crossover) parameter.
At a Spread value of 0, no stereo information is added (although the inherent stereo
information of a stereo signal and reverb will be retained). At a value of 100 the left and
right channel divergence is at its maximum.
The Xover parameter is set in Hertz. Any synthesized IR that falls below this threshold
value will be processed by adjustments over 0 for the Spread parameter.
The effect enhances the perceived width of the signal, without losing the directional
information of the input signal, normally found in the higher frequency range. Low
frequencies are spread to the sides, reducing the amount of low frequency content in
the center—allowing the reverb to nicely wrap around the mix.

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Creating Impulse Responses
This section discusses briefly the different methods for creating your own Impulse
Response files for use with Space Designer.

About Impulse Responses
Impulse Responses are recordings (stored as AIFF, SDII or WAV files) made in acoustic
spaces. To create an impulse response the sound of a starter pistol, digital spike or sine
sweep is recorded inside the desired room together with the resulting reflections.
Starter Pistols
Starter pistol shots are quite loud allowing a good signal to noise ratio to be recorded.
However, a perfect, undistorted recording of a starter pistol shot is difficult to produce.
In addition starter pistol shots contain very little high or bass frequency information.
Digital Spikes
A digital spike contains a wide frequency spectrum. You can create a spike by using the
Pencil tool in Logic’s Sample Edit window to draw a single sample in a silent sound file
(a recording of silence). The catch with this method is that, although fast, the output
level is often too low to be usable. You can, however, boost the signal in Logic.
Sweeps
The downside with starter pistols or digital spikes is that they release their (high level)
energy in a very short time period. As an alternative, you can playback, and record a
broadband audio sweep with optimal recording levels. For this a sine sweep covering
the whole audible frequency range could be used.
The recording of a sine sweep that is played back in an acoustic environment is known
as a sine sweep response, rather than an impulse response.
By means of a deconvolution process the sine sweep is removed from the sine sweep
response recording, leaving you with an impulse response file that can be loaded into
Space Designer. This method of creating impulse responses often results in a superior
reverb sound quality.

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Using the Deconvolution Facility
Now that you know about sine sweep responses you’d probably like to know how to
use it to roll your own. It couldn’t be easier.
Accessing the Deconvolution Facility

The Deconvolution facility is necessary only in combination with a sweep response
recording.
To access the Deconvolution facility:
1 Bypass the plug-in by clicking on the Bypass button at the top of the of the Plug-in
window. This conserves CPU resources.
2 Click on the Deconvolution button towards the top right of the Space Designer GUI.
This will launch a File Selection box.
3 The File Selector menu bar will show Load Coded Impulse Response for Deconvolution.
Click Choose and select the wet, reverberated sweep response recording.
4 The File Selector menu bar will change to display Load Testsignal. Click Choose. Now
open the same dry sweep audio file you used as a source for the wet response
recording.
5 You will be prompted to save the decoded audio file. Select the desired path and file
name. Now the new IR will be calculated.
6 Any silence at the end and beginning of the new IR file should be removed using a
sample editor.
7 This file can now be loaded into the Space Designer.

About Convolution
This section looks at the technicalities behind convolution (in a reverb sense).
The use of reverberation on dry sounds is commonplace in computer music
production. Reverbs (and delays) can be used to simulate room and other acoustic
spaces or entirely new sounds, which may (or may not) be related to an actual physical
space.
There are several methods used to simulate and model different reverbs and physical
environments. One technique involves the actual recording of the ambience of a room,
and overlaying this recording (the impulse) onto another sound. This technique is
called convolution.
Before we get into convolution, a few basics need to be covered on reverberation itself.

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So, Just What Is Reverberation?
Put simply, reverberation is basically a delay of the source signal x number of times (by
a very small time value), which is then fed back onto itself, simulating the way sound
bounces around a room.
Prior to the invention of digital systems, engineers used a variety of techniques to
create reverb-type effects. As examples:
Reverberant chambers—sending sound via a loudspeaker into a tiled room and rerecording the reverberant sound—ever recorded a vocal in the toilet?
Reverb units with springs and metal plates hung from the walls—with the latter being
close and distant-miked then re-recorded alongside the original signal.
There are some essential elements for creating a reverberant sound, with the key thing
to remember being that reverb is simply a number of (delayed) copies of the original
signal—which emulates the multiple versions of a sound bouncing around a room.
In a software-based reverb plug-in there may be one copy of the signal dedicated to
making the first reflection: the very first reflected sound we hear when a signal is
introduced into a reverberant space. Other algorithms within the plug-in may be
dedicated to early reflections—the first sounds we hear after the initial reflection. The
(delayed) reflections are fed back, and added, to the original signal at a lower level. This
creates an effect known as comb filtering (a short feedback delay that emphasizes
specific harmonics).
Following on from the delayed signals (first and early reflections) you’ll generally find
that filters are used to make the reverb tails sound as if they are far away. Filtration of
higher frequencies—say 5 kHz and up—makes a signal sound farther away. This is
because high frequencies have very little energy, and don’t travel far. Filtration may also
be used to soften any harmonics introduced by the comb filtering mentioned above.
Needless to say, emphasis of particular harmonics can make a reverb sound very
artificial, which may be just what you’re after … but not if you’re simulating a room!
Note that not all reverb units/plug-ins offer this sort of filtration. Another approach is to
both feed back and feed-forward the same signal simultaneously, phase inverted, to
create an all-pass filter. With this type of filter/delay the comb filtering effect doesn’t
occur (because of the phase inversion of the feed-forward signal—this fills in the
missing frequencies created by the comb filter of the feedback signal). This creates a
blended reverb.

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Typical reverb algorithms have parameters for:
• room size (church, club, closet, bathroom, and so on),
• brightness (hard walls, soft walls or curtains and so on) and
• feedback or absorption coefficient (are there people, carpets, and so on in the
room?—how quickly does the sound die?).
• Upmarket reverbs may also contain several filter parameters.
When trying to simulate rooms, take the following into consideration:
• How large is the room? How long will it take for the first reflection to get back to
your ears?
• What are the surfaces like in the room? What material are they made of? Not just the
walls, ceiling, and floor, but are there objects in the room that can reflect sound
waves? As examples: plastic chairs, tables, and so on.
• Are there surfaces in the room that can absorb sound? As examples: curtains, soft
furnishings such as lounges. Even people in the room can also remove energy from
reflected sounds!

Convolving Reverb
The Space Designer is, as you know, a convolving reverb. So, just how does convolution
work? You’ve already learnt how to use the effect parameters, and how to record (or
create) an impulse, and use the impulse response, but now we’re going to look at the
maths behind the process.
Convolution is a fairly complicated software process. It takes each sample in the
impulse response file (of a given room) and multiplies that sample by each sample in
the sound file that we want to place in the room. So each sample input sound file, such
as a vocal sound file that you want to add reverb to, is multiplied by each sample in the
impulse response file.
As an example, imagine a three second impulse file, and a one minute sound file that
you want to add the reverberant characteristics of some space to. At 44.1 kHz, that’s:
3 (sec. − IR) × 44,100 (Samples) × 60 (sec.—audio) × 44,100 (Samples)
= 180 × 680,683,500,000
= 122,523,403,030,000 (Wow!)
This multiplication of each point in each function by every other point in the other
function (in the time domain)—called a cross multiply—produces what we refer to as
the convolution in the frequency domain.
As we’re sure that few of you have applied mathematics degrees, we’ll just say that
there’s a whole lot of computation going on! This computationally intensive (and
remember that we only looked at a 60 second sound!) has not been widely adopted for
the simple reason that the garden variety computer just simply wasn’t fast enough to
do it—until now, that is.

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The reason why we can actually perform the necessary calculations in real-time these
days is due to a mathematical operation known as the FFT (Fast Fourier Transform). For
filters (and we use the term filters here as this is what is effectively happening—the
input sound is being filtered by the impulse response) with lots of nonzero values, it is
easier to compute the convolution in the spectral domain. Here’s how:
If we want to convolve our music function (the sound you wish to process) against our
filter function (the impulse response), this results in another sound (the convolved
results). This convolved sound has a spectrum that is equal to the product of the
spectrum of the music function and the filter function.
Put another way, the Fourier coefficients of the convolution can be computed by
simply multiplying together each of the Fourier coefficients of the music and filter
functions.

The End Result?
Knowing the ins and outs (mathematics) behind convolution doesn’t really matter too
much. The important thing is to have a large library of great impulse responses—the
best sounding cathedrals, recording studios, concert halls, railway tunnels, electronic
reverb units or even resonance bodies of instruments … you name it!—and you can
simulate any space for any sound.
Thankfully, we’ve created and included a number of Impulse Responses to get you
started, and you may find that they’re all you’ll ever need.
If you wish to create your own, the Internet is a great place to share them with other
Space Designer users from around the globe.
Whatever way you decide to go, either using the factory IRs, downloading another
user’s efforts or rolling your own, Space Designer makes it easy to get that perfect
reverb sound.

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Special

This chapter introduces Logic’s special plug-ins.
This includes the Spectral Gate, Pitch Shifter II, SubBass, Denoiser, Exciter, and Stereo
Spread plug-ins, amongst others.

Spectral Gate

The Spectral Gate separates the signals above and below the Threshold level into
frequency ranges that can be independently modulated. It does this via a Fast Fourier
Transformation (FFT) of the entire signal. Technical jargon aside, the Spectral Gate is a
tool that lets you come up with some pretty wacky filtering effects.
The frequency range that you wish to process is defined by the Center Freq. and
Bandwidth knobs. This frequency band is separated by steep slopes. Within this band,
you can use the Threshold slider to determine a level that separates the frequencies
above and below it. The frequencies above the Threshold are made audible with the
Super Energy knob, the frequencies below it with Sub Energy. The original signal, outside
the defined frequency band, can also be added to the mix: Low Level blends in the
frequencies that lie below the frequency band (bass frequencies,) and High Level, the
higher frequencies that lie above the defined frequency band.

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The actual frequency band can be modulated by three parameters: Speed determines
the modulation frequency, CF Mod. (Center Frequency Modulation) defines the
intensity of the center frequency modulation, and BW Mod. (Band Width Modulation)
controls the bandwidth modulation.
The Gain slider lets you adjust the level of the generated effects signal.
We suggest you use a drumloop when you begin experimenting with this plug-in. Set
Center Freq. to its minimum, and Bandwidth to its maximum, values (the entire
frequency range is processed at these settings). Turn up the Super Energy or Sub Energy
knobs—one at a time—and experiment with the Threshold knob. You’ll soon get a feel
for how different Threshold levels affect the sound of Super Energy and Sub Energy.
When you’ve come across a sound that you consider particularly nifty (or even just
useful), you can narrow the Bandwidth drastically, gradually increase the Center Freq.,
and use the Low Level and High Level sliders to mix in some treble and bass from the
original signal. At lower Speed settings, turn up the CF Mod. or BW Mod. knobs.

Pitch Shifter II

The Pitch Shifter II takes a minimalist approach—with just a few parameters available in
the Editor view.
Semi Tones is used to set the transposition value—in semi-tone increments, within a
range of one octave upwards or downwards. Cents controls detuning in increments
equivalent to 1/100th of a semi-tone step. Use the Mix slider to control the desired
balance between the original and processed signals.
The Drums, Speech, and Vocals buttons allow you to choose between three presets that
optimize the Pitch Shifter II to deliver the best results for different audio material.
• When you select Drums, the groove of the original track remains intact.
• With Vocals, the intonation of the original is retained unaltered. Hence Vocals is wellsuited for any signals that are inherently harmonic or melodious, such as string pads.
• Speech mode is a compromise between the two—the program attempts to retain
both the rhythmic and harmonic aspects of the signal, which is desirable for complex
signals, such as spoken-word recordings or rap music. Speech is thus also suitable for
other hybrid signals, such as rhythm guitar.

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Note: When in doubt, Speech is a good place to start. A/B the options to compare
them, and use the one that suits a given recording best. When auditioning and judging
settings for quality, it’s a good idea to temporarily turn the Mix knob up to 100%. Keep
in mind that Pitch Shifter II artefacts are a lot harder to hear when you mix a smaller
percentage of a transposed audio to the overall signal.
In the Pitch Shifter II’s Controls View you can create your own presets, using the Delay
and Crossfade parameters. These parameters are only effective when you select the
Manual option in the Timing menu. You can also select the Auto option here—the Pitch
Shifter will then automatically create presets by analyzing the incoming signal. The
Stereo Link parameter allows you to invert the stereo channel’s signals, with processing
for the right channel occurring on the left and vice versa.

Vocal Transformer

The Vocal Transformer plug-in allows you to manipulate vocal tracks in many different
ways. If you want to transpose the pitch of a vocal line, to augment or diminish the
range of the melody, or even reduce it to a single note to mirror the pitches of a
melody—the Vocal Transformer will be your plug-in of choice. No matter how you
change the pitches of the melody, formants remain the same. You can shift the formants independently, which means that you can turn a vocal track into a “Mickey
Mouse” voice, while maintaining the original pitch.
The Vocal Transformer is well suited to extreme vocal effects. The best results are achieved with monophonic signals, including monophonic instrument tracks. The plug-in is
not designed for polyphonic voices (a choir on a single track, for example) or other
“chordal” tracks.
The Vocal Transformer operates in two modes: Robotize on or off.
The Robotize function is used for augmenting, diminishing or mirroring the melody.
Let’s first have a look at how the Vocal Transformer operates when Robotize is switched
off.

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Pitch, Formant, and Mix
Mix defines the level ratio between the original (dry) and effect signals.
Note: In order to get an idea of what the plug-in does, it may be helpful to listen to the
original dry signal in the background. A Mix setting of about 75% should suffice.
The Pitch parameter transposes the pitch of the signal (up to) two octaves upwards or
downwards. Adjustments are made in semitone steps. Incoming pitches are indicated
by a vertical line.
Note: Transpositions of a fifth upward (Pitch = +7), a fourth downward (Pitch = −5) or
by an octave (Pitch = ±12) are the most useful, harmonically.
As you alter the Pitch parameter, you might notice that the formants don’t change.
Note: Formants are characteristic emphases of certain frequency ranges. They are static, and do not change with pitch. Formants are responsible for the specific timbre of a
given human voice.
The Pitch parameter is expressly used to change the pitch of a voice, not its character. If
you set negative Pitch values for a female soprano voice, you can turn it into an alto
voice, without changing the specific character of the singer’s voice.
The Formant parameter shifts the formants, while maintaining the pitch or while independently altering the pitch. If you set this parameter to positive values, the singer
sounds like Mickey Mouse. By altering the parameter downwards, you can achieve
sound effects reminiscent of Darth Vader from Star Wars.
Note: If you set Pitch to 0 semitones, Mix to 50% and Formant to +1 (while Robotize
remains switched off ), you can effectively place a singer (with a smaller head) next to
the original singer. Both will sing with the same voice—in a choir of two. This choir
effect is quite effective, and is easily controlled with the Mix parameter.

Robotize
If you switch Robotize on, the Vocal Transformer can augment or diminish the melody.
You can control the intensity of this distortion with the Tracking parameter.
Note: The four −1, 0, 1, and 2 switches set the Tracking slider to values of −100%, 0%,
100% and 200%. These switches aren’t individual parameters. They are merely additional controls that make faster to set the Tracking parameter to its most useful settings.
• At a value of 100% (switch 1), the range of the melody is maintained. Higher values

augment, and lower values diminish the melody.
• At a setting of 200% (switch 2) the intervals are doubled.
• The reduction to 0% (switch 0) delivers interesting results, with every syllable of the
vocal track being sung at the same pitch. Low values turn sung lines into spoken language.

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• At a setting of −100% (switch −1), all intervals are mirrored.

The Pitch Base parameter is used to transpose the note that the Tracking parameter is
following. As an example: the note which is spoken, if Tracking is set to 0%.

Pitch Correction

The Pitch Correction plug-in enables you to correct the pitch of audio tracks. Improper
intonation is a common problem with vocal tracks, and this can be easily fixed with the
Pitch Correction plug-in. The sonic artefacts of the process are minimal and can barely
be heard, as long as the corrections are moderate. The natural articulation of the performance is preserved. Should large transposition intervals be used, you can achieve a
static pitch, thus creating an interesting effect, known as the “Cher” effect. Any scale
can be defined as a pitch quantization grid. Improperly intonated notes will be corrected to this grid. The keyboard at the center of the Plug-in window is used to define the
scale that will serve as the pitch quantization grid.
Note: Polyphonic Recordings (choirs on one track) and highly percussive signals, with
prominent noisy portions, can’t be corrected to a specific pitch. Despite this, feel free to
try the plug-in on drum tracks!

Function Principle
The Pitch Correction algorithm is designed for corrections of smaller pitch shifting
intervals. The correction works in a similar fashion to an audio tape, with the playback
speed accelerated and slowed down in a way that ensures the singing voice always
matches the correct note pitch. If you force the algorithm to correct larger intervals,
you can create special effects.

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Range
The Normal/Low parameter determines the pitch range that you wish to search for
notes that need correction. Simply select Normal or Low for the track. Normal is the
default range, and works for most audio material. Low should only be used for audio
material that contains extremely low frequencies (below 100 Hz) which may cause the
pitch dectection to work incorrectly. The parameter does not affect the sound. It is designed to optimize tracking in the target pitch range.
Keyboard
The central element of the plug-in is the 12-note Keyboard. The Keyboard is used to
exclude notes from the pitch quantization grid. When you first open the plug-in, all
notes of the chromatic scale are selected. This means that every incoming note will be
altered to fit the next semitone step of the chromatic scale. If the intonation of the
singer is poor, this might lead to notes being incorrectly identified, and corrected. As an
example, the singer might have intended to sing an e, but the note is actually closer to
a d#. If you don’t want the d# in the song, the d# key can be disabled on the Keyboard.
Given that the original pitch was sung closer to an e than to a d, it will be corrected so
that it becomes an e.
Note: The settings are valid for all octave ranges. Individual settings for different octaves aren’t provided.
Show Input/Output
These two buttons below the Keyboard will display the pitch of the input or output
signal, respectively, on the notes of the keyboard. This provides a real time, at-a-glance
overview of either the original or corrected signal.
Programmable Scales (Root and Scale)
Clicking the Scale field allows you to select different pitch quantization grids. The scale
that is set manually (with the Keyboard) is called the User Scale. The default setting is
the Chromatic scale. The other scale names are self-explanatory. If you’re unsure of the
intervals used in any given scale, simply select it in the Scale field and check out the
values shown in the Keyboard grid. You can alter any note in the scale by clicking on
the Keyboard keys. Any such adjustments will overwrite the existing User Scale settings.
Note: There is only one User Scale per song. You can, however, create multiple User Scales, and save them as Pitch Correction plug-in Settings files.
The Drone scale uses a fifth as a quantization grid, the Single scale defines a single note.
Both of these scales aren’t meant to result in realistic singing voices, so if you’re after
interesting effects, you should give both of them a try.
Root allows you to select the root note of the scale (with the exception of the User and
Chromatic Scale, where there is no root note (none)). You may freely transpose the
major and minor scales as well as the scales named after chords.

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Bypass
Use of the small bypass buttons (byp) above the green (black) and below the blue
(white) keys excludes notes from correction. This is useful for blue notes. Blue notes are
notes that slide between pitches, making the major and minor status of the keys difficult to identify. As you may know, one of the major differences between C minor and C
major is the e flat and the b flat, instead of the e and the b. Blues singers glide between
these notes, creating an uncertainty or tension between the scales. Use of the bypass
switches allows you to exclude particular keys from changes, leaving them as they
were.
Bypass All
With Bypass All active, the input signal is passed through unprocessed and uncorrected.
The Bypass All function is useful for spot corrections to pitch through use of Logic’s
automation system. Bypass All is optimized for seamless bypass switching in all
situations.
Note: You’ll often find that it’s best to only correct notes with the most harmonic gravity. As an example, select a Scale of sus 4 and the Root note of the song’s key. This will
limit correction to the root note, the fourth and the fifth of the key scale. Switch all
other notes to Bypass, and only the most important and sensitive notes will be corrected, while all other singing remains untouched.
Reference Tuning
The File > Song Settings > Tuning settings determine the tuning reference for all software instruments. If you engage Use Global Tuning in the Pitch Correction Plug-in window, the Global Tuning settings will be used for the pitch correction process. If this
parameter is switched off, you can use this section to freely set the desired reference
tuning in cents.
Note: Tunings that differ from software instrument tuning can be interesting, when you
want to individually correct the notes of singers in a choir. If all voices were individually
and perfectly corrected to the same pitch, the choir effect would be partially lost. You
can avoid this by (de)tuning the pitch corrections individually.
Response
Response determines how quickly the voice reaches the corrected destination pitch.
Singers use portamenti and other gliding techniques. If you choose a Response that’s
too high, seamless portamenti turn into semitone-stepped glissandi, but the intonation will be perfect. If the Response is too slow, the pitch of the output signal won’t
change quickly enough. The Response of pitch changes is indicated in milliseconds.
The optimum setting for this parameter depends on the singing style, tempo, vibrato,
and quality of the original performance.

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Correction Amount
The amount of pitch change is indicated in the horizontal bar display. If you keep a
close eye on this display, you can use it for two important tasks: To better understand
the inner workings of the algorithm works, and adjust the Response accordingly. You
can also use the display when discussing (and optimizing) the vocal intonation with a
singer during a recording session.
Automation
As with almost all Logic plug-ins, Pitch Correction can be fully automated. This means
that you can automate the Scale and Root parameters to follow the harmonies of the
song. Simply select the Touch or Write automation modes for the track, and change the
Scale and Root parameters by key during playback. Depending on the quality of the original intonation, setting the key Scale might suffice. Weaker intonations might need
more significant changes to the Scale and Root parameters.
Note: The keys of the scale can’t be switched by MIDI notes.

SubBass

This plug-in generates frequencies below those in the original signal—in other words
an artificial bass.
Warning: This process is known as a loudspeaker killer! Choose moderate monitoring
levels, and never try to play back sub-bass frequencies with loudspeakers which aren’t
capable of doing so. Never try to force a loudspeaker to output these frequency bands
with an EQ.

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The simplest application of the SubBass plug-in is as an octave divider, similar to
“Octaver” effects pedals for electric bass guitar. A simple frequency division circuit in
such pedals requires a monophonic input sound source, with a clearly defined pitch.
This type of device is only capable of producing an output signal which is one or two
octaves lower than the input signal. An octave is a frequency division by two. A ratio of
four means two octaves, and a ratio of eight equals three octaves.
As opposed to a pitch shifter, the waveform of the signal synthesized by the SubBass
plug-in has nothing to do with the waveform of the input signal. It’s shape is
sinusoidal—but a pure sine wave is rarely achievable in complex arrangements. It is
mixed together with the original signal. The mix ratio is defined by the Mix parameter.
SubBass creates two bass signals, derived from two freely definable portions of the
incoming audio signal. Therefore, you are not limited to monophonic signals with a
defined pitch, but complex summed signals may be processed as well. Center High and
Center Low define the center frequency of the band that the transposed signal will be
derived from. Bandwidth sets the bandwidth of the frequency band.
Within these frequency bands, the filtered signal should have a reasonably stable pitch,
in order to be analyzed correctly. The graphic shows the frequency bands of a typical
boom box, which transposes two frequency bands, with the width of a fifth, by one
octave each. If you set the analyzed frequencies a little higher, the SubBass plug-in
plays these frequencies, much like a bass guitar player doubling the lower notes of a
guitar player.
In real life, narrow bandwidths lead to the best results as they avoid unwanted
intermodulations. Set Center High a fifth higher than Center Low, which means a factor
of 1.5 for the Center Frequency. Derive the sub-bass to be synthesized from the existing
bass portion of the signal, and transpose by only one octave in both bands (Ratio = 2).
Do not overdrive the process, or you will introduce distortion. If you recognize
frequency gaps, move the Center Frequencies, or widen the Bandwidth a little.
Be prudent when using the SubBass plug-in, and compare the extreme low frequency
content of your mixes with other productions.

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Denoiser

The Denoiser eliminates or reduces almost any kind of noise floor.

Denoiser Parameters
Threshold
The value of this parameter determines how high you think the noise floor of the
material is.
Tip: Find a passage where only noise can be heard in isolation, and set the Threshold
value so that only signals of this volume will be filtered out.
Reduce
Reduce determines the level that the noise floor should be reduced to. A CD
theoretically has a maximum signal to noise ratio of 96 dB. Each 6 dB reduction is
equivalent to halving the volume level (a 6 dB increase equals a doubling of the
volume level).
If the noise floor of your recording is very high (on recordings from cassette—more
than −68 dB), you should be content with reductions of 83 to 78 dB, provided that
there aren’t any plainly audible side effects. After all, you have reduced the noise by
more than 10 dB, which is less than half of the original volume.
Noise Type
This value effectively states what type of noise you think the material contains:
• A value of 0 means white noise (equal frequency distribution);
• Positive values change the noise type to pink noise (harmonic noise; greater bass
response),
• Negative values change the noise type to blue noise (Hiss—tape noise).
Smoothing
The Denoiser uses FFT (Fast Fourier Transform) analysis to recognize frequency bands of
a lower volume and less complex harmonic structure, and then reduces them to the
desired dB value. In principle, this method is never exact, as neighboring frequencies
will also be affected.

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If you use the Denoiser too aggressively, the algorithm will produce artefacts, such as
“glass noise” which—in most cases—are less desirable than the existing noise.
Therefore, there are three parameters for reducing this effect in all three dimensions of
sound:
• Time Smoothing
This is the simplest form of smoothing. This parameter sets the time required by the
Denoiser to reach (or release) maximum reduction.
• Frequency Smoothing
This parameter sets a factor for smoother transition of the denoising to neighboring
frequencies. More precisely: If the Denoiser recognizes that only noise is present in a
certain frequency band, the higher the Frequency Smoothing parameter is set, the
more it will also change the neighboring frequency bands to avoid “glass noise”.
• Transition Smoothing
This parameter sets a factor for smoother transition of the denoising to neighboring
volume levels. More precisely, if the Denoiser recognizes that only noise is present in
a certain volume range, the higher the Transition Smoothing parameter is set, the
more it will also change similar level values to avoid “glass noise”.
The Graphic Display
The graphic displays how the lowest volume levels of your audio material (which
ideally is only noise) will be reduced.

Exciter

An Exciter generates high frequency components that are not part of the input signal.
It’s a nonlinear distortion process, that resembles overdrive and distortion effects. As
opposed to these processes, however, the distorted signal is mixed with the dry signal.
In addition, the harmonics generator is fed by a highpass filtered version of the input
signal. This means that the artificial harmonics have frequencies at least one octave
above the lowest frequency being allowed through by the highpass filter. Human
hearing can not easily distinguish artificial harmonics in very high frequency ranges.

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The cutoff frequency parameter of the high pass filter is called Frequency. The graphic
displays the frequency range that is used as the source signal for the process.
Harmonics controls the level of the effect signal mixed to the original signal. If you
disable Input, the original signal won’t be fed through. This should be your approach
when using the Exciter plug-in on a Bus channel, being fed by Sends from several
channel strips simultaneously, or should you wish to listen to the soloed effect signal.
The lower you tune the filter, the more harmonics the effect will produce, the lower the
frequencies of the artificial harmonics will be, and the less natural they will sound.
You can choose between two sound characteristics for the distortion. Color 1 is less
dense in spectrum than Color 2. The effect of Color 2 is more intense, but it also
introduces more unwanted intermodulation distortion.
Basically, higher settings for Frequency and Harmonics are preferable, because we
cannot easily distinguish between artificial and original high frequencies. Exciters add
life to digital recordings. They are especially well suited to audio tracks where the treble
is weak. Exciters are also popular for guitar tracks.

Stereo Spread

The Stereo Spread plug-in is a standard effect for mastering. There are several ways to
extend the stereo base, including complex algorithms of the Ghetto blaster type
(“wide” or “space”), or tampering with the signal’s phase. They all sound great, but most
of them could be very dangerous to your mix, ruining the transient response, for
example.
The Stereo Spread follows a sure and simple method: it extends the stereo base by
alternately distributing a (selectable) number of frequency bands from the middle
frequency range to the left and right channels. This greatly increases the perception of
stereo width, especially when applying a stereo effect to monaural recordings—
without making them sound totally alien.

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Parameters of the Stereo Spread
Order
The Order knob determines the number of frequency bands that the signal is to be
divided into.
Upper Int.
This parameter controls the intensity of the base extension of the upper frequency
bands.
Lower Int.
This parameter controls the intensity of the base extension of the lower frequency
bands.
Note that
• the stereo effect shows up mainly in the middle and higher frequencies, and
• if very low frequencies are distributed between the left and right speakers, the
energy distribution for both speakers will be significantly reduced.
Therefore, it is always best to select a lower Intensity setting for the lower frequency
bands, and to avoid setting the Lower Freq. below 300 Hz.
Upper Freq.
This sets the upper limit of the highest frequency band to be distributed in the stereo
image.
Lower Freq.
This sets the lower limit of the lowest frequency band to be distributed in the stereo
image (see the note in the Lower Int. section).
The Graphic Display
The graphic display shows the selected order (how many bands the signal is divided
into), and how strong the Stereo Spread effect will be in the upper and lower
frequency bands.
The upper section represents the left channel, the lower section the right channel, and
the frequency scale shows the frequencies in ascending order from left to right.

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Helper

This chapter introduces you to Logic’s Helper plug-ins.
This includes Test Oscillator, Tuner, Gain, I/O, Direction Mixer, MultiMeter, Levelmeter,
and Correlation Meter.

Test Oscillator

The Test Oscillator generates a static frequency or a sine sweep. The latter is a userdefined frequency spectrum tone sweep that can be used for the creation of Impulse
Responses, for use with Logic’s Space Designer (see “Convolution Reverb:
Space Designer”, on page 117). Here’s a quick run-down on the parameters:
The Waveform radio buttons select the type of waveform to be used for test tone
generation. Selection of any button will generate a fixed oscillation with the selected
waveform. Some further information about the Waveform section:
• the square and needle waveforms are available as either aliased or anti aliased
versions, when pressed in conjunction with the anti aliased button between them.
• needle is a single needle impulse waveform.
• if the Sine Sweep button is active, the fixed oscillator settings in the Waveform section
above are disabled.
Frequency determines the frequency of the oscillator. Level determines the overall
output level of the Test Oscillator.

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The Sine Sweep section generates a user-defined frequency spectrum sine wave sweep.
The Time field determines the duration of the sweep. The Start and End Freq(uency)
parameters define the oscillator frequency at the beginning and end of the sine sweep.
The Trigger button behavior can be switched via the edit field below:
• single: pressing the x1 symbol triggers the sweep once.
• continuous: pressing the Infinity symbol triggers the sweep indefinitely.

Tuner
The ET1 Tuner plug-in can be used to tune acoustic instruments. This ensures that
software instruments, existing samples or recordings are perfectly tuned to any new
acoustic recordings you may make.
You would normally insert the ET1 Tuner into an Input fader channel.

Use couldn’t be simpler. There is a single tuning control at the bottom of the ET1 Tuner
interface. To adjust, simply click and drag it to the desired pitch for A. By default, the
ET1 Tuner is set to concert pitch A = 440 Hz.
The Keynote and Octave panels display the incoming note pitch, and the octave that
the incoming note belongs to. This matches the MIDI octave range, with the C above
middle C displayed as C4, and middle C displayed as C3.
The numeric semicircle around the top of the ET1 interface displays the amount of
variance—in cents—from the original pitch. The range is displayed in single semitone
steps ±6 cents, and then in larger increments to a maximum of ±50 cents.
If the incoming note is slightly flat, the indicator segments to the left will be
illuminated. If the incoming note is slightly sharp, the indicator segments to the right
will be illuminated.
When the pitch is perfect, the center segment is lit.

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Adjust the pitch of your instrument—using the tuning nuts on your guitar, for
example—until the center segment (at the very top of the ET1) is illuminated. This
indicates that the incoming note/string pitch is perfectly tuned.

Gain

This plug-in allows a constant amplification or reduction, by a specific decibel amount,
of an Audio Track or Bus Object. It is ideal for use in situations where you’re working
with automated tracks during post-processing, and you want to quickly adjust master
levels. This could be the case when you’ve inserted an additional plug-in that doesn’t
feature a dedicated gain control, or you want to change the basic level of a track for a
remix version.
Note: The Gain plug-in replaces the previous Volume and Gainer plug-ins.
• The old Gainer plug-in will remain in Logic to retain compatibility with older songs,

but it is no longer directly accessible from the plug-in menu.
• Automation data is upwardly compatible for any existing Gainer instances.
• Any Volume plug-in instances in older songs will automatically be replaced by the

Gain plug-in. Any existing Volume plug-in automation data will be understood and
used by the Gain plug-in.
• Similarly, the Settings files used by the Volume and Gainer plug-ins can be read by
the Gain plug-in. If such Settings are used in a loaded song file, the Gain plug-in will
replace the Volume plug-in, and equivalent parameters will be set.

Parameters
The following parameters are available in the Gain plug-in:
Gain
This control adjusts levels from −96 to +24 dB, in steps of 0.1 dB. Press Shift while
dragging on the Gain parameter to adjust in fine increments.

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Phase Invert
These buttons invert the phase of the left and right channels. This allows you to
combat time alignment problems, particularly those caused by running multiple
microphones at the same time. When you invert the phase of a signal, it sounds
identical to the original. Only when the signal is heard in conjunction with other signals
does phase inversion have an audible effect. As an example, if you mic a snare drum
from the bottom as well as from the top, you should invert the phase of the bottom
microphone’s signal so that it is in phase with the top mic signal.
Stereo Balance
The Stereo Balance control distributes the incoming signal between the left and right
channels.
Swap Left/Right
This button swaps the left and right output channels. It is placed after the Stereo
Balance in the signal path.
Mono
Activation of the Mono button outputs the summed mono signal on both the left and
right channels.
Note: The Gain plug-in is available in m → m, m → s and s → s configurations.
In m → m and m → s modes, only one Phase Invert button is available. In the m → m
version, the Stereo Balance, Swap Left/Right and Mono parameters are disabled.

I/O
The I/O insert plug-in allows you to insert external effect processors into the Logic
mixer.
This only makes sense with audio cards that have more than two outputs, thus
providing discrete inputs and outputs (analog or digital) for sending signals to/from
the external effect processor.
The I/O plug-in offers assignment pull-down menus by which it can access the input/
outputs provided by your audio hardware. In and output levels can be adjusted with
the respective volume sliders.
• Output—assigns the output (or output pair) of the plug-in.
• Output Volume—adjusts the volume of the output signal.
• Input—assigns the input (or input pair) of the plug-in
• Input Volume—adjusts the volume of the input signal.

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Direction Mixer

The Direction Mixer plug-in offers the following features:
• MS Decoder
• The option of influencing the stereo base
• Variable pan positioning of a stereo recording

Parameters of the Direction Mixer Plug-in
Input
The LR and MS radio buttons determine whether the input signal is a standard left/right
signal, or if you’re dealing with an MS encoded (middle and side) signal, for example
when the two sides of an MS stereo mic setup were recorded directly.
Direction
This parameter determines the direction from which the middle of the recorded stereo
signal will emanate from within the mix, or in less complicated terms, its pan position.
At a value of 0, the middle of the stereo recording will be dead center within the mix.
Positive values shift the middle of the stereo recording towards the left, negative values
towards the right. At 90˚, the middle of the stereo recording is panned hard left, at −
90˚, hard right. Higher values shift the recorded signal back to the center of the stereo
mix, except that the stereo sides of the recording are swapped. At values of 180˚ or −
180˚, the middle of the recording is yet again dead center of the mix, although the left
side of the recording is audible on the right side of the mix, and vice versa.
Basis
This parameter determines the spread of the stereo base. At a neutral value of 1, the
left side of the signal is positioned precisely on the left, and the right side precisely on
the right. As the values decrease, the two sides increasingly move towards the center of
the stereo image. A value of 0 produces a mono signal (both sides of the input signal
are routed to the two outputs at the same level—a true middle signal). At values
greater than 1, the stereo base is extended out to an imaginary point beyond the
spatial limits of the speakers. In terms of MS levels, this is an involved way of saying
that the level of the side signal is increased so that it is higher than the level of the
middle signal. At a value of 2, only the side signal remains audible (on the left you’ll
hear L-R and on the right R-L).

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If you chose to use the Direction Mixer simply to spread the stereo base, please keep in
mind that as the Basis values increase beyond 1, monaural compatibility decreases
accordingly. Once you process a stereo signal with an extreme setting of 2, when you
play back the track in mono, the signal will be cancelled out completely—after all, L-R
plus R-L doesn’t leave you with much.

What is MS?
Relegated to obscurity for a good long while, MS stereo (middle-side as opposed to
left-right) has recently enjoyed a renaissance of sorts. To explain; two microphones are
stacked on a stand or suspended from the ceiling so that they are positioned as closely
together as possible. One of the two is a cardioid (or omnidirectional) mic which faces
the sound source that you want to record directly—in straight alignment. The other is a
bidirectional mic, the sensitive axes of which point to the left and right at 90˚ angles.
The cardioid mic delivers the middle signal, the bidirectional mic the side signal. Simply
record the middle signal to the left side of a stereo track and the side signal to the
right. This configuration enables the Direction Mixer to easily decode MS recordings.
The advantage that MS recordings have over XY recordings (two cardioid microphones
aligned so that they are directed to a point halfway to the left and right of the sound
source) is that the stereo middle is actually located on the on-axis (main recording
direction) of the cardioid mic. This means that slight fluctuations in frequency response
that occur off the on-axis—as is the case with every microphone—are less
troublesome.
In principle, MS and LR signals are equivalent and can be converted at any time. When
“-” signifies a phase inversion, then the following applies:
M = L+R
S = L−R
In addition, L can also be derived from the sum of and R from the difference between
M and S. Here’s some interesting trivia for you: Radio (FM) broadcasts feature M and S
stereo. The MS signal is actually converted to a signal suitable for the left and right
speakers by the receiver.

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Multimeter

This plug-in consists of a collection of professional gauge and analysis tools, namely;
• a 1/3 octave Analyzer
• a Goniometer for judging the phase coherency in the stereo sound field
• a Correlation Meter to spot mono phase compatibility
• an integrated Level Meter
The control panel to the left of the display allows you to switch between the Analyzer
and Goniometer, and contains parameter controls for the Multimeter.
The Stereo level and Phase Correlation meter is always displayed.

1/3 Octave Spectrum Analyzer
This tool analyses the level of 31 independent frequency bands. Each frequency band
represents one third of an octave. The Analyzer complies with the specifications
outlined by lEC document 1260.
Analyzer Button
The Analyzer button activates the Analyzer (and deactivates the Goniometer). The four
buttons below determine the portion of the Input signal that the Analyzer displays.
You can choose to view the Left or Right channel independently.
LR max shows the maximum Band levels of either channel, while Mono displays the
levels of the stereo signal as a summed mono entry.
View
The View options determine the level represented by the top line of the scale shown in
the display (Top; range: −40 to +20 dB) and the overall dynamic range of the Analyzer
(range; range: 20 to 80 dB).
These two parameters can also be set directly in the bar graph. To do so, click-hold and
drag on the bar graph to move the top line of the display. Click-holding and dragging
directly on the dB scale allows you to compress or expand the range of the scale.

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These features are useful when analyzing highly compressed material, as you can
identify smaller level differences more easily by moving and/or reducing the display
range.
Mode
There are three display response modes: RMS slow, fast, and Peak.
The two RMS modes with slow and fast response settings show the effective signal
average (“Root Mean Square”), and provide a good representative overview of the
perceived volume levels. The Peak mode shows level peaks accurately.

Goniometer

The idea of the Goniometer was born with the advent of early two channel
oscilloscopes. To use such devices as Goniometers, users would connect the left and
the right stereo channels to the X and Y inputs, while rotating the display by 45
degrees. This would result in a useful visualization of the signal’s stereo phase.
A Goniometer helps you to judge the coherence of the stereo image. Phase problems
are easily spotted as trace cancellations along the center line (M—mid/mono).
The signal trace slowly fades to black. Not only does this imitate the retro glow of the
tubes found in older Goniometers, it also significantly enhances the readability of the
tool.
Goniometer Button
This button activates the Goniometer (and deactivates the Analyzer).
Auto Gain
In order to obtain a higher readout on low-level passages, you can use the Auto Gain
display parameter. When activated, this parameter allows the display to automatically
compensate for low input levels. The amount of compensation can also be set with the
Auto Gain parameter. The Auto Gain parameter can also be adjusted by click-holding
and dragging directly in the display area of the Goniometer.

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Note: Auto Gain is a display parameter only, and increases display levels in order to
enhance readability. The actual audio levels are not affected by this parameter.
Decay
The Decay parameter determines the time that it takes for the Goniometer trace to fade
to black.

Correlation Meter
This is another phase measurement tool that gauges the phase relationship of a stereo
signal. This meter is also available as a separate plug-in.

The Correlation Meter’s scale ranges from −1 to +1.
• A +1 correlation value indicates that the left and right channels correlate 100%. In
other words, the left and right signals are in phase.
• Correlation values in the blue zone (between +1 and the middle position) indicate
that the stereo signal is mono compatible.
• The middle position indicates the highest permissible amount of left/right
divergence, which is often audible as an extremely wide stereo effect.
• Once the correlation meter moves into the red area to the left of the center position,
out-of-phase material is present. This will lead to phase cancellations if the stereo
signal is combined into a mono signal.

Level Meter (Peak/RMS Meter)
The stereo Level Meter shows the current signal level on a logarithmic scale—using
two blue bars. If the level is higher than 0 dB, the portion of the bar above the 0 dB
point will turn red.
This meter is also available as a separate small plug-in.
The current peak values are displayed numerically (in dB increments), above the Level
Meter. The values are reset by clicking into the display.
While the separate Level Meter plug-in is switchable between Peak and RMS
characteristics, the Level Meter of the Multimeter shows Peak and RMS values
simultaneously. The RMS level is represented by a dark blue bar while the Peak level bar
is light blue in color.

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Peak Hold
This section controls the peak hold behavior of all metering tools.
Hold
The Hold button activates the Multimeter’s peak hold function. The duration of the
Hold time is set in the parameter field alongside the Hold button.
• In the Analyzer and Level Meter a small yellow segment above each 1/3 octave level
bar, and stereo level bar, labels the most recent peak level.
• In the Goniometer, all illuminated pixels of the display are held in place during peak
hold.
• In the Correlation Meter, a growing horizontal area around the white correlation
indicator represents any phase correlation deviations—in both directions. A vertical
red line to the left of the correlation indicator permanently shows the maximum
negative phase deviation value. This line can be reset by clicking on it during
playback.
(Value)
This parameter field allows the hold time for all metering tools to be set to 2 s, 4 s, 5 s,
6 s or, infinite by click/dragging on the value.
Reset
During operation, the Reset button resets the peak hold segments of all metering tools.
It should be noted that all metering tools are automatically reset when playback is
stopped.

Correlation Meter
The Levelmeter is also part of the MultiMeter plug-in. A detailed description of its
parameters can be found in the Multimeter’s “Correlation Meter” section, on page 165.

Levelmeter
The Levelmeter is also part of the MultiMeter plug-in. A detailed description of its
parameters can be found in the Multimeter’s “Level Meter (Peak/RMS Meter)” section,
on page 165.

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Vocoder—Basics

If you are new to vocoders you should read this chapter.
It provides you with basic knowledge about vocoders and their functionality. You will
also find tips on using vocoders, and achieving good speech intelligibility.

What Is a Vocoder?
The word Vocoder is an abbreviation for VOice enCODER. As with many technologies in
this otherwise beautiful world, it is a child of war. The Vocoder was initially developed
for secure speech transmission over telephone lines which couldn’t be tapped. To
achieve this, the speech signal was analyzed and only the cryptic results of the analysis
were transmitted over telephone lines. On the receiving side, these results were used to
synthetically rebuild the original voice signal.
Fortunately, Vocoders are used nowadays for altogether more peaceful purposes—
namely for music. A Vocoder analyses and transfers the sonic character of the audio
signal arriving at its analysis input to the audio signal present at its synthesis input. The
result of this process is heard at the output of the Vocoder.
The classic vocoder sound uses speech as the analysis signal and a synthesizer sound as
the synthesis signal. This classic sound was popularized in the late 70’s and early 80’s.
You’ll probably know it from tracks such as “O Superman” by Laurie Anderson, “Funky
Town” by Lipps Inc. and numerous Kraftwerk pieces—from “Autobahn” and “Europe
Endless” up to “The Robots” and “Computer World”.
Away from these “singing robot” sounds, vocoding has also been used in many films. As
examples: the Cylons in Battlestar Galactica, and most famously, on the voice of Darth
Vader from the Star Wars saga.
Vocoding, as a process, is not strictly limited to vocal performances. You could use a
drum loop as the analysis signal to shape a string ensemble sound arriving at the
synthesis input.

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How Does a Vocoder Work?
The speech analyzer and synthesizer referred to above are actually two filter banks of
bandpass filters. Bandpass filters allow a frequency band (a slice) in the overall
frequency spectrum to pass through unchanged, and cut the frequencies which fall
outside of the band’s range.
In Logic’S EVOC 20 plug-ins, these filter banks are named the Analysis and Synthesis
sections. These filter banks have a matching number of corresponding bands—if the
Analysis filter bank has five bands (1, 2, 3, 4, and 5), there will be a corresponding set of
five bands in the Synthesis filter bank. Band 1 in the Analysis bank is matched to band 1
in the Synthesis bank, band 2 to band 2, and so on.
The audio signal arriving at the analysis input passes through the Analysis filter bank
where it is divided into up to 20 bands.
An envelope follower is coupled to each filter band. The envelope follower of each band
tracks (follows) any volume changes in the portion of the audio source allowed to pass
by the associated bandpass filter. In this way, the envelope follower of each band
generates dynamic control signals.
These control signals are then sent to the Synthesis filter bank where they control the
levels of the corresponding Synthesis filter bands. This is done via VCAs—Voltage
Controlled Amplifiers. This allows the volume changes of the bands—and thus the
changes of the original sound—in the Analysis filter bank to be imposed on the
matching bands in the Synthesis filter bank.
The more bands a Vocoder offers, the more precisely the original sound’s character will
be re-modeled.
Envelope Follower 1—5

Analysis
Audio Source

Analysis
Filter Bank
Band 1—5
U/V
Detection

Synthesis
Audio Source

Env.
Follower
Synthesis
Filter Bank
Band 1—5

Control Signal 1—5
Audio
Output
VCA
1—5

How Does a Filter Bank Work?
If you removed all circuits responsible for transferring the sonic characteristics from the
analysis to the synthesis signal from a Vocoder, and dispensed with the detection of
voiced or unvoiced signals, you’d be left with two filter banks—the analysis and
synthesis filters. To use these musically, you would need to ensure that you could
control the output level of each bandpass filter. With this level of control, you can apply
unique and dramatic changes to the frequency spectrum.

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Analyzing Speech Signals
The principles you’ve been introduced to thus far are insufficient for the transmission of
speech signals.
The reason is that human speech consists of a series of voiced sounds (tonal sounds)
and unvoiced sounds (noisy sounds). The main distinction between voiced and
unvoiced sounds is that voiced sounds are produced by an oscillation of the vocal
cords, while unvoiced sounds are produced by blocking and restricting the air flow
with lips, tongue, palate, throat, and larynx.
Should speech containing voiced and unvoiced sounds be used as a Vocoder’s analysis
signal, but the synthesis engine doesn’t differentiate between voiced and unvoiced
sounds, the result will sound rather toothless. To avoid this, the synthesis section of the
Vocoder must produce different sounds for the voiced and unvoiced parts of the signal.
In Logic’s EVOC 20 PS and the EVOC 20 TO Vocoder plug-ins, there is an Unvoiced/
Voiced detector. This unit detects the unvoiced portions of the sound in the analysis
signal and then substitutes the corresponding portions in the synthesis signal with
Noise, a mixture of Noise + Synth or with the original signal (Blend). If the U/V Detector
detects voiced parts, it passes this information to the Synthesis section, which uses the
normal synthesis signal for these portions. Control over unvoiced/voiced sound
detection, type, and level is found in the U/V Detection section of Logic’s vocoder plugins.

Tips for Better Speech Intelligibility
The classic vocoder effect is very demanding, with regard to the quality of both the
analysis and synthesis signals. Furthermore, the vocoder parameters need to be set
carefully. Following, are some tips on both topics.

Editing the Analysis and Synthesis Signals
Compressing the Side Chain
The less the level changes, the better the intelligibility of the vocoder. We therefore
recommend that compression be used in most cases.
Enhancing High Frequency Energy
The vocoder, in a way, always generates the intersection point of the analysis and
synthesis signals. To explain: If there’s no treble portion in the analysis signal, the
resulting vocoder output will also lack treble. This is also the case when the synthesis
signal features a lot of high frequency content. This is true of each frequency band. As
such, the vocoder demands a stable level in all frequency bands from both input
signals, in order to obtain the best results.

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Due to the way human beings hear, the intelligibility of speech is highly dependent on
the presence of high frequency content. To aid in keeping speech clear, it may be
worthwhile using equalization to boost or cut particular frequencies in analysis signals
before processing them with the vocoders.
If the Side Chain (analysis) signal consists of vocals or speech, a simple shelving filter
should be sufficient. It doesn’t require much processing power, and efficiently boosts
the high-mid and treble range, which is so important for speech intelligibility.
If the synthesis signal lacks treble energy, it can be generated with a distortion effect.
The overdrive plug-in is perfect for this purpose.

Avoiding Sonic Artefacts
A common problem with vocoder sounds are sudden signal interruptions (ripping,
breaking sounds) and rapidly triggered noises, during speech pauses.
Release Parameter in the Analysis Section
The Release parameter defines the speed that a given synthesis frequency band can
decrease in level, if the signal level of the respective analysis band decreases abruptly.
The sound is smoother when the band levels decrease slowly. To achieve this smoother
character, use higher Release values in the analysis section of the interface. Longer
release times result in a washy sound.

Short Attack values are no problem. They may, in fact, even be desirable when a fast
reaction to impulse signals by the vocoder is required.

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Gating Background Noises in the Side Chain

If the Side Chain signal is compressed, as recommended, the level of breath, rumble,
and background noises will rise. These background noises can cause the Vocoder bands
to open, but this is normally not intended. In order to eliminate these noises, it’s
therefore a good idea to employ a noise gate before compression and treble boosting.
If the Side Chain signal is gated appropriately, you may find that you want to reduce
the analysis Release value. When gating speech and vocals, the Hysteresis parameter is
important. Threshold defines the level, above which the gate will open. Hysteresis
defines a lower Threshold level, under which the gate will close. The value is relative to
the Threshold level. The graphic shows a Threshold setting, which is well-adapted to
compressed speech. Unwanted triggering by low or high frequency noise is avoided by
the Noise Gates’ dedicated sidechain filters. The Hold, Release, and Hysteresis values
shown are typical level envelopes, suitable for most vocal and speech signals.

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Treatment of the Side Chain signal (speech) with Compressor, Shelving Filter and Gate.
The Silver Compressor, Silver Gate or another EQ are well-suited for these purposes.

Achieving the Best Analysis and Synthesis Signals
For good speech intelligibility, please keep these points in mind:
• The spectra of the analysis and synthesis signals should overlap almost completely.
Low male voices with synthesis signals in the treble range do not work well.
• The synthesis signal must be constantly sustained, without breaks. The track should
be played legato, as breaks in the synthesis signal will stop the vocoders output.
Alternatively, the Release parameter of the synthesis signal (not to be confused with
the Release time of the analysis section) can be set to a longer time. Nice effects can
also be achieved by the use of a reverberation signal as a synthesis signal. Note that
the two latter methods can lead to harmonic overlaps.
• Do not overdrive the Vocoder. This can happen easily, and distortion will occur. Lower
the Gain of the compressor in the Side Chain track, to avoid this problem.
• Enunciate your speech clearly, if the recording is to be used as an analysis signal.
Spoken words, with a relatively low pitch, work better than sung vocals—even if the
creation of vocoder choirs is your goal! Pronounce consonants well.

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Note: A nice example is the rolled R of “We are the Robots”, by Kraftwerk, a classic
vocoder track. This pronunciation was specifically made to cater to the demands of the
vocoder.
Feel free to do what you like when setting the Formant parameters. The intelligibility of
speech is surprisingly little affected by shifting, stretching or compressing the formants.
Even the number of frequency bands used has a minimal impact on the quality of
intelligibility. The reason for this is our ability to intuitively differentiate the voices of
children, women, and men, whose skulls and throats vary vastly by nature. Such
physical differences cause variations in the formants which make up their voices. Our
perception (recognition) of speech is based on an analysis of the relationships between
these formants. In Logic’s Vocoder plug-ins, these stay intact, even when extreme
formant settings are used.

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The EVOC 20 PS

The EVOC 20 PS combines a vocoder with a polyphonic
synthesizer, and can be played in real-time. This chapter
explains the use of the EVOC 20 TO and its parameters.
The EVOC 20 PS is a sophisticated vocoder, equipped with a polyphonic synthesis
engine, and capable of receiving MIDI note input. This allows the EVOC 20 PS to be
played, resulting in classic vocoder choir sounds, for example. Single notes and chords
played with the polyphonic EVOC 20 PS will sing with the articulation of the Analysis
audio source. During this process, the sonic characteristics and changes of the audio
signal arriving at the Analysis input (the audio track selected as a side chain) are
imposed onto the output signal of the integrated synthesizer (the Synthesis section).
The signal path of the EVOC 20 PS is shown in the block diagram on page 190.

Using the EVOC 20 PS
To make use of the EVOC 20 PS, you need to insert the EVOC 20 PS into an Audio
Instrument channel instrument slot, and you also need to provide an audio signal as
the Analysis audio source.
You can do this by following these steps.
1 Select or create a new audio track in the Arrange window.
2 Insert (or record) an audio file—we’ll use a vocal part to start with—onto this audio
track by Shift-clicking onto the Arrange area with the Pencil tool. This will launch a file
browser, allowing you to select the desired audio file.
3 Click once on the name of the audio file you wish to use, and press Open. The file will
be inserted at the selected location. It may be worthwhile setting up a cycle region in
the Arrange window, allowing you to continually cycle the audio part. This will make
experimentation easier.
4 Insert the EVOC 20 PS into the Instrument slot of an Audio Instrument channel.
5 In the gray area at the top of the Plug-in window, click-hold on the Side Chain pulldown menu, and select the audio track that contains the audio file.

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6 Ensure that the corresponding Audio Instrument track is selected in the Arrange
window. The EVOC 20 PS is now ready to accept incoming MIDI data, and has been
assigned to see the output from the selected audio track via a Side Chain.
7 In the Track Mixer or Environment Audio layer (not the Arrange!), mute the audio track
(the vocal track) serving as the Side Chain input.
8 Press the Play button on the Transport Bar, or use the Play key command (0 on the
numeric keypad).
9 And now … as the audio file is playing back, play your MIDI keyboard.
10 In the Track Mixer (or Environment Audio layer), adjust the volume levels of the
EVOC 20 PS and the audio track used for the Side Chain to taste.
11 Do a little experimentation with the knobs, sliders, and other controls. Have fun, and
feel free to insert further effect plug-ins on the channel or busses to further enhance
the sound.

EVOC 20 PS Parameters

The EVOC 20 PS interface is divided into six main sections. These are the Synthesis,
Sidechain Analysis, Formant Filter, Modulation, U/V Detection, and Output areas.

Synthesis Parameters
The EVOC 20 PS is equipped with a polyphonic synthesis engine. It is capable of
accepting MIDI note input. The parameters of the Synthesis section are described
below.
Mode Switches
These switches determine the number of voices used by the EVOC 20:

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• When Poly is selected, the maximum number of voices is set via the numeric field

alongside the Poly button. To change the value, click and hold with your mouse, and
drag up or down to increase/decrease polyphony.
Note: Increasing the number of voices also increases processor overhead.
• When Mono or Legato is selected, the EVOC 20 PS is monophonic, and uses only one

voice.
• In Legato mode, Glide (see page 180) is only active on tied notes. Envelopes are not
retriggered when tied notes are played (single trigger).
• In Mono mode, Glide is always active and the envelopes are retriggered by every
note played (multi trigger).
• The Unison button enables/disables Unison mode. In this mode, each EVOC 20 PS
voice is doubled, which will cut polyphony in half (to a maximum of 8 voices) as
indicated by the numeric Voices field. The doubled voices are detuned by the amount
defined with the Analog parameter. (Also see the “Analog Tuning”, from page 180
onwards.)
• In Unison-Mono mode (both the Unison and Mono or Legato buttons are active), up to
16 voices can be stacked and played monophonically. In this mode, the Voices field
displays the number of stacked voices that sound at the same time.
Warning: Stacking voices in Unison-Mono mode will increase the EVOC 20 PS’s output
volume. To avoid overloading the Audio Instrument channel output, adjust the
EVOC 20 PS’s Level slider accordingly.
Oscillator Section
The EVOC 20 PS is equipped with a two oscillator digital synthesizer which features a
number of waveforms, and FM (Frequency Modulation). Further to these soundgenerators in the Synthesis section is an independent Noise generator.
There are two oscillator modes.
• Dual: Where two oscillators make use of single-cycle digital waveforms to provide

the Synthesis sound source(s).
• FM: A two operator FM engine, with Oscillator 1 as a sine wave carrier, and Oscillator

2 as the modulator. Oscillator 2 can use any of the single-cycle digital waveforms.
You can switch between Dual and FM modes by clicking on the Dual or FM label(s) to
the top left of the section shown in the diagrams.

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As you can see, there are some subtle differences between the two modes. We will look
at the common parameters first, and will then look at the mode-specific options.
Wave 1 Parameters
The footages below the Wave 1 label in both modes harks back to the days of pipe
organs. The longer the pipe, the deeper the tone. This also applies to Wave 1. Simply
click on the 16, 8 or 4 foot value to select the range in which Wave (oscillator) 1
functions. Your selection will be illuminated.
The numerical value beside the Wave 1 label (shown as 41 in the diagrams) indicates
the currently selected waveform type. The EVOC 20 PS features 50 waveforms with
different sonic characteristics. To switch between them, simply click-hold on the
numerical field and drag up or down. When the desired waveform number is visible,
release the mouse button.
Note: When in FM mode, the waveform of Wave 1 is a fixed sine wave. The waveform
parameter of Wave 1 does not have an effect in this mode.
Wave 2 Parameters
The numerical value beside the Wave 2 label (shown as 41 in the diagrams) indicates
the currently selected waveform type. The EVOC 20 PS features 50 single-cycle digital
waveforms with different sonic characteristics. To switch between them, simply clickhold on the numerical field and drag your mouse up or down. When the desired
waveform number is visible, release the mouse button.
Noise Parameters

The Noise generator provides a further sound source which can be used in addition to
the two oscillators (Wave 1 and Wave 2).
The Level knob controls the amount of noise added to the signals of the two oscillators,
and the Color knob controls the timbre of the noise signal. When the Color knob is
turned fully-left, the Noise generator creates a pure white noise. When turned fully-right,
it generates blue noise (high-passed noise). White noise has always been used to create
wind and rain sound effects. It has the same energy in each frequency interval. Blue
noise sounds brighter, because its bass portion is suppressed by a high pass filter.
It is important to note that the Noise generator in the Oscillator section is independent
of the Noise generator in the U/V Detection section. For further information on voiced
and unvoiced signals, refer to “Analyzing Speech Signals”, from page 169 onwards.

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Note: Turn Color full-right and Level a tiny bit up to achieve a more lively and fresh
synthesis signal.
Dual Mode Parameters

The parameters specific to Dual mode are found in the Wave 2 section, and the Balance
slider to the right.
• The Semi parameter adjusts the tuning of the second oscillator (Wave 2) in semitone
steps. Adjustment is made by using the mouse as a slider directly on the numerical
field. Its range: ±24, or up/down two octaves.
• The Detune parameter fine-tunes Wave 1 and Wave 2 in cents. 100 cents equals a
semitone step. Doing so will detune Wave 1 in conjunction with Wave 2 around the
tuning zero point. The range is ±50 cents, or up/down half a semitone. Adjustment is
made by using the mouse as a slider directly on the numerical field.
• The Balance slider allows you to blend the two oscillator signals (Wave 1 and Wave 2).
FM Mode Parameters
The parameters specific to the FM mode are found in the Wave 2 section, and the FM
Int slider to the right.

• The Ratio c(oarse) parameter adjusts the coarse frequency ratio of the second

oscillator in relation to the first oscillator. Adjustment is made by using the mouse as
a slider directly on the numerical field. Range: 0–32.
• The Ratio f(ine) parameter adjusts the fine frequency ratio of the second oscillator in
relation to the first oscillator. The range is 0–99. Adjustment is made by using the
mouse as a slider directly on the numerical field
• The FM Int slider determines the intensity of Wave 1’s sine wave modulation by Wave
2. Higher FM Int settings will result in a more complex waveform with more
overtones.
When combined, the Ratio and FM Int parameters form the resulting complex FM
waveform, thus defining the harmonic content.

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Analog Tuning
The Analog tuning parameter simulates the instability of analog circuitry found in
vintage vocoders. Analog alters the pitch of each note randomly. This behavior is much
like that of polyphonic analog synthesizers. The Analog knob controls the intensity of
this random detuning.

Tuning
The range of detuning is defined in the Tune window. Adjustments are made by using
the mouse as a slider. The range is from 425 to 455 Hz.
Glide
The effect of this knob depends on the setting made in the Bend Range window. Glide
determines the time it takes for the pitch to slide from one note to another
(portamento). The maximum value is +5000 ms.

Bend Range
Bend Range determines the pitch bend modulation range in semitones. The range is
±12 semitones.
Cutoff
This parameter sets the cutoff frequency of the lowpass filter. As you turn this knob to
the left, an increasing number of high frequencies is filtered from the signal.
Resonance
Turning up Resonance leads to an emphasis of the frequency area surrounding the
frequency defined by the Cutoff parameter. The filter is used for rough signal shaping,
before the signal is articulated by the vocoding circuits.
Note: Set Cutoff as high as possible, and add a little Resonance to achieve a nice,
brilliant high-end.

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Envelope

The EVOC 20 PS features an Attack/Release envelope generator used for level control of
the Oscillator section.
• The Attack parameter determines the amount of time that it takes for the Oscillators
of the Synthesis section to reach their maximum level.
• The Release parameter determines the amount of time that it takes for the Oscillators
of the Synthesis section to reach their minimum level.

Sidechain Analysis In Parameters
Attack
The Attack parameter determines how quickly each envelope follower (coupled to each
Analysis filter band) reacts to rising signals. Longer Attack times result in a slower
tracking response to transients of the Analysis input signal.
Note: A long Attack time on percussive input signals (a spoken word or hi-hat part, for
example) will translate into a less articulate vocoder effect. Set Attack to the lowest
possible value to enhance articulation.
Release
The Release parameter determines how quickly each envelope follower (coupled to each
Analysis filter band) reacts to falling signals. Longer Release times cause the Analysis
input signal transients to sustain longer at the vocoder’s output.

Note: A long Release time on percussive input signals (a spoken word or hi-hat part, for
example) will translate into a less articulate vocoder effect. Note that Release times that
are too short result in rough, grainy vocoder sounds. Release values of around 8 to
10 ms have proven to be useful starting points.

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Freeze
The Freeze button holds the current Analysis sound spectrum infinitely.
The frozen Analysis signal can capture a particular characteristic of the source signal
which is then imposed as a complex sustained filter shape on the Synthesis section.
Using a spoken word pattern as a source, for example, the Freeze parameter could
capture the attack or tail phase of an individual word within the pattern—the vowel a,
for example.
With Freeze engaged, the Analysis filter bank ignores the input source until it is
disengaged.
Another use of the Freeze parameter (which can be automated) could be to
compensate for people’s inability to sustain sung notes for a long period, without
taking a breath. If the Synthesis signal needs to be sustained, when the Analysis source
signal (a vocal part) isn’t, Freeze can be used to lock the current formant levels (of a
sung note)—even during gaps in the vocal part—between words in a vocal phrase.
Note: When the Freeze parameter is used, the Attack and Release parameters have no
effect.
Bands
The Bands window determines the number of frequency bands used by the
EVOC 20 PS. It ranges from 5 to 20.

The greater the number of bands, the more precisely the sound can be reshaped. As
the number of bands is reduced, the source signal’s frequency range is divided up into
fewer bands—and the resulting sound will be formed with less precision by the
Synthesis engine.
Note: Increasing the number of Bands also increases the processor overhead.
You may find that a good compromise between sonic precision—allowing incoming
signals (speech and vocals, in particular) to remain intelligible—and resource usage, is
around 10 to 15 bands.

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Formant Filter Parameters
The Formant Filter Window
The Formant Filter window is divided into two sections by a horizontal line. The upper
half applies to the Analysis section and the lower half to the Synthesis section. Changes
made to the High/Low frequency parameters, the Bands parameter or the Formant
Stretch and Shift parameters will result in visual changes to the Formant Filter window.
This provides you with invaluable feedback on what is happening to the signal as it is
routed through the two Formant Filter banks.

High/Low Frequency
The blue bar shown just beneath the EVOC 20 PS logo is a multi-part control which is
used to determine the lowest and highest frequencies allowed to pass by the filter
section. The length of the blue bar represents the frequency range for both analysis
and synthesis. Frequencies of any audio input which fall outside these boundaries will
be cut. All filter bands are distributed evenly across the range defined by the High/Low
Frequency values.
• To adjust the Low Frequency value, simply click-hold on the silver slider to the left of
the blue bar, and drag to the right (or left). The value range is 75–750 Hz.
• To adjust the High Frequency value, simply click-hold on the silver slider to the right
of the blue bar, and drag to the left (or right). The value range is 800–8000 Hz.
• To adjust both sliders simultaneously, click on the area between the slider halves
(directly on the blue bar) and drag to the left or right.
• You can make changes to the High/Low Frequency values directly by using your
mouse as a slider on the numerical entries—270 and 7100 Hz in the diagram.
Lowest/Highest
These parameters can be found in the two small windows on either side of the
Formant Filter window. These switches determine whether the lowest and highest filter
bands act as bandpass filters (like all of the bands between them), or whether they act
as lowpass/highpass filters, respectively. Click once on them to switch between the two
curve shapes available.
• In the Bandpass setting, the frequencies below/above the lowest/highest bands are
ignored for both analysis and synthesis.
• In the Highpass (or Lowpass) setting, all frequencies below the lowest (or above the
highest) bands will be considered for analysis and synthesis.

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Formant Stretch
This parameter alters the width and distribution of all bands in the Synthesis filter bank,
extending or narrowing the frequency range defined by the blue bar (Low/High
Frequency parameters) for the Synthesis filter bank.
With Formant Stretch set to 0, the width and distribution of the bands in the Synthesis
filter bank is equal to the width of the bands in the Analysis filter bank. Low values
narrow the width of each band, while high values widen them. The control range is
from 0.5 to 2 (expressed as a ratio of the overall bandwidth).
Note: You can jump directly to a value of 1 by clicking on its number.
Formant Shift

Formant Shift moves the position of all bands in the Synthesis filter bank up and down.
With Formant Shift set to 0, the position of the bands in the Synthesis filter bank is
equal to the position of the bands in the Analysis filter bank. Positive values will move
the bands up in frequency, while negative values will move them down in respect to
the Analysis filter bank.
Note: You can jump directly to the values −0.5, −1, +0.5 and +1 by clicking on their
numbers.
Note: When combined, Formant Stretch and Formant Shift alter the formant structure of
the resulting vocoder sound, and can lead to some interesting timbre changes. As an
example, using speech signals and tuning Formant Shift up results in Mickey Mouse
effects.
Note: Formant Stretch and Formant Shift are also useful if the frequency spectrum of the
Synthesis signal does not complement the frequency spectrum of the Analysis signal.
You could create a synthesis signal in the high frequency range from an analysis signal
that mainly modulates the sound in a lower frequency range, for example.
Resonance

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Resonance is responsible for the basic sonic character of the vocoder: low settings give
it a soft character, high settings will lead to a more snarling, sharp character. Increasing
the Resonance value emphasizes the middle frequency of each frequency band.
The use of either, or both, of the Formant Stretch and Formant Shift parameters can
result in the generation of unusual resonant frequencies—when high Resonance
settings are used.

Modulation Parameters

The Modulation (LFO) area offers two LFOs to control the Formant Shift and Pitch
parameters of the EVOC 20 PS. The LFOs can run free or synchronized to the song’s
tempo.
• Pitch LFO, on the left-hand side, controls Pitch modulation (Vibrato) of the built-in
synthesizer’s oscillators. It is hardwired to accept data from the modulation wheel of
your MIDI keyboard (or corresponding MIDI data) to control modulation intensity.
• Shift LFO controls the Formant Shift parameter of the Synthesis filter bank to produce
dynamic phasing-like effects.
Wave
These two switches allow the selection of the waveform type used by Pitch LFO and
Shift LFO. A selection of Triangle, falling and rising Sawtooth, Square up and down
around zero (bipolar, good for trills), Square up from zero (unipolar, good for changing
between two definable pitches), a random stepped waveform (S&H), and a smoothed
random waveform is available for each LFO.

Intensity/Int via Whl
The Intensity slider controls the amount of Formant Shift modulation by the Shift LFO.
The Int via Whl slider for the Pitch LFO features a multi-part slider. The intensity of LFO
pitch modulation can be controlled by the modulation wheel of an attached MIDI
keyboard. The upper half of the slider determines the intensity when the modulation
wheel is set to its maximum value, and the lower half when set to its minimum value.
By clicking and dragging in the area between the two slider segments, you can
simultaneously move both.

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Rate Knobs
These knobs determine the speed of modulation. Values to the left of the center
positions are synchronized with the sequencer’s tempo and include bar values, triplet
values and more. Values to the right of the center positions are non-synchronous, and
are displayed in Hertz (cycles per second).
Note: The ability to use synchronous bar values could be used to perform a formant
shift every four bars on a cycled one bar percussion part. Alternately, you could
perform the same formant shift on every eighth note triplet within the same part.
Either method can generate interesting results, and can lead to new ideas, or a new
lease of life on old audio material.

Unvoiced/Voiced (U/V) Detection
Please refer to “Analyzing Speech Signals”, from page 169 onwards, for an explanation
of the U/V Detection principle.

Speech intelligibility is highly dependent on high frequency content, as human hearing
is reliant on these high frequencies to determine syllables within words. Bear this fact
in mind when using the EVOC 20 PS, and take care with filter frequency settings in the
Synthesis and Formant Filter sections.
To aid intelligibility, it may be worthwhile using equalization to boost particular
frequencies in the mid to high frequency range, before processing the signal with the
EVOC 20 PS. Please see the “Tips for Better Speech Intelligibility” on page 169 for
further information.
Sensitivity
This parameter determines how responsive U/V detection is. By turning this knob to
the right, more of the individual unvoiced portions of the input signal are recognized.

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When high settings are used, the increased sensitivity to unvoiced signals can lead to
the U/V source—determined by the Mode parameter—being used on the majority of
the input signal, including voiced signals. Sonically, this results in a sound that
resembles a radio signal which is breaking up, and contains a lot of static or noise.
Mode
Mode selects the sound source(s) that can be used to replace the unvoiced content of
the input signal. Possible settings are Off, Noise, Noise + Synth, or Blend.

• Noise—uses noise alone for the unvoiced portions of the sound.
• Noise + Synth—uses noise and the synthesizer for the unvoiced portions of the

sound,
• Blend—uses the Analysis signal after it has passed through a highpass filter, for the

unvoiced portions of the sound. This filtered Analysis signal is then mixed with the
EVOC 20 PS output signal. The Sensitivity parameter has no effect when this setting is
used.
Level
The Level knob controls the volume of the signal (Noise, Noise + Synth, or Blend) used to
replace the unvoiced content of the input signal.

Warning: Care should be taken with this control, particularly when a high Sensitivity
value is used, to avoid internally overloading the EVOC 20 PS.

Output Parameters
Signal

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This pull-down menu offers the choice of Voc(oder), Syn(thesis) and Ana(lysis). These
options allow you to determine the signal that you wish to send to the EVOC 20 PS
main outputs. To hear the vocoder effect, the Signal parameter should be set to Voc.
The other two settings are useful for monitoring purposes.
Ensemble
The three Ensemble switches switch the ensemble effect(s) on or off. Ensemble I is a
special chorus effect. Ensemble II is a variation, creating a fuller and richer sound by
using a more complex modulation routine.
Level
The Level slider controls the volume of the EVOC 20 PS output signal.
Stereo Width
Stereo Width distributes the output signals of the Synthesis section’s filter bands in the
stereo field.
• At the left position, the output of all bands are centered.
• At the centered position, the output of all bands ascends from left to right.
• At the right position, the bands are output—alternately—to the left and right
channels.

MIDI Controllers Received
The following tables show the CC numbers used when the following MIDI preference is
active: Options > Settings > MIDI Options > (Version 4.x behavior).
Sidechain Analysis

Keyboard

Synthesis

Noise

Analog

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Attack

CC #77

Release

CC #78

Freeze

CC #76

Mode

CC #89

Voices

CC #83

Unison

CC #108

Oscillator Mode

CC #88

Osc 1 Octave

CC #90

Osc 1 Waveform

CC #94

Osc 2 Waveform

CC #95

Osc 2 Semitone

CC #93

Detune/Ratio Fine

CC #91

Osc Mix/FM Intensity

CC #100

Level

CC #99

Color

CC #98

Tune

CC #84

Intensity

CC #85

Glide

Oscillator Filter

Osc. Envelope

Formant Filter

Time

CC #87

Bend Range

CC #86

Cutoff

CC #96

Resonance

CC #97

Attack

CC #106

Release

CC #107

FF Low Freq

CC #65

FF Hi Freq

CC #66

Formant Shift

CC #67
CC #71

LFO 1 (Shift)

LFO 2 (Pitch)

U/V Detection

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Chapter 14 The EVOC 20 PS

FF Resonance

CC #72

FF Low/Bandpass Select

CC #73

FF High/Bandpass Select

CC #74

LFO 1 Rate

CC #68

LFO 1 Waveform Select

CC #69

LFO 1 Intensity

CC #70

LFO 1 Rate

CC #102

LFO 1 Waveform Select

CC #103

LFO 1 Low Intensity

CC #104

LFO 1 High Intensity

CC #105

Sensitivity

CC # 81

Mode

CC # 80

Level

CC # 82

Signal Out Select

CC #109

Ensemble

CC #112

Level

CC #110

Mono/Stereo Select

CC #111

Stereo Width

CC #79

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Block Diagram
This block diagram illustrates the signal path in the EVOC 20 TO and EVOC 20 PS.
L
Side
Track Chain

stereo to mono

R
Analysis Source
Frequency range between Highest/Lowest
1

Filter bank with 5 bands
(example)
Sensitivity
A

Freeze

U/V Detection
B
TO: Pitch
Analysis

ANALYSIS Section
SYNTHESIS Section

TO: Max/
Quant./
Glide

Noise,
N+Synth,

VCA VCA VCA VCA

1
EVOC TO:
• Tracking Oscillator
• Track or Side Chain

VCA

Blend

Level
Filter bank with 5 bands

Stereo
Width

Synthesis Source
EVOC PS:
• Poly Synth
pitch
PS: MIDI keyboard

Filter bank input

Level

LFO
Shift

LFO

Stretch
R
Resonance

Legend

Audio signal

Control signal
EF = Envelope Follower
VCA = Voltage-Controlled Oscillator Parameter control

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Vocoder History

The vocoder is over 50 years old. This chapter discusses
its history.
You may be surprised you to learn that the Voder and Vocoder date back to 1939 and
1940, respectively.
Homer Dudley, a research physicist at Bell Laboratories, New Jersey (USA) developed
the Voice Operated reCOrDER as a research machine. It was originally designed to test
compression schemes for the secure transmission of voice signals over copper phone
lines.
It was a composite device consisting of an analyzer and an artificial voice synthesizer.
These were the:
• Parallel Bandpass Vocoder—a speech analyzer and resynthesizer, invented in 1940.
• The Voder speech synthesizer—a voice model played by a human operator, invented
in 1939. This valve-driven machine had two keyboards, buttons to recreate
consonants, a pedal for oscillator frequency control, and a wrist-bar to switch vowel
sounds on and off.
The analyzer detected the energy levels of successive sound samples, measured over
the entire audio frequency spectrum via a series of narrow band filters. The results of
this analysis could be viewed graphically as functions of frequency against time.
The synthesizer reversed the process by scanning the data from the analyzer and
supplying the results to a number of analytical filters, hooked up to a noise generator.
This combination produced sounds.

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The Voder was demonstrated at the 1939 World Fair, where it caused quite a stir:

In World War II, the Vocoder (now called VOice enCODER) proved to be of crucial
importance, scrambling the transoceanic conversations between Winston Churchill and
Franklin Delanore Roosevelt.
Werner Meyer-Eppler, the director of Phonetics at Bonn University, recognized the
relevance of the machines to electronic music after Dudley visited the University in
1948. Meyer-Eppler used the vocoder as a basis for his future writings which, in turn,
became the inspiration for the German “Elektronische Musik” movement.
In the 1950’s, a handful of recordings ensued.
In 1960, the Siemens Synthesizer was developed in Munich. Among it’s many oscillators
and filters, it included a valve-based vocoding circuit.
In 1967, a company called Sylvania created a number of digital machines that used
time-based analysis of input signals, rather than bandpass filter analysis.
In 1971, after studying Dudley’s unit, Bob Moog and Wendy Carlos modified a number
of synthesizer modules to create their own vocoder for the Clockwork Orange sound
track.
Peter Zinovieff’s London-based company “EMS” developed a standalone—and
altogether more portable—vocoder. EMS are probably best known for the “Synthi AKS”
and VCS3 synthesizers. The EMS Studio Vocoder was the world’s first commercially
available machine, released in 1976. It was later renamed the EMS 5000. Among it’s
users were Stevie Wonder and Kraftwerk. Stockhausen, the German “Elektronische
Musik” pioneer, also used an EMS vocoder.
Sennheiser released the VMS 201 in 1977, and EMS released the EMS 2000, which was a
cut-down version of it’s older sibling.

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1978 saw the beginning of mainstream vocoder use, riding on the back of popularity
created through the music of Herbie Hancock, Kraftwerk, and a handful of other artists.
Among the manufacturers who jumped into vocoder production at this time
are: Synton/Bode, Electro-Harmonix, and Korg, with the VC-10.
In 1979, Roland released the VP 330 ensemble/vocoder keyboard.
The late 70’s and early 80’s were the heyday of the vocoder. Artists who used them
included: ELO, Pink Floyd, Eurythmics, Tangerine Dream, Telex, David Bowie, Kate Bush,
and many more.
On the production side, vocoders could (and can still) be picked up cheaply in the form
of kits from electronics stores.
From 1980 through to the present, EMS in the UK, Synton in Holland and PAiA in the
USA were, and remain, the main flyers of the vocoding flag.
In 1996, Doepfer in Germany and Music and More joined the vocoder-producing
fraternity.
Throughout the 1990’s, a number of standalone software-based vocoders have
appeared.
As you can see, the history of the vocoder is long and varied, and we’re sure that the
vocoder will be with us for a while to come.

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Synthesizer Basics

If you are new to synthesizers, you should read this
chapter.
It covers important facts about the synthesizer and explains the difference between
analog, digital and virtual analog synthesizers. Important synthesizer terms such as
cutoff, resonance, envelope, and waveform are also introduced.

Analog and Subtractive
An analog synthesizer signal is an electrical signal, measured in volts. To give you a
brief comparison with a technology you’re probably familiar with, we’ll look at speakers.
The speaker “coils” move when the voltage—amplified by a power amplifier and output
to the speaker—changes. When the voltage rises, the speaker coil moves forward. If the
voltage falls, the speaker coil moves backwards.
In a digital synthesizer, the signal flow is digital. Binary descriptions of the signal (a
string of zeros and ones) are fed from one algorithm to another. This is an important
distinction to make. It is not the signal itself that is fed from a virtual oscillator to a
virtual filter and so on.
A virtual analog synthesizer is a digital synthesizer which mimics the architecture,
features, and peculiarities of an analog synthesizer. It includes the front panel with all
controls, which provides direct access to all sound generation parameters.
Logic’s ES1 is an example of a virtual analog synthesizer. Its virtual signal flow is as per
that found in analog synthesizers. It includes some of the desirable idiosyncrasies of
particular analog circuits—in cases where they tend to sound nice, such as high
oscillator levels overdriving the filter. The ES1 also features a graphical control surface
on your computer screen. Its signal processing (those “virtual” oscillators and so on) is
performed by the central processing unit (CPU) of your computer.

195

Undesirable analog synthesizer phenomena, such as the habit of going completely out
of tune, are not simulated by virtual analog synthesizers. You can, however, set the
voices of the ES1 to randomly detune, adding “life” to the synthesizer’s sound. Unlike its
analog counterparts, the ES1 is also; completely programmable (you can save sound
settings), can be completely automated (you can record and playback fader
movements), polyphonic (you can play up to 16 notes at the same time), multitimbral
(you can play different sounds at the same time—on different Audio Instrument
channels), and velocity sensitive.
These are important benefits, which overcome the limitations of old synthesizers. If you
find it more inspirational to avoid the use of these features, you can always switch
them off.

What Is Synthesis?
Before we start, synthesis in this context, is the (re)production of a sound which
emulates, or synthesizes the sound of another instrument, a voice, helicopter, car, dog
bark—in fact, any sound you can think of!
This synthetic reproduction of other sounds is what gives the synthesizer its name.
Needless to say, synthesizers can also produce many sounds which would never occur
in the “natural” world. This ability to generate sounds which cannot be created in any
other way is what makes the synthesizer a unique musical tool. Its impact on modern
music has been enormous, and will continue well into the future—although it is more
likely to live on in “virtual” form, rather than as hardware.

Subtractive Synthesis
Subtractive synthesis is synthesis using filters. All analog and virtual analog synthesizers
use subtractive synthesis to generate sound. In analog synthesizers, the audio signal of
each voice is generated by the oscillator. The oscillator generates an alternating current,
using a selection of waveforms which contain differing amounts of (more or fewer)
harmonics. The fundamental (or root) frequency of the signal primarily determines the
perceived pitch, its waveform is responsible for the basic sound color, and the
amplitude (level) determines the perceived volume.

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Cutoff and Resonance—illustrated with a sawtooth wave

This picture shows an overview of a sawtooth wave (a = 220 Hz); The filter is open, with
cutoff set to its maximum, and with no resonance applied. The screenshot shows the
output signal of Logic’s ES1, routed to a monophonic Logic Output Object. The
recording was performed with the Bounce function of this Audio Object, and is
displayed in Logic’s Sample Editor at a high zoom setting.
When Michelangelo was asked how he would manage to cut a lion out of a block of
stone, he answered, “I just cut away everything that doesn’t look like a lion”. This, in
essence, is how subtractive synthesis works: Just filter (cut away) those components of
sound which should not sound—in other words, you subtract parts of the oscillator
signal’s spectrum. After being filtered, a brilliant sounding sawtooth wave becomes a
smooth, warm sound without sharp treble. Analog and virtual analog synthesizers are
not the only devices that make use of subtractive synthesis techniques. Samplers and
sample players also do so, but use modules which play back digital recordings
(Samples) in place of oscillators (that supply sawtooth and other waveforms).
The picture below shows a sawtooth wave with the filter half closed (24 dB/Fat). The
effect of the filter is somewhat like a graphic equalizer, with a fader set to a given cutoff
frequency (the highest frequency being fed through) pulled all the way down (full
rejection), so that the highs are damped. With this setting, the edges of the sawtooth
wave are rounded, making it resemble a sine wave.

The wave length here is not really higher, but the zoom setting is.

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Fourier Theorem and Harmonics
“Every periodic wave can be seen as the sum of sine waves with certain wave lengths
and amplitudes, the wave lengths of which have harmonic relations (ratios of small
numbers)”. This is known as the Fourier theorem. Roughly translated into more musical
terms, this means that any tone with a certain pitch can be regarded as a mix of sine
partial tones. This is comprised of the basic fundamental tone and its harmonics
(overtones). As an example: The basic oscillation (the first partial tone) is an “A” at
220 Hz. The second partial has double the frequency (440 Hz), the third one oscillates
three times as fast (660 Hz), the next ones 4 and 5 times as fast, and so on.
You can emphasize the partials around the cutoff frequency by using high resonance
values. The picture below shows a sawtooth wave with a high resonance setting, and
the cutoff frequency set to the frequency of the third partial (660 Hz). This tone sounds
a duodecima (an octave and a fifth) higher than the basic tone. It’s apparent that
exactly three cycles of the strongly emphasized overtone fit into one cycle of the basic
wave:

The effect of the resonating filter is comparable to a graphic equalizer with all faders
higher than 660 Hz pulled all the way down, but with only 660 Hz (Cutoff Frequency)
pushed to its maximum position (resonance). The faders for frequencies below 660 Hz
remain in the middle (0 dB).
If you switch off the oscillator signal, a maximum resonance setting results in the selfoscillation of the filter. It will then generate a sine wave.

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Other Oscillator Waveforms
Waveforms (waves) are named sawtooth, square, pulse, or triangular because of their
shape when displayed as an oscillogram (as in Logic’s Sample Editor). This is the
triangular wave:

The triangular wave has few harmonics—which is evident by the fact that is shaped
more like a sine than a sawtooth wave. This wave contains only odd harmonics—which
means no octaves.

Envelopes
What does the term envelope mean in this context? In the image, you see an
oscillogram of a percussive tone. It’s easy to see how the level rises immediately the
top of its range, and how it decays. If you drew a line surrounding the upper half of the
oscillogram, you could call it the envelope of the sound—a graphic displaying the level
as a function of time. It’s the job of the envelope generator to set the shape of the
envelope.

The screenshot shows a recording of an ES1 sound created with these ADSR (attack
time, decay time, sustain level, and release time) parameter settings: attack as short as
possible, medium value for decay, zero for sustain, medium value for release.

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When you strike a key, the envelope travels from zero to it’s maximum level in the
attack time, falls from this maximum level to the sustain level in the decay time, and
maintains the sustain level as long as you hold the key. When the key is released, the
envelope falls from its sustain level to zero over the release time. The brass or string-like
envelope of the following sound—the envelope itself is not shown in this graphic—has
longer attack and release times, and a higher sustain level.

The envelope generator can also control the rise and fall of the cutoff frequency. You
can also use envelope generators to modulate other parameters. In this context,
modulation can be thought of as a remote control for a given parameter. There are
more sources that can serve as a modulation source: the pitch (note number), velocity
sensitivity or the modulation wheel, for example.

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Chapter 16 Synthesizer Basics

EFM 1

The 16-voice polyphonic EFM 1 is a powerful synthesizer
based on frequency modulation.
It produces the typically rich bell and digital sounds that FM synthesis has become
synonymous with.

Concept and Function
At the core of the EFM 1 engine, you’ll find a multi-wave Modulator oscillator and a sine
wave Carrier oscillator. The Modulator oscillator modulates the frequency of the Carrier
oscillator within the audio range, thus producing new harmonics. These harmonics are
known as sidebands.
The EFM 1 is divided into three areas. The top ring contains the global Transpose, Tune,
Randomize, and Unison parameters.
The raised T-shaped FM engine in the center consists of the Modulator, Carrier, and FM
controls, including the Modulation Envelope and LFO.
The bottom section of the ring houses the Output section, and features the Sub Osc
Level and Stereo Detune parameters, plus the Volume Envelope, Main Level and Velocity
controls.

201

Global Parameters
Transpose
The base pitch is set with the Transpose parameter. You can transpose the EFM 1 by ±2
octaves.

Tune
Tune will fine-tune the EFM 1 ± 50 cents. A cent is 1/100th of a semitone.

Randomize
The Randomize facility generates new sounds with each mouse click. Click the
Randomize button to create a new randomized sound, based on the Intensity value.
Higher Intensity values—set in the numeric field by click-dragging up/down—will
produce more random sounds. Experiment with values below 10% for small variations
of a given sound.

Unison
Clicking on the Unison button will layer two complete EFM 1 voices, making the EFM 1
sound larger and fatter. In Unison mode, the EFM 1 can be played with 8-voice
polyphony.

Voices
The number of simultaneously playable voices (polyphony) is determined by the Voices
parameter. Available values are: Mono (one voice), Legato (one voice) and 2–16 voices.
In the monophonic Legato mode, playing overlapping notes will not retrigger the
EFM 1 envelopes.

Glide
Glide is used to introduce a continuous pitch bend between two consecutively played
notes. The Glide value (in ms) determines the time it takes for the pitch to travel from
the last played note to the next. Glide can be used in both of the monophonic Mono
and Legato Voices modes, as well as with the polyphonic Voices settings (2–16).

FM Parameters
FM (Intensity)
The Modulator oscillator modulates the Carrier frequency, resulting in newly generated
sidebands that add new overtones. Turning up the FM (Intensity) control (the large dial
in the center) produces increasing numbers of overtones—and the sound becomes
brighter. The FM (Intensity) parameter is sometimes called the FM Index.
Note: Although the technology behind it is very different, you could compare the FM
(Intensity) parameter to the Filter Cutoff parameter of an analog synthesizer.

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Modulation Env(elope)
To control the FM (Intensity) parameter dynamically, the EFM 1 provides a dedicated
ADSR (FM) Modulation Envelope, consisting of four sliders: A (Attack time), D (Decay
time), S (Sustain level) and R (Release time). The envelope is triggered every time a MIDI
note is received. The Attack slider sets the time needed to reach the maximum
envelope level. The Decay slider sets the time needed to reach the Sustain level
(determined by the Sustain slider). The Sustain level is held until the MIDI note is
released. The Release slider sets the time needed to reach a level of zero, after the MIDI
note has been released.

FM Depth
The strength, or impact, of the Modulation Envelope on the FM intensity is determined
by the FM Depth control.
Turning the FM Depth control clockwise increases the effect of the Modulation
Envelope. Turning the FM Depth control counter clockwise inverts the effect of the
Modulation Envelope, meaning that the envelope slopes down during the Attack
phase, and slopes up during the Decay and Release time phases.
In the center (0) position, the envelope does not have an effect. You can easily center
the FM Depth dial by clicking on the 0.

Modulator Pitch
The impact of the Modulation Envelope on the pitch of the Modulator oscillator is
determined by the Modulator Pitch control.
Turning the Modulator Pitch control clockwise increases the effect of the Modulation
Envelope. Turning the Modulator Pitch control counter-clockwise inverts the effect of
the Modulation Envelope, meaning that the envelope slopes down during the Attack
phase, and slopes up during the Decay and Release time phases.
In the center (0) position, the envelope does not have an effect. You can easily center
the Modulator Pitch dial by clicking on the 0 button.

LFO
The LFO (Low Frequency Oscillator) serves as a cyclic modulation source for FM
Intensity or Vibrato. Turning the LFO control clockwise increases the effect of the LFO
on FM Intensity. Turning it counter clockwise introduces a vibrato.
In the center (0) position the LFO does not have an effect. You can easily center the LFO
dial by clicking on the 0.

Rate
The speed/rate of the LFO cycles is set with the Rate parameter.

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203

Modulator and Carrier
Harmonic
In FM synthesis, the basic overtone structure is determined by the tuning relationship
of the Modulator and Carrier. This is often expressed as a tuning ratio. In the EFM 1, this
ratio is achieved with the Modulator and Carrier Harmonic controls. Additional tuning
control is provided by the Fine (Tune) parameters.
You can tune the Modulator and Carrier to any of the first 32 harmonics. The tuning
relationship (or ratio) greatly changes the base sound of the EFM 1, and is best set by
ear.
As a rule of thumb: even tuning ratios tend to sound more harmonic or musical, while
odd ratios produce more inharmonic overtones—which are great for bell and metallic
sounds.
As an example, the Modulator and Carrier set to the First Harmonic (a 1:1 ratio) will
produce a sawtooth-like sound. If the Modulator is set to the Second Harmonic, and
the Carrier to the First Harmonic (a 2:1 ratio), the tone produced will sound similar to a
square wave. In this respect, the tuning ratio is somewhat similar to the waveform
selector of an analog synthesizer.
The Harmonic dial of the EFM 1 Carrier can be set to a value of zero. This, in effect,
produces a DC (Direct Current) signal. In this scenario, the Carrier actually acts as a
wave shaper.
Fine
Fine tune adjusts the tuning in-between two adjacent harmonics (as determined by the
Harmonic control). The range of this control is ±0.5 harmonic. Dependent on the
amount of detuning, this will create either a subtle “beating” of the timbre or—if high
detuning amounts are used—adds new harmonic and inharmonic overtones.
In the center (0) position Fine tune does not have an effect. You can easily center the
Fine tune control by clicking on the 0.
Fixed Carrier Button
This button allows you to disconnect the carrier frequency from keyboard, pitchbend,
and LFO modulations.

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Modulator Wave
In classic FM synthesis, sine waves are use as Modulator and Carrier waveforms. To
extend its sonic capabilities, the EFM 1 Modulator provides a number of additional
digital waveforms.
When turned completely counter clockwise the Modulator produces a sine wave.
Turning the Wave parameter clockwise will step/fade through a series of complex
digital waveforms. These digital waveforms add a new level of harmonic richness to the
resulting FM sounds.

The Output Section
Sub Osc Level
For added bass response, the EFM 1 features a sine wave sub oscillator. This operates
one octave below the FM engine (as determined by the Transpose parameter). Turning
up the Sub Osc Level control mixes the sub oscillator sine wave with the EFM 1’s FM
engine output.

Vol(ume) Envelope
The Volume Envelope shapes the overall volume contour. The Volume Envelope consists
of four sliders: Attack time, Decay time, Sustain level, and Release time. The Volume
Envelope is triggered every time a MIDI note is received. The Attack slider defines the
time needed to reach the maximum volume level. The Decay slider sets the time
needed to reach the Sustain level (as determined by the Sustain slider). The Sustain
level is held until the MIDI note is released. The Release slider controls the time needed
to reach a volume level of zero, after the MIDI note has been released.

Stereo Detune
Stereo Detune adds a rich and diverse chorus-like effect to the sound of the EFM 1. This
is achieved by doubling the EFM 1 voice with a detuned second FM engine. The
amount of detune is adjusted using the Stereo Detune dial. A wide stereo effect is also
added, increasing the “space” and “width” of your sound.

Velocity
The EFM 1 is able to respond to MIDI velocity, and reacts with dynamic sound and
volume changes—harder playing will result in a brighter and louder sound. The
sensitivity of the EFM 1 in response to incoming velocity information is determined by
the Velocity parameter.
Set the Velocity control all the way to the left (counter-clockwise) if you don’t want the
EFM 1 to respond to velocity. Turning the control clockwise will increase velocity
sensitivity, and with it, dynamic changes to the sound that the EFM 1 is able to
produce.

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205

Main Level
The Main Level control adjusts the overall output level of the EFM 1. Turning it clockwise makes the EFM 1 output louder. Turning it counter clock-wise will decrease the
output level.

Pitch Bend, Modulation Wheel, Aftertouch
The EFM 1 responds to pitch bend, modulation wheel and aftertouch controller data.
Pitch bend is hardwired to pitch. The modulation wheel introduces vibrato while
aftertouch offers control over FM intensity.

Randomize
The Randomize facility generates new sounds with each mouse click. Click the
Randomize button to create a new randomized sound, based on the Intensity value.
Higher Intensity values will produce more random sounds. Experiment with values
below 10% for small variations of a given sound.

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ES M

This chapter introduces you to Logic’s ES M synthesizer.
The monophonic ES M (ES Mono) is a good starting point if you’re looking for bass
sounds that punch through your mix.

Parameters of the ES M
8, 16, 32
The 8, 16, and 32 buttons set the ES M’s octave transposition.

Glide
The ES M permanently works in a fingered portamento mode, with notes played in a
legato style resulting in a glide (portamento) from pitch to pitch. The speed of the glide
is set with the Glide parameter. At a value of 0, no glide effect occurs.

Mix
Mix crossfades between a sawtooth wave and a 50% rectangular wave, which is heard
one octave lower.

Cutoff
This parameter sets the cutoff frequency of the resonance-capable dynamic lowpass
filter. Its slope is 24 dB/Octave.

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Resonance
This parameter sets the resonance of the dynamic lowpass filter. Increasing the
Resonance value results in a rejection of bass (low frequency energy) when using low
pass filters. The ES M compensates for this side-effect internally, resulting in a more
bassy sound.

Int
The ES M features two very simple envelope generators with a single Decay parameter.
Int enables modulation of the cutoff frequency by the filter envelope.

Decay (Filter)
This parameter sets the decay time of the filter envelope. It is only effective if Int is not
set to 0.

Velo (Filter)
Velo determines the velocity sensitivity of the filter envelope. This parameter is only
effective if Int is not set to 0.

Decay (Volume)
This parameter sets the decay time of the dynamic stage. The attack, release, and
sustain times of the synthesizer are internally set to 0.

Velo (Volume)
This parameter determines the velocity sensitivity of the dynamic stage.

Vol
This parameter sets the master volume of the ES M.

Overdrive
This parameter sets the overdrive/distortion level for the ES M output. Caution: The
overdrive effect significantly increases the output level.

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ES P

This chapter introduces you to Logic’s eight-voice
polyphonic ES P (ES Poly) synthesizer.
Functionally, (despite its velocity sensitivity) this flexible synthesizer is somewhat
reminiscent of the affordable polyphonic synthesizers produced by the leading
Japanese manufacturers in the 1980s: Its design is easy to understand, it is capable of
producing lots of useful musical sounds, and you may be hard-pressed to make sounds
with it that can’t be used in at least some musical style. The creation of classic analog
synthesizer brass sounds are just one of its many strengths.

Parameters of the ES P
8, 16, 32
The 8, 16, and 32 buttons determine the ES P’s octave transposition.

Waveform Faders
The faders on the left side of the panel allow you to mix several waveforms, output by
the oscillators of the ES P. In addition to triangular, sawtooth, and rectangular waves,
the rectangular waves of two sub-oscillators are also available. One of these is one
octave lower than the main oscillators, and the other, two octaves lower. The pulse
width of all rectangular waves is 50%. The right-most fader adds white noise to the mix.
This is the raw material for classic synthesizer sound effects, such as ocean waves, wind,
and helicopters.

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Vib/Wah
The ES P features an LFO which can either modulate the frequency of the oscillators
(resulting in a vibrato), or the cutoff frequency of the dynamic low pass filter (resulting
in a wah wah effect). Turn the control to the left in order to set a vibrato, or to the right
to cyclically modulate the filter.

Speed
Speed controls the rate of the oscillator frequency or cutoff frequency modulation.

Frequency
This parameter set the cutoff frequency of the resonance-capable dynamic low pass
filter.

1/3, 2/3, 3/3
The cutoff frequency can be modulated by MIDI note number (keyboard position); you
may know this parameter as Keyboard Follow on other synthesizers. You have the
choice of: no modulation, one third, two thirds, or full keyboard follow (3/3). When set
to 3/3, the relative harmonic content of each note is the same, independent of its pitch.

ADSR Int
The ES P features one ADSR envelope generator per voice. ADSR Int sets the amount of
cutoff frequency modulation by the ADSR envelope generator.

Velo Filter
The cutoff frequency modulation by the ADSR envelope generator is velocity sensitive.
The amount of velocity sensitivity is set by this parameter.

Volume
This parameter sets the master volume of the ES P.

Velo Volume
This parameter determines the amount of velocity sensitivity, with each note being
louder if struck more firmly.

A
The A slider determines the attack time of the envelope generator.

D
The D slider determines the decay time of the envelope generator.

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S
The S slider determines the sustain level of the envelope generator.

R
The R slider determines the release time of the envelope generator.

Chorus
This parameter sets the intensity of the integrated chorus effect.

Overdrive
This parameter sets the overdrive/distortion level of the ES P output. Caution: The
overdrive effect significantly increases the output level.

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ES E

This chapter introduces Logic’s eight-voice polyphonic ES
E synthesizer.
The ES E (ES Ensemble) is designed for pad and ensemble sounds. It is great for adding
atmospheric sounds to your music.

Parameters of the ES E
4, 8, 16
The 4, 8, and 16 buttons determine the ES E’s octave transposition.

Wave
The left-most setting of the Wave parameter causes the oscillators to output sawtooth
signals, which can be modulated in frequency by the integrated LFO. Across the
remaining range, the oscillators output pulse waves, with the average pulse width
being defined by the Wave parameter.

Vib/PWM
If Wave is set to sawtooth, this parameter defines the amount of frequency modulation,
resulting in a vibrato or siren effect, depending on LFO speed and intensity. If Wave has
been set to a pulse wave, this parameter controls the amount of pulse width
modulation (PWM). When the pulse width becomes very narrow, the sound sounds like
it is being interrupted. Given this potential artefact, set the PWM intensity with care,
and select the Wave parameter’s 12 o’clock-position (50% rectangular) for the pulse
width, if you want to achieve the maximum modulation range.

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Speed
Speed controls the frequency of the pitch (sawtooth) or pulse width modulation.

Cutoff
This parameter sets the cutoff frequency of the resonance-capable dynamic lowpass
filter.

Resonance
This parameter sets the resonance of the ES E’s dynamic lowpass filter.

AR Int
The ES E features one simple envelope generator per voice. It features an Attack and a
Release parameter. AR Int, defines the amount of cutoff frequency modulation applied
by the envelope generator.

Velo Filter
This parameter sets the velocity sensitivity of the cutoff frequency modulation applied
by the envelope generator. This parameter is only effective if AR Int is not set to 0.

Attack
This parameter sets the attack time of the envelope generator.

Release
This parameter sets the release time of the envelope generator.

Velo Volume
This parameter determines the amount of velocity sensitivity, with each note being
louder if struck more firmly.

Volume
This parameter sets the master volume of the ES E.

Chorus/Ensemble
The ES E features a chorus/ensemble effect, with three switchable variations (plus off ).

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ES1

This chapter introduces Logic’s virtual analog ES1
synthesizer.
The ES1’s flexible tone generation system and interesting modulation options place an
entire palette of analog sounds at your disposal: punchy basses, atmospheric pads,
biting leads, and sharp percussion.

Parameters of the ES1

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2', 4', 8', 16', 32'
These footage values allow you to switch the pitch in octaves. 32 feet is the lowest, and
2 feet, the highest setting. The origin of the term feet to measure octaves, comes from
the measurements of organ pipe lengths.

Wave
Wave allows you to select the waveform of the oscillator, which is responsible for the
basic tone color. You can freely set any pulse width in-between the square wave and
pulse wave symbols. The pulse width can also be modulated in the modulation section
(see the “Router” section, on page 219). Modulating the pulse width with a slowly
cycling LFO, for example, allows periodically mutating, fat bass sounds.

Sub
The sub oscillator delivers square waves (one and two octaves below the frequency of
the main oscillator), as well as a pulse wave (two octaves below the frequency of the
main oscillator). In addition to pure square waves, the waveform switch allows
selections between different mixes, and phase relationships of these waves, resulting in
different sounds. You can also use white noise, or switch the sub oscillator OFF. You can
feed a Side Chain signal (from any track!) into the synth’s filter (select EXT). You can
select the Side Chain source track from the Side Chain panel in the gray area at the top
of the Plug-in window.

Mix
This slider defines the mix relationship between the main and sub oscillator signals.
When the sub oscillator wave is switched to OFF, its output is completely removed from
the mix. As a tip, high resonance values allow the filter to self-oscillate, which can be
useful if you want to use the filter like an oscillator.

Filter Parameters
Drive
This is an input level control for the lowpass filter, which allows you to overdrive the
filter. Its use changes the behavior of the Resonance parameter, and the waveform may
sound distorted.

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Cutoff and Resonance
The Cutoff parameter controls the cutoff frequency of the ES1’s lowpass filter.
Resonance emphasizes the portions of the signal which surround the frequency
defined by the Cutoff parameter. This emphasis can be set so intensively, that the filter
begins to oscillate by itself. When driven to self-oscillation, the filter outputs a sine
oscillation (a sine wave). If key is set to 1, you can play the filter chromatically from a
MIDI keyboard.
There is another way to set the cutoff frequency: click-hold on the word Filter
(surrounded by the slope selectors), and move the mouse vertically to adjust the cutoff
frequency. Moving the mouse horizontally adjusts the resonance.
Slope buttons
The lowpass filter offers four different slopes of band rejection above the cutoff
frequency.
• The 24 dB classic setting mimics the behavior of a Moog-style filter: turning up the
resonance results in a reduction of the low-end of the signal.
• The 24 dB fat setting compensates for this reduction in low frequency content.
Turning up resonance doesn’t diminish the low-end of the signal, and thus resembles
an Oberheim-style filter.
• 18 dB tends to resemble the filter sound of Roland’s TB-303.
• The 12 dB setting provides a soft, smooth sound which is reminiscent of the early
Oberheim SEM.

Key
This parameter controls the amount of cutoff frequency modulation by the keyboard
pitch (note number). If Key is set to zero, the cutoff frequency won’t change, no matter
which key you strike. This makes the lower notes sound relatively brighter than the
higher ones. If Key is set to maximum, the filter follows the pitch, resulting in a constant
relationship between cutoff frequency and pitch.

ADSR Via Vel
The main envelope generator (ADSR) modulates the cutoff frequency over the duration
of a note. The intensity of this modulation can be set to positive or negative values, and
can respond to velocity information. If you play pianissimo (Velocity = 1), the
modulation will take place as indicated by the lower arrow. If you strike with the
hardest fortissimo (Velocity = 127), the modulation will take place as indicated by the
upper arrow. The blue bar between the arrows shows the dynamics of this modulation.
You can adjust the modulation range and intensity simultaneously by grabbing the bar
and moving both arrows at once. Note that as you do so, they retain their relative
distance from one another.

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Level Via Vel
The upper arrow works like a main volume control for the synthesizer. The greater the
distance from the lower arrow (indicated by the blue bars), the more the volume is
affected by incoming velocity messages. The lower arrow indicates the level when you
play pianissimo (velocity =1). You can adjust the modulation range and intensity
simultaneously by grabbing the bar and moving both arrows at once. Note that as you
do so, they retain their relative distance from one another. In order to retain the
maximum possible resolution for velocity sensitivity, even when set to a low volume,
the ES1 has an additional Out Level parameter—available in the Controls view.

Amplifier Envelope Selector
The AGateR, ADSR, and GateR switches define which of the ADSR envelope generator
controls have an effect on the amplifier envelope.
• AGateR activates the attack and release time controls, but allows the level to remain
constant between the time the peak level is reached, and the release of the key—
regardless of the decay and sustain settings.
• ADSR activates all controls for the amplifier section.
• GateR sets the attack time for the amplifier section to zero, with only the release
control still having an effect on the envelope level.
All ADSR parameters will always remain active for the filter (ADSR via Vel). A stands for
attack time, R for release time, while Gate is the name of a control signal used in analog
synthesizers, which tells an envelope generator that a key is pressed. As long as an
analog synth key is pressed, the gate signal maintains a constant voltage. Used as a
modulation source in the voltage controlled amplifier (instead of the envelope itself ), it
creates an organ type envelope without any attack, decay or release.

Glide
The Glide parameter defines the amount of (portamento) time applied to each
triggered note. The Glide trigger behavior depends on the value set in Voices (see
“Voices” on page 221). A value of 0 disables the Glide function.

LFO Waveform
The LFO offers several waveforms: triangle, ascending and descending sawtooth,
square wave, sample & hold (random), and a lagged, smoothly changing random wave.
You can also assign a Side Chain signal (any audio track) as a modulation source (EXT).
Select the Side Chain source track via the Side Chain pull-down menu in the gray area
at the top of the Plug-in window.

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Rate
This defines the speed (frequency) of modulation. If you set values to the left of zero,
the LFO phase is locked to the tempo of the song—with phase lengths adjustable
between 1/96 bar and 32 bars. If you select values to the right of zero, it will run freely.
When set to zero, the LFO will output at a constant (and full) level, allowing you to use
the modulation wheel to modulate, say, the pulse width: moving the mod wheel
changes the pulse width in accordance with the “Int via Whl” setting, without
introducing LFO modulation.

Int Via Whl
The upper arrow defines the intensity of the LFO modulation if the modulation wheel
(MIDI Controller 1) is set to its maximum value. The lower arrow defines the amount of
LFO modulation if the modulation wheel is set to zero. The distance between the
arrows (indicated by a green bar) indicates the range of your keyboard’s modulation
wheel. You can simultaneously adjust the modulation range and intensity by grabbing
the bar and moving both arrows at once. Note that as you do so, they retain their
relative distance from one another.

Router
The router defines the modulation target for LFO modulation and the modulation
envelope. Only one target can be set for the LFO, and another one can be set for the
modulation envelope. You can modulate:
• the pitch (frequency) of the oscillator
• the pulse width of the pulse wave
• the mix between the main and sub oscillators
• the cutoff frequency of the filter
• the resonance of the filter
• the main volume (the amplifier)
The following two targets are only available for the modulation envelope:
• Filter FM (the amount of cutoff frequency modulation by the triangle wave of the
oscillator)
The modulation characteristics are non-linear. Thus, you can achieve a pseudo
distortion of existing sounds, or, if only the self-oscillation of the resonating filter is
audible, create metallic, FM style sounds. Switch Sub to off and Mix to Sub in order to
do so.
• LFO Amp (the overall amount of LFO modulation)
As one application, you can create a delayed vibrato by modulating the LFO
modulation intensity if the LFO router is set to pitch. The shape of the modulation
envelope will control the intensity of the vibrato. Select an attack style setting (High
value for form).

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Int Via Vel
The upper arrow controls the upper modulation intensity setting for the modulation
envelope, if you strike a key with the hardest fortissimo (velocity = 127). The lower
arrow controls the lower modulation intensity setting for the modulation envelope, if
you strike a key with the softest pianissimo (Velocity = 1). The green bar between the
arrows displays the impact of velocity sensitivity on the (intensity of the) modulation
envelope. You can simultaneously adjust the modulation range and intensity by
grabbing the bar and moving both arrows at once. Note that as you do so, they retain
their relative distance from one another.

Mod Envelope
The modulation envelope itself only has one parameter. You can set a percussive type
of decay envelope (low values), or attack type envelopes (high values). A full setting of
the modulation envelope delivers a constant, full level. This is useful if you want a
parameter to be modulated solely by velocity: select a modulation destination, (LFO
Amplitude, for example), set the modulation envelope to full, and adjust Int via Vel as
needed, in order to obtain a velocity sensitive, yet non time-variable amount of LFO
Amplitude modulation.

ADSR
The ADSR envelope affects the filter (ADSR via Vel) and the amplifier (if set to ADSR).
The parameters are attack time (A), decay time (D), sustain level (S) and release time (R).
If you’re unfamiliar with these parameters: set the amplifier to ADSR, the Cutoff to a low
value, Resonance to a high value, and move both of the “ADSR via Vel” arrows upwards,
in order to check out what these parameters do.

Tune
Tune sets the pitch of the ES1.

Analog
Analog slightly alters the pitch of each note, and the cutoff frequency, in a random
manner. Similar to polyphonic analog synthesizers, Analog values higher than zero
allow the oscillators of all triggered voices to cycle freely. Note that if Analog is set to a
value of zero, the oscillator cycle start points of all triggered voices are synchronized.
This may be useful for percussive sounds, when looking to achieve a sharper attack
characteristic. For a warm, analog type of sound, the Analog-Parameter should be set to
higher values, thereby allowing subtle variations for each triggered voice.

Bender Range
Bender Range selects the sensitivity of the pitch bender in semitones.

Out Level
Out Level is the master volume control for the ES1 synthesizer.

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Voices
The number displayed is the maximum number of notes which can be played
simultaneously. Each ES1 instance offers a maximum of 16 voice polyphony. Fewer
played voices require less CPU power.
If you set Voices to legato, the ES1 will behave like a monophonic synthesizer with
single trigger and fingered portamento engaged. This means that if you play legato, a
portamento corresponding to the Glide setting will occur, but if you release each key
before you press a new one, there will be no portamento at all. The envelope will not
be triggered by the new note. This allows for pitch bending effects without touching
the pitch bender. Don’t forget to select a higher Glide value when using the Legato
setting.

Chorus
The ES1 offers classic stereo Chorus/Ensemble effects. There are four possible
settings: Off, C1, C2, and Ens.
Off deactivates the Chorus. C1 and C2 are typical Chorus effects. C2 is variation of C1
and is characterized by a stronger modulation. In comparison, the Ensemble effect (Ens)
employs a more complex modulation routing, creating a fuller and richer sound.

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ES2

The virtual-analog ES2 synthesizer offers an exciting and
extensive array of features and functions.
This chapter covers all details of the ES2’s powerful tone generation system. A brief
summary is followed by an in-depth description of its parameters. At the chapter’s end
you will find tutorials, where well-known sound designers explain how to program ES2
sounds.

Concept and Function
The ES2 is one of the most versatile virtual-analog synthesizers ever designed. The
features and sonic potential of many well-respected, rare, and vintage analog
synthesizers pale in comparison to the power offered by the ES2.
Don’t be fooled by the compact and easy-to-understand user interface of the ES2. Its
lack of additional menu items, windows, and lengthy parameter lists may give you the
impression that its sonic power might not be that immense. Belying this simple
appearance, the ES2 offers facilities that exceed those found in most of the legends of
analog synthesizer history.
So, for the synthesizer connoisseur, here’s the “brief” list of key features to whet your
appetite:
The ES2 features an advanced modulation matrix (known as the Router) in addition to a
number of “hard-wired” modulation routings. Basically, the concept of combining any
modulation source with any modulation target, is almost as old as the synthesizer itself.
Most important to the concept is a huge set of modulation targets, sources, a sufficient
number of modulation channels and modulation processes which can be inserted into
modulation channels. The ES2—featuring 10 modulation channels—represents the new
standard when it comes to matrix modulation. You can modulate parameters such as
filter resonance or the intensity of cutoff frequency modulation by Oscillator 1.

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The ES2’s three Oscillator concept is reminiscent of the Minimoog and EMS VCS 3. The
oscillators can be synchronized and ring-modulated. Pulsewidth modulation is also
possible. Oscillator 1 can be modulated in frequency by Oscillator 2, and is capable of
producing FM synth sounds.
Further to the classic, standard waveforms, the ES2’s Oscillators also feature 155 singlecycle waveforms, known as DigiWaves. Each has a totally different sonic color. The
DigiWave Parameter can be modulated, allowing dynamic cross-fades between the
waves. Intense use of such modulations will scroll through several waves, resulting in
sounds somewhat like those of wavetable synthesizers.
Two dynamic multi-mode filters, with a number of selectable slopes, guarantee the
fattest analog sounds. The filters can be mixed freely in parallel and series modes. The
filters can be overdriven by the Drive control.
Distortion, Chorus, Phaser, and Flanger effects are included. The unique Unison mode
delivers a density of sound normally only associated with big analog synthesizers,
including classic machines such as the Roland Jupiter 8, the SCI Prophet V and
Oberheim OB 8, amongst others. The great thing about the ES2’s Unison mode is that it
can be used in both monophonic and polyphonic modes. The unison voices are
intelligently spread across the entire stereo spectrum, but you can still modulate the
panorama position of the voices with any modulation source. This latter facility is
unprecedented.
As with the classic Yamaha CS Series analog synthesizers, a sine wave derived from
Oscillator 1 can be mixed directly into the dynamic stage to fatten up the sound. This
makes the penalties normally associated with the use of highpass, bandpass, and band
rejection filters much less of an issue.

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The ES2 Parameters

If given just a few words to explain the principles behind a subtractive synthesizer, it
would go something like this: The Oscillator generates the oscillation (or waveform),
the Filter takes away the unwanted overtones (of the waveform), and the Dynamic
Stage sets the volume of the permanent oscillation (the filtered waveform) to zero as
long as no key is pressed.
In an analog synthesizer, these three sections are commonly called the VCO, VCF, and
VCA, with VC being the abbreviation for Voltage Controlled, and the other letter(s)
standing for Oscillator, Filter, and Amplifier, respectively.
The basic parameters of a synthesizer are controlled (modulated) by voltages: pitch in
the oscillator, timbre in the filter, loudness in the amplifier.
These voltages are generated by modulation sources. In the ES2, the Router controls
which parameter is controlled by which modulation source.
Finally, the synthesizer’s sound is refined by effects such as distortion or chorus.
Following this simple signal path, we would like to introduce you to the modules of the
ES2.

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Oscillators
Tune
Tune sets the pitch of the ES2 in cents. 100 cents equals a semitone step. The range is
±50 cents. At a value of 0 c (zero cents), a' is tuned to 440 Hz or concert pitch.

Analog
Analog alters the pitch of each note, plus the cutoff frequency in a random fashion.
Much like polyphonic analog synthesizers, all three oscillators used by each synthesizer
voice maintain their specific deviation, but are shifted by the same amount randomly.
Medium values simulate the tuning instabilities of analog synthesizer circuitry which
can be useful in achieving that much sought after “warmth” of the ES2’s analog
hardware counterparts.

If the ES2 is set to mono or legato, Analog is only effective with unison activated. In this
situation, Analog sets the amount of detuning between the stacked (unison) voices.
If voices is set to 1 and Unison is not activated, the Analog parameter has no effect.
Read more on these parameters in the “Keyboard Mode (Poly/Mono/Legato)” section,
on page 227.
CBD
Fine Tuning detunes Oscillators 1, 2, and 3 in cents (1/100th percentages of a semitone).
The detuning results in “beats” (phasing), the speed of which is determined by the
difference between the two oscillator frequencies (given that these frequencies are
nearly identical). The higher the pitch, the faster the phasing beats are. High notes may
therefore seem to be further out of tune than lower notes.

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The CBD (Constant Beat Detuning) parameter matches this natural effect by detuning
the lower frequencies in a ratio proportionate to the upper frequencies. Besides
disabling CBD altogether, four values are at your disposal: 25%, 50%, 75%, 100%. If you
choose 100%, the phasing beats are (almost) constant across the entire keyboard
range. This value, however, may be too high, as the lower notes might be overlydetuned at the point where the phasing of the higher notes feels right. Try lower values
for CBD (and detune, of course) in cases where the bass appears to be a little too far out
of tune.
The reference pitch for CBD is C3 (middle C): Its detuning stays the same, regardless of
the CBD value.
Glide
The Glide parameter controls the portamento time. This is the time it takes for the pitch
to travel from one note to another. Glide behavior depends on the value of the
Keyboard Mode parameter setting.

If Keyboard Mode is set to Poly or Mono, and Glide is set to a value other than 0,
portamento is active. If Keyboard Mode is set to Legato, and Glide is set to a value other
than 0, you need to play legato (press a new key while holding the old one) to activate
portamento. If you do not play legato, portamento will be inactive. This behavior is also
called fingered portamento.
Bend Range
Bend Range determines the pitch range for pitch bend modulation. The range is ±36
semitones. There are separate range settings for upwards and downwards bends, plus
an optional link mode.

Keyboard Mode (Poly/Mono/Legato)
A polyphonic instrument allows the simultaneous playing of several notes—an organ or
piano, as examples. Many synthesizers are monophonic, especially the older ones. This
means that only one note can be played at a time, much like a brass or reed
instrument. This shouldn’t be viewed as a disadvantage in any way, because it allows
playing styles that are not possible with polyphonic keyboard instruments.

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You can switch between monophonic and polyphonic modes by clicking on the Poly
and Mono buttons. Legato is also monophonic, but with one difference: The envelope
generators are only retriggered if you play staccato (release each key before playing a
new key). If you play legato (press a new key while holding the old one), the envelope
generators are only triggered with the first note you play legato, and then continue
their curve until you release the last legato played key. If you switch to Mono, legato or
staccato playing does not have this impact: The envelope generators are retriggered
with every new note that is played.
Note: If you switch to Legato, you need to play legato to actually hear the Glide
parameter taking effect.
Note: On several monophonic synthesizers, the behavior in Legato mode is referred to
as single trigger, while Mono mode is referred to as multi trigger.
Voices
The maximum number of notes that can be played simultaneously is determined by
the Voices parameter. Maximum value for Voices is 32.
The value of this parameter has a significant impact on the computer processing
resources demanded by the ES2 when played at its maximum polyphony. Reduce this
value to the number of voices that you actually require for the part. Setting it to a
higher value places higher overheads on the CPU, and wastes resources.
Unison
A forte of polyphonic analog synthesizers has always been unison mode. Traditionally,
in unison mode, classic analog polysynths run monophonically, with all voices playing a
single note simultaneously. As the voices of an analog synthesizer are never perfectly in
tune, this results in an extremely fat chorus effect with great sonic depth. Switch the
ES2 to Mono or Legato and switch on Unison in order to achieve and hear this effect.
The intensity of the unison effect depends on the number of Voices selected.
Remember that the amount of processing power required is correlated to the number
of voices! The intensity of detuning (voice deviation) is set via the Analog parameter.

In addition to this classic monophonic unison effect, the ES2 also features a polyphonic
unison effect. In Poly/Unison, each note played is effectively doubled, or more correctly,
the polyphony value of the Voices parameter is halved. These two voices are then used
for the single triggered note.
Switching on Poly and Unison has the same effect as setting the ES2 to Mono, Unison,
and Voices = 2, except that it can be played polyphonically.

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Osc Start
The oscillators can run freely, or they can begin at the same phase position of their
waveform cycle each time you hit a key (every time the ES2 receives a note on
message).
When Osc Start (Oscillator Start) is set to free, the initial oscillator phase startpoint is
random, with each note played. This gives the sound more life and a less static feel—
just like an analog hardware synthesizer. On the other hand, the output level may differ
each time you play a note, and the attack phase may sound less punchy.

If you set Osc Start to soft, each initial oscillator phase will start at a zero crossing every
time a note is played. This mimics the sonic character of a normal digital synthesizer.

If Osc Start is set to hard, each initial oscillator phase begins at the highest possible
level in its waveform cycle every time a note is played. This punch is only audible if the
ENV3 Attack Time is set to a minimal value—a very fast attack. This setting is highly
recommended for electronic percussion and hard basses.

Note: Osc Start soft and hard result in a constant output level of the initial oscillator
phase every time the sound is played back. This may be of particular importance when
using Logic’s Bounce feature, at close to maximum recording levels.
Flt Reset
If you increase the filter’s Resonance parameter to higher values, it will begin to
internally feed back and will start to self-resonate. This resonance results in a sine
oscillation (a sine wave), which you may be familiar with, if you’ve used subtractive
synthesizers before.

In order to start this type of oscillation, the filter requires a trigger. In an analog
synthesizer, this trigger may be the noise floor or the oscillator output. In the digital
domain of the ES2, noise is all but eliminated. As such, when the oscillators are muted,
there is no input signal routed to the filter.
When Filter Reset is engaged, however, each note starts with a trigger that is used to
make the filter resonate immediately.

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Frequency Switch
Switches the pitch in semitone steps over a range of ±3 octaves. As an octave consists
of 12 semitones, the ±12, 24, and 36 settings represent octaves. You can click on these
options to quickly set the corresponding octave.

The value display works as follows: the left numbers show the semitone (s) setting, the
right numbers show the cent (c, 1 cent = 1/100th semitone) setting. You can adjust
these two values independently. As an example: an oscillator with the value 12 s 30 c
sounds an octave (12 semitones) and 30 cents higher than an oscillator with the value 0
s 0 c.
Note: The fifth (seven semitones), and all settings that correspond with harmonics of
an oscillator set to 0 semitones (19, 28 semitones) result in harmonic sounds.
Muting Oscillators
By clicking on the green numbers to the right of the oscillators, you can mute and unmute them independently. This saves processor power.
Wave
Each of the three oscillators features a rotary knob that allows you to select a
waveform. This is responsible for the basic harmonic content and tone color of a sound.
Oscillators 2 and 3 are almost identical to each other, but differ from Oscillator 1.
Oscillator 1 is capable of generating a sine wave, the frequency of which can be
modulated in the audio range, for true FM synthesis sounds. Oscillators 2 and 3 can be
synchronized to, or be ring-modulated with, Oscillator 1. They also feature rectangular
waves with freely definable fixed pulse widths plus pulse width modulation (PWM)
facilities. Via the Router, the rectangular and pulse waves of Oscillator 1 can be
modulated in width in conjunction with the synchronized and ring-modulated
rectangular waves of Oscillators 2 and 3.
Note: The Filter button disables the entire filter section. This makes listening to the pure
oscillator waveforms easier.

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Oscillator 1 Waveforms
Oscillator 1 outputs standard waveforms—pulse, rectangular, sawtooth, and triangular
waves—or, alternately, any of the 155 available DigiWaves. It can also output a pure sine
wave.

The sine wave can be modulated in frequency by Oscillator 2 in the audio frequency
range. This kind of linear frequency modulation is the basis on which FM synthesis
works. FM synthesis was popularized by synthesizers such as Yamaha’s DX7 (the
architecture of which is much more complex, when it comes to FM synthesis).
A click on the Oscillator number disables the output of Oscillator 1.
Note: Even when Oscillator 1 is turned off, it is still available for use as a modulation
and synchronization source for Oscillators 2 and 3.
We will now take a closer look at the different waveforms available to Oscillator 1.

Screenshot of ES2’s Oscillator 1, sine wave selected. The sample was created with
Logic’s Bounce function and is displayed in the Sample Editor. The sine wave shown is
in its basic oscillation form. It contains no harmonics. According to the theorem of Jean
Baptiste Fourier, all regular waveforms can be interpreted as the sum of sine oscillations
with defined frequency, amplitude, and phase position, with their frequencies being
“harmonic”—having integer frequency ratios, in other words.

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During a Fourier transformation, complex oscillations can be divided into their basic
sine components. In additive synthesis, complex oscillation forms can be resynthesized. The most simple additive synthesizer is the drawbar organ (the Hammond
organ, for example). With such an organ, you can mix nine sine choirs with drawbars.
Try selecting sine waves for all three oscillators and the following semitone
settings: −12 (16'), 0 (8') and +7 (5 1/3'), and set all oscillators to the same level. Select
an organ envelope, and voila—a classic organ sound!

Screenshot of the ES2’s triangular wave, created and shown as above. The triangular
wave only contains odd harmonics (no octaves), the amplitudes of which decrease
square-proportionately to their number. This means that its sound has few overtones.
This corresponds with its appearance, which is reminiscent of a sine wave.
Classic synthesizer literature encourages the use of the triangular wave for the creation
of flute-like sounds. In the age of sampling, however, it’s pretty hard to sell a triangular
wave as a flute sound to anyone.

Screenshot of the ES2’s sawtooth wave. The sawtooth wave contains all harmonics, the
amplitudes of which decrease proportionately with their number.

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Classic synthesizer literature indicates the use of the sawtooth wave to create a sound
similar to that of a violin. The rich and full sound of the sawtooth wave is the most
popular synthesizer waveform, and serves as a basis for synthetic string and brass
sounds. It is also handy for synthesized bass sounds.

Screenshot of the ES2’s rectangular wave. The 50% rectangular wave contains all odd
harmonics, the amplitudes of which decrease proportionately with their number. The
pulse width can be set to any value and serves as a modulation address.
Classic synthesizer literature likens the rectangular wave to the sound of a clarinet, as
the clarinet doesn’t feature any even harmonics in a certain frequency range either. The
typical hollow sound of the rectangular wave is achieved with its pulse width value set
to 50%. Heavily lowpass-filtered rectangular waves are popular as techno bass sounds.
Changing the pulse width to something around 75% results in quite a nasal sound,
resembling an oboe.
Digiwaves
The ES2, as we’ve mentioned, not only features the most popular synthesizer
waveforms, but also a selection of 100 additional waveforms, called Digiwaves.
Note: You can select the Digiwave by touching its name and moving the mouse
vertically. You can select it numerically when you hold Shift.
The number of the Digiwave is a parameter that can be modulated. By modulating the
OscWave target, you can scroll through the list of Digiwaves. Choose a sufficiently low
modulation intensity and speed to hear the single Digiwaves being crossfaded. The
Digiwaves of the three oscillators can be modulated individually or together. The
modulation targets are discussed in “OscWaves”, on page 255, through to “OscWaveB”,
on page 256.
Given the ability to modulate Digiwaves, the ES2 can produce sounds resembling those
of the famous wavetable synthesizers from PPG and Waldorf (and also the Korg
Wavestation).

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Linear Frequency Modulation
The principle of linear frequency modulation (FM) synthesis was developed in the late
sixties and early seventies by John Chowning. It’s such a flexible and powerful method
of tone generation that synthesizers were developed, based solely on the idea of FM
synthesis. The most popular FM synthesizer ever built is Yamaha’s DX7. FM synthesis is
also found in other models of the Yamaha DX range and some Yamaha E-Pianos. In the
discipline of pure FM synthesis, the ES2 can’t be compared with these synthesizers, but
it can certainly achieve some of their signature sounds.

Note: The DX7 featured six oscillators (called operators) per voice, with each controlled
by a separate envelope generator. It’s bell-like, synthetic electric piano sounds became
de rigeur in popular music, especially in soul ballads (Whitney Houston and others).
Between the Sine setting (when the sine symbol is selected) and the FM symbol of the
Oscillator 1 knob, a range is available which allows stepless control over the amount of
frequency modulation. This parameter is also available as a modulation address.
Note: Osc1Wave is optimized for subtle FM sounds, utilizing lower FM intensity
amounts. For more extreme FM modulations, the Router offers the Osc1WaveB target.
See the “Modulation Targets”, on page 254.
The frequency of Oscillator 1 can be modulated by the output signal of Oscillator 2.
Whenever it outputs a positive voltage, the frequency of Oscillator 1 will increase.
Whenever it is negative, its frequency will decrease.
The effect is similar to an LFO modulation, being used to create a vibrato (a periodic
modulation of the frequency) or a slow siren effect. In comparison to an LFO, Oscillator
2 does not oscillate slowly. In the audio domain, it oscillates a little faster than Oscillator
1 itself. Thus, Oscillator 1’s oscillation also accelerates and slows down within a single
phase, resulting in a distortion of the basic sine shape of Oscillator 1. This sine wave
distortion has the side benefit of a number of new harmonics becoming audible.

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Oscillator 1’s frequency modulated sine wave, modulated by Oscillator 2 set to sine
wave. Oscillator 2 was set to three times the frequency of Oscillator 1 (+19 semitones).
The modulation intensity is low (Wave control at about 12 o’clock). As the wavelength
(the duration period) of the modulating Oscillator is a third of that of the modulated
Oscillator, the sine is accelerated and slowed down three times within a phase.

Oscillator 1’s frequency modulated sine wave, modulated by Oscillator 2 set to sine
wave. Oscillator 2 was set to three times the frequency of Oscillator 1 (+19 semitones).
The modulation intensity is much higher (Wave control at about 3 o’clock). The
distortion of the basic sine wave is much stronger, resulting in more harmonics
becoming audible.
The effect of frequency modulation depends on both the modulation intensity and
frequency ratio of both Oscillators.

The upper graphic shows a slightly deformed sine wave of Oscillator 1, modulated in
frequency by Oscillator 2 at double the speed of the carrier (Oscillator 1). The resulting
waveform resembles a rectangular wave or clipped sine wave.

The upper graphic shows a slightly frequency modulated sine as the output signal of
Oscillator 1, with the modulator frequency being identical to the carrier frequency. The
resulting waveform resembles a lowpass filtered sawtooth wave.
The resulting spectrum not only depends on the frequency modulation intensity and
frequency ratio, but also on the waveform used by the modulating oscillator (Oscillator
2). The modulation that takes place varies according to the waveform selection for
Oscillator 2—it might even be an oscillation that is synchronized to Oscillator 1! Given
the 100 available Digiwaves, countless combinations of modulation intensities and
frequency ratios, the frequency modulation of the two oscillators delivers an infinite
pool of spectra and tone colors.

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Waveforms of Oscillators 2 and 3
Basically, Oscillators 2 and 3 supply the same selection of analog waveforms as
Oscillator 1: sine, triangular, sawtooth, and rectangular waves. The pulse width can be
scaled steplessly between 50% and the thinnest of pulses, and can be modulated in a
number of ways (see “Pulse Width Modulation” section, on page 236).
Oscillators 2 and 3 also offer the selection of:
• a rectangular wave, synchronized to Oscillator 1,
• a sawtooth wave, synchronized to Oscillator 1,
• a ring modulator, which is fed by the output of Oscillator 1 and a square wave from
Oscillator 2,
• colored noise for Oscillator 3.
Synchronization and ring modulation afford the creation of very complex and flexible
harmonic spectra. The principle of Oscillator synchronization is described on page 237,
and ring modulation on page 238.
Pulse Width Modulation
Oscillators 2 and 3 allow you to scale the width of the pulses to any value. The
spectrum and tone color generated by these oscillators depends on the pulse width.
The pulse width can be modulated. You can even modulate the pulse width of the
square and pulse wave of Oscillator 1, the pulse width of the synchronized pulse waves
of Oscillator 2 and 3, and the square wave of Oscillator 2’s ring modulator.

This width modulation is controlled in the Router (the Modulation Matrix). The pulse
width is defined by the waveform rotary control. The graphic below shows a pulse
wave, with the pulse width modulated by an LFO. You can clearly see how the width of
the pulses changes over time.

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Screenshot of an ES2 sample, created with the Bounce function of Logic. The pulse
width is modulated. It is easy to see how the width of the pulses varies between a
rectangular shape and very thin pulses. An LFO is selected as the modulation source,
and its waveform is a sine wave. You can see about half a phase of the sine wave. If a
rectangular wave had been selected for the LFO, you would see the pulse width
periodically changing between the two fixed extreme values.
Note: A pulse wave (with its width modulation controlled by an LFO set to a sine wave)
makes a single oscillator sound vivid, undulating, and rich with overtones. Sonically,
this is somewhat like the sound of two slightly detuned, phasing oscillators. The effect
sounds great with sustained bass and pad sounds. Select the intensity and speed of the
modulation with care, as the overall volume (and level of the first partial) decreases,
and slight detuning occurs, when the pulses become very thin (below 10%).
Note: Pulse width modulations via velocity sensitive envelope generators sound very
dynamic—a great effect, especially for percussive bass sounds.
Sync
You can see that the rectangular and sawtooth waveforms also feature a Sync option. In
this mode, the frequency of Oscillator 2 (or 3, respectively) is synchronized to the
frequency of Oscillator 1. This does not mean that their frequency controls are simply
disabled. They still oscillate at their selected frequencies, but every time that Oscillator
1 starts a new oscillation phase, the synchronized Oscillator is also forced to restart its
phase from the beginning. This is practical while the frequency of the synchronized
Oscillator is significantly higher than that of Oscillator 1.
The graphic shows the output signal of Oscillator 2, set to a synchronized sawtooth
wave. Its frequency is about three and a half times that of Oscillator 1. Oscillator 1 was
set to 0 semitones, and Oscillator 2 to +22 semitones. The oscillogram clearly shows
how the phase of Oscillator 2 is forced to restart after about three and a half phases.
Between the pulses of Oscillator 1, Oscillator 2 runs freely.

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Synchronized sawtooth wave as output by Oscillator 2. Oscillator 1 is set to 0, Oscillator
2 to +22 semitones. The dots in the graphic indicate the phases of Oscillator 1.

Synchronized rectangular wave as output by Oscillator 2. Oscillator 1 is set to 0,
Oscillator 2 to +22 semitones. The dots in the graphic indicate the phases of Oscillator
1.

Output signal of Oscillator 2 (sawtooth), synchronized to Oscillator 1. The frequency of
Oscillator 2 is modulated by an envelope generator, changing the duration of the
phases over time. On the right-hand side, the sawtooth shapes are broader, due to the
sinking frequency of Oscillator 2.
At regular intervals, defined by the phase duration of Oscillator 1, the waveform is
forced back to its beginning. The dots in the graphic indicate the phases of Oscillator 1.
The distance between these dots remains constant, as the frequency of Oscillator 1 is
not being modulated.
Synchronized oscillator sounds are especially cool when the frequency of the
synchronized oscillator is modulated by an envelope generator. This way, the number
of phases within a section (phase) of the synchronization cycle always changes, and so
does the spectrum. Typical oscillator sync sounds tend towards the aggressive,
screaming leads that synthesizer manufacturers like to talk about.
Ring (Ring Modulation)
The waveform control of Oscillator 2 also features the Ring setting. In this mode,
Oscillator 2 outputs the signal of a ring modulator. This ring modulator is fed with the
output signal of Oscillator 1 and a square wave of Oscillator 2. The pulse width of this
square wave can be modulated.

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A ring modulator has two inputs. At it’s output you will find the sum and difference
frequencies of the input signals.

The graphic shows the output signal of the ring modulator, appearing as the output
signal of Oscillator 2. The amplitude (or the elongation, to be more exact) of the output
of Oscillator 2 changes with the phase of Oscillator 1. Oscillator 2 is set to a higher
frequency than Oscillator 1. As the frequency ratio is odd (irrational), the resulting
waveform always changes over time. The resulting spectrum is inharmonic and you
won’t hear a clearly defined pitch.
If you ring-modulate a sine oscillation of 200 Hz with a sine oscillation of 500 Hz, the
output signal of the ring modulator will consist of a 700 Hz (sum) and a 300 Hz
(difference) signal. Negative frequencies result in a change of the phase polarity of the
output signals. With sawtooth and rectangular input signals, the output signal is much
more complex, as these harmonically-rich waveforms produce a number of extra side
bands.
Note: Ring modulation is a powerful tool for inharmonic, metallic sounds as the spectra
resulting from its use are inharmonic with almost every frequency ratio. The ring
modulator was the tool of choice for bell-like sounds, and dates back to the early days
of the synthesizer (see the “RingShifter—Ring Modulator/Frequency Shifter” section, on
page 101).

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White and Colored Noise (Oscillator 3 only)
Unlike Oscillator 2, Oscillator 3 is not capable of producing ring modulated signals nor
sine waves. Its sonic palette however, is bolstered by the inclusion of a noise generator.
By default, Oscillator 3’s noise generator generates white noise. White noise is defined as
a signal that consists of all frequencies (an infinite number) sounding simultaneously, at
the same intensity, in a given frequency band. The width of the frequency band is
measured in Hertz. Its name is analogous to white light, which consists of a mixture of
all optical wavelengths (all rainbow colors). Sonically, white noise falls between the
sound of the consonant F and breaking waves (surf ). Synthesis of wind and seashore
noises, or electronic snare drum sounds, requires the use of white noise.

Screenshot of a white noise sample, showing the output signal of Oscillator 3.
Oscillator 3 has more up its sleeve than the output of neutral sounding white noise—
you also can make it hiss or rumble. Even better, you can modulate this sound color in
real-time without using the main filters of the ES2.
If the waveform of Oscillator 3 is modulated, (Osc3Wave) the color of the noise will
change. It can be filtered by a dedicated highpass or lowpass filter with a descending
slope of 6 dB/octave. At negative values, the sound becomes darker (red); The lowpass
filter can be tuned down to 18 Hz with a setting of −1. When Osc3Wave is modulated
positively, the noise becomes brighter (blue): At a value of +1 for Osc3Wave, the
highpass filter’s cutoff frequency is set to 18 kHz. This filtering of the noise signal is
independent of the main filters of the ES2, and can be automatically changed in realtime.
Oscillator Mix Field—the Triangle
By grabbing, and moving the cursor in the Triangle, you can crossfade between the
three oscillators. This is pretty self-explanatory. Moving the cursor along one of the
Triangle’s sides will crossfade between two oscillators, with the third oscillator being
muted.

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The position of the cursor can be controlled via the vector envelope, just like the cursor
position in the Track Pad (the Square), which we’ll look at in “The Square” section, on
page 272.
Note that the vector envelope features a loop function. This addition extends its
usefulness, allowing you to view it as a luxurious pseudo-LFO with a programmable
waveform. It can be used for altering the positioning of the Triangle and Square
cursors. Read more about this in the “Vector Mode” section, on page 272, and “The
Vector Envelope” section, on page 273.
Triangle Values in Control View
Internally, the position of the cursor in the Triangle is described by two parameters (coordinates, actually) which are relevant when it comes to automating the oscillator mix.
These parameters, called OscLevelX and OscLevelY—are visible in the Controls view.
Don’t confuse them with the X and Y positions of the Square.
If you intended to edit a Region containing oscillator mix automation data in the Hyper
Editor, you would use the following MIDI controller values. Take a look at the
information below to get a feel for how they operate.
In order to listen to Oscillator 1 only …
• choose OscLevelX= 1.0 (MIDI: 127) and OscLevelY= 1.154 (MIDI: 127).
In order to listen to Oscillator 2 only …
• choose OscLevelX= 0.0 (MIDI: 0) and OscLevelY= 0.577 (MIDI: 64).

In order to listen to Oscillator 3 only …
• choose OscLevelX= 1.0 (MIDI: 127) and OscLevelY= 0.0 (MIDI: 127).
In order to listen to Oscillator 1 and 2 only …
• choose OscLevelX= 0.5 (MIDI: 64) and OscLevelY= 0.866 (MIDI: 95).
In order to listen to Oscillator 1 and 3 only …
• choose OscLevelX= 1.0 (MIDI: 127) and OscLevelY= 0.577 (MIDI: 64).
In order to listen to Oscillator 2 and 3 only …
• choose OscLevelX= 0.5 (MIDI: 64) and OscLevelY= 0.288 (MIDI: 32).

All Oscillators have the same level at …
• OscLevelX= 0.667 (MIDI: 85) and OscLevelY= 0.577 (MIDI: 64).

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Filters
The ES2 features two dynamic filters which are equivalent to the Voltage Controlled
Filters (VCF) found in the world of analog synthesizers. The two filters are not identical.
Filter 1 features several modes: lowpass, highpass, bandpass, band rejection, peak. Filter
2 always functions as a lowpass filter. Unlike Filter 1, however, Filter 2 offers variable
slopes (measured in dB/octave).

The Filter button bypasses (switches off ) the entire filter section of the ES2. Disabling
the filters makes it easier to hear adjustments to other sound parameters, as the filters
always heavily affect the sound. If the word Filter is green, the filters are engaged.
Disabling the filters reduces processor load.

Series and Parallel Filter Signal Flow
You can rotate the entire (circular) filter segment of the ES2 user interface. Press the
button labelled Parallel or Series to do so. The label and position/direction of the filter
controls clearly indicate the current signal flow.

In the position displayed above, the filters are cabled serially. This means that the signal
of the Oscillator Mix section (Triangle) passes through the first filter, and then this
filtered signal passes through Filter 2—if Filter Blend is set to 0 (middle position). See
the “Filter Blend: Cross-Fading the Filters” section, on page 243 for a detailed
description of this parameter.
The mono output signal of Filter 2 is then fed into the input of the dynamic stage (the
equivalent of a VCA in an analog synthesizer), where it can be panned in the stereo
spectrum, and then fed into the effects processor.

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In the graphic below, the filters are cabled in parallel. If Filter Blend is set to 0, you’ll hear
a 50/50 mix of the source signal routed via Filter 1 and Filter 2, which is fed into the
mono input of the dynamic stage. There it can be panned in the stereo spectrum, and
then fed into the effects processor.

Filter Blend: Cross-Fading the Filters
You can cross-fade the two filters. When wired in parallel, you’ll find this quite obvious
to look at and understand: If Filter Blend is set to its top position, you’ll only hear Filter 1.
If Filter Blend is set to its lowest position, you’ll only hear Filter 2. In between these
positions, the filters are cross-faded.
More often, you’ll want to cable the filters serially. It should be noted that even in series
mode, it’s possible to cross-fade the filters! This is achieved through the use of
controllable side chains (bypassing lines). In this serial cabling scenario, the distortion
circuits controlled by the Drive parameter also need to be considered, as the distortion
circuits are positioned before or in-between the filters, dependent on Filter Blend. All
fades occur flawlessly, in very fine steps.
The Filter Blend parameter can be modulated dynamically in the Router!
Filter Blend and Signal Flow
No matter whether parallel or series filter configurations are chosen, a Filter Blend
setting of −1 always results in only Filter 1 being audible. A Filter Blend setting of +1 will
limit audibility to Filter 2. This is reflected in the user interface.

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In conjunction with the overdrive/distortion circuit (Drive) and a series wiring
configuration, the ES2’s signal flow is anything but commonplace. The graphics
illustrate the signal flow between the Oscillator Mix stage (the Triangle) and the
dynamic stage, which is always controlled by ENV 3. The signal flow through the filters,
the overdrives and the bypassing sidechains is dependent on the Filter Blend setting.

Filter Blend in parallel filter mode.

Filter Blend in Serial filter mode. Between 0 and −1 two distortion
circuits are active—one “pre” each filter. Filter Blend cross-fades up to
three bypassing lines simultaneously.

Filter Blend and Serial Filter Configuration Tips
• With positive values for Filter Blend, Filter 1 is partially bypassed.

• With negative values for Filter Blend, Filter 2 (the low pass) is partially bypassed.
• At zero and positive values for Filter Blend, there is only one overdrive circuit for both

filters.
• With negative values, another overdrive circuit is introduced, distorting the output

signal of the Oscillator Mix stage before it is fed into the first filter—where Filter Blend
= −1.
Note: If Drive is set to 0, no distortion occurs.

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Filter Blend and Parallel Filter Configuration Tip
The overdrive/distortion circuit is always wired after the Oscillator Mix stage and before
the filters. The filters receive a mono input signal from the overdrive circuit’s output.
The outputs of both filters are mixed to mono via Filter Blend.

Note: If Drive is set to 0, no distortion occurs.
Drive
The filters are equipped with separate overdrive modules. Overdrive intensity is defined
by the Drive parameter. If the filters are connected in parallel, the overdrive is placed
before the filters. If the filters are connected serially, the position of the overdrive
circuits depends on the Filter Blend parameter—as described above.

Polyphonic Distortions In the Real World
The ES2 features a distortion effect, equipped with a tone control, in the Effects section.
Given the inclusion of this effect, you may be wondering what benefit the Drive
function in the Filter section brings?
The distortion circuit in the Effects section affects the sum of the entire polyphonic
synthesizer performance. Thus, more complex chords (other than major chords, parallel
fifths and octaves) sound rough, when using distortion. Every rock guitarist knows this.
Due to these intermodulation distortions, distorted guitar playing is generally
performed by using few voices or parallel fifths and octaves.
The Filter Drive affects every voice individually—and when every voice in the ES2 is
overdriven individually (like having six fuzz boxes for the guitar, one for each string),
you can play the most complex harmonies over the entire keyboard range. They’ll
sound clean, without unwanted intermodulation effects spoiling the sound.
Furthermore, appropriate Drive parameter settings lead to character. To explain, the
way analog filters behave when overdriven forms an essential part of the sonic
character of a synthesizer. Each synthesizer model is an individual with regard to the
way its filters behave when overdriven. The ES2 is extremely flexible in this respect,
allowing for the most subtle fuzz through to the hardest distortions.
Finally, in series mode, the distortion always takes place before the lowpass filter (Filter
2). As the lowpass filter (Filter 2) can filter (cut) away the overtones introduced by the
distortion, the Drive feature can be seen (and used) as another tool for deforming the
oscillator(s) waveforms.

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To check out how the overdrive circuit between the filters works, program a sound as
follows:
• Simple static waveform (a sawtooth)
• Filter set to Serial mode
• Filter Blend set to 0 (center position)
• Set Filter 1 to peak Filter Mode
• Set a high Resonance value for Filter 1
• Modulate Cutoff Frequency 1 manually or in the Router
• Set Drive to your taste
• Filter away (cut) the high frequencies with Filter 2 to taste
The sonic result resembles the effect of synchronized oscillators. With high resonance
values, the sound tends to scream. Modulate the Resonance of Filter 1, if you wish.
Filter Parameters
Cutoff and Resonance
With every lowpass filter (in the ES2: Lo mode for Filter 1 and all of Filter 2’s modes), all
frequency portions above the Cutoff Frequency (Cut) are suppressed, or cut off, hence
the name. The Cutoff Frequency controls the brilliance of the signal. The higher the
Cutoff Frequency is set, the higher the frequencies of signals that are allowed to pass
through the lowpass filter.

Resonance (Res) emphasizes the portions of the signal which surround the frequency
defined by the Cutoff Frequency value. This emphasis can be set so intensively in Filter 2,
that the filter begins to oscillate by itself. When driven to self-oscillation, the filter
outputs a sine oscillation (sine wave). This self-oscillation can be supported by the Filter
Reset parameter. See “Flt Reset” on page 229 for details.

Note: If you are new to synthesizers, experiment with a simple saw wave, using
Oscillator 1, and Filter 2 (lowpass filter, Filter Blend = +1) on its own. Experiment with
the Cutoff Frequency and Resonance parameters. You’ll quickly learn how to emulate a
number of recognizable sounds, and will pick up the basic principles of subtractive
synthesis intuitively.

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Note: The dynamic lowpass filter is the most essential module in any subtractive
synthesizer. This is why Filter 2 always operates in lowpass mode.
Note: As opposed to the filter and EQ effect plug-ins in Logic, the ES2’s filters are
dynamic, which means that the Cutoff Frequency parameter can be modulated
extremely quickly and severely in real-time—even on modulation signals in the audio
frequency range.
The Chain Symbols
Manipulating the Cutoff and Resonance controls in real-time is one of the key
ingredients in the creation of expressive synthesizer sounds. You’ll be pleased to know
that you can control two filter parameters at once by dragging on one of the three little
chain symbols in the filter graphic.
• The chain between Cut and Res of Filter1 controls Resonance (horizontal mouse
movements) and Cutoff (vertical mouse movements) of the first filter simultaneously.
• The chain between Cut and Res of Filter2 controls Resonance (horizontal mouse
movements) and Cutoff (vertical mouse movements) of the second filter
simultaneously.
• The chain between Filter1’s Cut and Filter2’s Cut controls Cutoff (vertical mouse
movements) of the first filter, and Cutoff (horizontal mouse movements) of the
second filter simultaneously.
Filter Slope
A filter can not completely suppress the signal portion outside the frequency range
defined by the Cutoff Frequency parameter. The slope of the filter curve expresses the
amount of rejection applied by the filter (beneath the cutoff frequency) in dB per
octave.

Filter 2 offers three different slopes: 12 dB, 18 dB and 24 dB per octave. Put another way,
the steeper the curve, the more severely the level of signals below the cutoff frequency
are affected in each octave.
Fat
Increasing the Resonance value results in a rejection of bass (low frequency energy)
when using lowpass filters. The Fat switch compensates for this side-effect, delivering a
bassier sound.

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Filter Mode (Lo, Hi, Peak, BR, BP)
Filter 1 can operate in several modes, allowing the filtering (cutting away) and/or
emphasis of specific frequency bands.

• A lowpass filter allows frequencies that fall below the cutoff frequency to pass. When

•

•
•

•

set to Lo, the filter operates as a lowpass filter. The slope of Filter 1 is 12 dB/octave in
Lo mode.
A highpass filter allows frequencies above the cutoff frequency to pass. When set to
Hi, the filter operates as a highpass filter. The slope of Filter 1 is 12 dB/octave in Hi
mode.
In Peak mode, Filter 1 works as a Peak Type Filter. This allows for a level increase in a
frequency band, the width of which is controlled by the Resonance parameter.
The abbreviation BR stands for band rejection. In this mode, the frequency band (a
range of adjacent frequencies) directly surrounding the cutoff frequency is rejected,
whilst the frequencies outside this band can pass. The Resonance parameter controls
the width of the rejected frequency band.
The abbreviation BP stands for bandpass. In this mode, only the frequency band
directly surrounding the cutoff frequency can pass. All other frequencies are cut. The
Resonance parameter controls the width of the frequency band that can pass. The
bandpass filter is a two-pole filter with a slope of 6 dB/octave on each side of the
band.

Impact of the Filters on the Waveform
Below, you’ll find a number of oscillograms of a sawtooth wave generated by Oscillator
1. These images illustrate the effect of the various modes of Filter 1. The cutoff
frequency was set so that it is equal to the frequency of the first overtone (twice the
frequency of Oscillator 1). The duration and wavelength of this overtone (the second
harmonic) is half as long as the duration and wavelength of the first harmonic (the
fundamental tone).

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Lowpass Filter

Lowpass-filtered sawtooth wave, Cutoff Frequency one octave above the frequency of
the sawtooth. As the slope isn’t infinite, harmonics are still visible. Note that the
waveform is rounded.
Highpass Filter

Highpass-filtered sawtooth wave, Cutoff Frequency one octave above the frequency of
the sawtooth. The basic harmonic is rejected by about 12 dB, as the slope is 12 dB/
octave. Try to mentally add a sine curve with the basic wavelength to the graphic: The
sum of both would result in a sawtooth wave.
Peak Type Filter

The second partial is not easy to see, even though all other partials are rejected.
Peak Type Filter with Resonance

Here, the second partial is strongly emphasized.

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Band Rejection

In this image, the second partial is rejected. The basic oscillation and all other
harmonics are present. Sawtooth is still the input waveform.
Bandpass Filter

The bandpass filter distorts the picture of the sawtooth wave.
Filter 2 FM
The cutoff frequency of Filter 2 can be modulated by the sine wave of Oscillator 1,
which means that it can be modulated in the audio frequency range.

The effect of such filter modulations in the audio spectrum is unpredictable, but the
results tend to remain harmonic as long as the modulation intensity doesn’t get too
high. FM defines the intensity of frequency modulation. This parameter can be
modulated in real-time: In the Router, this modulation is abbreviated as LPF FM.
A clean sine wave, at the frequency of Oscillator 1, is always used as the modulation
source.
Note: Don’t confuse this type of filter frequency modulation with the FM feature of
Oscillator 1, which can be modulated by Oscillator 2, as described in the “Linear
Frequency Modulation” section, on page 234.
Note: If a frequency modulation of Oscillator 1 by Oscillator 2 is used, it does not
influence the (sine wave) signal used to modulate the cutoff frequencies.

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Note: Filter 2 can be driven to self-oscillation. If you set a very high value for Resonance,
it will produce a sine wave. This self-oscillating sine wave will distort at the maximum
Resonance value. If you mute all oscillators, you’ll only hear this sine oscillation. By
modulating the Cutoff Frequency, you can produce effects similar to those produced
by modulating the frequency of Oscillator 1 with Oscillator 2.
Handling Processing Power Economically
The ES2 is designed to make the most efficient use of computer processing power.
Modules and functions that are not in use don’t use processing power. This principle is
maintained by all elements of the ES2.
As examples: If only one of the three oscillators is in use, and the others are muted, less
processing power is required. If you do not modulate Digiwaves, or if you disengage
the filters, processing power is saved. When it comes to filtering, here are some hints
that will help you to use processing power most efficiently:
• If you can achieve the same lowpass-filtered sound with Filter 1 as with Filter 2, use
Filter 1. Filter 1 uses less processing power, although it differs a little, sonically.
• Filter FM uses additional processing power. If you don’t need it, don’t use it.
• Modulation of the Filter Blend parameter requires quite a bit of additional processing
power, as soon as it is engaged in the Router.
• Drive requires additional processing power. This is especially the case when it comes
to filters wired in series and Filter Blend settings with two distortion circuits. See “Filter
Blend and Signal Flow”, on page 243 for details.
Remember that you’ll never be forced to make compromises in your sound! You can
always make use of Logic’s Bounce features in order to convert a processor-intensive
Audio Instrument track into an audio track, playing back a bounced audio file. To do so,
route the Audio Instrument (the ES2) to an Output Object. Switch the ES2 track solo.
Set the locators in the Transport window. Press Bounce in the Output Object (in the
Audio Mixer). Select Bounce & Add. After the Bounce procedure, drag the resulting file
from the Audio window into the Arrange window, onto a stereophonic audio track.
Save the ES2 setting. Mute the bounced Audio Instrument track. Don’t delete anything.
Save the song. If you want to change the notes, or tempo, recording level, or sound,
repeat the entire procedure.
You can also use Logic’s Freeze facility to perform individual offline bounce processes
for each track. To Freeze, simply click on the Freeze button (the ice crystal) of the
desired Audio Instrument track(s) in the Arrange window track list. The next time you
hit Play, Logic will Freeze the tracks, saving massive amounts of processing power.

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Dynamic Stage (Amplifier)
The dynamic stage defines the level—which means the perceived volume—of the
played note. The change in level over time is controlled by an envelope generator.
ENV3 and the Dynamic Stage
ENV3 is hard-wired to the dynamic stage—envelope generator 3 is always used for
control over the level of the sound. For detailed explanations of the envelope
parameters, see “The Envelopes (ENV 1—ENV 3)” section, on page 268.

Router Modulation Target: Amp
The dynamic stage can be modulated by any modulation source in the Router. The
modulation target is called AMP in the Router.
Note: If you select AMP as the target, LFO1 as the Source, and leave via set to Off in the
Router, the level will change periodically, based on the current Rate of the LFO—and
you’ll hear a tremolo.
Sine Level

Sine Level allows the mixing of a sine wave (at the frequency of Oscillator 1) directly into
the dynamic stage, independent of the filters. Even if you have filtered away the basic
partial tone of Oscillator 1 with a highpass filter, you can reconstitute it through use of
this parameter. Please note:
• In cases where Oscillator 1 is frequency modulated by Oscillator 2 (if you have turned
up FM with the waveform selector), only the pure sine wave is mixed into the
dynamic section, not the distorted FM waveform.
• Low frequency modulations of Oscillator 1’s pitch, set in the Router, affect the sine
wave frequency mixed in here.
Note: Sine Level is well suited for adding warmth and a fat bass quality to the sound.
Thin sounds can be fattened up with this function, given that Oscillator 1 actually plays
the basic pitch.

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The Router
The ES2 features a modulation matrix, called the Router. Any modulation Source can be
connected to any modulation Target—much like an old-fashioned telephone exchange
or a studio patchbay. The modulation intensity—how strongly the Target is influenced
by the Source—is set with the associated vertical slider.
Note: To set the modulation intensity to zero, just click on the little zero symbol (the
small circle) right beside via.
The intensity of the modulation can also be modulated: The via parameter determines
yet another modulation source, which defines the amount or intensity of the
modulation.
Ten such modulations of Source, via, and Target can take place simultaneously, in
addition to those which are hard-wired outside the Router. The bypass (b/p) parameter
allows the disabling/enabling of individual modulation routings without losing
settings.
Note: The bypass parameter is not available in Logic 5.x. If you would like to exchange
ES2 settings that utilize bypassed modulation routings with Logic 5.x users, you should
set these Targets to off.
Note: Some modulations aren’t possible, due to technical reasons. As an example, the
envelope times can be modulated by parameters that are only available during a noteon message. Therefore, there are situations where the envelopes are not available as
Targets. Furthermore, the LFO 1 can’t modulate its own frequency. Values that are not
available are grayed out.
Note: You may need to switch from the Vector Envelope display to that of the Router.
The Range of Via
The modulation intensity is set with the vertical slider. This is self-explanatory, as long
as the via parameter is set to off. This ensures that the modulation intensity is constant,
if not affected by any other controller (such as the modulation wheel or aftertouch).
As soon as you select a value other than off for via, the slider is divided into two halves.
The lower half defines the minimum intensity of the modulation, when the via
controller is set to its minimum value. The upper half defines the maximum modulation
intensity when the via modulator (the modulation wheel, in this case) is set to its
maximum value. The area between the two slider halves defines the range that is
controlled by the via controller.
You can grab the area between the two slider halves with the mouse and drag both
halves simultaneously. If this area is to small to be grabbed with the mouse, just click
on a free part of the slider track and move the mouse up or down to move the area.

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In the example below, the lower half of the slider knob defines the vibrato intensity
when the modulation wheel is turned down. The upper half defines the vibrato
intensity that takes place when the modulation wheel is set to its maximum value.

Note: To invert the effect of the via modulation source, simply activate the Via invert
(inv) parameter in the Router.
A Modulation Example
Say you’ve chosen these settings:
• Target = Pitch 123
• via = Wheel
• Source = LFO1
• Modulation intensity = slider position, set as desired
In this configuration, the modulation source—LFO1—is used to modulate the
frequency (pitch) of all three Oscillators (Pitch 123). (Pitch 123) is the modulation target
in this example. You’ll hear a vibrato (a modulation of the pitch) at the speed of LFO 1’s
Rate. The modulation intensity is controlled by the (modulation) wheel, which is
determined by the via parameter. This provides you with control over the depth of
vibrato (pitch modulation) via the modulation wheel of your keyboard. This type of
configuration is well-known in countless sound settings (patches).
It does not matter which of the ten Router Channels you use.
You can select the same target in several Router Channels, in parallel. You can freely use
the same Sources as often as you like, and the same via controllers can be set in one or
multiple Router channels.
Modulation Targets
The following targets are available for real-time modulation.
Note: These modulation targets are also available for the X and Y axes of the X/Y
modulator (the Square). See “The Square” section, on page 272.
Pitch 123
This target allows the parallel modulation of the frequencies (pitch) of all three
oscillators. If you select an LFO as the source, this target leads to siren or vibrato sounds.
Select one of the envelope generators with zero attack, short decay, zero sustain, and
short release as the source for tom tom and kick drum sounds.

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Pitch 1
This target allows modulation of the frequency (pitch) of Oscillator 1. Slight envelope
modulations can make the amount of detuning change over time, when Oscillator 1 is
sounding in unison with another (unmodulated) Oscillator. This is useful for synthesizer
brass sounds.
Pitch 2
This target allows modulation of the frequency (pitch) of Oscillator 2.
Pitch 3
This target allows modulation of the frequency (pitch) of Oscillator 3.
Detune
This target controls the amount of detuning between all three oscillators.
Note: The sensitivity of all the pitch modulation targets described above depends on
the modulation intensity. This sensitivity scaling allows for very delicate vibrati in the
cent range (one cent equals 1/100 semitone), as well as for huge pitch jumps in octave
ranges.
•
•
•
•
•
•

Modulation intensity from 0 to 8: steps are 1.25 cents.
Modulation intensity from 8 to 20: steps are 3.33 cents.
Modulation intensity from 20 to 28: steps are 6.25 cents.
Modulation intensity from 28 to 36: steps are 12.5 cents.
Modulation intensity from 36 to 76: steps are 25 cents.
Modulation intensity from 76 to 100: steps are 100 cents.

This leads to the following rules of thumb for modulation intensity settings.
• Modulation intensity of 8 equals a pitch shift of 10 cents.
• Modulation intensity of 20 equals a pitch shift of 50 cents, or one quarter tone.
• Modulation intensity of 28 equals a pitch shift of 100 cents, or one semitone.
• Modulation intensity of 36 equals a pitch shift of 200 cents, or two semitones.
• Modulation intensity of 76 equals a pitch shift of 1,200 cents, or one octave.
• Modulation intensity of 100 equals a pitch shift of 3,600 cents, or three octaves.
OscWaves
Dependent on the Waveforms set in the three Oscillators, this target can be used to
modulate:
• the pulse width of rectangular and pulse waves,
• the amount of frequency modulation (Oscillator 1 only),
• Noise color (Oscillator 3 only)
• and the position of the Digiwaves.
OscWaves affects all oscillators simultaneously. The Osc1Wave, Osc2Wave, and Osc3Wave
targets only affect the specified Oscillator. Check out the ensuing paragraphs to see
what wave modulation does in the three Oscillators.

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For further information on the effects of these modulations, please read the “Pulse
Width Modulation” section, on page 236. Also take a look at the “Linear Frequency
Modulation” section, on page 234, “White and Colored Noise (Oscillator 3 only)”, on
page 240, and the “Digiwaves” section, on page 233.
Osc1Wave
Dependent on the waveform selected, you can control the pulse width of rectangular
and pulse waves of Oscillator 1, the amount of frequency modulation (with Oscillator 1
being the carrier and Oscillator 2 being the modulator), or the position of the Digiwave.
The pulse width of the rectangular and pulse waves is not restricted to two fixed values
in Oscillator 1.

Note: In classic FM synthesizers, the amount of FM is controlled in real time by velocity
sensitive envelope generators. Select one of the ENVs as the source for such sounds.
Osc2Wave
As per Osc1Wave, except that Oscillator 2 does not feature FM. Please note that pulse
width modulation also works with both the synchronized rectangular and ring
modulated rectangular waves.

Osc3Wave
Oscillator 3 is as per Osc1Wave and Osc2Wave, but it does not feature FM or ring
modulation. Oscillator 3 features Noise, the color of which can be modulated with this
parameter.

OscWaveB
The transitions between Digiwaves during a wavetable modulation are always smooth.
Depending on the modulation intensity, an additional OscWaveB target can be used to
continuously modulate the shape of the transitions from smooth to hard. It applies to
all Oscillators.

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Osc1WaveB
If wavetable modulation is active for a DigiWave using Osc1Wav, you can use this target
to modulate the shape of the transition.
Osc1 FM mode: When compared with the hardwired Osc1-FM and the Osc1Wave
modulation target, the Osc1WaveB modulation target offers much higher FM
intensities.
Osc2WaveB
If wavetable modulation is active for a Digiwave using Osc2Wav, you can use this target
to modulate the shape of the transition.
Osc3WaveB
If wavetable modulation is active for a Digiwave using Osc3Wav, you can use this target
to modulate the shape of the transition.
SineLevl
SineLevl (Sine Level) allows modulation of the sine wave level of Oscillator 1, which can
be mixed directly into the input of the dynamic stage—without being affected by the
filters. The parameter defines the level of the first partial tone of Oscillator 1. See the
“Sine Level” section, on page 252.
OscLScle
OscLScle (Osc Level Scale) allows modulation of the levels of all three oscillators
simultaneously. A modulation value of 0 mutes all oscillators, while a value of 1 raises
the gain of the entire mix by 12 dB. The modulation is applied before the overdrive
stage, allowing for dynamic distortions.
Osc1Levl
Osc1Levl (Osc 1 Level) allows modulation of Oscillator 1’s level.
Osc2Levl
Osc2Levl (Osc 2Level) allows modulation of Oscillator 2’s level.
Osc3Levl
Osc3Levl (Osc 3Level) allows modulation of Oscillator 3’s level.
Cutoff 1
This target allows modulation of the Cutoff Frequency of Filter 1. See the “Cutoff and
Resonance” section, on page 246.

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Resonance 1 (Reso 1)
This target allows modulation of the Resonance of Filter 1. See the “Cutoff and
Resonance” section, on page 246.
Cutoff 2
This target allows modulation of the Cutoff Frequency of Filter 2.

Resonance 2 (Reso2)
This target allows modulation of the Resonance of Filter 2.
LPF FM
A sine signal, at the same frequency as Oscillator 1, can modulate the Cutoff frequency
of Filter 2 (which always works as a low pass filter). LPF FM (Lowpass Filter Frequency
Modulation) allows the modulation of Filter 2’s FM modulation intensity. This filter FM
parameter is described in the “Filter 2 FM” section, on page 250.
Cut 1+2
Cut 1+2 (Cutoff 1 and 2) modulates both filter’s Cutoff frequencies in parallel, much like
applying the same modulation to Cutoff 1 and Cutoff 2 in two Router channels.
Cut1inv2
Cut1inv2 (Cutoff 1 normal and Cutoff 2 inverse) simultaneously modulates the Cutoff
frequencies of the first and second filters inversely (in opposite directions). Put another
way, while the first filter’s Cutoff frequency is rising, the Cutoff of the second filter will
fall—and vice versa.
Note: In cases where you have combined Filter 1, defined as a highpass filter, and Filter
2 (which always works in lowpass mode) in Serial mode, both will serve as a bandpass
filter. Modulating the Cut1 inv 2 target will result in a modulation of the bandpass filter’s
bandwidth in this scenario.
FltBlend
FltBlend (Filter Blend) modulates the FilterBlend (the cross-fading of the two filters), as
described in the “Filter Blend and Signal Flow” section, on page 243.
Note: If FilterBlend is set as a target in one or several Router channels, the modulation
data for both filters will be calculated—even if the FilterBlend parameter is set to 1.0 or
+1.0. As such, we advise caution when choosing FilterBlend as a modulation target
because it may increase the need for processing power.

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Amp
This target modulates the dynamic stage, or level (the voice’s volume). If you select
Amp as the target and modulate it with an LFO as the Source, the level will change
periodically, and you will hear a tremolo.
Pan
This target modulates the panorama position of the sound in the stereo spectrum.
Modulating Pan with an LFO will result in a stereo tremolo (auto panning).
In Unison Mode, the panorama positions of all voices are spread across the entire
stereo spectrum. Nevertheless, pan can still be modulated, with positions being moved
in parallel.
Lfo1Asym
Lfo1Asym (Lfo1 Asymmetry) can modulate the selected waveform of LFO 1. In the case
of a square wave, it changes its pulse width. In the case of a triangle wave, it sweeps
between triangle and sawtooth. In the case of a sawtooth wave, it shifts its zero
crossing.
Lfo1Curve
This target modulates the waveform smoothing of the square and random wave. In the
case of a triangle or sawtooth wave, it changes their curves between convex, linear, and
concave.
Scaled Modulations
All following modulation targets result in a scaled modulation, which means that the
modulation value isn’t simply added to the target parameter’s value, but rather that the
target parameter value will be multiplied. This works as follows: a modulation value of
0.0 results in no effect, while a modulation value of +1.0 equals a multiplication by 10,
and a modulation value of −1.0 equals a multiplication by 0.04.
LFO1Rate
This target modulates the frequency (speed, rate) of LFO 1.
Note: Say you’ve created a vibrato with another Router channel by modulating the
Target Pitch 123 with LFO 1. If desired, you can have LFO 1’s speed (the vibrato speed)
automatically accelerated or slowed down. To do so, modulate the LFO1Rate target
with one of the envelope generators (ENV). Select LFO 2 as a source and reduce it’s Rate,
in order to periodically accelerate and slow down the vibrato.
Env2Atck
Env2Atck (Envelope 2 Attack) modulates the Attack time of the second envelope
generator.

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Env2Dec
Env2Dec (Envelope 2 Decay) modulates the Decay time of the second envelope
generator.
In cases where you’ve selected ENV2 Dec as the target and Velocity as the source, the
duration of the decaying note is dependent on how hard you strike the key. Selecting
Keyboard as the source will result in higher notes decaying more quickly (or slowly).
Env2Rel
Env2Rel (Envelope 2 Release) modulates the Release time of the second envelope
generator.
Env2Time
Env2Time (Envelope 2 All Times) modulates all of ENV2’s time parameters: Attack time,
Decay time, Sustain time, and Release time.
Env3Atck
Env3Atck (Envelope 3Atck) modulates the Attack time of the third envelope generator.
Env3Dec
Env3Dec (Envelope 3 Decay) modulates the Decay time of the third envelope generator.
Env3Rel
Env3Rel (Envelope 3 Release) modulates the Release time of the third envelope
generator.
Env3Time
Env3Time (Envelope 3 All Times) modulates all of Env3’s time parameters: Attack time,
Decay time, Sustain time, and Release time.
Glide
This target modulates the duration of the Glide (portamento) effect.
Note: If you modulate Glide with Velocity selected as the source, the velocity used to
(how hard) strike the key will define how long it takes for the played notes to “find their
way” to the target pitch. See the “Glide” section, on page 227.
Modulation Sources
Some modulation sources are unipolar, delivering values between 0 and 1. Others are
bipolar, and output values between −−1 and +1. The following modulation sources are
available:
LFO1
… LFO 1 is described in “The LFO’s” section, on page 265.
LFO2
… LFO 2 is described in “The LFO’s” section, on page 265.

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ENV1
… Envelope Generator 1 is described in “The Envelopes (ENV 1—ENV 3)” section, on
page 268.
ENV2
… Envelope Generator 2 is described in “The Envelopes (ENV 1—ENV 3)” section, on
page 268.
ENV3
… Envelope Generator 3 is described in “The Envelopes (ENV 1—ENV 3)” section, on
page 268.
Note: Envelope Generator 3 always controls the level of the overall sound.
Pad-X, Pad-Y
These modulation sources allow you to define the axes of the Square, for use with the
selected modulation target. The cursor can be moved to any position in the Square,
either manually or controlled by the vector envelope. See “The Square” on page 272
and “The Vector Envelope” on page 273.
Max
If you select Max as a source, the value of this source will permanently be set to +1. This
offers interesting options in conjunction with via, as the possible values available for via
control the modulation intensity.
Kybd
Kybd (Keyboard) outputs the keyboard position (the MIDI note number). The center
point is C3 (an output value of 0). Five octaves below and above, an output value of −−
1 or +1, respectively is sent.
Note: This could be used to control the Cutoff frequencies of the filters in parallel with
the keyboard position—as you played up and down the keyboard, the Cutoff
frequencies would change. Modulate the Cut 1+2 target with the Keyboard source to do
so. At a modulation intensity of 0.5, the Cutoff frequencies scale proportionally with the
pitches played on the keyboard.
Velo
If you select Velo (Velocity), velocity sensitivity serves as modulation source.
Bender
The pitch bender serves as a bipolar modulation source, if Bender is selected. This is also
true when the Oscillators’ Bend Range parameter is set to 0.
ModWhl
The modulation wheel serves as an unipolar modulation source, if ModWhl is selected.

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Note: For most standard applications, you’ll probably use the wheel as the via
controller. Traditionally, it can be (and is) used for control over the intensity of periodic
LFO modulations. Used here, it can be employed for direct, static modulations, such as
controlling the Cutoff frequencies of the filters (Target = Cut 1+2).
Note: The Least Significant Bit (LSB) controller for the modulation wheel is recognized
correctly, as well.
Touch
Aftertouch serves as modulation source. The ES2 reacts to poly pressure (polyphonic
aftertouch). It uses the sum of channel pressure and the note-specific poly pressure
value.
Note: If you set the Target to Cut 1+2, the Cutoff frequencies will rise and fall, dependent
on how firmly you press a key on your touch-sensitive MIDI keyboard after the initial
keystrike.
Whl+To
The modulation wheel and aftertouch serve as modulation sources.
MIDI Controllers A–F
MIDI controllers available in the mod matrix are named Ctrl A–F and can be assigned to
arbitrary controller numbers (via the MIDI Controllers Assignment menus at the bottom
of the GUI).
Note: Earlier ES2 versions offered the Expression, Breath, and MIDI Control Change
Messages 16–19 as modulation sources. These MIDI controllers are the default values for
the assignment and guarantee backwards compatibility.
The values of the MIDI Controllers Assignment menus are only updated if the default
setting is loaded, or if a setting that was saved with a song is loaded. If you simply step
through settings, the assignment will remain unchanged.
The MIDI Controllers Assignment menus allow you to assign your favorite MIDI
controllers as Ctrl A, Ctrl B, and so on.
All MIDI Controller Assignment menus feature a Learn option. If this is selected, the
parameter will automatically be assigned by the first appropriate incoming MIDI data
message.
Note: If none of the controller assignments (Ctrl A–F) is assigned to Expression, the
Expression CC message (Ctrl #11) controls the output volume.
Note: The “Vector Stick” (Joystick) of the Korg Wavestation synthesizer generates
Controllers 16 and 17, for example. If you use this instrument as your master keyboard,
you can control any two ES2 parameters directly with its Joystick.

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Note: In the MIDI specification, for all controllers from 0 to 31, there also is a LSBController defined (32 to 63). This “Least Significant Bit”-controller allows for a
resolution of 14 bit instead of 7 bit. The ES2 recognizes these control change messages
correctly; the breath or expression controllers, for example.
RndN01
RndNO1 (Note On Random1) outputs a random modulation value between −1.0 and 1.0
(same range as an LFO), that changes when a note is triggered or re-triggered. The
(random) note-on modulation remains constant throughout the duration of the note
until the next note-on trigger.
Note: There is no value change when playing legato, while in legato mode.
RndN02
RndNO2 (Note On Random2) behaves like Note On Random1 with the exception that it
glides to the new random value using the Glide time (inclusive of modulation). It also
differs from NoteOnRandom1 in that the (random modulation) value changes when
playing legato, while in legato mode.
SideCh
SideCh (Side Chain modulation) uses a Side Chain (tracks, inputs, busses) to create a
modulation signal. The Side Chain source can be selected in the upper gray area of the
window. It is fed to the internal envelope follower, which creates a modulation value
based on the current Side Chain input signal level.
Via—Controlling the Modulation Intensity
Some modulation sources are unipolar, delivering values between 0 and 1. Others are
bipolar, and output values between −−1 and +1. The following sources may be used to
modulate the modulation intensity.
LFO1
The modulation undulates at the speed and waveform of LFO1, which controls the
modulation intensity.
LFO2
The modulation undulates at the speed and waveform of LFO2, which controls the
modulation intensity.
ENV1
ENV1 controls the modulation intensity.
ENV2
ENV2 controls the modulation intensity.
ENV3
ENV3 (the level envelope) controls the modulation intensity.

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Pad-X, Pad-Y
Both axes of the Square (the Vector Envelope) are available as via sources as well,
enabling you to control modulation intensities with them.
Kybd
Kybd (Keyboard) outputs the keyboard position (the MIDI note number). The center
point is C3 (an output value of 0). Five octaves below and above, an output value of −1
or +1, respectively is sent.
If you select Pitch 123 as the target, modulate it with the LFO1 source, and select
Keyboard as the via value, the vibrato depth will change, dependent on key position.
Put another way, the vibrato depth will be different for notes higher or lower than the
defined Keyboard position.
Velo
If you select Velo (Velocity) as the via value, the modulation intensity will be velocity
sensitive—modulation will be more or less intense dependent on how quickly (hard)
you strike the key.
Bender
The pitch bender controls the modulation intensity.
ModWhl
If you select ModWhl (Modulation Wheel) as the via value, the modulation intensity will
be controlled by your MIDI keyboard’s modulation wheel.
The Least Significant Bit (LSB) Controller for the modulation wheel is recognized
correctly, as well.
Touch
If you select Touch (Aftertouch) as the via value, the modulation intensity will be touch
sensitive—modulation will be more or less intense dependent on how firmly you press
the key of your touch-sensitive MIDI keyboard after the initial keystrike (aftertouch is
also known as pressure sensitivity).
Whl+To
Both modulation wheel and aftertouch control the modulation.
MIDI Controllers A–F
MIDI controllers available in the mod matrix are named Ctrl A–F (rather than
Expression, Breath, General Purpose 1–4. MIDI Control Change Messages 16–19 are also
known as “General Purpose Slider 1/2/3/4”.) and can be assigned to arbitrary controller
numbers (via menus at the bottom of the GUI). The default values for the assignment
guarantee backwards compatibility. The values of these assignment menus are only
updated if the default setting is loaded, or if a setting that was saved with a song is
loaded. If you simply step through settings, the assignment will remain unchanged.

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This facility is especially helpful if you’ve always wanted to use Controller #4 (foot
pedal), for example, as a modulation source. This feature allows you to assign your
favorite MIDI real-time controllers as Ctrl A, Ctrl B, and so on.
All parameters that allow you to select a MIDI controller feature a Learn option. If this is
selected, the parameter will automatically be assigned by the first appropriate
incoming MIDI data message.
Note: As the new entry is added to the top of the list, existing automation data is
increased by one. Should further assignments be made, this will increment by one for
each entry.
Note: If none of the controller assignments (Ctrl A–F) is assigned to “Expression”, the
Expression CC message (Ctrl #11) controls the output volume.
Note: The “Vector Stick” (Joystick) of the Korg Wavestation synthesizer generates
Controllers 16 and 17, for example. If you use this instrument as your master keyboard,
you can control modulation intensities directly with its Joystick.
Note: In the MIDI specification, for all controllers from 0 to 31, there also is a LSBController defined (32 to 63). This “Least Significant Bit”-controller allows for a
resolution of 14 bit instead of 7 bit. The ES2 recognizes these control change messages
correctly, for instance the controllers for breath or expression.
RndN01
RndN01 (NoteOnRandom1) controls the modulation intensity (See RndNO1 on page
263).
RndN02
RndNO2 (NoteOnRandom2) controls the modulation intensity (See RndNO2 on page
263).
SideCh
A Side Chain source (tracks, inputs, busses) is used to create the modulation signal.

The LFO’s
LFO is the abbreviated form of Low Frequency Oscillator. In an analog synthesizer, LFO’s
deliver modulation signals below the audio frequency range—in the bandwidth that
falls between 0.1 and 20 Hz, and sometimes as high as 50 Hz. LFO’s serve as modulation
sources for periodic, cyclic modulation effects. If you slightly modulate the pitch of an
audio oscillator at a rate (speed, LFO frequency) of, say, 3–8 Hz, you’ll hear a vibrato. If
you modulate the cutoff frequency of a lowpass filter, you’ll hear a wah wah effect, and
modulating the dynamic stage results in a tremolo.
The ES2 features two LFO’s, the outputs of which are available as sources in the Router.

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• LFO 1 is polyphonic, which means that if used for any modulation of multiple voices,

they will not be phase-locked. Furthermore, LFO 1 is key-synced: Each time you hit a
key, the LFO 1 of this voice is started from zero. To explain, when used on polyphonic
input (a chord played on the keyboard) the modulation is independent for each
voice (note). Where the pitch of one voice may rise, the pitch of another voice might
fall and the pitch of a third voice may reach its minimum value.

• LFO 2 is monophonic, which means, that the pitch of all voices will rise and fall

synchronously, if you modulated the Pitch123 target with the LFO2 source, for
example.

Both LFO’s feature a number of waveforms. LFO 1 can fade in or out automatically,
without the need to engage a separate envelope generator. The LFO parameters are
detailed below:
EG (LFO1)
At its center position—which can be accessed by clicking the middle mark—the
modulation intensity is static: it won’t be faded in or out at all. At positive values, it is
faded in. The higher the value, the longer the delay time is. At negative values, it is
faded out. The lower (onscreen) the slider is positioned, the shorter the fade out time is.
The function is abbreviated as EG because the fading in or out is internally performed
by an ultra-simple Envelope Generator.
Note: Chaotic and fast modulations of the oscillator(s) frequencies (Pitch123) by LFO 1
with a delayed Sample&Hold selected as the waveform, a high Rate, and short fade out,
make the attack phase of the note sound Moog “Rogue-ish”—and quite similar to the
attack phase of brass instruments.
Note: Most commonly, this is used for delayed vibrato—many instrumentalists and
singers intonate longer notes this way. To set up a delayed vibrato: Place the slider at a
position in the upper half (Delay) and modulate the Pitch123 target with the LFO1
source. Set a slight modulation intensity. Select a Rate of about 5 Hz and the triangular
wave as the LFO waveform.

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Rate
This parameter defines the frequency or speed of the modulation. The value is
displayed in Hertz (Hz) beneath the slider.
Wave
This is where you select the desired LFO waveform. Check out the waveforms while a
modulation of Pitch123 is engaged and running. You should find the symbols quite selfevident.
Triangular Wave
The triangular wave is well suited for vibrato effects.
Sawtooth Wave and Inverted Sawtooth
The sawtooth is well suited for helicopter and space gun sounds. Intense modulations
of the oscillator(s) frequencies with a negative (inverse) sawtooth wave leads to
bubbling and boiling, underwater sounds. Intense sawtooth modulations of lowpass
filters (such as Filter 2) create rhythmic effects.
Rectangular Waves
The rectangular waves make the LFO periodically switch between two values. The
upper rectangular wave switches between a positive value and zero. The lower wave
switches between a positive and a negative value set to the same amount above/
below zero.
Note: An interesting effect you may wish to try out is achieved by modulating Pitch123
with a suitable modulation intensity that leads to an interval of a fifth. Choose the
upper rectangular wave to do so.
Sample & Hold
The two lower waveform settings of the LFO’s output random values. A random value is
selected at regular intervals, defined by the LFO rate. The upper waveform delivers
exact steps of randomization. At its lower setting, the random wave is smoothed out,
resulting in fluid changes to values.
Note: A random modulation of Pitch123 leads to the effect commonly referred to as a
random pitch pattern generator or sample and hold. Check out very high notes, at very
high rates and high intensities—you’ll recognize this well-known effect from hundreds
of science fiction movies!
Note: The term Sample & Hold (abbreviation—S & H) refers to the procedure of taking
samples from a noise signal at regular intervals. The voltage values of these samples are
then held until the next sample is taken. When converting analog audio signals into
digital signals, a similar procedure takes place: Samples of the voltage of the analog
audio signal are taken at the rate of the sampling frequency.

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Rate (LFO 2)
The LFO2 Rate (frequency) control allows for the free (in the upper half of the slider
range) or song-tempo synchronized (in the lower half of the slider range) running of
LFO 2. The rate is displayed in Hertz or rhythmic values, dependent on whether the
song tempo synchronization is engaged or not. Rates range from speeds of 1/64-notes
through to a periodic duration of 32 bars. Triolic and punctuated values are also
possible. LFO 2 is ideally suited for rhythmic effects which retain perfect synchronicity,
even during tempo changes to the song.

The Envelopes (ENV 1—ENV 3)
In addition to the complex Vector Envelope, described in “The Vector Envelope” section,
on page 273, the ES2 also features three envelope generators per voice. On both the
front panel and as a source in the Router, they are abbreviated as ENV 1, ENV 2 and ENV
3, respectively.
Note: The roots of the term envelope generator and its basic functionality are described
in the “Envelopes” section, on page 199.
The feature sets of ENV 2 and ENV3 are identical. ENV 3 defines the changes in level
over time for each note played. You can regard ENV 3 as being “hard-wired” to the
Router’s AMP modulation target.

The parameters of ENV 2 and ENV 3 are identical—but ENV 3 is always used for control
over level.
Unlike many other synthesizers, there is no hard-wired connection between any of the
envelope generators and the cutoff frequencies of the ES2 filters. Modulation of the
cutoff frequencies must be set separately in the Router. In the default setting, this is
already the case—in the Router channel just below the Filter (see graphic).

Note: Set up a Router channel as follows, in order to establish this type of
modulation: Set target to Cutoff 1, Cutoff 2 or Cut 1+2, set source to, say, ENV 2. Once set
as described, the slider of the Router channel will function as the filter’s EG Depth
parameter.

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Note: Both ENV 2 and ENV 3 are velocity sensitive, making it unnecessary to set via to
Velo in the Router channel: You can leave via switched off.
The Parameters of ENV 1
At first glance, ENV 1 appears to be rather poorly equipped. Its few parameters,
however, are useful for a vast range of synthesizer functions.
Trigger Modes: Poly, Mono, Retrig
In Poly mode, the envelope generator behaves as you would expect on any polyphonic
synthesizer: Every voice has its own envelope.

In Mono and Retrig modes, a single envelope generator modulates all voices in
parallel—identically, in other words.
• If ENV 1 is set to Mono, all notes must be released before the envelope can be

triggered again. If you play legato, or any key remains depressed, the envelope won’t
start its attack phase again.
• In Retrig mode, the envelope will be triggered by any key you strike, no matter
whether other notes are sustained or not. Every sustained note is affected by the
retriggered envelope.
Note: The design of early analog polysynths led to polyphonic instruments where all
voices passed through a single lowpass filter. This design was primarily due to cost
factors. The best known example of these polyphonic instruments were the Moog
Polymoog, the Yamaha SK20 and Korg Poly 800. The sole lowpass filter of such
instruments is controlled by a single envelope generator. To simulate this behavior, use
the Mono or Retrigger modes.
Note: Say you’ve modulated the Cutoff 2 target with a percussive source, such as ENV1,
which is set to Retrig. If you play and sustain a bass note, this note will receive a
percussive filter effect every time you hit another key. The newly struck key is also
shaped by the same filter. Playing a sound set up in this way feels like you’re playing a
polyphonic synthesizer with one filter. This is despite the fact that the ES2’s filters
remain polyphonic and can be simultaneously modulated by different polyphonic
sources.
Note: If you want to simulate the percussion of a Hammond Organ, you will also need
to use the Mono or Retrigger modes.

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Decay/Release
ENV 1 can be set to act as an envelope generator with an Attack time and Decay time
parameter or with an Attack time and Release time parameter.
Switching between both modes is achieved by clicking on the D or the R above the
right ENV 1 slider.
• In its Attack/Decay mode, the level will fall to zero after the attack phase has
completed, no matter whether you sustain the note or not. It will decay at the same
speed even if you release the key. The decay time is set with the Decay time slider,
abbreviated as D.
• In its Attack/Release mode, the envelope level remains at its maximum after the
attack phase is over, as long as the key remains depressed. Following the release of
the key, its level decreases over the time set with the R slider—the abbreviation for
Release time.
Attack Time and Attack via Vel
The Attack time slider is divided into two halves. The lower half defines the attack time
when the keys are struck hard (at maximum velocity). The upper half defines the attack
time at minimum velocity.
You can grab the area between the two slider halves with the mouse and drag both
halves simultaneously. If this area is to small to be grabbed with the mouse, just click in
a free part of the slider track, and move the mouse up or down to move the area.
The Parameters of ENV 2 and ENV 3
The feature sets of ENV 2 and ENV 3 are identical, but it is always the task of ENV 3 to
define the level of each note—to modulate the dynamic stage. ENV 3 is available for
simultaneous use as a source in the Router as well. The envelope’s time parameters can
also be used as modulation targets in the Router.
Note: See the “Envelopes” section, on page 199 for information on the basic
functionality and meaning of envelope generators.
Attack Time
As per the Attack slider of ENV 1, the Attack time sliders of ENV 2 and ENV 3 are divided
into two halves. The lower half defines the attack time when the keys are struck at
maximum velocity. The upper half defines the attack time at minimum velocity.
You can grab the area between the two slider halves with the mouse and drag both
halves simultaneously. If this area is to small to be grabbed with the mouse, just click in
a free part of the slider track, and move the mouse up or down to move the area.

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Decay Time
The Decay time parameter defines the length of time it takes for the level of a sustained
note to fall to the Sustain level, after the attack phase is over. If Sustain level is set to its
maximum, the Decay parameter has no effect. When the Sustain level is set to its
minimum value, Decay defines the duration or fade-out time of the note.
The Decay parameter appears as a modulation target in the Router for ENV 2 and ENV 3
individually (ENV2Dec, ENV3Dec).
Note: On pianos and plucked string instruments, high notes decay faster than low
notes. In order to simulate this effect, modulate the Decay Time target with the Kybd
source in the Router. The Router channel slider should be set to a negative value.
Sustain and Sustain Time
When the Sustain Time parameter is set to its center value—which can be achieved by
clicking the ∞ symbol—the Sustain level behaves like the Sustain parameter of any
synthesizer ADSR envelope. In this position, the Sustain level (abbreviated as S) defines
the level that is sustained for as long as the key remains depressed, following the
completion of the Attack time and Decay time phases.
The Sustain Time slider defines the time it takes for the level to rise to its maximum—or
to fall to zero—after the decay phase is over. Settings in the lower half of its range (Fall)
determine the speed at which the level decays from the Sustain level to zero. The lower
the slider position, the faster the decay speed. Settings in the upper half of its range
(Rise) determine the speed at which the level rises from the Sustain level to its
maximum value. The higher the slider position, the faster the speed of the rise.
Release Time
As with any synthesizer ADSR envelope, the Release time parameter (R) defines the time
the level takes to decay to zero after the key is released.
Vel
The Vel (Velocity Sensitivity) parameter defines the velocity sensitivity of the entire
envelope. If it is set to maximum, the envelope will only output its maximum level
when the keys are struck at maximum velocity.

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The Square
The Square has two axes: The X and Y axes have positive and negative value ranges.
They are bipolar, in other words. By touching and moving the cursor with the mouse,
the values of both axes are continuously transmitted. As you can modulate one freely
selectable parameter with the X value, and another freely selectable parameter with
the Y value, you can use the mouse like a Joystick.

As an alternative to this real-time control, the position of the cursor can be modulated
by the Vector Envelope—just like the mix between the three Oscillators in the Triangle.
The Loop function of the Vector Envelope generator allows for cyclic movements. This
opens a number of doors, allowing it to operate as a two-dimensional, luxurious
pseudo-LFO with a programmable waveform. More on this is found in “The Vector
Envelope” section, on page 273.
Vector Mode
Vector Mode allows you to disable the control of the Square cursor by the vector
envelope. The same parameter also defines whether or not the Triangle (the oscillator
mixer) shall be controlled by the vector envelope.
• Vector Mode Off—The vector envelope does not influence the Triangle or the Square.
It’s simply switched off. This allows you to set and control the cursors of the Triangle
and the Square in real-time.
• Vector Mode Mix—The vector envelope controls the Triangle (the oscillator mix), but
not the Square.
• Vector Mode XY—The vector envelope controls the Square, but not the Triangle.
• Vector Mode Mix+XY—The vector envelope controls both the Square and Triangle.
Note: Like all of the ES2’s parameters, the movements of the cursors in the Triangle and
Square can be recorded and automated by Logic. This automation data can be edited
and looped in Logic. This is completely independent of the cyclic modulations of the
Vector Envelope. Vector modulation of the Square and Triangle should be disabled for
this type of use (Vector Mode = off ).
Vector Target—Modulation Destinations
The Vector X and Vector Y target parameters determine the effect of cursor movements
in the Square. The modulation targets are identical to those available in the Router, so
we won’t repeat ourselves here. Please see the “Modulation Targets” section, on page
254 for descriptions. The position of the cursor in the Square is also available in the
Router, as the Pad-X and Pad-Y Source and Via options.

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Vector Int—The Modulation Intensity
The maximum intensity, sensitivity, and polarity of the modulation is set with the Vector
X Int and Vector Y Int parameters.

The Vector Envelope
The Triangle and Square are the most special and unusual elements of the ES2’s
graphical user interface. Whilst the Triangle controls the mix of the three oscillators, the
X and Y axes of the Square can modulate any (modulation) target.

The Vector Envelope can control the movement of the cursors in the Triangle and the
Square in real-time. Each voice is equipped with its own Vector Envelope, which is
triggered from its startpoint with every new keystrike (or with every MIDI note-on
message, to be more precise).

The concept of the Vector Envelope, the Square and the Triangle may seem strange at
first glance, but believe us: Combined with the other synthesis possibilities of the ES2,
and you will end up with sounds that are truly unique and literally, moving.
Envelope Points, Times, and Loops
The Vector Envelope consists of up to 15 points on the time axis. Each point can control
the position of both the Triangle and Square’s cursors.
The points are numbered sequentially. Point 1 is the starting point. In order to edit a
point, simply select it—by clicking on it.
Sustain Point
Any point can be declared the Sustain Point. Given that the note played is sustained
long enough and there’s no loop engaged, any envelope movement will stop when
this Sustain Point is reached. It will be sustained until the key is released (until the MIDI
note-off command).

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To define a point as the Sustain Point, click on the turquoise strip above the desired
point. The selected point will be indicated by an S between the point and its number,
on the turquoise strip.

Loop Point
Any point can be declared Loop Point. Given that the note is sustained long enough,
the envelope can be repeated in a loop.
The looped area is the time span between Sustain Point and Loop Point. In between,
you can define several points which describe the movements of the Square and
Triangle cursors.
In order to define a point as the Loop Point, click on the turquoise strip below the
desired point. A Loop Point is indicated by an L in the strip below.

Note: In order to see and define the Loop Point, the loop must be activated. See the
“Loop”, on page 277.
Note: With loop activated, the Vector Envelope works like a multi-dimensional,
polyphonic LFO with a programmable waveform.
Vector Envelope Times
With the exception of the first point, which is tied to the beginning of each played
note, every point has a Time parameter. This parameter defines the period of time
required for the cursor to travel from the point which preceded it. The times are
normally displayed in milliseconds (ms).
To adjust a time value, you can click directly on the numerical value and use your
mouse as a slider.

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Default Setting of the Vector Envelope
The default setting of the Vector Envelope consists of three points. Point 1 is the
startpoint, point 2 is defined as the Sustain Point, and point three is the end point, by
default.
The impact of the Vector Envelope on the Oscillator Mix or on the Square is switched off
by default. This allows the ES2 to behave as a synthesizer without a Vector Envelope
generator. This traditional starting point is more convenient when creating patches
from scratch.
There are two ways to switch off the Vector Envelope:
• You can switch on the Solo Point parameter (described on page 276). If it is on, only
the Triangle and Square cursor positions of the currently selected point are active.
• You can disable the Vector Envelope altogether (or only the Triangle or Square) as
described in the “Vector Mode” on page 272.
Setting and Deleting Points
The more points you set, the more complex the Vector Envelope movements that can
be designed. You can:
• Create a new point by Shift-clicking between two existing points. The segment that
previously existed between the two old points is divided at the mouse position. The
sum of the two new segment times is equal to the time used by the old undivided
segment. As such, the ensuing points retain their absolute time positions. In addition,
the existing cursor positions in the Triangle and Square are fixed, ensuring that the
creation of new points does not affect any previously defined movements.
• Delete points by clicking on them while holding Control.
Setting Vector Envelope Times
By clicking a time value and moving the mouse, you will alter the envelope time—the
time it takes for the Vector Envelope to travel from the point before this time value to
the point after this time value. You have two ways of doing this.
• Simple vertical dragging of the time parameter results in reaching all following
points later (or sooner, respectively) in time.
• Dragging with Control held, you will shorten or lengthen the time of the following
point by the same amount. This ensures that the adjacent, and all following, points
retain their absolute time positions.
Resetting the Values of a Point
Reverting to default cursor positions in the Triangle and the Square is done by:
• Clicking in the Triangle while holding down the Option key. This sets all Oscillators to
output the same level. The cursor is set to the middle of the Triangle.
• Clicking onto the Square while holding down the Option key sets the cursor to the
center of the Square. Both axis values are set to zero.

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Solo Point
This function basically switches off the entire Vector Envelope generator. If Solo Point is
set to on, no dynamic modulations are applied by the Vector Envelope. In this scenario,
the currently visible cursor positions of the Triangle and Square are permanently in
effect. These cursor positions match the currently selected Vector Envelope point.
If you select another Vector Envelope point (by clicking on it), you will engage its
Triangle and Square cursor positions immediately. If Solo Point is set to on, the newly
selected point will become the solo point.
Note: You can independently switch off the vector modulation of the Square by setting
Vector Mode off, as outlined on page 272.
Env Modes Normal and Finish
If Env Mode is set to normal, the release phase (the phase after the sustain point) will
begin as soon as you release the key (note off ). The release phase will start from the
Vector Envelope point where you released the key.

• If the loop is switched off, and the Vector Envelope reaches the Sustain Point S, the

Sustain Point S will be played for as long as you hold the key.
• If the loop is switched on (see “Loop”, on page 277), and the Loop Point L is

positioned before the Sustain Point S, the loop will be played for as long as you hold
a key.
• If the loop is switched on, and the Loop Point L is positioned after the Sustain Point S,
the loop will be played after the release of a key.
In Env Mode finish, the Vector Envelope will not immediately commence the release
phase when you release the key. Rather, it will play all points (for their full duration)
until the last point is reached, regardless of whether you hold the key or release it.

• If the loop is switched off, the Sustain Point S will be ignored. The Vector Envelope

will end on its last point, regardless of whether you hold the key or release it.
• If the loop is switched on, the Vector Envelope will play all points until it reaches the

Loop Point, and then play the loop for as long as the sound ends. It does not matter
if the Loop Point L is before or after the Sustain Point S.
• If the loop is switched on, and Loop Count is set to a value other than infinite, the
Vector Envelope will continue on to following points after the selected number of
loop repeats. If Loop Count is set to infinite, the number of segments after the loop is
irrelevant. See “Loop Count”, on page 279.

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Curve
The Curve parameter sets the shape of the transition from point to point. You can
choose between nine convex and nine concave shapes. There are also the two extreme
forms; hold+step and step+hold, which allow stepped modulation. Where step+hold
jumps at the beginning of the transition time, hold+step jumps at the end.

Note: You can use hold+step to create stepped vector grooves with up to 15 steps.
Vector Envelope Loops
The Vector Envelope can—like any envelope—run in one shot (as long as the note is
sustained) mode. It can also run several times or in an infinite cycle, much like an LFO.
You can achieve this through the use of loops.
Note: The loop parameters might remind you of the loop parameters available for
samples. Just to avoid any misunderstandings: The Vector Envelope only supplies
control signals used for moving the cursor positions of the Triangle and Square. The
audio of the ES2 is not looped at all.
Loop
The ES2 features these loop options:
• Off
If Loop Mode is switched Off, the Vector Envelope runs in one shot mode from its
beginning to its end—given that the note is held long enough. The other loop
parameters are disabled.

• Forward

When Loop is set to Forward, the Vector Envelope runs to the Sustain Point and
begins to repeat the section between the Loop Point and Sustain Point periodically,
always in a forward direction.

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• Backward

When Loop is set to Backward, the Vector Envelope runs to the Sustain Point and
begins to repeat the section between the Sustain Point and Loop Point periodically,
always in a backward direction.

• Alternate

When Loop is set to Alternate, the Vector Envelope runs to the Sustain Point and
returns to the Loop Point and back to the Sustain Point periodically, alternating in
both a backward and forward direction.

Loop Rate
Just as every LFO has a speed (or Rate) parameter, the loop can be set to cycle at a
defined Loop Rate. And just like an LFO, the Vector Envelope Loop Rate can be
synchronized to the song tempo automatically.

• If you switch the Loop Rate to as set, the duration of the loop cycle is equal to the

sum of the times between the Sustain and Loop Points. Click on the field labeled as
set (below the Rate slider) to select.
• If you set the Loop Rate to one of the rhythmic values (sync, left half of the slider, 32
bars up to 64th Triplet Note), the Loop Rate fits to the song tempo.
• You also can set the Loop Rate in the small panel to the right half of the slider (free).
The value indicates the number of cycles per second. Use the mouse as a slider to
adjust.
Note: If Loop Rate is not switched to as set, and the loop is activated (Loop Mode
Forward, Backward or Alternate), the times of the points between the Loop and Sustain
Points as well as the value for Loop Smooth are indicated as a percentage of the loop
duration, rather than in milliseconds.
Loop Smooth
When Loop Mode is set to Forward or Backward, there will inevitably be a moment
when a transition from the Sustain Point to the Loop Point occurs. In order to avoid
abrupt cursor position changes, this transition can be smoothed through use of the
Loop Smooth parameter.
• Set the value by touching and dragging it with the mouse.

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• If Loop Rate is set to Sync or Free, the Loop Smoothing Time will be displayed as a

percentage of the loop cycle duration.

• If Loop Rate is set to as set, the Loop Smoothing Time will be displayed in

milliseconds (ms).

Loop Count
The loop cycle of the Vector Envelope doesn’t need to be performed infinitely—you
can have it cycle just a few times. Following the defined number of repetitions, the
Vector Envelope will run from the sustain point onwards, just as with Loop Mode off.
• Use the mouse as a slider to set the Loop Count value. Possible values are 1 to 10 and
infinite.

Time Scaling
You can stretch and compress the entire Vector Envelope. As an example, to double the
Vector Envelope’s speed, it’s not necessary to halve the time values of every point. All
you need to do is set Time Scaling to 50%.
• Adjust Time Scaling by using the mouse as a slider.

• The range for the Time Scaling parameter is from 10% to 1000%. It is scaled

logarithmically.
• If the Loop Rate is as set, the scaling also affects the loop. If not (Loop Rate = free or
sync), the setting will not be affected by Time Scaling.
Fix Timing—Normalizing Time Scaling and Loop Rate
By clicking Fix Timing, the Time Scaling value will be multiplied by all time parameters,
and Time Scaling will be reset to 100%. There will be no audible difference. This is simply
a normalizing procedure, most similar to the normalizing function of the Region
playback parameters in Logic.

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In cases where a loop which was synchronized to the song tempo had been engaged
(Loop Rate = sync), pressing Fix Timing will also switch the Loop Rate to as set, thus
preserving the absolute rate.

Effect Processor
The ES2 is equipped with an integrated effect processor. Any changes to this
processor’s effects settings are saved as an integral part of each sound program. The
entire output of the dynamic stage is processed in true stereo.
Despite the inclusion of this integrated effects processor, please feel free to process the
ES2 with the other effect plug-ins included in Logic. The sound and parameter set of
the integrated effects unit is reminiscent of classic pedal effects, designed for the
electric guitar. The use of guitar pedal effects on classic analog synthesizers was a
standard practice amongst gigging musicians.
Distortion
At its Soft setting, the distortion circuit is somewhat like a tube overdrive, whilst Hard
sounds like a fully-transistorized fuzz box. The Distortion control defines the amount of
distortion, and Tone controls the treble portion of the output of the distortion process.

Chorus, Phaser, Flanger
These classic modulation effects and their parameters (Intensity and Speed) won’t need
any explanation. These sophisticated algorithms simulate the sound of analog effects
of this kind, with one exception: They don’t produce as much noise.

Note: A chorus effect is based on a delay line, the output of which is mixed with the
original, dry signal. The short delay time is modulated periodically, resulting in pitch
deviations. The modulated deviations, in conjunction with the original signal’s pitch,
produce the chorus effect.
Note: The flanger works in a fashion similar to that of a chorus, but with even shorter
delay times, and the output signal being fed back into the input of the delay line. This
feedback results in the creation of harmonic resonances which wander cyclically
through the spectrum, giving the signal a metallic sound.

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Note: The phaser is based on a mix of a delayed and an original signal. The delayed
element is derived from an all-pass filter, which applies a frequency-dependent delay to
the signal. This is expressed as a phase angle. The effect is based on a comb filter, which
is basically an array of inharmonic notches (rather than resonances, as with the flanger),
which also wander through the frequency spectrum.

Random Sound Variations
The ES2 offers a unique feature that allows you to vary the sound parameters randomly.
You can define the amount of random variation, and can restrict the variations to
specific sonic elements. The random sound variation feature will inspire and aid (or
occasionally amuse) you when creating new sounds.

Pressing the RND button randomly alters the sound. The process is triggered by a single
click and can be repeated as often as you like.
Note: To avoid possible misunderstandings: This feature has nothing to do with
random real-time modulations. The random feature changes the parameters randomly
with each mouse click. Real-time random modulations are performed with the random
waveforms of the LFO’s and by the Analog parameter, for random pitch settings.
RND Int
RND Int (Random Intensity) defines the amount of random parameter alteration. As you
move the slider to the right, you will increase the amount of random variation.
The random sound variation feature always alters the parameters as they are currently
set, not based on the memorized Setting file. As such, clicking RND repeatedly will
result in a sound which increasingly differs from the original setting. If want to check
out several slight alterations of the current setting, you can reload the original Setting,
after each random alteration.
RND Destination
Some aspects of your sound may already be ideal for the sound you had in mind. As
such, it may not be desirable to alter them. Say your sound setting has a nice
percussiveness, and you’d like to try a few sonic color variations while retaining this
percussive feel. To avoid the random variation of any attack times, you can restrict the
variation to the oscillator or filter parameters, with the envelope parameters excluded
from the variation process. To do so, set the RND Destination to Waves or Filters.
Please note:
• The Master Level, Filter Bypass as well as the three Oscillator On/Off parameters and
the Vector/Router display options are never randomized.
• During randomizations of the Vector Envelope, the Solo Point will always be set to off.

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You can restrict the random sound variation to the parameter groups listed below:
All
All ES2 parameters, with the exception of the parameters mentioned above, will be
altered.
All except Router and Pitch
All ES2 parameters, with the exception of all Router parameters and the basic pitch
(semitone settings of the oscillators), will be altered. The oscillator fine tuning will be
varied. This will result in more musically useful sounds.
All except Vector Env
All ES2 parameters, with the exception of all Vector Envelope parameters, will be
altered. This maintains the rhythmic feel of a given setting.
Waves
Only the Wave and DigiWave parameters of the Oscillators will be altered. The other
Oscillator parameters (tuning, mix, and modulations in the Router) are excluded.
DigiWaves
DigiWaves will be selected in all Oscillators. The DigiWave number will be altered. The
other Oscillator parameters (tuning, mix, and modulations in the Router) are excluded.
Filters
The filter parameters are varied. The parameters included are: Filter Structure (series or
parallel wiring), Blend, Filter Mode, Cutoff Frequency, and Resonance for Filter 1 and 2,
Fatness, Filter FM of Filter 2.
Envs
All envelope parameters of all three envelopes ENV 1, ENV 2 and ENV 3 are varied. The
Vector Envelope is excluded.
LFOs
All LFO parameters of all LFOs are varied.
Router
All Router parameters of all Router channels are varied with all Intensities, Target, via,
and Source parameters.
FX
All effects parameters are varied.
Vector Envelope
All Vector Envelope parameters are varied, including the XY routing of the Square.

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Vector Env Mix Pad
The Oscillator mix levels (Triangle cursor positions) of the Vector Envelope points are
altered. The rhythm and tempo of the modulation (the time parameters of the points)
will not be altered.
Vector Env XY Pad
The Square cursor positions of the Vector Envelope points are altered. The XY routing
won’t be altered. The rhythm and tempo of the modulation (the time parameters of the
points) will not be altered.
Vec Env Times
Only the time parameters of the Vector Envelope points will be altered.
Vec Env Structure
The Vector Envelope structure will be altered: All times, the Sustain Point, the number
of points and all loop parameters.
Note: It’s recommended that you save any good sounds resulting from the RND
process, as you work. Do this under a new name (Setting > Save Setting as…) in the
Plug-in window.

Tutorials
You will find the settings for these tutorials in the Tutorial Settings folder in the settings
menu (in the head of the ES2 Plug-in window).

Sound Workshop: Logic ES2
Tutorial Setting: Analog Saw Init
Topics: Sound Design from Scratch, Filter Settings, DigiWaves
This is designed for use as a starting point when programming new sounds from
scratch. Professional sound designers like to use such scratch settings when
programming entirely new sounds, usually as follows: An un-filtered sawtooth wave
sound without envelopes, modulations or any gimmicks. This type of setting is also
well-suited to the purpose of getting to know a new synthesizer. It allows you to access
all parameters without having to consider any pre-set values.
• Start with the filters, the heart of any subtractive synthesizer. Check out the four
lowpass filter types 12 dB, 18 dB, 24 dB and fat (Filter 2) with different values for Cut
(Cutoff Frequency) and Res (Resonance). Define Env 2 as filter envelope. This
modulation wiring is pre-set in the router.

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• Set Filter Blend to its left-most position, which will allow you to listen to Filter 1 in

isolation. In many circumstances, you’ll probably prefer to use Filter 2, but Filter 1 has
its advantages. In addition to the lowpass filter with 12 dB/octaves slope (Lo), Filter 2
also offers: a highpass, peak, bandpass (BP), and band rejection mode (BR). Filter 1’s
lowpass sounds softer in comparison to Filter 2. It is best-suited to sounds where the
filter effect is/should be less audible (Strings, FM-Sounds). Distorted TB-303 style
sounds are more easily achieved with Filter 1.
• This setting is also ideal for checking out the oscillator waveforms. The analog
waveforms can be set in the Editor view. In order to select the DigiWaves, set Osc 1
Wave to DigiWave.
Tutorial Setting: Analog Saw 3Osc
Topics: Three Detuned Sawtooth Oscillators, Unison Mode
Fat synthesizer sounds have always been popular, and are likely to remain so, given
their use in modern trance and techno styles. This setting features three detuned
oscillators, and sounds fat as it is. The following will introduce you to some additional
tools to fatten the sound even more.
• Check out the 3-oscillator basic sound with different filter and envelope settings.
• Check out the chorus effect at different Intensities and Speeds.
• Engage Unison mode and select a higher setting for Analog. As the sound is
polyphonic, each note is doubled. The number of notes that can be played
simultaneously will be reduced from 10 to 5. This will make the sound rich and broad.
Combining Unison and higher values for Analog will spread the sound across the
stereo spectrum.
In many factory settings, the Unison mode is active. This demands a lot of processing
power. If your computer isn’t fast enough, you can switch off the Unison mode and
insert an Ensemble effect in a bus, for use with several plug-ins. This will save lots of
processing power. Another way to save CPU resources is to bounce several Audio
Instrument tracks—which place high demands on the processor—to an audio track.
Tutorial Setting: Analog Unison
Topics: Extremely Detuned Monophonic Analog Sounds, Effects
There’s nothing fatter than this heavily detuned, un-filtered basic sound. As with the
example above, three sawtooth oscillators are used, but they are detuned further. The
combination of Unison and Analog (set to a high value) is essential again, but this time,
monophonic mode is used to stack ten voices. Without further effects, the result is an
extremely fat lead sound, as used in countless dance and trance productions. With
appropriate filter and envelope settings, trendy arpeggio and sequencer sounds can
easily be set up.
• Set the Cutoff Frequency of Filter 2 to 0. This will activate the preset filter envelope.
Feel free to check out different envelope settings.
• Switch Osc 1 to sound one or two octaves lower.

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• Increase Drive or Distortion.
• Set Env 2 to be velocity sensitive. This allows for velocity sensitive filter modulations.
• Insert a stereo delay effect in the audio instrument channel strip of the ES2. In order

to delay several Audio Instruments, you might prefer to insert the effect into a Bus,
which is accessed via each channel’s Send.
Logic incorporates reverb and delay effects which are essential for many synthesizer
sounds. These aren’t integrated into the ES2, ensuring that no processing power is
unnecessarily wasted.
Tutorial Setting: Analog Bass clean
Topic: Clean Bass Settings with One Oscillator Only
Not every sound needs to comprise of several oscillators. There are numerous simple
and effective sounds which make use of a single oscillator. This is especially true of
synthesizer bass sounds, which can be created very quickly and easily through use of
this basic setting.
The basic sound is a rectangular wave, transposed down by one octave. The sound is
filtered by Filter 2. What’s special about this sound is its combination of Legato and Glide
(portamento). As long as you play staccato, no glide effect will occur. If you play legato,
the pitch will smoothly glide from one note to another. All keys must be released
before striking a new key, in order to retrigger the envelopes.
• Check out different filter and envelope settings.
• Replace the rectangular wave with a sawtooth.
• Vary the Glide settings.
Editing works best while a bass line is running. Create a monophonic MIDI Region, with
most notes played staccato and some legato. This can provide some interesting results
with very long Glide values.
Tutorial Setting: Analog Bass distorted
Topic: Distorted Analog Basses
Filter 1 is engaged, with high settings for Drive and Distortion. This filter is better suited
to the creation of distorted analog sounds than Filter 2.
• Check out Filter 2 by setting Filter Blend to its right-most position. You’ll hear that
Filter 1 works better with distorted sounds.
• In order to control the filter modulation, move the green sliders of the first
modulation channel in the router. This controls the modulation intensity.

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Tutorial Setting: FM Start
Topic: FM Intensity and Frequency
These are the first steps to take when learning about linear Frequency Modulation (FM)
synthesis. You’ll first hear an un-modulated sine sound, generated by Oscillator 1.
Oscillator 2 is switched on and set to produce a sine oscillation as well, but its level is
set to 0: Just push the cursor in the triangle in the uppermost corner.
In the ES2, Oscillator 1 is always the carrier, and Oscillator 2, the modulator. So
Oscillator 2 modulates Oscillator 1.
• Adjust the intensity of the frequency modulation by slowly moving the wave selector
from Sine to FM. You will hear a typical FM spectrum, with the carrier and modulator
set to the same frequency.
• Alter the modulator frequency (Oscillator 2) by adjusting Fine Tune from 0 c to 50 c.
You’ll hear a very slow frequency modulation that can be compared to the effect of
an LFO. The frequency modulation, however, takes place in the audio spectrum. It is
adjusted in semitone steps by the frequency selector. Check out the entire range
from −36 s to +36 s for Oscillator 2. You’ll hear a broad spectrum of FM sounds. Some
settings will remind you of classic FM synthesizer sounds.
• Select other waveforms for Oscillator 2. Sine is the classic, standard FM waveform, but
other waveforms lead to interesting results as well, especially the DigiWaves.
• You will achieve further interesting results by altering the carrier (Oscillator 1)
frequency. Check out the entire range: from −36 s to +36 s semitones here, as well.
The odd intervals are especially fascinating. Note that the basic pitch changes when
doing so.
Tutorial Setting: FM Envelope
Topic: Controlling FM Intensity by an Envelope and FM scaling
In this setting, you can control the FM intensity with an Envelope, generated by
Envelope 2. The modulation target is the range which falls between Sine and FM in the
Oscillator wave selector. The first Router channel is used for this. You can control a wider
range through the use of additional modulations, which have been pre-prepared for
you. All you need to do is set their values. As these modulations work without velocity
sensitivity, you can set them in the editor view by moving both the lower and upper
fader halves to their top-most positions.
• Set the second modulation channel to 1.0. You’ll hear how the modulation will now
“wander” through a broader sound range.
• Set modulation channels 3 and 4 to a value of 1.0 as well, and listen to the increase in
the sound range.

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• Following such drastic augmentations to the modulation range, the sound will have

become uneven. In the lower and middle ranges, it sounds nice, but in the treble
range the FM intensity appears to be too severe. You can compensate for this effect
by setting the target Osc 1 Wave to be modulated by the keyboard position (kybd) in
modulation channels 5 and 6. This results in a keyboard scaling of the FM intensity.
• As the sound range is so vast (due to the 4 modulations), two modulation channels
are required to compensate for this. Set the lower slider halves to their lowest
positions. Good keyboard scaling is essential for any FM sound.
Tutorial Setting: FM Drive
Topic: FM with Drive and Filter-FM
You can dramatically alter the character of FM sounds by applying Drive and Filter FM.
The results are reminiscent of the feedback circuits of classic FM synthesizers.
• Check out different Drive and Filter FM settings.
• Lower the Cutoff Frequency of Filter 2 to 0. Envelope 2 modulates Filter 2. This
modulation routing is preset in the setting.
Tutorial Setting: FM DigiWave
Topic: FM with Digiwaves
In this example, a DigiWave is used as an FM modulator. This results in a bell-like
spectra out of only two operators. Normally, this type of timbre could only be produced
through the use of a larger number of sine oscillators.
In order to create a fatter, undulating, and atmospheric quality to the sound, the
polyphonic unison mode has been engaged. Filter and amplitude envelopes have been
preset to shape the sound.
• Check out the variety of DigiWaves, as FM modulation sources.
• Check out different Analog parameter values.
Tutorial Setting: FM Wavetable
Topic: FM with Wavetables
You can program the most vivid FM sounds when the modulation source “morphs”
between different Digiwaves. The morphing in this setting is controlled by LFO 2 in this
setting. The tempo of LFO 2 (and therefore the morph) depends on the sequencer
tempo (here: 2 bars).
• Set LFO 2 to different waveforms. Lag S/H (smooth random), in particular, should be
fun.
• Check out different FM intensities and Oscillator frequencies.
• Alter the modulation intensity of the first modulation channel (LFO2 modulates Osc2
Wave) and the LFO 2 rate.

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Tutorial Setting: FM Megafat
Topic: Distorted FM in monophonic Unison
This sound is hard core, and is well-suited for distorted basses and guitar-like sounds. In
its treble range, this sound gets rather “rude”. This cannot be compensated for by
scaling, but not every sound has to be “nice” over the entire keyboard range!
• Check out extreme detunings by adjusting the Analog parameter.
• Check out the Flanger with this sound.
• Engage the filter envelope by lowering the Cutoff Frequency of Filter 2 down to 0.
• Add some Glide to lead sounds.
• As always, when it comes to FM: You can dramatically alter the sound by varying the
frequencies of the oscillators. Make sure you check out the odd intervals, as well.
Tutorial Setting: FM Out of Tune and FM Tuned
Topic: FM with Unusual Spectra
If you’re unconcerned with the pitch of your sound, you can get the weirdest spectra
out of odd frequency ratios (oscillator intervals).
This setting offers a bell-like sound, reminiscent of a ring modulator. It was achieved
through a setting of 30 s 0 c, with the modulator set to a value of 0 s 0 c. Sounds like this
were very commonly used in the electronic music of the eighties, and have undergone
a resurgence in popularity in modern ambient and trance music styles.
You can further develop the sound by applying filtering, envelope modulations and
effects. There is, however, one little problem—the sound is out of tune.
• Use Oscillator 3 as a reference for the tuning of the FM sound, by moving the cursor
in the Triangle.
• You’ll notice that the sound is 5 semitones too high (or 7 semitones too low,
respectively).
• Transpose both oscillators 1 and 2 five semitones (500 ct) lower. Transposing them
upwards is not practical, as you’d need to select 37 s 0 c for Oscillator 1, which maxes
out at 36 s 0 c.
• It’s important to maintain the frequency ratio (interval) between Oscillators 1 and 2.
This means that Oscillator 1 will sound at 25 s O c and Oscillator 2 at −5 s 0 c.

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Tutorial Settings: PWM Start, PWM Slow, PWM Fast, and PWM Scaled
Topic: Slow and Fast Pulse Width Modulations with Oscillator 2
Pulse Width Modulation (PWM) is one of the most essential features of any
sophisticated analog synthesizer. Use this setting to manually control the pulse width
of a rectangular wave, set via the Wave control
Note: Avoid Drive and Distortion with PWM sounds.
• Select the “PWM Start” setting, and move the Wave control slowly back and forth

•
•
•

•

between the rectangular and the pulse wave symbols. Both are green. What you will
hear is a (manual) pulse width modulation.
Select the “PWM Slow” setting. Here, LFO 1 controls the pulse width modulation
source, not your manual movements. The result should be quite similar.
Raise the LFO 1 rate from its pre-set value of 0.230 to 4.400. The result is a classic, fast
PWM.
In this, and the next step, the PWM shall be set so that it sounds slower in the lower
keyboard range, and faster in the upper range. This is desirable for many sounds,
such as synthetic strings. First, reduce the LFO 1 Rate to 3,800.
Change the modulation intensity of the second router channel (target = LFO1 Rate,
Source = Kybd) to 0.46. This will alter the scaling of the PWM, making it sound faster
in the treble range. You can also hear this type of effect in the “PWM Scaled” setting.

Tutorial Settings: PWM 2 Osc and PWM Soft Strings
Topics: Pulse Width Modulation with Two Oscillators, PWM Strings
In order to make the sound fatter, add Oscillator 3, which can also be modulated in
pulse width. In fact, even the first oscillator can deliver PWM. In the “PWM 2 Osc”
setting, both oscillators are detuned in a relatively strong way. Develop your own
personalized PWM string sound, using this setting as your “base”.
• Adjust the Chorus intensity. You’ll probably choose higher values which make the
sound rather broad.
• Program Envelope 3 according to your taste. You should, at the very least, raise the
attack and release times. Define it to react to velocity, if you prefer. If you do not
intend to solely use the sound as a simple pad, a shorter Decay Time and a lower
Sustain Level of about 80 to 90% may be more appropriate.
• Reduce the Cutoff Frequency and Resonance of Filter 1 to make the sound softer.
• Save the new setting.
• Compare the result with the “PWM 2 Osc” setting. You’ll hear that the sound has
undergone a remarkable evolution.
• Compare it to “PWM Soft Strings”, which was created as described above. You’ll
probably notice a few similarities.

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Tutorial Setting: Ringmod Start
Topic: Ring Modulation
A ring modulator takes its two input signals and outputs the sum and difference
frequencies of them.
In the ES2, Oscillator 2 outputs a ring modulator, which is fed with a square wave of
Oscillator 2 and the wave of Oscillator 1, when Ring is set as Oscillator 2’s waveform.
Odd intervals (frequency ratios) between the oscillators, in particular, result in bell-like
spectra, much like those heard in the “RingMod Start” setting.
As discussed in the “FM Out of Tune” section, on page 288, the third oscillator can be
used as a tuning reference, in order to maintain a kind of basic tuning. On occasion,
you may find that it is nice to leave the sound out of tune, for use as a source of
overtones and harmonics for another basic wave, supplied by Oscillator 3.
Try to program an atmospheric bell sound, using your own imagination. Some hints:
• Experiment with the various frequency ratios of Oscillators 1 and 2. You may want to
use the 29 s 0 c/21 s 0 c ratio, which doesn’t sound out of tune at all. Ring modulation
is not only useful for bell-like sounds, It’s also good for a great variety of spectra
which tend to sound pretty “weird” at lower frequency settings. Also try alterations to
the fine tuning of the Oscillators.
• Check out an Intensity of 50% and a Rate, set to around 2/3 of the maximum value,
for the Chorus effect.
• Set the Attack and Release Times of Envelope 3 to taste.
• Check out Drive and Filter FM, if you like your sounds a little “out of control”.
• The rest is up to you!
Tutorial Setting: Sync Start
Topic: Oscillator Synchronization
If you select the synced square and sawtooth waveforms for Oscillators 2 and 3, they
will be synchronized with Oscillator 1. In the “Sync Start” setting, only Oscillator 2 is
audible and Oscillator 3 is switched off.
“Typical” sync sounds feature dynamic frequency sweeps over wide frequency ranges.
These frequency modulations (the “sweeps”) can be applied in various ways.
• Try the pre-programmed pitch modulation, assigned to the modulation wheel, first.
• In the second router channel, an envelope pitch modulation has been preprogrammed (target = Pitch 2, Source = Env 1). Setting the minimum value to 1.0
results in a typical sync envelope. Also check out shorter Decay Times for Envelope 1.
• In order to avoid a “sterile”, and lifeless sound after the decay phase of the envelope,
you may want to modulate the oscillator frequency with an LFO as well. Use the third
router channel and set the minimum modulation applied by LFO 1 to about 0.50.
• Check out the synchronized square wave in place of the synced sawtooth wave.

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Note: Pulse width modulation is also available via the synchronized square wave of
Oscillators 2 and 3. A modulation of the wave parameters of these two Oscillators
results in a PWM if the synced square is selected.
Tutorial Setting: Vector Start and Vector Envelope
Topic: First Steps in Vector Synthesis
In this tutorial section you’ll find some useful hints for the programming of vector
envelopes. In the “Vector Start” setting, the mix of the Oscillators is controlled by the
vector envelope. Each Oscillator has been set to a different waveform.
• Switch from the Router view to Vector view.
• In its basic (default) setting, the vector envelope has 3 envelope points. Point 1 is the
start point, Point 2 the sustain point, and Point 3 is the target in the release phase. By
clicking the points, you can see that the mix is always set to 100% for Oscillator 1, in
the Triangle.
• Click Point 2, and move the Triangle cursor to Oscillator 2. You’ll hear a square wave,
instead of Oscillator 1’s sawtooth.
• Engage the Vector Envelope by switching the Solo Point parameter off. As long as it is
switched on, you will only hear the selected point, with no dynamic modulation.
Having switched Solo Point off, you’ll hear the sound moving from saw to square,
with every triggered note.
• Alter the pre-set time of 498 ms, between points 1 and 2.
• While holding Shift, click between points 1 and 2. This will create a new Point 2, and
the point formerly known as “Point 2” will become Point 3. The total time span
between Point 1 and Point 3 is divided into the times between Points 1 and 2, and 2
and 3. The division takes place at the click location. If you clicked at the exact midpoint, the new time spans are equal.
• Grab the newly created Point 2, and move its cursor in the Triangle to Oscillator 2.
• Grab Point 3, and move its cursor in the Triangle to Oscillator 3. Listen to the three
oscillators morphing from sawtooth to square to a triangular wave at the final sustain
point.
• Grab Point 4 (the end point) and move its cursor in the Triangle to Oscillator 1 (if it is
not already there). Listen to how the sound returns to Oscillator 1’s sawtooth wave,
following the release of the key.
Tutorial Settings: Vector Envelope and Vector XY
Topic: Vector Synthesis—XY Pad
This example starts where the first one left off. You have a simple Vector Envelope
consisting of 4 points, which is set to modulate the oscillator mix (the Triangle).
In this example, the Vector Envelope will be used to control two additional
parameters: The Cutoff Frequency of Filter 2 and Panorama. These are pre-set as the “X”
and “Y” Targets in the Square. Both have a value of 0.50.

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• Switch on Solo Point, in order to more easily listen to the settings for the single

points.
• Click Point 1. You will only hear Oscillator 1’s sawtooth.
• Move the cursor in the Square to the hard left, which results in a low Cutoff Frequency
•
•
•
•
•

for Oscillator 2.
Click Point 2. You will only hear Oscillator 2’s rectangular wave.
Move the cursor in the Square all the way down, which results in the right-most
Panorama position.
Click Point 3. You will only hear Oscillator 3’s triangular wave.
Move the cursor in the Square all the way up, which results in the left-most
Panorama position.
Switch on Solo Point. The sound begins with a strongly filtered sawtooth wave and
turns into an-unfiltered square wave. It initially sounds from the right, and then
moves to the left while morphing into a triangular wave. After releasing the key, the
saw sound will be heard.

This example sound isn’t very dramatic or interesting, but we wanted it to be as clear as
possible for the purposes of the tutorial.
Tutorial Settings: Vector Loop
Topic: Vector Synthesis Loops
This example is much more spectacular. The basic sound, without the Vector Envelope,
consists of three elements:
• Oscillator 1 delivers a metallic FM spectrum, modulated by Oscillator 2’s wavetable.
• Oscillator 2 outputs cross-faded DigiWaves (a wavetable), modulated by LFO 2.
• Oscillator 3 plays a PWM sound at the well-balanced, and keyboard-scaled, speed of
LFO 1.
Unison and Analog make the sound fat and wide.
These heterogenic sound colors shall be used as sound sources for the vector loop.
A slow forward loop is pre-set. It moves from Oscillator 3 (PWM sound, Point 1) to
Oscillator 1 (FM sound, Point 2), then to Oscillator 3 again (PWM, Point 3), then to
Oscillator 2 (Wavetable, Point 4) and finally, it returns to Oscillator 3 (PWM, Point 5).
Points 1 and 5 are identical, avoiding any transition from Point 5 to Point 1 in the
forward loop. This “transition” could be smoothed out with Loop Smooth, but this would
make the rhythmic design more difficult to program.
The distances between the points of the Vector Envelope have been set to be
rhythmically exact. Given that Loop Rate has been engaged, the time values are not
displayed in ms, but as percentages. There are four time values (each at 25%), which is
a good basis for the transformation into note values.

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• Switch off the Vector Envelope by setting Solo Point to on. This allows you to audition

the individual points in isolation.
• Take the opportunity to alter the cursor positions in the Square according to your

taste. As in the example above, the X/Y axes of the Square control the Cutoff
Frequency of Filter 2, and the Panorama position. Adjustments to these make the
sound more vivid.
• Activate the Vector Envelope by setting Solo Point to off. Check the result, and finetune the cursor positions in the Square.
• Alter the Loop Rate from the pre-set value of 0.09 up to 2.00. You will hear a periodic
modulation, much like that of an LFO. At this point, the modulation is not
synchronized with the song tempo. To synchronize the loop speed with the song
tempo, move the Rate cursor to the very left, and set a note or bar value.
• You can create faster rhythmic note values by clicking between two points and
setting the new time values (resulting from the division which occurs) to, say, 12.5%.
Tutorial Setting: Vector Kick
Topic: Bass Drum with Self-Oscillating Filter and Vector Envelope
Electronic kick drum sounds are quite commonly created with modulated selfoscillating filters. This approach can also be taken with the ES2, particularly when the
Vector Envelope is used for filter modulation. An advantage of the Vector Envelope, in
comparison with conventional ADSR envelopes, is its ability to define/provide two
independent decay phases. The distortion effect applies the right amount of “drive”
without losing the original sonic character of the drum sound.
In order to make this setting really “punchy”, you must make sure to activate Flt Reset.
Because all Oscillators are switched off in this setting, the filter needs some time to
start oscillating. At the start of each note, Flt Reset sends a very short impulse into the
filter to make it oscillating right from the start.
Through “tweaks” to the “Vector Kick” setting, you’ll probably be able to create any
dancefloor kick drum sound your heart desires. These are the parameters which allow
for the most efficient and significant variations:
• Filter2 slopes 12 dB, 18 dB, 24 dB,
• Distortion Intensity, Soft/Hard,
• Envelope 3’s Decay Time (D),
• Vector Envelope Time 1 > 2 (Pre-set: 9.0 ms),
• Vector Envelope Time 2 > 3 (Pre-set: 303 ms),
• Vector Time Scaling.

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Tutorial Settings: Vector Perc Synth and Vector Punch Bass
Topic: Percussive Synthesizers and Basses with Two Filter Decay Phases (Vector
Envelope)
As with the “Vector Kick”, this setting uses the Vector Envelope to control the Filter
Cutoff Frequency (with two independently adjustable decay phases). This would not be
possible with a conventional ADSR envelope generator. Try creating further percussive
synthesizers and basses by varying these parameters:
• Vector Envelope Time 1 > 2 (= Decay 1),
• Vector Envelope Time 2 > 3 (= Decay 2),
• Vector Time Scaling,
• Cursor positions in the Square for points 1, 2, and 3 (= Cutoff Frequency),
• Choosing other waveforms.

Templates for Logic’s ES2
Welcome to a brief programming tour of the ES2!
While working on the factory preset programming for the ES2, a number of beta
testers, sound programmers and other people involved in the project indicated that it
would be nice to start their programming work from templates, rather than entirely
from scratch.
Needless to say, creating templates which covered all sound genres is something of a
mission impossible. As you spend time familiarizing yourself with the ES2’s architecture,
you’ll start to understand why …
Nevertheless, we kept this basic goal in mind, and included this programming tour for
the ES2 as a part of the “toolbox” to help you learn and understand the ES2’s
architecture through experimentation. You’ll find that this approach is fun. You’ll also
discover, as you’re working through a number of simple operations, that results will
come quickly when starting to create your personal sound library.
As you become more familiar with the ES2, and what its myriad of functions and
parameters do, you can create your own templates for use as starting points when
designing new sounds.
-1- Clean Stratocaster (Slap Strat)
The target of this preset was the sound of a Stratocaster, with the switch between
bridge and middle pickup in the middle position (in phase). We tried to emulate the
noisy “twang”, typical of this sound’s characteristics.
This might be a useful template to start work on emulations of fretted instruments,
harpsichords, clavinets, and so on.

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Let’s have a look at its architecture:
Osc 1 and 3 provide the basic wave combination within the DigiWave field. Changing
the DigiWaves of both (in combination) delivers a huge number of basic variations—
some also work pretty well for electric piano-type keyboard sounds.
Osc 2 adds harmonics with its synced waveform, so you should only vary its pitch or
sync waveform. We kept the “noisiness” we were after in mind. There are a couple of
values which can be changed here, which will give you a much stronger, more
balanced signal.
An an old trick which delivers a punchy attack that the use of a “naked” wave wouldn’t
deliver was used—even with the best and fastest filters available: You use an envelope
(in this case No. 1) for a quick “push” of a wavetable’s window (or all wavetables
together, where it makes sense).
So set up Envelope 1’s Decay time for this short “push”, moving the wave selectors for all
Oscillators on the attack. (… actually it makes no sense on the synced sawtooth
Oscillator, No. 2, but what an effort to exclude it from the selection!—it just works this
way …)
So you can vary the “punchiness” of the content between:
• envelope 1’s contribution to overall attack noise, changing decay speed (a slow one
gives you a peak, a long one gives you a growl, as it is reading a couple of waves
from the wavetable).
• modulation destination: you can always assign this to each of the Oscillators
separately.
• start point (you vary the wave window start with minimum/maximum control of
EG1/Osc.waves modulation: negative values for a startwave before the selected
wave, positive starts from a position behind the selected wave and rolls the table
back) …
• Feel free to try out a couple of experiments with this wavetable-driving trick. The
growl effect works pretty well for brass sounds, and some organs absolutely shine
with a little “click”, courtesy of a wavetable “push”.
Envelope 2, which controls the filter, provides a slight attack used for the slapped
characteristics. Setting it to the fastest value eliminates the wah-like attack (and don’t
worry, there’s enough “punch” left).
For playing purposes, you’ll find that LFO 2 is used as a real-time source for vibrato. It is
assigned to the mod wheel and pressure.
Don’t concern yourself too much with the different settings for wheel and pressure—
this is only how we felt most comfortable bringing in the controls. Feel free to change
them!

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Velocity is set up to be very responsive because many synthesizer players don’t touch
keys in the manner of a piano player’s weighted-action-punch. As such, we ask that you
play this patch softly, or you may find that the “slap” tends to sweep a little. Alternately,
you can adjust the sensitivity of the filter modulation’s velocity value to match your
personal touch.
Also feel free to increase the Voices to maximum; we thought six strings would be
enough for a guitar, but for held or sustained notes, a few extra voices may come in
handy.
-2- The big twirl, basically (Wheelrocker)
This quite ordinary organ patch doesn’t hold any deep, high-end sound design
secrets: it is just a combination of three Oscillators with their wave levels mixed
together. we are positive that you’ll easily find a different combination which more
closely matches your vision of what an organ sound is like. Check out the DigiWaves.
Focus your attention on the mod wheel’s response: please hold a chord, and bring the
wheel in by moving it slowly upwards, until you reach the top (maximum).
What we intended to program with this mod wheel modulation was a simulation of an
accelerating rotor speaker (or “Leslie”).
The modulation routings do the following jobs:
• Modulation 1 (Cutoff 1) assigns envelope 2 to Filter 1 (the only one used for this
patch), and produces a little organ key “click” with the envelope. We also opened the
filter (with Keyboard as via) when playing the higher ranges of the keyboard (with
the maximum value).
• Modulation 2 and 3 (Pitch 2/Pitch 3) bring in LFO 1 vibrato, and both Oscillators are
modulated out of phase.
• Modulation 5 reduces the overall volume—according to personal taste, the organ’s
level shouldn’t increase too drastically when all modulations are moved to their
respective maximums.
• Modulations 6 and 7 (Pitch 2/Pitch 3) detune Oscillators 2 and 3 against each other,
within symmetrical values (to avoid the sound getting out of tune, overall). Again,
both work out of phase with modulations 2 and 3; Oscillator 1 remains at a stable
pitch.
• Modulation 8 brings in LFO1 as a modulator for panorama movement—this patch
changes from mono to stereo. If you would prefer a full stereo sound, with a slowly
rotating Leslie in its idle position, just set an amount equal to the desired minimum
value, thereby achieving a permanent, slow rotation. Another modification you may
wish to try is a higher value, resulting in more extreme channel separation.
• Modulation 9 speeds up LFO 2’s modulation frequency.
• Modulation 10: for increasing the intensity of the “big twirl” we added a little Cutoff
to Filter 1.

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Please feel free to find your own values for customization of this setup. While doing so,
keep in mind the fact that there are two modulation “couples”, which should only be
changed symmetrically (Mod. 2 and 3 work as a pair of “twins”, and also Mod. 6 and 7).
So, if you change Pitch 2’s maximum to a lower minus value, remember to set Pitch 3’s
maximum value to the same positive amount (same goes for modulation pair 6 and 7).
You can also bring in LFO 2 to increase the pitch diffusion against LFO 1’s pitch and pan
movements. Just exchange it for LFO 1 on modulation 2 and 3—but note that there
will be no modulation source for the Leslie acceleration—so you’ll need to use it in a
static way, just fading it in. Alternately, you’ll need to sacrifice one of the other
modulations in favor of a second twirl.
For another stereo modification of the idle sound, you can use the patch in Unison
mode with a slight detune (please adjust the “analog” parameter for this).
-3- Something Horny (Crescendo Brass)
First of all—the tasks of the Oscillators:
• Oscillator 1 provides the basic brass wave—“sawtooth”.
• Oscillator 2 provides a—not that brassy—“pulse” wave, which brings in the

“ensemble”. It is pulse-width modulated by LFO 1 (Modulation 4).
Please note that for any modifications, the following critical point should be taken into
account. There are four (4) parameters, which behave in an entirely different fashion,
once one of them is changed. As such, all four must be changed when making
adjustments:
• You may adjust the initial pulse width of Oscillator 2’s wave parameter. We selected a
sort of “fat” position, close to the ideal square wave because we wanted to program a
full, voluminous synth-brass sound.
• Modulation 4 adjusts the modulation intensity, which means: how far does the
range differ from “fat” to “narrow”, when being pulse width modulated? Set with the
Minimum parameter.
• The rate of LFO1 directly controls the speed of the movement of the pulse width
modulation. For this patch, both LFO’s are used, to achieve a stronger diffusion effect
at different modulation speeds. As a bit of general advice, we suggest that you use
LFO1 for all permanent, automatic modulations because you are able to delay its
“job” with its EG parameter. You may use LFO 2 for all real-time modulations, which
you intend to access via modwheel, pressure or other controls while playing.
• Additionally, we set up a keyboard assignment as the Modulation 4 source because
all pitch or pulse-width modulations tend to cause a stronger detuning in the lower
ranges, while the middle and upper key zones feature the desired diffusion effect.
When using this parameter, you should initially adjust the lower ranges until an
acceptable amount of detuning resulting from the modulation is reached. Once set,
check whether or not the modulations in the upper zones work to your satisfaction.
Adjust the relationship between intensity (Max) and scaling (Min) values.

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Oscillator 3 generates a DigiWave, which we considered “brassy” enough within the
overall wave mix. As an alternative to the DigiWave, we could have used another
modulated pulse wave to support the ensemble, or another sawtooth wave to achieve
more “fatness” when detuning it with Oscillator 1’s sawtooth wave.
What we were after, however, was to have a little bit of growl, achieved through a short
wavetable “push”, as described for the Stratocaster patch, on page 294. This
configuration is set up in Modulation 3 (Oscillator 3 Wave moved by Envelope 1’s
Decay).
Other controls
Envelope 1 also effects the pitch of Oscillator 2 against Oscillator 3. This results in the
clash of both pitches against each other and also against the stable pitch of Oscillator
1, in the attack phase of the sound.
The filter envelope’s design closes with a short stab in the attack phase and then opens
again for a slower crescendo phase.
We’ve assigned another real-time crescendo to the mod wheel, which also brings in an
overall pitch modulation, controlled by LFO 2.
In addition to all of this, we programmed a sort of “contradictory” real-time modulation
(by pressure) which closes the filters. This allows you to play with an additional
decrescendo, remotely controlled by touch. Try to get a “feel” for the patch’s response.
You’ll find that it offers quite a few controls for “expression”: velocity, pressure after
note on, and pressure in advance. Listen to what happens when pressing with the left
hand before hitting a new chord with the right hand, and allowing the swell come in.
-4- … making our job redundant …? (MW-Pad-Creator 3)
This is an attempt to create a patch which is able to create patches by itself—actually
we are still working on the question of how we can get it to save its own results …
The Basics
Again, Oscillator 2 is used for a pulse width modulation to create a strong ensemble
component (please refer to “-3- Something Horny (Crescendo Brass)” section, on page
297, for further information).
Oscillators 1 and 3 are set to an initial start wave combination within their respective
DigiWave tables. You can modify these and start with a different combination of
DigiWaves from the outset.
On Modulation 3, we assigned a wavetable “drive” to all three Oscillators via the mod
wheel. What this actually means is that you scroll through Oscillator 1 and 3’s
wavetables, and change Oscillator 2’s pulse width by moving the mod wheel.

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Try a careful, very slow movement of the modwheel, and you’ll hear drastic changes
within the wave configuration. Each incremental position of the wheel offers a different
complete digital pad sound. No frantic movements, please, otherwise this will sound
like an AM-radio.
Another potential modification procedure is hidden in the modulation intensity of
Oscillator 1, 2, and 3 wave’s parameter. As already mentioned with the Stratocaster
patch, the value of this intensity parameter assigns step width and direction through
the wavetables. You may try modifications to the amount and positive or negative
values.
We also discovered an interesting side-effect of FM assignment to Filter 2 (Modulation
4/Lowpass Filter FM). Moving to higher positions of the mod wheel, we increased the
frequency modulation on the filter, causing all cyclical “beats” (vibrating pitches,
detunes, pulse width) to be emphasized. This also results in a rough and hissing touch
to the overall sound character.
FM offers vast scope for experimentation, and you can decide between:
• An initial FM, using Filter 2’s FM parameter, which you can “redraw” (set a negative
modulation amount for Modulation 4’s maximum) by moving the mod wheel to its
top position.
• Or you may have permanent FM (and another modulation setup, saved for a different
assignment). You can also switch off FM, if you consider its effect too “dirty”
sounding.
Real-time control is via pressure for a vibrato (Modulation 10), and also for a slight
opening of the Cutoff to emphasize the modulation (Modulation 9).
-5- Another approach to “Crybaby” (Wheelsyncer)
Never obsolete—and undergoing a renaissance in new popular electronic music: Sync
Sounds
The technical aspects of forcing an Oscillator to sync are described in “Sync” on
page 237. Here’s the practical side of the playground.
Wheelsyncer is a single-oscillator lead sound, all others are switched off.
Although Oscillator 2 is the only one actively making any sound, it is directly
dependent on Oscillator 1.
If you change Oscillator 1’s pitch or tuning, the overall pitch of the sound will go out of
tune, or will be transposed.
The pitch of Oscillator 2 provides the tone-color (or the harmonics) for the sync sound.
Pitch changes are controlled by modulation 7’s setup, where we assigned Oscillator 2’s
pitch to the mod wheel.

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If you move the wheel, you can scroll through the spectrum of harmonics that we’ve
programmed for real-time changes. Any modification here starts with the pitch of
Oscillator 2 itself, which we’ve set to three semitones below the overall pitch. Feel free
to start with a different pitch for Oscillator 2; it won’t effect the patch’s tuning.
The next modification may be modulation 7’s intensity (or the interval). We have
selected the maximum value—maybe this is too extreme for your needs, so feel free to
reduce it.
Another modification lies in the tone color of the lead sound itself. We have switched
Oscillator 1 off, as we are already satisfied with the result. If you switch it on, it will offer
you the entire range of Oscillator 1’s waveforms; from DigiWaves, through the standard
synth stuff, up to a sine wave, which can be further modulated by FM.
All real-time controls are via the mod wheel: It is used for opening the filter on
modulation 6, a panning movement on modulation 8, and acceleration of panning
movement on modulation 9. If you have deeper modulation interests, please refer to
“-2- The big twirl, basically (Wheelrocker)” section, on page 296, where we used a
similar setup for the Leslie simulation.

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Ultrabeat

Ultrabeat is a rhythm synthesizer with integrated step
sequencer.
Ultrabeat’s synthesis engine is optimized for creating electronic and acoustic drum and
percussion sounds.
The sonic diversity possible with Ultrabeat is due to its various synthesis engines. In
addition to a new type of Phase Oscillator, sample playback, FM, and physical modeling
are also put to use.
Special attention has been paid to achieving the greatest possible range of dynamics
for its sounds. Ultrabeat features many extremely versatile functions that can vary the
timbre of the sound, dependent on the dynamics of the performance or other
modulation sources.
Ultrabeat is integrated into Logic as a software instrument. Its synthesis engine can be
directly controlled from within Logic. Ultrabeat also features an integrated step
sequencer, used to create rhythmic grooves comprised of patterns. The sequencer
displays “running light” type controls like those of classic drum machines.
In addition to the entry and playback of note information, the sequencer plays an
important role in the dynamic shaping of rhythms and sounds produced with
Ultrabeat.

The Structure of Ultrabeat
Before taking a closer look at the user interface, here are a few further comments on
the structure of Ultrabeat:
Most software synthesizers offer one synthesizer per plug-in instance. Ultrabeat,
however, places 25 independent synthesizers at your disposal. These synthesizers—
called drum voices in Ultrabeat—are optimized for the generation of drum and
percussion sounds.

301

The distribution of drum voices across the MIDI keyboard is simple and easily
explained: the first (starting from the bottom) 24 MIDI keys are each assigned a single
drum voice. The 25th drum voice is an exception, and can be played chromatically over
three octaves.
You can compare Ultrabeat to a drum machine that features 24 drum pads plus a builtin three octave keyboard.
Ultrabeat’s 24 drum pads are assigned to the first 24 keys of a standard MIDI keyboard
(corresponds to MIDI notes C1-B2). The three octave keyboard for the 25th synthesizer
begins (lowest note in the range) at C3.
For the sake of simplicity (and to stay with the drum machine analogy), we’ll refer to
the independent synthesizers (drum voices) as sounds which, combined, form an
Ultrabeat drum kit.

Overview of Ultrabeat

Ultrabeat’s user interface is divided into three functional sections. The physical
arrangement of these sections illustrates the signal flow within the plug-in:
The assignment section of the plug-in is found on the left hand side. Here, the
individual sounds are selected, mixed, and organized.
The parameters of an Ultrabeat sound are displayed to the right of the assignment
section. A sound’s parameters are displayed once a drum name (in the assignment
section) has been clicked with the mouse or is triggered via MIDI.

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The step sequencer is located at the bottom of the Ultrabeat window. It can be used in
place of, or in addition to, MIDI notes entering Ultrabeat’s input section (from Logic) to
control sounds.

Saving and Loading Settings
In order to save and reload Settings, Ultrabeat uses the procedure common to all Logic
plug-ins: as is customary, settings are saved and loaded via the Settings menu.

An Ultrabeat setting contains:
• The drum kit, which consists of 25 sounds, inclusive of assignment and mixer
settings.
• The complete settings of all parameters for all 25 sounds.
• The sequencer settings and all 24 patterns, including the trigger and velocity/gate
rows for all 25 sounds.
Note: The joint recall of all of this data when you load an Ultrabeat setting makes a lot
of sense as the musical effect of the patterns, especially those with sequenced gate and
velocity parameters, are often intimately connected to the sounds being used.

The Assignment Section
The assignment section of Ultrabeat is found on the left hand side. Individual sounds
are selected, mixed, and organized here.
MIDI Control
The first 24 voices of an Ultrabeat instrument are triggered by MIDI notes C1 to B2.
Each of these notes triggers a different sound. The first drum voice is assigned to the
lowest note (C1) and is displayed at the bottom of the onscreen keyboard. The keys
found above this correspond to the ensuing drum voices and MIDI notes, in ascending
order. While the first 24 drum voices are each triggered by a single key, the 25th sound
is chromatically playable over a span of three octaves (from C3 upwards). The 25th
sound is represented by the C3 key on the onscreen keyboard.
The assignment of MIDI notes to drum voices is pre-configured and can’t be changed.
In order to quickly adapt an Ultrabeat drum kit to play a finished pattern/MIDI Region,
the individual drum voices can be exchanged or copied within Ultrabeat, or reassigned
by using a Mapped Instrument Object in the Logic Environment.
Selecting the Sounds
Click on the name of the sound in the assignment section to select the relevant drum
voice. The sound’s parameters will be displayed in the synthesizer section to the right.
Click the corresponding note of the onscreen keyboard to play the drum voice.

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You can also select the sound by using MIDI note input. To do so, activate the Voice
Select button in the upper left hand corner of the Plug-in window.
Note: The automatic sound selection function activated by the Voice Select button is
useful for quick selection of different sounds, which are then displayed for editing.
Note: Given the obvious situation of Ultrabeat receiving numerous trigger notes from
Logic or the integrated step sequencer, automatic sound selection would result in
constant, visually disturbing changes to the parameter display. To avoid this issue, the
Voice Select function turns off automatically when a rapid succession of trigger notes
occurs.
Naming and Organizing the Sounds
Double-clicking on the name of a drum voice opens its text entry field, allowing you to
(re)name it. Press Return or click anywhere outside the text entry field to complete the
naming operation.
Control-clicking on the sound name opens a contextual menu that allows the copying
and replacement of sounds.
• Copy: This command copies the selected sound to the Clipboard.
• Paste: This command replaces the selected sound with the sound from the
Clipboard.
• Swap with Clipboard: This command exchanges and replaces the selected sound
with the sound from the Clipboard.
Note: You can also open this menu by right-clicking on the drum sound name.
Note: Naturally, the contextual menu’s Paste and Swap with Clipboard commands
require an initial Copy command (to place data in the Clipboard) before functioning.
Copying and replacing drum sounds within an individual drum kit always includes any
step sequencer data that may be present. This ensures that copied drum sounds are
not separated from their sequencer data, which often contains velocity or gate
information for a specific sound.

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The Drum Mixer
The assignment section contains a mixer for the 25 sounds found in an Ultrabeat drum
kit. It allows you to adjust each sound’s volume and pan position, and also offers a
Mute and Solo button.

Volume
The individual volumes of all sounds are indicated by blue bars, providing a complete
overview of all levels within the kit. You can adjust the volume of the sound, in relation
to Ultrabeat’s total output level, by dragging the blue bar beneath the sound name.
The Master (Volume) control is found above the 25th drum voice and controls the total
output of the Ultrabeat kit at the main output.
Mute
You can mute individual sounds in a drum kit by pressing the Mute button (M) to the
right of the name.
Solo
You can listen to sounds in isolation by pressing the Solo button (S), found beside the
Mute button.
Pan
The rotary knob to the right of the Mute and Solo buttons controls the placement of
the signal in the stereo field (Panorama).
Individual Outputs
Ultrabeat features eight separate stereo outputs, and can be inserted as a multichannel instrument. In this situation, each drum voice can be independently routed to
individual outputs (or output pairs) by using the Out(put Selection) pull-down menu,
found beside the Panorama knob. The following routing assignments are
possible: Main, 3–4, 5–6, and so on through to 15–16. Drum voices that are routed to an
output pair other than Main are automatically removed from the main output(s).
Master Volume

The slider above the assignment section controls the levels of all drum voices in the kit.
Put another way, the overall mix level of all drum voices.

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The Synthesizer
The majority of Ultrabeat’s user interface is dedicated to creating and shaping
individual drum sounds; in short, Ultrabeat’s synthesizer. The parameters of the drum
sound selected in the assignment section are displayed in this synthesizer section.
Note: Despite its vast feature set, Ultrabeat’s user interface only requires a single Plugin window. Depending on the function selected, only a few parts of the synthesizer
section will change to display different parameters and operational elements, as
necessary.
The Signal Flow
Ultrabeat’s synthesis engine is based on classic subtractive synthesis principles.
If you look at the synthesizer section from left to right, you’ll recognize the classical
structure and signal flow of a subtractive synthesizer. First, the basic tonal material is
created by the oscillators and noise generator. A filter then takes away certain
frequencies from the raw sound, followed by volume shaping (envelopes). The details
of Ultrabeat’s functions and their importance become more apparent when you look at
the three dimensional interface, and recognize the different levels from front to back:
The elevated Filter section is in the middle. It’s a large, round control object. Its
placement and design are symbolic, as the filter section plays a central role in
Ultrabeat.
The filter receives its signal from the following sound sources: Oscillator 1, Oscillator 2,
the Noise Generator and the Ring Modulator. Their output sections are displayed by four
objects that sit adjacent to the filter (three round objects and the smaller, rectangular
ring modulator to the right of the filter). One level down, you’ll find the control
elements for these sound sources.
On each of the objects that adjoins the filter you’ll find a small, red signal flow button
which indicates whether the signals should proceed through the filter or bypass it on
their way to the output section of the synthesizer. Along the output path to the right,
the signals pass through two equalizers and a stage for stereo expansion or panoramic
modulation.
The output of the drum voice is then passed along to the mixer that is integrated into
the assignment section (see “The Drum Mixer” section, on page 305).
You can find a detailed description of all synthesis parameters in “The Synthesizer
Parameters” section, on page 309.

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Modulation
Ultrabeat was developed with special attention paid to dynamic sound shaping
possibilities. To this end, almost every sound parameter can be modulated. Ultrabeat
provides two powerful LFOs, four new types of envelope generators (Env 1–4), velocity,
and four freely-definable MIDI controllers as modulation sources.
The settings for the LFOs and envelope generators are represented graphically. They
are located above and below Ultrabeat’s output section.
Each modulation routing is set at the desired destination (at the sound parameter
itself ). Ultrabeat distinguishes between two types of modulation: Mod (standard
modulation) und Via (indirect modulation).
Mod and Via Modulations
You can modulate a sound parameter using an adjustable modulation value (called
modulation depth) with Mod. You can choose between two LFOs, four envelope
generators and the Max source, as sources for this modulation.
Via allows you to further tailor the modulation effect. To explain, the modulation depth
of the first modulation (Mod) can be modulated by a separate, independent source. The
intensity of this effect is set with the Via parameter. The sources for Via modulations
include velocity and four freely definable MIDI controllers.
A typical application for Via function usage is; increasing a pitch sweep as you play at
higher velocities, for example. To this end, an envelope (Env) is chosen as the Mod
source for the Pitch of an oscillator, and velocity (Vel) is chosen as the Via source. The
firmer the key is played, the higher (in pitch) it will sound—this is typical of synthesizer
tom tom sounds.
Exceptional Modulation Features
The design of the Mod and Via modulation options afford substantial differences
between Ultrabeat and other, more traditional, synthesizer designs. Ultrabeat’s Mod
and Via are given a target value that can be reached by modulation of the respective
target parameters, rather than indicating a modulation or effect intensity as a
percentage. The result of such modulation routing—its minimum and maximum effect
on the modulated parameter—can be set simply, and grasped at a glance, making the
complex subject of primary and secondary modulations an intuitive task.
As this method differs greatly from earlier approaches used in synthesis, we
recommend that you carefully read the “Modulation” section, on page 325, to fully
benefit from this innovative new functionality.

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Control Elements
All parameters can be set by clicking or grabbing them, and moving in an upward or
downward direction.
Note: If you hold Shift before clicking and moving a control, its value can be finetuned.
Repeated clicks on buttons steps through different operating states.
Move the mouse vertically while holding down the mouse button to adjust values in
number fields.
You can adjust envelopes graphically by grabbing the envelope handles (or the outer
edge of the envelope) with the mouse, and moving it/them.
The equalizer can also be adjusted graphically by dragging on the EQ graph.
The effect of modulations are determined by the small modulation controls found at
the control elements of the target parameter. If no source is selected in the Mod and
Via menus (they’re set to Off ), the modulation controls will remain hidden.
Note: All parameters can be reset to their default values by Option-clicking on the
respective numerical labels, rotary knobs or sliders.

The Step Sequencer
In the step sequencer section, you can see two rows—each consisting of 32 steps. In
the upper row, trigger information (note on) is either inserted or removed, the lower
row is used for controlling note lengths and velocity. The rows always correspond to
the sound currently selected in the assignment section (synthesizer parameters can be
seen in the main working area of the plug-in). Choosing another sound (via the
onscreen keyboard or MIDI) updates the sequencer display to show the rows that
correspond to the newly-selected sound.
The step sequencer is described more fully in “The Step Sequencer” section, on page
332.

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The Synthesizer Parameters
In this section, you’ll find a description of the individual parameters found in Ultrabeat’s
synthesizer section. A discussion of the signal flow can be found in “The Signal Flow”
section, on page 306.

The Sound Sources
A drum voice in Ultrabeat has four sound sources: two multi-synthesis capable
oscillators, a noise generator and a ring modulator.
Oscillator 1
To use Oscillator 1, you need to first turn it on. This can be done with the On/Off button
in the upper left corner of the Oscillator 1 section. When in an active state, the button
is red.

Note: When you program a drum sound, you can turn the individual sound sources on
or off with the corresponding On/Off buttons. You can also listen to the individual
components of the sound separately this way, and remove them from the patch if
necessary.
The volume of Oscillator 1 is controlled by the Volume knob on the right edge of the
Oscillator 1 section.

Volume can be modulated by the sources found in the Mod and Via menus. If a Mod
source is activated, the effect it has on Volume is set by the (Mod) ring that surrounds
the knob. If a Via source is activated, its effect can be set by moving the slider that
appears on the Mod ring. Colored areas between the Volume knob and its surrounding
ring clearly show the values of the Mod modulation (blue) and the Via modulation
(green), compared with the mean Volume value (red).
If neither a Mod or Via source is selected (both set to Off ), the Mod ring, its slider and
the colored areas remain hidden.

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The curved slider to the left of the Volume knob controls the pitch of the oscillator in
half step intervals. If you press Shift, you can adjust the pitch of Oscillator 1 in cent
intervals.

The pitch value is displayed numerically to the left of the slider. You can change the
displayed value by click-holding directly on the value field, and moving the mouse
vertically.
Pitch can be modulated by the sources found in the Mod and Via menus. If sources are
selected, small blue (Mod) and green (Via) sliders appear beside the pitch control. These
allow control over the effect of the Mod and Via modulation routing. The range affected
by parameter modulation is colored blue and green, and runs alongside the pitch slider.
Oscillator 1 can be switched between two different types of synthesis engines: Phase
Oscillator and FM. This can be done by clicking the appropriately labeled buttons at the
upper edge of the Oscillator 1 section.

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The Asym parameter can be modulated by the sources found in the Mod and Via
menus. This allows you to create dynamic sound changes at the oscillator level. The
effect of the Mod and Via modulations are adjusted with the small sliders to the left
and right of the Asym slider. The range affected by the modulations is colored blue
(Mod) and green (Via). If no source is selected in the Mod and Via menus (set to Off ),
the Mod and Via sliders remain hidden.
Note: The classic basic waveforms of analog synthesizers can be easily reproduced with
the Phase Oscillator: sine, rectangular, and sawtooth waves are each the result of
setting the Slope, Saturation, and Asym parameters to their minimum or maximum
values, in different combinations.
FM (Frequency Modulation)
In FM mode, Oscillator 1 generates a sine wave. Its frequency is modulated by the
waveform of Oscillator 2. Please remember that Oscillator 2 must be switched on to do
this. The more complex the Oscillator 2 waveform, the more partials will be created (by
increasing the FM Amount) during the FM process. You can watch the display to see
how the sine wave takes on an increasingly complex shape.
The FM Amount parameter can be modulated by the sources found in the Mod and Via
menus. If a Mod source is activated, the effect it has on FM Amount is altered by moving
the ring that surrounds the rotary knob. If a Via source is activated, its effect can be set
by the moveable slider that appears on the Mod ring. Between the rotary knob and its
surrounding Mod ring, colored areas clearly show the values of the Mod modulation
(blue) and the Via modulation (green), compared with the mean FM Amount value
(red).
If neither a Mod or Via source is selected (both set to Off ), the Mod ring and slider
remain hidden.
Note: While the Phase Oscillator is well-suited for simulating analog waveforms and
analog-style sounds, FM mode offers bell-like digital tones and metallic sounds.
Filter Bypass Button
Between Oscillator 1 and the filter section you’ll find a signal flow switch that controls
the routing (Filter Bypass button). Repeated mouse clicks will send the signal to the
filter (Filter Bypass button turns red), or bypass the filter and send it directly to the EQ
section (Filter Bypass button remains gray).

The direction of the arrow on the Filter Bypass button illustrates the routing.

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Note: The Filter Bypass button simply determines the signal flow. It doesn’t turn the
oscillator on or off. Use the oscillator On/Off button for this (see above).
Oscillator 2
To use Oscillator 2, you first need to turn it on. This can be done with the On/Off button
in the lower left corner of the Oscillator 2 section. When active, the button is red.
Note: When you program a drum sound, you can turn the individual sound sources on
or off with the corresponding On/Off buttons. You can also listen to the individual
components of the sound separately this way, and remove them from the patch if
desired (or necessary).
The volume of Oscillator 2 is controlled by the Volume knob on the right edge of the
Oscillator 2 section.
Volume can be modulated by the sources found in the Mod and Via menus. If a Mod
source is activated, the effect it has on Volume is set by the (Mod) ring that surrounds
the knob. If a Via source is activated, its effect can be set by moving the slider that
appears on the Mod ring. Colored areas between the Volume knob and its surrounding
ring clearly show the values of the Mod modulation (blue) and the Via modulation
(green), compared with the mean Volume value (red).
If neither a Mod or Via source is selected (both set to Off ), the Mod ring, its slider and
the colored areas remain hidden.
The curved slider to the left of the Volume knob controls the pitch of the oscillator in
half step intervals. If you press Shift, you can adjust the pitch of Oscillator 1 in cent
intervals.
The pitch value is displayed numerically to the left of the slider. You can change the
displayed value by click-holding directly on the value field, and moving the mouse
vertically.
Pitch can be modulated by the sources found in the Mod and Via menus. If sources are
selected, small blue (Mod) and green (Via) sliders appear beside the pitch control. These
allow control over the effect of the Mod and Via modulation routing. The range affected
by parameter modulation is colored blue and green, and runs alongside the pitch slider.
Oscillator 2 can be switched between three different types of synthesis engines: Phase
Oscillator, Sample, and Model. This can be done by clicking the appropriately labeled
buttons at the lower edge of the Oscillator 2 section.

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Phase Oscillator
The waveform of the Phase Oscillator can be “twisted” with the Slope, Saturation, and
Asymmetry parameters, and shaped into almost any basic synthesizer waveform. The
effects of these three parameters are graphically illustrated in the waveform display
within the oscillator section. Setting all three parameters to zero values will cause the
oscillator to produce a sine wave.
The Slope parameter determines the slope or steepness of the waveform. The higher
the Slope value, the steeper the waveform. The resulting sound takes on an increasingly
nasal character as steepness is increased.
Increasing Saturation values clip the waveform, gradually molding its shape towards a
rectangular waveform. This results in a corresponding increase in odd numbered
overtones.
Sine and rectangualar waves (and all variations in-between) are achieved with the Slope
and Saturation parameters. Asym (Asymmetry) “tilts” the waveform in the direction of a
sawtooth wave, making the sound more edgy.
The Saturation parameter can be modulated by the sources found in the Mod and Via
menus. This allows you to create dynamic sound changes at the oscillator level. The
effect of the Mod and Via modulations are adjusted with the small sliders to the left
and right of the Saturation slider. The range affected by the modulations is colored blue
(Mod) and green (Via). If no source is selected in the Mod and Via menus (set to Off ),
the Mod and Via sliders remain hidden.
Note: Oscillator 1 differs from Oscillator 2 as it is Saturation not Asymmetry that can be
modulated. This difference means that when both oscillators are in Phase Oscillator
mode they can produce different types of sounds.
Sample
A selection of multi-layer drum and percussion samples that were specially created for
Ultrabeat and its function set are included with Ultrabeat. You can load these via the
sample function in Oscillator 2. You can also load your own samples in AIFF, WAV or SDII
stereo interleaved format.
To accomplish this, activate the button labeled Sample in Oscillator 2. You can see how
the control elements of Oscillator 2 change—among other things, a waveform display
appears.

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Samples are selected in a dialog box, which can be reached by clicking on the arrow
(or no sample loaded text) in the upper left corner (or top) of the waveform display. In
addition to the supplied Ultrabeat multi-layer samples, it is also possible to use this
dialog to select and load an audio file of your own choice. It should be noted, however,
that the velocity layering function is not available for such samples.
Note: When saving a drum kit using the Settings menu, the location of the sample is
saved with the setting. The Ultrabeat setting doesn’t actually save the audio files
themselves—only a reference to their location. If you load a setting that contains a
reference to a sample that has been moved or erased, Ultrabeat will present you with a
dialog box that requests you to find it. To avoid this problem, it is highly recommended
that you use a dedicated Ultrabeat sample folder.
The Reverse arrow changes the playback direction of the sample (forwards/backwards).
The two Min/Max (Velocity) horizontal sliders below the waveform display determine
the start point of the sample—dependent on the dynamics of the performance. Min
determines the start point of the sample at the minimum velocity level (velocity = 1),
Max at the maximum level (velocity = 127). If Min and Max are set to the same value,
this corresponds to a static setting of the sample start point.
Every internal Ultrabeat sample consists of different layers that are velocity switched—
dependent on the dynamics of your performance. The layers that are switched to, in
accordance with incoming velocity values, is determined by the small Vel Layer slider on
the right. This slider determines which layer is triggered at the minimal level (velocity =
1). The second small slider on the left determines which layer sounds at the maximum
level (velocity = 127).
Note: User-supplied samples loaded into Ultrabeat cannot be separated into velocitydependent layers, and therefore the Vel Layer slider has no effect on samples you have
created and imported.

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Model
This oscillator type offers a physical model of a string instrument for the creation of
percussive sounds. The parameters at your disposal are based on the physical
properties of a real string.

Two contrasting exciters, each with different sound characteristics, are available. You
can toggle between them with the corresponding buttons (Type 1 and Type 2).
Note: In Ultrabeat’s oscillator 2 Model, an exciter is the agent or triggering device used
to initiate the vibration of the string. It should not be confused with the effect plug-in
of the same name.
In the Material Pad, you can set the Inner Loss and Stiffness string parameters. This
parameter actually determines the material qualities of the physical model.
Inner Loss determines the damping of the string which, in the real world, depends on
the material the string is made of (for example steel, glass, nylon, or wood). Damping
primarily affects high frequencies, and forces the sound to become more muffled and
smooth during the decay phase.
Stiffness controls the stiffness or rigidity of the string. In the real world, this depends on
the material the strings are made of and their diameter (or, more precisely: their
sluggishness). Rigid strings create an inharmonic vibration where the overtones do not
represent whole number multiples of the fundamental frequency. These overtones are,
in fact, slightly higher. Marked increases in rigidity (stiffness) ultimately transforms the
string into a metal rod.
Along the x-axis of the Material Pad you’ll find the value range for the Stiffness
parameter, and the value range for the Inner Loss parameter along the y-axis. To adjust
the parameters, click-hold on the dot in the Material Pad and move it.
Note: Click on the dot in the Material Pad while holding down Option if you want to
return the string parameters to their default values.
To the right of the Material Pad you’ll find the Resolution parameter. You can also use
this parameter to influence the overtone structure of the sound.

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In contrast to the other parameters of the Model oscillator, Resolution does not
reproduce a pre-defined real-world property of the physical model, but affects the
modelling process itself: higher values lead to an improved calculation resolution
which results in more overtones. Lower values reduce the precision of the calculations,
leading to fewer overtones and often to inharmonic spectra.
Filter Bypass Button
Between Oscillator 2 and the filter section you’ll find a signal flow switch that controls
the routing (Filter Bypass button). Repeated mouse clicks will send the signal to the
filter (Filter Bypass button turns red), or bypass the filter and send it directly to the EQ
section (Filter Bypass Switch remains gray). The direction of the arrow on the Filter
Bypass button illustrates the routing.
Note: The Filter Bypass button simply determines the signal flow. It doesn’t turn the
oscillator on or off. Use the oscillator On/Off button for this (see above).
The Ring Modulator
The ring modulator functions as its own sound source; its signal can bypass or be sent
into the filter, independent of Oscillators 1 and 2. Its volume can also be regulated.
Please note that both oscillators need to be switched on to use it.
The sound of the ring modulator is largely dependent on both of the oscillators, as it
modulates the output signals of both. Parameter changes, especially the tuning
relationships of each oscillator, have a direct effect on the sound of the ring modulator.
Note: The individual volumes of the oscillators have no effect on the process of ring
modulation.
The ring modulator doesn’t have an On/Off button like the oscillators. It is activated by
clicking directly on the Ring Mod label itself. When switched on, the label is red and
when off, it’s gray.

Note: As the ring modulator needs the signals of both oscillators to produce its output,
the ring modulator is muted when one of the oscillators is switched off. If you want to
hear the ring modulator’s signal in isolation (in order to better judge your settings),
temporarily set the volume of both oscillators to 0.
The slider adjusts the output volume of the ring modulator. Volume can be modulated
by the sources found in the Mod and Via menus. If sources are selected, small blue
(Mod) and green (Via) sliders appear beside the volume control. These allow control
over the effect of the Mod and Via modulation routing. The range affected by
parameter modulation is colored blue and green, and runs alongside the volume slider.

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Between the ring modulator and the filter section you’ll find a signal flow switch that
controls the routing (Filter Bypass button). Repeated mouse clicks will send the signal
to the filter (Filter Bypass button turns red), or bypass the filter and send it directly to
the EQ section (Filter Bypass Switch remains gray). The direction of the arrow on the
Filter Bypass button illustrates the routing.
Note: The Filter Bypass button determines the signal flow. It doesn’t turn the ring
modulator on or off. Use the Ring Mod field for this (see above).
The Noise Generator
The fourth synth engine is the noise generator. Noise contains—in a technical sense—
all tonal frequencies; that’s why our ears can’t recognize any tonality in a noise signal.
Despite this (or as a direct result of it), noise is an indispensable ingredient when
creating drum sounds. For this reason, Ultrabeat’s noise generator is outfitted with
extensive features.
To use the noise generator, you first need to turn it on. This can be done with the On/
Off button. When in an active state, the button is red.
Volume can be modulated by the sources found in the Mod and Via menus. If a Mod
source is activated, the effect it has on Volume is set by the (Mod) ring that surrounds
the knob. If a Via source is activated, its effect can be set by moving the slider that
appears on the Mod ring. Colored areas between the Volume knob and its surrounding
ring clearly show the values of the Mod modulation (blue) and the Via modulation
(green), compared with the mean Volume value (red).
If neither a Mod or Via source is selected (both set to Off ), the Mod ring, its slider and
the colored areas remain hidden.
The noise generator has its own filter which functions independently of Ultrabeat’s
multimode filter. The four Type buttons LP, HP, BP, and Byp allow you to switch the filter
between lowpass, highpass or bandpass modes, or deactivate it (Byp).

The names of the filter types illustrate how they work: A lowpass (LP) filter allows
frequencies that are lower than the Cutoff frequency (see below) to pass. This filter type
dampens higher frequencies, and makes the sound less sharp and bright.
A highpass (HP) filter has exactly the opposite effect. It filters out the lower frequencies
while leaving the higher frequencies untouched.

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The bandpass (BP) filter only allows a certain frequency range (a frequency band)
centered around the Cutoff frequency to pass. It can be used in the upper, as well as at
the lower, end of the frequency spectrum to reduce the highs and lows of a sound.
The Cut knob determines the Cutoff frequency, and defines the point in the frequency
spectrum where reduction begins. Depending on the type of filter you select, you can
make a sound darker (LP), thinner (HP) or more nasal (BP) by adjusting the Cut value.
Cutoff can be modulated by the sources found in the Mod and Via menus. If a Mod
source is activated, the effect it has on Cutoff is set by the (Mod) ring that surrounds the
knob. If a Via source is activated, its effect can be set by moving the slider that appears
on the Mod ring. Colored areas between the Cutoff knob and its surrounding ring
clearly show the values of the Mod modulation (blue) and the Via modulation (green),
compared with the mean value (red).
If neither a Mod or Via source is selected (set to Off ), the ring and slider remain hidden.
Increasing Resonance boosts frequencies that surround the Cutoff frequency. Values
range from 0 (no increase) to self-oscillation of the filter at high Resonance values.
Note: Self-oscillation is typical of analog filter circuits. It occurs when the filter feeds
back into itself and begins to oscillate at its natural frequency, when high resonance
values are used.
Dirt is a parameter developed especially for the noise generator. Turning up the Dirt
knob roughens up the pure, white noise appreciably, making it more grainy.
Note: The Dirt parameter is especially effective at high Resonance values.
Between the noise generator and the filter section you’ll find a signal flow switch that
controls the routing (Filter Bypass button). Repeated mouse clicks will send the signal
to the filter (Filter Bypass button turns red), or bypass the filter and send it directly to
the EQ section (Filter Bypass Switch remains gray). The direction of the arrow on the
Filter Bypass button illustrates the routing.
Note: The Filter Bypass button determines the signal flow. It doesn’t turn the noise
generator on or off. Use the On/Off button for this (see above).
In the noise generator, the Filter Bypass button operates as per the oscillators and ring
modulator: it determines whether the signal is sent to Ultrabeat’s main filter or
bypasses it. It has no effect on the independent filter contained in the noise generator.
This is deactivated with the Byp button in the noise generator filter section.
It is therefore possible to filter the noise generator signal twice. In many instances, you
may want the noise generator signal to bypass the main filter, freeing the main filter for
other duties—an important element when programming drum sounds.

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The Filter Section
The output signals of both oscillators, the ring modulator and the noise generator are
passed on to Ultrabeat’s central filter section (if they haven’t bypassed it through use of
the various Filter Bypass buttons). The filter section offers a multimode filter and a
distortion unit.
The order that sounds are passed through the filter and distortion unit is determined
by the red arrow found at the “equator” of the filter section. Repeated clicking on the
arrow switches between the ▲ (distortion, then filter) and ▼ (filter, then distortion)
settings.

Note: A description of Ultrabeat’s filter parameters and a discussion on the basic
concepts of subtractive synthesis and analog filters follows. If you’re new to
synthesizers, please read the “Synthesizer Basics” chapter for more information.
The Multimode Filter
A click on the word Filter in the upper middle section activates or deactivates the
multimode filter. In a deactivated state (the word Filter is gray: red when active), all
synthesis engine signals pass the filter unprocessed and are forwarded to the distortion
unit.
The multimode filter offers the following filter types: lowpass (LP), highpass (HP),
bandpass (BP), and band rejection (BR).
You can switch between the filter types by pressing the corresponding button directly
beneath the word Filter.

The names of the individual filters illlustrate their function: A lowpass (LP) filter allows
frequencies lower than the Cutoff frequency to pass. It removes (cuts) the highs of a
sound, making it darker and less bright.
A highpass (HP) filter allows frequencies higher than the Cutoff frequency to pass. The
lows of the sound are cut.
A bandpass (BP) filter allows a frequency band centered around the Cutoff frequency to
pass. Frequencies that lie further away (the lows and highs outside the band) are
filtered out. A sound with a lot of mid frequency content results.

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The abbreviation BR stands for Band Rejection filter. In this mode, the area (the
frequency “band”, to be more exact) around the Cutoff frequency is filtered out while
frequencies that lie further away (from the Cutoff frequency) are allowed to pass. The
mid frequencies become softer and the low and high frequencies remain unchanged.
Below the filter type buttons, you’ll find two buttons labeled 12 and 24. These allow you
the select the slope of a filter. All of Ultrabeat’s filter types offer a filter slope of either 12
or 24 dB/octave.
Note: Filters don’t completely remove the parts of the signal that you wish to filter out
and always work with limited precision in the selected “band”. The steepness or slope is
measured in decibels of damping per octave (dB/oct). Frequencies that are located
close to the Cutoff frequency are generally reduced less than those that are farther
away. The higher the slope value, the more apparent the level difference is between
frequencies that are nearer the Cutoff frequency and those that are further away from
it.
The Cut knob determines the Cutoff frequency of the filter.
Note: Adjusting the Cutoff frequency can make a sound darker (LP), thinner (HP), more
nasal (BP) or more transparent (BR), dependent on the type of filter chosen.
Cutoff can be modulated by the sources found in the Mod and Via menus. If a Mod
source is activated, the effect it has on Cutoff is set by the (Mod) ring that surrounds the
knob. If a Via source is activated, its effect can be set by moving the slider that appears
on the Mod ring. Colored areas between the Cutoff knob and its surrounding ring
clearly show the values of the Mod modulation (blue) and the Via modulation (green),
compared with the mean value (red).
If neither a Mod or Via source is selected (set to Off ), the ring and slider remain hidden.
Increasing Resonance boosts frequencies that surround the Cutoff frequency. Values
range from 0 (no increase) to self-oscillation of the filter at high Resonance values.
Note: Self-oscillation is typical of analog filter circuits. It occurs when the filter feeds
back into itself and begins to oscillate at its natural frequency, when high resonance
values are used.
If a Mod source is activated, the effect it has on Resonance is set by the (Mod) ring that
surrounds the knob. If a Via source is activated, its effect can be set by moving the
slider that appears on the Mod ring. Colored areas between the Resonance knob and
its surrounding ring clearly show the values of the Mod modulation (blue) and the Via
modulation (green), compared with the mean value (red).
If neither a Mod or Via source is selected (set to Off ), the ring and slider remain hidden.

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The Distortion Unit
Depending on the order determined by the red arrow in the filter section, the
distortion unit is inserted either before or after the multimode filter. It provides either a
bit crusher or distortion effect.
The desired mode is activated by clicking on the Crush or Distort button. The active
effect is indicated in red. If neither button is red the distortion unit is bypassed
Note: The distortion effect is modeled on an analog distortion unit, which distorts the
sound by overdriving the level. The bit crusher uses a digital process that deliberately
reduces the digital resolution of the sound (measured in bits), achieving an intentional
digital coloration of the sound. Both methods lead to distortions that are as tonally
divergent as the two approaches. Distortion offers a more “analog” feel while the bit
crusher can’t hide its digital origins (nor is it supposed to!).
The bit crusher and distortion effect are adjusted with the same three dials:
• Drive: Turning this parameter up increases the degree of distortion.
• Color: This parameter determines the basic sound of the distortion. Higher values

help you achieve a brighter sound and lower values lead to a darker, warmer tone.
• Level/Clip: The output volume is set here (Level) when in distortion mode. In bit

crusher mode, this dial determines the level required before distortion (Clip) begins.

Output Section
Depending on the status of each Filter Bypass button, the output signals of both
oscillators, the ring modulator and the noise generator are routed either; directly or via
the filter section to the output section of Ultrabeat. The output section passes signals
through both equalizers (EQ) and the pan/stereo spread section (in a pre-configured
order) before the final level is set, and the trigger behavior (of the signals) is adjusted.
2 Band EQ
Both equalizer bands have almost identical features. Their parameters are explained
jointly, but you can, of course, adjust band 1 (the upper EQ in the output section) and
band 2 separately.
Clicking on the Band 1 and Band 2 labels turns the individual band on or off. When
active, the field is red. If neither EQ is activated, the signal passes through unaffected.
The EQ Type buttons switch between two different types of EQs: shelving and peak.

In shelving mode, all frequencies above or below the set frequency are either increased
or reduced. In peak mode, only frequencies located near the set frequency are affected.

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The shelving EQ is activated by pressing the upper of the two EQ Type buttons. The peak
EQ is activated by pressing the lower of the two EQ Type buttons.
Note: The shelving filter in band 1 offers a low shelving EQ while the shelving filter in
band 2 features a high shelving EQ. Low shelving means that the frequencies below the
set frequency are affected. High shelving affects frequencies above the set frequency.
Note: Shelving EQs function similarly to synthesizer lowpass and highpass filters. The
fundamental difference: while lowpass and highpass filters merely dampen certain
frequencies (filter them out), shelving EQs also allow these frequencies to be boosted.
The EQ Gain knob is bipolar. Positive values (turned to the right) boost a certain
frequency range as determined by the EQ Type and Hz settings. Negative gain values (to
the left) lower the gain of the frequency range. If the Gain knob set to the mean value
of 0, the EQ has no effect.
Note: You can also return this knob to its neutral position by Option-clicking on it.
Alternately, you can click on the tiny 0 above the EQ Gain knob.
The frequency (measured in scale units called Hertz) is set by click-dragging vertically
on the Hz parameter field. This determines the frequency range to be boosted or
reduced.
Note: Option-clicking the Hz parameter returns its value to a neutral position. This is
200 Hz for the first band and 2000 Hz for the second. The selection of these default
frequencies was made in accordance with the different shelving characteristics of each
frequency band. Band 1 is designed to filter low frequencies and band 2, high
frequencies.
The Q factor is regulated by click-dragging vertically on the Q parameter field. The
effect of Q on the sound is heavily dependent on the selected EQ Type:
• With shelving filters, as the Q value goes up, the area around the threshold frequency
becomes more pronounced.
• With the peak EQ, Q determines the width of the frequency band selection: low Q
values select a broad band while high Q values select a very narrow band to be
boosted or reduced with the Gain control.
Editing the EQ Bands Graphically
The EQ bands each have their own display which shows changes on a frequency
response curve. The display provides immediate access to the Gain, Hz, and Q EQ
parameters. Just grab the graphical frequency response curve with the mouse, and
alter it by moving the mouse vertically and/or horizontally! Horizontal moves change
the EQ frequency, and vertical moves influence the Gain. At the peak (maximum point)
of the EQ, a handle can be dragged vertically to change the Q factor. This procedure is
much like that used in Logic’s Channel EQ.

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Pan Modulation/Stereo Spread
The EQ’s output signal is passed along to the pan/spread section. In the pan/spread
section, the placement of the sound in the stereo field (set in the assignment section’s
mixer) can be modulated (Pan Modulation mode), or the stereo basis of the sound can
be broadened (Stereo Spread mode). Activate the desired mode by clicking on the
appropriate button (Pan Mod or Spread). If neither mode is activated, the signal passes
through unaffected.
Pan Modulation
Pan Modulation varies the panorama position of a drum sound dependent on a Mod
and Via source. The modulation set here is relative to the panorama position set in
Ultrabeat’s mixer.
The panorama position set in the mixer is represented here by a thin, red line. To the
left and right of the line, small sliders (and corresponding menus) allow the adjustment
of the Mod and Via modulation routings.
Note: You cannot directly grab and move the red line (that represents panorama
position) shown in this section. In order to move the line, rotate the pan knob in the
mixer section (see the GUI detail below).

Stereo Spread
Stereo Spread broadens the stereo image, making it wider and more spacious.
Low Frequency applies the (spreading) effect to the bass frequencies: the higher the
value, the more prominent the effect becomes. Hi Frequency allows you to apply the
effect to the high frequencies.
Voice Volume
This rotary knob adjusts the output volume of the individual drum sounds. To be more
exact, you are controlling the Voice Volume with Env 4, thereby adjusting the maximum
volume level attained after the attack phase of Env 4.
The effect the envelope has on Voice Volume can also be modulated by a Via source.
Note: The leveling stage for Voice Volume precedes the sliders in the mixer. This
approach allows the starting volume of the individual drum voices to be set
independently of their relative levels in the drum kit mix.

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Trigger and Group Menus
The manner in which Ultrabeat reacts to a succession of incoming notes is individually
defined for each sound. These parameters are found in the output section, below the
Voice Volume knob.

Clicking the button below the Trigger label opens the Trigger menu, allowing you to
choose between Single and Multi trigger modes.
• Single: A new trigger note cuts off the (same) note that is currently playing.
• Multi: When a new note is played, preceding (currently playing) notes continue to

decay in accordance with their respective amplitude envelope settings (Env 4).
Clicking the button below the Group label opens the Group menu, allowing a choice
between the Off and group 1 to 8 settings. If two different sounds are assigned to the
same group, they will cut each other off. A typical use of this facility is when you’re
programming hi-hat sounds: when playing a real hi-hat, the closed hi-hat note cuts off
and mutes the ringing of the open hi-hat. This function is often referred to as “hi-hat
group” mode.
Note: While in Single Trigger mode, only the currently sounding note of the same
sound is cut off. A sound that is assigned to a group cuts off all other sounds
(regardless of note) in the group.
Clicking on the Gate button turns the Gate function on and off. If active, the sound is
immediately cut off as soon as the MIDI note is released (MIDI note off ), regardless of
envelope settings.
Note: The Gate function ensures that a specific sound does not play beyond a note off
event, as defined in the sequencer. A corresponding rhythmic definition of the exact
note off time is achieved with the Gate Length parameter in Ultrabeat’s step sequencer.
Logic’s sequencer allows you to quantize note off events, or precisely edit them
manually. Note length can be an important creative element when programming
rhythm tracks.

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Modulation
Numerous sound parameters can be controlled dynamically (modulated) in Ultrabeat.
The setting of modulation routings follows a universal principle that is explained in this
chapter.

The Principle of the Modulation Routings
Ultrabeat’s modulation routings feature three key players:
• The synthesizer parameter that you want to modulate (the modulation target)
• The source of the modulation (the modulation source)
• A second modulation source that can influence the intensity of the first modulation
(we call this a Via modulation)
Let’s look at an example to better understand how this works:
The Cut (Cutoff) paramater has a mean (default) value of 0.50. It’s not being modulated
yet as no modulation source has been selected in either the red Mod or blue Via menu
(set to Off ).

As soon as a modulation source is selected in the Mod menu (Env 1 in this example),
the ring around the rotary knob is activated. Grabbing and moving this ring with the
mouse allows you to set the value that this parameter will be increased to by the Mod
source (0.70 in the example).

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As soon as a modulation source is selected in the Via menu (Vel in this example), a
movable slider appears on the Mod ring. Grabbing and moving this slider with the
mouse allows you to set the maximum modulation value that can be reached through
use of the Via source (0.90 in this example).

So much for the settings. What do the markings found around the Cut knob mean, and
what is happening to the sound?
The Mod and Via controls indicate the minimum and maximum values that the
modulated parameter can attain (in comparison to the mean value). Put another way,
the result of the modulation. These controls don’t show us—as is customary with other
synthesizers—a percentage value describing their intensity, but state very
clearly: “These are the minimum and maximum possible values of the modulated
parameter”.
Back to our example; the frequency of the filter is set to the mean value of 0.50. When
the Mod source Env 1 enters the equation, the Env 1 envelope generator drives the Cut
value up from 0.50 to 0.70 (during the attack phase) and back down to 0.50 (during the
decay phase).
Note: You can view the exact values in the Help Tags that appear when you grab the
individual handles of various parameters.
If the Via source Ctrl A is introduced, the following interplay occurs: when Ctrl A
remains at its minimum value, nothing changes (yet); Cutoff continues to be modulated
between values of 0.50 and 0.70 by the envelope. A maximum value for Ctrl A causes
the envelope generator to vary the parameter between the values of 0.50 (the mean
value) and 0.90 (the Via amount).
You can see, at a glance, the degree of maximum influence on basic parameters by the
Mod and Via modulation sources: the area between the Mod and Via points shows the
amount that the modulation depth can be (further) altered by the Via modulation
source. In our example, the Cutoff can reach values between 0.70 and 0.90 depending
on the value being sent by Ctrl A.

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Here’s another example:

Cutoff is again set to 0.50, Env 1 now drives the value down to 0.25, and a maximum Ctrl
A value reduces the Cutoff frequency down to 0.
Here is another example that illustrates the simplicity and speed of Ultrabeat’s
modulation options:

In this example, you won’t just be changing the modulation intensity of Env 1 (which
affects Cutoff ) with the dynamics of your performance (Vel), but you’ll also control its
direction as well. Try this setting in Ultrabeat to create some extremely interesting
sounds.

Setting the Modulation Routing
Clicking on the Mod label opens the Mod menu. This is where you can choose one of
the LFOs or envelope generators (Env) as a modulation source.

The Off setting deactivates the Mod routing, and the Mod ring can no longer be
adjusted. In this situation, no Via modulation can occur either (this is because Via no
longer has a modulation target) and the Via slider disappears.

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Note: The Max setting produces a static modulation at maximum level. When the Mod
value is set to Max, the Via parameter is routed directly to the modulation target. This
way, velocity can be used as a direct modulation source, even though Vel is not
available as a source in the Mod menu.
Another example would be to set up an external MIDI fader unit with Ctrl A, B, C, or D
(see below). You could then use the Max item in the Mod menu to route the respective
Via source—Ctrl A, B, C, or D—to the parameter you’d like to control with one of the
faders on your MIDI fader box.
Clicking on the Via label opens the Via menu. This is where you choose the Vel or Ctrl A
to Ctrl D parameters.

Vel represents velocity.
Ctrl A to Ctrl D are four continuous controllers that can be assigned to four external
MIDI controllers. These assignments are made in the MIDI Controller Assignments area at
the upper right edge of the Plug-in window (see below). The assignments apply to all
sounds in the current Ultrabeat plug-in instance.

MIDI Controllers A–D
In the MIDI Controller Assignments area at the upper edge of the Plug-in window you
can assign a standard MIDI controller to each of the four controller slots: Ctrl A, B, C, or
D. Ctrl A, B, C, and D can be used as Via modulation sources within Ultrabeat. Use these
assignments to set up your external MIDI controller hardware to operate with
Ultrabeat. As examples; aftertouch or the modulation wheel of your MIDI keyboard.
Note: All MIDI Controller Assignment menus feature a Learn option. If this is selected,
the parameter will automatically be assigned to the first appropriate incoming MIDI
data message.

LFO 1/2
Among other items, two LFOs are available as modulation sources in the Mod menu.
LFO is the abbreviation for Low Frequency Oscillator. The LFO signal is used as a
modulation source. In an analog synthesizer, LFO frequency generally ranges between
0.1 and 20 Hz, which is outside our audible frequency spectrum.

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Note: The speed of the LFO in Ultrabeat can reach up to 100 Hz which, when compared
to analog synthesizers, offers a number of far-reaching possibilities.
Ultrabeat has two LFOs that offer identical feature sets. The parameters for each are
described jointly; you can, of course, adjust LFO 1 and LFO 2 completely independently
of each other.
The buttons labeled 1 and 2 select the corresponding LFO, allowing adjustments of
each LFO’s parameters. The blue On/Off button activates and deactivates the selected
LFO.

The LFO section display shows the LFO waveform, the shape of which is governed by
the Shape slider located underneath it. Dragging the slider from left to right causes the
waveform to fluidly morph from a sine to a triangle, and then finally to a square wave
(with variable pulse width), including all variations in-between. At the far right hand
position of the Shape slider, the LFO produces random waveforms.
The LFO speed (Rate) can be set independently (Free) or synchronised (Sync) to Logic’s
song tempo. Clicking either button once activates the corresponding mode.
The Rate knob determines the speed of the LFO. Depending on the Free/Sync setting,
Rate is displayed in either Hz or musical (measure) units.
The Ramp knob determines whether the output signal of the LFO is faded in or out.
Ramp works in a bipolar fashion: turning it to the right increases the Attack time of the
LFO, turning it to the left decreases the Decay time. In its middle position, Ramp has no
effect on the LFO. The Ramp value is displayed in milliseconds in its parameter field.
An LFO normally oscillates constantly. On percussive signals it can, however, be
interesting to limit the LFO cycles to a defined number. Ultrabeat allows you to set the
number of LFO cycles with the Cycle parameter. After completing the defined number
of cycles, the LFO stops oscillating.
Note: Try small Cycle values, and route the LFO to Osc Volume to create typical drum
flams or hand claps.
Note: The Cycle = 1 setting allows the LFO to function as an additional (albeit simple)
envelope generator.
The range of Cycle values extends from 1 to 100. Turning the knob to its maximum
value (all the way to the right) results in a permanent oscillation (an infinite number of
cycles).

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The Cycle parameter can also determine whether the LFO (waveform) is started from
the beginning (at a zero-crossing point) with each note trigger, or whether it simply
continues oscillating. A Cycle value of Inf (Infinity) forces the LFO to run freely. It is not
reset by incoming MIDI note on messages. When Cycle is set to values under 100, the
LFO will be reset by each new MIDI note on message (Note On Reset).
It’s really a question of taste as to whether or not you choose to trigger an LFO cycle
from the same spot or just allow it to oscillate freely, regardless of phase. The random
element of free-running LFOs can make many sounds fatter. This, however, can come at
the expense of a percussive attack—an often undesirable quality in a drum synthesizer.
Note: You can, of course, use minor shifting of the LFO phase—with the Cycle value set
to Infinity—to your advantage, adding an “analog” character to a drum sound, for
example.

Env 1 to 4
Further modulation sources available to you in the Mod menu include four identically
specified envelope generators. Envelope parameters are described in this section.
Note: In addition to it’s potential use in the Mod menus of various sound parameters,
Env 4 is permanently connected to the Voice Volume. In other words, Ultrabeat has a
“hard-wired” volume envelope generator.
Structure of the Envelope Generators
The default setting of the envelope generators is known as the one shot envelope
mode: after a key is pressed (note on message), the envelopes run their course,
regardless of how long the note is held. This setting is ideal for percussive signals
because it allows simple emulations of the behavior of natural percussion sounds. For
special cases such as sustained pad or cymbal sounds, you can activate a sustain mode
where the envelopes take the lengths of the played notes into account.
Editing the Envelopes Graphically
Before delving into the individual parameters, please take the time to familiarize
yourself with the graphical depiction of an envelope, shown below.
Ultrabeat’s envelope window provides a new type of envelope design, consisting of
Bezier curves in which two segments—attack and decay—make up the entire
envelope.

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In the envelope graphic, you can see various junction points of two different sizes. Both
of the larger handles on the x-axis (the horizontal, or time axis) control the attack and
decay times, respectively. A vertical line extends up from the first of the two handles,
and divides the envelope into an attack and decay phase. Both segments each have
two small curved junction points. You can move these in any direction to deform the
contour of the envelope, and freely shape its amplitude.
In order to move the curved junction points, simply grab them and drag them to a new
position. Experiment with the different junction points and you’ll quickly realize that
handling is very intuitive. You can also directly click-hold and drag any curve edge on
the envelope itself.
Envelope Parameters
In order to edit the envelope parameters, first select an envelope by clicking on the
desired 1 to 4 buttons. The parameters of the corresponding envelope can now be
directly changed in the envelope display window.
Attack Time
Attack time defines the period of time the envelope needs to reach its maximum value.
This is measured from the instant you press a key (note on). This period is called the
attack phase.
Grab the attack junction point (the left-most of the two handles found on the x-axis)
with the mouse, and move it to shorten or lengthen the attack time.
Note: To change the shape of the envelope in the attack phase, you can edit both
junction points found in this segment. It is also possible to directly grab the curve with
the mouse and alter it.
Decay Time
Decay time defines the period of time the envelope needs to fall back to a zero
amplitude, after it has reached its maximum value (defined in the Attack phase).
Moving the second junction point on the x-axis shortens or lengthens the decay phase.
Note: To change the shape of the envelope in the decay phase, you can edit both
junction points found in this segment. It is also possible to directly grab the curve with
the mouse and alter it.
Envelope Modulation
The time and shape of the envelopes can be modulated by velocity. Clicking in the
menu field below the envelope 1 to 4 buttons opens the Env Mod menu. Choose either
Time or Shape of the (A)ttack or (D)ecay phase as the modulation target.
The intensity of modulation is adjusted with the mod slider below the envelope display.
Note: When you modulate Shape, low velocity values lead to a “sagging” envelope
shape, while higher values cause the selected envelope segment to “bulge”.

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Note: When you modulate Time, increasing velocity values lead to a reduction in length
of the envelope segment. Lower velocity values increase the length of the envelope
segment.
Sustain
Activation of the Sustain button causes a red handle (and vertical line) to appear on the
x-axis. This can be moved horizontally—but only within the decay segment area. The
amplitude that the envelope reaches at the Sustain junction point is retained until the
MIDI note is released. After receiving the MIDI note off command, the envelope
continues for the remaining decay time.
Note: If the Sustain button is not activated, the envelope functions in “one shot” mode,
and the note length (MIDI note off command) is disregarded.
Zoom (to fit)
When you select the Zoom button, the envelope is enlarged to fill the entire width of
the display, making it easier to adjust junction points and curves. The graphic display is
quickly redrawn after any change is made to the Attack or Decay values.
Note: When the Zoom function is selected, the decay junction point can be dragged
beyond the right-hand edge of the display area, in order to lengthen the decay time.
After you release the mouse button, the envelope graphic is automatically resized to fit
the display area.
Zoom A/D
The Zoom A button only shows the attack phase across the entire width of the display,
and the Zoom D button only shows the decay phase. This allows easier and more
accurate edits to envelope shapes, even down to millisecond values.

The Step Sequencer
The integrated step sequencer allows all Ultrabeat sounds to be combined in
sequences, based on patterns. It’s design and use (step programming input) are based
on analog predecessors.
Dependent on your personal taste and favored musical style, you’ll want to control
Ultrabeat from either the integrated step sequencer or from Logic, when programming
rhythms. Combining both sequencers is also possible; they both can be active at the
same time, and are automatically synchronized with each other. Logic’s song tempo
stipulates the tempo of Ultrabeat’s internal step sequencer.
Before we turn our attention to Ultrabeat’s step sequencer, let’s take a brief look back at
the early days of sequencing.

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The Step Sequencer Principle
The basic idea behind analog step sequencers was to set up a progression of control
voltages, and output these step by step. In early analog sequencers, three control
voltages were usually created per step, in order to drive different parameters. The most
common usage was control of a sound’s pitch, amplitude, and timbre (Cutoff ) per step.
The control surface of analog sequencers often contained three rows of knobs or
switches on top of (or beside) each other, each row with 16 steps. Each row had its own
control voltage output, and the parameter it controlled was determined by the control
input (on the synthesizer) it was connected to. A trigger pulse determined the step
tempo. A “running light” (an LED) indicated the current step. This principle helped to
create an electronic music style whose appeal stems from the mesmerizing effect that
repeating patterns can have.
The “running light” programming concept also appeared in later drum computers. The
most well-known representatives of this category being the very popular Roland TR
series drum machines.
The introduction of the MIDI standard and increasing use of personal computers in
music led to a rapid decline in the step sequencer and related technology. More
modern concepts that didn’t adhere to the step and pattern principle came into vogue.
Despite this, step sequencers haven’t disappeared completely. Hardware “groove boxes”
have experienced a renaissance over the last couple of years. Their intuitive nature has
made them a favorite tool for rhythm programming.
Ultrabeat provides an integrated step sequencer of the newest design, that time-warps
the advantages of its analog predecessors to the present day. As part of the “dynamic
duo” with Logic, it raises modern rhythm programming to a new level.

Step Sequencing with Ultrabeat
Ultrabeat’s step sequencer contains 24 patterns—each consisting of 32 steps. You can
play 25 Ultrabeat sounds per step, each sound completely independent of the others.
The Control Surface of the Sequencer
The sequencer is divided into three sections.
• On the left you’ll find the parameters that globally control the pattern and sounds,
independent of the individual steps and patterns. These parameters are known as
the global parameters.
• At the bottom you’ll find the parameters of the currently selected pattern (pattern
parameters).

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• The actual sequencing takes place in the step grid above. In this section, a pattern of

32 steps is shown for each sound. The pattern grid of the sound that is currently
selected in the assignment section is shown. You can add or remove events to the
grid by simply clicking at the desired step position. Parameter values in the individual
steps are altered by grabbing and moving them with the mouse.

Global Parameters
A description of the parameters that apply globally to all internal sounds follows.
On/Off button
This button turns the step sequencer on or off.
Transport Button
The Transport button starts and stops the sequencer pattern. This allows you to inspect
the step sequencer pattern while the Logic song is stopped, for example. The Transport
button changes appearance, dependent on its current operating status.
The Transport button changes color when the step sequencer is, or is not, waiting for
incoming MIDI notes:
• Gray: The sequencer functions independently of incoming MIDI information.
• Blue: The sequencer interprets incoming MIDI notes between C-1 and B0 as

performance information. In this mode, MIDI messages can modify numerous
functions that control how the patterns are played.
Note: The step sequencer is always synchronized to Logic’s song tempo.
Swing
This rotary knob globally determines the swing intensity for all sounds that have the
Swing function activated (see “Swing Enable” on page 337).
Swing changes the distance between notes: notes on odd-numbered steps remain
unchanged, while even-numbered notes are slightly shifted. At a 0 setting (the knob all
the way to the left), the Swing function is not active. Turning the Swing knob up shifts
the affected notes towards the following note.
Note: Swing is only active at grid resolutions of 1/8 and 1/16.

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Pattern Parameters
A pattern has a maximum number of 32 steps and contains the total events of all 25
sounds. At the bottom edge of the Plug-in window you can select from one of 24
patterns, and set global parameters (for each pattern) that apply to all sounds.
Pattern # (Pattern Number)
Click the field next to the Pattern # label to open the Pattern menu, which allows you
to choose one of the 24 patterns.
If you’ve activated the function that allows you to switch between patterns via MIDI
notes (the Transport button turns blue), Pattern # indicates which pattern is currently
active (being played).
Control-clicking in the Pattern menu field opens a context menu where you can access
further pattern-specific commands (Copy, Paste, Clear).
Resolution
This parameter field displays the resolution of the pattern. It defines the metric unit of a
measure that is represented by the individual steps. Clicking the Resolution button
opens a pop-up menu that allows you to switch between different settings.
The 1/8 setting means that each step of the grid represents an eighth note. Given a
pattern length of 32 steps, the pattern would run for 4 measures (32 ÷ 8).
The Resolution parameter applies to the entire grid, and therefore, equally to all sounds.
Length (Number of Steps, Pattern Length)
This parameter defines the length of the pattern. The length of the grid can be
adjusted by dragging the value in the Length parameter field or the bar beneath the
swing buttons.
Note: The interplay between the Length and Resolution values allows the creation of
different kinds of time signatures. Here are a few examples: The values Length = 14 and
Resolution = 1/16 result in 7/8 time, Length = 12 and Resolution = 1/16 in 3/4 time, and
Length = 20 and Resolution = 1/16 in 5/4 time.
Accent
Individual steps can be strongly emphasized, or accentuated.

In order to program an accent, click on the blue LED above the desired step. The accent
applies to all sounds that fall on this beat, and the step in question is simply played
louder. The Accent slider to the left of the chain of blue LEDs globally determines the
volume of the programmed accents.

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Step Grid (The Heart of the Sequencer)
In the step grid area, the pattern is rasterized and displayed in numerous rows and
steps. The displayed grid applies to the sound that is currently selected in the
assignment area. Choosing a different sound (either via the onscreen keyboard/
assigment section or via MIDI) switches the grid display to match the newly chosen
sound.
The step grid area contains two rows—each with 32 fields:
Trigger On/Off (The Trigger Row)
In this row, consisting of the 1 to 32 buttons, triggers are placed on corresponding
steps. In other words: this is where you designate when (on which beat) the selected
sound plays.

A click on one of the 1 to 32 buttons activates or deactivates the sound on that
corresponding beat. In the example shown above, these are steps 1, 4, 8, 9, and 14.
Note: Dragging the mouse across the buttons allows corresponding triggers to be
quickly turned on or off.
Trigger Context Menu
Control-clicking on one of the Trigger buttons opens the context sensitive Trigger
menu, which offers the following options:
• Alter: Randomly reorders the sequencer steps while retaining the number of
activated triggers.
• Randomize: Randomly reorders the sequencer steps; the trigger row is also altered
by randomly erased and added events. In other words, a brand new sequence is
created.
• Shift Left: Shifts the sequencer data one step to the left.
• Shift Right: Shifts the sequencer data one step to the right.
• Copy: Copies all activated triggers to the Clipboard.
• Paste: Pastes all activated triggers from the Clipboard.
• Clear: Turns off all activated triggers.

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Swing Enable
Activation of the blue Swing button to the left of the trigger row stipulates that the grid
of the currently selected sound will be played in accordance with the Swing knob
setting.

Only even-numbered steps are affected by the Swing parameter; exactly which beats
this corresponds to depends on the selected Resolution parameter setting, as
demonstrated by the following example.
At a Resolution of 1/8 and a Length of 8, the notes on steps 1, 3, 5, and 7 represent
quarter notes in the measure. These remain unchanged. Only the eighth notes found
between them (steps 2, 4, and so on) are shifted by the Swing function. The amount of
shift is equal to the swing intensity (see “Swing” on page 334).
Note: Swing is only active for grid resolutions of 1/8 and 1/16.
Velocity/Gate Row (The Row for Velocity and Note Length)
In this row, you set the length (Gate Time) and the velocity of the notes entered in the
trigger row. Both parameters are displayed as a single graphical bar. The bar’s height
represents the velocity, it’s length depicts the note length (Gate Time).

Click-dragging on the desired bar allows you to change the length and velocity values
for each step.
Note: The Gate time is divided into four equal sections, making it easy to set
rhythmically accurate note lengths. In order for the “one shot” envelope to react to gate
time, it is necessary to either; activate the Gate function in the sound itself (see “Trigger
and Group Menus” on page 324) or use envelopes in sustain mode (see “Sustain” on
page 332), in conjunction with rhythmically useful (short) decay times.
Reset
The Reset button located to the left of the Velocity/Gate row returns all velocity and
gate values to their default settings.
Velocity/Gate Context Menu
Control-clicking on steps in the Velocity/Gate row opens a context-sensitive menu
which offers the following options:
• Alter Vel(ocities): Randomly changes the velocity values of all steps while retaining
the selected beats (the trigger row remains unchanged).

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• Alter Gate (Time): Randomly changes the note lengths of all steps while retaining the

selected beats (the trigger row remains unchanged).
• Randomize Vel(ocities): Same as Alter Velocities, but random parameter alteration is

more pronounced.
• Randomize Gate (Time): Same as Alter Gate, but random parameter alteration is more

pronounced.

Using MIDI to Control the Sequencer

As mentioned earlier, pattern performance can be influenced by incoming MIDI notes.
This allows you to spontaneously interact with the step sequencer, making Ultrabeat an
excellent live performance instrument. The manner in which Ultrabeat reacts to MIDI
control is determined by the Pattern and Mute Mode buttons and the Playback menu.
Pattern Mode
The On setting allows patterns to be switched or started via incoming MIDI note on/off
commands.

In this mode, the Transport button turns blue to indicate that it is ready to receive
incoming control commands.
MIDI notes C-1 to B0 switch between patterns: C-1 selects pattern 1, C#-1 pattern 2
and so on up to pattern 24, selected when MIDI note B0 is received.
Playback Mode
Pattern reactions to incoming MIDI notes is set in the Playback Mode menu.

The menu is opened by clicking on the field to the right of the Pattern Mode label on
the bottom-right edge of the Plug-in window. You will find the following options here:
• One Shot Trigger—The reception of a MIDI note starts the pattern, which plays once
through its cycle, then stops. If the next note is received before the pattern has
reached its final step, the new note stops playback of the first pattern and the next
pattern begins playing immediately (this can be a different pattern or the same
pattern, depending on the MIDI note received).

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• Sustain—The reception of a MIDI note starts the pattern and it continues playing in

an infinite loop until the corresponding MIDI note is released.
• Toggle—The reception of a MIDI note starts the pattern and it continues playing in

an infinite loop until the next note is received. If it is the same note, the pattern stops
immediately. If it is a different note, the sequencer immediately switches to the new
pattern.
Note: In Toggle mode, you can confidently switch between patterns in the middle of a
bar—the sequencer will stay in time and will automatically jump to the corresponding
beat of the new pattern. This isn’t the case in One Shot Trigger mode, however, because
the moment you switch in this mode, the new pattern is started from the beginning.
• Toggle On 1—The behavior is as per Toggled mode except that the pattern change or

stop occurs the next time beat 1 is reached—at the beginning of the next pattern
cycle.
Voice Mute Mode

• Off—This is the default setting. It means nothing more than the fact that Ultrabeat’s

sounds can be triggered by MIDI notes. The sounds can be played from a MIDI
keyboard, starting with note number C1, upwards.
• On—Playing a MIDI note starting at C1 and upwards mutes the corresponding sound
in Ultrabeat’s mixer. A subsequent MIDI note of the same pitch un-mutes it.
This setting is ideal for spontaneous arranging of pre-programmed patterns, and
muting single elements of a pattern without deleting them. This is especially useful in a
live performance—but not only there. Triggering the step sequencer via MIDI notes
opens up a number of remixing possibilities. Ultimately, all of the creative pattern
switching options discussed in this section is achieved through the use of MIDI note
messages and can therefore be simply recorded, edited, arranged, and automated in
Logic.

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Importing Sounds
Ultrabeat allows you to import individual sounds.
Please proceed as follows:
1 Click on the Import field to the right of the Voice Auto Select function.

2 Navigate the dialog box as you would normally until you find the setting that you want
to import sounds from.
3 After selecting the setting, a list of all the sounds found in this setting will open up in a
“drawer” next to the mixer section.

4 Select a sound in the list and click on its name while holding down the Control key.
5 A context menu opens. Use the Copy command to copy the selected sound to the
Clipboard.
6 Now select the sound you want to replace in the current drum kit in the assignment
section to the left.
7 Click on its name while holding down the Control key. A context menu opens. Use the
Paste command to replace the sound you selected (also see “Naming and Organizing
the Sounds” on page 304). This facility allows you to create a new drum kit that
contains sounds from several other drum kits.
Note: In the sound import process described above, existing step sequencer data is
neither copied nor replaced.
Previewing Sounds
Before importing a sound, you can preview it by clicking on its name in the import list.

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Tutorial: Creating Drum Sounds in Ultrabeat
Now that you’re acquainted with all of Ultrabeat’s features, we’d like to offer you a few
specific sound creation tips in the following section. Please take the time to explore the
vast and complex possibilties available to you in Ultrabeat, using the following
programing tips as a starting point. You’ll discover that there is hardly a category of
electronic drum sound that Ultrabeat can’t create easily.
Before we jump into sound programming, we’ll briefly discuss how sounds are selected.

To select sounds, proceed as follows:
The 25 sounds of an Ultrabeat drum kit are mapped to the onscreen keyboard found
on the left hand side of the Plug-in window. The order of sounds on the keyboard
corresponds to notes on a connected MIDI keyboard, starting with C1 for the first
(bottom) sound. Clicking on the name of a sound selects it for editing. The sound’s
parameters are displayed to the right of the keyboard, and can be edited
Note: Make sure that the sound you’re playing via MIDI is also the one you’re
editing: you can recognize the selected sound by the frame that appears around its
name in assignment area. The corresponding key on the onscreen keyboard to the left
of the sound name turns blue when it receives appropriate MIDI information. Clicking
on these keys with the mouse will play the sounds directly in the Plug-in window.

In this example, drum voice 2 is being played (the blue key) while drum voice 4 is
selected (the red frame).
When the Voice Auto Select function is switched on, the sound played via MIDI is
selected for editing.
Time to move on to sound programming! First, we’ll analyze several classic electronic
drum sounds, and show you how you to accurately recreate them using the many
features available to you in Ultrabeat. We’ll also take a look at additional refinements
you can make.
Note: In Ultrabeat’s Settings folder, you will find a drum kit called Tutorial Kit. This drum
kit contains all drum sounds discussed in this tutorial. It also includes a drum sound
called “Default Tut(orial)”, which is a default set of “neutral” parameters that provide an
excellent starting point for many of the following examples.

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Kick Drum
Electronically produced kick drum sounds are primarily based on the sound of a deeply
tuned sine wave.
To program this type of sound in Ultrabeat, please proceed as follows:
1 Load the Default Tutorial setting. Note that Oscillator 1 is in Phase Oscillator mode.
2 Find a suitably tuned pitch in the lower octaves by soloing the bass drum along with
other important tonal elements of the song (a bass or pad sound, for example). Use the
Osc 1 Pitch slider to adjust the pitch until appropriate.
3 Use Env 4 to shape the volume of the bass drum. For slower beats you’ll want a longer
Decay phase, while at faster tempos you’ll choose a shorter Decay time. The Attack time
of Env 4 should be very short in any case (zero, in most cases) or the sound will lose its
percussiveness, and its ability to be clearly heard in the mix.
Our kick drum still sounds very soft and is somewhat reminiscent of the famous TR 808
bass drum. It’s still missing a clearly defined attack. In order to give the bass drum more
“kick”, we’ll control the pitch of the oscillator with an envelope.
To accomplish this, please proceed as follows:
1 Ensure that Env 1 is chosen in the Mod menu of Oscillator 1’s Pitch parameter.
2 Set the degree of modulation by moving the blue Mod slider approximately 3–4
octaves above the original pitch.

3 Set the Attack time in Env 1 to zero by sliding the leftmost of the two junction points
that sit on the x-axis all the way to the left.

4 Now experiment with the Decay time by moving the rightmost of the two junction
points that sit on the x-axis; you’ll discover that higher Decay values (shifting the Bezier
handle to the right) result in sounds similar to synth toms, while shorter Decay values
(shifting to the left) provide the “kick” character that we’re after.

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5 Change the Mod amount (the blue control) of Osc 1 Pitch again (see step 1). The
interaction of this parameter with the envelope’s Decay time provides numerous
possibilties for shaping the “kick” or “punch” of the bass drum sound.
Note: This simple bass drum sound is called “Kick 1” in the tutorial set, at a pitch of C1.
Removing Tonality
One advantage of bass drums based on sine waves is that their sound can be precisely
tuned to match the song. The disadvantage: a recognizable pitch is not always
desirable. Ultrabeat offers several methods to reduce the tonality of the sound. A very
effective tool is the 2 Band EQ.
Try the following setting:
1 Band 1 is used in shelving mode with negative Gain, high Q value and a frequency of
about 80 Hz.
2 For band 2, choose the Peak mode at a frequency of around 180 Hz, a medium Q value
and also a negative gain value.
3 On the EQ graph, you can see how the frequencies around 80 Hz are boosted, while
the surrounding frequencies are reduced.

4 Vary the frequency of Band 2 now (easily recognizable in the blue part of the EQ graph)
and you can influence the extent of bass drum tonality.
A further method for reducing the tonality of a drum sound that is rich with overtones
is to use a lowpass filter. Control the cutoff frequency of the filter with an envelope.
Please proceed as follows:
1 Reload the Default Tutorial sound, choose A#0 as the basic pitch in Oscillator 1 and
modulate it (as shown in the example on page 342) using Env 1.
2 An increase in the Saturation parameter value enhances the overtones of the drum
sound.
3 Note, that the output of Osc 1 is directed to the filter, as the Filter Bypass button (the
arrow between Osc 1 and the filter) is activated.

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4 Make the settings shown in the following graphic to the filter section: Filter type = LP
24, Cutoff value = 0.10, Mod Source for Cut = Env 3, Mod Amount for Cut = 0.60,
Resonance = 0.30.

5 Set the Attack time of Env 3 to zero. Use the Decay time of Env 3 to shape the sound of
the filtered bass drum.
6 You may also choose to control the filter resonance with an envelope. Make sure you
dedicate a single envelope to this function (in this case, use Env 2 as a Mod source for
Res). Choose a Mod amount for Res of about 0.80. Select a longer Decay time in Env 2
than in Env 3 and listen carefully to the fatter and more atonal bass drum sound
achieved through this Res modulation (due to the higher filter resonance).
Note: The bass drum described in the above example is called “Kick 2” in the tutorial
set, at a pitch of C#1. It also features an interesting EQ setting (see paragraph below).
More Bass…
Use the “Kick 2” filtered bass drum sound as a starting point, and try out the remaining
parameters in the Phase Oscillator. You will discover that high saturation values make
the sound rounder and add more bass, for example. The character of our example is
beginning to head in the direction of a TR-909.
More Kick…
To get even closer to the 909, we recommend an EQ setting as shown in the following
graphic. Note the low frequency pressure point around 60 Hz (which can be seen in the
red area on the EQ graph) as well as the assertive punch or kick (the blue area starting
at 460 Hz and up) of a 909 bass drum are strengthened. (This EQ setting is already part
of the “Kick 2” setting.)

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More Contour…
In our example, all four envelopes are being used. Take some time to play with the
shapes of the envelopes, while maintaining the Attack and Decay settings. Experiment
with the junction points of the Decay phase in the different envelopes to familiarize
yourself with the sound shaping options available. Start with the Decay phase of Env 4
(which controls the volume of Oscillator 1 as well as filter resonance) and observe how
reshaping the “belly” of the envelope can change the character of the sound from crisp
and short to round and voluminous.
The Ultrabeat Kick
Let’s move on to bass drums that are uniquely “Ultrabeat”, rather than analog models.
Try modulating pitch with an LFO, rather than an envelope, for example.
An appropriate LFO setting could look something like this:
1 Start with the Default Tutorial sound at a pitch of A#0 (Osc 1 Pitch), and choose LFO 1 as
the Mod source in the Osc 1 Pitch section.
2 Set the degree of modulation by moving the blue Mod control to a value of A3.
3 LFO 1 should be set to a low number of Cycles (25 to 35), a high Rate (starting with
70 Hz and higher) and a medium value for Decay (Ramp rotary knob at about −190).
4 Experiment with the LFO waveform and you’ll discover that you can attain different
nuances in the character of the bass drum attack.
5 Now modulate the Asym (Asymmetry) parameter with the same LFO, and also vary the
Slope and Saturation values. This method enables you to create very different bass
drum sounds with a single oscillator, one LFO and one envelope (for volume). The
character of the sounds can range from soft to punchy and the degree of tonality in
the sound can be adjusted to taste.
Note: The bass drum sound described is called “Kick 3” in the tutorial set at a pitch of
D1.
Use the second oscillator (with similar settings or with a sample) or use the filter and
the ring modulator—the sky’s the limit as far as your imagination is concerned, so get
on with it, and create that next “gotta have it” drum sound.
Note: You can find an “emulation” of the legendary 808 bass drum under the name
“Kick 4” in the tutorial set, at a pitch of D#1.

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Snare Drum
The sound of an acoustic snare drum primarily consists of two sound components: the
sound of the drum itself and the buzzing of the snare springs. Try to approximate this
combination in Ultrabeat with a single oscillator and the noise generator.
Start programming your snare drum as follows:
1 Begin again with the Default Tutorial setting. Deactivate Oscillator 1, and switch
Oscillator 2 on (in Phase Oscillator mode).
2 To get rid of the sine wave (which is not especially desirable for a snare sound, in
contrast to the bass drum), modulate Osc 2 Pitch with a rapidly vibrating LFO with a
medium Ramp Decay value. To accomplish this, select LFO 1 in the Mod menu of Osc 2
Pitch. The pitch value for Osc 2 Pitch should be around G#2 and the Mod amount (the
blue Mod control) should be about 3–4 octaves higher.
3 LFO 1 should be set to a high Rate. Choose a value of 20 for Cycles and −20 for Ramp.
The LFO waveform parameter should be set to a value of about 0.58, which is a square
wave.
4 Use Env 1 to control the volume of Oscillator 2 by setting Vol to the lowest possible
value (−60 dB), selecting Env 1 in the Mod menu and adjusting the modulation intensity
to a point just below its maximum value.
The GUI detail shows the settings of Oscillator 2 and Env 1 described in steps 2 and 3.

5 Experiment with different Slope and Asym values to impart a more or less “electronic”
character on the sound.
6 Now turn on the noise generator and control its volume with the same quick envelope
used in Osc 2 Volume.
Use the filter parameters of the noise generator to either roughen up, refine or add
bright frequencies to the noise component of the snare drum sound. Select a LP filter
type, and try a filter frequency between 0.60 and 0.90. Modulate it with LFO 1 that
you’re already using to control the pitch of Oscillator 2.
Note: The snare drum sound is called “Snare 1” in the Tutorial Kit, at a pitch of E1.

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Now refine the snare drum sound using FM synthesis:
1 Turn on Oscillator 1 in FM mode. Use Env 1 to control the volume of Osc 1 as well.
2 Choose a pitch for Oscillator 1 that’s about an octave lower than Oscillator 2.
Consciously avoid even intervals between the oscillators and detune them slightly from
each other. As an example, try a pitch setting of F#2 in Osc 2 and E1 in Osc 1, then fine
tune Osc 1 a few cents higher by holding Shift while adjusting its Pitch slider.
3 Now experiment with FM Amount, and add more tone (low FM Amount value) or noise
(more FM Amount) as desired. Also try modulating the FM Amount with a fast LFO.
Note: An exemplary snare drum sound that makes use of FM can be found in the
tutorial set at a pitch of F1. It is called “Snare 2”.
Higher FM Amount values lead to considerably more overtones and a very electronic
sound character. If you want to make the sound more acoustic, we recommend feeding
Oscillator 1 (and possibly Oscillator 2 as well) into the main filter. Use these settings to
start: LP 24 mode, a Cutoff value of about 0.60.
The 808 Snare
The famous 808 snare is based on two resonating filters and a noise generator, fed
through a highpass filter. The mix ratio of the two filters and the volume of the noise
generator can be adjusted. This structure cannot be 100% replicated in Ultrabeat.
To “clone” the 808 snare sound, proceed as follows:
1 Begin again with the Default Tutorial setting.
2 Both of the resonating filters of the 808 snare can be replicated by two cleverly
programmed Phase Oscillators. They are given slightly different Slope values and are
detuned by almost an octave. The tonal relationship between the oscillators should
also be uneven here—from E3 to F2, for example.
3 The volume of each oscillator should be controlled by a different envelope. The
envelope for the lower tuned oscillator should have a longer decay time than the very
“snappy” envelope setting for the higher oscillator.
4 Feed the output of both oscillators into Ultrabeat’s main filter, and “hollow out” the
sound with a highpass filter. Activate the Filter Bypass button in both oscillators (the
arrow between the oscillator and the filter). Choose the HP 12 setting in the filter, a
Cutoff value around 0.40 and a Resonance value of about 0.70.
You have just very cleverly emulated both of the 808’s resonating filters. Shifting the
pitch of both oscillators simulates the behavior of the 808’s Tone control by the way.

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To complete the 808 emulation, add some noise:
1 Switch the noise generator on, and activate the highpass mode in its filter (HP). Set the
Cutoff value to about 0.65, Resonance to 0.35 and add a little Dirt (around 0.06).
2 The noise generator provides the sustained snare sound. It should be shaped by its
own envelope, independent of the decay phases of both oscillators, in order to get
808-like results. Changing the volume of the noise generator simulates the “snap”
parameter of the 808.
Note: The 808 snare drum described is called “Snare 3-808” in the Tutorial Kit, at a pitch
of F#1. It also features an interesting EQ setting.
Dynamics through Velocity
Using the 808 snare drum sounds in our tutorial kit, we’d like to explore the possibilities
Ultrabeat offers for implementing velocity.
To implement velocity, proceed as follows:
1 Select the sound “Snare 3-808”.
2 Click on the word Off below the Volume knob in Oscillator 1.

3 In the Via menu that appears, choose Vel.
4 A slider appears on the ring around the knob. Move it in a clockwise direction. When
you grab the marking, a help tag displays the value. Set it to −0 dB.

5 Repeat steps 1–4 in both Oscillator 2 and the noise generator. You can now dynamically
play the sound using velocity.

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You can increase the performance dynamics with the following steps:
1 First, reduce the values of the individual volumes by turning down the Volume knobs in
both oscillators and the noise generator. Note how the Mod ring and its Via sliders also
move back. Change the Via slider positions until all three Volume knobs look like this:

2 If you use differing intensities for each Volume knob when completing step1, you’ll
have the potential of individual velocity reactions for each sound component.
3 The dynamics of the sound as a whole can be further increased by the following
setting to the Voice Volume knob:

You now have an 808 snare that is exceptionally responsive to velocity. As you may
know, this wasn’t possible with the original—not even an 808 sample could offer the
dynamic volume control of individual sound components that’s demonstrated here. A
sample only offers you the whole sound, not it’s constituent parts.
In the next step, you’ll use velocity to control the character of the sound—individually
for each component—plus volume, of course:
4 In the Saturation Mod menu of Oscillator 2, choose Max and then velocity in the
corresponding Via menu.

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5 Set the additional control that appears (as shown), to control the character of the
sound with velocity:

6 Repeat this with the other parameters of Oscillator 2, as well as pitch:

7 Within the noise generator, you’ll be working with negative modulation (the position of
the marking is below that of the base parameter value), and will modulate one
parameter directly (Max setting in the Cut Mod menu), and another indirectly (the LFO
2 setting in the Dirt Mod menu):

Our sound is now nothing like an 808 snare—and that’s exactly what we wanted to
achieve. Keep experimenting with velocity and figure out when it makes sense to use it
as a direct or indirect modulation source, in either its positive or negative form.
Note: The dynamic 808 snare drum described above is called “Snare 4 - vel” in the
tutorial set, at a pitch of G1.

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The Kraftwerk Snare
A further classic electronic snare drum sound is the highly resonant lowpass filter of an
analog synthesizer that quickly closes with a “snap”. This sound was used extensively by
Kraftwerk.
Proceed as follows to recreate this sound with Ultrabeat:
1 Choose the “Snare 1” sound to start with.
2 Direct the signals of both oscillators and the noise generator to the main filter.
3 Modulate Cutoff with Env 1 (this is already modulating the volume of the noise
generator).
4 Modulate the filter resonance with Env 2.
5 Now begin to experiment with the parameters described in steps 1–5 (especially the
envelopes), introduce EQ into the sound, and discover how much “playing room” these
basic settings allow you.
Note: An exemplary sound called “Snare 5 - KW” is in the tutorial set, at a pitch of G#1.
Analyze, and compare it to your own creation.

Toms and Tonal Percussion
Tonal percussion sounds such as toms or congas are relatively easy to emulate
electronically with sine or triangular wave oscillators. Ultrabeat’s Phase Oscillator offers
you a broad spectrum of suitable basic sounds with which to start. Control the pitch of
the oscillators with envelopes, and use the programming techniques discussed in the
bass and snare drum sections to alter tonality. You should find it easy to create a broad
range of toms and similar sounds.
Note: At the pitches A1 to B0 in the tutorial set you’ll find typical 808 toms. Analyze
these sounds and modify them as you see fit.
At this point, spend some time experimenting with the Model mode of Oscillator 2. Try
to familiarize yourself with the effect of each parameter, and create some of your own
tonal percussion sounds, ranging from small tabla drums to glass bowls.
Note: The “Tabla” and “Glass” sounds (at pitches C2 and C#2) of the tutorial set combine
both Osc 2 Model and FM. They are also good examples of the complex use of velocity
as a modulation source.

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Hi-Hats and Cymbals
Electronic hi-hat sounds are very easy to create in Ultrabeat.
Please proceed as follows:
1 Load the Default Tutorial sound.
2 Switch off Oscillator 1 and turn on the noise generator. Choose the following settings
for the noise generator:

3 In the GUI detail above, you can see, that the Cutoff parameter is modulated by Env 1.
The modulation is negative, the position of the Mod slider is below that of the base
parameter value.
4 Use rather short Decay values for Env 1 and Env 4 ().
5 Set the Attack time of Env 4 to a value of 0. The Attack time of Env 1 should also be
rather short, but not equal to zero.
Note: You’ll find a similarly constructed sound called “HiHat 1” at a pitch of F2 in the
tutorial set. Also analyze the hi-hat sound Hihat 2 at a pitch F#2.
It’s not far from the hi-hat to the crash cymbal: the main difference between a hihat
and crash cymbal sound is the length of the decay time. Correct assignment of the
envelopes is the key to producing different cymbal sounds.
Select the “Cym 1” and “Cym 2” sounds in the tutorial kit and try different envelope
assignments and settings for Cutoff and Volume in the noise generator, Cutoff and
Volume in the main filter and so on.

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Metallic Sounds
If you want to create metallic sounds with Ultrabeat, the ring modulator and the Model
oscillator are the ideal tools.
To use the ring modulator, proceed as follows:
1 Load the Default Tutorial sound.
2 Activate a Phase Oscillator and the Model oscillator. Choose a pitch for each oscillator
above C3 so that a slightly detuned interval is created.
3 In the Material Pad of the Model oscillator, choose a setting with plenty of overtones as
in the graphic below.

4 Set the volume of each oscillator to a value of −60 dB and turn the ring modulator on
by clicking on its name.
You’ve just created a bell-like sound that you can filter (with a high resonance value) if
required.
Note: You can find a similar sound called “Ring Bell” at a pitch of A2 in the tutorial set.

Clicks and Cuts
Ultrabeat features extremely fast envelopes and uncommonly powerful LFOs. Use these
modulation sources to perform extreme modulations of the oscillator and filter
parameters. The key to creating “out of the ordinary” sounds is to try modulating as
many targets as possible, and not to be afraid of using extreme settings: use a quick
envelope to drive the filter to self oscillation for a fraction of a second, use a few LFO
cycles at a much higher rate than other cycles, or experiment with the Dirt parameter
or the bitcrusher.

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Programming in Building Blocks
As you become familiar with drum sound programming, you may begin thinking in
“building blocks”. By this we mean that you might realize that drum sounds usually
consist of different components. Once you’ve mentally, or physically, written down your
“list” of components, you should try to emulate each component that contributes to the
sound’s character—making use of the different sound generators available in Ultrabeat.
Assigning dedicated (amplitude) envelopes to the different components allows you to
control their temporal behaviour individually. As an example: You can emulate the
body of a drum with Oscillator 1, the sound of the stick hitting the skin (or first
transient) with the noise source, additional overtones and harmonics can be provided
by Oscillator 2 and/or the ring modulator. Once you begin thinking that drum sounds
consist of several building blocks or layers, the design of the Volume controls in the
individual sound generators might make more sense to you, as this is the place where
the blocks are combined, balanced, and controlled.
We hope to have given you a few insights into the interplay of Ultrabeats’s functions
and parameters. We also hope that you’ve gained some inspiration and insights for
programming your own sounds. We would also ask that you further explore all of
Ultrabeat’s functions at your leisure. The included settings will afford you some
interesting insights and hopefully, further stimulation. Spend some time analyzing
Ultrabeat’s presets and ask yourself “How did they do that?”—then figure it out.
Have fun creating your own Ultrabeat sounds and sequences!

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Sculpture

Sculpture is a synthesizer plug-in that generates its sound
based on a simulated vibrating string or bar.
To keep things clear, we will always refer to the “String” throughout this chapter, even
though many of the possible sounds that you can create with Sculpture have nothing
in common with what you’d expect from a stringed instrument!

The Graphical User Interface (GUI) of Sculpture is broken down into three main areas.
The silver section at the top contains the sound engine. The blue/gray area below
houses the polyphonic modulation sources and Envelopes. Global control sources,
incorporating the Morph Pad, are located on the bottom, gray ledge.
Most software synthesizers emulate real-world hardware, and generate sound based on
samples or emulations of the signals generated by analog oscillators.

355

Sculpture uses a method of synthesis called component modelling. This approach to
tone generation shares some aspects and parameters with other synthesis techniques,
such as those found in additive and subtractive synthesizers. As such, many of the
parameters used by Sculpture will be immediately familiar to you, such as LFO’s,
Vibrato, Envelopes, and so on. Many others, however, will be very new.
Fortunately, the GUI of Sculpture is incredibly elegant and intuitive to use, so you
should find yourself creating amazing sounds in next to no time.
Sculpture’s recordable envelopes provide massive scope for sonic animation, with the
payoff being some just plain astonishing sounds! Sculpture’s flexibility in this area
provides a new level of control that we’re sure will amaze and delight you.
We encourage you to make full use of every control and parameter that is available to
you—both when initially auditioning some of the supplied factory sounds, and when
creating new ones of your own. Don’t be afraid to experiment with sound design—
that’s what Sculpture was created to do!
Before taking a look at the product features, we’d like to briefly touch on the synthesis
core of Sculpture. We will also take a look at the creation of particular types of sounds
in “Programming: Quick Start Guide” on page 409.

The Synthesis Core of Sculpture
We have included this section at the top of this chapter to give you a feel for the way
Sculpture works. Component modelling, as you’ll discover, is quite different to
“traditional” synthesis methods—and so are the results!
We have followed the signal path (shown in the diagram) of the core synthesis engine
in the layout of the manual’s parameter descriptions. We invite you to check out each
parameter’s options as you read about them. This will give you a better feel for where
things are, and what’s available. So let’s get to it …
Objects ↔ Strings

Pickups

Stereo
Delay

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Amplitude
Envelope

Wave
Shaper

Filter

Body EQ

Level

Limiter

The String is the central synthesis element. It offers a range of parameters that allow
you to adjust its material—what it’s made of, in other words.
Up to three Objects of different types are used to excite or disturb the vibration of the
String. These Objects can be positioned anywhere along the String, and offer multiple
parameters for adjustments to their properties.
The String itself doesn’t make a sound unless it is stimulated by the Objects.
The vibration of the String is passed on to the ensuing processing section—via two
movable Pickups (you can view these as being similar to the electromagnetic pickups
found on guitars or electric pianos and clavinets).
The processing section consists of the ADSR-equipped Amplitude stage, a Waveshaper
(with selectable types) and a multi-mode Filter.
Note: All elements described above exist on a per voice basis.
The sum of all voice signals is then processed by an integrated Stereo Delay effect.
From there, the signal is sent to an EQ-like module (Body EQ), which globally simulates
the spectral shape/body response of your instrument. There are several body types to
choose from.
The resulting signal is then fed to a Level Limiter section.
A vast number of modulation options are also available, from tempo-synced LFOs to
Jitter generators and newly-developed recordable Envelopes.
A (recordable) Morph function also allows for smooth or abrupt transitions between (up
to) five Morph snapshots.

A String as a Synthesis Element
As you can see, Sculpture, as a component modelling synthesizer, provides you with a
String—rather than the oscillators found in traditional synthesizers.
The String is considerably more sophisticated in concept than simple oscillators.
Basically, you are actually creating the waveform, or base timbre from nothing. This is
achieved by (mathematically) describing the string’s properties, and that of its
environment. These include, amongst others: the material the string is made of, the
thickness and tension of the string, its characteristics over time, the atmosphere (water,
air, and so on) it is being played in, and the way it is being played—struck, bowed, and
so on.
In Sculpture, the general timbre/sound sculpting is set by the String parameters.
Modules such as the Waveshaper, Filter, and Body EQ are useful additions.

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Sculpture goes far beyond the mere creation of an infinite number of base timbres,
however. One of the key differences between Sculpture’s String and a traditional
synthesizer’s waveform is that the base timbre (provided by the String) is in a constant
state of flux.
By this, we mean that if Sculpture’s String is still vibrating for a specific note,
retriggering that same note will interact with the ongoing vibration. This is not
dissimilar to the effect of repeated plucking of a guitar string, where the string is still
vibrating when the next note is played. This will alter the harmonic spectrum each
time—which is why acoustic guitars sound organic when a note is played repeatedly,
and sampled guitars don’t.
As you can see, this is quite different to other synthesis methods where the base timbre
(waveform), even if modulated, does not harmonically interact when retriggered. What
usually happens in traditional synthesizers is that the waveform is restarted—from mid
cycle, or from the beginning—with the result being an increase in volume, or a slight
cyclical wave shift.
Beyond this aspect of base timbre control and interaction, Sculpture also provides you
with the means to model (emulate) a number of other properties of acoustic
instruments. Among these, you’ll be able to emulate a bow dragging across a string, a
piano hammer strike, or even dropping a coin onto the bridge of a guitar.
There would be no point in bowing or plucking a string, however, if you didn’t have a
neck or a body to your violin, cello or guitar. You’d probably want some way to change
the tension of the strings, or try steel, rather than nylon or catgut. In other words, to
control the physical properties of the string, and its excitation.
All of this is possible with Sculpture.
It is an instrument that will require some investment of your time, but it will reward you
with beautifully warm organic sounds, evolving soundscapes or a harsh and metallic
“Hell’s Bells” patch—if you’re after that sort of thing.
Think of it as your own personal sound design and sculpting toolkit.

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Sculpture’s Parameters

The following sections discuss the parameters of Sculpture.
Before touching on them, however, a key factor with component modelling is the
interaction between various sections of the synthesis engine. This can lead to some
truly unique sounds, but can also lead to unexpected results.
Sculpture is very different to traditional synthesizers, and requires a more measured
approach to achieve a particular end result. We have compiled a collection of tips in
“Programming: Quick Start Guide” on page 409, that will assist you in the creation of
particular types of sounds. Please ensure that you read this section.
Note: Keep the Sculpture flowchart diagram (on page 356) handy while familiarizing
yourself with the interface/programming. If your approach is methodical—and you
follow the flowchart, you shouldn’t encounter too many surprise results.

Parameter Use
Some additional parameters are accessible via Control-clicking (or clicking with the
right mouse button, if you have a two button mouse) on particular areas of Sculpture’s
front panel. This will launch a contextual menu, where you can make your choice by
highlighting the desired entry and releasing the mouse button.
Sculpture’s envelopes (and other fields, such as the Material or Morph Pads) can be
adjusted graphically, by click-holding on the nodes (shown as hollow or filled dots,
balls or diamonds), and dragging to the desired position in the respective panels on
Sculpture’s interface.

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Most parameters can be reset to their default value by clicking on the desired control,
while holding down Option.
Note: When adjusting most parameters, keep an eye on the small help tag that pops
up while the mouse button is depressed. This will provide you with an accurate
indication of the current parameter value, allowing you to make precise changes.

Global Parameters
These are found across the top of the Sculpture GUI, unless otherwise specified.
Transpose
Transpose is used for coarse tuning of the entire instrument (range: ±2 octaves). Given
the ability of component modelling to radically alter pitch with certain settings, coarse
tuning is limited to octave increments.

Tune
Tune is used for fine tuning of the entire instrument (range: ±50 cents). A cent is
1/100th of a semitone.

Warmth
Warmth is used to slightly detune the different voices, much like the random
fluctuations caused by the components and circuitry of analog synthesizers. As the
parameter name suggests, this “warms up” or “thickens” the sound.

Glide Time
Values above 0.0 lead to a softer pitch change between played notes (range: 0.00 to
5000 milliseconds).

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Voices
When Keyboard Mode is set to Poly, this parameter limits the number of simultaneously
sounding voices to the set value. A value of 16 voices is the maximum polyphony of
Sculpture.

Keyboard Mode
Here, you can select between the Mono, Legato or Poly Keyboard Modes.

You can switch between mono and poly modes by clicking on the Poly and Mono
buttons. The Portamento, the duration of which is set by the Glide Time parameter,
affects legato performances. As long as you don’t release the key of the previously
struck note before pressing a new key, you’ll hear the glide or portamento. If you
release every note before striking another key (portato or even staccato style), the glide
effect will not be audible.
When the Legato button is activated, a further side-effect occurs. The Amplitude
Envelope is not retriggered—the sound is sustained and the Attack phase of newly
played notes is not retriggered. These behaviors are sometimes referred to as Single
Trigger and Multi Trigger modes.
All modes simply retrigger a (potentially sounding) voice with the same pitch, instead
of allocating a new one. As such, multiple triggering of a given note results in slight
timbral variations, depending on the current “state” of the model at note-on time.
If Sculpture’s String is still vibrating for a specific note, retriggering that same note will
interact with the ongoing vibration, or current “state” of the string.
Important: A “true” retrigger of the vibrating string will only happen if both Attack
sliders of the Amplitude Envelope are set to zero. If either slider is set to any other value,
a new voice will be allocated with each retriggered note.

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Bender Range Up/Down
These parameters are found below Object 3, on the left-hand side of the Sculpture GUI.

Separate settings are available for upwards and downwards pitch bends—using your
MIDI keyboard’s pitch bend controller.
When Bender Range Down is set to Linked, the Bender Range Up value is used for both
(up/down) directions.
The range is set in semitone increments: +12 and−36 respectively.
Note: Bending the string will, just like the real deal on a guitar, alter the shape of the
model, rather than merely act as a simple pitch bend.

String and Object Parameters
The string and object parameters discussed in this section apply on a per-voice basis.
You will note a number of parameter names followed by “(morphable)”. This indicates
that the parameters can be morphed between up to five snapshots, called Morph
points. More detail on morphing can be found in “Morph” on page 400.
Hide/Keyscale/Release Buttons
Before beginning, we’d like to briefly cover some common parameters that you’ll
encounter when manipulating the String. The String’s default note position is C3
(middle C).

Three buttons are available for your use that will activate/deactivate and hide the
various Keyscaling and Release parameters. Simply click on the Keyscale, Release or Hide
button, dependent on the adjustments you would like to make. The corresponding
parameters will become visible in the ring surrounding the Material Pad.
Keyscale parameters can be set for notes that fall below C3, or notes that are positioned
above it. In simple terms, the impact of these parameters can be controlled across the
keyboard range. As an example: a parameter such as string stiffness (which we’ll
discuss shortly) could be more intense for high notes, and less intense for low notes. In
practical terms, this would result in more harmonic (“sweeter”) sounding bass notes,
and inharmonic overtones in treble notes (notes above C3).

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The string Release parameters impact on the vibrations of the string once the key has
been released.
The Hide button is handy for avoiding accidental parameter changes, and simplifies the
interface.

The Material Pad
The following two string material parameters determine the general timbre, and are
controlled by the “ball” (which correlates to the X and Y co-ordinates) within the
Material Pad.

The “crosshair” is a handle for the Key Scale/Release Scale diamonds in cases where
these are hidden by the ball. It also allows you to independently change the keyscaling
for one of the two axes (X/Y positions—Inner Loss or Stiffness).
In general synthesizer terms, this could be viewed as a base timbre generator in the
Oscillator section.
Note: Option-clicking resets all string parameters to their default values.
InnerLoss (morphable)
This parameter is used for damping of the string, as caused by the string material—
steel, glass, nylon, or wood. These are frequency-dependent losses which cause the
sound to become more mellow during the decay phase. Simply click-hold, and drag,
the Inner Loss and Stiffness “ball” to the desired position.
Inner Loss Scale Low/High
Allows independent adjustments to the key tracking of inner “losses” for notes above
and below C3.
To adjust Inner Loss keyscaling, first select the Keyscale button, then click-hold on the
green horizontal line for low notes or the blue horizontal line for high notes, and drag
up/down to the desired position.
A diamond indicates the intersection between the Inner Loss and Stiffness Low/High
Scaling positions. You can click-hold and drag this diamond directly, to adjust both
parameters simultaneously.

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Inner Loss Scale Release
Values above 1.0 cause the inner losses to increase when the key is released. This is
quite unnatural, as this would mean that the string material would change after the
note was released.
In practice, however, the use of this parameter in combination with Media Loss Scale
Release allows a natural simulation of strings that are dampened at note-off time.
To adjust, first select the Release button, then click-hold on the blue Release line, and
drag up/down to the desired position.
Stiffness (morphable)
Controls the stiffness (rigidity) of the string. In reality, this is determined by the string
material and diameter (or to be more precise: by its geometrical moment of inertia).
Stiff strings exhibit an inharmonic vibration, where overtones are not integer multiples
of the base frequency. Rather, they have higher frequencies.
In effect, an increase in Stiffness turns the string into a solid metal bar.
Simply click-hold, and drag, the Inner Loss and Stiffness “ball” to the desired position.
Note that as you do so, the thickness of the “string”—the green horizontal line in the
Pickup window—will change.
Stiffness Scale Low/High
Allows independent adjustments to the key tracking of the stiffness parameter, for
notes above and below C3.
To adjust Stiffness keyscaling, first select the Keyscale button, then click-hold the green
vertical line for low notes or the blue vertical line for high notes, and drag left/right to
the desired position.
You can also simultaneously adjust both Stiffness and Inner Loss keyscaling by dragging
the diamond that intersects the green lines.
Low Stiffness values combined with low Inner Loss values lead to “metallic” sounds.
Increasing the Stiffness makes the sound become more “bell” or “glass”-like.
Increasing the Inner Loss value, while maintaining a low Stiffness level, corresponds to
nylon or catgut strings.
High Stiffness values combined with high Inner Loss values simulate “wood”-like
materials.

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The String Parameters Around the Material Pad
Resolution (Harmonics)
This parameter determines the maximum number of harmonics contained in (and
spatial resolution of ) the sound at C3. This is roughly proportional to the required CPU
power, so the more harmonically-rich/the higher the Resolution setting of the sound,
the more processing muscle will be required.

Note: As you alter the Resolution value, you are changing the interaction of the string
with the Objects. This also changes the frequency of the overtones: very low Resolution
values result in inharmonic spectra—even with Stiffness set to zero.
Resolution Scale Low/High
Allows you to set the key tracking resolution (the accuracy of key tracking) separately
for notes above and below middle C (C3). The green “low” slider (down to C0), inside
the Material Pad ring, is visible when the Keyscale button is clicked. To adjust, click-hold,
and drag. The blue “high” slider (up to C6) runs along the left-hand side of the outer
ring. Click-hold, and drag, to adjust.
Media Loss (morphable)
Controls dampening of the string caused by the surrounding media, for example; air,
water, pea, and ham soup and so on. These “losses” are independent of frequency. This
approach allows control over the duration of the (exponential) amplitude decay, once
the excitation of the string has stopped.

Media Loss Scale Low/High
Allows independent key tracking adjustments of media losses for notes above and
below middle C (C3). The green low slider (down to C0), inside the Material Pad ring, is
visible when the Keyscale button is clicked. To adjust, click-hold, and drag the green
arrow. The blue high slider (up to C6) runs along the left-hand side of the outer ring.
Click-hold, and drag the blue arrow, to adjust.

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Media Loss Release
The blue slider (in the outer ring of the Material Pad) controls the Media Loss release
time. To activate it, you must first click on the Release button, to the bottom right of the
Material Pad. Values above 1.0 cause media losses to increase when the key is released.
This parameter can be used to simulate a string that is dropped into a bucket of water
after initially vibrating in air, for example. Obviously, this is not what the average
violinist or pianist would do, but it can be useful for a number of interesting sound
variations.
Tension Mod (morphable)
Strings, such as those of a guitar, exhibit a particularly prominent non-linear
behavior: if the string excursion is large, the string is detuned upwards. As this
detuning is caused by the momentary, rather than the average excursion of the string,
the detuning occurs very quickly. This phenomenon is known technically as “tension
modulation non-linearity”. Non-technically, setting or modulating the Tension Mod
parameter to values above 0.0 emulates this momentary detuning effect in Sculpture.

Please note that this non-linear effect can produce some surprising results, and can
also make the entire model unstable, especially when combined with low Media Loss
and Inner Loss values. So, if you have a sound that “spikes” or “drops out” during the
decay phase, try reducing Tension Mod (and perhaps Resolution).
Tension Mod Scale Low/High
Allows independent adjustments to tension modulation key tracking, for notes above
and below middle C (C3). The green “low” slider (down to C0), inside the Material Pad
ring, is visible when the Keyscale button is clicked. To adjust, click-hold, and drag. The
blue “high” slider (up to C6) runs along the left-hand side of the outer ring. Click-hold,
and drag, to adjust.
Note: If you find that your instrument seems slightly sharp or flat as you play up/down
the keyboard, look at making adjustments to the Tension Mod and perhaps Media Loss
keyscaling parameters.

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Excite/Disturb Object Parameters

The following parameters are used to excite, disturb or dampen the String.
Important: At least one Object must be used, as the String itself does not make any
sound!
As you’ll discover shortly, there are a number of different string excite “models” such as
blow, pluck, bow, and so on. Needless to say, these quite radically alter the general
timbre of the String’s attack phase, resulting in bowed or plucked “flute” and “bell”
sounds, or guitars with a flute-like “blown” sound.
Judicious use of the Object parameters can deliver very accurate emulations of realworld instruments, or sounds that are altogether more “other-worldly”.
A particular aspect of component modelling to note is that each additional disturb/
damp Object that is activated will impact on the String. This will, in turn, alter the
interaction of any other active Object with the String, often resulting in a completely
different character to your sound.
Obviously, changing the sonic character is the reason why you would use a new Object,
but the pluck and blow combination you selected may sound like fingernails on a
blackboard—rather than the “plucked pan flute” you were going for—depending on
other String settings.
As such, you need to pay special attention to the Type and Strength of Objects. You may
find that the “flavor” of the excite Object (1 or 2), for example, has changed
significantly—and you may need to adjust or change the parameters of all Objects
(and perhaps several String parameter values) after introducing a new disturb/damp
Object (2 or 3). Similarly, the selection of a different Type of excite Object will impact on
the disturb/damp Objects (and the string, obviously), and therefore the character of
your sound.

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The repositioning of Objects changes the timbre of the String. If emulating, say a guitar,
changing an Object position could be viewed as similar to picking or bowing a string at
various spots along a fretboard.
About Objects and Velocity Sensitivity
It is important to note that:
• Object 1 is velocity sensitive.
• Object 2 is only velocity sensitive when a Type that actively excites the string is
selected.
• When damping Objects are used, Object 2 is not velocity sensitive.
• Object 3 is not velocity sensitive. This is because Object 3 can only be used as a
disturber of the string and not as an active excitation element.
On/Off (1, 2, and 3 buttons)
Enables/disables the respective Object. When active, each button is lit in a fetching
shade of aqua blue.
Type
The following three tables list all types available for Sculptures Objects.
Note: Object 1 can only make use of the excite Types found in the first table. Object 2
can make use of any of the Types available in either table. Object 3 can only make use
of the disturb/damp Types found in the second table.

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The following table lists all excite Types available for Objects 1 and 2, and information on
the “controls” available for each. Click-hold on the Type panel for these objects, and
select from the list.
Name

Description

Strength controls Timbre controls:

Variation controls:

Impulse

a short impulse excitation

impulse amplitude width

velocity
dependency of
width

Strike

short excitation like piano hammer or mallet

hammer start
speed (velocity
dependent)

hammer mass

felt stiffness

Grav Strike

like hammer but with gravitation towards the string,
leading to multiple hammer-string interactions and
disturbed string vibrations

hammer start
speed

felt stiffness

gravitation

Pick

finger or plectrum picking

pickup force and
speed

force/speed ratio

plectrum stiffness

Bow

bowing of the string

bow speed

bow pressure

slip stick
characteristics

Bow Wide

same as bow, but wider, resulting in a more mellow
tone, esp suited for smooth bow position changes

bow speed

bow pressure

slip stick
characteristics

Noise

noise injected into the string

noise level

noise bandwidth/ noise resonance
cutoff frequency

Blow

blow into one end of the “string” (an air column, or
tube). At various positions, starting from 0.0 (far
left): move the blowing direction and position from
“along the string”, towards one end. The string is
“blown sideways” at the chosen position.

lip clearance

blow pressure

noisiness

level

cutoff frequency
of lowpass filter
being used to
process sidechain
signal

width (size) of the
string area being
affected by the
sidechained signal

External
feeds sidechain signal into string.
(Only avail.
for Object
2)

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The following chart lists all Disturb/Damp Types available for Objects 2 and 3. Click-hold
on the Type panel of these Objects, and select from the list.

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Name

Description

Strength controls Timbre controls:

Disturb

A disturb object that is placed at a the hardness of
fixed distance from the string’s
the object
“resting” position

the distance from the resting
position.
Negative values: push the string
away from the resting position
Positive values: string is
unaffected when in the resting
position

Disturb
2-Sided

This parameter is somewhat akin
to a “ring” placed around the
string, that limits the string’s
vibration in all directions

the hardness of
the ring

the “clearance” of the ring (the
no impact
distance between the ring and
string).
(Negative values: the sides of the
damping ring overlap, influencing
the string if any movement occurs
Positive values: there is an amount
of clearance inside the ring. The
string will only be influenced if
moved sufficiently to actually
touch the ring

Bouncing Emulates a loose object laying or
bouncing on, and interacting
with, the vibrating string. This is
very random, by nature, and can’t
be sync-ed.

controls the
gravity constant
for the object
laying/bouncing
on the string

the stiffness of the object

Bound

A boundary that limits and
reflects string movement. This is
much like a fingerboard that
limits string movement when the
string is plucked very firmly

the distance from
the boundary
“center” position
to the string’s
“resting” position

the slope (steepness) of the
the amount of
boundary.
reflection at the
A value of 0.0 places the boundary boundary limits
parallel to the string.
Other values will position the
boundary closer to the string on
one end, and further away on the
other.

Mass

Used to model an additional mass the mass size/
attached to the string. This can
weight
lead to inharmonic sounds, and
very interesting results, if the
position of this mass is modulated
“along” the string

no effect

Damp

Localized damper, which is useful
for soft damping

the damping characteristics

the width of the
damped string section

the intensity of the
damping

Chapter 24 Sculpture

Variation controls:
controls width.
Negative values: only
a small section of the
string is affected
Positive values: a
broader section of the
string is affected

the damping of the
object

Gate
Determines when the Object is active—in other words, when it disturbs/excites the
String. Options are:
• KeyOn—between note on and note off.
• Always—between note on and the end of the release phase.
• KeyOff—triggered at note off, and remaining active until the voice is released.

Note: If using an Object Type such as Gravity Strike, the note may retrigger when you
release the key. To avoid this artefact, set Gate mode to Always.
Strength (morphable)
The central dial adjusts the intensity of the excitation/disturbance, dependent on Type.
See table above. A value of 0.0 means no excitation/disturbance at all. In contrast to the
On/Off switch, however, it is possible to fade in the Strength (intensity of excitation/
disturbance) via modulation and/or morphing.
Timbre (morphable)
The fader to the left of each Object’s controls is responsible for the timbre (tonal color)
of the excitation/disturbance, which is type dependent. See the table above. 0.0 is the
“normal” value for the Object. Positive values make the sound brighter, while negative
values lead to a more “mellow” sound.
Variation (morphable)
The Variation slider to the right of each Object’s controls is an additional timbre
parameter, which is again type dependent. Please see the table above for details of its
impact on the sound.
VeloSens (Objects 1+2 only)
The excite/disturb Objects are velocity sensitive, but this may not be appropriate for all
sounds. This parameter, found at the bottom of Objects 1 and 2, allows you to reduce
the velocity sensitivity to zero.

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Position (morphable)

Determines the position of each Object along the String (A value of 0.0 means one end,
and a value of 1.0, the other end of the String).
To adjust, simply click-hold and drag the corresponding numerical slider handle (the 1,
2 or 3 “arrows”) for each Object. Adjustment of these Object pickup positions will
disturb/excite a given portion of the string.
Object 1 can be an exciter. Object 3 can be a damper. You’ll note that Object 2 has two
arrows. This indicates that this Object can be used as either an exciter or damper.
• As you move the Object pickups through the Pickup A and B ranges (shown as
transparent bell curves), you’ll find that the intensity of the Object disturbance
increases significantly. This gives rise to a number of changes which can completely
alter the general timbre of your sound.
• The green horizontal line within the Pickup window represents the string. As the
Stiffness of the string is increased, the line will become thicker.
• You can Control-click on the green horizontal line (the String) to activate/deactivate
string animation. When active, this graphical string will vibrate, making it easier to
visualize the impact of the Objects and Pickups. Note that string animation increases
the CPU overhead, so disable it if your computer is struggling to process all data in
realtime.
• The vertical orange lines represent the positions of disturb/excite/damp Objects 1, 2,
and 3. The thickness and brightness of these lines indicate the Strength of the
Object(s).

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Processing
The “processing” tools covered in this section act on a per-voice basis, as do the String
parameters discussed in the previous chapter.

Pickup Parameters

• The transparent bell curves represent the position and widths of Pickups A and B.
• The green horizontal line within the Pickups window represents the String. As the

Stiffness of the String is increased, the line will become thicker.
• The vertical orange lines represent the positions of disturb/excite Objects 1, 2, and 3.

The thickness and brightness of these lines increases as the Strength level of each
Object is raised.
Note: View these pickups as being like the electromagnetic pickups found on an
electric guitar. Obviously, changing their positions will alter the tone of your axe, and
they’ll do the same in Sculpture.
Pickup A Position (morphable)
The Pickup A slider at the top of the Pickup window determines the position of Pickup
A along the string. Just click-hold, and drag, the slider “cap” to adjust.
Values of 0.0 and 1.0 determine the two ends of the string.
Pickup B Position (morphable)
The Pickup B slider at the bottom of the Pickup window determines the position of
Pickup B along the string. Just click-hold, and drag, the slider “cap” to adjust.
Values of 0.0 and 1.0 determine the two ends of the string.
Invert (Pickup B Phase)
The Invert button is found at the bottom left of the Pickup window. Options
are: normal or invert(ed).
Note: If the phase of Pickup B is inverted, the sound becomes thinner because portions
of the Pickup A and Pickup B signals cancel each other out.
While not found in the actual Pickup window, two further pickup parameters are
available …

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Stereo (Key)
Panning position (left/right) is determined by MIDI note number. Dependent on
settings, the further up/down the keyboard you play, the more the voice signal is
panned left/right.

Stereo (Pickup)
Spreads the two pickups across the stereo base. In other words, the pickup position,
combined with this parameter, will be spread further towards/from the left/right stereo
channels.
Note: You can create animated “width” and “chorus” effects by modulating the Pickup
Position parameters with an LFO or other modulator.

Amplitude Envelope Parameters
This is a “classic” analog synthesizer ADSR envelope that scales the Pickup signals before
passing them on to the Waveshaper and Filter. This approach/positioning of the
Amplitude Envelope at this point in the signal path produces more natural sounding
results when the Waveshaper is used.

Note: Even with long decay/release times, the sound may decay quickly. This can be
caused by high Inner or Media Loss values (in the String material section) or by Objects
(2 or 3) that are used to “damp” the string.
Attack—Soft
The Attack parameter features two sliders. The lower slider determines the Attack time,
when the keyboard/incoming note data is played at minimum velocity.
Attack—Hard
The upper Attack slider determines the Attack time when the keyboard/incoming note
data is played at maximum velocity.

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You can adjust both slider halves simultaneously by dragging in the space between
them.
Important: The Attack time parameters of the Amplitude Envelope have a major
impact on the way a single note is retriggered. When both Attack Soft and Hard are set
to value of zero, the vibrating string is retriggered. If either of these parameters is set to
a value above zero, a new note will be triggered.
Decay
Defines the Decay time. The Decay time is the amount of time that it takes for the
signal to fall to the Sustain level, following the initial strike/Attack time.
Sustain
This parameter sets the Sustain level. The Sustain level is held until the key is released.
Release
This parameter determines the length of time that it takes for the signal to fall from the
Sustain level to a level of zero. Short Release values help to reduce CPU load, as the
voice is no longer processed once the Release phase has completed.

Waveshaper Parameters
These parameters control and determine any non-linear processing/polyphonic
distortion for the stereo signal, resulting from the panning and level scaling of the two
Pickup signals.
The Waveshaper provides a non-linear shaping curve (per voice) that shapes the signal
coming from the Pickups (and Amplitude Envelope) and passes this reshaped signal on
to the Filter. This is quite similar to the waveshaping of oscillators in synthesizers, such
as Korg’s O1/W.

Waveshaper On/Off Button
The Waveshaper button enables/disables the Waveshaper.
Type
The Type pull-down menu above the Waveshaper button allows you to select different
types of waveshaping curves from the list below.
• Soft Saturation
• Vari Drive
• Tube-like distortion
• Scream

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Input Scale (morphable)
This is a bipolar parameter. Negative values attenuate, and positive values amplify, the
input signal prior to processing by the Waveshaper. When set positively, this results in a
richer harmonic spectrum. The level increase introduced by the parameter is
automatically compensated for by the Waveshaper.
Given its impact on the harmonic spectrum, Input Scale should be viewed/used as a
timbral control, rather than a level control.
Note: At extreme Input Scale values, processing noise can be introduced at the
Waveshaper output.
Variation (morphable)
The Variation parameter is a bipolar control. Its impact is dependent on the type of
Waveshaper selected.

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Type

Variation controls:

Value of 0.0

Negative Values:

VariDrive

wet/dry ratio

provides shaped
signal only

reduce shaped signal raise shaped signal
and add dry signal
and add phase
inverted dry signal,
making sound
sharper

SoftSat
Tube Dist.
Scream

Bias—which alters
the symmetry of the
shaping curve

results in
symmetrical
shaping

alter symmetry

Chapter 24 Sculpture

Positive Values:

alter symmetry

Filter Parameters
These parameters offer further timbral/spectral control over your sound. They should
be pretty familiar to you if you have any experience with synthesizers.
On/Off
The Filter button activates/deactivates the filter.

Filter Type Buttons
The five buttons at the bottom of the filter section determine the filter mode. Choices
are:
• Hipass—Allows frequencies above the Cutoff Frequency to “pass”. As frequencies
below the Cutoff Frequency are suppressed, it’s also known as a Low Cut Filter. The
slope of the filter is 12 dB/octave in Highpass mode.
• Lowpass—Allows frequencies which fall below the Cutoff Frequency to “pass”. As
frequencies above the Cutoff Frequency are suppressed, it’s also known as a High Cut
Filter. The slope of the filter is 12 dB/octave in Lowpass mode.
• Peak—This mode allows for an increase in level of a frequency band, the width of
which is controlled by the Resonance parameter.
• Bandpass—In this mode, only the frequency band directly surrounding the Cutoff
Frequency can pass. All other frequencies are cut. The Resonance parameter controls
the width of the frequency band that can pass. The band pass filter is a two-pole
filter with a slope of 6 dB/octave on each side of the band.
• Notch—In this mode, the frequency band directly surrounding the Cutoff Frequency
is cut. All other frequencies are allowed to pass. The Resonance parameter controls
the width of the frequency band that is cut.
Cutoff (morphable)
Determines the filter cutoff frequency. As an example of its use: In a low pass filter, all
frequency portions above the Cutoff Frequency are suppressed, or “cut off”, hence the
name. The Cutoff Frequency controls the brilliance of the signal. The higher the Cutoff
Frequency is set, the higher the frequencies of signals that are allowed to “pass”
through the low pass filter.

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Resonance (morphable)
Sets the filter resonance value.
For highpass and lowpass modes, the Resonance parameter emphasizes the portions of
the signal which surround the frequency—as defined by the Cutoff value.
In Peak, Notch, and Bandpass modes, Resonance controls the bandwidth.
Key
This knob adjusts the key tracking of the Cutoff Frequency. Put plainly, the further up/
down the keyboard you play, the more bright/mellow the sound becomes. Put more
technically, the Cutoff Frequency is modulated by keyboard position.
• A value of 0.0 disables key tracking.
• A value of 1.0 allows the Cutoff Frequency to proportionately follow the fundamental
of the note across the entire keyboard range.
Velo Sens
Determines the velocity sensitivity of the Cutoff Frequency. The harder you strike the
keyboard you’re playing, the higher the cutoff frequency (and generally, the brightness
of the sound) becomes.
• A value of 0.0 disables velocity sensitivity.
• A value of 1.0 results in maximum velocity sensitivity.

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Post Processing
The “post processing” tools covered in this section impact on the summed stereo signal
of all voices, rather than on a per-voice basis.

Stereo Delay
This is a (song) tempo-syncable stereo delay. It may also be set to run freely (not
synchronized).
On/Off
The Stereo Delay button enables/disables the Delay section.
Wet Level
The Wet Level knob sets the level of the Delay output (wet signal). The parameter value
is expressed as a percentage (%). Dry level is 100%.

Feedback
Defines the amount of delay signal that is fed from the left out (of the delay unit) to the
left in (of the delay unit), and from right out to the right in. Negative values result in
phase-inverted feedback.
Crossfeed
As above, but this parameter defines the amount of delay signal that is fed from the
left out to right in and right out to left in (of the delay unit). Negative values result in
phase-inverted feedback of the crossfed signal.
LoCut
Determines the cutoff frequency of the highpass filter at the delay line output/
feedback loop.

HiCut
Determines the cutoff frequency of the lowpass filter at the delay line output/feedback
loop.

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Input Balance
This parameter provides true stereo panning. Adjustments allow you to move the
(stereo) center of the Delay input to the left or right, without the loss of any signal
components. This makes it ideal for “ping-pong” delays.

Delay Base Time
This parameter, coupled with the Sync setting, enables you to set the delay base time.
This can be in either: musical note values—1/4, 1/4t (t = triplet) and so on—or in
milliseconds.

Sync
The Sync button allows you to select either tempo-synced, or tempo-independent
Delay modes.
Stereo Base
This parameter allows a reduction of the stereo base of the wet signal.

• A value of 0.0 = mono output
• A value of 1.0 results in full stereo output (the left delay line output is panned hard

left, and the right delay line output is panned hard right).
This parameter is used to achieve “pure” delay grooves, without hard left/right “pingpong” panning.
The following two parameters are combined in the two-dimensional Delay Time Pad.
Drag the diamond in the center of the crosshair to adjust.
Note: You can independently adjust the Spread and Groove parameter values by
directly dragging the lines that intersect the “diamond”.

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Spread
Positive values on the “y” axis (above the default, centered position) increase the delay
time of the right delay line and decrease the delay time of the left delay line—in effect,
“smearing” the delay times of the left and right channels. Negative values reverse this.
Spread is useful for “wide” stereo delay effects.

Groove
This parameter (on the “x” axis) allows you to reduce the delay time of one delay line by
a given percentage, while keeping the other delay line constant. Basically, Groove
distributes the delay taps to the left/right channels, rather than “smearing” them, like
Spread. Keep an eye on the small Help Tag while adjusting.
As an example, a value of +50% reduces the right delay time by half. If a value of 1/4
was used as the Delay Base Time, the right delay would equal 1/8th of a note and the
left delay would remain at 1/4 of a note. Needless to say, this parameter is perfect for
the creation of interesting rhythmic delays—in stereo.
Note: You can create some truly wide “chorus” and “modulated delay” effects by
modulating the Pickup Position and Pickup Stereo parameters (with an LFO or other
modulator), and then feeding this into the Delay unit.

Body EQ/Spectral Shaping
Model-based spectral shaping. It can work as a simple EQ, as a complex spectral shaper
or as body response simulator. In effect, the Body EQ can emulate the resonant
characteristics of a wooden or metallic “body”—such as that of a guitar or violin.
The various models are derived from Impulse Response recordings of instrument
bodies. These recordings have been separated into their general formant structure and
fine structure, allowing you to alter these properties separately.
Body EQ On/Off Button
This button (to the left of the image) enables/disables the spectral shaping section.

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Model
The Model pull-down menu allows you to select the model from the following list. Any
selection will be reflected in the graphic window to the right:
• Low Mid Hi—A broadband equalizer module, with individual controls (via the knobs)
of Low, Mid, and High frequency ranges.
• Guitar 1—Emulates the body of an acoustic guitar.
• Guitar 2—Emulates the body of another acoustic guitar.
• Violin 1—Emulates the body of a violin.
• Violin 2—Emulates the body of another violin.
Body EQ Controls
Please note that the following parameters are labeled differently for the Basic EQ (Low
Mid Hi Model). All other model parameters are discussed below.
• Low—gain of a low shelving filter—set at 80 Hz
• Mid—gain of a peak filter—set at 2.5 kHz (but is sweepable—see below)
• High—gain of a high shelving filter—set at 12 kHz
In addition, when the Lo Mid Hi model is selected, the slider above the response graph
is labeled Mid Frequency, and allows you to sweep the center frequency of the mid
band between 100 Hz and 10 kHz.
You can directly control the Lo Mid Hi model by click-dragging in the Body EQ graph:
• Click-dragging vertically on the left third of the graph allows you to control the Low
parameter.
• Click-dragging vertically on the center third of the graph allows you to control the
Mid parameter.
• Click-dragging horizontally on the center third of the graph allows you to control the
Mid Frequency parameter.
• Click-dragging vertically on the right third of the graph allows you to control the Hi
parameter.
For all other Models:
For all other BodyEQ models you have the following parameters:
Formant—Intensity
Scales the intensity of the model’s formants. In other words, any formants (harmonics)
in the model will become louder, or will be inverted, dependent on how this parameter
is used.
• A value of 0.0 results in a flat response.
• A value of 1.0 results in strong formants.
• Negative values invert the formants.

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Formant—Shift
This parameter shifts the formants logarithmically. A value of −0.3, for example, shifts all
formants one octave downwards, and a value of +0.3 shifts the formants up one octave.
A value of +1.0 shifts up by a factor of 10—from 500 Hz to 5000 Hz, for example.
Formant—Stretch
Stretches the formant frequencies, relative to each other. In other words, this parameter
alters the width of all bands being processed by the Body EQ, extending or narrowing
the frequency range.
Low Formant Stretch values move the formants closer together (centered around 1 kHz)
while high values move the formants further apart from each other. The control range
is expressed as a ratio of the overall bandwidth.
When combined, Formant Stretch and Formant Shift alter the formant structure of the
sound, and can result in some interesting timbral changes.
Fine Structure
This parameter enhances the spectral (harmonic) fine structure, making the overall
harmonic makeup of the sound more “precise”. This results in a more detailed sound
that is harmonically richer and—dependent on the model selected—more guitar or
violin-like, for example. Put another way, the resonant cavities of the instrument
become more resonant—somewhat like the increased “depth” of tone provided by a
large-bodied guitar.

• A value of 0.0 denotes no fine structure.
• A value of 1.0 results in enhanced/full fine structure of the selected model.

Please note that heavy use of Fine Structure may be quite CPU intensive.
You should also note that the use of Fine Structure may not actually result in too much
of a difference to your sound. This is dependent on the Waveshaper and Body EQ
modes, plus other String parameter settings. As always, use your ears!
Adjusting Model
You can directly control the model by click-dragging in the Body EQ graph:
• Click-dragging vertically on the graph allows you to control the Formant Intensity
parameter.

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• Click-dragging horizontally on the graph allows you to control the Formant Shift

parameter.

Level/Limiter
Level
Controls the overall output level for the instrument. Drag to adjust.

Level Limiter Mode
Clicking on the desired button activates/deactivates the integrated limiter. Options are:
• off—disables the limiter.
• mono—a monophonic limiter on the summed signal of all voices.
• poly—a polyphonic limiter, that processes each voice independently.
• both—a combination of both limiter types.
This function is very useful for “domesticating” some of the more “aggressive” aspects of
Sculpture’s component modelling synthesis engine.

Modulation Generators
Sculpture features a comprehensive collection of modulation generators. There are:
• Two assignable LFOs, with (song) tempo-syncable rates.
• An additional LFO that is hard wired to Vibrato.
• Random variations can be created via two Jitter generators (with adjustable
bandwidth).
• Two Randomizers that only change values at note start/on.
• Two Control Envelopes can either be used as “standard” envelopes or as MIDI
controlled modulators—with the ability to record, polyphonically play back (on a
per-voice basis) and modify incoming MIDI controller movements.
All modulation assignments take place within the generators.
“First order” modulations are where the modulation generator modulates core synthesis
parameters.
“Second order” modulations are where one modulation generator modulates the
parameters of the other modulation generator. These include: LFO Rate Modulation,
VariMod, Morph Envelope Modulation, and A Time Velosens.

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Each modulation generator allows you to select one (or in most cases, two) of the core
synthesis parameters as a modulation “target”. The same target can be selected for all
modulators, if desired.
Modules such as the LFOs and Control Envelopes also offer “via” modulations (a freely
modulatable amount/intensity—the scaling factor—of the modulation source output
level). These are “sidechain” modulations.
Some modulation generator parameters—for example LFO rates, can be modulated by
selecting the desired modulation “source” and amount.
To access the desired modulation generator, click on the corresponding button in this
section of the Sculpture GUI. Once a modulation source has been activated, the
corresponding button label will be lit.

LFO 1 and 2
The two full-featured LFOs 1+2 offer several possibilities that go beyond the Vibrato
LFO and Jitter generators described below.

Waveform
This pull-down menu allows you to select the waveform used for LFO modulation,
Options are: Sine, Triangle, Sawtooth, Rectangle Unipolar, Rectangle Bipolar, Sample&Hold,
Sample&Hold with Lag, Filtered Noise.
The triangular wave is well suited for vibrato effects.
The sawtooth is well suited for helicopter and “space gun” sounds. Intense modulations
of incoming frequencies leads to “bubbling and boiling, underwater” sounds. Intense
sawtooth modulations of lowpass filters create rhythmic effects.
The rectangular waves make the LFO periodically switch between two values (e. g.—a
positive value and zero—Unipolar). The Bipolar Rectangular wave switches between a
positive and a negative value set to the same amount above/below zero.

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The two Sample&Hold waveform settings output random values. A random value is
selected at regular intervals, defined by the LFO rate. A modulation of Pitch leads to the
effect commonly referred to as a “random pitch pattern generator” or “sample and
hold”. Check out very high notes, at very high rates and high intensities—you’ll
recognize this well-known effect from hundreds of science fiction movies! If the
Sample&Hold with Lag setting is used, the random wave is delayed, resulting in a “fluid”
changing of values.
The term “Sample & Hold” (abbreviation—S & H) refers to the procedure of taking
“samples” from a noise signal at regular intervals. The voltage values of these “samples”
are then “held” until the next sample is taken. When converting analog audio signals
into digital signals, a similar procedure takes place: Samples of the voltage of the
analog audio signal are taken at the rate of the sampling frequency.
Rate
The Rate knob determines the rate of LFO modulation, which can be either synced to
the current song tempo or set independently in Hz (Hertz) values.
• In “Hz” mode, the rates range from DC (Direct Current) to 100 Hz.
• In “sync” mode, rates range from a periodic duration of 32 bars through to speeds of
1/64-triplets. Triolic and punctuated values are also possible.
The LFO’s are ideally suited for rhythmic effects which retain perfect synchronicity even
during tempo changes to the song.
Sync/Free Buttons
Part of the Rate parameter, these buttons allow you to select either synchronized or
free-running LFO rates. When toggling between modes the value is derived from the
song tempo and meter.
Curve
Allows you to define a freely-variable number of waveform variations, resulting in
subtle/drastic changes to your modulation waveforms.
The Curve parameter can even influence the Sine waveform type.
• Curve value of 0.0—pure sine wave
• Curve values above 0.0—wave is smoothly changed into a nearly rectangular wave.
• Curve values below 0.0—the slope at the zero crossing is reduced, resulting in
shorter soft pulses to +1 and −1.
Note: The waveform displayed between the Curve knob and the Waveform pull-down
menu shows the results of these two parameter settings.
Envelope
A simple LFO envelope control that allows either;
• constant modulation
• a ramping up to the full amount

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• a decay to a zero amplitude level.

Phase

Allows the choice between strictly monophonic or polyphonic LFO modulations with
either; similar phases, completely random phase relationships, key-synced phase … or
anything in-between.
• If used polyphonically for modulation of multiple voices, the modulations will not be
phase-locked. To explain, when used on polyphonic input (a chord played on the
keyboard) the modulation is independent for each voice (note). Where the pitch (for
example) of one voice may rise, the pitch of another voice might fall and the pitch of
a third voice may reach its minimum value.
• When used monophonically, the pitch of all voices will rise and fall synchronously.
• If used randomly, some notes will be modulated synchronously, and others won’t.
Note: If you move the Phase knob slightly away from the mono position, you’ll get nonlocked modulations for all voices running at similar, but not identical, phases. This is
ideal for string-section vibratos!
RateMod Source and Amount
The LFO rate can be modulated by the Source and amount that you determine with
this pull-down menu and vertical slider (found to the right of the abovementioned LFO
controls).

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Target1/2
These two modulation destinations can be assigned per LFO, with an optional,
additional “via” modulation. These parameters determine modulation destinations 1
and 2. To activate, click on either the 1 or 2 buttons (which will highlight the Target and
via text/pull-down menus), then click-hold on the Target pull-down menu(s). Select the
desired target, and release the mouse button.

Via (source) 1/2
These parameters determine the source that controls the modulation scaling for each
LFO.
Modulation amount sliders: Amt and Via (Amount)
In cases where the Via source is set to off, only one amount slider is visible (the Via
(Amount) slider is hidden):
• Amt 1/2—The Amt slider determines the modulation amount.
In cases where any Via source other than off is selected, there are two sliders:
• Amt 1/2—The Amt slider determines the modulation amount in cases where the
incoming Via signal is zero, e. g. a modulation wheel at its minimum position/value.
• Via (Amount) 1/2—The Via (Amount) slider determines the modulation amount in
cases where the incoming Via signal is at full level, e. g. a modulation wheel at its
maximum position/value.

Vibrato
One LFO is hard-wired to pitch, for vibrato effects. The strength of the vibrato effect can
be adjusted via the MIDI controller assigned in the VibDepth Ctrl menu. This is set in the
MIDI controller assignment section. More information can be found in the “MIDI
Controller Assignments” on page 408.
Waveform
Allows you to select the waveform used for vibrato, for example sine, triangle, sawtooth,
and so on.

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There are two “special” rectangular waves: rect01 and rect1—the former switching
between values of 0.0 and 1.0 (unipolar), and the latter between values of −1.0 and +1.0
(bipolar, like the other waveforms). See the “LFO 1 and 2” section, on page 385.

Curve
Allows you to define a freely-variable number of waveform variations, resulting in
subtle/drastic changes to your modulation waveforms.
The Curve parameter can even influence the Sine waveform type.
• Curve value of 0.0: pure sine wave
• Curve values above 0.0: wave is smoothly changed into a nearly rectangular wave.
• Curve values below 0.0: the slope at the zero crossing is reduced, resulting in shorter
soft pulses to +1 and −1.
Note: The waveform displayed between the Curve knob and the Waveform pull-down
menu shows the results of these two parameter settings.
Phase
Allows the choice between strictly monophonic or polyphonic vibrato with either;
similar phases, completely random phase relationships, key-synced phase—or any
value in-between. See the “LFO 1 and 2” section, on page 385, for more details.
Rate
Determines the rate of vibrato, which can be either synced to the current song tempo
or set independently in Hz (Hertz) values. See the “LFO 1 and 2” section, on page 385,
for more details.
Depth via Vib Ctrl—Max
This slider determines the maximum possible modulation amount.
Depth via Vib Ctrl—Min
This slider determines the minimum possible modulation amount.

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Random Variations
Jitter 1+2
The two jitter generators are special LFOs, designed to produce continuous, random
variations—such as those of smooth bow position changes. The jitter generators are
equivalent to general purpose LFOs set to a “noise” waveform.
To activate the routing of the Jitter generators, click on the 1 or 2 buttons.

Note: Jitter modulation of pickup positions as the Target produces great chorus-like
effects.
Rate/Bandwidth
These knobs set the bandwidth/speed of the modulation (jitter) signal for each
respective Jitter generator.
Target 1/2
These pull-down menus define modulation destinations 1 and 2.
Amount 1/2
These sliders determine the amount of modulation for each Jitter generator.
Note On Random
The two note on random generators are intended for random variations between
different notes/voices. Their values are randomized for each note, and remain constant
until the voice is released. Such randomizations are useful for adding interest/
thickening the sound when played polyphonically. It is also useful for emulating the
random variations a player introduces when playing an instrument—even when
repeating the same note.
To activate the routing of the Note On Random modulators, click on the 1 or 2 buttons.
Target
Determines the modulation destination—what parameter will be randomly modulated
when a note is played.
Amount
Sets the modulation amount—the depth/strength of the modulation.

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Velocity Modulations
The Excite Objects and the Filter have dedicated velocity sensitivity controls. Many other
modulation routings also allow you to select Velocity as a via input source.
In some cases, it may be of use/interest to directly control other synthesis core
parameters by velocity. This can be done in this section—where two independent
destination/amount/velocity curve slots are available. To activate either, click on the 1
or 2 buttons.

Target 1/2
Click-hold on the respective Target pull-down menu to select the destination parameter
that you wish to modulate by velocity.
Amount 1/2
The slider below each Target determines the amount of modulation.
Curve 1/2
Click on the appropriate radio button to select from concave, linear, and convex velocity
curves.

Controller A and B

These parameters allow you to define two discrete modulation targets, and the
strength/intensity of modulation for both Controller A and B.
To use, simply click on the 1 or 2 buttons, select the desired Target (and Target mode),
and adjust the Intensity slider.
Each Target features a two-state button:
• Cont: continuous modulation
• Note On: modulation value is only updated when a note on message is received.

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The Control Envelopes

The two Control Envelopes are somewhat special, as they can be used as:
• “traditional” four segment envelopes.
• MIDI controller modulations.
• a combination of both: as MIDI controller movement recorders (with ADSR-like
macro parameters), for polyphonic playback.
To select Envelope 1 or 2, click on one of these buttons.

Mode—Ctrl/Env Buttons
Selects either controller (“run” mode) or envelope functionality. If both are activated, the
controller value is added to the envelope output, resulting in a modulation offset.

The Envelopes can act as polyphonic modulation recorders and playback units. Each
voice is handled independently, with a separate Envelope being triggered/running as
each note is played.

Modulation Routing
As in the LFO section, each envelope offers two modulation Target selectors with
amount and via amount controls, and a separate via modulation option.
The following target/amount/via settings are available for all “run” modes.
Target 1 and 2
Two modulation targets can be assigned per Envelope, with an optional, additional via
modulation. These parameters determine modulation destinations 1 and 2.
Either the 1 or 2 buttons must be clicked in order to activate these modulation options.

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A large number of possible Targets are available, including; string, object, pickup,
waveshaper, and filter parameters. To select, simply click-hold on the panel below the
word Target, and make your choice from the list.

Via (source) 1/2
The Via parameter defines the modulation amount for Envelopes 1 and 2. The Via
panels allow the selection of sources that are used to scale the modulation amount of
the envelopes.
Modulation amount sliders: Amt and Via (Amount)
In cases where the Via source is set to off, only one amount slider is visible (the Via
(Amount) slider is hidden):
Amt 1/2
The Amt slider determines the modulation amount.
In cases where any Via source other than off is selected, there are two sliders:
Amt 1/2
The Amt slider determines the modulation amount in cases where the incoming Via
signal is zero, e. g. a modulation wheel at its minimum position/value.
Via (Amount) 1/2
The Via (Amount) slider determines the modulation amount in cases where the
incoming Via signal is at full level, e. g. a modulation wheel at its maximum position/
value.

Envelope Curve Window
The Envelope curve is displayed in the window to the lower right of the Sculpture GUI.
The envelope curve window is only active if the envelope functionality is engaged
(Mode set to either Env or Ctrl+Env).

• The overall time/length of the Envelope is indicated by the numerical entry at the

top right of the window (1790 ms in the graphic).
• The maximum time/length of the Envelope is 48 bars/40 seconds.

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• The lines on the background grid are placed 100 milliseconds apart.
• The background lines are placed 1000 ms apart for very long displayed envelope

times. In sync mode, this is displayed as 1 quarter.
• The envelope is zoomed automatically after releasing the mouse button. This allows

the display of the entire envelope at the highest possible resolution for the graphic
Envelope window.
• This behavior can be disabled/enabled by pressing the Autozoom button—the small
magnifying glass.
• Autozoom is automatically disabled when you perform a manual zoom—by clickholding on the Envelope Curve window background, and dragging horizontally. As a
reference, the current window width is displayed by the numerical entry at the top
right of the window. You can re-engage autozoom by clicking on the Autozoom
button.
• If you click on the handles (nodes) or lines between the nodes in the Envelope
window, the current envelope segment will be highlighted. A small Help Tag also
indicates the millisecond value of the current segment.
Envelope Handling
When first launched, a default Envelope “curve” is automatically created for each
Envelope. To view either, press the Env button in the Mode section.
If a recorded Envelope exists, the parameters discussed in this section will allow you to
fine tune positions and levels.
You will see a few handles (nodes) placed—from left to right—along a straight line
within the Envelope. These are indicators of the following parameters.
• Node 1—Start level.
• Node 2—Attack time position/level.
• Node 3—Loop time position/level (can be freely positioned).
• Node 4—Sustain time position/level (can be freely positioned).
• Node 5—End time position/level.
As you move your mouse cursor along the line, or hover over the nodes directly, the
current Envelope segment is highlighted.
You can create your own envelopes manually, by manipulating the nodes and lines, or
you may record an envelope, as discussed in the “To record an Envelope…” section, on
page 396.
To adjust the time between nodes, click on the desired handle, and drag it left or right.
As you do so, the overall length of the Envelope will change—with all following nodes
being moved.

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You cannot move a node beyond the position of the preceding node. You can,
however, move nodes beyond the position of the following node—even beyond the
right-hand side of the Envelope window—effectively lengthening both the envelope
segment and the overall envelope.
When you release the mouse button, the Envelope window will automatically zoom to
show the entire envelope.
To adjust the level of each node, click on the desired handle, and drag it up or down.
To adjust the shape of the curve between nodes, click directly on the line that connects
them, and drag it up or down.
In the case of recorded envelopes, you may have a more complex “curve” between
nodes. To adjust, Control-click and drag the curve. An example of this is shown in the
next section.
Experiment with each node/parameter to get a feel for its operation. You’ll find that the
Envelopes are very intuitive to use.

Envelope Recording
Before we begin…
It is important to note that you can only record MIDI controller movements of the
assigned MIDI controller. MIDI controller assignments for the Envelopes must be set in
this section of the Sculpture GUI:

R(ecord) Button
Enables the record functionality for the Envelope. This button works in a similar fashion
to the record arm buttons in Logic. To stop recording, simply click on the R button a
second time, or use the “trigger” mode functionality described below.

Record Trigger Mode
The pull-down menu to the right of the R button is used to select different record
trigger modes to start recording (when R(ecord) is active):
• NoteOn: recording starts when a note is played
• Note+Ctrl: recording starts when MIDI control change messages (for the assigned
controllers—see “MIDI Controller Assignments” on page 408) arrive while a note is
held.
• Note+Sus: recording starts when the sustain pedal is depressed while a note is held.

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To record an Envelope…
To provide you with an Envelope recording example:
• Set the Record Trigger Mode to Note+Ctrl.
• Enable record by clicking on the R button.
• Play, and hold, a key—and start moving the modwheel or whatever controllers are
assigned to Envelope controls 1 and/or 2.
To end an Envelope recording…
An Envelope recording ends as soon as at least one of the following conditions is met:
• The record button (R) is disengaged manually by clicking on it.
• All voices are released.
• A new note is played after releasing all keys.
Following the recording of a controller movement, R(ecord) is automatically set to off
and Mode is set to Env. This ensures that only the recorded movement will be active,
regardless of the “stop” position of the recorded controller.
To play back a recorded Envelope…
Polyphonic playback of the recorded envelope occurs when you play a key. The Mode
parameter must be set to Env and the R(ecord) parameter must be set to off.
You can also activate both the Env and Ctrl buttons of the Mode parameter, as this will
allow you to use controllers assigned to Ctrl Env1 or Ctrl Env2 to manipulate the
Envelope in realtime, alongside playback of the recorded Envelope.
Note that if both Env and Ctrl are activated, however, the controller value is added to
the envelope output, resulting in a modulation offset.
Preparing the recorded envelope for editing
The envelope segments and handles are set automatically after recording. To change
the interpretation of the envelope you can grab and drag the vertical lines that
intersect the handles (also see the VariMod paragraph below). Note that this will not
change the shape of the envelope.

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Envelope Parameters
The following parameters are only active if the envelope functionality is engaged
(Mode set to either Env or Ctrl+Env).
A-Time Velosens
This slider is used to set the velocity sensitivity for the Attack time of the envelope.
Positive values will make the Attack time shorter at minimum velocities, and negative
values will make the Attack time shorter at maximum velocities.

Timescale
This parameter allows you to scale the duration of the entire envelope between 10%
(ten times faster) and 1000% (ten times slower). This will also visually impact on the
appearance of the Envelope curve displayed as it is shortened (sped up) or lengthened
(slowed down).
Sustain Mode
Allows you to select the behavior of the Envelope while a note is held. Choices are: the
usual sustain mode, finish mode or one of three loop modes (forward, backward,
alternate). When in any of the loop modes, the loop always cycles between userdefined Envelope “handles” (simply drag them to the desired position) that indicate the
Loop start point, and the Sustain point.
The Envelope can—like any envelope—run in one shot (as long as the note is
sustained). It can also run several times or in an infinite cycle, much like an LFO. You can
achieve this through the use of loops.
• When set to Finish, the Envelope runs in “one shot mode” from its beginning to its

end—even if the note is released before the envelope has come to its end. The other
loop parameters are disabled.
• When set to Loop Forward, the Envelope runs to the sustain point and begins to
repeat the section between the loop point and sustain point periodically—always in
a forward direction.
• When set to Loop Backward, the Envelope runs to the sustain point and begins to
repeat the section between the sustain point and loop point periodically—always in
a backward direction.
• When set to Loop Alternate, the Envelope runs to the sustain point and returns to the
loop point and back to the sustain point periodically, alternating in both a backward
and forward direction.
Note: If the loop point lies behind the sustain point, the loop will start after the key has
been released.
Loops can be synchronized to the song tempo automatically via the sync/ms buttons.

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Compare
Following the initial (original) recording of an envelope, and any subsequent edits, this
button allows you to toggle between the original recording and the edited version.
Obviously, this is only available as an option if an envelope curve has actually been
recorded.

Sync/ms Buttons
These parameters allow you to select between a free-running envelope (with segment
times displayed in milliseconds) or a tempo-synced envelope with note value options,
such as 1/8th or 1/4.
Switching between values forces a recalculation of times to the nearest note value or
ms time, respectively, based on the current song tempo.
VariMod—Source and Amount
VariMod is only available for recorded envelopes.
“Variation” in the Envelopes means the deviation of a recorded envelope path from
“straight” interconnecting lines between the points. After having recorded an envelope
you can reduce or exaggerate the amplitude-jitter of the recording by Controldragging the curves down (to reduce) or up (to exaggerate).
The variation parameter can be modulated by using these source and amount
parameters. To adjust the amount/level of variation, simply move the VariMod slider up/
down. Source options can be accessed by click-holding on the Source button, and
making your selection from the pull-down menu. Choices include: Off, Velocity Concave,
Velocity, Velocity Convex, KeyScale, Ctrl A, and Ctrl B.

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VariMod allows you to select a modulation source, and amount, to control the strength
of the “deviation”. As an example; if you recorded a tremolo, as shown.

You could trim the loop by dragging the envelope “handles” to fit the Attack peak, and
the start and endpoints of a tremolo loop.
• To do so, grab and drag the vertical lines that intersect the handles. (These are also
highlighted when the mouse pointer touches them). This will result in something like
this:

• If you now reduce the variation depth by Control-dragging down the curve of the

loop segment, you’ll end up with the following:

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• Now, set up the VariMod source to CtrlA (which might be set to Touch, for example),

and an amount above 0. You’ll end up with an interesting tremolo (or whatever
you’re modulating with this envelope) with an arbitrary waveform, and a modulation
depth that can be controlled by Key Pressure (Touch) or whatever controller is
assigned to CtrlA.

Copying Envelopes
If you Control-click on the Envelopes, a context menu opens, offering Copy, Paste, and
Clear options. These allow you to copy and paste (or clear) envelopes between
Envelopes 1 and 2, between settings and/or multiple open Sculpture instances.

Morph
Sculpture includes “morph” functionality that allows you to smoothly change the
sound—in a subtle or radical way—between up to five snapshots, called morph points.
Each morph point includes separate settings for more than 20 important synthesis core
parameters, including; string material parameters, parameters and positions of the
excite/disturb objects, pickup positions, filter, and waveshaper parameters.
All morphable parameters can be automated independently for each morph point.
The current morph point position within the Morph Pad can be controlled via MIDI
controllers (for example by a vector stick). Such movements can be recorded and
played back independently—each voice can be morphed differently.
The morph section consists of two parts:
• The Morph Pad, featuring 5 morph points (center and 4 corners), plus options for
randomizing (via the Randomize parameters) and copy/pasting morph points and
Morph Pad “states” via a contextual menu.

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• The Morph Envelope, that can be edited either by segment (with the mouse), or

recorded via MIDI controller movements. With a vector stick (Morph X/Y controllers)
or mouse movements of the Morph Cursor (the “ball”), for example, on the Morph
Pad.

Morph Point Selection and Randomization

There is always one of the five Morph Pad points (A/B/C/D/Center) that is selected for
editing. This selected point is indicated by two concentric circles that surround it.
All morphable parameters (all parameters which have an orange value bar, rather than
a blue or turquoise one) are shown, allowing you to edit the values of the selected
Morph Pad point.
Auto Select
Movements of the morph cursor automatically select the nearest morph point when
this parameter is active.

You can also click into the circles around A, B, C, D, or Option to select a Morph Pad
point manually.

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Randomizing
The randomize feature allows you to create random variations of selected morph
points. When combined with the copy/paste functionality that’s also available,
randomizing lends itself to using the Morph Pad as a kind of sound cell culture device.
Use of the Morph Pad can yield an interesting/inspiring composite sound. You can copy
this sound to a corner of the Morph Pad (or several corners) and randomize it by a
definable amount.
The morphed sound then becomes a new timbral element, creating permutations that
can, in turn, be moved to the corners, randomized, and so on …
In effect what you are doing is “breeding” a sound, while maintaining some control by
selecting parent and child sounds.
This approach can result in new, complex sounds without the need to be a
programming guru.
Points
Selects the number of morph points that are to be used for randomization, and
indicates which points will be randomized.

Randomize Button
The Rnd button generates a randomized collection of Morph points.

Int(ensity)
The Int(ensity) slider determines the grade of randomization from 1% (slight deviation)
to 100% (completely random values).

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A Brief Randomizing Tutorial
• Select the top button (5 points) in the Points section. Ensure that Auto Select is active.
• Select the Int(ensity) slider, and drag it to a value of say 25%.
• Now press the Rnd button. As you’re doing so, take a look at the parameters in the
core synthesis engine. You will see a number of them move after the mouse click.
• Now, click-hold on the Morph Cursor (the “ball”), and drag it to each of the corners in
the Morph Pad. Do this along the edges, as well as through the center of the Morph
Pad, and take a mental note of how this impacts on the “morph”.
• Oh, and don’t forget to strike a few notes on your MIDI keyboard while doing so.
As you’re moving the morph cursor around, you’ll see the realtime movement of the
“ghost” controls in the Pickup window, the “ghost” ball in the Material Pad, and—if you
look closely—you’ll also notice a number of red dots moving about in the various
string and object parameters.
Each of these indicates the “current” Morph position. This is handy tool for seeing what
parameters have changed, and how they have been changed.
You’ll also note that positions on the Morph Pad that fall in-between the various morph
points cause the randomized parameters to interpolate between values—which is
what “morphing” is all about.
Check out the next section on the copy/paste options, as these will help you to make
use of those “in-between” settings.

Morph Menu Parameters
You can access this contextual menu by Control-clicking on the Morph Pad.

Copy/Paste
These parameters allow you to copy:
• the current morph point (Copy selected Point) or the
• current morph state (Copy current Pad Position) into a local clipboard, and
• Paste to a selected Point
• Exchange a selected Point (basically, swaps previously copied data with another point
of your choice) or
• Paste to all Points

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The remaining entries in the Morph Pad contextual menu are to do with grouping of
random parameters. Put another way, the following options allow you to determine
which type of parameters you would like to randomize (via the Rnd button and
Int(ensity)) slider.
• All (Random Group)—this is your ticket to “wacky” sounds as all parameters in the

•

•
•
•

following three groups are randomized. This can lead to some interesting results, but
can be uncontrolled. This is less useful for the “sound cell culture” idea that we
discussed in the Randomizing section above.
All except TensionMod—basically the same as All (Random Group) but excluding the
somewhat dangerous TensionMod parameter from randomization. This option is the
default “random” group.
String (parameters)—in isolation, including; Material Pad position, stiffness, inner loss,
media loss, resolution, and Tension Modulation.
Objects + Pickups—in isolation. Alters the positions of Objects and Pickups, plus the
various Object parameters.
Filter + Waveshaper—Randomizes the positions of all Waveshaper and Filter
parameters.

Morph Envelope Window
The Morph Envelope offers 9 points/8 segments, and recording functionality that is
much like that of the Controller Envelopes.
Selected Point
In the diagram below, you will see a selected point in the lower panel (the Timeline)
and a corresponding selected point handle of the trajectory in the Morph Pad.
These are shown as an orange diamond in the Timeline and an orange diamond (or
ball) in the Morph Pad.

• The overall time/length of the Morph Envelope is indicated by the numerical entry at

the top right of the window.
• The maximum time/length of the Morph Envelope is 48 bars/40 seconds.
• The lines on the background grid are placed 100 milliseconds apart.
• If you click on the handles (nodes) or lines between the nodes in the Envelope

window, the current envelope segment will be highlighted. A small Help Tag also
indicates the millisecond value of the current segment.

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• As you move your mouse cursor along the line, or hover over the nodes directly, the

current Envelope segment is highlighted.
• You can create your own envelopes manually, by manipulating the nodes and lines,

or you may record an envelope, as discussed in the “Morph Envelope/Record Path”
section, on page 405.
• To adjust the time between nodes, click on the desired handle, and drag it left or
right. As you do so, the overall length of the Morph Envelope will change—with all
following nodes—being moved.
You cannot move a node beyond the position of the preceding node. You can,
however, move nodes beyond the position of the following node—even beyond the
right-hand side of the Envelope window—effectively lengthening both the envelope
segment and the overall envelope.

Morph Envelope/Record Path
R(ecord) Enable
The R button enables the Morph Envelope record functionality. This button works in a
similar fashion to the record arm buttons in Logic.
To make a Morph Envelope recording, just press the R button, play a note, and start
moving the silver ball in the Morph Pad with the mouse. You can also make use of an
external controller (see the “Morph X/Morph Y” section, on page 409).
The Mode is automatically set to Pad only as soon as the R button is pressed (for more
information on “Modes”, see the “Morph Envelope Parameters” section, on page 406).

Record Trigger Mode
The pull-down menu to the right of the R button is used to select different “trigger”
modes to start recording (when R(ecord) Enable is active):
• NoteOn: recording starts when a note is played.
• Note+Ctrl: recording starts when MIDI control change messages (for the assigned
Morph X/Y controllers) arrive while a note is held.
• Note+Sus: recording starts when the sustain pedal is depressed while a note is held.
Recording is stopped by pressing the R(ecord) Enable button (or trigger) a second time.
Once all keys are released, and all voices have completed their decay phase, the
recording ends. You can stop recording early by releasing all keys, and then pressing a
single key.
Following the recording of a controller movement, R(ecord) Enable is automatically set
to off and Mode is set to Env only. This ensures that only the recorded movement will be
active, regardless of the “stop” position of the recorded controller.

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Morph Envelope Parameters
Mode
Activates the Morph Envelope, and allows you to select from between the following
modes:

• Off—Morph functionality is disabled.
• Pad only—envelope is deactivated, and morph functionality is controlled by the
•
•
•

•

morph cursor (the “ball”) and/or X/Y MIDI controllers only.
Env only—envelope is running, but the morph cursor and X/Y MIDI controllers are
deactivated.
Env+Pad—envelope is running, and the position of the morph cursor and/or X/Y
MIDI controllers is used as an offset for any envelope movements.
Env+Point Set—envelope is running. The selected envelope point can be edited by
moving the morph cursor in the pad or via MIDI (“MorphX” and “MorphY” Controller
Assignments).
Point Solo—envelope in a kind of “still picture” mode. the selected envelope point
can be edited by moving the morph cursor in the pad.

Time Scale
This parameter scales the duration of the entire envelope between 10% and 1000%.

Sustain Mode
Allows you to select the behavior of the Morph Envelope while a note is held. Choices
are: the usual sustain mode, finish mode or one of three loop modes (forward,
backward, alternate).
When in any of the loop modes, the loop always cycles between the loop and sustain
Envelope “handles” (the nodes indicated by the small “L” and “S” icons). The Morph
Envelope can—like any envelope—run in one shot (as long as the note is sustained). It
can also run several times or in an infinite cycle, much like an LFO. You can achieve this
through the use of loops.
Note that if one of the three loop modes is selected, and the loop point is positioned
before the sustain point, the loop will be active until the key is released. Following key
release, the envelope will then continue beyond the sustain point, as per usual.
If the loop point is positioned after the sustain point, the loop will be entered as soon
as the key is released, and will be cycled infinitely (until the complete voice is released
by finishing the release phase of the amplitude envelope).

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The loop and sustain point icons (the small “L” and “S”) can be directly grabbed and
repositioned. Note that doing so can potentially alter the loop (and the overall Morph
Envelope) length.
• When set to finish, the Morph Envelope runs in “one shot mode” from its beginning
to its end—even if the note is released before the envelope has come to its end. The
other loop parameters are disabled.
• When set to forward, the Envelope runs to the sustain point (indicated by the blue “S”
icon) and begins to repeat the section between the loop point (indicated by the blue
“L” icon) and sustain point periodically—always in a forward direction.
• When set to backward, the Envelope runs to the sustain point (“S” icon) and begins to
repeat the section between the sustain point and loop start point (“L” icon)
periodically—always in a backward direction.
• When set to alternate, the Envelope runs to the sustain point (“S” icon) and returns to
the loop point (“L” icon) and back to the sustain point periodically, alternating in
both a backward and forward direction.
Loops can be synchronized to the song tempo automatically via the sync/ms buttons.
Sync/ms
These parameters allow you to select between a free-running envelope (with segment
times displayed in milliseconds) or a tempo-synced envelope with note value options,
such as 1/8th or 1/4.

Switching between values forces a recalculation of times to the nearest note value or
ms time, respectively, based on the current song tempo.
Depth
This parameter scales the amount of morph movement caused by the Morph Envelope.
The effect of the Depth parameter is visually displayed in the Morph Pad. As you
increase/decrease the value, the morph trajectory will also be scaled.

Modulation (Source and Depth)
These parameters allow you to select a modulation source and amount, which are used
to scale the movement of the Morph Envelope.

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Transition
Provides control over the “transitions” between the morph points. This can range from
the original (possibly recorded) movement to linear connections, and beyond this to
“stepped” transitions.

By the latter, we mean remaining at one morph state during the entire Morph Envelope
segment, and then abruptly switching to the morph state set at the following envelope
point.
This parameter (and the Morph Envelope itself ) can lead to interesting evolving sounds
or even rhythmic patches.

MIDI Controller Assignments
This section allows you to define the MIDI controllers you wish to use for vibrato depth
control or morph pad movements, for example. These parameters are saved with each
setting. They are only updated if the default setting (that is loaded on instantiating the
plug-in) is used, or if the setting was saved with a song.
This approach helps you to adapt all MIDI controllers to the keyboard, without having
to edit and save each setting separately.

Learn
All parameters that allow you to select a MIDI controller (VibDepth Ctrl, CtrlA, CtrlB,
CtrlEnv1, CtrlEnv2, MorphX, Morph Y) offer a Learn option. If selected, the parameter
will automatically be set to use the first appropriate incoming MIDI message.
Vib Depth Ctrl
Defines the MIDI controller used for vibrato depth control.
Ctrl A/Ctrl B
Allows the assignment of two controllers that can be used as “via” modulation signals,
for direct modulation routings found on the CtrlA/CtrlB tab or for side chain
modulations.

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CtrlEnv 1/CtrlEnv 2
Sets the controller assignments for the two Control Envelopes—used as a modulation
signal or an offset—in cases where the Control Envelope is set to Ctrl only or Ctrl+Env
modes. It also is used to define the source for recording controller movements.
Morph X/Morph Y
Determines the controller assignments for the X and Y co-ordinates of the Morph Pad.
Once assigned, the controller can be used to: manually move the morph point,
program single morph envelope points, shift the entire morph envelope and as source
for recording morph movements.
Mode
The two Mode menu entries allow you to select whether the MIDI controller
assignments shall be taken from a setting or left as they were, when loading a setting.
Switching between modes toggles between the original assignments saved with the
setting and the default assignments (taken from #default.pst setting—which is loaded
on instantiation of Sculpture, if it exists).

Programming: Quick Start Guide
This section of the manual contains a collection of programming guidelines, tips, tricks,
and information to assist you in creating particular types of sounds. A more involved
look at programming can be found in “Programming: In Depth” section, on page 426.

Approaches to Programming
Given the flexibility of Sculpture’s synthesis core, you can take a number of different
approaches to sound design.
By this, we mean that if you’re the type who prefers to sculpt a sound from scratch—
parameter by parameter—you can.
If you prefer to make use of Sculpture’s morphing capabilities to create new sounds,
you can also do this. This is discussed in the “A Brief Randomizing Tutorial” section, on
page 403.
If you’re more of a “tweaker” of factory or user patches, then the parameters that affect
the entire instrument may be more your style. These include, the Body EQ and Filter
sections, plus the Modulators, for example.
Whatever camp you fall into, you’ll be able to achieve new (and hopefully interesting)
results.
At the end of the day, however, we encourage you to experiment, and familiarize
yourself, with each approach. You will find that each has its strengths and weaknesses,
and that a combination of methods may strike the balance between satisfying sounds
and a social life.

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Basics
Throughout the manual, we have followed the signal flow of the core synthesis engine.
When programming “from scratch”, this is the approach you should also take, working
on each component of the sound in isolation.
Obviously, when you’re starting out with Sculpture, you won’t be familiar with the
impact of each parameter on your end results. Don’t sweat it, we’ll provide some
pointers for particular types of sounds in a few moments … but let’s get back to it.
First up, you’ll need a “plain vanilla” or “from scratch” patch. When you first launch
Sculpture, this is exactly what you’ll get—a default set of “neutral” parameters.
Sonically, this patch won’t set your heart racing, but it will provide you with a starting
point for all of the examples in this chapter.
Note: Save this patch as a Settings file. Name it as desired—maybe “neutral”, “plain
vanilla” or “from scratch”? This Setting can be reloaded as you work through the
examples.

The Core Engine
We discussed the signal flow in the “The Synthesis Core of Sculpture” section, on page
356. To recap, and explain in a more “hands on” way, please follow these steps. This
section is intentionally simplified, but we ask for your forbearance. Knowing the
“mechanics” of Sculpture is essential to your success:
The String
The “string” is the central synthesis element. It offers a range of parameters that allow
you to adjust its material—what it’s made of and what environment it’s being played in
(water, air, and so on).
Each parameter can be explored further in the “String and Object Parameters” on
page 362.
Before starting, we recommend that you Control-click on the “string” (the green
horizontal line in the Pickup section) to activate/deactivate string animation. When
active, the string will vibrate, making it easier to visualize the impact of the objects and
pickups.

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• Press the Keyscale button at the bottom of the Material Pad “ring”, as shown in the

graphic.
• Strike and hold and/or repeatedly strike middle “C” on your keyboard.

Middle “C” is the default pitch/key of the string.
• While doing so, grab hold of the “ball” on the Material Pad by click-holding on it with
your mouse, and move it around. Listen to the sonic changes as you move between
the Nylon, Wood, Steel, and Glass materials. Keep an eye on the String (the green
horizontal line in the Pickup section, to the left) as you’re doing so.
• Release the mouse button once you’ve found a basic tone that you like.
• Now, experiment with the values of each of the sliders that surround the Material
Pad—namely the Media Loss, Tension Mod and Resolution parameters (while
continuing to strike middle “C”). Take note of the changes that each makes to the
sound, and also the string animation in the Pickup section. Play a few notes above
and below middle “C”, again keeping your eye on the string.
• You probably noticed that moving the Media Loss, Tension Mod and Resolution sliders
also had an impact on the green and blue Keyscale sliders inside and outside the
“ring”. Grab, and drag each of these “keyscale” slider arrowheads to different
positions—one by one—and play a few notes either side of middle “C” as you’re
doing so. Note the changes that happen up/down the keyboard range.
• Once done, press the Release button at the bottom of the Material Pad “ring”, and
adjust the blue Media Loss Release slider while striking notes.
The Objects
Up to three “objects” of different types are used to excite or disturb the vibration of the
string. Each parameter can be explored further in the “String and Object Parameters”
on page 362.
• Please reload your default or “plain vanilla” Setting.
• Now press the (Object) 1 button (so that it turns gray), while striking a key. You’ll note
that you hear nothing. The string itself doesn’t make a sound unless it is “stimulated”
by the objects. Reactivate the button by clicking on it again.
• Now, click-hold on Object 1’s Type pull-down menu, and select each entry in the list.
Strike a note repeatedly while doing so to hear the impact of each object “type” on
the string. As always, keep an eye on the String animation.

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• Adjust the Strength knob, by click-holding it, and moving your mouse vertically for

large changes, or horizontally for fine adjustments. Strike a note repeatedly while
doing so.

The three string Object dials/controls are
shown, along with the Pickup section at
the center left.

• Drag the Timbre and VeloSens arrowheads to different positions while striking a key

•
•
•
•

to audition the changes that they bring. The impact of the Variation parameter is
indicated on the charts found in the “Type” section, on page 368.
Try out each of the Gate options.
Once you’ve settled on a particular group of settings for Object 1, activate Object 2 by
pressing on the 2 button.
Change parameters for this object as desired, and note the interaction of the two
Objects with each other and the string.
Do the same for Object 3.

The Pickups
The vibration of the string is captured by two movable “pickups”. The Pickup section also
houses three sliders—one for each object.

• Reload the “plain vanilla” Setting.
• Click-hold on the Object 1 Pickup “handle”—the down arrowhead with the number

“1” on it—and drag it left/right while striking a key. You’ll note that adjustments to
the Object pickup position alters the tonal characteristics of the string.
• Adjust Object 1’s Strength control to hear things better, or adjust the tone, if desired.
You may also wish to make use of the Object’s Timbre and Variation parameters to
alter the tone. Use the table in the “Type” section, on page 368.

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• Do the same with the Pickup A and Pickup B sliders. You’ll note that changes to the

Pickup positions result in quite different String vibrations (and tonal qualities). Adjust
the Level control (directly opposite the Pickup section, on the right of Sculpture’s GUI)
to increase volume, if desired.
• Adjust the pickups of the other two Objects. Adjust each Object’s Strength, Timbre,
and Variation parameters to alter the tone. Make use of the tables in the “Type”
section, on page 368.
Note that Object 1 can only make use of the excite Types found in the first table. Object
2 can make use of any of the Types available in either table. Object 3 can only make use
of the disturb/damp Types found in the second table.
• You can disable any of the Objects at any time by pressing on their numerical

buttons (1, 2, and 3).
About Parameter Interactions…
As you’re probably discovering, each parameter has an impact on the overall tone of
the string—and more often than not—an impact on the string interaction of other
parameters.
In effect, each parameter that you introduce or make changes to, will affect the
modelled string.
This will, in turn, affect the interaction of each parameter with the modelled string.
As such, parameter settings that you have already made for, say Object 1, may need to
be adjusted when Object 2 is activated.
Generally, such adjustments won’t need to be radical, and may only involve a small
tweak to the Strength parameters, or perhaps the Pickup positions of each Object, for
example. These parameters have the greatest impact on the tone and level of the
Objects, and should be the first things you look at if the introduction of Object 2 results
in an unwanted change to the color of your sound.
You may want to “fine tune” the Objects further through use of the Timbre and Variation
controls.
Small changes—rather than radical ones—will retain the general tonal character of the
string (and Object 1), while introducing the new flavor of Object 2.
Back to it …

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Processing
From the Pickups, the signal is sent to the processing section, which consists of the
ADSR-equipped Amplitude stage, a Waveshaper (with selectable types of waveshaping
curves) and a multi-mode Filter.
These processors and the Pickup parameters are covered in “Processing” on page 373.
Feel free to experiment with these, while referring to the individual parameter
descriptions.
All elements that we’ve covered thus far exist on a per-voice basis.
Additional Processing
All voice signals coming from the Pickups are summed, and then processed by an
integrated Stereo Delay effect.
From there, the signal is sent to an EQ-like module (Body EQ), which globally simulates
the spectral shape/body response of your “instrument”. There are several “body types”
to choose from.
The resulting signal is then fed to a Level/Limiter section.
We invite you to explore all of the parameters available in these processing sections on
your own—using the “plain vanilla” patch each time. This will give you a general “feel”
for each parameter, and its impact on the sounds you hear.
All other parameters on the lower portions of the Sculpture GUI (Modulation, Morph,
Envelope, and Controller Assignments) are not part of the core synthesis engine,
although they can obviously impact upon it. We’ll discuss some tips and uses of these
parameters shortly.

Creating Basic Sounds
This section covers the creation of basic types of sounds, such as organs, basses,
guitars, and so on.
The idea here is to provide you with a starting point for your own experimentation, and
to introduce you to different approaches for tone creation with Sculpture.
As you become familiar with the synthesizer, and component modelling, you’ll find that
there are many ways to achieve an end result. By this, we mean that each “component”
of the sound can be modelled using different techniques and tools available in
Sculpture.
This flexible approach allows you to create, say a “brass” sound, in several ways—using
the Waveshaper as a major tonal element in one patch. In another brass patch, the
Filter and Body EQ can be used to emulate the same sonic component that the
Waveshaper provided in the first patch.

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A good understanding of the physical properties of the instrument that you are trying
to emulate is obviously advantageous. This type of knowledge, however, is not
common to most people, but it can be found online.
You can certainly do some detailed research, but for most sound creation tasks with
Sculpture, you can follow this general breakdown formula when creating your “string”.
How is the sound of the instrument created?
By this, we mean:
• is it a string that is vibrating and resonating in a “box” (guitar or violin, as examples)?
• is it a column of air that is vibrating in a tube (flute, trumpet)?
• is it a solid object that is struck, causing vibration (woodblock)?
• is it a hollow object that is struck, causing vibration/resonance? (drum, bell)?
What are the physical properties of the instrument?
In other words, what is it made of?
When answering this question, don’t just consider the body of the instrument. Take
into account the “string” material—nylon or steel on a guitar, or perhaps the thickness
and material of the reed in a clarinet or oboe, or a mute in a trumpet.
Is it polyphonic or monophonic?
This is a pretty significant factor, that ties into the next question. Apart from the
obvious things such as the inability to play chords on a flute, a modelled “string” will
interact with any currently active string. This, of course, can’t happen in a flute. It’s
strictly a one-note instrument.
How is it played?
Is it bowed, blown, struck, plucked?
Are there other characteristics that contribute to the sonic character of the instrument?
Examples of this are:
• changes to lip pressure and mouth position with brass and wind instruments.
• breath or mechanical noises.
• momentary pitch changes—as an example, when fingers are pressed into a
fretboard, or when a string is plucked.
• momentary tonal or level changes—such as when brass players are running out of
breath, or “fluttering” the valves.
Once you’ve mentally, or physically, written down your “list” of properties, try to
emulate each component that contributes to the sound’s character. This is what
component modelling is all about.

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Important: Before beginning, we’d like to stress that the following examples are just
that—examples. There are many ways to model each component of the sound. We
encourage you to experiment with the settings that are suggested to create your own
“versions” of patches. You’ll note that specific parameter values are rarely given, and if
they are, feel free to use another.
Just to balance the ledger a little. Subtle changes—particularly when it comes to
Keyscaling parameters—result in more “controlled” sounds. Take your time, and try
everything as you’re following these examples.
You should make use of other users patches, and the factory Settings that ship with the
synth. Close study of these will provide you with an insight into how the sound was
created. Enable/disable different parts of Sculpture to see what each does, and then
use this knowledge for your next sonic masterpiece.
Have fun and take risks—you can’t break anything!
Basses
Creating bass sounds with Sculpture is pretty straightforward.
• Load up your “plain vanilla” Setting.
• Set the Transpose parameter to 1 Oct., and play a few notes around C2. You’ll note
that the general color of an acoustic bass is already there.
• You can certainly adjust the ball on the Material Pad towards the “Nylon” entry, but
before doing so, we recommend that you change Object 1’s Type to “Pick”.
• Have a play on the keyboard, and adjust the ball position while doing so.
• Now take a look at the Strength, Variation, Timbre, and VeloSens parameters of Object
1. Adjust each in turn, to taste.
• You may also wish to adjust the Amplitude Envelope Release parameter.
• To make your bass more “woody”, adjust Object 1’s Pickup position towards the right.
At extreme positions (left or right-hand end), the bottom-end of your bass will be
gutted. Try it out!
• Now, adjust the position of Pickup A and Pickup B. As you’ll hear, you can quickly
recreate a picked acoustic or electric bass sound.
• To instantly make it a hybrid (or full-on) synthesizer bass, press the WaveShaper
button, and select from the different Types.
• Save Setting as… with new names as you go. We’re sure that you’ll come up with
several in just a few minutes. Once again, these can be used as templates for future
bass sounds.

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Bells
At a basic level, bell-like sounds are quite easy to produce with Sculpture. The creation
of truly interesting bells involves a little more effort, but the harmonic richness and
detuning during the decay/release phase makes all the difference …
• Load your “plain vanilla” patch.
• Set Object 1’s Type to Strike.
• Move the Material Pad ball to the very bottom of the pad, and place it pretty much
halfway between the Steel and Glass entries. If you play a few notes, you’ll hear that
the sound is already more “bell” like.
• Now grab the Media Loss slider, and drag it nearly all the way down. Again, play a few
notes, and you’ll hear that the release phase of the sound is considerably longer.
• Drag the Resolution slider all the way to the right.
• Drag Pickup A’s position to around halfway (0.48).
• Drag Object 1’s pickup position to a value of 0.10. We’re starting to get pretty bells
now … play a few notes.
• Now click on the Stereo Delay button to activate the Delay unit.
• Click the (Delay) sync button, and drag the Delay Base Time slider to a value of 20 ms.
• Increase the (Delay) Wet Level to 66%.
• Activate the Body EQ by pressing the Body EQ button. Ensure that the Lo Mid Hi
model is selected.
• Adjust the Low level to 0.55, the Mid to 0.32, and the Hi to 0.20.
• At this point, you will have a working bell sound—BUT—you’ll probably find that
there is a tuning issue below C3, in particular. We took this approach to
programming as the harmonics of the sound are most noticeable after all other
parameters have been set. The solution to the tuning issue primarily lies in the Inner
Loss and Stiffness KeyScaling parameters. To adjust, first select the Keyscale button,
then click-hold on the green horizontal line (within the Material Pad) for low notes, or
the blue horizontal line for high notes—and drag up/down to the desired position.
• Save Setting as… with a new name, and use it as the basis for new bell sounds, or
your next Christmas album.

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Brass
Brass instruments are notoriously difficult to recreate with electronic instruments.
Samplers do a reasonable job in the right hands, and with the right sample library, but
they lack the “organic warmth” of a real live brass player. This is a simple and generic
brass patch that can be played as a solo instrument or as a brass section.
• Load your “plain vanilla” patch.
• Set Object 1’s Type to Blow.
• Activate Object 2, and set its Type to Noise.
• Adjust the Strength of Object 1 to around 0.90.
• Set Object 1 VeloSens to around 0.30.
• Drag the Material Pad ball to a position that is diagonally between the “I” of Inner
Loss, and the “l” of the word Steel, while playing middle “C”. The sound should be
quite “brassy”.
• Now play the “E” above middle “C”, and you’ll hear a weird “mandolin meets a
telephone” kind of sound.
• Drag the Resolution slider to the left/right while playing middle “C” and a few notes
down an octave or so. You’ll discover a range of sounds that cover everything from
sitars to flutes is possible, just through manipulation of this parameter.
• Now click on the Keyscale button and—while playing up/down the keyboard—
independently adjust the Resolution slider, plus the Resolution Low/High Keyscale
sliders until the range of the keyboard that you wish to play (say an octave or so
around middle “C”) doesn’t suffer from those mandolin/phone artefacts. Oh, and
make sure that your sound is still “brassy”.
• Move Pickup A’s position to around 77%.
• Turn on the Waveshaper and select Scream as your preferred Type. Adjust the Input
Scale and Variation parameters to taste.
• Turn on the Filter. Select HiPass mode, and adjust the Cutoff, Resonance, and other
filter parameters to taste. (As a suggestion, Cutoff at 0.30 and Resonance at 0.41).
• Save Setting as… with a new name.
Please explore this patch much further. There are a great number of directions that it
could be taken in—as a muted trumpet, french horns and even sitars or flutes. The
Waveshaper has a significant impact on this sound, and this is the first place you
should look to radically alter it.
Use the Stereo Delay to create “space” and the Body EQ to cut the lows and boost the
Mids and Hi’s.
Adjust the Material Pad ball position towards the Nylon entry, select Blow as Object 2’s
Type, and then experiment with the Object 1 and 2 positions. This can also result in
different “brass” sounds.

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Flute
This can be used as the basis for most instruments in the “wind” family, including flutes,
clarinets, shakuhachi, pan pipes, and so on.
• Load your “plain vanilla” patch.
• Keyboard Mode should, theoretically at least, be set to “mono”, as flutes and other
wind instruments are monophonic. After you’ve set up the patch, experiment with
this parameter while playing, and make your choice.
• Set Object 1’s Type to Blow.
• Set Object 2’s Type to Noise.
• Set the Gate of both Objects to Always.
• Adjust Object 2’s Strength to a value of around 0.25.
• Adjust Object 1’s Velosens parameter to a value around 0.33.
• Move the Material Pad ball to a position that is pretty much at the end of the Inner
Loss entry (below the word “Nylon”).
• Play the keyboard and you should hear a flute-like sound, but with a long release—
which we obviously don’t want. Drag the Amplitude Envelope Release slider down to
around 0.99 ms.
• Pickup A should be set to a value of 1.00 (far right).
• Set Object 1’s pickup position to around 0.27.
• Set Object 2’s pickup position to around 0.57.
• Now activate the Waveshaper by pressing on the Waveshaper button, and select the
Tube-like Distortion Type.
• Play a few notes, and adjust the Waveshaper Input Scale and Variation parameters to
taste. (In. Scl = 0.16/Var 0.55, as examples).
• As you play sustained notes, you’ll probably notice a distinct lack of interesting
timbral shifts (typical of “real” flute sounds—due to changes in the player’s breath, lip
position and so on) as the note is held.
• A number of approaches can be taken to add interest to the sustained sound. These
include; using the Vibrato modulator (assigned to aftertouch, perhaps), or perhaps
recording or drawing in an Envelope, and controlling the Waveshape Inner Scale via
Velocity and/or String Media Loss. You could even use the Loop Alternate Sustain
Mode. Feel free to experiment!
• Save Setting as… with a new name.

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Guitar
Guitar, lute, mandolin, and other plucked type instruments, including harps, can be
created from this basic patch.
• Load your “plain vanilla” patch.
• Set the Voices parameter to a value of 6—there’s only 6 strings on a guitar. Obviously,
pick 7 for a banjo, as many as possible for a harp.
• Set Object 1’s Type to Impulse, if it’s not already there.
• Activate Object 2 and set its Type to Pick.
• Now move Pickup A’s position to the extreme right.
• Move Object 2’s Pickup position to a value of 0.14.
• Activate the Body EQ, and select one of the Guitar models.
• Adjust the various Body EQ parameters. These have a major impact on the overall
brightness and tone of your guitar sound. (Suggestions—Guitar 2, Int—0.46, Shift—
0.38 and Stretch—0.20).
• Finestructure should be at a value around 0.30 to 0.35—but use your ears to judge
this. Remember that an increase in the Finestructure value results in a greater load
on the computer CPU.
• Click-hold on the Stereo Pickup button, and drag your mouse upwards—to increase
the perception of stereo width (a value around the 10 o’clock/2 o’clock mark is nice).
• Activate the Filter, and select Lo Pass mode.
• Adjust the Cutoff and Resonance parameters to taste. (suggestion—both at 0.81).
• Adjust the Tension Mod slider upwards, and play the keyboard to see how the
momentary detuning effect caused by this parameter affects the sound. Set it to an
appropriate amount.
• Set the Level Limiter to Both.
• Save Setting as… with a new name.
You may have noticed that we departed slightly from the signal path of the Core
synthesis engine in the creation of this Setting. The reason for this is the major impact
that the Body EQ model has on the sound.
In some cases, like this one, it may be better to work slightly out of sequence, rather
than strictly follow the signal flow.
Obviously, this is but one “guitar”. You can make use of the Object Strength, Variation,
and Timbre parameters, not to mention repositioning the Material Pad ball to create a
completely different “tone” to your guitar.
For quick and easy mandolins, make use of the Stereo Delay (or Vibrato) to emulate the
double-strike picking that is associated with the instrument.

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Organ
Organ sounds are amongst the easiest and quickest sounds to emulate in Sculpture as
they have no release phase. This simplifies programming as there is no real need to set
up Keyscaling parameters to create the basic tone. You may, however, wish to do so at
a later stage—for modulation routing or specific sound design purposes.
We encourage you to play notes/chords while you are adjusting parameters. This way,
you can hear what each parameter is doing to the sound.
• Load your “plain vanilla” patch.
•
•
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•
•
•
•
•
•
•

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(Object 1’s Type should be set to Impulse. If it isn’t, change it now.)
Increase the Voices parameter to a value of 8 (or higher if you wish).
Move the Material Pad ball into the top left corner.
Activate Object 2 by pressing the 2 button.
Change Object 2’s Type to Bow.
Set Object 2’s Gate Time to Always.
Pull the Release slider of the Amplitude Envelope all the way down.
Play a “C” chord, and you’ll hear a “flute-like” sound.
Drag Pickup A to the extreme right.
Play a “C” chord, and you’ll hear a cheesy “organ” sound. As you can see Pickup A’s
position has a significant impact on the overall sonic character of the sound.
Grab Object 2’s Pickup, and move it around while holding down the “C” chord. Once
you’ve found a position that meets your “that sounds like an organ” criteria, release
the Object pickup.
Now, very slightly adjust Object 2’s Timbre parameter upwards.
Adjust Object 2’s Variation parameter slightly downwards, and upwards, until you
find a tone that you like.
You may, at this point, want to move the Object 2 Pickup parameter to another
position. Hold down a chord while doing so.
Further “tweaks” can be made to the Variation and Timbre parameters of Object 2.
To introduce a little “key click”, change Object 1’s Type to Strike, and adjust its Strength
and Timbre.
To add a little of that “detuned organ” vibe, set the Warmth parameter between 0.150
to 0.200.

At this point, you should have a basic organ tone. Save Setting as… with a new name.
You can use this as the basis for your next organ patch.
You’ll probably notice some intermodulations that are introduced when you’re playing
chords. Apart from the pitch differences between notes in the chord, this is a result of
the interactions between each voice being produced by Sculpture. These slight
variations between each voice (or string, if you will), and their harmonic interaction
with each other are not dissimilar to the harmonic interactions of a violin section in an
orchestra—even when playing identical lines.

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Percussion
Percussive sounds, such as drums, tend to share a similar type of envelope. They
contain a “strike” element, where most of the sonic character is exhibited, followed by a
short decay phase. The release phase will vary—dependent on the instrument itself (a
snare drum as opposed to a woodblock), and the ambient space it is placed in—a
cavern, a bathroom and so on.
• Load your “plain vanilla” patch.
• Set Object 1’s Type to Strike.
• Activate Object 2, and set its Type to Disturb 2.
• Set Object 2’s Gate to Always.
• Object 1’s Strength should be about 0.84.
• Object 2’s Strength should be about 0.34.
• Adjust the Media Loss value upwards/downwards while playing to hear its impact.
Find a suitable setting.
• Similarly, you can alter the Material Pad ball position—although its impact on the
overall tone of the sound is heavily reliant on the Media Loss value.
• Activate the Body EQ and/or Filter, and adjust to your heart’s desire.
• Save Setting as… with a new name.
This sound can be used as the starting point for a vast range of percussive sounds—
including drums, blocks, “industrial” percussion and even rhythmic sequenced synth
sounds.
Just adjust the ball position in the Material Pad, and alter the Media Loss slider position.
Solo Strings—Cello Bowed
Solo stringed instruments, such as violins and cellos, that are played with a bow can be
created in much the same way. This sound can also be played polyphonically.
• Load your “plain vanilla” patch.
• Set Transpose to −1 Oct.
• Set Object 1’s Type to Bow.
• Play the lower half of your MIDI keyboard, and you’ll hear a viola/cello like sound.
This could obviously be improved.
• Set the Object 1 Velosens slider to match your playing style and that of the music, as
you’re playing the keyboard. Adjust later, if desired.
• Grab the Tension Mod slider, and move it slightly upwards, so that the arrowhead
covers the “D”. This emulates the momentary detuning effect of the bow stretching
the string.
• Move Pickup A to a position around 0.90.
• Move Object 1’s Pickup position to a value around 0.48.
• Activate the Body EQ, and select the Violin 1 model.
• Set the Body EQ parameters as follows: Int—0.73, Shift—+1.00, Stretch—+1.00.
• Adjust the Fine Structure to taste.

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• Click-hold on the Stereo Pickup dial, and drag upwards until the 10.30/1.30 positions

are reached.
• Set the Level Limiter to both.
• Save Setting as… with a new name.

We encourage you to set up your own modulations for this sound. The most common
thing that springs to mind is the introduction of vibrato into the sound after a short
period.
The creation of higher pitched solo string instruments is much the same as the
example above but special attention must be given to ALL Keyscaling parameters.
The Body EQ also has a large bearing on the upper octaves, so take care with its
parameters.
Setting Bonus:
Simply change Object 1’s Type to “Pick”, and you’ll have a round and rubbery synth bass
sound in the lower octaves, and a passable “harp” across the rest of the keyboard.
Synthesizers
One of Sculpture’s great strengths is the creation of endlessly evolving pad and
atmospheric sounds. It can also easily do fat synth basses (which you hopefully
discovered while following the “Basses” section, on page 416), powerful leads and other
types of typical synthesizer sounds.
Sculpture has an advantage over traditional synthesizers as its core synthesis engine
produces a wider variety of basic tones, and these tones have an “organic” quality and
richness to them.
Basic Pad
• Load your “plain vanilla” patch.
• Set the Voices parameter to 16.
• Set Object 1’s Type to Bow.
• Set Object 2’s Type to Bow Wide.
• Grab the Material Pad ball, and position it at the extreme left of the Pad, exactly
halfway between the top/bottom—on the “Material” line.
• Play a “C” chord (middle “C”), and you’ll hear a pad sound.
• Move Pickup A to a position around 0.75, and the pad will become a little sweeter.
• Move Object 1’s position to a value of 0.84.
• Move Object 2’s position to a value of 0.34.
• As a final step, click on the Points icon that features five dots in the Morph Pad
section.
• Set the “Int” slider in the Morph Pad Randomize section to a value of say 25%.
• Click once on the Morph “Rnd” button.
• Save Setting as… with a new name—say “vanilla_pad”.

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We’ll be using this basic pad sound for the following examples. Don’t be shy about
doctoring the “vanilla pad”—anything goes, so make use of the Filter, the Delay, EQ,
and Waveshaper to create new sounds.
Evolver
• Load your “vanilla_pad” patch.
• Click on the LFO 1 tab at the bottom of the GUI.
• Press the 1 button, and play the keyboard. Not much difference, there, huh?
• Now, click-hold on the Min/Max sliders, and drag left and right, while holding down a
chord. Finally settle on a value of 0.15.
• Click-hold on the Target (Pitch), and select Object 1 Strength. You’ll hear a “fluttering”
sound.
• Now click on the “sync” button, and adjust the Rate knob to a value of 1/8t.
• Activate the second LFO 1 object by pressing the 1 button, and select Object 1
Position as the Target.
• If you play the keyboard, there’s not much that’s different.
• Set the via option of the second LFO 1 object to Velocity.
• Play the keyboard at different velocities, and you’ll hear some shifting of the Object 1
Pickup position … and now, to make it interesting …
• Change the Waveform to Sample&Hold, and play the keyboard at different velocities.
If you’ve got a sustain pedal, use it. Listen to the endlessly evolving sound.
• You may like to experiment with the song tempo and LFO Rate.
• You may want to alter the Stereo Pickup value, and introduce LFO 2 or the other
modulators.
Morpher
• Load your “vanilla_pad” patch.
• Click on the R(ecord) button in the Morph Trigger section.
• Play a chord on the keyboard, and drag the Morph Pad ball in a circle.
• Once you’re done, press the R(ecord) button.
• Now change the Morph Mode to Env only, and you should see your Morph circle.
• Play the keyboard. There’s your morphed pad!
• Feel free to adjust the Morph Envelope parameters.
Remember when you were asked to use the Morph Points, Intensity, and Rnd
parameters while setting up the “vanilla_pad”?
This was to ensure that there were several Morph points already available for your
morphing pleasure.
You can, if you wish, retain the path of your morphed pad, and continue to press the
RND button and adjust the Intensity slider for an endless variety of sounds.

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Modulations
The modulation options can be very important for the emulation of acoustic
instruments. As a simple example, the introduction of vibrato into a trumpet sound
over time.
Many “classic” synthesizer sounds also rely as much on modulation as they do on the
basic sound source components—the VCO, VCF, and VCA.
Here’s a number of quick modulation tips …
• Let’s say you want to modulate the timbre of Object 2, with the LFO, for example. To
do so, click on the LFO 1 or 2 Tab, press the 1 or 2 button, select the desired Source/
Target and value. That’s it!
• To control any modulation with an external controller, such as your keyboard’s
modulation wheel, set the desired “Via” parameter to CtrlA or CtrlB respectively. By
default, the Mod Wheel is set to CtrlA.
• The “Bouncing” damp Type available to Object 3 affects the sound in a very
interesting way, but it cannot be run synchronously with the song tempo. To create a
similar effect to the “Bouncing” Object—but in-sync, you could use a “Disturb” object
Type, and move it by modulating its vertical position (Timbre) with an LFO.
“Breath” control is available to all users of Sculpture, even if you don’t own a breath
controller. To do so, record breath controller modulations into the recordable envelopes
(using a mod wheel or other controller), and then reassign the recorded modulation
path (use the CtrlEnv 1 and/or 2 parameters) with each NoteOn.

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Programming: In Depth
This tutorial explains how you can program sounds with Sculpture from scratch. Based
on Sculpture’s string model, you’ll learn how to use the individual sound shaping
parameters in order to recreate the physical properties of an instrument in detail.
Note: You will find the settings for these tutorials in the Tutorial Settings folder in the
settings menu (in the head of the Sculpture Plug-in window).

Programming Electric Basses with Sculpture
We’ll be concentrating on a single instrument type: the electric bass, including all of its
important variations and articulations. The physical nature of electric basses is not as
complex—and encumbered with acoustic issues—as is the case for many acoustic
instruments. This instrument is therefore an excellent choice for our sound
programming tutorial, the goal of which is to acquaint you with the art of using
Sculpture to accurately reproduce detailed sounds.
In order to build a bass (including all components) in Sculpture, it is necessary to
understand the basic, physical process of sound production within the instrument.
Before we look at the practical programming process within Sculpture, you’ll find detailed information on the construction of electric basses in the next section.
The Most Important Aspects of Electric Basses
In general, the electric bass has four strings. The lowest string is usually tuned to E 0 or
E (MIDI note number 28). The strings above the low E are tuned in fourths, thus A, D,
and G. You can, of course, find basses today that have five, six, and even more strings.
As Sculpture has no tonal limits, this is of little concern to us.
What is much more important for sound programming is the overtone content of the
bass sound. This depends primarily on the qualities of the strings.
• Round wound strings: a very fine wire is wound around a steel cable core which
results in a wiry, metallic sound that’s full of overtones.
• Flat wound strings: (which are much less popular today) the fine wire wrapping is
ground down or polished smooth, and the sound has far fewer overtones in
comparison.
In contrast to guitar strings, the structure and workmanship are the same for all strings
in a set. Sets combining wound and non-wound strings do not exist.
The relationship between string length and string tension has a significant impact on
the overtone content. Disregarding basses that can be adjusted to different scale
lengths (different vibrating string lengths), the actual playing position that is used plays
an important role. When you play D at the tenth fret on the low E string, it sounds more
muffled than the same pitch played on the open D string.

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The number of frets differs from bass to bass and depends on the scale length. We
don’t need to worry about pitches higher than a single ledger line C; the actual functional range of this instrument is primarily in its two lower octaves—between E 0 and E 2.
We should also mention the fretless electric bass. Like all instruments of this type, it is
freely tunable and possesses a distinctive, individual sound. Over the course of this
tutorial you will discover how to program this type of instrument in Sculpture.
There are three types of articulations that we will be discussing:
• Fingered: the strings are played with the alternating index and middle fingers.
• Picked: the strings are played with a pick.
• Thumbed/Slapped: the strings are either played with the (side of the) thumb on the
fingerboard or plucked strongly with the fingers.
The vibration of the strings is captured by an electromagnetic pickup. When the string
is vibrating, its steel core affects the magnetic field. The pickups are almost always
found some distance to the side, nearer to the bridge and stop tailpiece. There are different pickup concepts for electric basses and often two or more pickups are combined to make the “sound”. Although we can’t take these things into detailed
consideration at this point, there is a rule of thumb that applies:
The further you move the pickup towards the middle of the string, the “bassier” the
sound will be and the more “hollow” it will sound. The further you move the pickup
towards the end of the string, the more the sound’s overtone content will increase,
becoming more dense and compact. The sound will have more mid-range frequencies
or “buzz” and less bass. If the pickup is positioned at the very end of the string, the
sound becomes very thin. We can find parallels to the actual playing position of a real
string here: If you play more towards the middle of the string, you’ll get a smooth,
even, and powerful sound which contains limited harmonic denseness (overtones),
from time to time. If the string is played at the bridge, the sound develops a nasal
twang and features more “buzz” and more overtones.
Now to the body of the instrument, and its resonant properties. Almost all electric
basses have a steel rod running through the neck, to strengthen it, and a body made of
solid wood. This construction allows the strings to vibrate relatively freely (sustain),
even though very little direct sound is generated. The pickups and the amplifier and
speaker systems are responsible for the actual sound of the instrument.
The acoustic interaction between body, strings, and external sound sources is much
less complex than with pure acoustic instruments.

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The vibration of the strings is, of course, naturally hampered by several physical
factors: the radius of motion of the string (antenode) is impeded by the left bridge or
by the first fret that’s pressed down upon (and the frets in between). This can lead to
the development of overtones which can take the form of anything from a slight humming or buzzing to a strong scraping or scratching sound.
In addition, factors such as the material properties of the strings and the instrument, as
well as the softness of your fingertips also serve to dampen the vibration of the string
Programming a Basic Bass
In the following section we will discuss the programming of a basic bass sound. This
will serve as the foundation for the different bass sounds which we will be creating
thereafter.
Heed the following steps to create the proper working environment for design
of your own “homemade bass”:
1 Make sure the range from C 0 to C 3 is available on your keyboard by either;
transposing your master keyboard, or using the Transpose function in Logic’s Region
Parameter box.
Note: You can, of course, transpose sounds within Sculpture but this isn’t the best
solution in this case. The reason is that sounds would not be compatible with MIDI
Regions in which note number 60 as middle C is considered to be the measure of all
things.
2 Choose the default setting within Sculpture.
In order to recreate the sound characteristics of a typical bass instrument:
1 Set the Attack value of the Amplitude Envelope to its minimum value (0.00 ms). You’ll
find the Attack slider just to the right of the Material Pad.
2 Shorten the Release time of the Amplitude Envelope to a value between 4 and 5 ms.
Play a key on your keyboard. The note should stop abruptly when you release the key
and should be free of artefacts (a digital crackle or snap). If you encounter any artefacts during the course of this tutorial, please carefully increase the Release time.
3 Play some sustained notes in the range above E 0. These will die away (too) quickly.
Correct this quick die-out with the Media Loss parameter by pulling the slider located to
the left of the Material Pad almost all the way down to the bottom. For your
information: the low E string on a high quality bass can sound for over a minute!
Our basic bass should simulate a fingered articulation, which means the sound is created by striking the strings with the fingers.
4 Choose the Pick entry from the Type menu of Object 1.
Don’t be confused by the name of the Object Type: despite its name, this model is
appropriate for simulating the playing of strings with your fingers.

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Play some notes in the lower range. you’ll note that the sound is very muffled, hollow,
and distorted. Before we adjust further parameters in Object 1, we need to set the position of the Pickup.

This is accomplished in Sculpture’s Pickup window located to the left of the Material
Pad (see the GUI detail above). You’ll find three trapezoidally shaped sliders, representing Objects 1 to 3. Both of the transparent bell-shaped curves help you to visualize the
position and width of Pickup A and Pickup B.
On electric basses, the pickups are found quite a way off to the side and near the
bridge. We also assume that our bass only has a single pickup.
We can simulate the behavior of a single pickup by placing both Pickups at exactly the
same position.
5 Keep an eye on the Help Tag, and drag Pickup B to the exact position of Pickup A. The
two thin orange lines should overlap perfectly.
Note: Make sure that the Invert switch to the lower left of the Pickup window isn’t
turned on, as this would cause the Pickups to completely cancel each other out.
6 As a suitable value for our example, set both Pickups to 0.10.
It’s now time to determine the playing position:
7 Grab the Object 1 slider in the Pickup window and move it in a horizontal direction.
Play the keyboard while doing so, to hear the changes it makes.
8 You’ll quickly realize that you can only achieve a precise, crisp sound when relatively far
away from the middle of the string. Move Object 1 closer to the Pickup (position 0.15—
see GUI detail below).

9 The low notes are still distorted. You can remedy this by adjusting the Level dial found
to the right of the Amplitude Envelope. Set a value of −10 dB.
Although you can already recognize the sound of an electric bass, it doesn’t sound
“wiry” enough yet. Let us now direct our attention to the bass strings themselves.

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In order to recreate the material properties of a set of round wound strings:
1 Move the ball in the Material Pad up and down at the left edge. Pay attention to how
the overtones react. Move the ball to the lower left hand corner. The sound should
vaguely remind you of the sound of a low piano string. As the overtones sustain too
long, the tone sounds somewhat unnatural.
2 Move the ball upwards until you hear an acceptable sound. As an example, we recommend the following position:

Note: In general, a splaying of the overtones in low wound strings is typical. You can
recognize it by the slightly impure, metallic sound. This occurs because the partials
(overtones) are not exact whole number multiples of the fundamental frequency, but
rather they are shifted somewhat higher. An example of this effect in the real world of
electro-acoustic instruments are the low strings on a Yamaha CP70. Although we don’t
want to take it that far, our bass model does need a small amount of this effect.
How to splay overtones in Sculpture:
1 Move the ball in the Material Pad gradually to the right. The sound takes on a more
impure and bell-like character.
2 To realistically simulate the splaying of overtones, we recommend the following
example setting:

The vibration of a bass string does not occur in a vacuum. The antenode of the string
frequently encounters the natural, physical limitations of the instrument. This is heard
as the typical buzzing and rattling that occurs when the strings touch the frets.

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We can simulate these disturbing elements with Object 2:
1 Activate Object 2, and choose the Bouncing Type menu item. The sound should now
vaguely remind you of a mandolin tremolo. This is way too strong an effect for the kind
of sound we’re after.
2 Move Object 2 all the way to the right (a value of 1.00).
3 Experiment with Object 2’s parameters. A discrete and realistic result can be achieved
with the following parameter values: Strength: 0.33, Timbre: −1.00 and Variation: −0.69.
Play some low notes, and you’ll find that, once again, the overtones sustain a little too
long, reminding us of the lowest strings on a piano.
We’ll use Object 3 to dampen these overtones:
1 Activate Object 3 and choose the Damp Type menu option.
2 Also move Object 3 all the way to the right (value 1.00).
3 Set the Strength parameter to 0.18.
Note: Experiment with how the Strength parameter of Object 3 interacts with the Inner
Loss Material Pad parameter. The higher the Inner Loss value, the smaller the Strength
value can be, and vice versa.
In order to more realistically replicate the different tonal ranges of the bass, we’ll use
Sculpture’s scaling function.

Firstly, we need to activate the scaling function display:
Click on the Keyscale button located at the bottom of the Material Pad. The key-scaling
below C3 is displayed in green, the range above in light blue. The Material Pad with its
key-scaling parameters activated:

Note: The most relevant performance range for basses is found exclusively below C3.
For this reason, we’ll be using the green sliders to set the actual timbre of the sound.
The “primary” sliders found around the ring determine the timbre of the sound above
C3. For the moment, we’ll disregard the blue sliders (which control high key-scaling)
and simply set them to the same positions as the main sliders.

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Once activated, the key-scaling function is used to adjust the timbre of the sound,
independent of pitch. To do this, we’ll initially employ the Resolution parameter. This is
normally used to set the balance between DSP load and sound quality. As the overtone spectrum is reduced at low Resolution values, this parameter can also be used to
shape the sound.
To use the Resolution parameter to control the timbre, independent of pitch:
1 Play some notes at the higher end of the bass’s range (around C2), then move the Resolution slider all the way to the right and then gradually back towards the left.
2 You can hear how the sound loses overtones, yet simultaneously becomes louder. At
low Resolution values, an inharmonic metallic “rattling” is heard in the sound.
3 Increase the Resolution value until the metallic rattling disappears. We recommend setting the slider to the following position:

4 Play some notes in the bottom range (around E 0). You’ll note that the sound is quite
muffled and vintage-like. Move the green Low Keyscale slider (found below the main
Resolution slider) all the way to the right; the low range should now sound a little more
“wiry”.
With most stringed instruments, the overtone content decreases as the pitch becomes
higher. Strictly speaking, this is only true of open strings, and even then, in a limited
sense. If the strings are fingered, the length of the string is shortened (especially in the
high register) and the effect becomes more significant.
This is why we use the Inner Loss parameter to scale the overtone content,
dependent on pitch:
1 Move the Material Pad ball above the words Inner Loss. Try to move the ball solely in a
vertical direction, in order to maintain a constant Stiffness value.
2 Grab the green line next to the ball, and pull it towards the bottom until the small
green diamond is located directly above the word Steel.

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When playing, you’ll recognize the smooth transition that takes place between the
wiry, overtone-rich sound at the bottom end and the extremely dampened sound in
the upper register. This (exaggerated) setting was chosen to clearly demonstrate the
scaling principle in stringed instruments. In order to achieve an authentic sound and
timbre, we recommend the following setting:

In basses in particular, low notes sustain far longer than the high notes. Sculpture
allows you to authentically and convincingly simulate this behavior with the Media Loss
parameter.
To use the Media Loss parameter to scale the fading phase of the note, dependent on pitch:
1 Play a few held notes in the range around C2 and above. You’ll hear that these notes
die out much too slowly. Move the Media Loss slider up until this range begins to fade
out quickly enough. The downside is that the lower notes now die out too quickly!
2 Pull the green Media Loss Key Scale slider down until the fade-out phase of the lower
range is sufficiently long enough.
3 Compare your results with these recommended values:

We’ve now completed our task for this section, and created a basic bass that’s articulated with your fingers. Save this as “E-Bass Fingered Basic”. In the following sections we’ll
be using this basic bass as a foundation for the construction of further basses.

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Modifying the Frequency Spectrum of our Basic Bass
The scope for sound design, by altering the frequency spectrum of electromagnetic
instruments, is far more flexible than that offered by acoustic instruments. In addition
to the number of pickups, the choice of amplifier, the equalization setting within the
amplifier and, last but not least, the physical properties of the speakers and their
enclosing cabinet also play a major role.
The central features of our electric bass sound are complete, but the sound can be
improved upon by paying close attention to some details. Here are a few general
suggestions:
• Vary the position of the Pickups. Try placing each of them in different positions. This
will cancel out certain frequencies, and others will be summed together.
• Try turning on the Invert switch, even though this effect is not typical for electric basses.
• What is typical for bass sounds is the placement of the Pickups in the outer left third
of the string model. The farther you move them to the left, the thinner and more
nasal the sound will become.
• Shifting Object 1 will have a similar effect. Try different combinations here as well.
The Body EQ is ideal for giving the bass sound that final, finishing touch. Our electric
bass sound could be a little less smooth, and be a bit more precise in its attack phase.
Bassists like to use the terms “drier” and “more bite” to describe this phenomenon.
To alter the frequency spectrum of our basic bass with the Body EQ:
1 Load the “E-Bass Fingered Basic” Setting.
2 Select the standard Lo Mid Hi Body EQ model.
3 Reduce the low bass frequencies by setting the Low dial to a value of −0.30.
4 Boost the mid-range frequencies substantially by setting the Mid dial to a value of 0.50.
Grab the Mid frequency dial and drag it to the graphical display of the frequency spectrum (to the right), and pull downwards. Experiment a little, then choose a value of
0.26.
5 You’ll probably find that the boosting of the low mid frequencies is a little too strong at
this point, so please return the Mid value to 0.30 (see the GUI detail below).

6 The sound could stand to be a little more “wiry”, so set the High dial to a value of 0.30.
7 To finish off, set the Level dial (to the right of the Amplitude Envelope) to a value of
−3 dB. The sound is now as loud as possible, without the low notes distorting.

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8 Save this sound Setting, as we’ll need it for further modifications later. Please name it
“E-Bass Fingered Basic EQ1”.
Pick Bass
Our basic bass is played with the fingers. In the following example we will simulate
playing the strings with a pick. We’ll also use the Pick Object Type for this sound. We’ll
make use of the Timbre parameter to adjust the relationship between the speed and
intensity at which the string is struck. We’ll also make use of the Variation parameter to
define the virtual material density or hardness of the pick.
If we imagine the fingers to be very soft picks, it makes sense to alter the Pick parameters so that a hard plastic pick is the outcome.
To simulate playing with a pick:
1 Please load the “E-Bass Fingered Basic” Setting.
2 Set the Timbre parameter of Object 1 to its maximum value of 1.00. You’ll note that the
attack is now stronger.
3 Try several different Variation settings to get a feel for the material qualities of the pick.
Note: Not all positions will deliver usable results for the entire range of the instrument.
4 You’ll get a consistent, working setting for the two octaves above E 0 with the following parameter settings: Position 0.17 (pickup window), Strength 1.00 (maximum),
Timbre 0.90, Variation 0.56.
When these settings are used, you’ll find that the sound has become softer and
very thin. In fact, it’s somewhat reminiscent of a clavinet.
We’ll compensate for this side effect with the Body EQ.
1 Activate the Body EQ and add a healthy portion of bottom end to the sound by setting the Low parameter to 0.60. Mid should be set to 0.33.
2 Set the High dial to −0.45 as the sound is now so bright that rolling off a few of the
highs can’t hurt.
3 Now bring the volume into line. If you adjust the Level dial to 2.5 dB, nothing should be
distorting. If this isn’t the case, try reducing some more of the bottom end with the Low
dial.
4 Save this setting as “Pick Open Roundwound”.

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Damping
Playing with a pick is often combined with a damping technique that employs the ball
of the thumb. The right hand, which also holds the pick, should physically lay on top of
the strings at the bridge. This technique results in the sound having less overtone
content but become more percussive and “punchy” at the same time. You can variably
control the timbre of the sound through the angle and pressure of your hand while
playing.
Object 3 will be used to emulate the virtual ball of the thumb in this example. The
Timbre parameter determines the kind of damping that occurs. Variation dictates the
length of the string section that is being dampened.
To achieve this effect, proceed as follows:
1 Set the Object 3 Type to Damp.
2 Set Object 3’s Strength parameter to 0.50.
3 Move Object 3 a little bit to the right in the Pickup window (to position 0.95) to simulate the width and position of the ball of the thumb lying on the bridge.
4 Set Timbre to its minimum value (−1.00) to achieve a very soft damping effect.
5 Set the Variation parameter to its maximum value of 1.00.
You’ll note a metallic ringing that occurs during the attack phase still can be heard in
the octave above E0.
To suppress it:
Note: Move the small green diamond on the Material Pad to a position directly under
the ball. In doing so, you’ve just increased the Inner Loss value for the low key range.
Note: In order to place the diamond exactly under the ball, you can also click it while
pressing the Option key.
6 Save this setting as “Pick Bass Half muted”.

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Harmonics
Harmonics are single partials (overtones) of the overall sound. They can be heard by
damping certain points along the string. This is done by lightly laying the fingers of the
left hand (assuming a right-handed bass player) on the string (not pressing down)
before the note is articulated. The first overtone, the octave, is achieved by placing your
finger at the exact middle of the string, in effect separating the string into two halves.
The next overtone is the fifth above the octave and the position of your finger should
divide the string into a ratio of one-third to two-thirds. The next highest overtone
separates the string into proportions of one-quarter to three-quarters and so on.
To simulate fingers lightly touching the strings:
1 Object 3 is used as a damper. Select the Damp Type.
2 Adjust Object 3’s Timbre parameter to its maximum value of 1.00.
3 Variation must be set to its initial value of 0.00. Simply click on the slider while holding
down the Option key to do so.
4 Move Object 3 to the exact middle (0.50) of the Pickup window. Play the keyboard, and
you’ll hear the first overtone as a harmonic.
5 While playing, very slowly move Object 3 towards the left of the Pickup window. In
doing this, you are effectively “scrolling” through the overtone series, so to speak.
6 Save this setting as “Flageolet Xmple”.
Vintage Flat Wound Pick Bass
Now, in just a few easy steps, we would like to transform our pick bass into a vintage
pick bass with flat wound strings. This bass sound is typical for the funk and soul music
of the 70’s, but you’ll also find it in many easy listening arrangements.
Proceed as follows:
1 First load the “Pick Bass Half muted” Setting.
2 Drag the Material Pad ball upwards and the sound becomes more muffled. Here are the
values we recommend:

3 Increase the Object 3 Strength parameter to 0.70. The result is a muted pick bass with
flat wound strings.

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Tip: If you turn Object 3 off, you’ll hear a sound that is reminiscent of a 1970’s Fender
Precision Bass.
4 Save this Setting as “Flatwound Pick Damped”.
To get a nice percussive sound a la “Bert Kaempfert”, proceed as follows:
1 Turn Object 3 back on.
2 Move both Pickups a little to the left (position 0.08).
3 Our virtual pick (Object 1) can also be moved a little further to the outside (position
0.10).
4 We can add the icing to the cake with the Body EQ. Turn the Low dial to its maximum
value (1.00).

5 To remove the smacking in the attack phase, we’ll use the graphical display to choose a
value of 0.48 for the Body EQ Mid frequency, then use the dial to increase this value to
0.51. Option-click on the Body EQ High parameter to set it to a value of 0.00.
6 Save this setting as “Easy Listening Pick Bass”.
Slap Bass
We’re actually dealing with two different articulations here. The low notes originate
when the thumb literally slaps the strings on the upper part of the fingerboard. The
high notes are produced when the strings are strongly plucked or “popped” with the
fingers. This is achieved by hooking a finger under the string, pulling it away from the
instrument and then allowing it to slap back onto the fingerboard. In conjunction,
these articulation methods make up the typically aggressive and overtone-rich “slap
bass” sound.
To simulate a slap bass:
1 Load the “E-Bass Fingered Basic EQ1” Setting.
2 Turn off the Body EQ.
3 Also turn off Object 2 and Object 3, for now.
As the basic sound of a slap bass is brighter than a standard fingered bass, we need to
adjust some Material Pad settings:
4 Return the Low Key Scale parameter to its initial value by clicking on the little green triangle, while holding down the Option key.

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5 Move the ball down a little, and the sound becomes more wiry. The ball should now be
located directly above the word “Steel” on the horizontal axis.

From the models at our disposal, Strike is the most suitable for simulating a thumb physically striking the strings from above. This model is not, however, as appropriate for
the slapped (popped) strings. It makes the most sense to choose the Pick model for this
purpose.
6 To be safe, turn the Level dial to −25 dB.
7 Select the Pick model for Object 1.
8 Move Object 1 to position 0.90 in the Pickup window. This position corresponds to a
playing position above or on the fingerboard.
Note: Given its universal concept, Sculpture will not react exactly like a bass, where one
would tend to play in the middle of the string on the upper part of the fingerboard. Try
moving Object 1 to this position and see how it sounds. You’ll find that the sound is
little too smooth.
Setting the parameters for Object 1:
1 Set Timbre to a value of 0.38; this corresponds to a rapid attack.
2 Set the Strength parameter to 0.53.
3 Set the Variation parameter to −0.69; this defines the softer “material” that constitutes
the fleshy part on the side of your slapping thumb.
You’re probably familiar with the sound of low notes when played with your thumb.
What’s missing, thus far, is the typical bright rattling that is created when the string strikes the fingerboard. We’ll use Object 2 to this end, and select the Bound Type menu
option. Bound limits the antenode of the string in exactly the same way as the fingerboard on a real electric bass.
Let’s review the functions of these parameters: Timbre determines the angle of the
obstacle to the string while Variation defines the type and degree of reflection.

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Let’s adjust Object 2’s parameters to the following settings:
1 Set Timbre to 0.39. This corresponds to a fingerboard that runs almost parallel to the
string.
2 Set the Strength parameter to 0.33.
Note: Try some higher values as well. You’ll see that the sound becomes softer and
softer until it’s completely dampened by the obstacle.
3 An appropriate value for Variation is 0.64. Despite the overtone-rich reflection, the
string can still vibrate freely.
Note: Try some negative values: you’ll see that the reflections can no longer develop in
an unhindered fashion.
4 Set the Level dial to −3 dB; the Bound obstacle has made the sound softer.
5 The sound is still too smooth for a real slap bass; so let us direct our attention to the
Body EQ again. Switch it on, and adjust the parameters to the following
settings: Low = 0.25, Mid Frequency = 0.59, Mid = 0.43, High = 0.51.
6 Save this sound as “Slap Bass Basic#1”.
Fretless Bass
With the exception of shared playing techniques, the fretless bass differs from a
“normal” bass through its buzzing, singing sound. As the frets on the fingerboard of a
standard bass function as a collection of “mini-bridges” and allow the string to vibrate
in an unobstructed fashion, the direct “collision” of the string’s antenode with the
fingerboard on a fretless bass is responsible for its “typical” sound. The string length on
a fretless bass is markedly shorter than that of an acoustic double bass. The upshot of
this is that a “controlled buzzing” is produced, even when played with a weak attack.
This “buzzing” can be consistently reproduced in the high register, even on fretless
basses that have very short string lengths. The use of the comparatively soft tip of your
finger, instead of a hard, metallic fret, to divide or shorten the string also plays a role.
This is how you program a fretless bass:
1 Load the “E-Bass Fingered Basic EQ1” Setting.
2 Turn Object 3 off. We’ll come back to it later.
3 Choose the Object 2 Disturb Type menu option.
Tip: The Disturb model functions in the following way: the Timbre parameter
determines how far the string is deflected from its resting position by the obstacle.
Positive values precipitate no deflection of the vibration from its resting position.
Variation defines the length of the string section that is “disturbed”: positive values
correspond to a longer section of string, negative values to a correspondingly shorter
string section.

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4 Adjust Object 2’s parameters to the following values: Strength 0.14, Timbre −0.05,
Variation −1.00.
5 Object 2’s Pickup position remains at the far right; please enter a value of 0.99. You’ll
note that the range between C2 and C3 already sounds quite acceptable, but the buzzing in the lower notes is still too strong. It is somewhat sitar-like, so keep this disturb
model in mind when it comes to creating a “home-spun” sitar.
6 Try different settings for the Strength parameter for both the higher and lower playing
ranges. You’ll see that, at best, only a compromise is possible. The buzzing is either too
loud in the low range or not present enough in the high range.
Obviously, we need to scale the effect over the relevant tonal range. Unlike the parameters for the string, Objects 1–3 don’t have a directly adressable key scaling function.
We’ve got to be a little clever here. Both LFOs offer a key scaling function. As we don’t
want the buzzing to be modulated by a periodic oscillation, we need to reduce the LFO
speed to “infinitely slow” or 0. In this way, we can deactivate the LFO itself but still use
its modulation matrix.
7 Activate LFO2 by clicking on the LFO2 button at the bottom left, and set the Rate dial
to a value of 0.00 Hz.
8 Click on the button marked “1” (next to the Rate slider, to the upper right) to activate
the first modulation target.

9 Choose Object2 Strength as the Target parameter.
10 Select the KeyScale entry in the via column.
11 Move the lower slider labeled Amt (amount) to the right while you are playing. You’ll
quickly realize that the “singing” buzzing fades out in the lower range, while gradually
being retained as you move towards C3. Set the slider to a value of 0.15. The buzzing
will now be far more moderate in the low range.

12 Switch Object 3 back on. Set Timbre to its minimum value (−1.00) and Variation to its
maximum value (1.00). Object 3 should be positioned all the way to the right, at a value
of 1.00.

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13 Vary the Object 3 Strength parameter. You’ll discover that the overtone content of the
buzzing can be controlled very effectively. A Strength value of 0.25 is recommended
here.
14 Save this Setting as “Fretless Roundwound#1”.
Modulation and Detuning
Detuning and ensemble effects are normally achieved using a modulation effect or by
combining doubling and detuning. When using a fretless bass for a solo part, a broad
chorus effect adds a nice touch.
As Sculpture can only synthesize one note at a time at any certain pitch, we can’t work
with doubling. There are, however, alternatives for bringing movement and life into the
sound. Almost all of the Type parameters of the different Objects can be modulated by
LFOs, resulting in a vast number of possible combinations.
Emulating a chorus effect by modulating the Pickup positions:
1 Make sure that you’ve got the “Fretless Roundwound#1” Setting loaded.
2 Adjust the positon of Pickup B to 0.20.
3 The Stereo control element is located to the left of the Level dial. Click-hold on the (Stereo) Pickup semicircle, and move the mouse in an upwards direction. Both of the light
blue dots move downwards towards the letters L and R. You can hear how the stereo
breadth of the fretless sound has increased. Pickup A is sent out on the right channel,
while Pickup B occupies the left channel.

Note: Although only modern basses offer such stereophonic features, it’s still fun to
process conventional sounds (such as those created in the previous examples) with this
effect. Note that not all pickup positions are monophonic compatible; you can check
this by returning the Stereo Pickup setting to mono (by clicking on the Pickup semicircle
while depressing the Option key).
Now we need to make the pickups move:
4 Select LFO1.
5 Activate the first modulation target by clicking on the “1” button (next to the Rate
slider, to the upper right).
6 Choose Pickup Position A-B as the modulation target.
7 Set the Rate dial to 1.00 Hz.

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8 In order to hear the effect, the modulation intensity (amount) has to be set. Familiarize
yourself with this effect by moving the slider labeled Amt (amount) gradually to the
right. Set it to a final value of 0.15, a moderate rate that doesn’t “wobble” too much.
9 Save this Setting as “Fretless Chorus Dry”.
Tip: At the maximum stereo breadth, effects based on detuning are not as prominent,
especially when the “beats” heard in the sound result from signal differences between
the left and right channels. This is only valid to a certain degree because the motion of
the pickup doesn’t create a true chorus or harmonizer effect. Try it out and see what
happens when the stereo breadth is reduced a little. Also test other modulation
targets: Pickup Pos A+B, Pickup Pan A+B, Pickup Pan A−B, and String Stiffness are recommended. Have fun!
Reverb and Reflections
As a rule, basses are mixed without effects (“dry”) and you probably haven’t missed any
reverb or delay effects in our examples, so far. Having said this, a little bit of reverb can
be quite appealing on a fretless bass, when it’s used as a solo instrument. We can use
Sculpture’s Stereo Delay section to emulate this.
In order to create an unobtrusive atmospheric space, proceed as follows:
1 Load the “Fretless Chorus Dry” Setting.
2 Turn on the stereo delay section by clicking on the Stereo Delay button.
3 Set the Input Balance slider to 1.00.
4 Switch off the tempo synchronization of the delay by deactivating the small Sync
button (found directly to the right of the Delay Base Time slider).

5 Set the Delay Base Time slider to 90 ms.
6 Set the Crossfeed dial to 0.30.
The individual reflections are still too brash. In order to make the effect more discrete
and unobtrusive, we are going to adjust the frequency spectrum and amplitude of the
reflections. let’s start with the frequency spectrum:
7 Set LoCut to 200 Hz and HiCut to 1000 Hz (in the stereo delay section).
The LoCut at 200 Hz excludes the low frequencies in the reflections, thus avoiding a
“muddy” sound. The comparatively drastic cut to the highs with the HiCut parameter
blurs the individual reflections, thereby creating the impression of a small room with
soft surfaces.

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8 Reduce the total level of the effect by setting the Wet Level dial to a value of 25%.
9 Save this Setting as “Fretless Chorus+Ambience”.
This example shows that the stereo delay section can be used as a substitute reverb for
small spaces. For sophisticated reverb effects, it’s best to process Sculpture’s output
with one of Logic’s reverb plug-ins.
Creating a “drowned” in delay effect:
1 Reload the “Fretless Chorus Dry” Setting.
2 Switch the stereo delay section on.
3 Move the Input Balance slider all the way to the right (1.00).
4 Set the Delay Base Time value to “1/4t” (quarter note triplet).
5 Set the Feedback dial to a value of 0.20.
6 Adjust the Crossfeed dial to a value of 0.30.
7 Set the LoCut to 200 Hz and the HiCut to 1600 Hz.
8 Now adjust the overall level of the effect; we recommend setting the Wet Level dial to a
value of 45%.
9 Now vary the stereo position and rhythmical structure of the delay by moving the small
light blue diamond around the Stereo Delay Pad.
10 Save this Setting as “Fretless Chorus+Wet Delay”.

Synthesized Sounds
In the preceding sections, you learned how to program natural bass sounds with
Sculpture: by authentically reproducing the real physical interaction that occurs
between a string and the exciting agent that acts upon it. While producing such lifelike
models is undoubtedly a forte of Sculpture’s architecture, its sonic capabilities extend
to the creation of very different sounds as well.
Sculpture contains a number of functions that you can use to create new and novel
synthesized sounds. This includes the Morph Pad, which can be automated, as well as
recordable and programmable Envelopes that can be used in a rhythmic context.
Such features are usually unnecessary when reproducing natural bass sounds, as no
electric bass that exists can alter the tonal characteristics of the string during the decay
phase of a note—perhaps from wood to metal—and rhythmically synchronize this
change to the tempo of the song. These functions are very useful, however, when creating sustained, atmospheric sounds where slow and interesting modulations help it to
“come alive”.

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In the following sections, we’ll be demonstrating Sculpture’s ability to create somewhat spacy and less “organic” sounds, using several pad patches as examples. After
having studied the modeling of bass sounds in the preceding sections, we’ll now introduce you to a totally different group of sounds. Provided that you’re willing to supply
the necessary level of curiosity and time investment for experimentation, you’ll discover a wide variety of interesting and animated sounds.
Within the framework of these short experiments, it’s of course impossible to comprehensively cover all of Sculpture’s possibilities. We’d like to expressly encourage you to
experiment with the suggested settings and closely observe the results of the changes
you make. In this way you can learn a lot about the instrument, and hopefully be
inspired to create new sounds and variations.
Note: You will find the settings for these tutorials in the Tutorial Settings folder in the
settings menu (in the head of the Sculpture Plug-in window).
The Sustained Sound
First, please load Sculpture’s default setting again; the very simple sound consisting of a
plucked string that vibrates and fades away. Obviously, we’ll need to edit this sound
drastically as we need a sustained or extended sound for pads, rather than one that
dies away.
Have a look at the three Objects: you can see that only Object 1 is active, and acts on
the string with an Impulse. As in the pick example in the bass section, the string is
briefly excited when the note is played, and then the sound decays. For a sustained pad
sound we require an exciting agent that constantly acts upon the string; the appropriate Object Types are Bow or Bow wide (the string is played with either short or long,
extended bow strokes), Noise (excited by a random noise signal) or Blow (excited by
being blown—much like a clarinet or flute).

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Please test the abovementioned Object Types one after the other, and move the Object
1 (Pickup) slider, responsible for the exact position of the exciting agent, up and down
the string while you’re playing. You will come to two conclusions. First, the sound is
now sustained for as long as you hold the key down. Secondly, shifting the Object 1
slider, with the Bow Type selected, results in the most pronounced sonic changes. This
setting promises the most rewarding possibilities for varying the sound, and that’s why
we’ve decided to use this Type.

Recording an Envelope
The sonic variations created by the Bow Type are very appealing when the virtual bow
stroke is moved along the string. We’d like to control this movement through the use of
an Envelope, thus creating the foundation of our pad sound.
It makes more sense (and it is more convenient) to record the Envelope, rather than
programming it, even if the latter is easily achieved with the graphic display.
Proceed as follows to record an envelope:
1 Move the Object 1 slider all the way to the left. Starting from this position (where it
only generates an overtone-rich scratch), we want to start animating it by using the
Envelope.
2 Locate the Envelope section in the lower right corner of the Sculpture window. Choose
the first of the two Envelopes by clicking on the Envelope 1 switch, if necessary. In the
left part of the Envelope section, you can see two routing possibilities that allow you to
assign a modulation target to the Envelope.
3 Activate the first routing link by clicking on the “1” button, and choose Object1 Position
as the modulation target in the Target menu. Set the modulation intensity to its maximum value by moving the horizontal slider all the way to the right.

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The Envelope can now be recorded. We assume that your MIDI keyboard has a modulation wheel which outputs the corresponding MIDI controller message (CC number 1)
and that option 1 ModWh is selected for control of Envelope 1 (CtrlEnv 1) in the dark
bottom edge of the Sculpture window.
4 Click on the button labeled R, located in the upper right corner of the Envelope section
below “Record Trigger”, to prepare the Envelope for recording. Select the Note+Ctrl
option for the recording. This option specifies that the recording of the controller messages from the modulation wheel will begin the instant the first note is played.

5 Play a note when you want to start the recording, and move the modulation wheel
slowly upwards while keeping the key on the keyboard depressed. Pay attention to the
sound variations you create while moving the modulation wheel.
6 At the end of the recording, return the wheel to its initial starting position and, after
you’ve released the depressed note, click on the “R” button to deactivate the recording
mode.
You can now see the curve you recorded in the graphic display. You’ll note that the
curve arches exclusively above the zero axis—this is no surprise as the modulation
wheel only sends unipolar values, which means between zero and up to a positive
maximum value. As we already moved the Object 1 slider all the way to the left end of
the string, it can only be shifted all the way to the right by the Envelope when the
maximum modulation intensity is reached.
Play a note, or better yet, an entire chord and listen to the modulation you recorded. If
you’re not satisfied, you can, of course, repeat the procedure described above as often
as you’d like. The maximum available recording time is 40 seconds. It is therefore possible, if you’d like, to control parameters and create modulations that extend far beyond
the well-worn path of a simple ADSR envelope progression.
Note: By moving the junction points, you can edit the shape of the envelope when the
need arises. By clicking in the empty part of the graphic display and moving the mouse
to the right or left, you can zoom in and out of the display. When you activate the small
Sync button, the junction points snap to a rhythmic grid (further information about
these parameters and their functions can be found on page 397).

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Select Loop Alternate in the Sustain mode menu below the Envelope graphic display. As
the sustain point is found at the end of the Envelope, the Envelope repeatedly travels
from the beginning to the end and backwards from the end to the beginning, creating
a continuous flux within the sound.

Summing up: we now have a rudimentary, but appealing and organic-sounding pad
which we will use as the foundation for further shaping and refinement. Save this
rough version of the sound as a basis for further experiments (Preset 0001 raw pad).
Increasing Stereo Breadth and Chorus
The next thing we’ll do is give our very dry sounding pad a little more stereo breadth
and chorus effect. We’ll use the trick we discussed while creating bass sounds. Namely,
modulating the Pickup positions, and assigning them to the left and right channels.
Here’s a quick description of the process:
1 Click-hold on the Pickup semicircle in the Stereo control element, and move the mouse
upwards to separate the stereo pan positions of the Pickups. Both of the light blue dots
should come to rest near the line that separates both semicircles.

2 Activate both of the modulation links in LFO1 by clicking on the buttons labeled 1 and
2. For the first link, modulate the position of Pickup A (Target: PickA Pos) with a small
positive value and with the second link, the position of pickup B (Target: PickB Pos) with
a small negative value (intensity at ±0.15). Reduce the Rate of the LFO to about 0.3 Hz.

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You will hear a pleasant “beating” or chorus effect in the sound, which makes it broader
and more full, alleviating the unpleasant, dry character (preset 0002 lfo>pick pad).
Another unpleasant aspect is that the sound is too strong in the mid frequency range
and could use some equalization. We’ll use the Body EQ to correct this.
Activate the Body EQ, and experiment with the Lo Mid Hi model (which is the standard
setting). Our favorite setting is when Mid is reduced 0.5 and Mid Frequency is set to 0.37
but this, of course, is a matter of taste…

Now we want to give the pad a little depth by using the Stereo Delay section. Activate
the Stereo Delay, set the Delay Time to 1/4 and adjust the Crossfeed dial to 30%. The pad
now has a pleasant and unobtrusive ambience; you can leave the other Stereo Delay
parameters at their original values (preset 0003 eqfx pad).

Finally, we want to optimize the sound so that it is a little more animated. This is nothing too spectacular, so we’ll employ the Jitter modulators for the job. The Jitter modulators are basically LFOs that use a random waveform.
Using the jitter modulators to make the sound become more lively:
1 Activate the display for both of the Jitter modulators by clicking on the Jitter switch
below the LFO.
2 Turn on the first link in Jitter 1 by clicking on the button labeled 1, and choose Object1
Timbre in the target menu.

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3 Adjust the Intensity to −0.40 with the slider below the Target menu, and reduce the Rate
parameter to 1 Hz. There should be subtle inconsistencies in the pressure applied by
the bow to the string. To better recognize the effect, temporarily increase the Intensity
level.

We’ll use the second Jitter modulator for random position deviations with the modulation target Pickup Pos A+B (pickup position A and B).
4 Activate Jitter 2 and choose the Pickup Pos A+B option in the Target menu.
5 Set the slider under the Target menu to an Intensity of about 0.2, and adjust the Rate
dial to 1.5 Hz. As you increase the Intensity, the sound develops a distinct clinking or
rattling—use this effect to taste.
We now have a satisfactory pad sound, so we’ll leave it at that, even though a few
Sculpture features such as; the Filter and the Waveshaper lie idle, not to mention the
two additional Objects—but it’s smart to quit while you’re ahead… We’ve saved an
especially exciting function, morphing, for the end and we’d like to use it to bend and
twist our pad sound a bit.
Morphing
In the middle of the lower part of Sculpture’s window you can see the Morph Pad. In
each corner, it can contain a different setting for a diverse number of parameters and
by moving the red ball—which can be seen in the center of the Morph Pad—you can
crossfade between these settings and “morph” the sound.
Choose the Paste to all Points command in the context menu (accessible via a Controlclick on the Morph Pad) to copy the current patch setting into all four corners of the
Morph Pad. If you move the ball in the Morph Pad now, you won’t hear any changes in
the sound because the settings (in each corner) are all the same. Not for long…

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To vary the sound with the Morph Pad:
1 When you move the ball to one corner, the corresponding “partial” sound is selected;
you can recognize this by the gray blue arches that light up in the graphic display.
Choose each of the four corners one after the other, and vary the sound by altering
several parameters directly in Sculpture’s GUI.
2 Use of the Material Pad allows you to achieve an especially noticeable
variation: carefully move the ball around in the Material Pad, and try to find a position
where our pad sound takes on a new and interesting character. Also try the extreme
corners, for example.

As soon as you’ve chosen different settings for the Morph Pad corners (A to D), moving
the morph ball will create marked sound variations—even though the intermediate
stages will not all exhibit a tonal character. You can automate the morphing process by
assigning two MIDI controllers to the MorphX and MorphY menus at Sculpture’s bottom
edge. You can also automate the Morph Pad using a recorded Envelope—you’ll find
further infomation about this on page page 395.

Surrounding the Morph Pad, you’ll find a randomizing function which randomly varies
sounds to a chosen intensity level (or amount of randomization). This is especially
useful for subtle changes to natural sounds, but it can also provide for rewarding
variations to synthesized sounds as well.
To use the randomizing function:
1 On the left side of the Morph Pad, choose the number of corners that are to be varied
by selecting one of the “cubes”.
2 Use the slider to the right of the Morph Pad to adjust the Intensity of the random
deviations.

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3 Perform the randomization by clicking on the Rnd button. When you next move the
morph ball, you’ll hear the variations you just created.

We have now reached the end of our programming tutorial. By demonstrating how to
create basic sounds, detailed emulations of various electric bass sounds and explaining
how to approach the generation of synthesized sounds, we hope to have given you a
few insights into the interplay of Sculpture’s functions and parameters. We also hope
that you’ve gained some inspiration and insights for programming your own sounds—
we wish you much fun and success while creating your own sonic sculptures!

KlopfGeist

KlopfGeist is an instrument that is optimized to provide a
metronome click in Logic.

KlopfGeist is inserted on Audio Instrument channel 64by default. Logic automatically
assigns this channel to the Metronome Object, making KlopfGeist the synthesizer
responsible for the metronome click.
Theoretically, any other Logic or third-party instrument could be used as a metronome
sound source on Audio Instrument channel 64. Similarly, KlopfGeist can be inserted on
any other Audio Instrument channel for use as an instrument.
A look at KlopfGeist’s Plug-in window will, however, clearly show that it is a synthesizer
designed to create the metronome’s clicking sound.

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KlopfGeist can operate as a monophonic or polyphonic (4 voice) instrument, as
determined by the Trigger Mode radio buttons. There are two tuning parameters; Tune
for coarse tuning in semitone steps, and one for fine tuning (Detune) in cents. The
Tonality parameter changes the sound of KlopfGeist from a short click to a pitched
percussion sound—similar to a Wood Block or Claves. Damp controls the release time.
The shortest release time is reached when Damp is at its maximum (1.00) value. Level
Via Vel determines the velocity sensitivity of KlopfGeist. It is a two part slider; the upper
half of the slider determines the volume for maximum velocity, the lower half for
minimum velocity. By clicking and dragging in the area between the two slider
segments, you can move both simultaneously.
The overall level of the virtual click sound is determined by the Audio Instrument
channel 64Volume fader.
Note: A Klopfgeist (knocking ghost) is a ghostly little fellow, usually German, who
restricts himself to producing knocking and tapping sounds, unlike his big brother, the
Poltergeist. Audio Instrument channel 64is fitted with a Ghost Buster facility—the
button labeled M.

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EVB3

This chapter introduces Logic’s EVB3 virtual Hammond
organ.
An electro-mechanical Hammond organ, with tonewheels and a Leslie sound cabinet,
cannot be replaced by anything other than an electro-mechanical Hammond organ,
with tonewheels and a Leslie sound cabinet! Having said that: The EVB3 offers a truly
portable alternative to this classic pop, rock, and jazz instrument.

Concepts and Function
The EVB3 software instrument mimics the sound, and use, of the Hammond B3 and
Leslie sound cabinet. Being a software instrument, the EVB3 is not only lighter and
more portable than its big, heavy counterpart, it’s also much easier to integrate into
Logic’s production environment. There’s no need for any cabling or microphone setup,
for live-performances. The same applies to the recording of realistic organ tracks, with
the added bonus of full editing facilities at any stage of production.
The EVB3 simulates an organ with two manuals (keyboards) and a pedalboard—each
of which can have its own registration (sound settings). The sound generation process
is not limited to the mere additive synthesis of sine choirs. Rather, it fully simulates the
tone wheel generators of an electro-mechanical Hammond organ, down to the
smallest detail. This includes certain charming flaws, such as the Hammond’s enormous
level of crosstalk and the scratchiness of the key contacts. You may adjust the intensity
of these peculiarities to meet your tastes. This flexibility allows for flawlessly clean,
through to dirty and raunchy sounds, and everything in-between. Further quirks of the
original were the robbing of sine choir voices (multiplexing) and the specific repetition
behavior. No need to tell you that the EVB3 emulates this too!
The EVB3 simulates three different types of Leslie sound cabinets with rotating
speakers, with and without deflectors. Beyond this, three tube overdrives—with
different tonal characteristics, an equalizer, a wah wah and a reverberation effect are
also incorporated. You may freely define the signal flow of these effects as you wish.
You can also set the stereo intensity of the microphone position as desired.

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If you’re familiar with the original B3, you’ll remember the inverted (black) keys of the
lowest octave on each manual. These inverted keys are switches that recall preset
registrations (a preset of your drawbar settings). This feature is emulated by the EVB3 as
well, but has been improved significantly, given that you won’t need a screwdriver to
change the registration settings of your presets. A morphing feature allows seamless
crossfades between two registrations, without the need for an external fader box. The
EVB3 can be played with two manuals and a MIDI pedalboard, if you wish (and own
these devices). It also offers functions which allow you to play all registers (Upper,
Lower, and Pedal) with a single-manual master keyboard.

MIDI Setup
If you want to fully exploit all features of the organ, you will need a MIDI (bass) Pedal
unit, and two 73-key MIDI keyboards. As the EVB3 also emulates the B3’s preset keys,
the lowest octave of attached MIDI keyboards can switch the EVB3 registrations, just
like the original B3. Please read “Playing Both Manuals and the Pedals Live” on
page 456 for more information.
The EVB3 can, of course, be played with single-manual keyboards with the standard 61
keys (5 octaves C to C). Please read “Keyboard Split” on page 458 for more information.

Playing Both Manuals and the Pedals Live
The EVB3 receives the notes for the Upper and Lower manuals, and the Pedalboard, on
three subsequent MIDI channels.
Note: The MIDI channel (MIDI Cha) of the Audio Instrument must be set to All, and
Keyboard Mode must be set to Multi.
The default Upper-manual receive channel is MIDI channel 1, channel 2 for the Lower
manual, and channel 3 for the Pedal registers. If your master keyboard sends MIDI
notes on channel 1, it will play the Upper manual, if it sends on channel 2, it will play
the Lower manual, and if it sends on channel 3, it will play the Pedal register.
You may use any of your MIDI interface (Unitor 8 MkII or AMT 8, for example) inputs for
your master keyboard(s) and/or Pedalboard. You can also use a single-manual master
keyboard—with different keyboard zones or a keyboard split feature—that sends data
on different MIDI channels. Regardless of the input device(s) used, the only relevant
factor is the MIDI send channel!

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Logic records the channel information of incoming notes. With most other MIDI and
software instruments, this information is not used at all. This is due to Logic’s MIDI
channel setting (in the instrument parameters), which has priority, and overrides the
original channel information. This can be circumvented by setting MIDI Cha = All, which
is recommended for the EVB3. This will force the original channel information to be
used. This enables you to make use of the two manuals, and Pedal register of the organ
directly and simultaneously—when playing live.
Note: Please read the users manual of your master keyboard, to learn how to set its
MIDI transmission channel, often abbreviated as TX Channel.
Changing MIDI Channels
You also can set the EVB3 to receive on MIDI channels other than 1 for Upper, 2 for
Lower and 3 for Pedal. This is done with the Basic MIDI Ch parameter in the General
section. The receive channel number for Lower is always one (channel) number higher
than the channel assigned to Upper with the Basic MIDI Ch parameter. The Pedal register
receive channel is always two (channel numbers) higher than the Basic MIDI Ch selected
for Upper.
Note: The Basic MIDI Ch parameter is only available in the Controls view. Click-hold the
Editor pull-down menu at the top of the Plug-in window to access.
Note: Basic MIDI Ch only works if Keyboard Mode is set to Multi.
Note: When the Basic MIDI Ch is set to 16, the Lower manual receives on channel 1, and
the Pedal register on channel 2. When the base MIDI Ch is set to 15, Lower receives on
channel 16, and the Pedal register receives on channel 1.
Note: Selection of different MIDI channels may be necessary in a live performance
situation, particularly if you need to change the MIDI transmission channel of your
master keyboard, in order to play other sound generators.
Keyboard Ranges of the Upper and Lower Manual
The lowest playable MIDI note is 36 (C1). The range of the preset keys is note # 24 to 35
(C0 to B0).
Note: 128 notes are defined in the MIDI specification, but even the largest master
keyboards are only equipped with 88 keys, just like a concert grand piano.
One example: If your master keyboard ranges from C to c (5 octaves—61 keys), and the
Region and Object parameters of Logic are set to zero (Transpose. = 0), you can play the
entire keyboard range—every possible note of the EVB3. The preset (registration) keys
are positioned one octave lower. If you set Transpose. = −12, you can use the lowest
octave to switch between presets.

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Keyboard Split
The EVB3 can also be played perfectly with a single MIDI keyboard (one manual) that is
only capable of transmitting on one MIDI send channel. You can split the keyboard in
order to play Upper, Lower, and Pedal sounds on different keyboard zones.
In the parameter field in the bottom center of the GUI, set Keyboard Mode = Split.

Set the keyboard zones with the UL Split and LP Split parameters, in conjunction with
the Set buttons. The abbreviations are for: Upper/Lower and Lower/Pedal. To do so,
simply click on the appropriate Set button (it will turn orange), and press the desired
note on your MIDI keyboard.
Note: If the LP Split is set to a value above the UL Split, the other splitpoint is moved
(and vice versa).
Note: If you select the same value for both splitpoints, the Lower manual is not heard/
active.
Note: In order to use preset switching in conjunction with Keyboard Mode Split, you
might need to adjust the transpositions for the Upper Manual (Trans UM), Lower
Manual (Trans LM) and Pedal (Trans Ped).

Transposition (Octave Range)
You can individually transpose the Upper (Trans UM), Lower (Trans LM) and Pedal (Trans
Ped) registers up/down one or two octaves, independent of the global transpose
functions. Transposition is useful for setting the various registers to a particular range,
matching your needs. This facility is of particular importance when using split mode.
Note: As with Logic’s global transposition functions (which can be set in semitones),
these manual-specific transpositions have no impact on the preset keys. Also see
“Preset Keys and Morphing” on page 464.

MIDI Mode
This parameter allows you to define how the drawbar settings will respond to remote
MIDI control change messages. Normally, you won’t need to change anything here. If
you own a MIDI drawbar organ, however, you’ll probably want to use its hardware
drawbars to control the EVB3. Most hardware drawbar organs utilize an independent
control change number for every drawbar. Some models allow you to freely define
these control change numbers. Lists of control change number assignments for the
following MIDI Modes can be found in the “MIDI Controller Assignments” section, on
page 477.

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If you select MIDI Mode = RK, every drawbar responds to a specific MIDI control change
number, commencing with CC #70. (Non-drawbar parameters may be set using control
change messages up to CC #118).
If you select MIDI Mode = HS, all of the EVB3 drawbars will be controlled by just a few
control change numbers—CC #80–82. Its values are intelligently mapped to all
drawbars. The resolution of this technique is not particularly high (much like the
original B3), but it works well. The Hammond-Suzuki XB-2 uses this controller
assignment method, allowing you to remotely control the EVB3’s drawbars.
If you select MIDI Mode = NI, the Native Instruments B4 settings will be replicated,
allowing the use of the Native Instruments B4D Drawbar controller.
If you select MIDI Mode = off, the EVB3 will not respond to MIDI drawbar messages.
Note: Before you commence a recording session with drawbar movements, performed
on a hardware organ and the EVB3 set to MIDI Mode HS, we recommend that you
disable MIDI Data Reduction in Logic. The parameter is located in File > Song Settings >
Recording. The function thins out the incoming data stream, which is hardly ever
noticed with normal controller assignments. The controller mapping of the Hammond
XB-2’s drawbars, however, exploits the controllers to their limits. Therefore, MIDI Data
Reduction can lead to incorrect values. Switch the parameter on after the drawbar
recording session, in order to regain memory and processing resources.

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The EVB3 Parameters

The graphical user interface (Editor view) opens when you double-click on the EVB3
Input slot of an Audio Instrument object. You can open and close the wooden lid by
clicking the button underneath the Volume control. Keep it open while reading this
section of the manual, because we’ll be discussing every parameter in detail.

Drawbars
The principles of additive synthesis with sine drawbars is further explained in “Additive
Synthesis With Drawbars” on page 481. You can intuitively pick up the basic principles
by playing a little with the drawbars. The further down you drag the drawbars, the
louder the selected sine choir(s) will be—the drawbars behave like reversed mixer
faders. MIDI control of the drawbars is also reversed, when using a standard MIDI fader
box.

Drawbars of the Upper and Lower manual, plus Pedal drawbars.

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Pedal Drawbars
The organ features two drawbars for the bass Pedals. The waveform of the bass is not a
pure sine wave, but a mixed waveform, that realistically simulates the Hammond B3
bass. The two registers differ in pitch, and in the following ways:
• the Lower 16' register contains more octaves
• the 8' register has a more prominent fifth portion.

Volume
Relative Volumes—Upper/Lower/Pedal
In the Organ parameters section, you can set the volume of the Lower manual, and the
Pedal, relative to the Upper manual. The parameters are called Lower Volume and Pedal
Volume.

Volume Control and Expression Pedal
The overall volume of the EVB3 is not only controlled by the Audio Instrument volume
fader and control change #7, but also with the Volume control in the EVB3 graphic user
interface.

Warning: The Volume must be lowered whenever crackling or other digital distortion
occurs in the Audio Instrument channel. Volume levels over 0 dB can occur if you
maximize all registers, play numerous notes, and make use of the Distortion effect.
You can control the volume in real-time with an Expression Pedal (Swell Pedal).
Extensive, and often rhythmic, use of the Expression (volume) Pedal forms part of the
style of many organ players. The expression control also emulates the tonal changes of
the B3 preamplifier. Bass and treble are not attenuated as strongly as the mids, much
like a Hi-Fi amplifier that features a loudness correction facility.
Normally, you would connect an Expression Pedal to the quarter inch Expression jack of
your master keyboard. Your master keyboard should transmit MIDI control change #11
when the pedal is moved.

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Tune
The simulated tone wheel generator can be tuned in cents (percentages of a semitone).
0 c is equal to A = 440 Hz.

Scanner Vibrato
The vibrato of the organ itself must not be confused with the Leslie effect, which is
based on rotating speaker horns. The EVB3 simulates both.

The Scanner Vibrato is based on an analog delay line, consisting of several lowpass
filters. The delay line is scanned by a multipole capacitor, which has a rotating pickup. It
is a unique effect that cannot be simulated with simple LFOs.
Like the Hammond B3, the EVB3 features three types of vibrato—with different
intensities (V1, V2, V3). Vibrato speed is set with the Rate parameter. In the V1, V2, and
V3 positions of the Type parameter, only the signal of delay line is heard.
The C1, C2, and C3 Chorus positions of the Type parameter mix the signal of the delay
line with the original signal. Mixing a vibrato signal with an original, statically-pitched
signal, results in a chorus effect. The Chorus parameter allows you to freely mix the dry
signal with the vibrato signal. The Chorus parameter is only active if one of the chorus
settings is engaged.
If the C0 setting is active, chorus and vibrato are disabled. Note that the treble portion
of the organ is boosted slightly if any vibrato setting is used. This treble boost is
maintained in the C0 setting.
The Upper and Lower buttons allow you to switch the scanner vibrato (and its
associated treble boost) on/off, individually, for the Upper and Lower manual. As the B3
mixes the bass register (Pedal) with the Lower manual, the Pedal register is affected by
the Lower manual’s scanner vibrato settings. This side-effect reflects the technical
limitations of the original B3.

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Note: Check out the chorus and vibrato effects, and compare them with the sound of
the rotor cabinet simulation! The organ’s chorus sounds different to modern chorus
effects (such as Logic’s Chorus plug-in). Many organ players rarely use the Scanner
Vibrato, preferring to work with a Leslie, in isolation. Others, like B3 virtuoso Brian
Auger, prefer the integrated organ vibrato over the Leslie.

Percussion
Percussion is only available for the Upper manual—as it does on an original B3. The
percussion of an electro-mechanical organ is polyphonic, but is only (re)triggered after
all keys have been released. If you release all keys, new notes or chords will sound with
percussion. If you play legato, or sustain other notes on the Upper manual, no
percussion will be audible.
On the original B3, percussion is only available if the “B” preset key is selected (see
“Preset Keys and Morphing” on page 464). If you want this restriction to be simulated,
set the Perc parameter to Only B. You find this parameter in the Organ section. If you
always want percussion to be available, select Always.

Activate percussion by pressing the On button in the Percussion section, to the topright of the GUI. The percussion is heard in the 4' register or in the 2 2/3' register, if the
2nd/3rd button is set. Set the percussion decay time with Time, and set its level with Vol.
This improves on the B3, where Time and Vol could only be switched on/off, whereas
these parameters can be varied.
Note: Time has a maximum setting called Paradise. In this position, the percussion
doesn’t decay at all. The name is derived from a famous Jimmy Smith recording
“Groovin’ at Small’s Paradise”, where Jimmy used a B3 that had a defect in the
percussion trigger. The cool thing about this technical flaw is that one harmonic
sounds without Chorus-Vibrato, while the drawbar harmonics feature Chorus-Vibrato.
While we recognize that this is very specialized, we thought it would be a nice
inclusion, especially for jazz-oriented players.
The EVB3 percussion register can be played with velocity sensitivity, unlike the B3. Set
the percussion velocity sensitivity with Vel.

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If you engage percussion on a B3, the volume of the normal, non-percussive registers is
reduced slightly. The Up Level parameter simulates this behavior, allowing you to define
the volume of the Upper manual, with percussion engaged.

Preset Keys and Morphing
The Hammond B3 is equipped with 12 switches, located below the lowest octave of
both keyboard manuals. These are the preset keys, and are laid out as an inverted
keyboard octave (black keys, white sharps). They are used to recall drawbar
registrations. These presets could only be altered with a screwdriver on the original B3.

Drawbars and Preset Section

MIDI notes 24 to 35, the octave below the lowest octave of a (non-transposed) 5-octave
keyboard, are used as the preset keys. See “Keyboard Ranges of the Upper and Lower
Manual” on page 457.
You can click these preset keys directly in the graphic user interface of the EVB3. They
are located to the left (Upper) and right (Lower) of the Morph wheel. The current
drawbar registrations are indicated by small vertical lines on each key.
The presets only relate to the registration of a single manual. The presets do not store
vibrato or other parameter settings. The preset keys work in real-time, thus overall
settings (including effects) can be stored and recalled via the Settings menu.
Note: On keys C# to A#, the percussion only works in Percussion Mode Always (see
“Percussion” on page 463).
Disabling MIDI Preset Switching
You can disable the switching of presets with MIDI notes 24 to 35, thus avoiding any
problems that may arise from transpositions. To do so, Disable the MIDI to Presetkey
parameter.
Cancel Key, Registering while Playing
The lowest preset key (C) is the cancel key. if you depress it, all drawbars are moved to
their minimum setting. The other 11 keys, from C# to B, recall registrations. You can edit
recalled presets immediately. The preset memorizes these alterations instantaneously,
with no further action required. This means that if you recall a new preset, the former
preset memorizes the drawbar settings at the time the new preset was recalled.

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If you hold the cancel key (C) on your master keyboard with the small finger of your left
hand, and sustain a chord with your right hand, you can trigger the chord with
different registrations, by pressing the preset keys with the other fingers of your left
hand. This results in an organ-specific gater type effect, which wouldn’t be possible
with the right hand alone.
Morphing
You can switch between the presets of the Upper manual with any keyboard. To be
more precise, you can switch between registers with any MIDI controller—such as the
modulation wheel. Choose the desired MIDI controller with the Midi CC parameter.
The step or linear Mode parameter options determine whether an abrupt switch, or a
seamless morph (cross-fade), occurs between presets.
Note: Switching the registers with this morphing function is only available for the
Upper manual.
Range
After deciding on a controller to use for switching or morphing the Upper manual
registers, you can determine the number of preset keys that are affected.
The morphing (or switching) always begins with the top preset key, the B. Range
defines the “end” preset key. If Range = A#, you will switch or morph between two
presets. If Range = G#, you will switch or morph between four presets (B, A#, A, and G).
Save To
In Linear Mode (morphing instead of switching), the seamless crossfades result in a
variety of new drawbar registrations you might want to save. Before saving, you may
wish to alter some drawbar positions manually. Click the words Save To, and select the
destination preset key. You do not need to do anything else, in order to save the setting
to another preset key.
Note: As soon as you morph, the Morph text turns orange, indicating changes—in
other words, you’re performing a morph. The morphing result(s) can be further
modified with the drawbars, but are lost if you don’t save it/them. The Morph text will
start to flash if any drawbar modifications are made, indicating changes to the Morph
values.

Organ
The Organ parameters adjust the overall behavior of your EVB3.
The Lower Volume and Pedal Volume parameters are discussed in “Relative Volumes—
Upper/Lower/Pedal” on page 461. The Perc parameter is discussed in “Percussion” on
page 463.

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Max Wheels

Calculating (emulating) all tone wheel generators consumes considerable CPU
processing power. A reduction of this parameter value reduces the EVB3’s hunger for
processing resources.
Note: This will diminish some overtones, so you should not reduce this value if you’re
after an ultra-realistic simulation.
Tonal Balance
Tonal Balance changes the mix relationship of the higher/lower sounding tone wheels.
Positive values result in a lighter and brighter sound. Experiment with different Tonal
Balance and Equalizer settings. See the “Equalizer” on page 471, for further information.
Shape
While the Hammond’s tone generators produce pure sine waves (despite technical
artefacts), other organs deliver distorted waveforms. You can produce sounds
resembling those of Farfisa, Solina or Yamaha organs with the Shape parameters.
You can subtly alter the waveform of the sounds emanating from the tonewheel
generator with the Shape parameter. Moving the parameter to the right will make the
tone brighter (and louder), and moving it to left will make it duller (and softer).
Note: The Shape parameter is placed after the filters which follow the sine generators.
Bass Filter
The sound of the Pedal drawbars often appears to be somewhat “brilliant”, within the
overall musical context. To circumvent this issue, and to suppress the treble of the bass
register, please make use of the Bass Filter. When active, you will only hear a solid bass
organ fundamental in the bass register.
Ultra Bass
If you switch on Ultra Bass, another low octave will be added to the playable range of
the Upper and Lower manuals. These additional low octaves, and the ability to
independently transpose both manuals (see “Transposition (Octave Range)” on
page 458), are not available on the original B3.

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Condition
Technical limitations of electro-mechanical drawbar organs with tonewheels cause
some strange tonal artefacts, such as crosstalk. These quirks form an integral part of the
B3’s charm. You can adjust the following parameters to define the age of your EVB3.

Note: Read more about the Click parameters in “Click” on page 468.
Drawbar Leak
Even if all drawbars are at their minimum position, the tonewheel generators of the B3
aren’t completely quiet. This is due to “leakage” of the tone wheels, cross-talking with
the output. Drawbar Leak allows you to set the minimum output level of the drawbars,
when set to minimum. At a value of 0, you can completely eliminate this leakage. At
the maximum setting, drawbar leakage is most definitely audible.
Leakage
Leakage adds a sound resulting from the crosstalk of all tone wheels—including those
of the notes that you don’t play. This provides “breath” to the organ sound.
Crosstalk
The Hammond’s tone wheels are divided into compartments of four—with the same
key, but in different octaves. There are two tonewheels, four octaves apart, on each
rotating shaft. The signal of the Lower wheel contains a small amount of signal,
induced by the higher wheel, and vice versa. This “crosstalk” can be adjusted with the
Crosstalk slider. Note that crosstalk is only audible on certain wheels, avoiding “rumble”
when chords are played.
Random FM
If the tonewheel generator is clean, all frequencies are “straight”—the frequencies are
even/in tune. The three-fold decoupling of the tonewheels—via springs, flexible
couplings and flywheels—is very effective, but can’t compensate for irregularities that
come with dirt and grease in the driving gear. This gradual build-up of grime in the
mechanism makes the revolution of the tonewheel assembly irregular on its axis. This is
transmitted to the tonewheels. The Random FM slider allows you to simulate this effect.
Note: The effect only becomes audible in the higher frequency ranges.

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Filter Age
The high frequency output signals of the B3’s tone wheel generators are filtered by
bandpass filters. The center frequency of these filters varies as the capacitors age. Filter
Age allows you to alter the center frequencies of the filters. This colors the sound of the
jitter applied by Random FM and the background noise resulting from Leakage. (See
“Leakage” on page 467 and “Random FM” on page 467.) This parameter also influences
the intonation of the organ, if you use the pitch bender.

Click
The key contacts of electro-mechanical tonewheel organs tend to “saw” a little on the
busbar, introducing a short click sound. If any corrosion occurs to the key contacts or
busbar, this will increase the length, and level, of this click. This aspect of the B3’s
design causes irregular scratching noises (commonly referred to as “keyclick”), when
striking and releasing keys. Hammond fans like these clicking noises, as they introduce
a transient, percussive quality to the note. The EVB3 allows you to adjust the volume
and sound of the key click. The click sound is altered randomly, and independently,
from the click on and click off (release) volume settings.

Click On/Click Off
These two knobs independently control the click volume for the beginning (Click On),
and release, of the note (Click Off ). The click off is quieter, even if both controls are set
to the same position.
Click Min/Click Max
Not only are the tone color, and volume, of clicks altered randomly, but also their
duration. Click duration can vary between a short “tick” and a longer “scratch”. Minimum
duration is defined by Click Min, and maximum duration with Click Max. The duration is
displayed in milliseconds, as you move the sliders.
Note: Even if both parameters have identical values, there is still a random variation in
sound. This variation makes some clicks seem shorter than the value set with Click Min.
Click Color
Set the tone color of the click here. Despite its random variation, you can define its
treble portion globally.

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Pitch

Compared to the original B3, the EVB3 offers several parameters to change its pitch
behavior.
Note: The Trans UM, Trans LM and Trans Ped parameters are explained in “Transposition
(Octave Range)” on page 458.
Stretch
The EVB3 is tuned to an equal-tempered scale. As a deviation from this standard
tuning, you can stretch the tuning in the bass and treble ranges, much like acoustic
pianos (especially upright pianos). If you select a value higher than 0 for Upper Stretch,
the pitch of the higher notes will be raised. If you select a value higher than 0 for Lower
Stretch, the pitch of the Lower notes will be lowered.
Note: The tones of clavinets, harpsichords, and pianos have inharmonicities in their
harmonic structure. The frequencies of these overtones (harmonics) are not exact,
whole-number multiples of the base frequency, as Pythagorean theory dictates. They
are only approximate and are, in fact, a little higher. The overtones of lower (tuned)
notes, therefore, are more closely related to the main frequencies of the upper notes.
Due to the lack of strings, this inharmonic relationship is not true of organs. The stretch
feature was included for situations where you may wish to use the EVB3 in an
arrangement alongside an acoustic or digital piano (EVP88 and EVD6 Clavinet). When
arranged in conjunction with an orchestra or synthesizers, the stretched tuning facility
should not be used. Please experiment!
Lower Stretch controls the amount of deviation from the equal-tempered scale in the
bass end of the sound. The higher the value, the further down the low notes are tuned.
At a setting of 0, the EVB3 is tuned to an equal-tempered scale, with each octave below
exactly halving the frequency. Upper Stretch controls the amount of deviation from the
equal-tempered scale in the treble end of the sound. The higher the value, the further
up the high notes are tuned. At a setting of 0, the EVB3 is tuned to an equal-tempered
scale, with each octave up exactly doubling the frequency.
Warmth
Warmth controls the amount of random deviation from an equal-tempered scale. High
values add “life” to organ sounds, but tend to sound a little out of tune.

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When applying Warmth and Stretch, you should consider that these parameters may
result in a detuned sound, which is similar to the overuse of a chorus effect. Straight
tuning is nice too, so set Warmth to 0 if you’re after a “pure” sound.
Pitch Bender, Brake Effect
The Hammond organ has no pitch bender. As such, use of the pitch bender is not
suitable for realistic simulations, but it does provide a number of creative options.
Pitch up/down bender sensitivity can be set independently, in semitone steps, with the
Pitchbend Up and Pitchbend Down parameters. The maximum sensitivity for upward
bends is one octave.
You can set Pitch Bend Down to Brake, which gradually slows the movement of the tone
wheels down to a total stop, at the pitch bender’s minimum position.
Note: The Pitch Bend Down = Brake setting recreates an effect that is audible at the end
of “Knife Edge” by Emerson, Lake and Palmer. Keith Emerson’s virtuoso Hammond work
was recorded on a reel-to-reel tape recorder. At the end of the song, you can hear the
tape recorder being gently slowed to a total stop.

Sustain
Synthesizer players call the time the note takes to fade out after the release of the key
the “Release Time”. The EVB3 allows you to control this parameter as well; it’s called
“Sustain” in the organ lexicon. The three controls allow for individual settings in the
Upper (Up), Lower (Low) and Pedal (Ped) registers.

If you select the Smart Mode, playing new notes will cut the sustain (release) phase of
released notes. Normal Mode allows polyphonic sustain phases—all released notes will
continue to sustain. Smart Mode allows long sustain times, even in the bass register,
which would normally cause rumbling dissonances.

Effects
The EVB3 features a three-band equalizer, a reverberation effect, a pedal-controllable
wah wah, and a distortion effect that simulates the sound of an overdriven tube
amplifier. Finally, the signal can be processed by the rotor effect.

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Effect Chain and Effects Bypass
Effect Chain
The EVB3’s signal flow is as follows: the organ’s signal runs through the equalizer, wah
wah and distortion effects. You can choose between four different signal flow routings
for the equalizer, wah wah and distortion effects in the Effect Chain pull-down menu.
This treated signal is then fed into Reverberation and finally passed to the Rotor effect.
A “classic” B3 patch would be: an EQ’ed organ, plugged into a wah wah pedal,
amplified by an overdriven Leslie. Select EQ-Wah-Dist.
The sound of the overdrive changes if the input signal is being filtered—be it by the
EQ, or the wah wah. If you patch the EQ before the overdrive, the sound of the
overdrive becomes much more flexible. The output signal of the distortion effect
always contains high frequency content. If you want to suppress these frequencies, the
wah wah must be the final effect in the chain—EQ-Dist-Wah.
If you wish to create a “screaming” sound (achieved by distorting the wah wah output),
you can minimize any “harshness” by choosing the Wah-Dist-EQ routing.
You can suppress the brutal overtones of extreme distortions with two filters: Select
Dist-EQ-Wah.
Effect Bypass
The Distortion, Wah, and EQ effects can be bypassed separately for the Pedal register.
Set Effect Bypass to Pedal to do so. This avoids the entire bass portion of your organ
being suppressed by the wah wah. It also avoids undesirable intermodulation artefacts,
when utilizing the overdrive effect.
If you select None, the entire output of the organ is processed, as if you had plugged
the B3’s mono output into a Leslie cabinet.
Equalizer
As a Logic user, you won’t have any questions about the EQ Low, EQ Mid and EQ High
controls. The EQ algorithm is derived from Logic’s Fat EQ. EQ Level is a master EQ
volume control, allowing you to specify any gain level to the distortion effect. See
“Effect Chain and Effects Bypass” on page 471 for details on effects routing options.

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Reverb
Box, Small, Medium, Large, Big, and Spring are the names of the reverb algorithms.

Reverb level is defined by the Reverb parameter. A Reverb =0 value conserves
processing resources.
You can also select Bypass in the Reverb Mode pull-down menu, if you want to disable
the reverb without changing its level.
The reverb is always patched after the EQ, wah wah and distortion effects, but before
the rotor effect. This means that the reverb always sounds as if it is played back
through the rotor speaker. To hear the reverb after the rotor, switch off the organ’s
reverb, and use an aux send to apply reverb to the Audio Instrument object.
Wah
The name “Wah Wah” comes from the sound it produces. It has been a popular effect
with electric guitarists since the days of Jimi Hendrix. The pedal controls the cutoff
frequency of a band pass, low pass, or—less commonly—high pass filter. The wah wah
pedal has also been used extensively with the Hammond organ.

MIDI Pedal Control
It is recommended that you permanently attach an Expression Pedal to your MIDI
master keyboard. Your master keyboard should transmit MIDI control change #11
(Expression real-time volume), if you attach an Expression Pedal to the Expression jack
and move it. This would normally be used to control the volume while playing.
If you program an EVB3 setting, set the Expression parameter to 0, and then define a
wah wah effect (with controller 11 controlling the wah wah’s cutoff frequency), you can
control the wah wah with the pedal—without having to program anything on the
master keyboard.
Read more on this in the “MIDI Setup” on page 456. You should also consult the users
manual of your keyboard.
Wah Wah Control with other MIDI-Controllers or Aftertouch
You can use any MIDI control change message to control the wah wah effect. You can
select any controller number and Channel Aftertouch (Touch) in the CC field.

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Mode
Mode allows you to enable/disable the wah wah effect. If you select Mode off, the effect
is disabled. There are six different filter types available:
• ResoLP (Resonating Low Pass Filter)
In this mode, the wah wah will work as a resonance-capable low pass filter. At the
minimum pedal position, only low frequencies can pass.
• ResoHP (Resonating High Pass Filter)
In this mode, the wah wah will work as a resonance-capable high pass filter. At the
maximum pedal position, only high frequencies can pass.
• Peak
In this mode, the wah wah will work as a peak (bell) filter. Frequencies inside the
center frequency, which is controlled by the selected MIDI controller, will be
emphasized.
• CryB
This setting mimics the sound of the popular Cry Baby wah.
• Morley 1
This setting mimics the sound of a popular wah pedal, manufactured by Morley. It
features a slight “peak” characteristic.
• Morley 2
This setting mimics the sound of the Morley distortion wah pedal. It has a constant
Q.
Range
Range controls the sensitivity of the wah to controller movements. If you only intend to
make slight alterations to the cutoff frequency, choose a small value.
Bite
Bite is the name for the resonance parameter of the wah filter. You’ll know the
parameter’s meaning, if you’re a synthesizer player. Anyway, check it out: The cutoff
frequency is boosted. High values make the wah sound more aggressive.
Distortion
The distortion effect simulates an overdriven two-stage tube amplifier. Its primary role
is the simulation of the Leslie amplifier—or whatever amp might be used to feed the
Leslie speaker cabinet.

Type offers the choice between three different tube amp types: Growl, Bity, and Nasty.
Growl simulates a two-stage tube amplifier, resembling the Leslie 122 Model, the classic
partner for the Hammond B3 organ. Bity is reminiscent of a bluesy guitar amp. Nasty
delivers hard distortions, and is well suited for very aggressive sounds.

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The Tone control only affects the distorted portion of the sound, while the dry signal
portion remains unaffected. This allows for very warm overdriven sounds that won’t
become “scratchy” if you try to get more treble out of the instrument.
Drive controls the amount of overdrive distortion. The output level is automatically
compensated for, so there’s no need for another master volume control adjustment
facility.
Rotor Cabinet
The Hammond story can’t be fully told without a chapter on the rotor sound cabinets,
manufactured by Leslie. In fact, playing the B3 organ without a rotor cabinet is viewed
as a “special effect” these days. The EVB3’s rotor cabinet section simulates not only the
speaker cabinet itself, but also the microphones which pick up the sound.

Cabinet
There are five settings available:
Off
In the Off setting, there’s no rotor effect at all and you’ll hear the direct output signal of
the organ, and/or the other effects. There’s an alternative to switching the rotor effect
off: in the Brake mode, the speakers don’t rotate, but are still picked up by the
simulated microphones, in a random position. (see paragraph “Rotor Speed” below).
Wood
The Wood setting mimics a Leslie with a wooden enclosure, and sounds like the Leslie
122 or 147 models.
Proline
The Proline setting mimics a Leslie with a more open enclosure, similar to a Leslie 760
model.
Single
In the Single setting, the sound of a Leslie with a single, full-range, rotor is simulated.
The sound resembles the Leslie 825 model.
Split
In the Split setting, the bass rotor’s signal is routed more to the left side, and the treble
rotor’s signal is routed more to the right side.
Wood & Horn IR
This setting uses an Impulse Response of a Leslie with a wooden enclosure.

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Proline & Horn IR
This setting uses an Impulse Response of a Leslie with a more open enclosure.
Split & Horn IR
This setting uses an Impulse Response of a Leslie with the bass rotor signal routed
more to the left side, and the treble rotor signal routed more to the right side.
Rotor Speed
The Rotor Speed switches work as follows: Chorale = slow movement, Tremolo = fast,
and Brake stops the rotor.
Speed Control
Organ players generally switch between Choral and Tremolo. With the EVB3, you can
switch speeds remotely via controllers as the Modulation Wheel (ModWhl), Channel
Aftertouch (Touch), Sustain Pedal (SusPdl), or, alternately, exclusively via mouse (off ).
ModWheel—If you choose the modulation wheel, you can set all three speed settings
in real-time. Brake is selected around the modulation wheel’s center position, while
Choral is selected in the lower, and Tremolo in the upper third of the modulation
wheel’s travel.
All other entries in the Speed Control pull-down menu work as follows: They toggle
between Tremolo and the speed set with the Rotor Speed radio buttons, that means
either between Choral and Tremolo, or between Brake and Tremolo. If Rotor Speed is set
to Tremolo, you will toggle between Tremolo and Choral. The difference between the
entries is how the toggle action takes place.
ModWhl Toggle—Toggles as soon as the Mod Wheel exceeds its center value on its way
from its low position to its high position. If the Mod Wheel passes its center value on its
way from its high to its low position, there will be no toggle This has been
implemented for Roland keyboards with combined Pitch Bend- and Modulation levers.
ModWhl Temp—Toggles as soon as the Mod Wheel passes its center value, regardless if
you move the Mod Wheel from high to low or from low to high. This has been
implemented for Roland keyboards with combined Pitch Bend- and Modulation levers.
Touch—Toggles as soon as you press Aftertouch. If you release Aftertouch, there will be
no toggle.
Touch Temp—Toggles as soon as you press Aftertouch. If you release Aftertouch, there
will be another toggle.
SusPdl Toggle—Toggles as soon as you press the Sustain Pedal. If you release the
Sustain Pedal, there will be no toggle.
SusPdl Temp—Toggles as soon as you press the Sustain Pedal. If you release the Sustain
Pedal, there will be another toggle.

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Note: The Hammond B3 isn’t equipped with a Sustain Pedal. This allows you to make
use of your MIDI master keyboard’s Sustain Pedal as a speed switch.
Rotor Fast Rate
Rotor Fast Rate defines the maximum possible rotor speed (Tremolo). While moving the
slider, the Tremolo rotation speed is displayed in Hertz.
Acc/Dec Scale
The Leslie motors need to physically accelerate and decelerate the speaker horns in the
cabinets, and their power to do so is limited. Acc/Dec Scale determines the speed at
which the motors can accelerate the rotors (time it takes to get the rotors up to a
determined speed), and the length of time it takes for them to slow down. If the slider
is set to its far left position, you can switch to the preset speed immediately. If you drag
the slider to the right, speed changes take more time to occur.
In the default position of 1, which, like any default in Logic, can be set by clicking the
slider while holding Option, the behavior is Leslie-like.
Horn Deflector
If you look inside a Leslie cabinet, you’ll see a double horn, with a deflector at the horn
mouth. This deflector “makes” the Leslie sound. Some people, however, removed it to
alter the Leslie sound, as removal of the deflector increases amplitude modulation, and
decreases frequency modulation. This parameter allows you to switch the deflectors on
and off, without needing to order spare deflectors.
Mic Distance/Mic Angle
The Mic Angle slider defines the stereo image, by changing the angle of the simulated
microphones. An angle of 0° results in a mono sound, while an angle of 180° causes
phase cancellations. Experienced sound engineers tend to avoid wide spreads.
Mic Distance makes the sound darker, and less defined, when set to higher values. This
is typical of microphones, when positioned further from the sound source.
Motor Control
The original Leslie amps have simple AC outlets for connection of the motor plugs.
Modification is easy, you just remove or swap the plugs, for the motors.
Note: With Single Cabinet, the Motor Ctrl setting is irrelevant, because there are no
separate bass and treble rotors in a Single Cabinet.
A “fancy” variation is the inverse mode (Motor Ctrl = inv). If you switch to Tremolo, the
bass compartment rotates with fast speed, while the horn compartment rotates with
slow speed, and vice versa in Chorale mode. In Brake mode, both rotors will stop.
The 910, or “Memphis” mode, stops the bass drum rotation at slow speed, while the
speed of the horn compartment can be switched. This may be desirable, if you’re after a
solid bass sound, but still want treble movement.

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Another Mode is Sync: The acceleration and deceleration of the horn and bass drum
are about the same. This sounds as if the two were locked, but is only clearly audible
during acceleration/deceleration.
Additional Parameters
A number of additional parameters are accessible via the 001/011 button at the top of
the EVB3 Plug-in window. The Upper/Lower Stop Position sliders allow you to set an
exact stop position for the Leslie horn or bass rotator, respectively. This is something
that the original Leslie could not do, sometimes resulting in a horn that was aimed at
the back of the cabinet when it came to a halt, and a less than desirable sound. The
Velo to Click slider allows you to set the velocity sensitivity of the Click parameters (see
“Click Parameters” section, from page 494 onwards).

MIDI Controller Assignments
MIDI controller assignments are only updated if the default setting is loaded, or if a
setting that was saved with a song is loaded.
All parameters that allow you to select a MIDI controller feature a Learn entry. If this
option is selected, the parameter will automatically be assigned to the first appropriate
incoming MIDI data message.
Note: As the new entry is added at the top of the list, existing automation data needs
to be increased by one.
In most circumstances, Logic’s track automation is your best option for the recording of
manual movements in the Plug-in window.
MIDI automation is the better option when you wish to make use of your hardware
MIDI drawbar organ as a remote control for the EVB3. The following tables outline the
MIDI controller assignments of the EVB3. The assignments are optimized for use with
several popular MIDI drawbar organ models. In order to match the default settings of
all popular organs, some parameters are assigned twice. You need not concern yourself
about this.
If you use a Hammond XB-series organ, switch to MIDI Mode HS. If you use a Roland or
Korg organ, choose RK.

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This table describes the MIDI Control Change Message number assignment when MIDI
Mode is set to RK. This is the correct setting if you use a Roland VK series or Korg CX-3
drawbar organ as a remote controller for the EVB3.
Controller Number

MIDI Mode RK: Parameter Name

Drawbar 16'

Drawbar 5 1/3'

Drawbar 8'

Drawbar 4'

Drawbar 2 2/3'

Drawbar 2'

Drawbar 1 3/5'

Drawbar 1 1/3'

Drawbar 1'

Rotor Cabinet
80, 92

Chorale/Brake/Tremolo

Chorale/Brake

Reverb
82

Reverb Level

Vibrato
85

Upper Vibrato on/off

Lower Vibrato on/off

Chorus Vibrato Type

Percussion
94

on/off

2nd/3rd

102

Percussion Volume

103

Percussion Time

Equalizer
104

EQ Low

105

EQ Mid

106

EQ Hi

107

EQ Level

Wah
108

Wah Mode

109

Wah Bite

Distortion
110

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Chapter 26 EVB3

Distortion Type

Controller Number

MIDI Mode RK: Parameter Name

111

Distortion Drive

112

Distortion Tone

Click Levels
113

Click On Level

114

Click Off Level

Balance
115

Main Volume

116

Lower Volume

117

Pedal Volume

Rotor Fast Rate
118

Rotor Fast Rate

Only the controller assignments 80 to 82 differ, when MIDI Mode is set to HS. This
setting matches the controller mapping of Hammond XB-series organs. The other
controllers remain as described above. Switch MIDI Data Reduction off while recording
these data from an XB-Series organ (File > Song Settings > Recording Settings).
Controller Number

MIDI Mode HS Drawbar Assignment

All Upper Drawbars

All Lower Drawbars

Pedal Drawbars, Scanner Vibrato, Bass Filter

This table describes the MIDI Control Change Message number assignment when MIDI
Mode is set to NI. This setting matches the controller mapping of the Native
Instruments B4D controller.
Controller Number

MIDI Mode 0: Parameter Name

Upper Drawbar 16'

Upper Drawbar 5 1/3'

Upper Drawbar 8'

Upper Drawbar 4'

Upper Drawbar 2 2/3'

Upper Drawbar 2'

Upper Drawbar 1 3/5'

Upper Drawbar 1 1/3'

Upper Drawbar 1'

Lower Drawbar 16'

Lower Drawbar 5 1/3'

Lower Drawbar 8'

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Controller Number

MIDI Mode 0: Parameter Name

Lower Drawbar 4'

Lower Drawbar 2 2/3'

Lower Drawbar 2'

Lower Drawbar 1 3/5'

Lower Drawbar 1 1/3'

Lower Drawbar 1'

Vibrato
31

Upper Vibrato on/off

Lower Vibrato on/off

Brightness

Vibrato

Attack Time

Chorus Intensity

Percussion
Sostenuto

Percussion on/off

Release Time

Percussion Harmonic (2nd/3rd)

Sound Variation

Percussion Volume

Harmonic Content

Percussion Time

Equalizer
90

EQ Low

EQ Mid

EQ High

Distortion/Click
76

Distortion Drive

Distortion Tone

Click On Level

Leslie

480

Pan MSB

Microphone Angle

Microphone Distance

GP 8

Leslie Accelerate/Decelerate

GP 7

Leslie Fast

ModWheel MSB

Leslie Speed

Controls brake functionality: If Value = 0.0, switch Leslie to Brake.
All other values switch Leslie to previous speed.

Chapter 26 EVB3

Additive Synthesis With Drawbars
The Hammond B3 is the classic drawbar organ. As with an acoustic pipe organ, the
registers (drawbars, or “stops” on a pipe organ) can be pulled out, in order to engage
them. But in contrast to a pipe organ, the B3 allows seamless mixing of any drawbar
registers. The more you drag the drawbars down, the louder they will become.
Despite characteristics such as key clicks, intonation undulations, distortions, and
crosstalk (which are emulated by the EVB3), playing a single note, with a single register,
results in a pure sine tone. Mixing harmonic sine tones results in more complex spectra,
and is known as “additive synthesis”. Organs—even acoustic pipe organs—can be
regarded as additive synthesizers. There are, however, several limitations that need to
be considered before viewing the instrument in this way. These limitations, on the
other hand, constitute the character of any real musical instrument, loaded with charm.
The naming of the drawbars is derived from the length of organ pipes, measured in
feet ('). This naming convention is still used with electronic musical instruments.
Halving the length of a pipe doubles its frequency. Doubling the frequency means
nothing other than: one octave up.
The lowest register, 16' (far left, brown drawbar), and the higher octaves 8', 4', 2' and 1'
(white drawbars) can be freely mixed, in any combination. 16' is commonly described as
the “sub-octave”. When we’re regarding this register as the fundamental, the octave
above 8' is the second partial, 4' the fourth, 2' the eighth and 1' the sixteenth partial.
With the 5 1/3' register—the second brown drawbar—you can add the third partial.
This is the fifth above the 8'. Basically, the drawbars are arranged by pitch, but there is,
however, an exception. The second drawbar (5 1/3') is sounding a fifth higher than the
third drawbar. See the “Residual Effect” on page 482, for an explanation.
2 2/3' gives the sixth, 1 3/5' the tenth and 1 1/3' the twelfth partial. So the
electromechanical tone-wheel organ gives you the partials 1 (16'), 2 (8'), 3 (5 1/3'), 4 (4'),
6 (2 2/3'), 8 (2'), 10 (1 3/5'), 12 (1 1/3') and 16 (1'). As you can see, the harmonic spectrum
is nowhere near “complete”. That’s the reason why overdrive distortion effects are so
popular with electromechanical tone-wheel organs—they enrich the harmonic spectra
by generating more partials.
Note: The term “partial” is basically the same as “harmonic”, but they are counted in a
slightly different way. The fundamental is counted as the first partial. its octave, twice
the frequency, is the second partial, but is known as the first harmonic. The fifth partial
oscillates at five times the frequency of the fundamental. The fifth partial is known as
the fourth harmonic, because with harmonics, the fundamental is not counted (which
makes the term “harmonic” less practical to use).

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Note: 2 2/3' is the fifth over 4'. 1 3/5', is the major third over 2'. 1 1/3' is the fifth over 2'.
In the bass range, this can lead to inharmonic tones, especially when playing bass lines
in a minor key. This is because mixing 2', 1 3/5' and 1 1/3' results in a major chord.

Residual Effect
The residual effect is a psychoacoustic phenomenon. Human beings can perceive the
pitch of a note, even when the fundamental is completely missing. If we didn’t “hear”
this way, it would make listening to music with a kitchen radio impossible. Its speaker
will never play back the fundamental of the bass line, as this frequency is far below the
range that the speaker can transmit. If you pull out all registers of the drawbar organ,
except for the fundamental—16', you’ll still perceive the same pitch. The sound
becomes thinner, with less bass and less warmth, but the pitch remains the same.
Setting drawbar registrations often involves this psychoacoustic phenomenon. In the
lower octaves, mixing the 8' and 5 1/3' sine drawbars creates the illusion of a 16' sound,
although the frequency is missing. Old pipe organs also make use of the residual effect,
by combining two smaller pipes, eliminating the need for long, heavy, air-hungry, and
expensive giant pipes. This tradition is continued in modern organs, and is the reason
for arranging the 5 1/3' under 8: The 5 1/3' tends to create the illusion of a pitch that is
one octave lower than 8'.

A Short Hammond Organ Story
Three inventions inspired Laurens Hammond (1895–1973), a manufacturer of electric
clocks, to construct and market a compact electro-mechanical organ with tone wheel
sound generation. The Telharmonium by Thaddeus Cahill was the musical inspiration,
Henry Ford’s mass production methods, and the domestic synchron clock motor were
the other factors.
The Telharmonium was the first musical instrument that made use of
electromechanical sound generation techniques. In the year 1900, its man-sized tone
wheel generators filled a two-storey building in New York. For a short period around
this time, subscribers could order Telharmonium music over the New York telephone
network (the streaming audio system of the time). The only amplification tool was the
telephone’s mechanical diaphragm, as a proper tube amplifer and acceptable speakers
had not yet been invented. The Telharmonium was a commercial flop but its historical
status as the predecessor of modern electronic musical instruments is undeniable. The
Telharmonium also introduced the principles of electronic additive synthesis (see
“Additive Synthesis With Drawbars” on page 481).
Laurens Hammond began producing organs in 1935, in Chicago, Illinois, making use of
the same sound generation method. The differences were; much smaller tone
generators, and fewer registers. The patent for his model A organ dates from 1934.

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Hammond also holds the patent for the electro-mechanical spring reverb, still found in
countless guitar amplifiers today!
The Hammond B3 was manufactured between 1955 and 1974. It is the Hammond
model preferred by jazz and rock organ players such as: Fats Waller, Wild Bill Davis,
Brother Jack McDuff, Jimmy Smith, Keith Emerson, Jon Lord, Brian Auger, Steve
Winwood, Joey DeFrancesco, and Barbara Dennerlein. In addition to the B3, there are a
number of smaller Hammond instruments, known as the “spinet” series (M3, M100,
L100, T100). Bigger console models, many of which were designed to suit the needs of
American (USA) churches or theatres (H100, X66, X77, E100, R100, G-100), were also
manufactured.
The production of electro-mechanical organs ceased in 1974. Thereafter, Hammond
built fully electronic organs. Today, people at Hammond-Suzuki are more conscious of
their glorious tradition and produce fine electronic drawbar organs. In 2002, they even
introduced a new digital B3 model which mimics the design and functions of the
classic B3 (except for the weight). The new B3 utilizes a real, mechanical, rotor speaker
cabinet.

Tonewheel Sound Generation
Tonewheel sound generation resembles that of a siren. Of course, there’s no air being
blown through the holes of a revolving wheel. Rather, an electro-magnetic pickup,
much like a guitar pickup is used.
A notched metal wheel, called a tone wheel, revolves at the end of a magnetized rod.
The “teeth” of the wheel cause variations in the magnetic field, inducing an electrical
voltage. This voltage/tone is then filtered, sent through the manuals, amplified, has
vibrato and expression applied to it, and is then amplified.
A long drive shaft is driven by an AC synchronous motor. 24 driving gears are attached
to the shaft, with 12 different gear sizes. These gears drive the tone wheels. The
frequency depends on the gear ratios, and the number of notches in the wheels. The
Hammond is tuned to an almost exact well-tempered scale.
As with pipe organs that feature “multiplexed registers”, the Hammond organ uses
certain generators for more than one purpose. Some high frequency wheels serve as
the fundamental for high notes, and provide harmonics for lower notes. This has a
positive impact on the overall organ sound, avoids detuning and stabilizes levels.

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The Leslie
Don Leslie developed his rotor cabinets in 1937, and began marketing them in 1940.
Laurens Hammond wasn’t keen on the concept of rotating speakers at all! Leslie’s
approach was to simulate a variety of locations in the pipes, resulting in a new spatial
perception for every note. The rotor speaker cabinets could simulate this effect, and the
sense of space that they impart is incomparable, when placed side-by-side with any
fixed speaker. The periodic undulations in sound and volume, and the vibrato caused
by the doppler effect aren’t all there is to the Leslie “sound”—it’s the space effect, too!
The first Leslie, the model 30, had no chorale, just tremolo and stop. The Chorale idea
(which came much later) was borne of a desire to add a vibrato to the organ. Chorale
offers far more than a simple vibrato, and was first introduced to the market with the
122/147 models. At this time, Leslie also added the “Voice of the pipe organ” label to his
cabinets.
It wasn’t until 1980 that the two companies and brand names came together, six years
after the last tonewheel organ was built. Mechanical Leslie rotor cabinets are still being
built today, by the Hammond-Suzuki company. Even the newest digital B3 model is
combined with a real, mechanical, Leslie cabinet.
As an interesting piece of trivia, Don Leslie never actually owned a Hammond organ!
If you’re interested in every detail of the Hammond organ’s history, models, facts, and
recommendations for buyers, you should take a look at “The Hammond Organ—Beauty
in the B”. This book, by Mark Vail, is highly recommended.

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EVD6

This chapter covers everything about the EVD6, a virtual
emulation of the Hohner Clavinet D6.
The sound of the Hohner Clavinet D6 is synonymous with funk, but was also
popularized in the rock, pop, and electric jazz of the 1970s, by artists and groups such
as: Stevie Wonder, Herbie Hancock, Keith Emerson, Foreigner, and the Commodores. If
you’ve heard “Superstition” or “Higher Ground” by Stevie Wonder, then you’ll know that
the D6 is the funkiest instrument alive!

The EVD6—Concept and Functions
The EVD6’s synthesis engine emulates the sound of the Hohner D6 Clavinet, and does
not make use of any sampling technology. It improves on the original in that it can be
used in stereo, and on a noise level, there’s no comparison.
The dynamics and scaling of the sounds, over the entire 60-key range (F to E) of the
original instrument, has been extended across the full MIDI range (127 notes).
The EVD6’s engine also simulates the various string buzzes, key clicks, and the tone of
the pickups found in the original instrument. It synthesizes the “pluck” and “bite” of the
attack phase, as well as the “sticking” of the hammer pads. The sound generator reacts
smoothly, musically, and precisely to the 127 steps of velocity sensitivity, as defined in
the MIDI specification. You can almost feel the strings beneath your keyboard!
The extensive set of String parameters allow you to radically alter the tone of the EVD6,
enabling you to simulate an ageing Clavinet, or to create new “instruments”. The EVD6 is
capable of some truly unique sounds, which you’ll discover when exploring it, and
auditioning some of the included settings.

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You’ll appreciate the perfect integration of the EVD6 into Logic. In use, it’s much easier
to handle than a real world Clavinet. There’s no need to transport a bulky and heavy
instrument, or to attach any cables to it. In addition, the EVD6 eliminates the problems
of reliability, getting new parts, and tuning—all of which are becoming increasingly
difficult with the original instruments.
You will also discover an integrated effects processor incorporated into the EVD6’s
luxurious front panel, which provides a number of classic effects popularly-used with
the original Clavinet. The algorithms featured in the effects processor have been
adapted, and optimized, for the EVD6. Included are: a great sounding Wah,
Modulation, and Distortion circuit.

Parameters of the EVD6

Most of the “slider” parameters on the EVD6 interface are mapped to zero-centered
ranges—if a slider is in its middle “neutral” position, it doesn’t affect the base sound of
the selected EVD6-model. If the slider is moved to the left or right, it will scale the
original parameter value positively or negatively by that amount.
The EVD6 front panel can be broken down into five main sections, namely: the Global
silver panel section at the bottom, the Excite/Click and String parameters to the top left,
the Pickup window in the top center, and the Effects section at the top right.

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Global Parameters
The Global Parameters are found in the lower-left portion of the EVD6 Graphical User
Interface (GUI).

Voices
The voices parameter allows you to set the maximum number of voices that can sound
simultaneously. Lowering the value of this parameter limits the polyphony and also the
processing requirements of the EVD6. Minimal CPU power is used when the instrument
is operated monophonically. There are two monophonic settings: mono and legato.
Each setting provides only one voice for playing the EVD6. In the mono setting, the
EVD6 voice is triggered each time a key is pressed. In the legato mode, the EVD6 sound
shaping processes is not triggered if the notes are played legato—only the pitch
changes. If the notes are played staccato, an EVD6 voice with all sound shaping
processes is triggered. The maximum setting is 24, allowing for sustained glissandi. A
setting of 24 will, of course, be more processor-intensive. 10 is the default. Click-hold,
and use your mouse as a slider to adjust.

Tune
The global Tune setting works in cent steps. A value of 0 equals concert-pitch A 440 Hz.
The range is ±50 cents or, in more “musical” terms, plus/minus half a semitone. For
transpositions in semitone or octave steps, please use the Instrument Parameter box in
the Arrange window, as per any standard MIDI instrument. Click-hold, and use your
mouse as a slider to adjust.

Bender
This parameter determines the bend range, in semitone steps. Click-hold, and use your
mouse as a slider to adjust.

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Warmth
Amount of random deviation from an equal-tempered scale. High values add “life” to
sounds. It can be useful for simulating an instrument which has not been tuned for a
while, or for slightly “thickening” a sound. When playing chords, the Warmth parameter
creates the warm detuning or beating effect between the chord’s notes. Click-hold, and
use your mouse as a slider to adjust.

Stretch
The EVD6 is tuned to an equal-tempered scale. As a deviation from this standard
tuning, you can stretch the tuning in the bass and treble ends of the sound. This
simulates the way stringed keyboard instruments such as pianos are tuned, attempting
to find a more constant tuning balance between high and low notes. The stretch
feature was included for situations where you may wish to use the EVD6 alongside
acoustic pianos. When arranged in conjunction with an orchestra or synthesizers, the
stretch tuning facility should not be used. Click-hold, and use your mouse as a slider to
adjust.

Note: The tones of clavinets, harpsichords, and pianos have inharmonicities in their
harmonic structure. The frequencies of these overtones (harmonics) are not exact,
whole-number multiples of the base frequency, as Pythagorean theory dictates. They
are only approximate and are, in fact, a little higher. The overtones of lower (tuned)
notes, therefore, are more closely related to the main frequencies of the upper notes.
Also see “Stiffness/Inharmonicity” on page 496.
Note: When applying Warmth and Stretch, you should consider that these parameters
may result in a detuned sound, which is similar to the overuse of a chorus effect.
Pressure
On an original D6, applying pressure (aftertouch) to a depressed key raises the pitch
slightly. The Pressure parameter allows you to do this, or alternately lower the pitch by
pressure. Click-hold, and use your mouse as a slider to adjust. Range: −1.00 to +1.00

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Filter Switches
The four filter switches emulate the original switches on the D6, with one exception.
When all switches are set to off, you’ll hear the unfiltered sound, rather than the original
D6’s “humming silence”. Simply click anywhere on each switch to toggle between its
on/off position. Active switches are indicated by pale green lettering, and by being
depressed towards the bottom of the Plug-in window. You may use the filter switches
in any combination of on/off positions.

•
•
•
•

Brilliant—makes the sound nasal/cuts bass.
Treble—makes the sound sharper/cuts bass more gently.
Medium—makes the sound thinner/slight bass reduction.
Soft—makes the sound softer/more muted.

Pickup Switches
As with the original D6, the two pickups can be used in different modes. The AB and CD
switches are used to change modes. The internal wiring of the two pickups is changed
in accordance with the different switch positions, and with it, the sound at the
combined pickup output. The EVD6 features an additional menu that displays the
current pickup mode above the pickup switches. More information on the use of these
parameters/the pickups is found in “Pickup Parameters” on page 497.

C/D Switch

A/B Switch

What It Does

down

“neck” pickup—warm sound

down

“bridge” pickup—bright sound

both pickups—full sound

down

both pickups out of phase—thin sound

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Stereo Spread
Pickup—While the original D6 only has a mono output, the EVD6 has stereo
capabilities. When both pickups are active (upper+lower and upper−lower modes), the
two pickup signals can be spread across the stereo spectrum. To adjust the position of
the stereo spread, click-hold on the up/down arrows in the lower half of the circular
button—in the Pickup section.

Turning the Pickup Spread control up will move the signals of both pickups away from
the center position—one to the right, and the other to the left. Extreme left/right
positions are reached when Spread is set to its maximum value. Range: 0.00 (center, no
effect) to 1.00 (full left/right stereo).
More information on the use of the pickups is found in “Pickup Parameters” on
page 497.
Key—This parameter allows a keyscale modulation of the panning position—panning
is determined by keyboard position. The center position is MIDI note 60 (Yamaha C3).
To adjust the Key(board) position, click-hold on the up/down arrows in the top half of
the circular button—in the Key section.
When fully turned up, the extreme left/right position will be reached at MIDI note 60
±30 semitones. Range: (center, no effect) to 1.00 (full left/right stereo).
Note: You can use both spread types at the same time. They will automatically be
mixed.
Note: The effect of both stereo spread parameters is reflected graphically in the area
around the circular Stereo Spread button.

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Model
The Model parameter allows you to select a basic type of tone, or “model”. Each model
has its own unique tonal characteristic, and each is suitable for the creation of very
different sounds. Each model is an instrument in its own right, and can immediately be
played, without any further editing. We will discuss each model below, and encourage
you to experiment with each.

To select a model, simply click-hold in the area between the Stereo Spread and Level
controls, and make your choice from the pop-up list. Release the mouse button once
your selection is made. All EVD6 parameters are available to further shape the tonal
character of the model.
In some respects, the Model parameter can be viewed much like making an oscillator
waveform selection in a synthesizer. As with raw synthesizer waveforms, the editing
parameters can affect the model quite differently. As an example, particular Excite
settings may make one model more “nasal” sounding, and another model more “noisy”.
These behavioral differences are a result of the unique harmonic structures used by
each model.
The Models
Class(ic) D6
An almost 1:1 emulation of the original D6. It includes string noises on long decays,
and realistic release behavior, following the release of the key(s). Each D6 was unique in
its way, so feel free to adjust the many sound shaping controls, in order to match the
sound of D6 units that you have used, or heard.
Old D6
This model emulates a well-worn D6. Hammers and strings are a bit aged, and worn.
The sound of the sticky hammer heads is emulated, as well as the typically “richer”
sound in the bass range.
Sharp D6
Very sharp with a lot of bite—nice with wah wah and phaser.
Mello(w) D6
As the name suggests, a mellow fellow.
Basic
Basic, simple clavinet

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Domin(ation)
A powerful model with a strong and punchy attack—reacts more aggressively to
velocity than other models.
GuruFnk (Guru Funk)
In the lower bass-octave ranges, the string oscillations become increasingly resonant
over time, until they finally collapse (after 20 to 30 seconds). Higher notes have a much
shorter decay, which also applies to their resonating behavior. This model invites heavy,
funk-style bass playing in the lower octaves. It’s nice with a little phaser, and sustained
chords, when playing low bass notes. Adding a Logic Delay plug-in is also a great
option!
Harpsi(chord)
Harpsichord-like model.
Pluck
Plucked string—changing the pick-up positions allows further modifications, making
the sound more guitar-like. “Harp” style sounds are also possible, by positioning the
lower pick-up around the mid position. To get a harp sound increase String Decay,
Release, and Excite Shape and decrease Excite Brilliance.
(Tuned) Wood
Somewhat wooden, thin, and with some inharmonic overtones. Can sound slightly
detuned in some contexts.
Ltl (Little) India
Sitar-like sound, rich in resonance.
S(tring) Bells
A bell model with strong inharmonic overtones (inharmonicities).
Dulcimer
Dulcimer-like model.
Picked
This model emulates a picked nylon string.
Special Notes about the Models
You may note some “zones” on the keyboard where the sound changes significantly
between adjacent keys. This is intentional, and reflects the behavior of some of the real
clavinet models emulated by the EVD6. The original D6 has some strong key-to-key
timbral differences, with the most obvious one being between the highest, wound
string, and the lowest, non-wound string.
If you’re a player who likes the original’s sound, but not the original’s mechanical timbre
jumps, the EVD6 offers a smoothed model—MelloD6.

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When using a setting with both pickups quite close to the upper end of the strings and
Brilliant + Treble filter switches active, the fundamental tone is quite weak in the output
signal. As such, you will mostly hear the overtones that are not exactly in tune for
inharmonic models (Wood, for example). Try moving the pickups to the center, and
deactivate all filter switches to circumvent this detuned effect.

Level
Sets the (post -Effects) level, in dB (decibels). Click-hold, and drag, to adjust. If the MIDI
controller used for Expression is not assigned to Wah or Damper, it is used to scale the
output level.

Damper Wheel and Damper Ctrl
The original D6 features a damper slider on the right-hand side of the keyboard that
allows the player to create muted string sounds. Click-hold, and drag, or make use of a
MIDI controller, such as your keyboard’s mod wheel, to adjust the Damper Wheel. The
Damper Wheel position is saved with the sound.

The Damper Ctrl (number) parameter allows you to select the MIDI controller that
moves the Damper Wheel. Click-hold, and make a controller number/name selection
from the pull-down menu. Release the mouse button, once your selection is made.
Note: You can use MIDI Velocity to control the Damper Wheel. Just select the Velocity
parameter from the pull-down menu.
The software-wheel is moved on-screen, when controlled via MIDI. MIDI control can be
disabled by selecting the off option, found in the pull-down menu.
Note: The Wah Ctrl parameter is discussed on page 501.

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Velocity Curve
There are nine preset velocity curves available for the EVD6. These allow you to set up a
curve which is suitable for your playing style, or the sound.
The nine curves available are: fix25%, fix50%, fix75%, fix100%, convex1, convex2, linear
(the default), concave1, and concave2.

Excite Parameters
Excite describes the string excitation, the physical power which stimulates the string to
oscillate.

Shape
Shape adjusts the attack shape, allowing you to simulate the hardness of the rubber
hammers in the original D6. As the instrument aged, the hammers would become
worn, split, and so on, which had an impact on the overall brightness/tone of the D6.
Negative values (to the left) provide a softer attack, while positive values result in a
harder attack. Range: −1.00 to +1.00
Brilliance
Controls the harmonic content of string excitation. Positive values (to the right) result in
a sharper sound. Negative values result in a more muted sound. Range: −1.00 to +1.00

Click Parameters
The rubber hammers of the original D6 age and decay, just like piano hammer felts.
Well-loved (worn out) D6’s produce a distinctive click when a key is released. This is due
to the string sticking to the rubber hammer, before being released. The characteristics
of this release click are part of each model, and can be finely adjusted with the
following parameters.

Intensity
Positive values increase the level of the release click above the original model setting.
Negative values reduce the level—a value of −1.00 equals no release click. If you’d like
to simulate an old D6, increase the value, by moving the slider to the right.
Range: −1.00 to +1.00

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Random
Controls the amount of random click level variations across the keyboard. This slider
simulates the wearing of some hammers, but not all of them, emulating the real-world
wear and tear of the original. The further to the right the slider is moved, the greater
the variation between key clicks on some keys. If all the way to the left, all keys have
the same level of key click. Range: 0.00 to +1.00
Velocity
The Velocity parameter controls the level of release click modulation by velocity—Note
On velocity or Note Off/release velocity. Range: 0.00 to 1.00. The selection of note on/
off information as the modulation source is determined by the KeyOn/Key Off buttons.
KeyOn/KeyOff Button
Press the appropriate button to select the type of velocity information that should be
used for release click level modulation—press the KeyOn button, if you wish to use
your attack velocity (how hard you hit the keyboard) as the value for the key click. If
you wish to use your release velocity (how quickly you release the keys on your
keyboard) to determine the value of the key click, press the KeyOff button. This requires
a keyboard with release velocity facilities.
Needless to say, the Velocity parameter must be set to a reasonable level in order for
the KeyOn/KeyOff modulation to be effective.

String Parameters
The behavior of the strings is basically determined by the Model, but the following
parameters allow you to modify several string characteristics, relative to the model
setting. See “Model” on page 491, for further information on model selection.

Decay
Positive values provide a longer Decay time after attacking a note. Negative values
reduce the decay time. Range: −1.00 to +1.00
Release
As per the Decay parameter, but for the Release time (following the physical release of a
key). Range: −1.00 to +1.00

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Damping
The Damping parameter allows you to modify the damping of strings. Damping is
essentially a faster decay for the higher partials/harmonics in a sound, and is a property
of the string material used (high damping for catgut strings, medium damping for
nylon strings, low damping for steel strings). Sonically, damping results in a more
mellow and rounded, or woody sound, dependent on the Model in use. A positive value
will make the sound more mellow, and a negative value will allow more of the higher
partials through, making the sound brighter. Range: −1.00 to +1.00
Tension Mod
Tension modulation is a non-linear effect on strings, which usually results in the pitch
being slightly higher, immediately after being plucked/struck/strummed. It is common
to all stringed instruments, like the D6, guitars, and so on. This string characteristic is
built into each model, but can be further modified with the Tension Mod parameter. The
range of this parameter is quite large, and can be used to obtain weird sound effects
from the EVD6. It can also be used to simulate an out-of-tune Clavinet, or as a quick
and dirty sitar sound, for those “Norwegian Wood” covers. Range: −1.00 to +1.00
Stiffness/Inharmonicity
These two parameters allow you to intensify/reduce the strength of inharmonicity in
the sound. When combined at different levels, these parameters can create metallic,
bell-like sounds, or DX-like electric piano style sounds. They can also be useful for wood
bass sounds. Experiment with both parameters, on each Model.
The higher the level of the Inharmonicity parameter, the lower its threshold to
incoming frequencies. In other words, the Inharmonicity parameter determines the
lowest harmonic, above which inharmonic spectral spreading becomes relevant.
Range: −1.00 to +1.00
Stiffness controls the intensity of this stretching/spectral spreading. Range: −1.00 to
+1.00
The keynote is not affected by these parameters.
Pitch Fall
Due to the physical construction of the original D6, the pitch of each note falls
immediately after releasing the key. The intensity of this effect, which varies with each
model, can be modified with this parameter. To completely deactivate the pitch fall,
regardless of the selected model, set this parameter to the leftmost position (−1.00).
Range: −1.00 to +1.00

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Pickup Parameters
The original D6 is equipped with two electromagnetic pickups, much like those found
in electric guitars: one below the strings (lower) and one above (upper).

Pickup Position
In contrast to the fixed pickups of the original instrument, the EVD6 pickups can be set
to arbitrary positions and angles. To do so, simply click-hold on one end of the desired
pickup (Upper or Lower) and drag the end to another position. Release the mouse
button when done. Both values can be moved simultaneously. To do so click and drag
the point in the middle of the pickup to a new position.
The numerical upper and lower panels, to the top-left of the window, indicate the
current position of each pickup—with respect to the string. A value of 50 (percent)
means that the specific pickup end is positioned above/below the centre of the string,
resulting in a full-bodied tone. When the pickup approaches either end of the string
(values near 0 or 99), the tone becomes thinner.
In the graphical pickup window the strings are aligned from left to right in respect to
pitch—low strings to the left, high strings to the right.
It is recommended that you repeatedly strike a note when moving pickup positions, in
order to hear the effect that the pickup position has on the overall tone of your sound.
Interesting, phaser-like effects can be achieved by automating the pickup positions.
Important: It is possible to cross-over the pickups in the Pickup Position window. This
may lead to a “hole” (non or very soft sounding notes) within your keyboard range. This
is due to a phase-cancellation between the pickups. If you encounter such
cancellations, adjust one (or both) of the pickups until the required notes are playable.

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Pickup Mode
Pressing the AB and CD switches will change the virtual wiring of the two pickups. The
current wiring, the EVD6 calls it Pickup Mode, is displayed in the Pickup Mode panel.
You can also click directly on the Pickup Mode panel, and select the desired mode from
a pull-down menu.

•
•
•
•

C + A = Lower
C + B = Upper
D + A = Lower−Upper
D + B = Lower+Upper

Also see “Stereo Spread” on page 490, and “Pickup Switches” on page 489.

Effects Parameters
No Clavinet simulation would be complete if it didn’t include a selection of effects
processors. The EVD6 doesn’t disappoint in this regard, incorporating three footpedal
effects that have formed an integral part of “classic” Clavinet sounds over the decades.
Each effect was painstakingly modelled on effects pedals that were available in the
heyday of the Clavinet—the 1970’s—ensuring that vintage sound in your
performances.
Needless to say, you can also take advantage of Logic’s extensive range of effect plugins, to further tailor your sound.

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Distortion
The integrated distortion effect can be adjusted in both intensity and tone.
Range: Tone −2000 Hz to 20,000 Hz, Gain −0 dB to 20 dB. Using low Tone and Gain
settings allows the Distortion unit to create warm overdrive effects. Bright and
screaming distortion effects are produced with high Tone and Gain settings.

Compressor
Please note that the Distortion effect is always preceded by a compression circuit
(shown in the panel above the Tone knob—with a ratio of 1:19.5) This allows you to
increase/decrease the perceived gain, to provide the desired input level to the
Distortion circuit.
The compressor allows for really “crunchy” distortions, coupled with wah, or phaser. It
can also be useful for enhancing the keyclick sound, and emphasizing harmonics in the
various models.

Compression Ratio
The Compression Ratio panel allows you to adjust the slope of the compression applied.
To adjust, simply click-hold on the panel, and use the mouse as a slider.
The Compressor is tied to the Distortion effect, and always precedes it. As such, the
Effects Order parameter is very important for placement of this compressor in the
effects chain. Please see “FX Order” on page 502 for further information.
Note: If the Compressor/Distortion is used as the last effect in the chain, and its gain is
turned down, but the Compression Ratio is high, you will effectively compress the
output signal of the EVD6.

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Wah
The typical Wah effect is generated by a dynamically moving filter. The EVD6 offers
simulations of several classic wah effects, as well as some basic filter types. Possible
values are: off, ResoLP, ResoHP, Peak, CryB, Morl1, Morl2. The abbreviations are for
Resonant Low and High Pass filters, Peaking filter, CryBaby, Morley 1, and Morley 2. The
latter three are famous effects pedal models that continue to be manufactured.

Note: The combination of wah, followed by distortion, delivers those sought-after
funky fuzz-wah results.
Wah Mode
Simply click-hold on the (Wah) Mode panel, and select the desired (pedal effect) model
from the pop-up menu.

Range
The Range setting determines the cutoff frequency of the filter (set with Wah Mode).
With Range set to the left, the cutoff will only move in a narrow range. To provide a
wider control range, turn the Range knob to the right.
Envelope (Depth)
An auto wah effect is produced by using an envelope follower to control the filter
cutoff automatically. The envelope shape follows the dynamics of your performance.
The sensitivity of the envelope in respect to your performance and thus the resulting
filter modulation depth is set with the Envelope parameter. Turn Envelope to the right to
increase the modulation depth.

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Wah Ctrl
The Wah Ctrl parameter allows you to define the MIDI Controller (number/name) used
as a manual wah effect control—a MIDI foot controller for example. You can also use
MIDI Velocity to control the wah effect. Just click into the respective parameter field
and select velocity from the ensuing pop-up menu. MIDI-control/coupling can be
disabled by selecting off.

Note: Both envelope and a manual controller can control the wah simultaneously. In
this situation, the effect of the envelope and manual controls are mixed.
As with the Damper, there is a slider named Wah Pedal Position that always represents
the current pedal position. This ensures that the most recent pedal position is saved
with the sound. The Wah Pedal Position Slider is only available in the Controls View. Use
the pull-down menu found in the upper right corner of the gray portion of the Plug-in
window to switch from the Controls view to the Editor view. The Pedal Position can be
automated by either recording the MIDI controller messages, or by using the Track
Automation system.
Modulation
The EVD6 features a Modulation unit with three switchable modulation effect types.

Mode
The Mode panel allows you to select either a Phaser, Flanger or Chorus as the
modulation effect. Click-hold, and make your choice from the pop-up menu.
Phaser
The Rate parameter adjusts the speed of phasing, and the Intensity parameter adjusts
the depth of phasing. Ranges: Rate 0.10 Hz to 10 Hz, Intensity 0 to 100

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High values lead to very deep, self-oscillating phase shifts, for those cutting (and ear
and speaker damaging, so take care!) sounds.
Chorus
The Rate parameter adjusts the speed of the Chorus effect, and the Intensity parameter
adjusts its depth. High Intensity values lead to ensemble-type effects.
Ranges: Rate −0.10 Hz to 10 Hz, Intensity −0 to 100
Flanger
The Rate parameter adjusts the speed of flanging, and the Intensity parameter adjusts
the depth of flanging. Ranges: Rate −0.10 Hz to 10 Hz, Intensity −0 to 100.
FX Order
The order of the serial effects combination can be selected here. The four choices are:

•
•
•
•

WDM—Wah > Distortion > Modulation
DWM—Distortion > Wah > Modulation
MDW—Modulation > Distortion > Wah
WMD—Wah > Modulation > Distortion

Just like the foot pedals which could be freely connected to each other in series, the
EVD6 effect section invites you to experiment.
The freely assignable effect routing is especially useful for selecting whether a distorted
signal shall be wah-filtered, or if the wah-filtered sound shall be distorted (for
screaming sounds), as one example.

Controlling the EVD6 via MIDI
It is possible to control and automate the parameters of the EVD6 and other plug-ins
using the MIDI controls provided by many master keyboards or MIDI fader boxes.
MIDI controller assignments (Wah Ctrl, Damper Ctrl) are only updated if the default
setting is loaded or a setting that was saved with a song is loaded, with one
exception: If Velocity is chosen in a setting, the assignment is updated. When selecting
a setting that has any controller other than velocity assigned, the default value or the
last manually selected value (other than velocity) is used.
All parameters that allow you to select a MIDI controller offer a Learn entry. If this
option is selected, the parameter will automatically be assigned to the first appropriate
incoming MIDI data message.
Note: As the new entry is added at the top of the list, existing automation data needs
to be incremented by one.

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A Brief History of the Clavinet
German Company, Hohner, was the manufacturer of the Clavinet. Hohner were known
mainly for their reed instruments (harmonicas, accordions, melodicas, and so on), but
had made several “classic” keyboards, prior to the first incarnation of the Clavinet,
known as the “Cembalet”.
Musician and inventor, Ernst Zacharias, designed the Cembalet in the 1950’s. This was
intended to be a portable, amplifiable version of the Cembalo, or Harpsichord. It’s
mechanism worked by plucking the end of a flat reed with the key, which was then
picked up and amplified in much the same way as an electric guitar.
A year or two after the Cembalet’s release, two “Pianet” models appeared. Both the “CH”
and “N” models used flat reeds for tone generation, but employed a very different
plucking/striking action. When a key was depressed, it engaged a “sticky pad” with a
foam backing, which actually stuck to the reed. When the key was released, the weight
of the key caused the pad adhesive to free itself from the reed. This made the reed
vibrate, and this vibration was then amplified.
The model “T” Pianet was released several years later, and utilized a soft rubber “suction
pad” on the reeds, rather than the adhesive of the “CH” and “N” models. This method
still had several drawbacks, however, as the dynamics available from the keyboard were
limited. As a further shortcoming, all reeds were damped on release, thus negating the
possibility of obtaining sustain via a foot pedal. Despite these glaring problems, the
sound of the model “T” Pianet was popularized by bands such as the Zombies and
Small Faces, in the 1960’s.
In the years between the releases of the Pianet “N” and “T” models, Zachariah invented
what was to become Hohner’s most successful, and certainly funkiest keyboard—the
Clavinet. The Clavinet was designed to replicate the sound of a Clavichord, but with an
altogether fuller sound. (The Clavichord was notoriously thin sounding)

Original image from the
D6 Users Manual.

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The early models—Clavinet I with built-in amp, Clavinet II with tonal filters, Clavinet “L”
with its bizarre triangular shape, all led to the Clavinet model “C”. This, in turn, was
refined into the D6—a portable, amplifiable keyboard. The D6 used a hammer striking
a string against a metal surface to produce its tone. It had a fully dynamic keyboard—
as the striker is directly underneath the key, meaning the harder you hit, the louder and
more vibrant the tone.
Mention the Clavinet today and most people will automatically think of Stevie
Wonder’s “Superstition”—a recording that owes as much to the D6 as it does to the
artist that wrote and performed it. The D6 was later superseded by the “E7” and the
“Clavinet/Pianet Duo”. These were basically the same as the D6, but more roadworthy,
quieter and better protected against proximity hums than previous models.

How the D6 Clavinet Works
Each D6 keyboard key forms a single arm lever. When a key is depressed, a plunger
underneath touches the string and presses it onto an anvil. The string impinges on the
anvil with a strength according to key velocity. This affects the dynamics of the
sounding string.
These mechanical vibrations are converted into electrical frequencies through
magnetic pick-ups which are amplified and reproduced through the loudspeaker.
As the key is released, contact between plunger and anvil is immediately broken,
leaving the wool-wound part of the string free, so that the string vibration is
immediately muted.

Double-Triggered Notes
When experimenting with the EVD6, or auditioning some of the included Settings, you
may encounter sounds which seem to be triggered on both the note on and the note
off.
This is actually a feature, which emulates the original D6. The real D6 has the “problem”
of the strings sticking to the hammers if they are worn out, producing a second trigger
when the key is released. You can adjust the intensity of this key-off click, with the
Intensity slider in the Click section (see the “Click Parameters” on page 494). Move the
slider to the left, and the second key off trigger will no longer be audible!

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EVP88

This chapter covers Logic’s EVP88 virtual e-piano.
The sounds of various Fender Rhodes pianos are among the most popular keyboard
instrument sounds used in the second half of the twentieth century. The various
Rhodes models have been popularized in a wide range of musical styles, ranging from
pop and rock, electric jazz, jazz rock, soul, and in countless ballads, plus recent house
and hip hop genres. Nearly as popular was the Wurlitzer piano, which enjoyed most of
its success in the seventies. The Rhodes, Hammond organ and subtractive analog
synthesizers were considered the “fundamental” instruments in the keyboard rigs of
rock musicians between 1965 and 1985, and they appeared to be incomparable and
unbeatable … until now.

The EVP88—Concept and Functions
The EVP88’s piano synthesis engine simulates the sound of different Rhodes and
Wurlitzer pianos, as well as the sound of the Hohner Electra piano. The piano synthesis
engine is designed solely for the simulation of electric pianos, and does not make use
of any sampling technology. As such, we can proudly claim that the EVP88 is most
definitely not a digital piano! The EVP88 does not feature acoustic piano or grand piano
sounds. Its only purpose is the ultra-realistic simulation of electric pianos. The dynamics
and scaling of the sounds over the entire 88-key range is silky smooth and do not
suffer from the abrupt changes in sound that typify sampled instruments. There are no
audible loops, and we promise that you’ll never hear any lowpass filters closing while
the sound of a note is decaying.
The EVP88’s engine also simulates the physical movement of the various electric piano
reeds, tines, and tone bars in the electric and magnetic fields of the pickups found in
the original instruments. It synthesizes the ringing, smacking, and bell-like transients of
the attack phase, as well as the hammer action and damper noises. The sound
generator reacts smoothly, musically, and precisely to the 127 steps of velocity
sensitivity as defined in the MIDI specification.

505

Incorporated into the EVP88’s front panel, you will discover an integrated effects
processor which provides a number of classic effects popularly used on electric piano
sounds. The algorithms featured in the effects processor have been specifically
designed, adapted, and optimized for the EVP88. Included are: a great sounding
equalizer, an overdrive, a stereo phaser, a stereo tremolo and stereo chorus.

Parameters of the EVP88

Global Parameters
Model

The big switch shown above allows you to choose the electric piano model. When
selecting a new model, all currently active (sounding) voices are muted, and all
parameters are reset to standard values. As such, it is advisable to select the model
before attempting to edit the effect and parameter settings. There are several Rhodes
models available, such as the Mark I, Mark II and the suitcase piano plus the Wurlitzer
and Hohner Electra Models. The EVP88 simulates the sound of these instruments which
have (re)written modern popular music history. You can read more about the simulated
instruments in “A Brief History of the Clavinet” on page 503.
The names of these instruments are registered trademarks and are protected by law.

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Voices

The voices parameter allows you to set the maximum number of voices that can sound
simultaneously. Lowering the value of this parameter limits the polyphony and
processing requirements of the EVP88. When the parameter is set to 1, the instrument
is monophonic, and uses minimal CPU power. The maximum setting is 88, allowing for
glissandi over the entire keyboard range with the sustain pedal depressed. A setting of
88 will, of course, be more processor-intensive.
Tune

The global Tune setting works in cent steps. A value of 0 equals concert-pitch A 440 Hz.
The range is ±50 cents or, in more “musical” terms, plus/minus half a semitone. For
transpositions in semitone or octave steps, please use the Region Parameter box in the
Arrange window, as per any standard MIDI instrument.

Model Parameters

Decay
Decay time of the piano sound. The lower the value, the less the sound is sustained,
and the higher the level of damping applied to the vibration of the “tines”. When short
values are used for this parameter, the main tone is more pronounced, and sounds
longer than the transient harmonics. The effect is somewhat reminiscent of an electric
guitar string being damped with the palm of the picking hand. Electric pianos can be
modified in a similar way. Longer settings result in more sustain and a less dynamic
feel.
Note: Check out Logic’s compressor plug-ins and experiment with different settings for
decay.

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Release
The release parameter determines the amount of “damper” applied after the keys are
released. Extremely long settings allow you to play the piano like a vibraphone.
Bell
Bell determines the level of the inharmonic treble portion of the tone. It is useful for
emulating a number of classic and typical electric piano sounds.
Damper
This parameter sets the level of the damper noise caused by the damping felt hitting
the vibrating tine.
Stereo
If Stereo is set to high values, bass notes sound from the left, and treble notes from the
right channel. The effect is nice and spacey, but it is not typical for vintage electric
piano sounds. Even with acoustic pianos, the effect is less intense than one might
expect.
Note: The stereo control is not restricted to the bass sound from the left, treble from
the right use outlined above. Using Logic’s plug-ins, you can process the upper notes
differently to the lower ones. With appropriate signal processing routings, you can, for
example, add some bass via an EQ in the left “bass” channel and apply a little echo to
the higher notes. As another option, you could listen to the summed post-effect signals
in mono.

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Stretch and Warmth

The EVP88 is tuned to an equal-tempered scale. As a deviation from this standard
tuning, you can stretch the tuning in the bass and treble ranges, much like acoustic
pianos (especially upright pianos). You can also modulate the tuning of each note
randomly. The main tuning parameter is Tune.
Note: The tones of upright pianos, and—due to their longer strings, less so—grand
pianos have inharmonicities in their harmonic structure. The frequencies of the
harmonics are not exact, even multiples of the base frequency as dictated by
Pythagorean theory. They are only approximate and are, in fact, a little higher. The
harmonics of lower (tuned) notes, therefore, are more closely related to the main
frequencies of the upper notes. Due to the lack of strings, this inharmonic relationship
is not true of electric pianos, nor the EVP88. The stretch feature was included for
situations where you may wish to use the EVP88 in an arrangement alongside an
acoustic piano. When arranged in conjunction with an orchestra or synthesizers, the
stretched tuning facility should not be used.
Lower Stretch
Deviation from the equal-tempered scale in the bass end of the sound. The higher the
value, the further down the low notes are tuned. At a setting of 0, the EVP88 is tuned to
an equal-tempered scale, with each octave down exactly halving the frequency.
Upper Stretch
Deviation from the equal-tempered scale in the treble end of the sound. The higher the
value, the further up the high notes are tuned. At a setting of 0, the EVP88 is tuned to
an equal-tempered scale, with each octave up exactly doubling the frequency.
Warmth
Amount of random deviation from an equal-tempered scale. High values add “life” to
sounds.
Note: When applying Warmth and Stretch, you should consider that these parameters
may result in a detuned sound, which is similar to the overuse of a chorus effect.

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Effects
Equalizer
Treble
This is a conventional filter for the high frequency range. Depending on the model
selected, shelving or peak type filters are utilized, with optimized frequency ranges for
each model pre-selected.

Bass
This is a conventional filter for the low frequency range. Depending on the model
selected, shelving or peak type filters are utilized, with optimized frequency ranges for
each model pre-selected.
Note: By defeating the treble and bass frequency ranges, you can achieve a very direct
and aggressive sound with a more dominant mid range. If you require more precise
equalization, remember that you can insert any of Logic’s equalizer plug-ins into the
Audio Instrument channel strip. There’s also a Tone control in the overdrive circuit
which can be used to further contour the sound.
Drive
Playing an electric piano is best when using tube amplifiers. They offer a wide range of
tones, ranging from the subtle warmth of crunchy guitar amplifiers through to
psychedelic, screaming rock distortion. The EVP88 features an overdrive effect, which
simulates the saturation characteristics of a tube amplifier stage. The overdrive process
is the first signal processing circuit in the effects chain available in the EVP88.

Tone
The Tone control is used to EQ the sound before being sent to, and distorted by, the
virtual tube amplifier circuit. You can choose a more mellow tonal color here, and still
boost the treble with the equalizer after the overdrive circuit. If you prefer harsh
distortion characteristics that come closer to overdriven transistor stages, use higher
tone parameter values. If the sound becomes too hard, you can defeat the treble via
the Treble control, post the overdrive process.

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Gain
The Gain control determines the amount of harmonic distortion.
Phaser

Phaser pedals used by electric guitarists are “classic” effect tools for electric pianos as
well—especially in the electric jazz, jazz-rock and pop styles of the seventies. Classical
four-stage phasing effects are based on phase shifting using modulated all-pass filters.
Mixing the phase-delayed signal with the original signal results in characteristic
notches in the frequency response curve, also known as the comb-filter effect. The
frequencies of the notches in the frequency range are not harmonic (as with the
resonances known from the flanger effect), and these notches are shifted up and down
through the sonic spectrum via LFO (low frequency oscillator) modulation.
Note: Logic offers more parameters in its Phaser and other modulation plug-ins. You
can use these effects alternately to, or in conjunction with, the EVP88’s Phaser. The
parameters found in the EVP88 Phaser have much in common with the best analog
phasers of the 60’s and 70’s, including subtle analog-style distortion. It offers the same
32 Bit internal processing and sound quality of the Logic Phaser plug-in.
Rate
Speed of the phasing effect. When set to 0, the Phaser is switched off.
Color
Intensity of sound coloration introduced by the Phaser, caused by feeding the Phaser
output signal back into its input.
Stereophase
Relative phase shift between the left and right channels, ranging from 0° to 180°. With
0° selected, the effect is most intense, but not stereophonic. With 180° selected, the
effect symmetrically rises in the left channel while simultaneously falling in the right
channel, and vice versa.

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Tremolo

A periodic modulation of the amplitude (level) of the sound is known as tremolo. The
modulation is controlled via an LFO. The Fender Rhodes suitcase piano features a
stereo tremolo and many other electric pianos have a simple, but quite obtrusive,
mono tremolo that can introduce a strange kind of polyrhythmic feel to performances.
Note: The original Wurlitzer piano has a mono tremolo with a fixed modulation rate of
5.5 Hz. For an authentic Wurlitzer sound, choose 0°. For Rhodes sounds, select 180°. The
settings in-between result in nice spacey effects, especially at low LFO rates.
Rate
Speed of the tremolo effect (LFO frequency).
Intensity
Amount of amplitude modulation. With 0 selected, the tremolo effect is switched off.
Stereophase
At a setting of 0°, the level undulates in phase on both channels. With 180° selected,
the modulation is perfectly out of phase, resulting in a stereo tremolo effect that is also
known as “auto panning”. The effect is similar to manually turning the pan pot from side
to side.
Chorus Intensity
The well-known chorus effect is based on a delay circuit, the delay time of which is
permanently modulated by an LFO, while the delayed effect signal is mixed with the
original dry signal. It is the most popularly used effect on electric piano sounds. This
parameter regulates the intensity (the amount of delay time deviation), while the LFO
rate is fixed at 0.7 Hz. Pay close attention when using high values as this may result in
the piano sounding detuned.

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Additional Parameters
The EVP88 features a number of additional parameters that are accessible via the 001/
100 button at the top of the EVP88 Plug-in window.
The Volume slider sets the overall output level of the EVP88 (Range: −20 to +20 dB).
The Bend Range Down/Up sliders determine the pitch bend range in semitone steps.
The Chorus Rate slider sets the speed of the Chorus effect, in Hz. The Delay PP/FF sliders
determine the delay time (in milliseconds) when the keys are struck pianissimo (PP—
soft) or forte (FF—hard).

The E-Piano Models Emulated
Rhodes
The most commonly known and widely used electric piano model was constructed by
Harold Rhodes (born 1910). Designed in 1946 as a piano surrogate for practice,
education, and army entertainment, the Rhodes piano was successfully marketed by
guitar manufacturer Fender from 1956. The Fender Rhodes has become one of the
most popular musical instruments in jazz, especially electric jazz. Its popularity in pop
and rock music occurred after CBS took over production of the Rhodes in 1965. Despite
further changes in ownership throughout the company history, the instrument is most
commonly called the “Fender Rhodes”. There are also a number of “Rhodes” synthesizers
(which were developed by the now-defunct synthesizer manufacturer ARP). Japanese
synth and music technology manufacturer Roland were the proprietor of the Rhodes
name for a while, and released several digital pianos which carried the Rhodes moniker.
From 1997, until his death in december 2000, Harold Rhodes again inherited the name.
The method of sound generation used by the Rhodes piano is based on metal reeds
which function much like a tuning fork. These are hit by a hammer action that works in
a similar fashion to that of a grand piano action. The asymmetrically designed “tuning
fork” consists of a thin tine and a massive tone bar, which are bolted together. Due to
construction considerations, some of the tone bars are rotated by 90 degrees. The
piano is kept in tune by the mass of a spring which can be moved along the tine. The
tine oscillates in front of an electric pickup, similar to that of an electric guitar. This
functions along inductive principles, with permanent magnets placed around the tine
having a damping effect on its movement, thereby affecting the sound.
Like the output signal of an electric guitar, the Rhodes output signal is rather weak and
needs quite a bit of pre-amplification. The Rhodes sound is not harmonically-rich. This
is why a treble boost or an overdrive effect, which can both add harmonics, is quite
welcome when it comes to playing the Rhodes. Playing the Rhodes is, as mentioned
earlier, at its best when using tube amplifiers.

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The Rhodes piano was also made available as a suitcase piano (with pre-amp and twochannel combo amplifier) and as a stage piano, without amplifier. Both of these 73-key
“portable” versions have a vinyl-covered wooden frame and a plastic top. In 1973, an 88
key model was introduced. Smaller “Celeste” and bass versions were less popular. The
Mk II (1978) had a flat top instead of a rounded one. This allowed keyboardists to place
extra keyboards on top of the Rhodes. In 1984 the Mark V was introduced, and even
sported a MIDI output. Around this time, Rhodes production decreased as most
keyboard players invested in the more flexible (and lighter) digital synthesizers
available. These keyboards could emulate the sound of older pianos, like the Rhodes,
and also had the bonus of a range of great new piano sounds.
The individual characteristic sound of each Rhodes piano depends more on the
adjustment and maintenance of the instrument than on the model. Early models had
hammers covered with felt, resulting in a smoother sound than the newer models,
which had neoprene-covered hammers. The suitcase piano featured a pre amplifier
which could create a sound with a very dominant mid range. But appropriate pre
amplifiers and equalizers can make a stage piano sound the same. The stage piano has
no power cord—just like an electric guitar.
The MkII has no resonance clamps in the treble range, unlike former models. This is
why it has a little less sustain in the treble range. The most significant differences in
terms of sound depend on how “deeply” the tine is adjusted. In cases where it is in a
deep position—closer to the pickup—the bell characteristic becomes more prominent.
In the eighties, many Rhodes pianos were adjusted so that they had more “bell”—the
taste of the time.
There is little use in naming the most prominent Rhodes players and styles. Practically
every keyboard player of the electric jazz, jazz rock, crossover, soul pop, and rock styles
used to play it, at least in the seventies. Many still do. One of them is Ray Charles, who
played the role of a blind music shop owner in the Blues Brothers movie. Negotiating
the price of a used Rhodes, the Blues Brothers mentioned the lack of keyboard “action”.
In the ensuing furious—and famous—Rhodes solo, Ray Charles proved that this
particular Rhodes had plenty of “action”. For those of you who have never played the
original instrument, the keyboard action feels a little smooth in travel and sticky when
fully depressed. This makes its “feel” a little unusual for many players unfamiliar with it.
Obviously, though, its feel is good enough for Ray Charles!

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Rhodes Models:
• Suitcase MkI
• Suitcase V2
• Bright Suitcase
• Stage Piano MkI
• Stage Piano MkII
• Bright Stage MkII
• Hard Stage MkII
• MarkIV
• Metal Piano
• Attack Piano
The Metal Piano and Attack Piano models feature sound qualities that can be “aimed
at” with the original Rhodes instruments, but not to the extent of these models. They
do not sound realistic, but they are included as sound “ideals” that the Rhodes
technicians might have had in mind when preparing their keyboards.

Wurlitzer Piano
This well-known manufacturer of music boxes and organs also built electric pianos, the
portable versions of which have written pop and rock music history. The 200 series
Wurlitzer pianos are smaller and lighter than the Rhodes pianos, with a keyboard range
of 64 keys from A to C and an integrated amplifier and speakers.
The action resembles that of a conventional acoustic piano. It can be played with
velocity sensitivity, just like the Rhodes. Its sound generation system is based on spring
steel reeds which can be tuned with a solder weight. The Wurlitzer has electrostatic
pickups: The reeds are supplied with a 0 volt current and move between the teeth of a
“comb”, connected to a 150 volt current. The tone of the Wurlitzer, which was first
manufactured in the early sixties, features many odd harmonics. If you were to ever try
to emulate its sound with an analog synthesizer, you would start by switching the
oscillator to output a 60% rectangular (PWM) wave.
The Wurlitzer is best known as the signature piano sound of the band “Supertramp”. You
will know it from their “Crime of the Century” album. It can be heard on “Bloody Well
Right”, “Dreamer”, “Hide in Your Shell” and also in “The Logical Song”. You might also
recognize the Wurlitzer sound when listening to Pink Floyd’s “The Dark Side of the
Moon” or “Wish You Were Here” (“Have a Cigar”, “Money”, “Time”) and “I am the Walrus”
by the Beatles.
Wurlitzer Models:
• Wurlitzer 200 A
• Wurlitzer 240 V
• Soft Wurlitzer
• Funk Piano

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The model Funk Piano does not sound realistic in the bass. We’ve added this special
synthetic sound of the piano engine as a bonus.

Hohner Electra Piano
Not to be confused with the all-electronic RMI Electrapiano, the extremely rare Hohner
Electra Piano offers striking hammers like those of the Rhodes, but a stiffer keyboard
action. It was designed to resemble the look of a conventional acoustic upright piano.
John Paul Jones of Led Zeppelin played it on “Stairway to Heaven”, “Misty Mountain
Hop” and “No Quarter”.
Hohner Electra Model:
• Electra Piano

EVP88 and MIDI
Adaptation of Your MIDI Keyboards Velocity Sensitivity
The EVP88 responds with extreme sensitivity to the velocity information transmitted
with MIDI note messages. It’s advisable to set Logic’s velocity and dynamic track
parameters with care. In Logic Pro, you can try the following tip to fine-tune the
velocity curve if you find that you’re not getting the right feel with your MIDI keyboard.
• Create a Transformer object in the Environment, and cable it between the Physical
Input and Sequencer Input objects on the Click and Ports layer.
• Set the transformer parameters so that all MIDI events with the condition “note” are
set to Use Map under Vel in the lower operation line.
You can then “draw” your own individual keyboard velocity curve. For more detailed
information, please refer to the Environment chapter of the Logic Pro reference manual.

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MIDI Controller List
E-Piano

Stretched Tuning

Equalizer

Overdrive

Phaser

Tremolo

Chorus

Chapter 28 EVP88

Volume

Model

Voices

Tune

Decay

Release

Bell

Damper

Intensity

Lower

Upper

Warmth

Treble

Bass

Gain

Tone

Rate

Color

Stereophase

Speed

Intensity

Stereophase

Intensity

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EXS24 mkII

This chapter introduces Logic’s EXS24 mkII sampler.

The EXS24 mkII offers all of the facilities that you would expect to find in a hardware
sampler, without the cost and bulk of this type of device. As a purely software-based
instrument, the EXS24 is perfectly integrated into Logic, and makes use of your
computer’s RAM and hard disks. This integration within the computer environment
offers instant access to all audio data and Sampler Instruments used in a Logic song
file. These files are stored on your computer’s hard disks. This integration simplifies
sample library management and eliminates the need for separate physical devices and
the cables required to connect them.
You can make use of the editing features of Logic, or another audio editor for your
samples. This is far more convenient, not to mention faster, on a computer monitor
than on the cramped display found on most hardware samplers. As the samples are
stored within the computer, the slow and often unreliable transmission of sample data
back and forth between your sampler and Macintosh is eliminated.

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The EXS24 is compatible with the EXS24, AKAI S1000 and S3000, SampleCell, ReCycle,
WAV, AIF(F), Gigasampler, and SoundFont2 sample formats, as well as the Vienna
Library, allowing access to large and comprehensive sampler libraries.
The EXS24 offers numerous sample processing and synthesis options, enabling you to
tailor sounds to meet your needs.
Last, but not least: as a highly optimized Logic instrument, the EXS24 offers great
performance, even on slower machines. The EXS24’s performance is scalable, so you
can look forward to enhanced functionality and increased polyphony on future
computer technology. The number of possible Sampler Instruments available for
simultaneous playback is directly related to the computer’s processing and RAM
resources. The more RAM you have, and the faster your CPU, the more Sampler
Instruments can be loaded and played.
And what of the sound?
As the EXS24 uses high-end algorithms with 32-bit internal processing, is completely
digital, and seamlessly integrates into Logic, you are guaranteed pristine, clear sample
playback—up to 24-bit and 96 kHz, if you wish (and your audio hardware is
appropriate). With the EXS24, there’s no need to concern yourself over sound quality or
compatibility issues with future audio formats.

Using Instruments
Folder Structure
The following items will be installed in the main Logic program folder:
• The Sampler Instruments folder—contains all of the Sampler Instruments received
with the EXS24. This folder will also be used for the storage of all Sampler
Instruments added or created in future. A Sampler Instrument contains all sample
mapping information plus the modulation, filter, volume, and pan settings needed
for a fine Grand Piano multisample, as an example.
• The EXSamples folder—contains all of the raw samples (audio files) that the Sampler
Instruments make use of.

Loading and Playing an Instrument
The EXS24 ships with a ready-to-play Sampler Instrument library. These Instruments can
be found within the Sampler Instruments subfolder of the Logic program folder. Once
the EXS24’s graphical interface is opened, you can select one of the Sampler
Instruments by clicking on the pull-down menu above the silver panel area (directly
above the Cutoff knob). The selected Sampler Instrument will then load.

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Change the sound by twisting the knobs, pressing switches and moving sliders—and
don’t worry—you can’t destroy the original Sample Instrument.

Creating and Editing Instruments in the Instrument Editor
Instruments are created and edited in the Instrument Editor. It is also used to organize
samples and to convert foreign sample formats (AKAI S1000/3000 and so on). You can
also assign samples to keys or key ranges and set start, end, and loop points plus all of
the normally tedious tasks associated with sampling in the EXS24 Instrument Editor.
Fortunately, the EXS24 Instrument Editor is much easier and more pleasant to work
with than that of a hardware sampler. This is illustrated by the transparent architecture
of the EXS24: samples are assigned to Zones, Zones are assigned to Groups. The end
product of these assignments is a Sampler Instrument.
Now that you know how to load a Sampler Instrument, it would be a good time to
briefly introduce you to the Instrument Editor. Please open the Instrument Editor
window via the Audio > EXS24 Instrument Editor menu. The parameters and functions of
the Instrument Editor window are described in this section.

The Instrument Editor shown above is empty as no Instrument has been loaded or
created. The keyboard in the upper window area can be used to trigger notes for the
EXS24 in the currently selected track. Below the keyboard, a number of Zones are
shown.
Zones
A Zone is a location into which a single sample (or audio file, if you prefer this term) can
be loaded from hard disk or CD ROM. The sample loaded into the Zone is memory
resident—it uses the RAM of your computer. A Zone offers various parameters for
controlling the playback of the sample. Each Zone allows you to determine the range
of notes over which the sample should be heard (Key Range), and the “root key” (Key
Note)—the note at which the sample sounds at its original pitch. In addition, sample
start, end, and loop points plus volume and several other parameters can be adjusted
within the Zone. You can define as many Zones as you wish. Each Zone requires at least
one EXS24 voice when played.

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To create a new Instrument and a Zone
1 Select Instrument > New from the Editor window’s menu. A new instrument is created.
Note: In order to hear your edits, please ensure that the correct Instrument is loaded
into the EXS24 instance assigned to the currently selected track, and is selected in the
editor.
2 Go to Zone > New Zone to create a new zone. A small window will appear to the left of
the editor window.

3 Click on the empty field alongside the Audio File label. A file selection dialog box will
launch, allowing you to select a sample from the hard disk or CD ROM.
4 Select the sample. It is loaded into the Zone.
Drag and Drop Zone creation
A new zone (and a new instrument, if none is currently displayed in the editor) can also
be created by dragging a file onto one of the keys of the onscreen keyboard. The start
key, end key and root key are all set to the note that the file was dropped on. This drag
and drop functionality works for audio files from the following sources: Project
Manager, Audio Window and the Finder.
You can create multiple zones by drag and dropping multiple files from these sources.
When you do so, a dialog window will launch, asking how you would like these
multiple files to be handled. See the “Load Multiple Samples” section, on page 561 for
further information.
Note: If an audio file is dragged and dropped onto an existing zone (in the lower
section of the EXS Instrument Editor window), the file referenced by that zone is
changed to the new (dropped) file.

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To move a Zone:
1 Move the mouse cursor (it will change to a two-headed arrow) over an existing zone
bar (this is one of the bars displayed directly under the keys).
2 Click-hold and drag the zone to the desired position below the onscreen keyboard.
To change the start/end note of a Zone:
1 Move the mouse cursor to the beginning or end of a zone bar. The cursor will change
to a left (start) or right (end) bracket, surrounded by arrowheads.
2 Click-hold and drag the start or endpoint of the zone to the desired length.
Note: You can also change the root key of the zone, by pressing Option and Command
simultaneously, while dragging the zone.
Adjusting the Zone Parameters
Set up a key range for the sample with the two Zone Range parameters; Key Note allows
you to determine the note used to trigger the sample at its original pitch. Reverse plays
the sample from its end to the beginning. This option works non-destructively, and
doesn’t change the audio data. Adjust volume and pan position for the sample with
the corresponding parameters. Negative Scale values make notes lower than the note
position defined by the Key Note parameter sound louder than higher ones; positive
values have the opposite effect. Use this parameter for balancing the volume of a
sample across the selected key range.

Negative Scale values increase the volume of lower notes (left of the Key Note), positive
values increase the volume of higher notes (right of the Key Note). If necessary, adjust
the playback start and end points for the sample with the Start Frame and End Frame
parameters.
Activate Loop if desired; the loop parameters are hidden when the loop parameter is
deactivated. You can set a start and end point for the loop and fine-tune the loop with
the Tune parameter, if necessary.

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Editing Samples
You may have noticed the small E buttons next to the start, end and loop point
parameters. Clicking on these will launch the selected sample in Logic’s Sample Editor,
allowing you to edit the sample borders graphically, and use all of the Sample Editor’s
functionality. When loop is activated, you can also edit the loop points graphically: the
LS marker indicates the loop start point and LE, the loop end point.

Groups
Imagine a drum kit has been created, with a number of different samples being used in
several Zones, mapped across the keyboard. In many musical circumstances, it would
be great to be able to treat each of the samples independently with the EXS24’s sound
editing parameters—to alter the decay of the snare, or to use a different cutoff setting
for the hi-hat samples, for example.
This scenario is where the Groups come in—they allow for the very flexible
organization of samples. You can define as many Groups as desired, and can assign
each Zone to one of these Groups. In a drum set, for example, you could assign all kick
drums to Group 1, all snares to Group 2, all hi-hats to Group 3 and so on.
Why might you want to do this?
A Group makes it possible to define a velocity range for all assigned Zones, allowing
you to specify a velocity window in which the grouped Zones should sound, as one
example. Each Group also features offset parameters for the amplitude envelope and
filter settings made in the Plug-in window.
It’s also possible to play all Zones without defining and assigning even a single
Group—in this case, the parameters in the Plug-in window work in an absolute manner
for all Zones. To clarify, all samples in all Zones will be affected equally by the parameter
adjustments made in the Plug-in window.

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Given that up to 64 EXS24 instruments (dependent on your version of Logic) can be
used simultaneously, opening several instances of the EXS24 provides the advantage of
a dedicated channel strip for each and every sound you use. This allows full control
over the sound (via EXS and effects parameters) during composition and mixdown.
To assign a Group to a Zone
1 Select Group > New Group in the editor’s menu to create a new Group. A Group window
will appear on the right-hand side of the editor.

2 Select the new Group as a target in the Zone’s Group pull-down menu. The Group
parameters will now affect the sample in the Zone.

Multiple Zones and Groups
You may create as many Zones and Groups as you wish, and can assign as many Zones
to a Group as desired. The Groups offer several parameters for simultaneous control
over all assigned Zones:
• The Voices parameter allows you to determine the maximum number of voices for a
Group. A practical use of this would be to set up a classic “hi-hat mode” within a full
drum kit, mapped across the keyboard. In this scenario, you could assign both an
open and closed hi-hat sample to a Group, and set the Voices parameter of the Group
to 1. In this example, the most recently triggered of the two hi-hat samples will mute
the other, as only one voice is allowed for the Group. This mirrors the real-world
behavior of hi-hats. When samples in Zones are assigned to another Group, the other
sounds of the drum kit can still be played polyphonically.
• The Group Volume and Pan parameters simultaneously affect the settings of all Zones
assigned to the Group. This works much like a sub group on a mixing console.
• The two Select Range parameters are used to set up a velocity window for the Group.
Use these parameters for sounds where you wish to mix, or switch between, samples
dynamically by playing your MIDI keyboard harder or softer—with layered sounds, or
when switching between different percussion samples, for example.

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• Each Group offers separate ADSR parameters for offsetting the ADSR volume

envelope settings made in the Plug-in window: The Attack, Decay, and Release time
parameters can be adjusted by ±9999 ms, the Sustain level by ±50%.
• Similarly, the Cutoff and Resonance settings of the Plug-in window can be offset by
±50% for each Group.
It is possible to play all Zones without defining and assigning a Group—in such cases,
the parameters defined in the Plug-in window work in an absolute (identical) manner
for all Zones.
Detailed descriptions of all Zone and Group parameters can be found in “Zone
Parameters” on page 564 and “Group Parameters” on page 566.

File Organization
File Types and File Organization
The EXS24 uses the following file types and hierarchical structures:
Audio File
A single sample on your hard disk. The EXS24 is compatible with all audio file formats
supported by Logic. Audio files are handled in the EXS24 Instrument Editor, where they
can be edited and organized into Sampler Instruments.
Sampler Instrument
A Sampler Instrument points to one or more audio files, and organizes them as multi
samples or drum maps, respectively. Within the Sampler Instrument you may assign
different samples to different key and velocity ranges, set loop points, and adjust other
playback parameters. You can also work with Zones and Groups (see the “Multiple
Zones and Groups” section, from page 525 onwards), which always belong to a
Sampler Instrument, and are not stored or loaded separately.
Note: Audio files are not contained in a Sampler Instrument. The Sampler Instrument
simply stores information about an audio file’s name, its parameter settings, and its
location on the hard disk. When you delete or rename an audio file, the Sampler
Instrument that makes use of it will be unable to find it, so take care when handling
audio files.
A Sampler Instrument is the file type that is loaded into the EXS24 for playing. When
you select a Sampler Instrument in the EXS24’s pull-down menu, the associated audio
files are automatically located on the hard disk, and are subsequently loaded into your
computer’s RAM.
In order to be visible within the EXS24’s Sampler Instrument pull-down menu,
Instruments must be stored in the Sampler Instruments sub-folder of the main Logic
program folder.

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Note: You can store your Sampler Instruments in any folder on any of your computer’s
hard drives. To do so, you must create an alias pointing to this folder within the Sampler
Instruments folder located in the Logic program folder. Please refer to “File
Organization” on page 526.
You can manually load Sampler Instruments from other locations into the EXS
Instrument Editor at any time. Such Instruments also appear in the EXS24’s Sampler
Instrument load pull-down menu.
Settings
Settings are used to store all parameter adjustments made in the Plug-in window.
Every Logic plug-in allows you to store and recall Settings, and the EXS24 is no
exception. The Settings for the EXS24 are stored in the EXS24 folder, which itself is
located in the Plug-In Settings folder within the main Logic program folder.

Important: The Settings that can be stored and recalled in the Plug-in window are not
part of the Sampler Instrument being loaded.
Settings reside above the Sampler Instruments in the hierarchy: A Setting contains a
pointer to a Sampler Instrument, and when a new Setting is selected, the Sampler
Instrument it points to is automatically loaded. As such, Settings are convenient for
organizing and accessing your favorite Sampler Instruments. Settings also recall any
changes made to parameters within the Plug-in window.

Management of Sampler Instruments
As your sample library grows, the list of Sampler Instruments will also expand. To aid
you in keeping the list of Sampler Instruments manageable, the EXS24 features a
sophisticated, but easy to use method of file management.
The Sampler Instrument pull-down menu directly reflects the folder structure within
the Sampler Instruments folder. You can choose to sort your Sampler Instruments in
groups such as “basses and guitars”, by sound type, alphabetically, or by song.

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To organize your Sampler Instruments into a preferred hierarchy:
1 Simply create a folder—“Basses” for example—within the Sampler Instruments folder,
with your operating system’s file management utilities.
2 Drag and drop the desired EXS24 Sampler Instruments into this newly created folder.
Their menu structure will be reflected when clicking on the EXS24 Sampler Instruments
pull-down menu.

Note: You will need to relaunch Logic after changes are made to the folder hierarchy in
the Sampler Instruments folder.
The menu is limited to the display of folder sub-menus that actually contain EXS
instrument files. Other folders are not added to the menu. Aliases pointing to folders
which contain EXS instrument files outside the Sampler Instruments folder can also be
added to the menu. Even the Sampler Instruments folder itself can be an alias to a folder
on a different drive or location.
When selecting a Sampler Instrument from a sub menu, a bold entry at the top of the
root menu is added, to indicate the current selection. The sub menu that contains the
selected Sampler Instrument is also shown in bold type, as are further sub menus. This
makes it easy to trace the file path of the currently loaded Sampler Instrument.

Saving of Project-related EXS24 Instruments
This feature allows all EXS24 Instruments associated with a Project to be saved/loaded
into/from a single folder location, which also contains the song file. These Sampler
Instruments will then be exclusively associated with this song.
This is useful for two reasons:
• It makes the archiving and handling of songs, including the associated Sampler
Instruments, easier.
• It makes it simpler to deal with a particular set of samples that will not be used in
another song—vocals, modified drum kits and so on.

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It works as follows: When opening a Logic Project, the EXS24 initially looks for a subfolder named “Sampler Instruments” in the folder that contains the song file. If such a
sub-folder exists, all Sampler Instruments found in this folder are added to the Sampler
Instrument pull-down menu in the EXS24 GUI. This new entry in the Sampler
Instrument pull-down menu will appear as a sub-menu item that matches the song file
name. This behavior ensures that the EXS24 will always locate any song-related
Sampler Instrument files before searching in the global Sampler Instrument folder,
found in the Logic program directory.
To save Instruments related to a particular song
1 Create a new folder for a song/Project and name it.
2 Save the song file itself into this folder.
3 Create a sub folder named “Sampler Instruments” within the Project folder.
4 Simply copy/move the Sampler Instrument files required into this folder. Note that only
the Sampler Instrument files, not the raw samples used by these Sampler Instruments
should be copied, except when archiving (or unique samples are used), as discussed
below. The “Used by EXS24” option could be useful.
Even simpler:
1 Save your song with the File > Save as Project function. More information on this can be
found in your Logic manual.
2 When Logic is booted, the song is loaded, and an EXS24 instance is opened; a new
hierarchical menu item will appear within the EXS24 Sampler Instrument pull-down
menu when clicked. This new menu item will retain the song’s name and contains all of
the Sampler Instrument entries copied to this folder earlier.
3 When saving any newly created or modified Sampler Instruments, ensure that you use
the “Save as” function and browse to the “Sampler Instruments” folder inside the new
song folder.
When saving on a per-song basis, you should observe the following folder hierarchy:
• The Project folder contains the song file and the “Sampler Instruments” folder.
• The “Sampler Instruments” folder contains all Sampler Instruments that are used in
this song exclusively.
As the EXS24 automatically locates the audio files associated with Sampler Instruments,
it generally does not matter where these audio files are stored. One circumstance,
however, where the storage location of the audio files does matter is as follows: Should
you need to archive the song with all related data, or wish to deal with a particular set
of samples that will not be used in another song, you will want to store the audio files
inside the Project folder as well.

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This will change the folder hierarchy as follows:
• The Project folder contains the song file and the Sampler Instruments folder.
• The Sampler Instruments folder contains all Sampler Instruments that are used in this
song exclusively—vocals, for example.
• A separate folder containing the audio files associated with the respective Sampler
Instrument for each Sampler Instrument used.
To assist you in doing this, the EXS24 Instrument Editor provides the following
functions:
• Instrument > Copy Audiofiles
Copies the audio files of any Sampler Instrument edited in the EXS24 Instrument
Editor to the target directory of your choice. A folder for the audio files associated
with this Sampler Instrument is created in the target location. The Sampler
Instrument file itself is also copied.
• Instrument > Move Audiofiles
Moves the audio files of any Sampler Instrument edited in the EXS24 Instrument
Editor to the target directory of your choice. A folder for the audio files associated
with this Sampler Instrument is created in the target location.
• Functions available as Key Commands
The Backup audiofiles of all USED and ACTIVE instruments of current song key command
copies the audio files of all (active) Sampler Instruments used by the current song to
the target directory of your choice. Folders for the audio files associated with these
Sampler Instruments are created in the target location. All used Sampler Instrument
files are also copied.
• The Move audiofiles of all USED and ACTIVE instruments of current song key command
moves the audio files of all (active) Sampler Instruments used by the current song to
the target directory of your choice. Folders for the audio files associated with these
Sampler Instruments are created in the target location.

Searching for Sampler Instruments
As a further navigational enhancement, the EXS24 features a built-in Find function,
which works in conjunction with the hierarchical menu structure discussed earlier.
In order to minimize the number of Sampler Instruments displayed in the Sampler
Instrument pull-down menu, you can make use of the Find function. This will limit the
Sampler Instrument pull-down menu to only display Sampler Instrument names that
contain the word “piano” or “bass”, as an example. This will also hide any sub-menus
that don’t contain the search word. Simply select Find in the Sampler Instrument pulldown menu and, in the ensuing dialog box, type in the character string (search term)
to search for.

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The Clear Find option in the Sampler Instrument pull-down menu will display the full
menu but does not clear the actual search term typed into the search dialog. To return
to the limited menu, simply select Enable Find. The selection of Enable/Clear Find allows
you to toggle between the two without re-typing the search term.
If you wish to use a different character string, select the Find option a second time and
type in the desired search term.

Sample File Import
The EXS24 is compatible with the EXS24, AKAI S1000 and S3000, SampleCell, ReCycle,
Gigasampler, and SoundFont2 sample formats, as well as the Vienna Library.

Using EXS24 Files
We strongly recommend that you copy any EXS24 Sampler Instruments shipped on CDROM to your hard drives—for two reasons: firstly, to always have direct, immediate
access to your Sampler Instruments without searching for and inserting CD-ROMs, and
secondly, to be able to sort your Sampler Instruments according to your needs.
To copy an EXS24-format Sampler Instrument, along with its associated audio
files, from CD-ROM to your hard drives:
1 Copy the Sampler Instrument files from the CD into the Sampler Instruments folder
within the Logic folder.
2 Copy the associated samples from the CD into the EXSamples folder within the Logic
folder.
Note: You can sort your Sampler Instruments to suit your own needs (see “File
Organization” on page 526). The EXS24 file system is able to work with aliases for
Sampler Instrument folders. Furthermore, a Sampler Instrument searches for, and finds,
all samples it uses on all active hard drives—as long as you do not delete or rename
the samples.
Using EXS24 Instruments directly from CD-ROM
Normally, the Sampler Instrument and associated samples (audio files) will be stored on
your hard disks, but on occasion, you may wish, or need, to load an EXS Sampler
Instrument from CD-ROM.
To use an EXS Sampler Instrument stored on CD-ROM:
1 Copy the Sampler Instrument file (not the associated samples) from the EXS format CDROM into the Sampler Instruments folder.

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2 When the Sampler Instrument is loaded, ensure that the appropriate CD-ROM is in the
computer’s CD-ROM drive. If the appropriate CD-ROM (the one that contains the
desired Sampler Instrument and its associated audio files) is in the drive, the EXS24 will
automatically search for the associated samples on all local media. It will locate the CDROM and will load the Sampler Instrument.
3 If the CD-ROM is not present, you will be required to insert the appropriate disc and reload the Sampler Instrument.
Note: Aliases/shortcuts may only be used for files stored on hard disk, not on CD ROM.

Importing SoundFont2 Files
To make use of this functionality, simply copy or move your SoundFont2 files into the
Sampler Instruments folder.
Select the file name in the EXS24 Sampler Instrument load flip-menu and the file will
automatically be converted. An EXS Instrument file will be created in the Sampler
Instruments folder which contains the original SoundFont2 file. The raw samples
associated with the Sampler Instrument will be placed in a SoundFont Samples folder
within the Logic program folder.
Should a SoundFont2 Bank file (a Bank contains multiple sounds—a General MIDI bank,
for example) be loaded, it will create a Bank folder and also a Samples folder. These new
folders will have the same name as the SoundFont2 Bank file, with the word “Bank” or
“Samples” appended.
All sounds contained in the bank will automatically have an EXS Sampler Instrument
file created and placed into the newly created Bank folder. The EXS24 Sampler
Instrument pull-down menu will automatically be updated to reflect the new folder
hierarchy. All samples associated with the Bank will automatically have a Samples folder
created inside the SoundFont Samples folder which resides in the Logic program folder.
As an example, a SoundFont2 bank file named “Vintage Drums” is imported by the
EXS24. It contains over 50 individual drum kits from several different vintage drum
machines. A new folder named Vintage Drums.Bank will be created in the Sampler
Instruments folder. A second folder named Vintage Drums.Samples will be created in the
SoundFont Samples folder. Both of these folders are found in the main Logic program
folder.
The Sampler Instrument pull-down menu hierarchy is updated and the original Vintage
Drums entry is replaced with a Vintage Drums.Bank entry. This new entry is a folder that
contains the individual Sampler Instruments, which can be selected and loaded as per
usual.
Once conversion is complete, the original SoundFont2 source files can be freely deleted
from the hard disks.

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Note: You can store your imported Sampler Instruments in any folder on any of your
computer’s hard drives. To do so, you must create an alias pointing to this folder within
the Sampler Instruments folder located in the main Logic program folder. Care should be
taken when importing samples to ensure that when a song is loaded, the associated
Sampler Instruments will be found. Sampler Instruments are only searched for in the
Sampler Instruments folder (or an alias to it). Any Sampler Instruments stored in other
locations will not be located, and must be loaded manually.

The folder hierarchy of the EXS24.

Importing SampleCell Files
The importation of SampleCell format files is as per that of SoundFont2 files. Simply
copy or move your SampleCell files into the Sampler Instruments folder.
Select the file name in the EXS24 Sampler Instrument load flip-menu and the file will
automatically be converted. An EXS Instrument file will be created in the Sampler
Instruments folder which contains the original SampleCell file. The raw samples
associated with the Sampler Instrument will be placed in a SampleCell Samples folder
within the main Logic program folder.
Once conversion is complete, the original SampleCell source files can be freely deleted
from the hard disks.
Should you import SampleCell or AKAI format Samples, they will appear as a
SampleCell Samples or AKAI Samples folder on the same level as the EXSamples, Sampler
Instruments and SoundFont Samples folders. Please refer to the EXS24 folder hierarchy
diagram above.

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Importing Giga Files
The importation of Giga format files is as per that of SoundFont2 files. Simply copy or
move your Gigasampler files into the Sampler Instruments folder.
Select the file name in the EXS24 Sampler Instrument load flip-menu and the file will
automatically be converted. An EXS Instrument will be created in the Sampler
Instruments folder which contains the original Giga file. The raw samples associated
with the Sampler Instrument will be replaced in a Giga samples folder within the main
Logic program folder.
Once conversion is complete, the original Giga source file/s can be freely deleted from
the hard disk.
Should you import Giga samples, they will appear as a Giga samples folder on the same
level as the EXSamples, Sampler Instruments and SoundFont Samples folders. Please refer
to the EXS24 folder hierarchy diagram above.

Converting ReCycle Files to EXS24 Instruments
ReCycle is a sample editing program from Propellerheads software. It slices sample
material into small segments (slices) over time, and can generate a number of file types
which can be read by Logic and the EXS24. This section covers the use of the various
ReCycle format files with the EXS24.
The ReCycle file types supported are listed below:
Provenance

Mac OS File Type

Old ReCycle File

RCSO

Old ReCycle Export File

REX

New (2.0) ReCycle File

REX2

To convert ReCycle files into EXS24 Sampler Instruments, simply browse to the desired
function in the Instrument > ReCycle Convert menu of the EXS24 Instrument Editor. The
individual functions are outlined in the “Recycle Convert” section, from page 555
onwards.
Velocity Factor
The velocity factor determines how the loudness of each “slice” of the imported
ReCycle file affects the velocity values of the MIDI note generated to trigger it.
• If a positive value (up to 100) is entered, louder slices will generate MIDI notes with
higher velocity values.
• The use of negative values on louder slices will result in the generation of lower MIDI
note velocities.

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Converting AKAI Files
This section discusses the AKAI import procedure. The EXS24 can import samples saved
in the AKAI S1000 and S3000 sample formats. The AKAI Convert function can be used
to import:
• an entire AKAI format CD ROM
• an AKAI Partition
• an AKAI Volume
• an AKAI Program
• an Individual Audio File (sample)
These options have been provided to give you the most flexible and efficient method
of dealing with your sample library. There may be a sample or two, or perhaps a
particular drum kit which you would like to import from an AKAI CD-ROM.
Similarly, you may wish to import the contents of an entire CD-ROM in one simple
operation, rather than spend the time dealing with individual Partitions, Volumes,
Programs, and Audio Files.
This way, you can load and audition all of an AKAI CD-ROMs programs and files within
Logic. Later, at your convenience, you can make use of your operating system’s file
management utilities to remove or reorganize your imported AKAI sounds, as
discussed in “File Organization” on page 526.

To convert AKAI files
1 Select Options > AKAI Convert. This will launch a window similar to that shown above,
with the text “Waiting for AKAI CD” spread across the four columns.
2 Insert an AKAI format sample disc into your CD-ROM drive and the AKAI Import
window will commence reading the data. Following the reading of the CD-ROM, the
display will update to show the contents of the CD-ROM. The Partition column will
display information, with Partition A, Partition B (and so on) entries listed.

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Note: Reading of a CD-ROM may take some time, dependent on the amount of sample
data and file structure of the disc. In addition, the speed of the CD-ROM mechanism,
bus speed, memory, and other factors can affect performance.
3 To view the contents of the Partitions, click once on the appropriate entry with the
mouse button. This will display the Volume information contained within the Partition.
4 To continue through the architecture of the CD ROM, click on the Volume entries to
view any Programs contained therein, and on the Program entries, to view the raw
audio files (samples).
5 Once you have made your selection of Partition, Volume or Program, click on the
Convert button beneath the appropriate column. The selected Partition, Volume or
Program will be imported along with all associated audio files.
6 Any audio files imported will be stored within a folder which matches the name of the
Volume. This folder is created within the Logic > AKAI Samples folder. The Sampler
Instrument(s) created by the import procedure matches the Program name(s). It is
placed inside the Sampler Instruments folder, or a sub folder as determined by the Save
converted instrument file(s) into sub folder parameter discussed in “AKAI File
Organization” on page 536.
Note: Should you wish to convert an entire AKAI CD ROM, click on the Convert entire CD
button found to the lower right of the AKAI Sample Import window.
Sub-folders named after the Volume are created when converting a partition. If a
Volume only contains one program, no sub-folder is created. Sub-folders named after
the Partition are created when converting more than one Partition.
AKAI File Organization
In the following graphic, the AKAI-Strings folder contains several Volumes, which contain
Programs.

The VOLUME 002 folder contains four patches—BCL PT M F, BCL PT M, BCL PT ST F, and
BCL PT ST. The according Sampler Instruments are stored in the Sampler Instruments >
Akai-Strings > VOLUME 002 folder.
The audio files associated with these Sampler Instruments appear in the AKAI Samples >
VOLUME 002 folder.

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When a Program is imported, these programs appear as Program.EXS in the Sampler
Instruments folder.
Sampler Instrument management works with AKAI samples imported from CD ROM, in
the same fashion as with other sample formats. Given the different file structures used
by many AKAI format discs, however, you should take care to follow these guidelines.
• Create a shortcut to any folder on your hard disk/s which contains your AKAI sample
library (or where you wish to store it). Name the shortcut “AKAI Samples” and all
converted AKAI CDs/samples will automatically be saved in this destination folder.
The “AKAI Samples” shortcut must be placed within the Sampler Instruments folder.
• If converting an entire CD ROM, you can create a shortcut with the sample CD’s
name—“Dance MegaSynth” for example. This can be placed in the Sampler
Instruments folder directly, or as a sub-folder within the AKAI Samples folder. The
advantage with the second method is that all imported AKAI Instruments will be
placed under the AKAI Samples sub-menu within the EXS24’s load window flipmenu.
Note: Assuming that an entire CD has been converted, you will find an AKAI Samples
folder (which actually contains the raw sample data) and several Partition folders within
the destination folder. The Partitions may contain several folders which bear the name
of the imported instruments. The .EXS files (the EXS Instruments) may be contained in
either the Instrument or Partition folders.
Additional AKAI Convert Parameters
Within the AKAI Convert window, you will find additional parameters listed below the
four gray column areas. We will discuss these in their order of appearance.
Save converted instrument file(s) into sub folder.
Entering a name into this parameter field is achieved by clicking once with the mouse
and typing in the desired name, followed by pressing Return or Enter respectively. In the
example shown within “AKAI File Organization” on page 536, an AKAI-Strings folder was
created.
All imported Volumes and Programs will automatically be added to this menu, and
folder structure, until the name is changed. This facility may be useful, particularly
when importing an entire CD, to create a folder name which reflects the CD-ROM’s
name. Alternately, you may wish to use a category name, such as Strings. This way, any
imported Programs or Volumes will be added to the Strings category.
Note: If an existing category name is used, the imported Sampler Instrument will be
added to the folder/menu. It will not create a new menu entry/folder of that name.

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Default instrument output volume (head room)
This parameter is extremely useful for many AKAI CD-ROMs. Please select this option
before converting a CD-ROM.
• For drum CDs, select a headroom value of −− 3 up to zero dB.
• For piano/string/pad CDs, a headroom value of −− 9 dB is recommended, or the
sound may/will clip with polyphonic use of these types of instruments.
• In cases where you’re not sure of which headroom value to select, choose − 6 dB
(average).
Merge programs (same MIDI cha. and prog. change number) into one EXS
instrument
This parameter is Off by default. Its use is dependent on the structure of program
material on the CD-ROM being imported.
To explain, many CD-ROMs created for AKAI samplers may feature several programs
that contain single velocity layers for an instrument. AKAI samplers require the loading
of an entire volume, or all necessary single programs, to be able to hear/play all
velocity layers. All of these single programs are automatically assigned to the same
MIDI channel and also react to the same MIDI program change number.
The EXS24 AKAI Conversion intelligently checks for these settings, and will build a
single EXS Sampler Instrument out of multiple single programs. In general, this type of
behavior is desirable with these types of CDs. When importing samples of this type, this
option should be set to ON.
The same is true for drum CD-ROMs where single programs contain one instrument
from a complete drum kit (kick/snare/hi-hat and so on as separate entities) You’ll
probably want these single AKAI programs to be merged into a single EXS Sampler
Instrument as a full drum kit.
There are, however, a number of AKAI CD-ROMs where a single program of an AKAI
Volume contains the entire instrument, and where other programs in the same Volume
have the same MIDI channel and MIDI program change number preset. On this type of
CD-ROM, use of the merge programs parameter is not desirable, and the option should
be set to OFF.
Create interleaved stereo files whenever possible
This option should always be left enabled, as interleaved files offer better performance
within the EXS24. When executing an AKAI conversion, some audio files are created as
split stereo and as interleaved stereo files.
The detection of when it is possible to build an interleaved file is based on information
stored with both the AKAI Program and audio files. Both the left and right files must
have the same settings; otherwise they can not be used to create an interleaved file/
multiple interleaved files.

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Prelisten Function
The AKAI Import window features a Prelisten button, which is found below the Audio
Files column. This facility allows you to individually audition AKAI audio files before
deciding whether or not to import them.
To prelisten an audio file:
1 Select an individual file (sample) within the Audio Files column:
2 Press Prelisten. This will start playback of the selected audio file and the Prelisten button
will update, with the word “Stop” appearing on the face of the button.
3 The selected audio file will loop continuously until you press the Stop button.

Vienna Library
The EXS24 features an additional interface for the Vienna Symphonic Library—
Performance Set. The Performance Tool software provided by VSL needs to be installed
to allow access to this interface. For details please refer to the VSL documentation.

Plug-in Window Parameters

Legato/Mono/Poly Buttons

These switches determine the number of voices used by the EXS24:
• When Poly is selected, the maximum number of voices is set via the numeric field
alongside the Poly button. To change the value, click and hold with your mouse, and
drag up or down to increase/decrease polyphony.
• When Mono or Legato is selected, the EXS24 is monophonic, and uses only one voice.
• In Legato mode, Glide is only active on tied notes. Envelopes are not retriggered
when tied notes are played (single trigger).
In Mono mode, Glide is always active and the envelopes are retriggered by every note
played (multi trigger).

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Voices
This parameter determines the number of voices (polyphony) that the EXS24 is
supposed to play. The used field indicates the number of voices that are actually used. If
both fields tend to show the same value most of the time (probably causing a
noticeable number of samples to drop out), you should set a higher voices value.
Unison Mode
This mode plays multiple EXS24 voices when each key is triggered:

• In Poly mode, two voices per note.
• In Mono or Legato mode, you can adjust the number of voices per note with the

voices parameter (this value is limited to 8—which is more than enough for fat
unison sounds!)
The voices are equally distributed in the panorama field and are symmetrically
detuned, dependent on the Random knob value.
Note: The number of voices actually used per note increases with the number of
layered sample zones.
Sampler Instrument Selection Pull-Down Menu
This menu allows the selection and loading of a Sampler Instrument into your
computer’s RAM. In order to appear within this list, a Sampler Instrument must reside in
the Sampler Instruments subfolder of Logic’s program folder.

You will find plus (+) and minus (−) buttons to the left and right of the Instrument Load
pull-down menu/display. These buttons allow you to browse to the next/previous
Instrument (sound) of your sound library (if necessary, this will change folders in
accordance with their order of appearance in the menu). Please note that the global
Next/Previous EXS Instrument key and MIDI commands also perform the same function.
Edit Button
This button to the right of the Sampler Instrument selection pull-down menu opens
the currently loaded Sampler Instrument in the EXS24 Instrument Editor. If none is
loaded, the Sample Editor will open, allowing the creation of a new Sampler
Instrument.

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Options Button
Clicking the Options button launches a menu that offers the following options:
• Recall default EXS24 settings recalls a neutral setting for all parameters in the Plug-in
window.
• Recall settings from instrument command manually recalls the original parameter
settings of the loaded Sampler Instrument. This parameter is extremely useful if
you’ve been over zealous with your tweaking.
• Save settings to instrument parameter stores the current settings of the Plug-in
window into the Instrument file. When the Instrument is reloaded, these settings are
restored in the Plug-in window.
• Delete Settings from instrument removes the stored settings from the Instrument.
• Rename instrument allows the renaming of the currently opened Sampler Instrument.
When invoked, a file dialog box will open. This will overwrite the existing Instrument
name.
• Save instrument as allows the storage of the currently opened Sampler Instrument
under a different name. When invoked, a file dialog box will open.
• Delete instrument will delete the opened Sampler Instrument.
• (Recall default EXS24 mkI settings) does almost the same as the first entry, but the
settings for the former version of the EXS are recalled for the selected Instrument,
especially the former modulation paths (see “EXS24 mkI Modulation Paths” section,
from page 552 onwards).
• Extract MIDI-Region(s) from Recycle Instrument allows you to extract the Regions
contained in a Recycle Instrument. If no Recycle Instrument is selected, this option is
not active.
• AKAI Convert launches the AKAI Convert window (see “To convert AKAI files” on
page 535). This menu option accelerates working with AKAI samples, as you do not
need to open the EXS24 Instrument Editor.
• SoundFont Convert
SampleCell Convert
DLS Convert
Giga Convert each will launch a dialog with instructions on performing these
conversions. In order to play back long Gigasampler audio files, the Virtual Sample
Memory option should be active (see below).
• Preferences opens a window with preferences for each Sampler Instrument (see
“Preferences” section, from page 558 onwards).

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• Virtual Memory opens a settings window for the EXS virtual memory functions.

Virtual memory allows samples of almost unlimited length to be played back using
streams that are fed directly from the hard disk. Switch off this option if you have
enough RAM for your current work.

The Active checkbox switches virtual memory on or off. In the General Settings, you can
set the Disk Drive Speed and the Hard Disk Recording Activity. The Requires Constant RAM
allocation of field displays the memory usage required by the two parameters
mentioned above. The Performance section contains two fields that show the current
Disk I/O Traffic and the data Not Read from Disk in Time. Should these values rise to high
levels you should change the General Settings to free up additional RAM for virtual
memory use. The Cancel button rejects any changes made in the window.
Hold Pedal and Crossfades
Hold via
This parameter determines the modulation source used to trigger the sustain pedal
function (hold all currently played notes, and ignore their note off messages until the
modulation source’s value falls below 64). The default is controller number 64 (MIDI
standard). You can change it if there are reasons to prevent Sustain from using CC 64,
or if you wish to trigger Sustain with another modulation source.

Crossfade (Xfade)
Xfade allows you to crossfade between layered sample Zones with adjacent Select
Range settings (Select Range was labeled Velocity Range in earlier versions). Please read
the “Sample Select” section which follows.

Crossfades are controlled by two parameters:

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Amount
This is the range of velocity (or other modulation source) values in which the crossfade
takes place. The Select Range setting of all Zones will be expanded by this value, with
the crossfade taking place in the expanded area. When the Amount parameter is set to
0, the EXS24 will switch between sample Zones in exactly the same fashion as earlier
versions (Velocity Switching).
Type
You can choose between three different fade types for the velocity crossfade:
• dB linear
• linear (gain linear)
• equal power
Sample Select
This is just another modulation Destination—but it is important to know a little bit
more about it. By default, Sample Select is controlled by velocity (via the default
Velocity to Sample Select modulation path). The velocity value determines which of the
layered Zones with different Select Range settings (velocity layers) is heard. You can
also use modulation sources other than velocity—even multiple sources (in multiple
modulation paths)!
If planning to do so, however, please keep in mind the fact that all sources (except
Velocity and Key) cause all velocity layers to run simultaneously—using up as many
voices as there are layered Zones. This also happens in cases where the Zones are not
audible at the current control level.
If a continuous controller (such as the modulation wheel) is chosen, you can step
through the velocity layers during playback. This is where the XFade parameter
becomes important, as it allows smooth transitions between velocity split points.
Keep in mind that you can also combine velocity and modulation wheel control by
using the Modulation Matrix (see “Modulation Matrix” on page 550).

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Pitch Parameters

Tune
Offsets the pitch of the sample(s) in semitones by up to ±2 octaves. The middle
position of the knob (which can be set by clicking the small 0 button) leaves the pitch
unaltered.
Transpose
This parameter allows you to transpose the EXS24. In contrast to the Coarse Tune
parameter, Transpose not only affects the pitch, but also moves the Zones in
accordance with the Transpose setting.
Random
This rotary knob controls the amount of random detuning which will apply to every
played note. Random ranges from 0 to ±50 cents.

You can use Random (detune) to simulate the tuning drift of analog synthesizers. This
parameter can also be effective in emulating a natural feel for some stringed
instruments.
Fine
Allows the EXS24 to be fine-tuned.
Pitch Bend Up (▲)
The amount of pitch bend (in semitones) that can be introduced by moving the pitch
bend wheel to its maximum position.
Pitch Bend Down (▼)
The amount of pitch bend (in semitones) that can be introduced by moving the pitch
bend wheel to its minimum position. When Linked is selected, the Pitch Bend Up value is
used.

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Remote
The Remote parameter allows you to easily pitch complete EXS24 Instruments in realtime. To do so, set the Remote parameter to the key of your MIDI Keyboard that you
would like to use as the original pitch. All keys in a range of ±1 octave around this key
will now pitch the entire Instrument. This two octave range is similar to the Pitch Bend
function, but is quantized to semitones.

Please note that the 2 octaves of remote keys don’t actually trigger the instrument—
they are used exclusively for semitone tuning.
Glide
The effect of this slider depends on the setting of the Pitcher slider: When Pitcher is
centered, Glide determines the time it takes for the pitch to slide from one note to
another (portamento). When the Pitcher parameter is set to a value above its centered
value, Glide determines the time it takes for the pitch to glide down from this higher
value back to its normal value. When Pitcher is set to a value below the centered value,
the pitch glides from this lower setting back up to the normal value.

Pitcher
The Pitcher slider works in conjunction with the Glide slider: When the Pitcher is
centered (which can be set by clicking the small Port(amento) button), Glide determines
the portamento time. When Pitcher is set to a higher or lower value, a pitch envelope is
activated. In this scenario, Glide determines the time it takes for the pitch to glide from
the higher/lower Pitcher setting back to the original value. The Pitcher parameter can be
modulated by velocity: the upper half of the slider determines the setting for
maximum velocity, the lower half for minimum velocity. By clicking and dragging in the
area between the two slider segments you can move both simultaneously.
Please note that the upper half of the Pitcher slider can be set above the center
position, and the lower half below the center position. When the Pitcher sliders are set
in this fashion, lower velocity values cause the pitch to rise from the lower setting back
to the original note pitch, while higher values cause it to fall from the higher setting
back to the original value. In other words: the polarity of the pitch envelope can be
changed according to velocity values.

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When both halves of the pitcher slider are set below or above the centered position,
either a low or high velocity will slide up/down to the original pitch. Dependent on the
position of the upper/lower halves of the slider in relation to the center position, the
time required for the slide up/down to the original note pitch can be adjusted
independently for both soft/hard velocities.
Filter Parameters

Filter On/Off Switch
This button switches the filter section on or off. Please note that the knobs and buttons
in the silver panel area and the Filter Envelope are only active when the filter is turned
on. When the filter section is turned off, the EXS24 is far less CPU-intensive.
Lowpass (LP)
The Lowpass Filter offers four different settings for its cutoff steepness: 24 dB (4 pole),
18 dB (3 pole), 12 dB (2 pole), and 6 dB (1 pole). The 24 dB setting can be used for
drastic sweep effects, such as cutting off all but a few notes, or for the creation of ultradeep bass sounds with just the necessary amount of overtones. The slope setting of
6 dB per octave is very useful in cases where you want a slightly “warmer” sound,
without drastic filter effects—to smooth “overly bright” samples, for example. The two
remaining values may be used for any purposes.
Fat (Fatness)
The Fatness mode is separate from the slope setting, and can be used with all available
slope values. Fatness preserves the bass frequency response, even when high
Resonance settings are used. Please note that this only applies to Lowpass filters.
Fatness is non-functional when used in conjunction with the High or Bandpass filters.

Highpass (HP)
The Highpass Filter is a 2 pole (12 dB/Oct.) design. A Highpass filter reduces the level of
frequencies that fall below the cutoff frequency. It is useful for situations where you
would like to suppress the bass and bass drum in a sample, for example, or for creating
classic highpass filter sweeps.

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Bandpass (BP)
The Bandpass Filter is a 2 pole (12 dB/Oct.) design. A Bandpass filter only allows the
frequency bands directly surrounding the cutoff frequency to pass. Frequencies which
fall outside these boundaries will be cut.
Drive
This knob allows the filter input to be overdriven. Turning Drive up leads to a more
dense and saturated signal, with additional harmonics being introduced/becoming
audible.
Cutoff
The cutoff frequency of the lowpass filter. As you turn this knob to the left, an
increasing number of high frequencies are filtered from the signal. The Cutoff value also
serves as the starting point for any modulation involving the filter.
Resonance
Turning up Resonance leads to an emphasis of the frequency area surrounding the
frequency defined by the Cutoff parameter. Very high Resonance values introduce self
oscillation, and cause the filter to produce a sound (a sine wave) on its own.
Simultaneous Control of Cutoff and Resonance

By clicking and dragging on the chain symbol located between the Cutoff and the
Resonance knobs, you can control both parameters simultaneously: vertical mouse
movements alter Cutoff, and horizontal mouse movements affect Resonance values.
Key
This knob defines the amount of filter cutoff frequency as determined by note number.
When Key is fully turned to the left, the cutoff frequency is not affected by the note
number, and is identical for all notes played. When Key is set fully right, the cutoff
frequency follows the note number 1:1—if you play one octave higher, Cutoff is also
shifted by one octave. This parameter is very useful in avoiding overly filtered high
notes.

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Volume and Pan Parameters
Level via Vel
Controls the volume of the sound. The Level parameter can be modulated by
velocity: the upper half of the slider determines the volume for maximum velocity, the
lower half for minimum velocity. By clicking and dragging in the area between the two
slider segments, you can move both simultaneously.
Volume
The main volume parameter for the EXS24. Move this knob to find the right balance
between avoiding distortion and getting the best (highest) resolution in the channel
fader and the Level via Vel slider.

Key Scale
This parameter modulates the sound’s level by note number (position on the
keyboard). Negative values increase the level of lower notes. Positive values increase
the level of higher notes.
Amp Envelope (ENV 2)
This is an ADSR envelope generator for controlling the sound’s level over time. It offers
Attack, Decay, Sustain, and Release parameters.

The attack time can be reduced by velocity: the upper half of the slider determines the
time for minimum velocity, the lower half for maximum velocity. By clicking and
dragging in-between the two slider segments, both can be moved simultaneously.

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LFO Parameters

LFO 1 EG
This knob allows LFO 1 to be faded out (Decay area) or faded in (Delay area). In the
centered position (which can be set by clicking on the small 0 button), the LFO
intensity is constant.
LFO 1 Rate
This is the frequency of LFO 1. It can be set in note values (left area), or in Hertz (right
area). In the centered position (which can be set by clicking on the small 0 button), the
LFO is halted and generates a constant modulation value at full level (DC = Direct
Current).
This allows you to perform a nice trick: Set up an LFO to modulate, say, Cutoff, with the
modulation wheel controlling the LFO’s intensity. Then set the LFO’s rate to DC. As the
LFO’s modulation intensity is controlled via the modulation wheel, you can now make
use of the modulation wheel to manually open the filter.
Waveform for LFO 1 and LFO 2
These two switches allow the selection of the waveform type used by LFO 1 and LFO 2.
A selection of Triangle, falling and rising Sawtooth, Square up and Square down, a
random stepped waveform, and a smoothed random waveform is available for each
LFO.
LFO 1 is a polyphonic LFO with key synchronization. This means that when LFO 1 is
used, each voice of the EXS24 has its own discrete LFO. When a note is played, the LFO
corresponding to that voice starts its cycle. This scheme means that the LFO cycles of
each voice played are not synchronous, and operate independently of each other. This
opens up a range of modulation possibilities. As an example—the LFO of one voice
could generate the maximum modulation value, while the LFO assigned to another
voice could output its minimum value. This extremely flexible approach can result in
some very lively modulations.
In contrast, LFO 2 is a monophonic LFO without key synchronization. This means that
LFO 2 runs continuously, and is not restarted by the triggering of a new note. All voices
are modulated by the sole LFO, so the degree of modulation at any given time is the
same for all voices. This results in a rather synthetic-sounding modulation.
Use these different characteristics to tailor the sound to your needs.

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LFO 2 Rate
The frequency of LFO 2. It can be set in note values (left area), or in Hertz (right area). In
the centered position (which can be set by clicking on the small 0 button), the LFO is
halted, and generates a constant modulation value with full level (DC = Direct Current).
Again, don’t overlook this feature if you want to control an LFO-modulated parameter
directly via the Modulation Matrix (see following section).
LFO 3 Rate
There is a third LFO available which always uses a triangular waveform. LFO 3 can
oscillate freely between 0 and 35 Hz, or can be tempo synchronized in values between
32 bars and 1/128 triplets.
Modulation Matrix

The Modulation Matrix is the dark horizontal band that spans the center of the EXS24
interface. It consists of ten modulation paths, each linking a modulation Source with a
modulation Destination (the sound parameter that you want to modulate). This is
similar to the use of patch cords on modular synthesizers, but with the additional
option of control over the modulation amount via another modulation source.
Creating a new modulation path is easy:

• first, choose the Destination (Dest)
• then choose the Source (Src)

The green triangular fader on the right side of each modulation path allows you to set
the modulation depth with a bipolar range (positive or negative value).
In this example, the LFO 1 Speed is modulated by channel pressure (aftertouch)
messages of a MIDI keyboard.

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You have the option of inserting another modulation source in the middle slot labeled
via. In this scenario, the green triangular fader will divide, allowing you to set a range
for modulation depth. The size of the modulation range depends on the possible
values allowed by the via modulation source.
In our example, the key number of the MIDI keyboard (Key) determines how strongly
channel pressure controls the Speed of LFO1. More experienced users would read the
picture like this: pressure to LFO1 Speed via key number.

Inverting Sources
You can also invert the direction of the source’s effect on modulation depth by clicking
the inv button (right of the word Src or via), depending on which of the sources you
would like to invert.
In this example, we inverted the via modulation source. You can see how the green and
orange triangles have swapped positions. The orange triangle always marks the
modulation depth for the maximum value of the via source, while the green triangle
always marks the modulation depth if the via source is at its minimum value. They are
reversed by inverting the modulation.

Bypassing Modulation Paths
You can temporarily disable the entire modulation path with the b/p button, found
alongside the word Dest.

0
In our example, both modulation sources—Pressure and Key—are disconnected from
the LFO1 Speed modulation destination. Clicking the b/p button a second time
reconnects the modulation path, restoring the old modulation depth settings.

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Second Order Modulations
The EXS24 also allows the use of second order modulation destinations (such as
envelope times, LFO speeds and so on)—functionally outperforming many analog
synthesizers. To explain:
• The same source can be used as often as desired to control different destinations.
• The same destination can be controlled by different sources. The different input
values are accumulated.
EXS24 mkI Modulation Paths
Many of the hard-wired modulation paths that were available as sliders on the original
EXS24 (mkI) are now part of the Modulation Matrix. In order to reconstitute the
modulation slider configuration of the mkI version, click on the options button in the
upper-right corner and choose (Recall default EXS24 mkI settings) from the pop-up
menu. This will load the mkI modulation paths into the Modulation Matrix, as follows:
• Velocity to Sample Select
• LFO 1 to Pitch via ModWheel (= Ctrl#1)
• Velocity to Sample Start (inv)
• LFO 2 to Filter Cutoff via ModWheel
• Velocity to Filter Cutoff
• Envelope 1 to Filter Cutoff via Velocity
• LFO 2 to Pan via ModWheel
You can, of course, freely alter the settings of these modulation paths. To exchange
modulation sources with sources that were not available in EXS24 mkI, for example (see
the complete list of sources and destinations below).
EXS24 mkI Backward Compatibility
For technical reasons, the settings of the Modulation Matrix can not translate
backwards to the EXS24 mkI. If you are a sound designer and want your products to be
compatible for Logic 4.x users, we recommend the use of Logic version 5.3
(Mac OS X: 5.4) with the EXS24 mkI.

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Modulation Sources

Modulation Destinations

Sample Select

Side Chain (level)

Sample Start

Maximum

Glide Time

Env1

Pitch

Env2 (Amp)

Filter Drive

LFO 1

Filter Cutoff

LFO 2

Filter Resonance

LFO 3

Volume

Release Velocity

Pan

Pressure

Relative Volume

Pitch Bend

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Modulation Sources

Modulation Destinations

Relative Volume (auto adjust)

Key

LFO1 Dcy./Dly

Velocity

LFO1 Speed

Control Nr. 1

LFO2 Speed

…

LFO3 Speed

Control Nr. 120

Env1 Attack
Env1 Decay
Env1 Release
Time
Env2 Attack (Amp)
Env2 Decay (Amp)
Env2 Release (Amp)
Hold

Note: Controllers 7 and 10 are marked as (not available). Logic uses these controllers for
volume and pan automation of the audio object. Controller 11 is marked as
(Expression). It has a fixed connection to this functionality, but it can also be used to
control other modulation sources.

Multiple Outputs
The EXS24 mkII offers five stereo pairs and six individual mono outputs. The first ten
EXS24 outputs are always configured as stereo pairs. Outputs 11 through 16 are mono
outputs.
The main output pair (1–2) of the EXS24 is assigned to its Audio Instrument channel.
Additional outputs, starting with output 3, are accessed via the Aux channels. Please
refer to the paragraph on Aux channels below.
Individual output assignments may be set for Groups, or Zones, within an EXS24
Instrument. If zones are assigned to their individual outputs, any existing group
outputs for the zones will be ignored. All output assignments are stored with the EXS24
Sampler Instrument.
If the EXS24 is inserted as a regular (not multi channel) mono or stereo plug-in, all
multiple output assignments stored with a Sampler Instrument are automatically
routed to the main stereo output pair, or mono output, respectively.
The multiple outputs of the EXS24 are managed intelligently: Multiple outputs
accessed via the Aux channels are automatically subtracted from the main output.
Conversely multiple outputs that are not accessed via Aux channels are automatically
routed to the main stereo output pair.

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Aux Channels for Instrument Plug-ins with Multiple Outputs
By default, all outputs of a “Multi Channel” Instrument plug-in are summed and routed
to the stereo output of the respective Audio Instrument channel at first. Signals of
individual outputs (starting with output 3) are automatically subtracted from the stereo
output sum after they have been assigned to Aux channel inputs.
Outputs 1 and 2 are always assigned to the respective Audio Instrument. This
assignment is fixed and can’t be changed.
Note: Additional instrument outputs are only available to plug-ins inserted into an
Audio Instrument channel via the Multi Channel sub menu. Aux channel Input
sources—being sent from suitable Instrument plug-ins—start from output 3.

Instrument Editor Parameters
Instrument Menu
New

Creates a new, empty Sampler Instrument. In order to load a sample, you will need to
create a new Zone (see below).
Open
Allows an existing Sampler Instrument to be loaded into the Instrument Editor.
Close
Closes the currently opened Sampler Instrument. If you have made any changes to the
Instrument, you will be asked if you would like to save them.
Save
Saves the currently loaded/edited Sampler Instrument. When you create a new
Instrument and save it for the first time, you will be asked to give it a name. If you have
edited an existing Sampler Instrument and save it via this command, the existing file
name is used and the old version is overwritten.

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Save As…
This command also saves the currently loaded/edited Sampler Instrument. When Save
As… is used, you will be prompted to give the file a name. Use this command when
you want to save a copy of an edited Sampler Instrument, rather than overwriting the
original version.
Rename
This command allows you to rename the loaded Sampler Instrument. The renamed
version replaces the previous version on the hard disk.
Delete
This deletes the currently opened Sampler Instrument.
Move Audio files
Moves the audio files of the selected Sampler Instrument into a desired folder location.
Use of this option will launch a standard operating system file navigation/browse
utility. You may browse to an existing folder or enter a new name, as desired. If no
folder name is entered, a new folder will be created which matches the Instrument
name, and all audio files will be moved into this folder.
Copy Audio files
Operation is as per the Move Audiofiles function, but files are duplicated, rather than
moved, to the nominated folder. This facility should be used as part of your working
methods when creating Logic songs, as discussed in “Saving of Project-related EXS24
Instruments” on page 528.
AKAI Convert
Opens a window that allows you to convert files from an AKAI format CD ROM (see “To
convert AKAI files” on page 535).
Recycle Convert
There are different options here (please also read “Converting ReCycle Files to EXS24
Instruments” on page 534):

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Extract MIDI region and make new Instrument
Use of this menu option will launch a file selection dialog, allowing the selection of a
ReCycle file.
• Following selection, a new EXS24 Instrument with a name which matches that of the
ReCycle loop will be created. Should an EXS24 Instrument of that name already exist,
a # sign and a number will be appended, ensuring that the filename is unique within
the Sampler Instruments folder.
• You will be prompted to enter a velocity factor (see page 534). This should be left at
the default value of zero in most cases.
• Press the OK button (after entering a value, if applicable). The EXS24 will generate a
number of Zones (one for each “slice” of the imported ReCycle file) and one Group to
which the Zones are assigned.
• In addition, a MIDI Region is generated on the currently selected track, at the current
song position (rounded to whole bars). This MIDI Region is used to trigger the
imported “slices” at the timing defined by the ReCycle file. You can generate new
MIDI Regions at any time from the imported EXS24 Instrument (see below), so feel
free to modify or delete it.
Extract MIDI region and add samples to current Instrument
This option allows the addition of a ReCycle loop to any EXS24 Instrument currently
opened in the Instrument Editor. This permits the use of several different loops in a
single Sampler Instrument, which can be recorded and played on a single Audio
Instrument track. If no Instrument is open in the Instrument Editor, this function
behaves identically to the Extract MIDI region and make new instrument function.
Slice Loop and make new Instrument/Slice Loop and add samples to current
Instrument
The first option creates a new EXS24 Sampler Instrument while the second option adds
the Zones (from the sliced loop) to the currently active Sampler Instrument.
The Recycle slices are rendered into a single audio file (the whole Recycle loop), played
back at the current Logic song tempo. Rather than importing and playing back a single
discrete Recycle slice, each Zone will play back the Recycle loop to its very end, starting
with the slice points originally assigned to the respective Zones. To explain: the lowest
Zone will play back the entire loop, and the highest zone will only play the last slice of
the loop.
This option allows for loop trigger techniques often used in “old school” Drum’n’Bass
tracks, with the sample loop start point determined by playing the respective notes on
the keyboard.

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Extract Region(s) from ReCycle Instrument
This option generates MIDI Regions from EXS24 Instruments which were originally
converted from ReCycle file(s). The MIDI Regions are created on the currently selected
track, at the current song position (rounded to bars). A single MIDI Regions is
generated for each imported ReCycle loop in the currently open Instrument. If no
imported ReCycle loop exists in the currently open Instrument, this menu option is
disabled. This function will also ask for a velocity factor (see page 534).
Paste loop from clipboard as new Instrument
Functionally identical to the Extract MIDI region and make new instrument option with
the contents of the Clipboard used, rather than an existing ReCycle file. A loop must
have been copied into the Clipboard via ReCycle’s Copy loop facility for this to function.
Paste loop from clipboard to current Instrument
Functionally identical to the Extract MIDI region and add samples to current instrument
option with the contents of the Clipboard used, rather than an existing ReCycle file. A
loop must have been copied into the Clipboard via ReCycle’s Copy loop facility for this
to function.
Edit Menu

Undo
Allows the most recent change to the Sampler Instrument to be undone.
Redo
Undoes the last Undo command.
Undo History…
Opens the Undo History. Here you can undo or redo arbitrary operation steps. More
information can be found in your Logic manual.
Delete Undo History
Deletes the Undo History. More information can be found in your Logic manual.

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Cut, Copy, Paste
The standard commands for cutting, copying, and pasting values. In addition to values
you may also cut, copy, and paste selected Zones and Groups.
Note: When multiple Zones and Groups are cut, copied, or pasted simultaneously, the
Group assignments of the Zones are retained.
Clear
Deletes the currently selected Zone or Group. Clear can be undone with the Undo
command.
Select All
Selects all Zones and Groups of the loaded Sampler Instrument.
Select Zones pointing to selected group(s)
This command automatically selects all Zones that point to one or multiple selected
Groups.
Preferences

The Preferences window allows you to:
Choose the interpolation quality used by the EXS24. When Sample Rate Conversion is
set to Best, the highest possible sound quality is maintained when transposing samples.
It should be noted that this option requires additional CPU cycles over the Normal
setting, which will be adequate in most cases.
Select the format in which the EXS24 handles the loaded sample data via the Sample
Storage parameter. When set to Original, the samples are loaded into RAM at their
original bit depth, and are converted to Logic’s internal 32 Bit floating point format on
playback. When 32 Bit Float is selected, the samples are stored and loaded in this
format. This eliminates the need for any realtime conversion, meaning that the EXS24
can handle the sample data more efficiently and can play back more voices
simultaneously. It should be noted that this requires twice as much RAM for 16 bit
samples, and a third more RAM for24 Bit samples.

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The Velocity Curve parameter determines the EXS24’s responsiveness to velocity values
received from your MIDI keyboard. Negative values increase the response to soft key
strikes, and positive values decrease it.
The Search Samples On parameter determines the location that instruments samples
should be searched in. You may either choose the drives normally used by the
operating system or external SCSI, FireWire or USB drives, accessible directly or over a
network.
Drives can be selected individually, or grouped as follows:
• Local Volumes internal storage media (hard disks and CD ROM mechanisms) attached
to or installed in the computer directly.
• External Volumes storage media accessible over a network.
• All Volumes both internal and network media are scanned for appropriate data.
Note: Selecting External or All Volumes may result in a dramatic increase in the time
required by the EXS24 to find and load Sampler Instruments and files.
The following two preferences are particularly useful when used in conjunction with
the function discussed in the “Load Multiple Samples” section, from page 561 onwards.
Read Root Key From:
• file/filename—initially reads information about the root key from the file itself (in the
header of the AIFF or WAVE file) when loading an audio file into a zone. If no
information of this type exists in the file header, a smart analysis of the filename may
detect a root key. If this second method doesn’t provide any useful results, C3 will be
used as the default root key in the zone.
• filename/file—as above, but vice versa, with the filename read first, and the header
read second.
• filename only—reads from the filename only. If no root key information exists, C3 will
automatically be assigned to the zone as the root key.
• file only—reads from the file header only. If no root key information exists, C3 will
automatically be assigned to the zone as the root key.
Root Key at File Name Position
When loading an audio file into a zone this option is used for the analysis of a root key
in the file name. Possible options in the pull-down menu are “Auto”, or numerical values
from 1 to 30.
• Auto is the recommended value. It provides a smart analysis of numbers and keys in
the file name. A number in the file name can be recognized, regardless of its
format—“60” or “060” are both valid. Other valid numbers can range between 21 and
127. Numerical values outside of these are generally just version numbers. A key
number is also a valid possibility for this use—“C3”, “C 3”, “C_3”, “A-1”, “A -1” or “#C3”,
“C#3”, for example. The possible range is “C-2” up to “G8”.

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Note: There may be cases where a sound designer has used multiple numbers in a
filename, which is common with loops, with one value being used to indicate tempo—
“loop60-100.wav”, for example. In this situation, it isn’t clear which, if either of the
numbers, indicates a root key or something else: 60 or 100 could indicate the file
number in a collection, tempo, root key, and so on. You can set a value of “8” to read
the root key at position (letter/character) eight of the filename—namely the 100 (E6).
Alternately, setting a value of “5” will select the 60 (C3) as the root key position.
Previous/Next Instrument
Previous/Next Instrument and the number fields alongside determine which MIDI event
type (and data value) will be used for selection of the previous or next
Instrument: Note, Poly Pressure, Control, Program, Channel Pressure, Pitch Bend.
In the number field (depending on the event type), either the Note Number or the
value of the first data byte can be entered. When Control is selected, the number field
determines the Controller number.
Giga Convert includes Release Trigger
Determines whether or not the Release Trigger function of the Gigasampler format will
be performed by the EXS. This is important if you want to stay compatible with the
EXS24 mkI, which doesn’t offer this functionality.
Ignore Release Velocity
This option also refers to the Release Trigger function and should always be set to on
for this purpose. Regardless of whether or not your keyboard is able to send Release
Velocity, you would want your samples played by the Release Trigger function to be
louder or softer than the original Sample, or at the same volume, regardless of the
initial velocity. When playing with Release Trigger, you would want the Release Velocity
value to have the same value as the Initial Velocity value. To accomplish this, you can
switch off Release Velocity.
Keep common samples in memory when switching songs
Determines whether or not the samples commonly used by two open song files are
reloaded when switching between songs.

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Zone Menu

New Zone
Creates a new Zone in the currently loaded Sampler Instrument.
Load Multiple Samples
Allows several samples to be loaded in one operation. The Instrument Editor reads the
key note from samples, and places the samples into new Zones. These Zones are
automatically created. The key note is located in the middle of the Zone, with Zone
borders being determined by the Zone borders of neighboring Zones. Zones with
identical key notes will be layered. A root key is automatically determined by use of the
settings described in the “Preferences” section, from page 558 onwards.
Three automatic mapping functions are available when loading multiple samples:
• Auto map (default) uses the key note information (root key) stored with the audio files
and places the samples (as zones) over the keyboard range. The number of keys that
constitute a zone is intelligently determined by the placement of neighboring zones.
• Drums uses the key note information (root key) stored with the audio files. Each zone
contains a single key on the keyboard determined by the key note information.
• Chromatic ignores all key note information (root key) of the audio files and places the
samples on the keyboard in chromatic order, starting at C1.
Move selected to the top
When this option is activated, the parameter window of a selected Zone is moved to
the top of the onscreen Zone listing, and is automatically opened.
Select zone of last played key
When active, this menu option allows you to switch between Zones by pressing a key
on the onscreen keyboard, or via a connected MIDI keyboard.
Show End as Length
When this option is activated, the sample length is shown in the Zone’s parameter
window, rather than the end point value.

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Sort by…
These commands determine the order in which the Zone windows are displayed:
Sort by None: The Zones are shown in the order in which they were created.
Sort by Name: The Zones are sorted alphabetically.
Sort by root key (high to low): The Zones are sorted according to their Root Key settings;
the higher key notes are displayed at the top of the list.
Sort by root key (low to high): The Zones are sorted according to their Root Key settings;
the lower key notes are displayed at the top of the list.
Sort by Velocity (high to low): The Zones are sorted according to their Velocity Range
settings; the higher velocity ranges are displayed at the top of the list.
Sort by Velocity (low to high): The Zones are sorted according to their Velocity Range
settings; the lower velocity ranges are displayed at the top of the list.
Sort by Group: The Zones are shown in the order of their Group assignment(s).
Sort by File Name: The Zones are sorted alphabetically according to the names of their
associated audio files.
Update selected zone(s) info from audio file
This option is used when editing loops in an external sample editor. It reads loop
settings from the audio file and updates the settings of the Zone accordingly. In
addition, this feature also validates the loop’s length and start position.
Group Menu

New Group
Creates a new Group.
Sort by…
These commands determine the order in which the Group windows are displayed:
Sort by None: The Groups are shown in the order in which they were created.
Sort by Name: The Groups are sorted alphabetically.

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Sort by Velocity (high to low): The Groups are sorted according to their Velocity Range
settings; the higher velocity ranges are displayed at the top of the list.
Sort by Velocity (low to high): The Groups are sorted according to their Velocity Range
settings; the lower velocity ranges are displayed at the top of the list.
Delete unused Groups
This command deletes all Groups that do not have a Zone assignment. This deletion
can be undone with the Undo command.
View Menu

This menu offers display options for the Zone and Group windows:
• View All shows all available parameters. When this option is activated, you can
manually prevent certain parameters from being displayed by deselecting the
corresponding option from the menu.
• When Toggle Mode is on, only one parameter will be displayed at a time. You can
select the desired parameter by selecting the corresponding option from the menu.
• When Show filename when Zone is closed is activated, the names of loaded audio files
are shown in the title bars of closed Zone windows.
The following menu entries allow you to select the parameters you wish to be
displayed.
Note: When working with the Zone and Group windows, it can be useful to close or
open all windows at once. To do so, click on one of the triangles while pressing Option.
Note: To set a switch parameter for a number of (selected) Zones simultaneously, hold
Command when clicking the switch.

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Zone Parameters

Zone Name
New Zones are numbered consecutively. Double-clicking on a Zone’s number allows
you to enter a name instead.
Audio File
A file selection dialog can be opened by clicking on the gray field next to the Audio File
label. This will allow you to select and load a sample into a Zone. When loaded, the
sample’s name is displayed in the gray field. Additional information about the sample is
displayed beneath its name (sample format, sampling frequency, bit depth, mono/
stereo status and sample length).
Group
Group allows you to assign a Zone to an existing Group. When No Group is selected, the
Zone is only affected by the parameters of the Plug-in window.
Key Note/Tune
Key Note allows you to determine the note at which the sample will sound with its
original pitch. The Key Note is displayed as both a note name and a numerical value.
The Tune and cent fields allow coarse and fine-tuning of the sample in semitone and
cent increments.
Zone Range/Disable Pitch
The two Zone Range parameters allow you to define a key range for the Zone. When
Disable Pitch is activated, the sample is always played at its original pitch, regardless of
the note number.
Velocity Range
Activation of the Select Range option, in conjunction with the use of the two Velocity
Range fields allows you to define a velocity range for the Zone.

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One Shot/Reverse
Activating One Shot causes the Zone to ignore the length of notes used to trigger the
sample—the sample is always played to the end. This option is useful for drum
samples, where you often don’t want the MIDI note length to affect sample playback.
Reverse plays the sample from its end to its beginning. This option works nondestructively, leaving the audio data in the sample unchanged.
Volume/Pan/Scale
Volume adjusts the volume of the Zone.
Pan adjusts the pan position of the Zone. This parameter only works when the EXS24 is
used in stereo.
Scale—Negative Scale values make notes lower than the note position defined by the
Key Note sound louder than higher ones; positive values have the opposite effect. Use
this parameter for balancing the volume of a sample across the selected key range.
EXS Output
This parameter determines the outputs used by the Zone. Choices include the main
outputs, and paired channels 3 and 4, 5 and 6, 7 and 8, 9 and 10, or individual outputs
11 through to 16. This allows individual zones to be routed independently to Aux
channels in a multi channel EXS instance.
Start/End Frame
The Start Frame and End Frame parameters set the sample’s start and end points,
respectively. Clicking on the small E button between the two values will launch Logic’s
Sample Editor, allowing you to set the start and end points graphically.

Loop
The loop parameters become visible when this option is activated, and the sample will
loop when sustained MIDI notes are received.
Loop Start, Loop End—You can define discrete loop start and end points in these fields,
allowing you to cycle (loop) a portion of the audio file. Data entry is via the mouse as
slider, or by double-clicking and directly typing in a value. Clicking on the small E
button between the two values will launch Logic’s Sample Editor, allowing you to set
the loop start and end points graphically: Loop Start is represented by the LS marker
and Loop End, by the LE marker.
Tune—This parameter allows the tuning of the looped portion of the audio file to be
different to that of the non-looped portion. It is adjustable in cent increments (±50
cents).

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Auto Crossfade
In a crossfaded loop, there is no hard “cut” between the loop end and loop start points.
Rather, the loop end and start points are crossfaded for a smooth transition. This is
especially convenient with samples that are hard to loop, and would normally produce
clicks at the transition point—the “join” in the loop.
The Auto Crossfade field allows a pre-determined value (in milliseconds) to be used as a
default when the option is enabled. The higher the value, the longer the crossfade, and
the smoother the transition between the loop end and start points.
EqPower—EqPower allows you to enable an exponential crossfade curve that causes a
volume boost of 3 dB in the middle of the crossfade range. This will fade out/fade in
the joined portions of a loop at an equal volume level.
Note: The “perfect” settings for the crossfade parameters depend on the sample
material. A loop which cycles reasonably smoothly is the best starting point for a
perfect crossfade loop, but a crossfaded loop does not always sound better. Just
experiment a little with the parameters, and you’ll soon find out how, when, and where
they work best.

Group Parameters

Volume/Pan/Bal
Volume—Adjusts the volume of the Group, and therefore the volume of all assigned
Zones, simultaneously.
Pan/Bal—Adjusts the pan position of the Group (stereo balance for stereo samples),
and the pan position of all assigned Zones simultaneously.
EXS Output
This parameter determines the outputs used by the Group. Choices include the main
outputs, and paired channels 3 and 4, 5 and 6, 7 and 8, 9 and 10, or individual outputs
11 through to 16. This allows individual Groups to be routed independently to Aux
channels in a multi channel EXS instance.
Voices/Velocity Range
Voices—Determines how many voices the Group is allowed to use. This parameter is
discussed in the “Multiple Zones and Groups” section, from page 525 onwards.

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Select Range—The two Velocity Range parameters allow you to set up a velocity range
for the Group. The settings made here override the settings in the Zones, if
necessary: When a Zone’s velocity range is larger than that allowed by the Group
setting, the Zone’s velocity range is limited by the Group setting.
Trigger on
Key Down (default)—Zones pointing to this group are triggered on key down.
Key Release—Zones pointing to this group are triggered on key release. This allows you
to trigger additional samples on key release. This is useful for emulating organ key
clicks, for example.
Select by
If this option is enabled, you can define a specific key. Whenever the defined key is
pressed, zones pointing to this group are played by ensuing MIDI note data, and other
groups (selected by a different key) are not played. The defined key is not played itself,
it simply acts as a remote.
This feature also works with other MIDI events (Controllers, Bends), and Groups.
As you select the different menu options, the display will update to allow the entry of
the desired continuous controller number (CC 19, for example) and range, or Group
selection, as appropriate.
Key Range
The Key Range allows you to define a key range within the Group that will be affected
by this functionality.
ADSR Offsets
The amp(litude) envelope settings from the Plug-in window can be offset separately for
each Group by these parameters. Each time parameter has an offset range of
±9999 ms, the sustain level can be varied by ±50%. The offset fields are for Attack,
Decay, Sustain, and Release—from left to right.
Filter Offsets
The Cutoff Offset setting of the Plug-in window can be offset separately for each Group
(±50%).
The Resonance Offset setting of the Plug-in window can be offset separately for each
Group (±50%).

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EXS24 Key Commands
A number of key commands are available for the EXS24 which accelerate editing in
Logic, and provide additional functionality. They are found in the Key Commands
window.
These key commands have no default keyboard assignments, so you will need to create
them, should you wish to take advantage of these shortcuts and facilities. Please
consult your Logic reference manual for information on accessing the Key Commands
window and on the assignment of keyboard shortcuts to functions.
Previous Instrument
Selects the previous Instrument (when multiple Instruments are opened for editing)
allowing you to quickly switch between several Instruments. An Instrument must be
selected for this function to work.
Next Instrument
Selects the following Instrument (when multiple Instruments are opened for editing)
allowing you to quickly switch between several Instruments.
The Next/Previous Plug-In Setting or EXS Instrument key commands are also available.
They perform the same functions as above, but only in the topmost window. If the EXS
is the topped window, the key command will select the next/previous Instrument. If
another Plug-in window is selected, the key command will select the next/previous
plug-in Setting file.
Select zones pointing to selected group(s)
Following the selection of a Group, use of this function will select all associated Zones,
ensuring that any edits made are only performed on Zones associated with the
selected Group.
Previous Zone/Group
Selects the previous Zone/Group (when multiple Groups/Zones exist) allowing you to
quickly switch between them. A Zone or Group must be selected for this function to
work.
Next Zone/Group
Selects the following Zone/Group (when multiple Groups/Zones exist) allowing you to
quickly switch between them. A Zone or Group must be selected for this function to
work.
New Zone
Creates a new Zone.
New Group
Creates a new Group.

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View: All/Toggle Mode
Toggles between viewing all parameters in Zones and Groups and a limited view which
displays the Audio File name in Zones and the Volume/Pan parameters in Groups.
View: Next Zone Parameter
This key command is designed to aid in the adjustment of the same parameters in each
Zone. It limits the Zone display(s) to individual parameters (or rows of parameters) and
steps through them from top to bottom. This allows you to adjust the appropriate
parameters of all Zones in a Group more easily as working on a reduced set of
parameters is simpler. It should be noted that if all Zone parameters are visible before
the Next Zone Parameter Key Command is invoked, use of the function will switch to a
reduced view mode.
View: Next Group Parameter
Operation and functionality is as per the Next Zone Parameter, for Group Parameters.
Move Audiofiles
Moves the audio files of the selected Instrument into a desired folder location. Use of
this option will launch a standard operating system file navigation/browse utility. You
may browse to an existing folder or enter a new name, as desired. If no folder name is
entered, a new folder will be created which matches the Instrument name, and all
audiofiles will be moved into this folder.
Copy Audiofiles
Operation is as per the Move Audiofiles function, but files are duplicated, rather than
moved, to the nominated folder. This facility should be used as part of your working
methods when creating Logic songs, as discussed in “Saving of Project-related EXS24
Instruments” on page 528.
Move audiofiles of all instruments…
Moves the audio files of all Sampler Instruments in the Sampler Instruments folder to
the target directory of your choice. In the target location, folders for the audio files
associated with these Sampler Instruments are created.
Backup/Copy audiofiles of all instruments…
Copies the audio files of all Sampler Instruments in the Sampler Instruments folder to
the target directory of your choice. In the target location, folders for the audio files
associated with these Sampler Instruments are created. In addition, the Sampler
Instrument files themselves are also copied.
Backup audiofiles of all USED and ACTIVE instruments of current song…
Copies the audio files of all (active) Sampler Instruments used by the current song to
the target directory of your choice. Folders for the audio files associated with these
Sampler Instruments are created in the target location. All used Sampler Instrument
files are also copied.

Chapter 29 EXS24 mkII

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Move audiofiles of all USED and ACTIVE instruments of current song…
Moves the audio files of all (active) Sampler Instruments used by the current song to
the target directory of your choice. Folders for the audio files associated with these
Sampler Instruments are created in the target location.

A Brief History of Sampling
The idea of an instrument that could change its sound at any time, and that could
imitate any other instrument, dates back centuries. By the 15th century, organ builders
had managed to simulate violins, flutes, trumpets, and even human-like sounds with
their instruments. Some years later, organs were perfected that could imitate birdsong.
Following the inception of film sound, several instruments were built that used film for
the storage and playback of sound. Motion picture sound was based on the concept of
recording sound onto the film itself as a separate track. Changes in brightness were
read via an opto-electrical mechanism, and sound was replayed. This meant that sound
was transferred to light and graphics in the widest sense. Creative musicians of the
time began to scratch these films manually, to draw waves on them, and to film
gearwheels and other things in order to produce interesting sounds from these
images.
The immediate next of kin to today’s samplers, however, was the Mellotron. This was a
very bulky keyboard instrument that used a separate tape recording of an acoustic
instrument for each and every key. Pressing a key started the playback of the
corresponding tape; after releasing the key, the tape was drawn back by a spring. Due
to the very complicated electro-mechanical mechanism used by the Mellotron, it was a
very heavy and frequently unreliable keyboard instrument.
Compared to this, the first digital samplers at the beginning of the eighties seemed
ultra-modern, but from today’s point of view they did not offer much for their five or six
digit price tag: a few seconds of sampling time, and sound quality that is surpassed by
today’s speaking toys. Nevertheless, early samplers like the Fairlight CMI and E-mu’s
Emulator are considered legendary. They had a great impact on music and on the
development of electronic musical instruments in the following years.
Nowadays, hardware samplers all sound good and are comparatively affordable.
However, this is not the end of development for samplers. With computers getting
faster and faster, it is now possible to build a fully-fledged sampler entirely in software,
making hardware samplers unnecessary. Your EXS24 is proof of this…

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Chapter 29 EXS24 mkII

MIDI Controller List
Common

Pitch

Filter

Chapter 29 EXS24 mkII

Mono Mode

Voices

Start Fixed

Start via Vel

Time via Key

Attack Curve

112

• Pitch Bend (up)

• Pitch Bend (down)

• Transpose

Coarse Tune

• Fine Tune

Glide

Pitcher

Pitcher via Vel

Modulation LFO

Mod. Depth Fixed

Mod. Depth Wheel

Filter (on/off )

Filter Type

Filter LFO

Filter LFO Fixed

Filter LFO Wheel

Filter Cutoff

Filter Resonance

Filter Drive

Filter via Key

• Filter via Vel

Filter ADSR Fixed

Filter ADSR via Vel

Filter Attack

106

Filter Att. via Vel

107

Filter Decay

108

Filter Sustain

109

Filter Release

110

571

Volume

LFOs

572

Chapter 29 EXS24 mkII

• Output Volume

• Key Scale +/−

Level Fixed

Level via Vel

Tremolo/Pan LFO

Pan Modulation

Tremolo

100

Amp Attack

113

Amp Att. via Vel

114

Amp Decay

115

Amp Sustain

116

Amp Release

117

LFO 1 Dec./Delay

101

GarageBand Instruments

GarageBand Instruments are automatically installed with
Logic. You can insert them as per other software instruments.
GarageBand Instruments are accessible from the Stereo > Logic > GarageBand
Instruments sub-menu.

About GarageBand Instruments
GarageBand Instruments are software instrument plug-ins that are used in Apple’s
GarageBand application. Their inclusion makes the importing of GarageBand files into
Logic a trouble-free experience.
GarageBand Instruments are actually small variations of equivalent Logic instrument
plug-ins. In the case of synthesizer sounds, the ES2 is the “big equivalent” of the
GarageBand Instrument. In the case of organ sounds, the EVB3’s is the big brother, in
the case of electric piano sounds, it’s the EVP88 and so on with Clavinet (EVD6) or other
sounds (EXS24).

573

The interface of GarageBand Instruments consists of a simple silver panel that contains
a number of parameter sliders and associated value fields. As an example, here is the
Digital Stepper instrument:

Many of these parameters are macro parameters, which address specific, useful
parameters in the EXS24, ES1 (or other equivalent Logic instrument) instance
simultanously.
This has two main benefits:
• as the GarageBand plug-in are smaller, they load faster than the equivalent software
instrument and use less processing power
• limitation to a few, but powerful parameters makes use of the instruments very easy.
Play around with the parameters to see how easy it is to get spectacular sounds!
The Macro parameter sliders available to each GarageBand Instrument are different.
This is because the Logic Instrument’s parameters they address may be different or
because there’s no need to include an organ’s Drawbars parameter on a GarageBand
Piano Instrument, for example—at least not unless you’ve been getting creative with
your Steinway in the garden shed!

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Chapter 30 GarageBand Instruments

External Instrument

The External Instrument plug-in provides a simplified way
of handling an external MIDI sound source if you’re
feeding its audio output directly into your audio
interface.
This facility allows you to use one Arrange track for both MIDI recording and audio
mixing with effect plug-ins.
The External Instrument plug-in can be inserted in Audio Instrument channels (Mono/
Stereo > Logic > External) in place of a software instrument.
The plug-in features a MIDI Destination pull-down menu, that allows you to choose one
of the MIDI Instrument Objects in your Environment. The Input field allows you to select
one (or a pair of ) audio inputs (connected to the audio output(s) of the external sound
source). Input Volume determines the incoming signal level.
Obviously, this facility is not as useful if the audio interface you are using with Logic is
not equipped with multiple physical inputs.
Additional things to bear in mind:
You might still have the corresponding Input Objects (Input 1, for example) in your
mixer. Any effects inserted into this Input channel are placed before the effects inserted
in the External Instrument channel (in the signal chain).
If using multitimbral MIDI sound sources, please be aware that each External
Instrument requires a separate audio output.
Freezing an External Instrument track cannot happen faster than realtime, as per any
Bounce operations where MIDI hardware is involved.

575

Glossayr

Glossary

AAF Abbreviation for Advanced Authoring Format. This file format, typically used for
data exchange with Digidesign ProTools software, can be imported and exported by
Logic. It allows multiple audio tracks to be imported, with reference to tracks and
Region position, volume automation included.
AD converter or ADC Short for analog/digital converter; a device that converts an
analog signal to a digital signal.
aftertouch MIDI data-type generated by pressure on keys after they have been struck.
There are two types: Channel aftertouch, the value of which is measured by a full
length keyboard sensor. It affects all played notes. Polyphonic aftertouch (rare) is
individually measured and transmitted for each key. Aftertouch is also known as
pressure.
AIFF Abbreviation for Audio Interchange File Format. A cross-platform file format
supported by a large number of digital video and audio editing applications. AIFF
audio can use a variety of bit depths, but the two most commonly used are 16 bit and
24 bit.
AKAI Common sample format that the EXS24 is compatible with.
alias A pointer to a MIDI Region in the Arrange window. An alias does not contain any
data. It simply points to the data of the original MIDI Region. You can create an alias by
Shift-Option-dragging the original MIDI Region to a new location. An alias can not be
edited directly. Any change to the original Region will be reflected in the alias.
aliasing A digital artefact that occurs when the sample material contains frequencies
higher than one-half of the sample rate.
allpass filter A filter that allows all frequencies to pass, providing only phase shift or
phase delay without appreciably changing the amplitude characteristic.
amplifier Device which controls the level of a signal.
amplitude This term is used to describe the amount of a signal. If you have an audio
signal, amplitude refers to the sound’s volume, measured in decibels (dB).

577

analog signal A description of data that consists of a constantly varying voltage level,
that represents audio information. Analog signals must be digitized, or captured, for
use in Logic. Compare with digital.
Arrange window The heart of Logic. The primary working window of the program
where Audio and MIDI Regions are edited and moved to create a song arrangement.
attack Start phase of a sonic event. Also part of an envelope (see envelope).
attenuate To lower an audio signal’s level.
Audio Configuration window Logic window that provides an overview of all audio
routing. Allows the copying of the entire audio configuration between Logic songs, and
assists in renaming tasks. You can open the Audio Configuration window by choosing
Audio > Audio Configuration.
audio file Any digital recording of sound, stored on your hard drive. You can store
audio files in the AIFF, WAV and Sound Designer II formats. All recorded and bounced
WAV files are in Broadcast Wave format.
Audio Instrument Logic supports the use of software based instruments. Software
instrument plug-ins are inserted into Audio Instrument Objects. Software instrument
recording takes place on Audio Instrument tracks in the Arrange window. Playback of
these tracks is routed via the Audio Instrument Object.
audio interface Device needed to get sound into and out of your computer. An audio
interface converts digital audio from your computer to analog waves that speakers can
broadcast, or, in the other direction, an audio interface converts analog waves into
digital audio your computer can work with.
Audio Mixer An Environment layer that shows all Audio Objects of a song. The Audio
Mixer is used for mixing multiple audio channels in real time. The Audio Mixer is also
known as Environment Mixer.
Audio Object Audio Objects are found in Logic’s Environment. They are the “building
blocks“ of the Audio Mixer. When expanded, Audio Objects look just like channel strips.
Audio Objects tell Logic where to send audio signals. The following Audio Object types
are available: Audio Track Object, Input Object, Audio Instrument Object, Bus Object,
Output Object, Master Object, Aux Object.
Audio Region Chosen area of an audio file which is registered in the Audio window for
use in the song and, can be placed on audio tracks in the Arrange window, just like a
MIDI Region can be placed on MIDI tracks. Audio Regions are aliases (or pointers) to
portions of audio files. They can be as short as a single sample, or as long as the audio
file itself. You can use all of Logic’s tools to edit Audio Regions. Editing is non
destructive on the original audio file, as the Region is only an alias of the audio file. See
also Region and MIDI Region.

578

Glossary

audio track A track in Logic’s Arrange window that is used for playback, recording and
editing of Audio Regions.
Audio Track Object Audio Object in the Environment’s Audio layer. Used to playback
audio tracks in Logic’s Arrange window. All data on the audio track is routed to the
Audio Object, that was assigned in the Arrange window’s Track List menu.
Audio Units (AU) Audio Units is the standard format for real-time plug-ins running on
Mac OS X. It can be used for audio effects and software instruments. The Audio Unit
format is part of the Mac OS X operating system. Once installed, Audio Unit plug-ins
can be accessed by all programs simultaneously. Logic supports all Audio Unit format
plug-ins.
Audio window Logic window used for a number of audio file handling and conversion
tasks.
Autoload Song Song with your favorite settings and preferences. It loads automatically
when you launch Logic, and serves as a starting point for your songs and projects.
automation Automation is the ability to record, edit, and play back the movements of
all knobs, controls and buttons, including volume faders and pan, EQ, and Aux send
controls plus almost all effect and instrument plug-in parameters.
Aux Object Audio Object in the Environment’s Audio layer. Aux Objects are similar to
the Bus Objects, but are more flexible. Unlike Bus Objects Aux Objects also have Sends
of their own, allowing you to form complex signal paths
bandpass filter This filter only allows the frequency band centered around the cutoff
frequency to pass, while frequencies that lie further away (the lows and highs) are
filtered out. A sound that contains lots of mid range frequencies is the result. Also see
filter.
band rejection filter This filter cuts the frequency band centered around the cutoff
frequency, while allowing the frequencies that lie further away to pass. The mid range
frequencies will become softer and the lows and highs remain unchanged.
bar In musical notation, a bar is a measure that contains a specified number of beats,
and establishes the rhythmic structure of a musical piece.
Bar Ruler Ruler found at the top of the Arrange, Matrix, Hyper and Score windows. It
displays musical time units including bars, measures, beats and beat divisions. It is used
to set and display the song position, the cycle and autodrop locators, as well as
markers.
beat A musical time interval: “the beat is the regular rhythmic pulse in a composition
that people tap their feet to”. Usually a quarter note.

Glossary

579

Beat Mapping track Component of the Global tracks that helps to make a rhythmically
meaningful display of recordings that do not correspond to a strict tempo throughout.
It does this by redefining the bar positions of existing musical events, without
changing their absolute time position, thereby preserving the audible result with its
original timing.
beats per minute See bpm.
bit depth The number of bits a digital recording or digital device uses. The number of
bits in each sample determines the theoretical maximum dynamic range of the audio
data, regardless of sample rate. Also known as bit resolution, word length or bit rate.
bit rate See bit depth
bit resolution See bit depth
blue noise Highpass-filtered white noise, sounds like tape hiss.
boosting The act of raising an audio level.
bounce To process recorded or streamed MIDI and/or Audio Regions with any applied
effects, such as delay or compression, combining them into one audio file. In Logic, you
can choose between Realtime and Offline bouncing. Offline bouncing is faster, but
doesn’t allow you to apply live automation or record real time audio input.
Bounce button You can bounce the output of any Output Object to an audio file by
clicking the Object’s Bounce button. See also bounce.
bpm Abbreviation for beats per minute, a measure of the tempo of musical piece. As
an example: 120 bpm means that in one minute, there will be 120 musical beats
(quarter notes).
bus The term bus is used to describe a send/return routing scheme for audio channels.
In Logic, effects can be sent to/from Bus Objects for processing or submixing tasks.
Bus Object Audio Object in the Environment’s Audio layer. Usually used to route the
signal of an individual send bus to Output Objects. See also bus.
bypass To deactivate a plug-in. Bypassed plug-ins do not drain system resources. In
Logic you can bypass a plug-in by either clicking its Bypass button in the plug-In
window or by Option-clicking on the appropriate plug-in slot.
cable In Logic the term cable is used to describe the virtual cables that represents the
MIDI connection between two Environment Objects.
carrier In FM synthesis, the carrier is the equivalent of an analog synthesizer oscillator
that is producing a sine wave. The carrier frequency is modulated by the modulator.

580

Glossary

Catch button The button in the Transport bar featuring the running man icon. Activate
this button (blue) to turn on automatic horizontal scrolling during playback. This
ensures that the current playback position is always visible.
Catch function A window function that makes the currently displayed song section
reflect the current song position. Also see Catch button.
CD Audio Short for Compact Disc—Audio; current standard for stereo music
CDs: 44.1 kHz sampling rate and 16 bit depth.
cent A tuning subdivision of a semitone. There are one hundred cents in a semitone.
Many of Logic’s software instruments contain a Fine parameter that allows sounds to
be tuned in cent steps.
channel strip A channel strip is a virtual representation of a channel strip on a mixing
console. Each channel strip contains a number of similar controls, such as a Mute
button, Volume fader, Pan/Balance knob, Output selector and Bus and/or Insert slots.
Channel Strip setting Logic allows the routing of a channel strip, including all inserted
effects or instruments (plus their settings) to be saved and recalled. This simplifies the
task of recreating complex serial effect routings between channels or songs.
checkbox A small box. You click a checkbox to select or deselect an option.
chorus effect Effect achieved by layering two identical sounds with a delay and
slightly modulating the delay time of one or both of the sounds. This makes the audio
signal routed through the effect sound thicker and richer, giving the illusion of multiple
voices.
click Metronome, or metronome sound.
Clipboard The Clipboard is an invisible area of memory, into which you cut or copy
selected objects, using the Edit menu. From there, you can paste these objects to
different positions. Logic’s Clipboard spans all songs, allowing it to be used to exchange
objects between songs.
clipping (in digital recording) Feeding too much signal through a channel strip,
thereby exceeding the limit of what can be accurately reproduced results in a distorted
sound known as clipping. Logic’s Audio Objects feature a clip detector, which indicates
signal level peaks above 0 dB.
comb filter effect A short delay of feedback that emphasizes specific harmonics in a
signal is generally termed a comb filter. The name is derived from the appearance of a
frequency spectrum graphic, which resembles the teeth of a comb.
compressor An effect that restricts the dynamic range of an audio signal.

Glossary

581

controller MIDI data type. As examples; sliders, pedals or standard parameters like
volume and panning. The type of command is encoded in the first data byte, the value
in the second data byte.
Controls view All Logic plug-ins (and Audio Units) offer a non-graphical alternative to
the Editor views of effect and instrument parameters. The Controls view is accessed via
the Controls pull-down menu at the top of each plug-in window. This view is provided
to allow access to additional parameters and to use less onscreen space.
Core Audio Standardized audio driver system for all Macintosh computers running
version 10.2 or higher. Core Audio is an integral part of Mac OS X, allowing access to all
audio interfaces that are Core Audio compatible. Logic is compatible with any audio
hardware that offers Core Audio drivers.
Core MIDI Standardized MIDI driver system for all Macintosh computers running
Mac OS X version 10.2 or higher. Core MIDI is an integral part of Mac OS X, allowing the
connection of all MIDI devices that are Core MIDI compatible.
cutoff frequency Frequency at which the audio signal passing through a low or
highpass filter is attenuated by 3 dB.
Cycle function A function in Logic which constantly repeats the area between the
Locator positions. To turn on Cycle mode, click the Cycle button in the Transport
window. The cycle function is useful for composing a part of a song or editing events,
as examples. The Cycle area is shown as a green stripe in the top part of the Bar Ruler.
DA converter or DAC Short for digital/analog converter; a device that changes an
analog signal into a digital signal.
DAW Acronym for Digital Audio Workstation. A computer used for recording, mixing
and producing audio files.
dB Abbreviation for decibels, a unit of measurement that describes the relationships of
voltage levels, intensity or power, particularly in audio systems.
decay An envelope parameter that determines the time it takes for a signal to fall from
the maximum attack level to the sustain level. See envelope.
Deesser A signal processor that removes hissing or sibilance in audio signals.
default The preset parameter value.
delay In the Environment, an Object that can create a series of repeats. In the Arrange
window, a Region parameter which can delay or advance a selected Region by a given
number of milliseconds. Delay is also an effect process that delays the incoming audio
signal, resulting in subtle chorusing effects through to endless repeats of the signal.

582

Glossary

destructive Destructive audio processing means that the actual data of an audio file is
changed, as opposed to just editing peripheral or playback parameters.
dialog A window containing a query or message. It must be cancelled or replied to
before it will disappear and allow you to continue.
digital A description of data that is stored or transmitted as a sequence of ones and
zeros. Most commonly, refers to binary data represented using electronic or
electromagnetic signals. All files used in Logic are digital. Also see analog for
comparison.
disclosure triangle A small triangle you click to show or hide details in the user
interface.
distortion The effect produced when the limit of what can be accurately reproduced
in a digital signal is surpassed, resulting in a sharp, crackling sound.
distributed audio processing See Logic Node application.
drag & drop Grabbing objects with the mouse, moving them, and releasing the mouse
button.
driver Drivers are software programs that enable various pieces of hardware and
software to be recognized by other programs in a computer, and also to have the
appropriate data routed to them in a format they can understand. In Logic, you can use
the Preferences > Audio > Drivers panel to select and configure your audio hardware
drivers. If you do not have the proper driver installed, your computer may not
recognize or work properly with a given piece of hardware.
DSP (digital signal processing) In Logic, the mathematical processing of digital
information to modify a signal. An example is the Insert slot of channel strips, which
assigns DSP effects such as dynamic compression and delay to a channel’s signal.
DTDM Mixer Logic Pro supports a number of Digidesign hardware devices via Direct
TDM. The DTDM mixer is created in Logic’s Environment window, and allows the use of
Logic “native” effects and instruments with suitable Digidesign hardware.
dynamics Refers to changes in volume or other aspects of a piece of music over time.
dynamic range The dynamic range of a sound system is the difference in level
between the highest signal peak that can be reproduced by the system (or device in
the system) and the amplitude of the highest spectral component of the noise floor.
The dynamic range is the difference between the loudest and softest signals that the
system can reproduce. It is measured in decibels (dB). See decibels.
editor Window for editing MIDI or audio data. Logic offers the Hyper, Matrix and Score
editors for MIDI event data, and the Sample Editor for audio data.

Glossary

583

Editor view Almost all Logic plug-ins (and Audio Units) offer a graphical view of effect
and instrument parameters. The Editor view is used by default, but can be accessed via
the Editor pull-down menu at the top of each plug-in window, should the Controls
view be visible.
effect A type of software algorithm that lets you alter the sound of a track in a variety
of ways. Logic includes a set of EQ, dynamics, time-based, modulation and distortion
effects in Logic’s native and Audio Unit plug-in formats.
envelope The envelope is the variation that a sound exhibits over time, an envelope
basically determines how a sound starts, continues and disappears. Synthesizer
envelopes usually consist of Attack, Decay, Sustain and Release phases.
Environment The Environment is Logic’s brain: it graphically reflects the relationships
between hardware devices outside your computer and virtual devices within your
computer. Beyond basic input and output handling, the Environment can be used to
process MIDI data in real-time, and can even be used to create processing “machines”,
such as virtual rhythm generators and step sequencers or complex synthesizer editors.
Environment layer A place in the Environment, used to organize Objects and making
usage easier. Objects of the same type (Audio Objects, for example) are generally
placed on the same layer.
Environment Mixer See Audio Mixer
EQ Shortened form of equalizer. Equalizers are used to boost or cut frequencies in an
audio signal. There are several types available in Logic.
equalization See EQ
Eraser A tool used for deleting items. Click a selected item to delete it. All other
currently selected items are also deleted.
ESB TDM The ESB TDM connects your TDM hardware with Logic’s audio engine. This
allows your computer’s CPU to perform processes in Logic’s native mixer, including
audio track playback, the use of software-based instruments and effect plug-ins.
event Individual MIDI command, such as a note on command. Continuous controller
movements (modulation wheel, for example) produce a quick succession of individual
events with absolute values.
export To create a version of a file, such as a Logic song, in a different format that can
be distributed and used by other applications.

584

Glossary

filter effect Filters are effects you can apply to Audio or MIDI Regions (when streamed
or recorded as audio). They are designed to reduce a signal’s energy at a specific
frequency. A true filter always acts as a subtractive device, and doesn’t add anything to
the signal. The names of the individual filters illustrate their function. As an example: A
Low Pass filter allows frequencies that are lower than the cutoff frequency to pass.
Filter button Buttons in the Event List/Track Mixer, that allow you to hide/show
specific event types/channel strip types.
filter slope The filter slope is the steepness, or severity, of filter attenuation. As
examples, a filter slope of 6 dB per octave would sound much softer than a filter slope
of 12 dB per octave.
flanger The flanger effect is similar to the chorus effect, where a slightly delayed signal
(which is shorter than that of the Chorus) is fed back into the delay line input. Flanging
makes a sound thicker, and slightly “out of phase”.
float window Window with special status which always “floats” on the surface above
all other windows, but can only be operated with the mouse. Any Logic window can be
opened as a float window by holding down Option while opening it.
frame Unit of time. A second in the SMPTE standard is divided into frames that
correspond to a single still image in a file or video.
Freeze function The Freeze function performs individual offline bounce processes for
each “frozen” track, saving almost 100% of the CPU power used for software
instruments and effect plug-ins. All plug-ins of a track (including software instrument
plug-ins, if applicable, along with all related automation data) are rendered into a
“Freeze file”. You can use the Freeze function on individual Audio or Audio Instrument
tracks.
frequency The number of times a sound signal vibrates each second, measured in
cycles per second, or Hertz (Hz).
grab (an object) Positioning the mouse cursor over an object, then pressing and
holding the mouse button down.
help tag A small text window that appears when the mouse cursor is placed over an
interface element that indicates the name or value. When editing operations such as
moving or cutting a Region are performed, a larger help tag will display the current
position of the Region or function—in realtime.
Hermode Tuning A microtonal tuning system that can be used on all Logic and Audio
Unit software instruments. Hermode Tuning can make your software instruments
sound harmonically richer by fine tuning thirds, fifths and sevenths to specific
intervals—in cents.

Glossary

585

hierarchical menu Structured menus where choosing an individual entry opens a
submenu.
high cut filter A high cut filter is essentially a lowpass filter that offers no slope or
resonance controls.
highpass filter A highpass filter allows frequencies above the cutoff frequency to pass.
A highpass filter that offers no slope or resonance controls is generally knows as low
cut filter.
icon Small graphic symbol. In Logic, an icon may be assigned to each track.
importing The process of bringing files of various types into a Logic project or song.
Imported files can be created in another application, captured from another device, or
brought in from another Logic project.
Input Object Audio Object in the Environment’s Audio Layer. The Input Object
represents the physical inputs of your audio interface and helps managing audio from
your audio interface into Logic.
Insert slot A point on Logic’s Mixers where you can patch in (insert) an effect plug-in.
All audio channel types in Logic’s Track and Audio Mixers (except the Master Object)
offer effect insert slots.
interface 1) A hardware component such as a MIDI or audio device that allows Logic
to “interface” (connect) with the outside world. You need an audio or MIDI interface to
get sound/MIDI into and out of your computer. Also see audio interface. 2) A term that
is used to describe Logic’s graphical elements that can be interacted with. An example
would be the Arrange window, where graphical interface elements such as Regions are
interacted with to create an arrangement, within the overall Arrange interface.
key The scale used in a piece of music, centered around a specific pitch. The specified
pitch is called the root of the key.
key command Function which can be executed by pressing a specific key (or key
combination) on your computer keyboard or MIDI controller.
latency You may notice a delay between playing your keyboard and hearing the
sound. This is a form of latency. A variety of factors contribute to latency including
audio interface, audio and MIDI drivers. One factor under you control, however, is the
I/O buffer size, which is set in the Audio > Audio Hardware & Drivers preferences.
legato Method of musical performance that smoothly connects one note to the next.
level meter A meter that lets you monitor audio output levels from your computer.
You use the level meters in Logic when recording, arranging and editing audio files.

586

Glossary

Link mode Link mode is activated by clicking the Link button. It determines the
relationships between windows. An editing window in Link mode shows the same
contents as the top window.
Link button Button featuring the chain link icon in the top left corner of most Logic
windows. It controls the linking between different windows.
local menu Menu in a window that only contains functions that are relevant to that
particular window.
Local Off mode Operating mode on a MIDI keyboard where the keyboard does not
directly play its own integrated sound generator. This is useful when using it as a
master keyboard in a MIDI setup with Logic.
Locators Lower two sets of numbers, displayed to the right of the Transport buttons in
the Transport window. The number on top is the left Locator; the number on bottom
the right Locator. The Left and Right Locators control the time-range which Logic’s
playback will cycle or skip during playback. The Locators also can be used to define the
editing area for certain functions.
Logic Setup Assistant A utility that guides you through the process of setting up your
Macintosh, audio and MIDI hardware to work with Logic. The Logic Setup Assistant can
be run at any time by choosing the Preferences > Start Logic Setup Assistant menu
item.
loop An audio clip that contains recurring rhythmic musical elements or elements
suitable for repetition. Logic also supports Apple Loops.
Loop function Loop is a Region parameter in Logic that creates “loop repetitions” for
an Audio or MIDI Region. These repetitions will repeat until the song end point, or until
another Region or folder (whichever comes first) is encountered on the same track in
the Arrange window.
LFO Abbreviation for Low Frequency Oscillator. An oscillator that delivers modulation
signals below the audio frequency range—in the bandwidth that falls between 0.1 and
20 Hz, and sometimes as high as 50 Hz or 400 Hz.
lowcut filter A low cut filter is essentially a highpass filter that offers no slope or
resonance controls.
lowpass filter The lowpass filter defines the maximum frequency that can pass
through without being affected, thus controlling the brightness of the sound. Every
signal above this frequency will be cut. The higher the cutoff frequency, the higher the
frequencies that can pass through. A lowpass filter that offers no slope or resonance
controls is a high cut filter.

Glossary

587

marker Markers serve three purposes in Logic: They mark time-positions in the
Arrange window. They hold text notes and they delimit song settings. Markers can be
placed in the Marker track, or they can be placed in the Bar Ruler. Markers are generally
used for indicating and navigating to different song sections.
main menu bar The bar at the top of the computer screen, offering global functions
such as opening, saving, exporting or importing songs. It does not offer access to local
functions.
merge Mix, or combine, two or more MIDI events or Regions into a single Region.
metronome A part of Logic that produces a sound that taps out the beat. It can set by
click-holding the Metronome button in the Transport bar.
MIDI Abbreviation for Musical Instrument Digital Interface. Standardized,
asynchronous, serial and event-oriented interface for electronic musical instruments.
MIDI is an industry standard that allows devices such as synthesizers and computers to
communicate with each other. It controls a musical note’s pitch, length, and volume,
among other characteristics.
MIDI channel A MIDI channel is a “tube” for MIDI data, which flows through MIDI ports
in channels. Up to 16 separate MIDI channels can pass through a port simultaneously.
MIDI message A message transmitted via MIDI consisting of one status byte and none,
one, two or many data bytes (with system exclusive commands). See event.
MIDI Multi mode Multi-timbral operating mode on a MIDI sound module where
different sounds can be controlled polyphonically on different MIDI channels. A Multi
mode sound module behaves like several polyphonic sound modules. General MIDI
describes a 16-part multi mode (the ability to control 16 different parts individually).
Most modern sound generators support multi mode. In Logic, multi mode sound
modules are addressed via Multi Instrument Objects.
MIDI Region Data container for MIDI events which is shown in the Arrange window as
a named horizontal beam. In earlier Logic versions MIDI Regions were called sequences.
mixing The process of shaping the overall sound of a song by adjusting the volume
levels, pan positions, adding EQ and other effects, and using automation to
dynamically alter aspects of the song.
modifier key Computer keyboard keys used in conjunction with alphabetical keys to
change functionality. Modifier keys include; Control, Shift, Option and Command.
modulation Generally, a slight, continuously varying change. Logic’s effects and
synthesizers contain a number of modulators.
modulation amount The strength, or intensity, of modulation.

588

Glossary

modulation matrix The EXS 24 and other Logic instruments contain a grid that allows
you to modulate a number of target parameters with a number of modulators. This
grid is referred to as the modulation matrix.
modulation path A modulation path determines which target parameter will be
modulated by a specific modulator (modulation source).
modulation wheel A MIDI controller found on most MIDI keyboards.
mono Short for monophonic sound reproduction. The process of mixing audio
channels into a single track, using equal amounts of the left and right audio channel
signals. Compare with stereo.
MP3 Abbreviation for MPEG-2 Audio Layer 3. A compressed audio file format,
frequently used to distribute audio files over the Internet.
MS stereo recordings Short for middle-side stereo. Two microphones are stacked on a
stand or suspended from the ceiling so that they are positioned as closely together as
possible. One microphone delivers the middle signal, the other the side signal. If you
want to use MS stereo recordings in Logic, you have to decode them—this can for
example be done with the Direction Mixer plug-in. Also see XY stereo recording.
MTC See MIDI Time Code.
Multi Instrument Object An Object in Logic’s Environment that represents a multi
timbral hardware or software device that reacts to MIDI. The Multi Instrument Object is
essentially 16 Instrument Objects rolled into a single package. Each of these, called subchannels (or part-instrument), has a fixed MIDI channel and shares the same port. All
other parameters can be set individually. The purpose of Multi Instrument Object is to
address multi-channel MIDI devices, which receive MIDI data and play different sounds
on separate MIDI channels.
multitimbral This term describes an instrument or other device that can play different
sounds at the same time, using several MIDI channels at the same time.
Multi Trigger mode This term is associated with synthesizers such as the ES 1. In this
mode, a synthesizer envelope usually is retriggered by every note played.
mute Switch off an Audio Object or track’s audio output. You can mute a track by
clicking the Track Mute button in the Track List. The output of an Audio Object is
disabled by clicking the Mute button at the bottom of the channel strip.
nodes Positions in Hyper Draw and automation tracks that mark the positions where
data manipulation begins or ends. Occasionally referred to as points.

Glossary

589

normalize This function applies the current Parameter box settings to the selected
MIDI events (by altering the actual events themselves), and clears the Parameter
settings. When it comes to audio, a different “Normalize” function raises the volume of a
recorded audio file to the maximum digital level without altering the dynamic content.
notch filter This filter type cuts the frequency band directly surrounding the cutoff
frequency and allows all other frequencies to pass.
note number Pitch of a MIDI note, controlled by the first data byte of a MIDI note
event.
Object If capitalized, the term Object is used to refer to the graphical representation of
all elements in Logic’s Environment. These elements can be used to create and process
MIDI data in real-time, and can even be used to create processing “machines“, such as
virtual rhythm generators or step sequencers. Examples for Objects are Instruments,
Multi Instruments, Faders, Arpeggiators and others. In the Environment’s Audio layer
you also find Objects used to process audio data. These Objects are correspondingly
named Audio Objects. Also see Audio Objects.
Object Parameter box The Object Parameter box displays the properties of any
selected Environment Object. In the Arrange window this Parameter box is located
below the Toolbox and displays the properties of the selected Track’s Object.
OMF Abbreviation for Open Media Framework, also known as OMFI—Open Media
Framework Interchange. This file format, typically used for data exchange with
Digidesign ProTools software, can be imported and exported by Logic. The OMF file
format only supports the exchange of audio data (audio media and the usage of this
audio media in a song). MIDI and automation data will simply be ignored when using
Logic’s export function. option 1) Alternative function, often in the form of a checkbox,
sometimes also available as a menu entry. 2) Modifier key, in Windows terminology this
key is also known as Alt key.
oscillator A synthesizer oscillator generates an alternating current, using a selection of
waveforms which contain different amounts of harmonics.
Output Object Audio Object in Logic’s Environment controlling the output level and
pan/balance for each output on your audio interface. They are assigned to a specific
hardware output in their Object Parameter box.
pan, pan position The placement of mono audio signals in the stereo field, by setting
different levels on both sides.
Parameter box Field on the left side of Logic’s windows used to adjust the parameters
of the selected Regions or Objects.

590

Glossary

peak 1) The highest level in an audio signal 2) portions of a digital audio signal that
exceed 0 dB, resulting in clipping. You can use Logic’s level meter facilities to locate
peaks and remove or avoid clipping. The Search Peak command in the Sample Editor’s
Functions menu searches for the sample bit with the greatest amplitude value in the
currently selected Audio Region.
pink noise A harmonic noise type that contains more energy in the lower frequency
range.
pitch The perceived highness or lowness of a musical sound. Corresponds to the
frequency of the sound wave.
pitch bend message MIDI message transmitted by a keyboard’s pitch bend wheel.
playback Playing an Audio or MIDI Region or an entire arrangement, allowing you to
hear it.
plug-in Software application that enhances the functionality of the main program (in
this case, Logic). Logic’s plug-ins are typically software instruments or effects.
Plug-in window A window that launches when a plug-in is inserted, or the Insert/
Instrument slot is double-clicked. Allows you to interact with the plug-in parameters.
post fader Sends in analog mixers are positioned either before (pre) or after (post) the
fader. Post fader means positioned after the fader in the signal flow, with the level of a
signal going to the Send changing along with the fader movements.
pre fader Sends in analog mixers are positioned either before (pre) or after (post) the
fader. Pre fader means positioned before the fader in the signal flow, so the level of a
signal routed pre fader to a Send remains constant, regardless of any fader movements.
Preferences window A window that is accessed via the Logic > Preferences menu. All
Logic preferences can be set in this window.
preset Set of plug-in parameter values that can be loaded, saved, copied or pasted via
the Settings menu in the Plug-in window header. See setting and Settings menu.
pressure See aftertouch
project In Logic, the top-level folder that holds all media associated with a song,
including audio files, Sampler Instruments and samples, Video and Settings of various
kinds.
Project Manager A window that allows you to manage all media and file types that
Logic can read/use. You can access the Project Manager by choosing Windows >
Project Manager.

Glossary

591

PWM Pulse Width Modulation. Synthesizers often feature this facility, where a square
waveform is deformed by adjusting it’s pulse width. A square waveform usually sounds
hollow, and woody, whereas a pulse width modulated square wave sounds more reedy
and nasal.
Q factor A term generally associated with equalizers. The Q factor is the “quality” factor
of the equalization, and is used to select a narrower or broader frequency range within
the overall sonic spectrum of the incoming signal.
quantization Time-correction of note positions by moving them to the nearest point
on a selectable grid. When quantization is applied to any selected event or Region,
Logic will move all note events non-destructively to align perfectly with the nearest
grid position.
RAM Abbreviation for random-access memory. A computer’s memory capacity,
measured in megabytes (MB), which determines the amount of data the computer can
process and temporarily store at any given moment.
real-time effects Effects that can be applied to Regions in real time, without requiring
rendering before playback. Real-time effects can be played back with any Macintosh
computer qualified to run Logic.
ReCycle ReCycle is the name of an application from software manufacturer
Propellerheads, which mainly serves as an editing and production tool for loops
(repeatedly looped audio samples). ReCycle uses specific file formats (.REX) which can
be imported by Logic.
Region Regions can be found in the tracks of the Arrange window: They are
rectangular beams that act as containers for audio or MIDI data. There are three
different types of Regions: Audio Regions, MIDI Regions and Folder Regions. Also
see: Audio Region, MIDI Region and Folder.
Region Parameter box Box in the upper left corner of the Arrange window, used to
non-destructively set the playback parameters for individual Regions, including;
quantization, transposition, velocity, compression and delay. These parameters do not
alter the stored data. Rather, they affect how the events are played back.
resonance A term generally associated with filters, particularly those of synthesizers.
Resonance emphasizes the frequency range surrounding the cutoff frequency. See
cutoff frequency.

592

Glossary

reverb Reverb(eration) is the sound of a space. More specifically, the reflections of
soundwaves within a space. As an example, a handclap in a cathedral will reverberate
for a long time as sound waves bounce off the stone surfaces within a very large space.
A handclap in a broom closet will hardly reverberate at all. This is because the time it
takes for the soundwaves to reach the walls and bounce back to your ears is very short,
so the “reverb”’ effect will probably not even be heard.
RMS Root Mean Square. A measurement of the effective audio signal average. Used in
Logic’s analysis tools.
root note The central note of a musical scale or key, which all other notes are related
to.
routing Generally refers to the way audio is sent through processing units. Also often
used to describe specific input and output assignments.
sample A digital recording of a sound at a particular instant in time.
Sample Editor Logic’s Sample Editor allows stereo or mono audio files to be
destructively cut, reversed, shortened, changed in gain and processed in a number of
other ways. It allows editing of individual samples within an audio file consisting of
thousands or millions of samples. The Sample Editor also provides access to a number
of special sample processing tools, collectively known as the Digital Factory.
sampler Device used for sampling. In Logic, this generally refers to the EXS24 softwarebased sampler.
sample rate When an analog audio signal is converted to a digital signal, this term
refers to the number of times per second the audio file is sampled. Logic can record
and edit audio at sample rates ranging from 44.1 kHz (44,100 times per second) up to
192 kHz (192,000 times per second).
sampling The process of converting analog audio into digital information. The sample
rate of an audio stream specifies the number of samples that are captured per second
(see sample rate). Higher sample rates yield higher quality audio.
saturation A term most commonly associated with a slight tape distortion or the
characteristics of tube amplifiers. It basically describes a very high gain level that causes
a slight distortion of the incoming signal, resulting in a warm, rounded sound.
scroll bar and scroll box Gray beam at the edge of a window. A movable box inside
the beam is used to select the displayed song section in the window.
self-oscillation Self-oscillation is a typical characteristic of analog filter circuits. It
occurs when the filter, at high resonance values, feeds back into itself and begins to
oscillate at its natural frequency.

Glossary

593

semitone Smallest interval between two pitches in the standard diatonic scale, equal
to a half tone. Correspondingly a semitone is also called half step or half tone.
send Abbreviation for auxiliary sends. An output on an audio device used for routing a
controlled amount of the signal to another device. Sends are for example often used to
send several signals to the same effect, which is rather advisable for computationallyintensive effects such as reverb.
sequencer A sequencer is a computer application that allows you to record both
digital audio and MIDI data and blend the sounds together in a software mixing
console. There are editing tools that let you control every aspect of the production,
down to the finest details. Effect processors and software instruments are increasingly
being added to these applications. Modern sequencers such as Logic now can fulfill
many, if not all, functions that were only possible in the most expensive and wellequipped recording studios a decade ago.
setting 1) A parameter value. 2) A set of plug-in parameter values that can be loaded,
saved, copied or pasted via the Settings menu. A plug-in setting is also known as
preset. Also see preset and Settings menu.
Settings menu Accessible via the Arrow button found in the gray header at the top of
all plug-in windows. Allows you to save, load, copy and paste settings: the parameter
values of effects and software instruments.
shelving EQ EQ type that allows reducing or increasing the frequency range above or
below the specified frequency.
side chain A side chain is effectively an alternate input signal—usually routed into an
effect—that is used to control an effect parameter. As an example, you could use a side
chained track containing a drum loop to act as the control signal for a Gate inserted on
a sustained pad track, creating a rhythmic gating effect of the pad sound.
Single Trigger mode This term is associated with synthesizers such as the ES 1. In this
mode, envelopes are not retriggered when tied (legato) notes are played.
Snap menu A pull-down menu found at the top of all linear editing windows in Logic.
Selection of items in the menu will cause events or Regions to “snap” to the specified
value when moved.
software instrument Logic’s software counterpart to a real sound source or a sampler
or synthesizer module.
solo A way to temporarily highlight one or more tracks or Regions or events, allowing
them to be heard in isolation.
Solo tool Click-holding on individual Regions or events with the Solo tool temporarily
allows them to be heard in isolation. All other objects are muted.

594

Glossary

song Main Logic file, containing all MIDI events and parameter settings (including
mixer automation data) plus information about the audio files to be played.
Song Settings The Song Settings, accessible from the File menu, are a collection of
program settings that are specific to the current song. These are different to the global
preferences that affect all Logic songs (see preference).
stereo Short for stereophonic sound reproduction of two different audio channels.
Compare with mono.
Sustain pedal A momentary footswitch that is connected to MIDI keyboards. It
transmits MIDI controller number 64, which is recorded and played back by Logic.
synthesizer A device (hardware or software) that is used to generate sounds. The word
is derived from early attempts with mechanical and electronic machines to emulate (or
synthesize) the sounds of musical instruments, voices, birdsong and so on. Logic
features several software synthesizers, including; the ES1, ES2, EFM 1, ES E, ES P and
ES M.
tempo The playback speed of a piece of music, measured in beats per minute. Logic
allows you to create and edit tempo changes in the Tempo track.
time signature Two numerals separated by a diagonal bar that appear at the
beginning of a song. Common time signatures are 4/4 and 2/4. The first number
denotes the number of notes in a measure, or bar. The second number denotes a unit
of time for each beat. In a 2/4 signature, each bar has two beats; each beat is a quarter
note long.
timing Measure of the ability to play notes at the right time. Timing can also refer to
synchronization between events, Regions and devices.
toggle To switch between two states such as on or off (applies to windows, parameter
values and so on).
Touch Tracks An Environment Object that allows MIDI Regions to be assigned to, and
triggered by, individual MIDI note events. This enables you to assign a number of
musical phrases to different MIDI keyboard keys, and trigger (or record) them in realtime, making the process of arrangement faster and more intuitive.
track A horizontal row in the Arrange window that contains either Audio or MIDI
Regions that can be played back over time. Each track has a specified destination that
data is routed to. Logic allows hundreds of tracks to be used in a song.
Track List Situated to the left of the Arrange window’s working area. Displays the
Objects assigned to various tracks as well as the Track buttons.

Glossary

595

Track Mixer Adaptive Mixer which automatically configures itself to show every audio
and MIDI track, in the order that they appear in the Arrange window or in an open
Folder. If you move the controls on the Track Mixer while recording, automation data is
stored in the relevant tracks as MIDI controller information.
transient Position in an audio recording where the signal becomes a lot louder—over
a short time span (a signal “spike”, in other words). As this is typical for drum recordings,
transients can be used to indicate where beats occur in an audio signal.
Transport window Window used to control recording and playback functions. The
Transport window offers Record, Pause, Play, Stop and Rewind/Forward buttons plus
other functions. You can also configure a fixed Transport window in the Arrange and
Matrix windows by selecting View > Transport. This Transport window variation is
named Transport field. The term Transport bar refers to both the Transport window and
Transport field.
transpositon Transposition is changing the pitch of a Audio or MIDI Region or event
by a number of semitones.
Undo function Function which reverses the previous editing operation.
velocity Force at which a MIDI note is struck; controlled by the second data byte of a
note event.
virtual memory Area of the hard disk used as an extension of RAM memory by the
computer. The disadvantage is its very slow access time, in comparison to physical
RAM.
WAV, WAVE The primary audio file format used by Windows-compatible computers. In
Logic, all recorded and bounced WAV files are in Broadcast Wave format, which include
a high-resolution timestamp.
waveform A visual representation of an audio signal.
wet/dry mix Refers to the ratio of a signal that effects have been added to (wet), and
the original, unprocessed signal (dry).
white noise Noise type that consists of all frequencies (an infinite number) sounding
simultaneously, at the same intensity, in a given frequency band. Its name is analogous
to white light, which consists of a mixture of all optical wavelengths (all rainbow
colors). Sonically, white noise falls between the sound of the consonant F and breaking
waves (surf ). Synthesis of wind and seashore noises, or electronic snare drum sounds,
requires the use of white noise.
window class Status of the window as a float window or a normal window. Float
windows always “float” in the foreground and can not be hidden by normal windows.
Also see float window.

596

Glossary

word length See bit depth.
XY stereo recordings Two cardioid microphones aligned so that they are directed to
the left and right of the sound source. Also see MS stereo recordings.
zero crossing A point in an audio file where the waveform crosses the zero amplitude
axis. If you cut an audio file at a zero crossing there will be no click at the cut point.
zoom An action that enlarges (zooms in on) or shrinks (zooms out from) the display in
a Logic window. The Magnifying Glass in the Toolbox, and the Zoom controls found in
the lower left and upper right corners of windows, are both used for zooming tasks.
Also see Zoom control and zoom level.
Zoom control The control that appears at the bottom left and top right of some
windows, such as the Arrange. The Zoom control slider allows you to navigate through
the entire length of the currently displayed song. The lines on the left and right of the
slider can be clicked to zoom in and out by a fixed percentage.
zoom level The amount that a window’s contents (tracks, Regions and Objects, for
example) are magnified. Zooming in to a high level allows you to make more precise
edits. Conversely, you can zoom all the way out to see the entire song and work on
very large sections.

Glossary

597

A
AAF 577
Adaptive Limiter 50
Gain 51
Input Scale 51
margin display 51
Out Ceiling 51
ADC 577
AD converter 577
additive synthesis 232, 481
aftertouch 577
channel 577
polyphonic 577
AIFF 577
AKAI 577
alias 577
aliasing 577
allpass filter 577
amplifier 577
amplitude 577
analog 578
analog synthesizer 195
Arrange window 578
attack 578
attenuating 578
AU
Audio Configuration window 578
Audio Instrument 578
Audio Instrument Object 13, 23, 24
Audio Mixer 578
Audio Object 13, 578
Audio Track Object 579
Aux Object 579
Bus Object 580
Input Object 586
Audio Track Object 579
Audio Unit. See AU
Audio window 579
AutoFilter 67
Attack 68
Coarse 69
Cutoff Freq. 67
Decay 68

Index

Delay 69
Distortion Input 68
Distortion Output 68
Dynamic Modulation 68
Envelope 68
Fatness 68
Fine 69
Frequency 69
LFO 69
Modulation 68
Pulsewidth 69
Release 68
Resonance 68
Slope 68
Speed Mod. 69
Stereo Phase 69
Sustain 68
Sync 69
Threshold 68
Volume 69
Autoload Song 579
automation 20, 579
Aux Object 579
aux send 23
AVerb 109
Density/Time 109
Pre Delay 109
Reflectivity 109
Room Size 109

B
bandpass filter 250, 579
band rejection 250
band rejection filter 579
bar 579
Bar Ruler 579
beat 579
Beat Mapping track 580
beat per minute. Seebpm
Bitcrusher 63
Clip Level 63
Downsampling 63
Drive 63

599

Mode 63
Resolution 63
bit depth 580
bit rate. See bit depth
bit resolution. See bit depth
blue noise 152, 580
boosting 580
bounce 580
Bounce button 580
Bounce function 9, 26
bpm 580
bus 23, 580
Bus Object 580
bypass 15, 17, 580

C
Carlos, Wendy 192
carrier 580
Catch function 581
Catch button 581
cent 581
Channel EQ 29
Analyzer 30
Analyzer Decay 31
Analyzer Mode 31
Analyzer Top 31
dB Warp 31
Fast Fourier Transformation 30
using as Default EQ 31
channel strip 581
Channel Strip setting 581
Channel Strip Settings 20
chattering effect 43
checkbox 581
Chorus 98, 280, 462
chorus 581
click 581
Clipboard 581
Clip Distortion 64
clip circuit graphic 64
Drive 64
Filter 64
Frequency 64
Gain 64
Input Gain 64
Mix 64
Sum Filter 64
Symmetry 64
Tone 64
colored noise 240
comb filter effect 99, 581
component modelling 356, 357
Compressor 39
Attack 40
Auto Gain 40

600

Index

Gain Reduction Meter 41
Knee slider 41
Output Clip 41
Peak 41
Ratio 40
Release 40
RMS 41
Threshold 40
controller 581, 582
Controls view 17, 582
conversion module
mono-stereo 15
stereo-mono 15
convolution 138
Convolution Reverb. See Space Designer
Core Audio 582
Core MIDI 582
Correlation Meter 166
cutoff frequency 197
Cycle 582

D
DAC 582
DA converter 582
DAW 582
dB
decibel. See dB
DeEsser 47
Detector Frequency 48
Detector Sensitivity 48
Monitor 48
Smoothing 48
Strength 48
Suppressor Frequency 48
Denoiser 152
blue noise 152
graphic 153
Noise Type 152
pink noise 152
Reduce 152
Smoothing 152
Frequency 153
Time 153
Transition 153
Threshold 152
white noise 152
destructive 583
dialog 583
digital 583
digital audio workstation. See DAW
digital signal processing. See DSP
digital synthesizer 195
Direction Mixer 161
Basis 161
Direction 161

Input 161
MS stereo 162
Distortion 62
Drive 62
Output 62
Tone 62
Distortion II 66
PreGain 66
DJ EQ 38
driver 583
DSP
Ducking 18
Dudley ,Homer 191
dynamic range 583

E
editor 583
Editor view 17, 584
effect 23
bus 23
insert 23
mono 15
plug-in 13, 23
stereo 15
EFM 1
aftertouch 206
Carrier 204
Fine 204
Fixed Carrier option 204
FM 202
FM Depth 203
Glide 202
Harmonic 204
LFO 203
Main Level 206
Modulation Env 203
modulation wheel 206
Modulator 204
Modulator Pitch 203
Modulator Wave 205
pitch bend 206
Randomize 202, 206
Rate 203
sideband 201
Stereo Detune 205
Sub Osc Level 205
Transpose 202
Tune 202
Unison 202
Velocity 205
Voices 202
Vol Envelope 205
EMF 1 201
Ensemble 105
Effect Volume 105

Index

graphic 105
Intensity 105
LFO 1 105
LFO 2 105
Mix 105
Phase 105
Random 105
Rate 105
Stereo Base 105
Voices 105
envelope 199, 584
attack 199
decay 199
release 199
sustain 199
Enveloper 67
Attack 45
Lookahead 46
Out Level 46
Release 45
Threshold 46
Time 46
EnVerb 115
Attack 116
Crossover 116
Decay 116
Density 116
High Cut 116
Hold 116
Low Level 116
Original Delay 115
Predelay 116
Release 116
Spread 116
Sustain 116
Environment 584
layer 584
Environment Mixer 584
equalizer 29
Eraser tool 584
ES1 215
2', 4', 8', 16', 32' (octave transposition) 216
ADSR 218, 220
ADSR via Vel 217
AGateR 218
Analog 220
Bender Range 220
Chorus 221
Cutoff 217
Drive 216
Filter 217
Filter FM 219
GateR 218
Glide 218
Int via Vel 220
Int via Whl 219

601

Key 217
Level Via Vel 218
LFO Amp 219
LFO Waveform 218
Mix 216
Mod Envelope 220
Out Level 220
Rate 219
Resonance 217
router 219
Sub 216
Tune 220
Voices 221
Wave 216
ES2 223
Amp 252
amplifier 252
Analog 226
Bend Range 227
Blend. See Filter Blend
bypassing modulation 253
CBD 226
chain symbol 247
Chorus 280
Curve 277
Cut (Cutoff Frequency) 246
delayed vibrato 266
Digiwave 233
Distortion 280
Drive 245
dynamic stage 252
effect processor 280
ENV 1 269
Attack 270
Attack via Vel 270
Decay 270
Mono 269
Poly 269
Release 270
Retrig 269
ENV 2 270
Attack 270
Decay 271
Release 271
Sustain 271
Sustain Time 271
Vel 271
ENV 3 252, 270
Attack 270
Decay 271
Release 271
Sustain 271
Sustain Time 271
Vel 271
envelope 268
Env Mode 276

602

Index

Fat 247
filter 242
Filter 2 FM 250
Filter Blend 243
filter mode 248
BP 248
BR 248
Hi 248
Lo 248
Peak 248
filter slope 247
Fix Timing 279
Flanger 280
Flt Reset 229
FM 234
frequency switch 230
Glide 227
Intensity 280
intermodulation effect 245
Legato 227
LFO 265
Rate 267
Wave 267
LFO 1 266
EG 266
LFO 2 266
Rate 268
Loop Count 279
Loop Mode 277
Loop Rate 278
Loop Smooth 278
modulation source 260
Bender 261
ENV1 261
ENV2 261
ENV3 261
Kybd 261
LFO1 260
LFO2 260
Max 261
MIDI Controllers A–F 262
ModWhl 261
Pad-X 261
Pad-Y 261
RndNO1 263
RndNO2 263
SideCh 263
Touch 262
Velo 261
Whl+To 262
modulation target 254
Amp 259
Cut 1+2 258
Cut1inv2 258
Cutoff 1 257
Cutoff 2 258

Detune 255
FltBlend 258
Lfo1Asym 259
Lfo1Curve 259
LPF FM 258
Osc1Levl 257
Osc1Wave 256
Osc1WaveB 257
Osc2Levl 257
Osc2Wave 256
Osc2WaveB 257
Osc3Levl 257
Osc3Wave 256
Osc3WaveB 257
OscLScle 257
OscWaveB 256
OscWaves 255
Pan 259
Pitch 1 255
Pitch 123 254
Pitch 2 255
Pitch 3 255
Reso 1 258
Reso 2 258
SineLevl 257
Mono 227
multi trigger 228
muting oscillators 230
noise 240
Oscillators 226
OscLevelX 241
OscLevelY 241
Osc Start 229
Parallel 242
Phaser 280
Poly 227
polyphonic distortion 245
processing power, handling economically 251
PWM 236
Random function 281
Res (Resonance) 246
Ring 238
RND 281
RND Destination 281
RND Int 281
Router 253
scaled modulation target 259
Env2Atck 259
Env2Dec 260
Env2Rel 260
Env2Time 260
Env3Atck 260
Env3Dec 260
Env3Rel 260
Env3Time 260
Glide 260

Index

LFO1Rate 259
Series 242
Sine Level 252
single trigger 228
Solo Point 276
Speed 280
Square 272
Sync 237
Target. See modulation target 254
template 294
Clean Stratocaster 294
Crescendo Brass 297
MW-Pad-Creator 3 298
Slap Strat 294
Wheelrocker 296
Wheelsyncer 299
Time Scaling 279
Tone 280
Triangle (Oscillator Mix Field) 240
Tune 226
tutorial 283
Analog Bass clean 285
Analog Bass distorted 285
Analog Saw 3Osc 284
Analog Saw Init 283
Analog Unison 284
FM DigiWave 287
FM Drive 287
FM Envelope 286
FM Megafat 288
FM Out of Tune 288
FM Start 286
FM Tuned 288
FM Wavetable 287
PWM 2 Osc 289
PWM Fast 289
PWM Scaled 289
PWM Slow 289
PWM Soft Strings 289
PWM Start 289
Ringmod Start 290
Sync Start 290
Vector Envelope 291
Vector Kick 293
Vector Loop 292
Vector Perc Synth 294
Vector Punch Bass 294
Vector Start 291
Vector XY 291
Unison 228
Vector Envelope 273
Curve 277
Default Setting 275
deleting point 275
editing point 275
envelope point 273

603

Env Mode 276
Fix Timing 279
Loop Count 279
Loop Mode 277
Loop Point 274
Loop Rate 278
Loop Smooth 278
resetting a point 275
setting point 275
Solo Point 276
Sustain Point 273
switching Vector Envelope off 275
Time Scaling 279
Vector Int 273
Vector Mode 272
Vector X Target 272
Vector Y Target 272
via 253
Via invert (inv) 254
via source 263
Bender 264
ENV1 263
ENV2 263
ENV3 263
Kybd 264
LFO1 263
LFO2 263
ModWhl 264
Pad-X 264
Pad-Y 264
RndNO1 265
RndNO2 265
SideCh 265
Touch 264
Velo 264
Whl+To 264
Voices 228
Wave 230
waveform Oscillator 1 231
waveform Oscillator 2 236
waveform Oscillator 3 236
wavetable synthesis 233
ESB TDM 584
ES E 213
4, 8, 16 (octave transposition) 213
AR Int 214
Attack 214
Chorus I 214
Chorus II 214
Cutoff 214
Ensemble 214
Release 214
Resonance 214
Speed 214
Velo Filter 214
Velo Volume 214

604

Index

Vib/PWM 213
Volume 214
Wave 213
ES M 207
8, 16, 32 (octave transpositon) 207
Cutoff 207
Decay (filter) 208
Decay (volume) 208
Glide 207
Int 208
Mix 207
Overdrive 208
Resonance 208
Velo (filter) 208
Velo (volume) 208
Vol 208
ES P 209
1/3, 2/3, 3/3 210
8, 12, 32 (octave transposition) 209
A (Attack time) 210
ADSR Int 210
Chorus 211
D (Decay time) 210
Frequency 210
Overdrive 211
R (Release time) 211
Resonance 210
S (Sustain level) 211
Speed 210
Velo Filter 210
Velo Volume 210
Vib 210
Volume 210
Wah 210
waveform fader 209
EVB3 455
Acc/Dec Scale 476
additive synthesis 481
Basic MIDI Ch 457
Bass Filter 466
Bite 473
Brake 475
Cabinet 474
Cancel Key 464
CC field 472
changing MIDI channels 457
Chorale 475
Chorus 462
Click Color 468
Click Max 468
Click Min 468
Click Off 468
Click On 468
Condition 467
Crosstalk 467
Distortion 473

Drawbar 460
Drawbar Leak 467
Drive 474
Effect Bypass 471
Effect Chain 471
EQ High 471
EQ Level 471
EQ Low 471
EQ Mid 471
Expression 461
Filter Age 468
Horn Deflector 476
Keyboard Mode 458
keyboard range of upper and lower manual 457
keyboard split 458
Leakage 467
Leslie 474, 484
Lower Stretch 469
Lower Volume 461
LP Split 458
Max Wheels 466
Mic Angle 476
Mic Distance 476
MIDI CC 465
MIDI controller assignment 477
MIDI Mode 458
MIDI Preset Switching 464
MIDI setup 456
MIDI to Presetkey 464
Mode (Morph) 465
Mode (Reverb) 472
Mode (Sustain) 470
Mode (Wah) 473
Morphing 465
Motor Control 476
Organ 465
pedal drawbar 461
Pedal Volume 461
Percussion 463
Pitch 469
Pitchbend Up/Down 470
playing both manuals and the pedals 456
preset key 464
Random FM 467
Range 465, 473
Rate 462
residual effect 482
Reverb 472
Rotor Cabinet 474
Rotor Fast Rate 476
Rotor Speed 475
Save To 465
Scanner Vibrato 462
Shape 466
Speed Control 475
Sustain 470

Index

Time 463
Tonal Balance 466
Tone 474
tonewheel sound generation 483
Trans LM 458
Trans Ped 458
Trans UM 458
Tremolo 475
Tune 462
Type 462
Type (Distortion) 473
UL Split 458
Ultra Bass 466
Up Level 464
Upper Stretch 469
Vel 463
Vibrato 462
Volume 461
Wah 472
controlling with MIDI-controller or
aftertouch 472
MIDI pedal control 472
Warmth 469
EVD6 485
AB 489
Bender 487
Brilliance 494
Brilliant 489
CD 489
Chorus 502
Intensity 502
Rate 502
Clavinet 503
Click 494
Comp 499
Damper 493
Damper Ctrl 493
Damping 496
Decay 495
Distortion 499
double-triggered notes 504
effect 498
Envelope 500
Excite 494
Filter 489
Flanger 502
Intensity 502
Rate 502
Freeze 182
FX Order 502
Gain (Distortion) 499
Inharmonicity 496
Intensity 494
KeyOff 495
KeyOn 495
Level 493

605

Medium 489
MIDI 502
Mode (Modulation) 501
Model 491
Basic 491
Class D6 491
Domin 492
Dulcimer 492
Guru Funk 492
Harpsi 492
Ltl India 492
Mello D6 491
Old D6 491
Picked 492
Pluck 492
Sharp D6 491
StrBells 492
Wood 492
Modulation 501
Phaser 501
Intensity 501
Rate 501
Pickup 489
Pickup Mode 498
Pickup Position 497
Pitch Fall 496
Pressure 488
Random 495
Range 500
Release 495
Shape 494
Soft 489
Stereo Spread 490
Stiffness 496
Stretch 488
String 495
Tension Mod 496
Tone 499
Treble 489
Tune 487
Velocity 495
Velo Curve 494
Voices 487
Wah 500
Wah Ctrl 501
Wah Pedal Position 501
Warmth 488
event 584
EVOC 20 FB 72
Bands 73
blue bar 74
Boost A/B 74
Fade AB 75
filter bank 168
Formant Filter window 73
Formant Shift 75

606

Index

High/Low Frequency 74
Highest 74
Intensity 76
Level 77
LFO Fade 75
LFO Shift 75
Lowest 74
MIDI controllers, received 78
Modulation 75
Overdrive 77
Rate 76
Resonance 75
Slope 74
Stereo Mode 77
Stereo Width 77
Waveform (LFO) 76
EVOC 20 PS 175
Ana 188
Analog 180
Attack (Sidechain Analysis In) 181
Attack (synthesizer) 181
Balance 179
Bands 182
Bend Range 180
block diagram 190
blue bar 183
Color 178
Cutoff 180
Detune 179
Dual 177, 179
Ensemble 188
FM 177, 179
FM Int 179
Formant Filter window 183
Formant Shift 184
Formant Stretch 184
Glide 180
High/Low Frequency 183
Highest 183
inserting in Logic setup 175
Intensity 185
Int via Whl 185
Legato 177
Level (output) 188
Level (U/V Detection) 187
LFO waveform 185
Lowest 183
MIDI controller 188
Mode 187
Blend 187
Noise 187
Noise + Synth 187
modulation parameters 185
Mono 177
multi trigger 177
Noise 178

Level 178
oscillators 177
Pitch LFO 185
Poly 177
portamento 180
Rate 186
Ratio c 179
Ratio f 179
Release (Sidechain Analysis In) 181
Release (synthesizer) 181
Resonance (Formant Filter) 184
Resonance (synthesizer filter) 180
Semi 179
Sensitivity 186
Shift LFO 185
Side Chain 175
Sidechain Analysis In 181
Signal 187
single trigger 177
Stereo Width 188
Syn 188
Tuning 180
U/V Detection 186
Unison 177
Unvoiced/Voiced detector 169
Voc 188
Voices 176
Wave 1 178
Wave 2 178
Waveform 178
EVOC 20 TO 79
Analysis In 81
Attack 80
Bands 82
block diagram 190
blue bar 85
Coarse Tune 82
Fine Tune 82
FM Int 83
FM Ratio 83
Formant Filter window 85
Formant Shift 86
Formant Stretch 86
Freeze 81
Glide 84
High/Low Frequency 85
Highest 86
Intensity 87
Level (Output) 90
Level (U/V Detection) 89
LFO 87
Lowest 86
Max Track 85
MIDI controller 91
Mode 89
Blend 89

Index

Noise 89
Noise + Synth 89
modulation parameters 87
output parameters 89
Pitch Quantize 83
pitch tracking oscillator 79
Rate 88
Release 80
Resonance 87
Root/Scale 84
Sensitivity 88
Signal 89
Stereo Mode 90
Stereo Width 90
Strength 83, 84
Synthesis In 81
tracking oscillator 82
Tune 82
Unvoiced/Voiced detector 88, 169
Wave 87
EVP88 505
Bass 510
Bell 508
Chorus Intensity 512
Color 511
Damper 508
Decay 507
Drive 510
emulated e-piano model
Hohner Electra piano 516
Rhodes 513
Wurlitzer piano 515
EQ 510
equal tempered scale 509
Gain (Drive) 511
Intensity (Tremolo) 512
Lower Stretch 509
MIDI 516
adapting velocity sensitivity 516
Model 506
Model parameters 507
Phaser 511
Rate (Phaser) 511
Rate (Tremolo) 512
Release 508
Stereo 508
Stereophase (Phaser) 511
Stereophase (Tremolo) 512
Tone 510
Treble 510
Tremolo 512
Tune 507
Upper Stretch 509
Voices 507
Warmth 509
Exciter 153

607

Color 1 154
Color 2 154
Frequency 154
graphic 154
Harmonics 154
Input 154
Expander 42
Auto Gain 42
Ratio 42
Threshold 42
EXS24 mkII 519
AKAI
Convert entire CD 536
Convert function 535, 555
Convert window 537
file organization 536
Partition 535
Prelisten 539
Program 536
Volume 536
Amount 543
Amp (Env 2) 548
b/p 551
BP 547
chain symbol (filter) 547
Clear Find 531
compatibility EXS24 mkI 552
Cutoff 547
Dest 550
Drive 547
Edit button 540
Enable Find 531
EXS24 mkI Modulation Path 552
Fat 546
file organization 526
audio file 526
Sampler Instrument 526
Setting 527
Filter On/Off button 546
Find function 530
Fine 544
Glide 545
Group 524, 525
creating 525
Group parameter 566
ADSR Offset 526
Cutoff Offset 526
Pan/Bal 566
Resonance Offset 526
Select Range 525
Velocity Range 567
Voices 525, 566
Vol 566
Volume 525
history of sampling 570
Hold via 542

608

Index

HP 546
Instrument Editor 521
Edit menu 557
Group menu 562
Instrument menu 554
View menu 563
Zone menu 561
inv 551
Key 547
key command 568
Key Scale 548
Legato 539
Level via Vel 548
LFO 1 EG 549
LFO 1 Rate 549
LFO 2 Rate 550
LFO 3 Rate 550
LP 546
MIDI controller 571
Modulation Matrix 550
b/p 551
bypassing modulation path 551
compatibility EXS24 mkI 552
creating modulation path 550
Dest 550
EXS24 mkI Modulation Path 552
inv 551
Second Order modulation 552
Src 550
via 551
Mono 539
Multiple Outputs 553
Aux Channel 554
Options button 541
Pitch Bend 544
Pitcher 545
Poly 539
Preferences 558
Random 544
ReCycle 534
ReCycle Convert 555
Velocity 534
Remote 545
Resonance 547
SampleCell 533
sample file import 531
AKAI 535
EXS24 mkII 531
ReCycle 534
SampleCell 533
SoundFont 2 532
using EXS 24 Instrument from CD-ROM 531
Vienna Library 539
Sampler Instrument 520, 526
closing Sampler Instrument 554
copying audio files of Sampler Instrument 555

creating 522
creating Sampler Instrument 554
deleting Sampler Instrument 555
loading Sampler Instrument 520, 540
managing Sampler Instruments 527
moving audio files of Sampler Instrument 555
opening Sampler Instrument 554
renaming Sampler Instrument 555
saving Sampler Instrument 554
saving Sampler Instrument song-related 528
searching Sampler Instrument 530
selecting Sampler Instrument 540
Sample Select 543
Second Order modulation 552
Setting 527
SoundFont 2 532
Src 550
Transpose 544
Tune 544
Type 543
Unison 540
via 551
Vienna Library 539
Voices 540
Volume 548
Wave (LFO 1, LFO 2) 549
Xfade 542
Zone 521
creating 522
Zone parameter 564
Audio File 522, 564
Cent 564
Disable Pitch 564
E button 524
End Frame 523, 565, 566
Group 564
Key Note 523, 564
Loop 523, 565
One Shot 565
Pan 523, 565
Reverse 523, 565
Scale 523, 565
Start Frame 523, 565, 566
Tune 523, 564
Velocity Range 564
Volume 523, 565
Zone name 564
Zone Range 523, 564
external effect processor, inserting 160
External Instrument 575
Input Volume 575
MIDI Destination 575

F
Fat EQ 37

Index

filter 197, 585
allpass 577
bandpass 579
band rejection 579
cutoff frequency 197
highpass 586
resonance 198
filter bank 168
Filter button 585
filter slope 320, 585
flanger 99, 280
float window 585
FM synthesis 234
formant 146
Fourier theorem 198
Freeze function 26, 585
Fuzz-Wah 70
Auto Gain 70
AutoWah Attack 72
AutoWah Depth 71
AutoWah Release 72
Comp Ratio 72
Fuzz Gain 72
Fuzz Tone 72
FX Order 70
Pedal Position 71
Pedal Range 71
relative Q 71
Wah Level 70
Wah Mode 70

G
Gain 159
Gain 159
Mono 160
Phase Invert 160
Stereo Balance 160
Swap Left/Right 160
GarageBand Instrument 573
GoldVerb 111
Balance ER/Reverb 111
Density 113
Diffusion 113
High Cut 113
Initial Delay 112
Predelay 112
Reverbtime 113
Room Shape 112
Room Size 112
Spread 113
Stereo Base 112
Guitar Amp Pro 57
Amp menu 58
Bass control 59
Centered button 60

609

Conderser button 60
Dynamic button 60
EQ menu 59
FX section 61
Gain 59
Link button 59
Master 60
Mid control 59
Off-Center button 60
Output 61
Presence parameter 59
Reverb section 61
Speaker menu 58
Treble control 59

H
Hammond organ 482
Hermode Tuning 27, 585
highpass filter 249, 586
High Shelving EQ 38, 92
Hohner Electra piano 516

I
I/O 160
Input 160
Input Volume 160
Output 160
Output Volume 160
impulse response 117
Input Object 586
insert 23
instrument plug-in 13, 23, 24
interface 586
intermodulation effect 245

K
KlopfGeist
Detune 454
Level Via Vel 454
metronome click 453
Semitone 454
Tonality 454

L
latency 586
legato 586
Leslie 484
Levelmeter 166
LFO 265
Limiter 49
Gain 49
graphic display 50
Lookahead 49
Output Level 50
Release 50

610

Index

Softknee 50
LinearPhase EQ 32
Link button 16
Link function 587
Link button 587
Local Off mode 587
lowpass filter 249
Low Shelving EQ 38, 92

M
Match EQ 33
Analyzer Position menu 34
Apply slider 36
Channel Link slider 34
Learn button 34
manual modifications 36
Match button 35
Phase menu 36
Select buttons 34
Smoothing slider 36
View menu 34
metronome click. See KlopfGeist
Meyer-Eppler, Werner 192
modifier key 588
modulation 200
Modulation Delay 97
Anti Pitch 97
Constant Mod. 97
Feedback 98
Flanger-Chorus 98
LFO 97
LFO Mix 97
Mix 98
Stereo Phase 98
Vol. Mod. 97
Width 97
mono
effect 15
mono Object 15
Moog, Bob 192
MP3 589
MS stereo 162
Multipressor 52
Attack 53
Bands 53
Comp. Ratio 54
Compression Threshold 54
crossover frequencies 53
downward expansion 52
Exp. Ratio 54
Expansion Threshold 54
Gain Change 55
graphic curve 54
level meter 55
Lookahead 53

Master Gain 55
multi-band graphic 53
Output 55
Peak/RMS 53
Reduction 54
Release 53
multitimbral 589
multi trigger 228, 361
Multi Trigger mode 589
mute 589

N
node 589
No HMT option 27
Noise Gate 43
Attack 43
chattering effect 43
Hold 43
Hysteresis 43
Lookahead 44
Monitor 44
Reduction 43
Release 43
Side Chain 44
Threshold 43

O
Object
mono 15
stereo 15
Object Parameter box 590
No HMT option 27
oscillator 590
Output Object 590
Overdrive 62
Drive 62
Output 62
Tone 62

P
Parallel Bandpass Vocoder 191
Parameter box 590
Parametric EQ 38
peak 591
peak type filter 249
Phase Distortion 65
Cutoff 65
Intensity 65
Max Modulation 65
Mix 65
Monitor 65
Phase Reverse 65
Resonance 65
Phaser 99
Color 100

Index

comb filter effect 99
Envelope Modulation 100
LFO 1 100
LFO 2 100
LFO Mix 100
Order 99
Stereo Phase 100
Sweep Ceiling 100
Sweep Floor 100
phaser 281
pink noise 152, 591
pitch 591
Pitch Correction 147
Byp button 149
Range 148
Response 149
Root 148
Scale 148
Pitch Shifter II 144
Cents 144
Drums 144
Mix 144
Semi Tones 144
Speech 144
Vocals 144
PlatinumVerb 114
Crossover 114
Low Level 114
Low Ratio 114
playback 591
plug-in
Adaptive Limiter 50
adding 13
adjusting parameters 16
Audio Units format 21
AutoFilter 67
automation 20
AVerb 109
Bitcrusher 63
button 16
bypassing 15, 17
Channel EQ 29
Chorus 98
Clip Distortion 64
comparing settings 16
Compressor 39
Correlation Meter 166
DeEsser 47
deleting 15
Denoiser 152
Direction Mixer 161
Distortion 62
Distortion II 66
DJ EQ 38
effect 13, 23
EFM 1 201

611

Ensemble 105
Enveloper 67
EnVerb 115
ES1 215
ES2 223
ES E 213
ES M 207
ES P 209
EVD6 485
EVOC 20 FB 72
EVOC 20 PS 175
EVOC 20 TO 79
EVP88 505
Exciter 153
Expander 42
EXS24 mkII 519
External Instrument 575
Fat EQ 37
fine-tunig parameters 16
Gain 159
GarageBand Instrument 573
GoldVerb 111
Guitar Amp Pro 57
High Shelving EQ 38, 92
I/O 160
instrument 13, 23, 24
Levelmeter 166
Limiter 49
loading multiple plug-ins 20
Low Shelving EQ 38, 92
Match EQ 33
Modulation Delay 97
Multipressor 52
Noise Gate 43
numerical panel 16
operation 16
Overdrive 62
Parametric EQ 38
Phase Distortion 65
Phaser 99
Pitch Correction 147
Pitch Shifter II 144
PlatinumVerb 114
resetting parameters 16
RingShifter 101
rotary knob 16
Rotor Cabinet 106
Sample Delay 93
Scanner Vibrato 106
Sculpture 355
Setting 19
setting
loading default automatically 19
Settings menu 17, 19
Silver Compressor 42
Silver EQ 38

612

Index

Silver Gate 45
SilverVerb 110
slider 16
Space Designer 117
Spectral Gate 143
Spreader 107
Stereo Delay 96
Stereo Spread 154
SubBass 150
Tape Delay 94
TDM 21
Test Oscillator 157
Tremolo 104
Tuner 158
Ultrabeat 301
up/down arrow 16
Vocal Transformer 145
plug-in menu 13
Plug-in window 14, 16, 591
additional parameters 18
common parameters 16
Controls view 17
Editor view 17
Link button 16
opening as non-floating window 20
open on insertion 14
Side Chain menu 18
switching contents 17
view modes 17
001/011 button 18
PMW 592
post fader 591
pre fader 591
preset 591
pressure 591

Q
quantization 592

R
RAM 592
Record function 9
rectangular wave 233
reflectogram 117
Region Parameter box 592
residual effect 482
resonance 198, 592
reverberation 139
Rhodes 513
RingShifter 101
Delay section 103
Dual 102
Env Follower 103
Frequency 102
LFO 104

OSC 102
Output section 103
Side Chain 102
Single 102
RMS 593
root note 593
Rotor Cabinet 106
routing 593

S
Sample & Hold 267
Sample Delay 93
sampler (history) 570
sample rate 593
sampling 593
saturation 593
sawtooth wave 232
Scanner Vibrato 106
Rate Left 106
Rate Right 106
Stereo Phase 106
Sculpture 355
1, 2 and 3 button 368
Always mode 371
amplitude envelope 374
Attack 374
Bandpass 377
Bender Range Down 362
Linked 362
Bender Range Up 362
Body EQ 381
aktivieren 381
component modelling 356
String 357
Control Envelope 392
Controller A/B modulations 391
Crossfeed 379
Cutoff 377
Decay 375
Delay Base Time 380
Envelope 392
copying 400
Ctrl/Env button 392
handling 394
modulation routing 392
parameters 397
recording 395
window 393
Feedback 379
filter 377
activating 377
setting mode 377
Fine Structure 383
Formant Intensity 382
Formant Shift 383

Index

Formant Stretch 383
Gate 371
Glide Time 360
Groove 381
HiCut 379
Hide 362, 363
High (Body EQ) 382
Hipass 377
Inner Loss 363
Inner Loss Scale
Low/High 363
Release 364
Input Balance 380
Input Scale 376
Invert button 373
Jitter 1/2 390
Key (filter) 378
Keyboard Mode 361
KeyOff 371
KeyOn 371
Keyscale 362
Legato mode 361
Level Limiter 384
LFO 385
Curve 386
Envelope 386
Phase 387
Rate 386
RateMod 387
Sync/Free button 386
Target 1/2 388
Waveform menu 385
LoCut 379
Lopass 377
Low (Body EQ) 382
Material Pad 363
Media Loss 365
Release 366
Media Loss Scale
Low/High 365
Mid (Body EQ) 382
Mid Frequency 382
MIDI controller assignment 408
Learn function 408
Model (Body EQ) 382
modulation generators 384
Mono mode 361
Morph function 400
Envelope window 404
Morph menu 403
randomizing 402
Multi Trigger 361
Notch 377
Object 367
activating 368
Type 368

613

velocity sensitivity 368
Peak 377
Pickup 373
setting position 373
Poly mode 361
Position 372
programming electric basses 426
Release 362, 363, 375
Resolution 365
Resolution Scale
Low/High 365
Resonance 378
signal path 356
Single Trigger 361
Spread 381
Stereo Base 380
stereo delay 379
activating 379
Stiffness 364
Stiffness Scale
Low/High 364
Strength 371
String 357
activating animation 372
Sustain 375
Sync (stereo delay) 380
Tension Mod 366
Tension Mod Scale
Low/High 366
Timbre 371
Transpose 360
Tune 360
Variation 371, 376
Velocity modulations 391
Velo Sens (filter) 378
Velo Sens (Object) 371
Vibrato 388
Curve 389
Depth via Vib Ctrl 389
Phase 389
Rate 389
Waveform 388
Voices 361
Warmth 360
Waveshaper 375
activating 375
Type 375
Wet Level 379
self-oscillation 593
sequencer 594
Setting 19
setting 594
loading default automatically 19
Settings menu 17, 19
sideband 201
Side Chain 18

614

Index

Ducking 18
Use Instrument as Side Chain 18
side chain 594
Silver Compressor 42
Silver EQ 38
Silver Gate 45
SilverVerb 110
Density/Time 110
graphic display 110
High Cut 110
LFO 111
Low Cut 110
Modulation Int 111
Modulation Phase 111
Predelay 110
Reflectivity 110
Room Size 110
sine sweep 157
sine wave 231
single trigger 228, 361
Single Trigger mode 594
software instrument
multiple outputs 26
Software Instrument Pitch parameter 27
Space Designer 117
automation 118
blue grace period bar 119
clickable key parameters 118
convolution 138, 140
Deconvolution facility 138
Density Envelope 133
End Level 134
Init Level 134
Ramp Time 134
Reflection Shape 134
digital spike 137
Direct Output 123
Envelope window 125
A and D buttons 127
All 126
Density Env 127
Envelope Mode buttons 127
Filter Env 127
node 126
Overview display 128
Reset 126
Reverse 127
Volume Env 127
Zoom to Fit 127
Filter 135
BP 135
Filter Mode 135
HP 135
LP (6 dB, 12 dB) 135
On 135
Reso 135

Filter Envelope 131
Attack Time 132
Break Level 132
Curve Form Node 133
Decay Time 132
End Level 133
Init Level 131
Impulse Response 137
creating 137
digital spike 137
loading 120
randomly generating 120
starter pistol 137
sweep 137
impulse response 117
Input (Crossfeed) 123
inserting Space Designer 118
IR Sample 120
IR Start 124
Latency Compensation 125
Length 122
Low Shelving EQ 136
Freq 136
Gain 136
operating parameters 118
Pre-Delay 124
Preserve Length 122
red progress bar 119
reflectogram 117
reverberation 139
Reverb Output 123
Rev Vol Compensation 124
Sample Rate 121
starter pistol 137
Stereo Spread 136
Spread 136
Xover 136
sweep 137
Synthesized IR 120
Volume Envelope 128
Attack Time 129
Curve Form node 130
Decay Time 129
End Level 130
Exp 130
Init Level 128
Lin 130
Volume Decay Mode 130
Spectral Gate 143
Bandwidth 143
Central Freq. 143
CF Mod. 144
Gain 144
High Level 143
Low Level 143
Sub Energy 143

Index

Super Energy 143
Threshold 143
Spreader 107
stereo
effect 15
Object 15
Stereo Delay 96
Crossfeed 96
Delay 96
Groove 96
High Cut 96
Left Feedback 96
Low Cut 96
Mix 96
Right Feedback 96
Stereo Spread 154
graphic 155
Lower Freq 155
Lower Int. 155
Order 155
Upper Freq. 155
SubBass 150
Bandwidth 151
Center High 151
Center Low 151
graphic 151
Mix 151
Ratio 151
subtractive synthesis 196
synthesis 196
additive 481
subtractive 196
synthesizer 595
analog 195
digital 195
virtual analog 195

T
Tape Delay 94
Delay 94
Feedback 94
Flutter Intensity 95
Flutter Rate 95
Freeze 94, 96
Groove 94
High Cut 95
LFO Depth 95
LFO Speed 95
Low Cut 95
Mix 95
Smooth 95
Sync 94
Tempo 94
tempo changes 94
TDM 21

615

tempo 595
Test Oscillator 157
Frequency 157
Level 157
Sine Sweep 157
Time 158
Trigger 158
Waveform 157
time signature 595
timing 595
toggle 595
Touch Track 595
Track List 595
Track Mixer 596
transient 596
transposition 596
Tremolo 104
graphic display 104
Rate 104
Smoothing 104
Stereophase 104
Symmetry 104
triangular wave 199, 232
Tuner 158

U
Ultrabeat 301
2 band EQ 321
editing graphically 322
808 snare 347
assignment section 302, 303
control element 308
fine-tuning 308
setting to default value 308
distortion unit 321
Clip parameter 321
Color parameter 321
Crush button 321
Distort button 321
Drive parameter 321
Level parameter 321
drum kit 302
drum mixer 305
individual outputs 305
Master Volume slider 305
Mute button 305
Pan knob 305
Solo button 305
Volume fader 305
Drum Sound
importing 340
drum sound
copying 304
naming 304
pasting 304

616

Index

selecting 303, 341
drum voice 301
distribution across MIDI keyboard 302, 303
envelope 330
Attack parameter 331
Decay parameter 331
editing graphically 330
modulation 331
one shot 330
selecting 331
Sustain button 332
volume 330
Zoom button 332
filter section 319
Cut knob 320
red arrow 319
Gated button 324
Group menu 324
kick drum 342
LFO 328
Cycle parameter 329
Ramp parameter 329
Rate parameter 329
turning on/off 329
waveform 329
Master Volume slider 305
MIDI control 303
MIDI Controller Assignment menu 328
Learn option 328
modulation 307, 325
Mod 307
setting routing 327
Via 307
multi-channel instrument 305
Mute button 305
noise generator 317
Cut parameter 318
Dirt parameter 318
filter 317
Filter Bypass button 318
Resonance parameter 318, 320
Oscillator 1 309
Asym parameter 310, 313
Filter Bypass button 311
FM Amount parameter 311
FM mode 311
Phase Oscillator 310
Pitch parameter 310, 312
Saturation parameter 310, 313
Slope parameter 310, 313
turning on/off 309
Volume parameter 309
Oscillator 2 312
Filter Bypass button 316
Inner Loss parameter 315
Min/Max (Velocity) parameter 314

Model button 315
Phase Oscillator 313
Resolution parameter 315
Reverse arrow 314
Sample mode 313
Stiffness parameter 315
Type button 315
Vel Layer parameter 314
Volume parameter 312
Oszillator 1
turning on/off 312
Output section 321
Pan knob 305
Pan Mod button 323
ring modulator 316
Filter Bypass button 317
Volume slider 316
setting 303
loading 303
saving 303
signal flow 302, 306
snare 346
Solo button 305
Spread button 323
step sequencer 308, 332
MIDI control 338
Pattern Mode button 338
Playback Mode menu 338
principle 333
starting/stopping 334
turning on/off 334
Voice Mute Mode button 339
structure 301
synthesizer 306
synthesizer section 309
Trigger menu 324
tutorial 341
Voice Volume control 323
Volume fader 305

V
velocity 596
virtual analog synthesizer 195
virtual memory 596
Vocal Transformer 145
formant 146
Mix 146
Pitch 146

617

Index

Pitch Base 147
Robotize 145
Tracking 146
Vocoder 167
analyzing speech signals 169
Carlos, Wendy 192
Dudley, Homer 191
filter bank 168
functionality 168
history 191
Meyer-Eppler, Werner 192
Moog, Bob 192
Parallel Bandpass Vocoder 191
sound
unvoiced 169
voiced 169
speech intelligibility 169
avoiding sonic artifacts 170
compressing the Side Chain 169
enhancing high frequency energy 169
gating background noise in Side Chain 171
Release parameter 170
suitable analysis/synthesis signals 172
Unvoiced/Voiced detector 169
Voder 191
Zinovieff, Peter 192
Voder 191
Voice Auto Select function 304

W
white noise 152, 240, 596
window
float 585
word length. See bit depth
Wurlitzer piano 515

X
XY stereo recording 597

Z
001/011 button 18
zero crossing 597
Zinovieff, Peter 192
zoom 597
control 597
level 597

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Linearized                      : Yes
Author                          : Apple Computer
Create Date                     : 2004:09:10 13:43:24Z
Keywords                        : kmanual, klogic
Modify Date                     : 2012:03:17 03:44:02+01:00
XMP Toolkit                     : Adobe XMP Core 4.2.1-c043 52.372728, 2009/01/18-15:56:37
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Metadata Date                   : 2012:03:17 03:44:02+01:00
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Format                          : application/pdf
Title                           : Logic Pro 7: Plug-In Reference (Manual)
Creator                         : Apple Computer
Subject                         : kmanual klogic
Page Mode                       : UseOutlines
Page Count                      : 618

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Apple Logic Pro 7 Plug In Reference Ref

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